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 <[EMAIL PROTECTED]> 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 > >>> [email protected] > >>> > 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 > [email protected] > http://openser.org/cgi-bin/mailman/listinfo/users > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Users mailing list [email protected] http://openser.org/cgi-bin/mailman/listinfo/users
