Re: [Asterisk-Users] three way call using sip

2005-01-22 Thread Philipp von Klitzing
Hi! According to this only the BT102D supports conferencing aka 3-way calling: http://www.grandstream.com/Product_Spec.pdf This is vaporware, the product was cancelled. Anyway, with a recent firmware you can do attended transfers with the BudgeTone, but no phone based

[Asterisk-Users] three way call using sip

2005-01-21 Thread mmiranda
Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] three way call using sip

2005-01-21 Thread Eric Wieling
[EMAIL PROTECTED] wrote: Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? The BT101 cannot to supervised transfers or 3-way calling. ___ Asterisk-Users mailing list

RE: [Asterisk-Users] three way call using sip

2005-01-21 Thread Paul Rodan
: [Asterisk-Users] three way call using sip Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] three way call using sip

2005-01-21 Thread Eric Wieling
Paul Rodan wrote: The BT100's do support conferencing, most SIP phones do. But how does your Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind; or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet, VoicePulse? You basically need to make sure your

RE: [Asterisk-Users] three way call using sip

2005-01-21 Thread mmiranda
PROTECTED] Behalf Of Paul Rodan Sent: Friday, January 21, 2005 12:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] three way call using sip The BT100's do support conferencing, most SIP phones do. But how does your Asterisk connect you to the PSTN

RE: [Asterisk-Users] three way call using sip - SOLVED -

2005-01-21 Thread mmiranda
PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, January 21, 2005 1:41 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] three way call using sip I connect to the PSTN using cisco as5400 gateways, this cisco devices have E1's to a DMS300 switch. I mean, i configured sip channels

Re: [Asterisk-Users] three way call using sip

2005-01-21 Thread C F
You are right that according to theri web site it does, however it just doesn't work. The following is an email I received from them: From me: Thanks for your reply, I figured it out. However I have another problem, is the Conference button suppose to work? Also when putting someone on hold if I