Dave Covert (Sailtech)
2004-Aug-25 07:15 UTC
[Asterisk-Users] YAAN (Yet Another Asterisk Newbie)
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Huddleston, Robert
2004-Aug-25 07:18 UTC
[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:dave@sailtechmarine.com] > Sent: Wednesday, August 25, 2004 10:16 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie) > > << Message: >> << File: ATT1093894.txt >>
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Covert (Sailtech) Sent: Wednesday, August 25, 2004 7:16 AM To: asterisk-users@lists.digium.com 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@
Huddleston, Robert
2004-Aug-25 07:33 UTC
[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:scott@evtmedia.com]
Sent: Wednesday, August 25, 2004 10:26 AM
To: dave@sailtechmarine.com; '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: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Covert
(Sailtech)
Sent: Wednesday, August 25, 2004 7:16 AM
To: asterisk-users@lists.digium.com
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@
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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: asterisk-users@lists.digium.com 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 dave@sailtechmarine.com | 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 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