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(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

Klaus Darilion schrieb: > this is quit difficult: Which SIP phones? Which

I use snom 360 and 200 phones, Asterisk 1.2.7.1

> > You have to make sure that Asterisk and the SIP

> There are several ways how to make a call

> > Also an often seen problem is the different

> Asterisk. Asterisk must be able to call B in the

Of course Asterisk is able to call A or B in the

Regards Bastian > > regards > klaus > > Bastian Schern wrote: >> Hello, >> >> does anybody got a working configuration to make

>> transfer" with a call through an Asterisk

>> >> Example: >> >> PSTN --> Asterisk --> SER --+-- A >> | >> +-- B >> >> The call will come from the PSTN Network and

>> sets the call on "Hold" and calls "B". After A

>> hangup an B got the call from PSTN. >> >> This in _not_ working at the moment. >> >> Attended call transfer only with OpenSER and

>> Problem. But if the is an Asterisk as PSTN-GW in

>> 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(a)openser.org >>

____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com

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