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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
