Re: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
You need two 4chan TDM cards with six total FXO modules to drive the six incoming lines on the Starplus. I am assuming that you will be taking out your POTS lines going into the StarPlus. ATA's or a small channel bank will also work for the * to analog conversion. And please don't post html to the list. Lyle - Original Message - From: Dave Covert (Sailtech) To: [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 9:15 AM Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) I plan to set up an Asterisk server later today or tomorrow to begin putzing and learning about it. Learn by doing... I would like to cut thru some of the confusion that such a flexible system tends to breed by quickly describing my end goal and getting some input from the 'group mind' as to the pieces I should concentrate my efforts on. We are a 5-person operation with 6 VOIP numbers an old-style POTS PBX (Vodavi Starplus 616EX) and a dozen 6-line desk phone stations. Rather than using a small bank of ATAs, we would like to use an Asterisk server to 'terminate' the VOIP lines and route them to both the Starplus desk phones and to softphones running on certain workstations. That is, a new incoming call would ring both the first unused line hooked to the Starplus and the first unused line on the softphones. So... The question is... to get that to work, what sort of hardware do I need in the Asterisk box to turn the incoming VOIP calls into a two-wire POTS input for the Starplus PBX and what is a suggested softphone we can use with Asterisk? Thank you for your time, Dave Covert, KB5GOG | Sailtech | Office 281-334-4690 | Fax 281-538-3270 | Email [EMAIL PROTECTED] | www.sailtechmarine.com Disclaimer INFORMATION PROVIDED IN THIS DOCUMENT IS PROVIDED AS IS WITHOUT WARRANTY REPRESENTATION OR CONDITION OF ANY KIND, EITHER EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO CONDITIONS OR OTHER TERMS OF MERCHANTABILITY AND/OR FITNESS FOR A PARTICULAR PURPOSE. THE USER ASSUMES THE ENTIRE RISK AS TO THE ACCURACY AND THE USE OF THIS DOCUMENT. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
Uh oh... Does that mean that my request for help - with opening statement take mercy on me - won't be reviewed =( -Original Message- From: Dave Covert (Sailtech) [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 10:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) Message: File: ATT1093894.txt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
You'll find the following web site to have a huge amount of information (too much really!) http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Covert (Sailtech) Sent: Wednesday, August 25, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) I plan to set up an Asterisk server later today or tomorrow to begin putzing and learning about it. Learn by doing... I would like to cut thru some of the confusion that such a flexible system tends to breed by quickly describing my end goal and getting some input from the 'group mind' as to the pieces I should concentrate my efforts on. We are a 5-person operation with 6 VOIP numbers an old-style POTS PBX (Vodavi Starplus 616EX) and a dozen 6-line desk phone stations. Rather than using a small bank of ATAs, we would like to use an Asterisk server to 'terminate' the VOIP lines and route them to both the Starplus desk phones and to softphones running on certain workstations. That is, a new incoming call would ring both the first unused line hooked to the Starplus and the first unused line on the softphones. So... The question is... to get that to work, what sort of hardware do I need in the Asterisk box to turn the incoming VOIP calls into a two-wire POTS input for the Starplus PBX and what is a suggested softphone we can use with Asterisk? Thank you for your time, Dave Covert, KB5GOG | Sailtech | Office 281-334-4690 | Fax 281-538-3270 | Email dave@ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
Recieving you loud and clear here on Moz. Thunderbird 0.7.2. I guess you're seeing the attachment because you sent an HTML mail to the list (doubtless you'll now get flamed by many). ;-) Cheers, Benjamin Huddleston, Robert wrote: Uh oh... Does that mean that my request for help - with opening statement take mercy on me - won't be reviewed =( -Original Message- From: Dave Covert (Sailtech) [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 10:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) Message: File: ATT1093894.txt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Benjamin Johnson Director thinktech Ltd. - Appropriate security solutions for business. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
Yah... that is where I started... which lead me to the asterisk.org site, which lead me to this list... and you are certainly right, it really is a huge amount of information (too much really!) I am hoping to get an idea of what I need for this project so I can go and read up on just those parts... Dave -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 9:26 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) You'll find the following web site to have a huge amount of information (too much really!) http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Covert (Sailtech) Sent: Wednesday, August 25, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) I plan to set up an Asterisk server later today or tomorrow to begin putzing and learning about it. Learn by doing... I would like to cut thru some of the confusion that such a flexible system tends to breed by quickly describing my end goal and getting some input from the 'group mind' as to the pieces I should concentrate my efforts on. We are a 5-person operation with 6 VOIP numbers an old-style POTS PBX (Vodavi Starplus 616EX) and a dozen 6-line desk phone stations. Rather than using a small bank of ATAs, we would like to use an Asterisk server to 'terminate' the VOIP lines and route them to both the Starplus desk phones and to softphones running on certain workstations. That is, a new incoming call would ring both the first unused line hooked to the Starplus and the first unused line on the softphones. So... The question is... to get that to work, what sort of hardware do I need in the Asterisk box to turn the incoming VOIP calls into a two-wire POTS input for the Starplus PBX and what is a suggested softphone we can use with Asterisk? Thank you for your time, Dave Covert, KB5GOG | Sailtech | Office 281-334-4690 | Fax 281-538-3270 | Email dave@ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
I've read almost everything on every site possible ever made on Asterisk =) I've posted to a gizzillion forums and email lists... I can understand not wanting to share proprietary information - so if someone is just able to tell me yes/no that this is capable of doing I would be happy... My original email Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do - and please someone let me know if this can be done... We have a large VoIP network (we are a communications carrier)... The gatekeeper (Lucent iMerge) supports MGCP/H.323 (soon SIP) and allows for calls to be made to the PSTN cloud via GR303 links in our class 5 switches. I would like to build Asterisks with H323 (or MGCP if need be - SIP availabe w/ future upgrades) and have it attach to our gatekeeper to access the PSTN. Instead of installing a T1/E1 or ISDN or POTS card we would like to use the existing VoIP network. Anyone ran into this before - can provide some direction? Asterisk would register itself against the Lucent iMerge Softphone would register with Asterisk And inbound/outbound calling could be completed to the PSTN cloud via the Lucent iMerge via Asterisk.. -- ---- -- PSTN Lucent Gatekeeper T1 (or broadband) Asterisk Softphone/endpoint -- ---- -- End My original email -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 10:26 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) You'll find the following web site to have a huge amount of information (too much really!) http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Covert (Sailtech) Sent: Wednesday, August 25, 2004 7:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) I plan to set up an Asterisk server later today or tomorrow to begin putzing and learning about it. Learn by doing... I would like to cut thru some of the confusion that such a flexible system tends to breed by quickly describing my end goal and getting some input from the 'group mind' as to the pieces I should concentrate my efforts on. We are a 5-person operation with 6 VOIP numbers an old-style POTS PBX (Vodavi Starplus 616EX) and a dozen 6-line desk phone stations. Rather than using a small bank of ATAs, we would like to use an Asterisk server to 'terminate' the VOIP lines and route them to both the Starplus desk phones and to softphones running on certain workstations. That is, a new incoming call would ring both the first unused line hooked to the Starplus and the first unused line on the softphones. So... The question is... to get that to work, what sort of hardware do I need in the Asterisk box to turn the incoming VOIP calls into a two-wire POTS input for the Starplus PBX and what is a suggested softphone we can use with Asterisk? Thank you for your time, Dave Covert, KB5GOG | Sailtech | Office 281-334-4690 | Fax 281-538-3270 | Email dave@ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
