I too have seen this problem using both Cisco and
Polycom phones.
I have tried a very slim openser config in order to
eliminate as many variables and still no success.
Cannot get inbound PSTN calls to "warm-transfer" from
UA1 to UA2. SIP to SIP transfer to PSTN is fine.
Interestingly, on Polycom (and I suspect Cisco too),
when a warm transfer is attempted, the transferring
party cannot retrieve the call after the transfer key
is hit the second time.
I have many traces but would be happy to do quite a
bit more testing and post results if anyone has
additional advice on some steps to investigate.
F
--- Klaus Darilion <klaus.mailinglists(a)pernau.at>
wrote:
then we will need some more SIP dumps to help you.
"ngrep -d any port 5060" on the SIP proxy.
regards
klaus
On Tue, April 25, 2006 20:00, Bastian Schern said:
Klaus Darilion schrieb:
> this is quit difficult: Which SIP phones? Which
version of Asterisk? ...
I use snom 360 and 200 phones, Asterisk 1.2.7.1
and OpenSER 1.0.1
>
> You have to make sure that Asterisk and the SIP
phones are
"compatible".
> There are several ways how to make a call
transfer.
>
> Also an often seen problem is the different
dialing plans on openser and
> Asterisk. Asterisk must be able to call B in
the
same way (same request
URI) then
A calls B.
Of course Asterisk is able to call A or B in the
same way.
Regards
Bastian
>
> regards
> klaus
>
> Bastian Schern wrote:
>> Hello,
>>
>> does anybody got a working configuration to make
an "attended call
>> transfer" with a call through an
Asterisk
gateway?
>>
>> Example:
>>
>> PSTN --> Asterisk --> SER --+-- A
>> |
>> +-- B
>>
>> The call will come from the PSTN Network and
will go through
"A". A
>> sets the call on "Hold" and
calls "B". After A
is connected with B, A
>> hangup an B got the call from PSTN.
>>
>> This in _not_ working at the moment.
>>
>> Attended call transfer only with OpenSER and
only SIP-Phones is no
>> Problem. But if the is an Asterisk as
PSTN-GW in
the game it will not
>> work.
>>
>> Regards
>> Bastian
>>
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