Re: [asterisk-users] Learn some terminalogy before mountingthistask.
Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
SLA is not BLF. The only thing you need to configure to have BLF is adding hint priority to your dial plan. On 8/8/07, James Collier [EMAIL PROTECTED] wrote: Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
Thank you for your reply as it is exactly what we would need. Sorry I didn't find it myself. I do have a question about configuration within Asterisk. I'm reading the PDF on the Cisco Expansion module and it says 'When used as a DN key buttons are illuminated ...' Is that what we are doing within Asterisk or Trixbox when we configure an extension? (A Directory Number??) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Monday, August 06, 2007 7:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mountingthistask. On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote: In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? You're probably looking for something like this: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html I have no experience integrating this specific piece of hardware with Asterisk, but I've done what you're trying to do with the Grandstream equivalent for our front reception: http://www.grandstream.com/gxp2000.html and http://www.grandstream.com/gxp2000ext.html As I understand it, so long as the device can do a SIP SUBSCRIBE for each extension you want to monitor and you configure hints in your Asterisk dialplan for those extensions, it should work. You may need to set 'subscribecontext' (in sip.conf) for the phone that will be watching the extensions unless your hints are in the same context as the phone uses for outbound dialing. Of course, what the device does with the various payloads contained in the SIP NOTIFY messages is going to be different for each phone. On the Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing red) distinctly, but 'unavailable' and 'in use' are both mapped to a solid red, which makes it somewhat useless for transiently connected user agents like softphones. Hopefully someone with experience will speak up and confirm that the 7900 series does interop properly with Asterisk for SUBSCRIBE and NOTIFY. If that doesn't work, you could always go with a software solution, like the Flash Operator Panel. voip-info has a list (look at the Operator section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI -- j. -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
On 8/6/07, James R. Stevens [EMAIL PROTECTED] wrote: I'm reading the PDF on the Cisco Expansion module and it says 'When used as a DN key buttons are illuminated …' Is that what we are doing within Asterisk or Trixbox when we configure an extension? (A Directory Number??) I suspect DN Key is just one way of describing a multi-function button that can both display extension status and serve as a speed dial / transfer destination. On the Grandstream I have to configure the expansion car buttons as Asterisk BLF buttons, even though BLF (busy lamp field) isn't an Asterisk setting that I turn on. To enable BLF functionality in Asterisk, I have to set up hints in the dialplan and configure the user agent to subscribe to status notitications for those extensions. I'd search for asterisk user testimonials to be safe (assuming nobody steps up and says I got that working). Often times you'll find someone's blog about how they got a feature working with a particular piece of hardware, along with configuration samples. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
The 7914 only works under SCCP; the SIP firmware does not support it at all (the expansion panel won't even power on fully.) The SCCP channel driver under Asterisk doesn't really support the 7914 very well, currently it will only show onhook/offhook state (though there has been much discussion recently about changing this.) If you want to do this with SIP then you're better off with something like the grandstream mentioned, or just use the Flash Operator Panel (IMO it gives you more flexibility at a much lower cost.) I have personally found receptionist phone functionality handled much better with FOP. I have a 7914 and its functionality (and usefulness) is very limited under Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James R. Stevens Sent: Monday, August 06, 2007 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mountingthistask. Thank you for your reply as it is exactly what we would need. Sorry I didn't find it myself. I do have a question about configuration within Asterisk. I'm reading the PDF on the Cisco Expansion module and it says 'When used as a DN key buttons are illuminated ...' Is that what we are doing within Asterisk or Trixbox when we configure an extension? (A Directory Number??) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Monday, August 06, 2007 7:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mountingthistask. On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote: In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? You're probably looking for something like this: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html I have no experience integrating this specific piece of hardware with Asterisk, but I've done what you're trying to do with the Grandstream equivalent for our front reception: http://www.grandstream.com/gxp2000.html and http://www.grandstream.com/gxp2000ext.html As I understand it, so long as the device can do a SIP SUBSCRIBE for each extension you want to monitor and you configure hints in your Asterisk dialplan for those extensions, it should work. You may need to set 'subscribecontext' (in sip.conf) for the phone that will be watching the extensions unless your hints are in the same context as the phone uses for outbound dialing. Of course, what the device does with the various payloads contained in the SIP NOTIFY messages is going to be different for each phone. On the Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing red) distinctly, but 'unavailable' and 'in use' are both mapped to a solid red, which makes it somewhat useless for transiently connected user agents like softphones. Hopefully someone with experience will speak up and confirm that the 7900 series does interop properly with Asterisk for SUBSCRIBE and NOTIFY. If that doesn't work, you could always go with a software solution, like the Flash Operator Panel. voip-info has a list (look at the Operator section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI -- j. -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner http://www.athensdistributing.com/ , and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
what you are reading on Cisco manual DN is a completely different concept that what we are dealing in asterisk. In CME you refer to each number as a DN, that concept does not exist on Asterisk. Although Asterisk support SCCP (Skinny) and H323, but its always easier and better to use SIP or IAX. if you like to have a reception phone with BLF, there are lots of options to choose from. Beside the fact that i don't like quality of Cisco Phones, I usually get better and professional results with Polycom. But again that is my opinion. On 8/6/07, Ryan Amos [EMAIL PROTECTED] wrote: The 7914 only works under SCCP; the SIP firmware does not support it at all (the expansion panel won't even power on fully.) The SCCP channel driver under Asterisk doesn't really support the 7914 very well, currently it will only show onhook/offhook state (though there has been much discussion recently about changing this.) If you want to do this with SIP then you're better off with something like the grandstream mentioned, or just use the Flash Operator Panel (IMO it gives you more flexibility at a much lower cost.) I have personally found receptionist phone functionality handled much better with FOP. I have a 7914 and its functionality (and usefulness) is very limited under Asterisk. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *James R. Stevens *Sent:* Monday, August 06, 2007 10:41 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Learn some terminalogy before mountingthistask. Thank you for your reply as it is exactly what we would need. Sorry I didn't find it myself. I do have a question about configuration within Asterisk. I'm reading the PDF on the Cisco Expansion module and it says 'When used as a DN key buttons are illuminated …' Is that what we are doing within Asterisk or Trixbox when we configure an extension? (A Directory Number??) *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *James FitzGibbon *Sent:* Monday, August 06, 2007 7:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Learn some terminalogy before mountingthistask. On 8/5/07, *James R. Stevens* [EMAIL PROTECTED] wrote: In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? You're probably looking for something like this: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html I have no experience integrating this specific piece of hardware with Asterisk, but I've done what you're trying to do with the Grandstream equivalent for our front reception: http://www.grandstream.com/gxp2000.html and http://www.grandstream.com/gxp2000ext.html As I understand it, so long as the device can do a SIP SUBSCRIBE for each extension you want to monitor and you configure hints in your Asterisk dialplan for those extensions, it should work. You may need to set 'subscribecontext' (in sip.conf) for the phone that will be watching the extensions unless your hints are in the same context as the phone uses for outbound dialing. Of course, what the device does with the various payloads contained in the SIP NOTIFY messages is going to be different for each phone. On the Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing red) distinctly, but 'unavailable' and 'in use' are both mapped to a solid red, which makes it somewhat useless for transiently connected user agents like softphones. Hopefully someone with experience will speak up and confirm that the 7900 series does interop properly with Asterisk for SUBSCRIBE and NOTIFY. If that doesn't work, you could always go with a software solution, like the Flash Operator Panel. voip-info has a list (look at the Operator section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI -- j. -- This message has been scanned for viruses and dangerous content by *Athens Hyperion Scanner*http://www.athensdistributing.com/, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
