[Users] Attended Transfer

Frogger froggerandbongo at yahoo.com
Thu Apr 27 08:06:54 CEST 2006


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 at 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
> >>>
> >>> ____________
> >>> Virus checked by G DATA AntiVirusKit
> >>> Version: AVK 16.7010 from 25.04.2006
> >>> Virus news: www.antiviruslab.com
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Users mailing list
> >>> Users at openser.org
> >>>
> http://openser.org/cgi-bin/mailman/listinfo/users
> >>
> >
> >
> > ____________
> > Virus checked by G DATA AntiVirusKit
> > Version: AVK 16.7010 from 25.04.2006
> > Virus news: www.antiviruslab.com
> >
> >
> >
> 
> 
> 
> _______________________________________________
> Users mailing list
> Users at openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
> 


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