mmiranda@americatel.com.sv
2005-Jan-21 11:20 UTC
[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
mmiranda@americatel.com.sv 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.
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 Asterisk server has access to more than 1 line. If it does, then you should be able to 3-way call without any problems. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mmiranda@americatel.com.sv Sent: Friday, January 21, 2005 1:20 PM To: asterisk-users@lists.digium.com Subject: [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://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
mmiranda@americatel.com.sv
2005-Jan-21 12:41 UTC
[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 (in and oout) to these gateways, i dont have any special hardware in the asterisk server. thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On 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? 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 Asterisk server has access to more than 1 line. If it does, then you should be able to 3-way call without any problems. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of mmiranda@americatel.com.sv Sent: Friday, January 21, 2005 1:20 PM To: asterisk-users@lists.digium.com Subject: [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://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi!> According to this only the BT102D supports "conferencing" aka 3-way > calling: > http://www.grandstream.com/Product_Spec.pdfThis is vaporware, the product was cancelled. Anyway, with a recent firmware you can do attended transfers with the BudgeTone, but no phone based 3-person-conferencing. Cheers, Philipp