The first 3 point I meant was "we have VOIP.MS ITSP and DID number
provide by them was assigned to AA of our server and external caller
calls the DID number and connected to AA.Then external user transfer to 200"
I never tried by your configuration as "FreeSWITCH as SBC".But in my
configuration if I try to answer the call from 200 after dialing *78200
from 201 then the call will be disconnected from 200..
Can you please try with latest build and let us know whether its
reproducible?
Regards,
Kumaran T
On 10/18/2011 6:39 PM, Fulvio Scapin wrote:
> Hello Kumaran.
>
> I had forgotten to mention my SipXECS version in my previous mail. I
> fear it's still a 4.2. Haven't found the time to upgrade yet.
> Your scenario appears mostly correct, although I didn't quite get the
> first three points you enumerated.
>
> Just to add a few more details, I use FreeSWITCH as SBC in a B2BUA
> configuration, connected to a few ITSP SIP providers.
> So what actually happens is that I receive a SIP invite through my
> ITSP account, which FreeSWITCH «bridges» with my SipXECS default AA.
> When the AA answers the caller digits an internal extension number and
> the call is transferred to that extension, which would be the 200 of
> your example.
> If nobody answers at 200 and user 201 dials *78200 as you imagined,
> the call is instantiated solely between the external caller and
> extension 201, as one might expect.
>
> However, if the user at 200 picks up the call after the user at 201
> has dialed *78200, they BOTH answer the call, in somethin akin to a
> three-way conference.
>
> I do realize it's not intended to behave that way, but it might become
> a useful feature rather than a bug for quite a few people, hopefully.
>
> Regards,
> Fulvio
>
> 2011/10/18 Kumaran T<thiru.venkateshwa...@ezuce.com>:
>> Hi Fulvio,
>> Which Build you seeing this issue?
>> I tried in latest 4.5.2 build by following scenario and its working
>> fine.Please let me know my scenario was right?
>> 1.DID no assigned to AA
>> 2.Call DID number from PSTN number
>> 3.AA will answer and dial 200 from pstn number
>> 4.The call will transferred to 200 and 200 will start ringing
>> 5.From 201 user dialed *78200
>> 6. Call is retrieved and call will disconnected from 200
>> 7.Call will established between pstn user and 201..
>>
>> Regards,
>> Kumaran T
>>
>> On 10/18/2011 1:22 PM, Fulvio Scapin wrote:
>>> Hello again.
>>> I've been finally able to reproduce in a repeatable way the phenomenon
>>> I described a few weeks ago.
>>>
>>> Situation:
>>>
>>> Call from outside to the SipXECS-based IVR, dialing the internal
>>> extension directly through the IVR, let's say extension 31.
>>>
>>> Extension 31 (polycom soundpoint 550) rings.
>>>
>>> The operator at extension 30 (also polycom soundpoint 550) dials *7831
>>> to pick up the call.
>>> After he has dialed *7831 the user at extension 31 picks up the call
>>> and begins talking.
>>> A second or two later the user at extension 30 picks up the call as well.
>>>
>>> End result:
>>>
>>> A three-way call in which each of three parties involved can speak
>>> with and hear the other two.
>>>
>>>
>>> Am I the only one finding this peculiar?
>>>
>>> Bye,
>>> Fulvio Scapin
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