Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). Some IP phones support this Which ones? With Thomson ST2030, using telnet, you can for instance : - check current forwarding status (is it forwarded ? toward which number ?), - and change forwarding status. In this case, forwarding is still distributed but its negative consequences are, IMHO, mitigated with dialplan magic. You should also be able to centralize forwarding with phone's StarCodes and ServiceMonitoring features (but I've not tried it yet) : instead of using phone GUI to turn on or off or monitor forwarding, the phone will send INVITE or SUBSCRIBE with appropriate data. With High end XML supporting business phones, it shouldn't be too hard to tune GUI. To illustrate what I meant, I recently asked a customer to send me an Alcatel user manual. Though I haven't implemented it yet, I couldn't any feature I couldn't mimic with Asterisk/SIP Phones. Sure, it's a huge effort for 1 customer. Could you easily have a Panasonic user manual in english ? If so, if you could send it to me off-list, I would be very curious to dig deeper into it and discover interactions we could or couldn't mimic with Asterisk. Cheers ___ -- 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] Panasonic x Asterisk if I can emulate Panasonic fast!
2008/10/16 C F [EMAIL PROTECTED] * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. What would be missing to integrate this feature ? With features.conf, it should be possible to map key combinations to an Asterisk application (maybe an AGI script ?) From there, it should be possible to drive SIP hardphones BLF status, don't you think ? ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 has this as well). ** Company internal directory on the phone updated on the PBX Some (most ?) IP phones support this ** System Speed Dial on the display updated by the PBX This one is interesting. I can't see a way to do it. Ant idea ? ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). Some IP phones support this ** On screen Voicemail (on the phone). high end ip phones (XML) should support ** Line assignment to buttons with LED indication, and hold indication. For this one, I don't know. SCA, maybe ? ** Hold ringback (some IP phones support it). There are many more features but I can't remember them at the moment. Granted in bigger installations there many more factors and usually more funding which makes the above list almost obsolete for the features that Asterisk does have. Again my advice do not go with Asterisk for this installation go with Panasonic. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- 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 ___ -- 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] Panasonic x Asterisk if I can emulate Panasonic fast!
** System Speed Dial on the display updated by the PBX This one is interesting. I can't see a way to do it. Ant idea ? P-Asserted Identity ? Most Business SIP phones support it. At the moment, I think that Asterisk wouldn't update caller's phone screen but hopefully, it should be one day. ___ -- 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] Panasonic x Asterisk if I can emulate Panasonic fast!
Am Mittwoch, den 15.10.2008, 21:03 -0400 schrieb C F: On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza [EMAIL PROTECTED] wrote: Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) Not really so. Depending on lots of factors, usually for a small office of only 5-10 users Panasonic is more feature rich. Since the main feature they are looking for in a PBX is to be able to yell across the hallway; hey boss call on 5 it's your wife which is not really possible with Asterisk (yeah I know call parking, but how many phones support it flawlessly with flashing LEDs?). Other features that are quite popular in small offices and not supported by Asterisk: * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 has this as well). ** Company internal directory on the phone updated on the PBX ** System Speed Dial on the display updated by the PBX ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). ** On screen Voicemail (on the phone). ** Line assignment to buttons with LED indication, and hold indication. ** Hold ringback (some IP phones support it). There are many more features but I can't remember them at the moment. Ok, those are to consider, thanks for being specific. Negatives, for me: Forwarding is an important issue for us. I'll read more, search for equivalent equipment before taking the decision. The same with line assignment to buttons. Ok, for me: Screening: do not need it. LEDs: Due to internal policies, we usually buy the essential, so we have just 1 phone with leds, the operator's. I'll buy the best phone for the operator. The rest must be handled manually by her. No queues. No problem with directories. Voicemail find I better on *. The rest, we will suffer, not important for us. Granted in bigger installations there many more factors and usually more funding which makes the above list almost obsolete for the features that Asterisk does have. Again my advice do not go with Asterisk for this installation go with Panasonic. Maybe this is the time for us to switch to *. In some point we must start this new tech. Anyway, thanks for the advice. Rodolfo. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- 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
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
If you want to wow them with GUI stuff (and make it easy for you, since the settings are generally correct out of the box) then download and install EVB (Easy Box Box). Another FreePBX/Asterisk based GUI with Webmin and lots of other good programs pre-installed. I would not use it for a very large install, at least without moving the DB (at the least) to separate box. To WOW them, open a web browser to EVB and and SSH window connected to the Asterisk console with a good amount of debugging. I don't know how meeting are done there, but have a meeting room with a projector, and MIC that is always better than five guys crowding around a computer screen. Presentation is key. The simplicity of the GUI coupled with the complexity of the CLI with verbose and sip debug turned on, will leave a lasting impression. Show them the ease an power of FreePBX (so learn it prior to the demo) so you feel confident and can answer questions on the fly. Show them the mouse over and even let the least computer savvy person create and activate a Holiday greeting/IVR. I find that driving the whole time is not so good, let everyone sit down and do something, add an extension, make an IVR, setup routing, followme, call screening, or whatever. Show them backup scheduling and Samba, so you can pull or push down backups on a regular schedule. EVB has quite a bit more than FreePBX and is much better than Trixbox as far as I am concerned. There is one add in EVB, but you can remove that with a little know how. You can also add your company's log which is always a nice touch. If someone has an Iphone, set them up as a user and show them that their message will show up in their email and they can play the attachment. There is plenty that Asterisk cannot do compared to a small key system, but I think that Asterisk more than makes up for those short comings with other features. If you get a firm understanding of it, you can show them how easy it is to create an IVR, Add an Extension, FollowMe, Call Screening, but base your presentation on what the Panasonic can do, then what Asterisk (FreePBX) can do. Focus on the positives, be confident, and if someone throws you a curve ball, don't fumble around for an answer. Do a quick Googe, or simply say, I don't know but I will find out. There is nothing wrong with not knowing something, but fumbling around or making up an answer leaves a bad taste. I respect I don't know, but I will get you an answer. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Oct 15, 2008 at 9:03 PM, C F [EMAIL PROTECTED] wrote: On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza [EMAIL PROTECTED] wrote: Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) Not really so. Depending on lots of factors, usually for a small office of only 5-10 users Panasonic is more feature rich. Since the main feature they are looking for in a PBX is to be able to yell across the hallway; hey boss call on 5 it's your wife which is not really possible with Asterisk (yeah I know call parking, but how many phones support it flawlessly with flashing LEDs?). Other features that are quite popular in small offices and not supported by Asterisk: * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 has this as well). ** Company internal directory on the phone updated on the PBX ** System Speed Dial on the display updated by the PBX ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). ** On screen Voicemail (on the phone). ** Line assignment to buttons with LED indication, and hold indication. ** Hold ringback (some IP phones support it). There are many more features but I can't remember them at the moment. Granted in bigger installations there many more factors and usually more funding which makes the above list almost obsolete for the features that Asterisk does have. Again my advice do not go with Asterisk for this installation go with Panasonic. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please:
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
On Thu, Oct 16, 2008 at 8:45 AM, Olivier [EMAIL PROTECTED] wrote: 2008/10/16 C F [EMAIL PROTECTED] * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. What would be missing to integrate this feature ? With features.conf, it should be possible to map key combinations to an Asterisk application (maybe an AGI script ?) From there, it should be possible to drive SIP hardphones BLF status, don't you think ? Yes and no, the real thing would be to be able to get a status feedback from Asterisk that it's actually recording, then based on that one would be able to use devstate from 1.6 to turn on BLF. For where to store the recordings, the best way would be if arg could be passed to the recording app indicating to which voicemail user to send it and follow the settings (email, pager etc.) for that user. ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 has this as well). ** Company internal directory on the phone updated on the PBX Some (most ?) IP phones support this The phones support it, but not from asterisk, what that requires is a separate provisioning system for each type of phone, that pulls the data from central database, not that hard to build and maintain in theory, just very costly to develop since such a provisioning system doesn't yet exist, at least AFAIK. ** System Speed Dial on the display updated by the PBX This one is interesting. I can't see a way to do it. Ant idea ? As far as SIP goes, no it's impposible, however as far as end users/admins are concerned the PBX is/could also be the system provisoining the phones, in which case what I wrote above could be done. ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). Some IP phones support this Which ones? ** On screen Voicemail (on the phone). high end ip phones (XML) should support Again XML, expensive development costs. ** Line assignment to buttons with LED indication, and hold indication. For this one, I don't know. SCA, maybe ? ** Hold ringback (some IP phones support it). There are many more features but I can't remember them at the moment. Granted in bigger installations there many more factors and usually more funding which makes the above list almost obsolete for the features that Asterisk does have. Again my advice do not go with Asterisk for this installation go with Panasonic. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- 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 ___ --
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Steve, I got to congratulate you on this one, very nicely written and you make a lot of sense. However to the OP my advice: As Steve has mentioned in his email so learn it prior to the demo and you have indicated as well: In some point we must start this new tech. The ideal way would be to first run it in test/dev mode in your own office or by an office you can take chances, not by a customer where you can sell them a system (in this case a Panasonic) that you know will do what they want and they'll be happy. Asterisk is not a hardware based system where features work just because they are advertised. If this customer happens to be willing to work with you on the chances then go for it, but if all they want is a system that works, then I suggest wait around until you have tested an asterisk system that you like for the solution you are trying to provide. On Thu, Oct 16, 2008 at 9:52 AM, Steve Totaro [EMAIL PROTECTED] wrote: If you want to wow them with GUI stuff (and make it easy for you, since the settings are generally correct out of the box) then download and install EVB (Easy Box Box). Another FreePBX/Asterisk based GUI with Webmin and lots of other good programs pre-installed. I would not use it for a very large install, at least without moving the DB (at the least) to separate box. To WOW them, open a web browser to EVB and and SSH window connected to the Asterisk console with a good amount of debugging. I don't know how meeting are done there, but have a meeting room with a projector, and MIC that is always better than five guys crowding around a computer screen. Presentation is key. The simplicity of the GUI coupled with the complexity of the CLI with verbose and sip debug turned on, will leave a lasting impression. Show them the ease an power of FreePBX (so learn it prior to the demo) so you feel confident and can answer questions on the fly. Show them the mouse over and even let the least computer savvy person create and activate a Holiday greeting/IVR. I find that driving the whole time is not so good, let everyone sit down and do something, add an extension, make an IVR, setup routing, followme, call screening, or whatever. Show them backup scheduling and Samba, so you can pull or push down backups on a regular schedule. EVB has quite a bit more than FreePBX and is much better than Trixbox as far as I am concerned. There is one add in EVB, but you can remove that with a little know how. You can also add your company's log which is always a nice touch. If someone has an Iphone, set them up as a user and show them that their message will show up in their email and they can play the attachment. There is plenty that Asterisk cannot do compared to a small key system, but I think that Asterisk more than makes up for those short comings with other features. If you get a firm understanding of it, you can show them how easy it is to create an IVR, Add an Extension, FollowMe, Call Screening, but base your presentation on what the Panasonic can do, then what Asterisk (FreePBX) can do. Focus on the positives, be confident, and if someone throws you a curve ball, don't fumble around for an answer. Do a quick Googe, or simply say, I don't know but I will find out. There is nothing wrong with not knowing something, but fumbling around or making up an answer leaves a bad taste. I respect I don't know, but I will get you an answer. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Oct 15, 2008 at 9:03 PM, C F [EMAIL PROTECTED] wrote: On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza [EMAIL PROTECTED] wrote: Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) Not really so. Depending on lots of factors, usually for a small office of only 5-10 users Panasonic is more feature rich. Since the main feature they are looking for in a PBX is to be able to yell across the hallway; hey boss call on 5 it's your wife which is not really possible with Asterisk (yeah I know call parking, but how many phones support it flawlessly with flashing LEDs?). Other features that are quite popular in small offices and not supported by Asterisk: * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. ** Login/Logout of queues,
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Being a Panasonic dealer and having more than 50 Asterisk system in production, I can tell you that if this is your first Asterisk project, then go with Panasonic, you'll safe yourself lots of aggravation and have a happier customer. Some features of the Panasonic you will never be able to emulate on Asterisk. While depending on the needs of that customer, and in some cases I would suggest dive into Asterisk, I gather from the subject (yes I have read the whole message, for those of you out there that might think that I did not) that a Panasonic will work nicely for them, therefore my advice stick with Panasonic. On Mon, Oct 13, 2008 at 9:15 PM, Rodolfo Alcazar Portillo [EMAIL PROTECTED] wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. * I need to get a formatted calls report for the administrators to charge the users. I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza [EMAIL PROTECTED] wrote: Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) Not really so. Depending on lots of factors, usually for a small office of only 5-10 users Panasonic is more feature rich. Since the main feature they are looking for in a PBX is to be able to yell across the hallway; hey boss call on 5 it's your wife which is not really possible with Asterisk (yeah I know call parking, but how many phones support it flawlessly with flashing LEDs?). Other features that are quite popular in small offices and not supported by Asterisk: * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 has this as well). ** Company internal directory on the phone updated on the PBX ** System Speed Dial on the display updated by the PBX ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). ** On screen Voicemail (on the phone). ** Line assignment to buttons with LED indication, and hold indication. ** Hold ringback (some IP phones support it). There are many more features but I can't remember them at the moment. Granted in bigger installations there many more factors and usually more funding which makes the above list almost obsolete for the features that Asterisk does have. Again my advice do not go with Asterisk for this installation go with Panasonic. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- 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
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Am Montag, den 13.10.2008, 22:54 -0500 schrieb Jorge Mendoza: Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) Yes, that's what I need help for. ... when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. Ok. I will try those. Now, my idea of dial plan: 1xx -- office extensions 92xx -- local city calls, which must be free 90800xx. -- national 0800 calls, which must be free 4___#9xxx. other calls, which must be paid: cell, national, international. the three ___ are each user account code. Is that ok? I don't know how I'll extract the 4___ calls account code, but that's the idea. Thanks a lot! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- 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] Panasonic x Asterisk if I can emulate Panasonic fast!
On Tuesday 14 October 2008 08:29:18 Rodolfo Alcazar Portillo wrote:
1xx -- office extensions
92xx -- local city calls, which must be free
90800xx. -- national 0800 calls, which must be free
4___#9xxx. other calls, which must be paid: cell, national,
international. the three ___ are each user account code.
Is that ok? I don't know how I'll extract the 4___ calls account code,
but that's the idea.
Set(CDR(accountcode)=${EXTEN:1:3})
Dial(Zap/g0/${EXTEN:6})
--
Tilghman
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[asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. * I need to get a formatted calls report for the administrators to charge the users. I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- 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] Panasonic x Asterisk if I can emulate Panasonic fast!
Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- 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
