[asterisk-users] Voicemail Platform

[Roger Workman]

Has anyone got * to work just as a voicemail platform?  The problems I'm having is when 

* answers the call 
Caller enters extension
* needs to issue hookflash dial extension the hang up

The last part is where I'm having the nag...getting * to do a hook flash on the analog ports and hanging up

Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
 Fax:   304.324.3801 
ICQ: 4447584
FWD Network: 56505
Website: http://www.upperclassman.net
Billing Questions: billing at upperclassman.net
Rental Questions: rentals at upperclassman.net
Maintenance: help at upperclassman.net 
 
 

This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Roger Workman
Sent: Monday, August 07, 2006 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FXS gateway/Channel Bank

Can someone recommend a good FXS gateway/Channel bank that will intergrate smoothly with *  I need to port over 158 analog lines

Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
 Fax:   304.324.3801 
ICQ: 4447584
FWD Network: 56505
Website: http://www.upperclassman.net
Billing Questions: billing at upperclassman.net
Rental Questions: rentals at upperclassman.net
Maintenance: help at upperclassman.net 
 
 

This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leon Sun
Sent: Monday, August 07, 2006 3:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to link 2 existing calls

Hi,

I searched web for few hours and couldn't find any solution about linking 2
calls from Asterisk. This is scenario.

1. A call has been connected from A pstn gateway to my Asterisk waiting with
music.
2. Meanwhile, B call has been connected from B pstn gateway to my asterisk
waiting with music.
3. My asterisk has an application that issues a request to link A call and B
call.
4. Asterisk should issue a re-invite to both A and B gateway and let them
exchange RTP directly. Asterisk should still be working as SIP proxy to
collect signaling(like bye).

Would please anyone suggest how to do step 3 and 4? I wouldn't prefer
conference room type since I like RTP packets go through gateway directly.



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Hadley Rich
Sent: Sunday, August 06, 2006 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ring Groups

On Monday 07 August 2006 06:36, Chris Hembrow wrote:
> I am new to asterisk, and learning as I plod along. Currently, I am
> trying to work out how to create a ring group without using AMP.

You should check out the book - 'Asterisk: The Future of Telephony' - 
published by O'Reilly it's available to buy or download. It will give you a 
good starting point.

> I set my inbound line to ring multiple lines by using
> Dial(SIP/101,SIP/102) but when I answered the call, the lines which
> didn't answer became locked with no dialtone, as though on a call.

That dial line should be Dial(SIP/101&SIP/102) - the comma (or a pipe, |) 
separates what to dial from the options to the dial command. typing 'show 
application dial' from the Asterisk CLI will get you all the gory details.

> I thought that setting up a ring group might help, but could only find
> references to creating them through AMP.

'Ring Group' is just an AMP term, you are going about it the right way
above.

HTH

hads

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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