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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Call Transfer



 
 
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  #1  
Old August 13th 05, 02:58 PM posted to uk.telecom.voip
sean
external usenet poster
 
Posts: 415
Default Call Transfer

Hi, when i call a PSTN line, then transfer the call to another PSTN
line.. How does the hangup/cutoff rules work?

I guess both have to be down a set amount of time to relese the line?
  #2  
Old August 13th 05, 03:40 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 209
Default Call Transfer


"Sean" wrote in message
...
Hi, when i call a PSTN line, then transfer the call to another PSTN line..
How does the hangup/cutoff rules work?


Not very well. You could end up with a long call charge. It's not too bad if
you remain in the conversation (3 way call) as you can verify or control the
disconnect. You could always mute yourself to give the appearance of having
transferred the call.


  #3  
Old August 13th 05, 04:01 PM posted to uk.telecom.voip
Ian
external usenet poster
 
Posts: 289
Default Call Transfer


"Jono" wrote in message
news

"Sean" wrote in message
...
Hi, when i call a PSTN line, then transfer the call to another PSTN

line..
How does the hangup/cutoff rules work?


Not very well. You could end up with a long call charge. It's not too bad

if
you remain in the conversation (3 way call) as you can verify or control

the
disconnect. You could always mute yourself to give the appearance of

having
transferred the call.

In theory
The clearing is controled by the calling party
So caller A calls caller B who transfers/diverts to C

A clears this will release call at B which should clear down the call to C

For ISDN this is controlled by the messaging and is reliable for POTS lines
its fairly reliable on correctly configured systems but can lead to calls
being help up and lines locking out.

Ian


  #4  
Old August 13th 05, 08:58 PM posted to uk.telecom.voip
sean
external usenet poster
 
Posts: 415
Default Call Transfer

Jono wrote:
"Sean" wrote in message
...

Hi, when i call a PSTN line, then transfer the call to another PSTN line..
How does the hangup/cutoff rules work?



Not very well. You could end up with a long call charge. It's not too bad if
you remain in the conversation (3 way call) as you can verify or control the
disconnect. You could always mute yourself to give the appearance of having
transferred the call.


Call cost is not a problem

Thanks for the advice tho
  #5  
Old August 13th 05, 09:00 PM posted to uk.telecom.voip
sean
external usenet poster
 
Posts: 415
Default Call Transfer


In theory
The clearing is controled by the calling party
So caller A calls caller B who transfers/diverts to C

A clears this will release call at B which should clear down the call to C

For ISDN this is controlled by the messaging and is reliable for POTS lines
its fairly reliable on correctly configured systems but can lead to calls
being help up and lines locking out.

Ian


That theory does not seem to work Using JD VoIP... Seems to keep the
call connected when hung up.
  #6  
Old August 14th 05, 10:07 AM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 73
Default Call Transfer


"Sean" wrote in message
...

In theory
The clearing is controled by the calling party
So caller A calls caller B who transfers/diverts to C

A clears this will release call at B which should clear down the call to
C

For ISDN this is controlled by the messaging and is reliable for POTS
lines
its fairly reliable on correctly configured systems but can lead to calls
being help up and lines locking out.

Ian


That theory does not seem to work Using JD VoIP... Seems to keep the
call connected when hung up.


You never mentioned VOIP in your original question. Obvoiusly I guessed
there was some involved, considering the group you posted in.

You make a call VOIP-----PSTN
put PSTN on hold
Ring another PSTN
Then connect the two PSTN parties.

Is that what your trying to do?


  #7  
Old August 14th 05, 01:15 PM posted to uk.telecom.voip
sean
external usenet poster
 
Posts: 415
Default Call Transfer

Jono wrote:
"Sean" wrote in message
...

In theory
The clearing is controled by the calling party
So caller A calls caller B who transfers/diverts to C

A clears this will release call at B which should clear down the call to
C

For ISDN this is controlled by the messaging and is reliable for POTS
lines
its fairly reliable on correctly configured systems but can lead to calls
being help up and lines locking out.

Ian


That theory does not seem to work Using JD VoIP... Seems to keep the
call connected when hung up.



You never mentioned VOIP in your original question. Obvoiusly I guessed
there was some involved, considering the group you posted in.

You make a call VOIP-----PSTN
put PSTN on hold
Ring another PSTN
Then connect the two PSTN parties.

Is that what your trying to do?


Yes, that's what i have done a few times now.
  #8  
Old August 14th 05, 01:32 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 209
Default Call Transfer



Sean wrote:
|| Jono wrote:
||| "Sean" wrote in message
||| ...
|||
||||| In theory
||||| The clearing is controled by the calling party
||||| So caller A calls caller B who transfers/diverts to C
|||||
||||| A clears this will release call at B which should clear down the
||||| call to C
|||||
||||| For ISDN this is controlled by the messaging and is reliable for
||||| POTS lines
||||| its fairly reliable on correctly configured systems but can lead
||||| to calls being help up and lines locking out.
|||||
||||| Ian
|||||
||||
|||| That theory does not seem to work Using JD VoIP... Seems to
|||| keep the call connected when hung up.
|||
|||
||| You never mentioned VOIP in your original question. Obvoiusly I
||| guessed there was some involved, considering the group you posted
||| in.
|||
||| You make a call VOIP-----PSTN
||| put PSTN on hold
||| Ring another PSTN
||| Then connect the two PSTN parties.
|||
||| Is that what your trying to do?
|||
|||
|| Yes, that's what i have done a few times now.

Even PBXs for PSTN are notoriously unreliable at detecting that the line has
cleared down, let alone a VOIP service provider. I'm surpised they even alow
it, especially if they're delivering the call(s) for [next to] nothing.



  #9  
Old August 14th 05, 02:54 PM posted to uk.telecom.voip
sean
external usenet poster
 
Posts: 415
Default Call Transfer

Jono wrote:

Even PBXs for PSTN are notoriously unreliable at detecting that the line has
cleared down, let alone a VOIP service provider. I'm surpised they even alow
it, especially if they're delivering the call(s) for [next to] nothing.




I'm suprised too, as i get geo calls in evening and weekend included. So
when i've finnished on phone to someone, i ask if they are wanting to
speak to anyone else, if so i put them through
  #10  
Old August 14th 05, 04:12 PM posted to uk.telecom.voip
Ian
external usenet poster
 
Posts: 289
Default Call Transfer


"Sean" wrote in message
...

In theory
The clearing is controled by the calling party
So caller A calls caller B who transfers/diverts to C

A clears this will release call at B which should clear down the call to

C

For ISDN this is controlled by the messaging and is reliable for POTS

lines
its fairly reliable on correctly configured systems but can lead to

calls
being help up and lines locking out.

Ian


That theory does not seem to work Using JD VoIP... Seems to keep the
call connected when hung up.


Ok You never menioned IP but anyway. I have reread the scenario again

what you are doing is

You are caller A who calls B then holds B and calls C, Then joins B& C

B & C are both *called* so do not control clear down so as you say the calls
stay up and so they should, Unless you use a PBX in the middle which may
sense the clear down or may not. This one of the reasons Trun to trunk
transfer was not allowed under BABT regs.

Ian



 




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