Re: [Asterisk-Users] sharing a line w/multiple extensions
On Thu, 2005-12-15 at 09:33 -0800, John Biundo wrote: I'm particularly worried about acceptance of this shared line (or lack thereof) aspect of the system. My wife will get the idea of extensions, transfers, parking, etc. because she uses a PBX at work, though I worry that the habits of how the phone is supposed to work at home may die hard with her. And the kids are a whole 'nuther story. IMHO, kids are the ones likely to use the more advanced features... ie, conferences, and anything else likely to tie up multiple in/outbound calls at the same time... You might consider permitting them to throw their own mp3's onto the server for their personal MoH I thought that having some common area phones share a single extension (wired into a single ATA FXS port) might ease the transition, but I'm also afraid it might be confusing (you can just pick up from these extensions, but you have to transfer or park to/from these extensions. Huh?). Program one of the speed-dial keys to transfer the call to a meetme with MoH. Then tell them that to put the call on hold, just press this button. Then label the next closest speed dial as Pickup which simply dials the meetme room. Of course, if you have multiple 'lines' you might need some 'custom' dialplan magic to ensure a hold will always add you to a new empty meetme, while an unhold will take you to the oldest meetme which hasn't been 'un-holded' yet :) Otherwise, you might as well teach them about parking Still, a button for Hold which does a #700, and then they just walk to any other phone and dial 701 (or whatever) should get you most of the way there. Just remember to set a short-ish parking timeout which will call-back the phone . The huge selling point, which I'm hoping will overcome any initial resistance, is the idea that one person will no longer tie up the whole phone system for the house when they make/take a call. And deploying one of my free DIDs to give my 16-year-old his own phone number that rings only in his bedroom is the real ace up my sleeve! Yep, neat + his own direct VM etc... Sure, Asterisk will come with a lot of other neat features, but frankly most of them have more geek appeal (though I have high hopes for my favorite feature -- announced caller id over the stereo/tivo while we're making dinner -- to revolutionize the way we deal with (or at least who answers ;-) ) phone calls at that hour), and in some cases I think may face similar that's not the way it's supposed to work objections. For example, while they will acknowledge that voicemail is cool, I suspect they'll miss the simplicity of walking into the kitchen, seeing if the answering machine is blinking, and just pressing the button. Use phones that have a VM indicator then program a speed dial for your VM extension. I'm excited AND anxious about starting a real beta test with them! Maybe that's why I'm already 3 weeks behind my original schedule. ;-) Well, looks like you are close, I think the biggest one to test thoroughly is the echo. That is probably the hardest one to ask people to live with... Regards, Adam ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? There might be a way for you to address your objective depending upon exactly what you're trying to do. The previous responses to your question _assume_ that each room in your case has a pbx extension (regardless of whether its a sip or analog phone). If their assumption is correct, then the responses are correct. However, if you want to use your existing analog phones and you group them together, several analog phones can share a single extension and those phones in the group can pick up and join the conversation whenever they want. Think in terms of using something like a Sipura sip adapter (or the equivalent from other vendors), and connecting all analog phones within your defined group to the rj11 analog jack of the adapter. For example, I have four analog pstn lines and multiple iax connections to various itsp's and clients. One of the analog pstn lines is a house line and connects directly to * via a TDM04b card. When an incoming call occurs on that line, it rings multiple sip phone/adapters. One of those happens to be a Sipura spa3000 that has most of the analog house phones attached. Anyone one of those phones can answer the call, others can join in, etc. The approach can work if you can define specific groups of interest such as kids vs adults, sales vs support, home vs business, etc. Combine that approach with carefull selection of analog phones (those with some form of line in use LED), and you end up with an approach that sort of looks like a poor-man's key system behind a pbx. Pay attention to the features within the sip adapter (eg, Sipura) and you're likely to find additional options that might address your needs. All depends upon exactly what it is that you're trying to engineer. ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
On Thu, Dec 15, 2005 at 06:11:07AM -0600, Rich Adamson said: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? There might be a way for you to address your objective depending upon exactly what you're trying to do. The previous responses to your question _assume_ that each room in your case has a pbx extension (regardless of whether its a sip or analog phone). If their assumption is correct, then the responses are correct. However, if you want to use your existing analog phones and you group them together, several analog phones can share a single extension and those phones in the group can pick up and join the conversation whenever they want. Think in terms of using something like a Sipura sip adapter (or the equivalent from other vendors), and connecting all analog phones within your defined group to the rj11 analog jack of the adapter. One system I found that works well in a home environment is using a two-line, multi-handset cordless phone system. Run 2 analog ports to the base station, and this handles most home needs. Two users can make or receive calls, join existing calls, etc rather easily. The dial plan is set so that either line makes outgoing calls over a VoIP service, line 2, or whatever, so that the main incoming line is always available to receive calls. The home office has a Polycom 601 with it's own lines and dial plan logic, plus the fact that the polycom user is much more likely to know how to answer, transfer, park, etc. Wife proofing a * system is non-trivial and takes careful planning. ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? Probably, but it would take some very creative dialplan programming and an external application to transfer the parties into a meetme room. You will not get 'pickup' behavior on the SIP phones regardless, they will have to press a speed dial button which would attempt to join the meetme. In other words: you can get there, but it will _not_ behave like a key system, and people will expect it to, so they will be frustrated when it doesn't. We've been down this road many times before, and many Asterisk installations have been taken out because the installers thought they could achieve key system behavior (or retrain the users) but failed. If you want to try, feel free... I'm only telling you what has happened before :-) Thanks. That's very helpful because being new to Asterisk, I don't know the history of what people have attempted to use Asterisk for. It's unfortunate that there's no way to do it because it sounds like others are looking for this same functionality. I wonder what it would take to implement this in Asterisk natively. Does Digium take feature requests? Certainly, this would have appeal for residential systems. Just a couple of comments on the subject of key systems verses pbx's The traditional pbx (from years ago) implemented exactly what Kevin mentioned above. Just about every company that deployed a pbx back then also had several key systems attached to their pbx. The key systems were typically limited to executives and their assistants (secretaries back then) primarily due to the additional cost of the older key systems. The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Rich Adamson wrote: The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Doubt they are losing much in sales. Sales of the digium cards are from geeks (like many of us on this list) and small companies selling asterisk into business accounts. Most 'friends family' wouldn't consider investing $1k for all the pieces necessary to have a reasonable system (even if they could use a retired PC) unless they're geeks as well. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? Not that I'm aware of, but I don't try to keep track of competing projects either. There's sort of a dichotomy thing going on where most development folks (whether its asterisk or some other I/T-type projects) are focused on programming some function/features that are system oriented (eg, odbc, sql support, echo cancellers, fax support, menues, jitterbuffers, architectural changes, scipts); and, another group without programming skills that would love to see additional basic pbx/key-system functions implemented that don't require someone to jump through hoops in the dialplan. But, that's the nature of open source projects. ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Fascinating discussion. The whole idea of acceptance of an asterisk based system by the rest of the family is probably worthy of its own thread. I'm in alpha test (I switch on asterisk after the wife leaves for work, switch it back before she gets home ;-) ) of my home asterisk system, so I've been thinking/worrying about a lot of similar issues. I'm particularly worried about acceptance of this shared line (or lack thereof) aspect of the system. My wife will get the idea of extensions, transfers, parking, etc. because she uses a PBX at work, though I worry that the habits of how the phone is supposed to work at home may die hard with her. And the kids are a whole 'nuther story. I thought that having some common area phones share a single extension (wired into a single ATA FXS port) might ease the transition, but I'm also afraid it might be confusing (you can just pick up from these extensions, but you have to transfer or park to/from these extensions. Huh?). The huge selling point, which I'm hoping will overcome any initial resistance, is the idea that one person will no longer tie up the whole phone system for the house when they make/take a call. And deploying one of my free DIDs to give my 16-year-old his own phone number that rings only in his bedroom is the real ace up my sleeve! Sure, Asterisk will come with a lot of other neat features, but frankly most of them have more geek appeal (though I have high hopes for my favorite feature -- announced caller id over the stereo/tivo while we're making dinner -- to revolutionize the way we deal with (or at least who answers ;-) ) phone calls at that hour), and in some cases I think may face similar that's not the way it's supposed to work objections. For example, while they will acknowledge that voicemail is cool, I suspect they'll miss the simplicity of walking into the kitchen, seeing if the answering machine is blinking, and just pressing the button. I'm excited AND anxious about starting a real beta test with them! Maybe that's why I'm already 3 weeks behind my original schedule. ;-) ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Rich, Kevin I just stubbled into one of these projects. I am making it work for the client but constantly running into walls. Client is very cool about the work and loved the history of the Key System from Rich. Current issues and solutions: Busy boxes = SNOM 360s with Sidecars Static Parking(orbit or 60) = Seperate MeetMe rooms (MOH if single user) Paging = Extra SIP line on the phones set to auto answer, ugly chanisavial check No Voicemail = a, bounce to Oper Queues = Hard Code procedural so that I can hint the phones right. answer at any phone = button hack for the blinking button. Maybe I should submit this job to Digium for the contest :-) Andrew On 12/15/05, Rich Adamson [EMAIL PROTECTED] wrote: The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Doubt they are losing much in sales. Sales of the digium cards are from geeks (like many of us on this list) and small companies selling asterisk into business accounts. Most 'friends family' wouldn't consider investing $1k for all the pieces necessary to have a reasonable system (even if they could use a retired PC) unless they're geeks as well. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? Not that I'm aware of, but I don't try to keep track of competing projects either. There's sort of a dichotomy thing going on where most development folks (whether its asterisk or some other I/T-type projects) are focused on programming some function/features that are system oriented (eg, odbc, sql support, echo cancellers, fax support, menues, jitterbuffers, architectural changes, scipts); and, another group without programming skills that would love to see additional basic pbx/key-system functions implemented that don't require someone to jump through hoops in the dialplan. But, that's the nature of open source projects. ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger
Re: [Asterisk-Users] sharing a line w/multiple extensions
Fascinating discussion. The whole idea of acceptance of an asterisk based system by the rest of the family is probably worthy of its own thread. I'm in alpha test (I switch on asterisk after the wife leaves for work, switch it back before she gets home ;-) ) of my home asterisk system, so I've been thinking/worrying about a lot of similar issues. I've sort of done the same by using a spa3k for the house phones, and its configured so the spouse doesn't have a clue its in the middle. No remedial training required! (But, if someone picks up a house phone and happens to dial an * extension, it handles it. By prefixing any dialed number with an 8, the call is routed via *. If * is down, no one notices. ;) I'm particularly worried about acceptance of this shared line (or lack thereof) aspect of the system. My wife will get the idea of extensions, transfers, parking, etc. because she uses a PBX at work, though I worry that the habits of how the phone is supposed to work at home may die hard with her. And the kids are a whole 'nuther story. I thought that having some common area phones share a single extension (wired into a single ATA FXS port) might ease the transition, but I'm also afraid it might be confusing (you can just pick up from these extensions, but you have to transfer or park to/from these extensions. Huh?). Just put all the phones on one ata and let everyone kind of step slowly into added functionality. Then add an itsp did number (or one for each person) and use something like distinctive rings for each. Add on some voicemail when their comfortable, a decent sip speakerphone, some analog phones with a VM LED, etc. Pretty soon they'll be asking if you can do a, b, or c with the system. The huge selling point, which I'm hoping will overcome any initial resistance, is the idea that one person will no longer tie up the whole phone system for the house when they make/take a call. And deploying one of my free DIDs to give my 16-year-old his own phone number that rings only in his bedroom is the real ace up my sleeve! Make that a two line cordless and he'll jump all over it. One up on his buddies. ;) Sure, Asterisk will come with a lot of other neat features, but frankly most of them have more geek appeal (though I have high hopes for my favorite feature -- announced caller id over the stereo/tivo while we're making dinner -- to revolutionize the way we deal with (or at least who answers ;-) ) phone calls at that hour), and in some cases I think may face similar that's not the way it's supposed to work objections. For example, while they will acknowledge that voicemail is cool, I suspect they'll miss the simplicity of walking into the kitchen, seeing if the answering machine is blinking, and just pressing the button. I'm excited AND anxious about starting a real beta test with them! Maybe that's why I'm already 3 weeks behind my original schedule. ;-) Have fun. :) ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Is there a way for another extension to join a call in progress? i.e. If I can't share the line with all extensions, it would be nice to have a single button (dial sequence) that allows any extension to join the call. How can this be configured? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
On 12/15/05, Robert La Ferla [EMAIL PROTECTED] wrote: Is there a way for another extension to join a call in progress? i.e. If I can't share the line with all extensions, it would be nice to have a single button (dial sequence) that allows any extension to join the call. How can this be configured? Probably through some creative use of meetme. But IMO once you start using hacks like that, you should probably either look at alternatives or adapt to a different way of doing things. It's also possible that there is a solution that would work well for you, but as always it depends on the details. You might post your exact setup, hardware, number of phones, what type of environment they are used in, etc.. For example as someone else mentioned in a home environment you can do something similiar to what you want using the sipura SPA-3000. It's possible, depending on the details, that a similar solution might exist in your case. I kind of doubt it, but it can't hurt to try. Chris ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
I'm particularly worried about acceptance of this shared line (or lack thereof) aspect of the system. My wife will get the idea of extensions, transfers, parking, etc. because she uses a PBX at work, though I worry that the habits of how the phone is supposed to work at home may die hard with her. And the kids are a whole 'nuther story. If you're using VoIP, then you don't have to worry about the shared line issue. Outgoing calls will work all the time (well, assuming your VoIP provider works which is a big IF I guess). You can keep your landline for 911 calls. The huge selling point, which I'm hoping will overcome any initial resistance, is the idea that one person will no longer tie up the whole phone system for the house when they make/take a call. And deploying one of my free DIDs to give my 16-year-old his own phone number that rings only in his bedroom is the real ace up my sleeve! Sure thing :) Sure, Asterisk will come with a lot of other neat features, but frankly most of them have more geek appeal (though I have high hopes for my favorite feature -- announced caller id over the stereo/tivo while we're making dinner -- to revolutionize the way we deal with (or at least who answers ;-) ) phone calls at that hour), and in some cases I think may face similar that's not the way it's supposed to work objections. For example, while they will acknowledge that voicemail is cool, I suspect they'll miss the simplicity of walking into the kitchen, seeing if the answering machine is blinking, and just pressing the button. You could use the voicemail to email feature. It's as nice as it gets. Who doesn't check their emails like 20 times a day nowadays? :) Cheers, Jean-Michel. ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Look at meetme, also FOP (www.asternic.org) can do that for you. On 12/14/05, Robert La Ferla [EMAIL PROTECTED] wrote: I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Let me revise this a little: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) C F wrote: Look at meetme, also FOP (www.asternic.org) can do that for you. On 12/14/05, Robert La Ferla [EMAIL PROTECTED] wrote: I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Sean Cook wrote: also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) I have that but once one extension picks up, others can't join in. Well, at least when I tried it with mixed SIP and Zap, it didn't work. Maybe all SIP does but I need it to work for all phones SIP and analog (via Zap). ___ --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
RE: [Asterisk-Users] sharing a line w/multiple extensions
Have you try first blind transfer to a meetme meeting room. Then multiple user can join in. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Thursday, December 15, 2005 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sharing a line w/multiple extensions Sean Cook wrote: also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) I have that but once one extension picks up, others can't join in. Well, at least when I tried it with mixed SIP and Zap, it didn't work. Maybe all SIP does but I need it to work for all phones SIP and analog (via Zap). ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Robert La Ferla wrote: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Kevin P. Fleming wrote: Robert La Ferla wrote: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Robert La Ferla wrote: phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? Probably, but it would take some very creative dialplan programming and an external application to transfer the parties into a meetme room. You will not get 'pickup' behavior on the SIP phones regardless, they will have to press a speed dial button which would attempt to join the meetme. In other words: you can get there, but it will _not_ behave like a key system, and people will expect it to, so they will be frustrated when it doesn't. We've been down this road many times before, and many Asterisk installations have been taken out because the installers thought they could achieve key system behavior (or retrain the users) but failed. If you want to try, feel free... I'm only telling you what has happened before :-) ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Kevin P. Fleming wrote: Robert La Ferla wrote: phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? Probably, but it would take some very creative dialplan programming and an external application to transfer the parties into a meetme room. You will not get 'pickup' behavior on the SIP phones regardless, they will have to press a speed dial button which would attempt to join the meetme. In other words: you can get there, but it will _not_ behave like a key system, and people will expect it to, so they will be frustrated when it doesn't. We've been down this road many times before, and many Asterisk installations have been taken out because the installers thought they could achieve key system behavior (or retrain the users) but failed. If you want to try, feel free... I'm only telling you what has happened before :-) Thanks. That's very helpful because being new to Asterisk, I don't know the history of what people have attempted to use Asterisk for. It's unfortunate that there's no way to do it because it sounds like others are looking for this same functionality. I wonder what it would take to implement this in Asterisk natively. Does Digium take feature requests? Certainly, this would have appeal for residential systems. ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Just as an aside... some Sip ATA's come with both Hotline Warmline settings with configurable timeouts. Do any software ones offer similar functionality? Regardless, it could achieve almost that outcome using Wamline (to auto-enter the meetme-finding-extension that interfaces with external program to find actual meetme for the conversation to join and then transfer the call into) and suitably short timeouts. It would have the corresponding effect of allowing people to 'optionally' join the call by just picking up and waiting 5 seconds.. or alternatively, dialling a differnet number and launching a seperate call. I could picture it working 'almost' like the author wants through the use of cordless phones/hardphones plugged into an ATA that supports wamline configured with the below said 'messy in the middle dealing-with-meetme' program Kevin P. Fleming wrote: Robert La Ferla wrote: phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? Probably, but it would take some very creative dialplan programming and an external application to transfer the parties into a meetme room. You will not get 'pickup' behavior on the SIP phones regardless, they will have to press a speed dial button which would attempt to join the meetme. In other words: you can get there, but it will _not_ behave like a key system, and people will expect it to, so they will be frustrated when it doesn't. We've been down this road many times before, and many Asterisk installations have been taken out because the installers thought they could achieve key system behavior (or retrain the users) but failed. If you want to try, feel free... I'm only telling you what has happened before :-) ___ --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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] sharing a line w/multiple extensions
Robert La Ferla wrote: Thanks. That's very helpful because being new to Asterisk, I don't know the history of what people have attempted to use Asterisk for. It's unfortunate that there's no way to do it because it sounds like others are looking for this same functionality. I wonder what it would take to implement this in Asterisk natively. Does Digium take feature requests? Certainly, this would have appeal for residential systems. Our list of feature requests is a mile long, but yes, we take them. Money makes them happen sooner, sometimes :-) Implementation of this in a proper way will require true 'shared line' functionality in Asterisk, which is a non-trivial thing to add. On top of that, there is no money in 'residential' Asterisk systems, because cheap key systems can do the job for less than the SIP phones alone would cost, let alone a box to run Asterisk on and TDM interface hardware. It has a high 'geek factor', but regular people would just get frustrated by it (in my opinion, of course). That means the people who really desperately want this behavior also have no resources to make it happen... so we are at a stand-still. ___ --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
