> hi .. > > I have an asterisk system with three TDM100P (single port FXO) cards > and 10 Grandstream 100 phones connected to it ..
The TDMx00P cards are FXS cards.. :) > > 1st question: > when i phone out > or receive a call from one of the SIP phones onto the PSTN, there is > a LOT of local echo in the handset .. the PSTN end of the call does not > here this echo, but it's VERY annoying on the SIP end of things .. > the echo seems to be about 0.3 seconds delayed to the speech .. > there is no echo on incoming voice, just an echo of my own voice > as I speak. What are you using to connect to the PSTN?? X100P, T100P, E100P, I4L, Chan_Capi.... > > 2nd question: > using a grandstream phone & asterisk, if I hear another phone ringing, > how can answer it from the phone infront of me? eg. if extension 6003 > is ringing, and i have phone number 6004, how can I answer it ? You need to setup call groups, search through the archives cos I rememeber a thread on this a short while ago.. > > 3rd question: > can someone give me some "starter hints" to configure call parking ? > I haven't managed to find a direct way to transfer a call from phone > to phone except using blind transfer and I want the person initiating > the transfer to speak to the receiving person before actually passing > the call. As far as I know there is no facility to do a consultative transfer on the GS phones.. Only a blind transfer.. Maybe it will come later.. > > can anybody help please ? > > cheers > Dave A Caruana > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users