I don't know of any sip carriers who will let you do refer. you will
need to find a carrier who supports it. FreeSWITCH will have no
problem sending it but I doubt you will find a carrier who will let
you do it easily.
Mike
On Aug 8, 2009, at 3:12 PM, Vladimir Rodionov wrote:
A call is coming on SIP trunk. From PSTN. I does not need to be
answered, actually - I need to do some logic before redirecting call
but I can answer call as well It won't break the app logic.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones
<[email protected]> wrote:
Are your calls coming in on TDM or SIP trunks? Are your calls answered
by FreeSWITCH before you need to redirect them?
On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
Rodionov<[email protected]> wrote:
> Actually, this is what I need
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>
> Will it work with PSTN? Can I redirect incoming PSTN call to
another PSTN
> number?
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <[email protected]
>
> wrote:
>>
>> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I
am pretty
>> sure it is doable because voxeo offers this
>> option for their Voice XML customers but I am not interested in a
hosted
>> solution at the time - it is quite expensive. As far as I
understood, Voip
>> provider MUST have pstn call transfer feature enabled by telecom
provider
>> (AT&T for example) and this should work fine with SIP.
>>
>> -Vladimir Rodionov
>>
>>
>> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <[email protected]
>
>> wrote:
>>>
>>> Hi there,
>>>
>>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>>> INVITE) and only pass back the media to the network, or pass back
>>> signaling also (SIP REFER)?
>>>
>>> I know several suppliers who support SIP re INVITE but none that
>>> support SIP REFER.
>>>
>>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>>
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<[email protected]
>
>>> wrote:
>>> > Good morning,
>>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID).
Calls goes
>>> > through PSTN Gateway (1) to freeSWITCH application server (AS)
(2).
>>> > AS does some logic and transfers call (or forward) out of Voip
provider
>>> > network to another PSTN number User2.
>>> >
>>> >
>>> > This is call bridge
>>> >
>>> >
>>> > UA1 (PSTN) - -> UA2
(PSTN)
>>> > - -
>>> > - (1) - (4)
>>> > -> PSTN Gateway->
>>> > - -
>>> > (2) - - (3)
>>> > -> FreeSWITCH ->
>>> >
>>> >
>>> > This is what I want to acomplish
>>> > (4)
>>> > UA1 (PSTN) ------------------------------- -> UA2
(PSTN)
>>> > -
>>> > - (1)
>>> > -> PSTN Gateway->
>>> > - -
>>> > (2) - - (3)
>>> > -> FreeSWITCH ->
>>> >
>>> >
>>> > 1. Can it be implemented in FreeSWITCH?
>>> > 2. Does anybody know Voip providers which support out of
network call
>>> > transfer/forwarding to PSTN?
>>> >
>>> > TIA
>>> >
>>> > -Vladimir Rodionov
>>> >
_______________________________________________
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
