Re: [asterisk-users] Bounty on Google Voice
On Wed, Mar 29, 2017 at 11:45 AM, Saint Michael wrote: > The channel motif and res_xmpp do not work. But there is one company that > does make it work and charges $US 6 for a lifetime connection to your own > free Google Voice number, from SIP. I wonder if anybody would be able to fix > Asterisks libraries so people of low income would not have to pay a third > party for this basic translation service. Hey, Sorry to hear about your difficulty with this code. At this time, I'm not aware of an active maintainer/owner of those modules. As I'm sure you're already aware, Asterisk is an open source project worked on by people with different interests and motivations. Some work on it purely for the fun, others work on it because they have a business interest of some sort. Bug bounties are a great way to get the attention of people in both categories, and are usually posted to the asterisk-dev mailing list. You can learn more about the guidelines prior to posting at this wiki page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties Hope that helps a bit, and best wishes in getting your bug fixed. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
On 29/03/17 16:18, Olivier wrote: Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complies with it ? Debian's versioning scheme is all their own. And I would not expect it to work with anything but a Debian-packaged Asterisk. Stretch is currently the "testing" distribution. This means that new versions of packages could appear at any time; but if a newly-introduced package breaks any other packages, they will be removed from "testing" (and replaced as soon as possible with newer, compatible versions) rather than allow packages to exist in the repository that cannot be co-installed. If you really want to use a newer Asterisk version, the Debian source will contain a file called "rules", which is really a Makefile "in disguise". This should give you a good clue as to how to hand-build an equivalent based on more up-to-date Source Code (if the compile-time options have not changed too much, then you might even get away with using it directly, but consider this a bodge). 2. From package description, is this package enough or not to allow transcoding with G711 ? For instance, in the following situation: SIP Phone < Opus > Asterisk < G111 ---> ITSP All codecs can input and output raw, uncompressed PCM; so as long as you build all the relevant modules, your Asterisk will be able to transcode between any two codecs it supports. (Is "G111" a typo for "G711" ?) 3. Can you share here any personal field experience with this codec, for home worker use case ? Is there a better user experience with Opus than with G729 or G711 ? Opus is, to the best of my knowledge, fully Open Source. G729 was encumbered by patents in some jurisdictions, though it's now patent-free. G.711 A-Law is what the PSTN uses natively, and that is unlikely to change anytime soon; though some VoIP providers are bringing Opus online already. If you have many phones connected to your Asterisk, then you may run into CPU limitations transcoding incoming and outgoing calls between G711 and Opus. But that depends on your Asterisk server. If you are recording calls, Asterisk will already have to convert both the incoming and outgoing legs to raw PCM anyway. In any case, if your provider supports Opus, you can offload the donkey work to them . 4. Does it work on ARM boxes (Raspberry, ...) ? The only thing that would prevent any software from working on ARM / Raspberry Pi would be if it contained any architecture-specific binary code without Source Code (which you could just about get away with, if you released it under LGPL plus exceptions or an Apache licence). And I suspect if any such code existed, it would be rewritten in fairly short order anyway. Also, it's Debian; and they really, really don't like binary blobs, only grudgingly banishing them to a special "non-free" section which is not even enabled by default. And that package was in the main repository, suggesting full Source Code availability. In any case, I see builds for armhf (R.Pi 1 and 2) and arm64 (R.Pi 3); so even if there is some sneaky binary-only component, you will be able to get it to work. [1] https://packages.debian.org/stretch/asterisk-opus Regards -- JM or AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bounty on Google Voice
The channel motif and res_xmpp do not work. But there is one company that does make it work and charges $US 6 for a lifetime connection to your own free Google Voice number, from SIP. I wonder if anybody would be able to fix Asterisks libraries so people of low income would not have to pay a third party for this basic translation service. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
On Wed, Mar 29, 2017 at 06:04:49PM +0200, Olivier wrote: > Is there any relation between this external patch and the binary mentioned > in [2] > [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/ > > The later one mentions a binary-only distribution to comply with legal > constraints. No, it is not. This package is in Debian's main archive, which tells you it is not based on any binary blob. Opus is widely implemented in software, including free software (Firefox, Chromium, Linphone, Jitsi and a host of others). See also https://en.wikipedia.org/wiki/Opus_(audio_format)#Software My understanding is that to Digium's best legal advice, there are still patent issues with the Opus codec. Even though many others disagree (as evident from above) and I also happen to disagree. But I certainly am not the one who runs Digium. And the powers that be there probably decided that whatever patent issues there are, have merit and need to be mitigated. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge function slight change from 11 to 13
On Wed, Mar 29, 2017 at 9:40 AM, Michaël Gaudette wrote: > Hi, > > > > I have been using ConfBridge since Asterisk 11, and I recently upgraded a > server to 13. While everything that needed fixing seems fixed, I have an > issue that does not seem documented anywhere. > > > > The way I used ConfBridge is that I have a standard bridge profile, user > profile and menu that (almost) everyone uses. I call the ConfBridge this > way: > > > > exten => s,1,Confbridge(some_id,bridge_basic,user_basic,admin_menu_basic) > > > > But, even though everyone uses the same basic config, each conference has > a different NIP to get it. So what I USED to do is this: > > > > exten => s,1,Set(CONFBRIDGE(user,pin)=123456) > > exten => s,2,Confbridge(some_id,bridge_basic,user_basic,admin_menu_basic) > > > > This, as far as I could tell on Asterisk 11, meant that the user_basic > profile was used, but whatever default PIN present in confbridge.conf was > “overwritten” by my on-the-fly to the ConfBridge(user,pin) value > > > > Now, on Asterisk 13.14 (I don’t have an Asterisk 12 I can play with) it > seems that the fact that I am referencing the user_basic profile in my call > to the ConfBridge app means that whatever PIN value I put in my dial plan > is ignore, since a user profile is present. While, before, the PIN value > overwrote the profile value by having that defined in my dialplan. > > > > · TBH this seems the legit way to use ConfBridge The wiki says: > > · > > · “user_profile - The user profile name from confbridge.conf. > When left blank, a dynamically built user profile created by the CONFBRIDGE > dialplan function is searched for on the channel and used. If no dynamic > profile is present, the 'default_user' profile found in confbridge.conf is > used” > > · > > · …but it suited me better to use it the way it worked in > Asterisk 11. > > Is there any way to make the channel values overwrite the profile values > instead of be ignored by the presence of a profile in the dialplan > application parameters? Or something that has a similar effect, i.e. an > easy to change overall default bridge user that can be slightly modified > through the dialplan? > You have to explicitly create your dynamic user profile based upon the desired user profile first. Otherwise, it uses the "default_user" profile as the basis of the dynamic profile. ; Use a non-default user profile as the basis of the dynamic user profile exten = ,1,NoOp() same = n,Set(CONFBRIDGE(user,template)=user_basic) same = n,Set(CONFBRIDGE(user,pin)=123456) same = n,ConfBridge(some_id,bridge_basic,,admin_menu_basic) ; Use the default "default_user" profile as the basis of the dynamic user profile exten = ,1,NoOp() same = n,Set(CONFBRIDGE(user,pin)=123456) same = n,ConfBridge(some_id,bridge_basic,,admin_menu_basic) If you explicitly specify a bridge, user, or menu profile to ConfBridge then that is what gets used regardless of any dynamic profile you created. See the online documentation: core show application ConfBridge or https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_ConfBridge config show help app_confbridge user_profile template or https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
2017-03-29 17:28 GMT+02:00 Tzafrir Cohen : > On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote: > > Hello, > > > > After reading [1] (in french), I would be very happy if I could get > answers > > to: > > > > 1. Does this 13.7+20161113-3 package version has any relation with > > asterisk's version it complements ? Current asterisk version in repo is > > 13.14.0. Does this 13.7 complies with it ? > > The opus codec was used as an external patch. It looked ugly and thus a > separate package was preffered. > Thank you very much for this informative answer. Is there any relation between this external patch and the binary mentioned in [2] [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/ The later one mentions a binary-only distribution to comply with legal constraints. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]
Basically, in some countries, regulation says: "caller MUST know in advance what call would given the number (s)he dialed". If regulation also states that waiting time shall not charged or shall not charged more than a local/national/whatever call, then an alternative is to separate telecommunication costs from service costs the former being billed by caller's telco based on whole duration, the later being billed byservice provider based on effective duration (whole duration minus waiting time). Of course, this requires that service providers are able to directly bill callers which, certainly, can't be easy or even possible in all situations. Some service providers require callers to enter a personal PIN code, before joining a queue. That could help to bind a caller to a paying customer and let the later one being charged for the services used by the caller. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote: > Hello, > > After reading [1] (in french), I would be very happy if I could get answers > to: > > 1. Does this 13.7+20161113-3 package version has any relation with > asterisk's version it complements ? Current asterisk version in repo is > 13.14.0. Does this 13.7 complies with it ? The opus codec was used as an external patch. It looked ugly and thus a separate package was preffered. Its version number is not directly related to Asterisk. It has originally been split from the Debian packaging of Asterisk, and starting from the same version number allowed easier upgrading. There is no version number for the upstream code (the patch). > > 2. From package description, is this package enough or not to allow > transcoding with G711 ? > For instance, in the following situation: > SIP Phone < Opus > Asterisk < G111 ---> ITSP Technically Asterisk codecs translate to/from (typically) linear and Asterisk combines codecs to do whatever transcoding needed. So the codec does not transcode directly to G.711. But Asterisk can transcode between opus and G.711. > > 3. Can you share here any personal field experience with this codec, for > home worker use case ? > Is there a better user experience with Opus than with G729 or G711 ? > > 4. Does it work on ARM boxes (Raspberry, ...) ? Should work just the same. > > > [1] https://packages.debian.org/stretch/asterisk-opus -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general
Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complies with it ? 2. From package description, is this package enough or not to allow transcoding with G711 ? For instance, in the following situation: SIP Phone < Opus > Asterisk < G111 ---> ITSP 3. Can you share here any personal field experience with this codec, for home worker use case ? Is there a better user experience with Opus than with G729 or G711 ? 4. Does it work on ARM boxes (Raspberry, ...) ? [1] https://packages.debian.org/stretch/asterisk-opus Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge function slight change from 11 to 13
Hi, I have been using ConfBridge since Asterisk 11, and I recently upgraded a server to 13. While everything that needed fixing seems fixed, I have an issue that does not seem documented anywhere. The way I used ConfBridge is that I have a standard bridge profile, user profile and menu that (almost) everyone uses. I call the ConfBridge this way: exten => s,1,Confbridge(some_id,bridge_basic,user_basic,admin_menu_basic) But, even though everyone uses the same basic config, each conference has a different NIP to get it. So what I USED to do is this: exten => s,1,Set(CONFBRIDGE(user,pin)=123456) exten => s,2,Confbridge(some_id,bridge_basic,user_basic,admin_menu_basic) This, as far as I could tell on Asterisk 11, meant that the user_basic profile was used, but whatever default PIN present in confbridge.conf was overwritten by my on-the-fly to the ConfBridge(user,pin) value Now, on Asterisk 13.14 (I dont have an Asterisk 12 I can play with) it seems that the fact that I am referencing the user_basic profile in my call to the ConfBridge app means that whatever PIN value I put in my dial plan is ignore, since a user profile is present. While, before, the PIN value overwrote the profile value by having that defined in my dialplan. · TBH this seems the legit way to use ConfBridge The wiki says: · · user_profile - The user profile name from confbridge.conf. When left blank, a dynamically built user profile created by the CONFBRIDGE dialplan function is searched for on the channel and used. If no dynamic profile is present, the 'default_user' profile found in confbridge.conf is used · · but it suited me better to use it the way it worked in Asterisk 11. Is there any way to make the channel values overwrite the profile values instead of be ignored by the presence of a profile in the dialplan application parameters? Or something that has a similar effect, i.e. an easy to change overall default bridge user that can be slightly modified through the dialplan? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]
In most instances the company being called is not charging the caller for their phone serves. That is the callers service provider, and once the answer is issued the call is up. This only makes senses if the company being called is providing services and charging a per min rate for that service. They would not charge the customer for the hold time waiting for a rep to come on the line. This could all be done by creating billing records from cel logs. These can log events such as channel start and answer by an extension, transfers and hangups. As Samy Go stated a good way to reduce charges to the caller would be to offer call back options. So when a rep is available the system would call the original caller back. Telecom networks around the world are just not designed to offer delayed billing. Legislating that requirement would require world wide overhauls of the networks as well as treaties. In some areas you also have to pay ring time. That is a novel idea to actually pay for a resource you are using when you use it. That is a little too capitalistic for some. Bryant From: "SamyGo" Sent: Wednesday, March 29, 2017 9:52 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED] Hi, Just trying to figure out how is this solved ? by involving multiple telcos in the loop and asking them to not charge based on 200 OK/Answer!? As far as I know people have designed Queue/CallCenter platforms who upon entering a number in queue just state them their number in queue and approx time before they'll be contacted and drop the call. This all can be done within Progress. As soon as their turn comes the CallCenter platform automatically triggers the call to them and get them connected with an agent. This is the way I can understand as nobody waiting in the queue but people in the waiting-list. Since Queue has to "answer" the call first before doing anything once the signal to Answer is triggered technically that marks the start of billing for everyone. Regards, Sammy On Wed, Mar 29, 2017 at 7:25 AM, Olivier wrote: Thank you very much, Max, for this valuable and informative answer. Offline billing must be quite complex to set up as several telco may be involved (or origination,transit or termination). Moving to normal landline fare seems much simpler ! Thanks again2017-03-28 21:41 GMT+02:00 Max Grobecker : Hi, in Germany, this kind of regulation is in effect for phone numbers which cost more than a normal landline call. The regulation states, that the waiting time must not be charged to the customer. Most companies implemented this by simply switching their telephone numbers to those, which are charged per call (so there's no difference in price between waiting for someone to pick up or being connected to someone) ;-) Or they decided to use a normal landline phone number for which this regulation does not apply. The second method was to not answer the call before really connected to a person on the queue and using Early Media as you mentioned. But: The maximum length of this Early Media stream is in most telephone networks limited to somewhat around 90 to 180 seconds, then the call gets disconnected by the network. I'm not very familiar with regulations and numbering plans in France, but maybe there's also something called "offline billing". Using this, your call is not billed by the caller's telephone company until you send them the amount of time that should be billed for a specific call. Your best choice will be, that - if you ever get those regulations - you should rely on what your telephone number provider tells you to do ;-) Greetings Max Am 28.03.2017 um 15:24 schrieb Olivier: > Hello, > > In France, years ago, there was some discussions about a new regulation forcing some providers to not charge anything to callers while those are waiting for a call center agent to become available. > Once caller and agent are on call with each other, nominal charging applies. > > No matter if those discussions ever did or didn't change current regulation, I wonder which dialplan statements could technically comply this dual billing requirement ? > > > same = n,Progress() > same = n,Queue(whatever,...,macro-option, ...) > > To me, coupling Progress app with Queue's macro or gosub option like above, would let a sysadmin answer a queued call. > Doing so, time spent before connection with queue agent should not be billed to anyone (caller nor callee), while time spent after connection is billed normaly. > > 1. Should this work ? Am I missing something ? > > 2. Is there an alternative way to implement this ? > > 3. Comments ? Suggestions ? > > Regards > > -- _ -- Bandwidth and Colocation Provided
Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]
Hi, Just trying to figure out how is this solved ? by involving multiple telcos in the loop and asking them to not charge based on 200 OK/Answer!? As far as I know people have designed Queue/CallCenter platforms who upon entering a number in queue just state them their number in queue and approx time before they'll be contacted and drop the call. This all can be done within Progress. As soon as their turn comes the CallCenter platform automatically triggers the call to them and get them connected with an agent. This is the way I can understand as nobody waiting in the queue but people in the waiting-list. Since Queue has to "answer" the call first before doing anything once the signal to Answer is triggered technically that marks the start of billing for everyone. Regards, Sammy On Wed, Mar 29, 2017 at 7:25 AM, Olivier wrote: > Thank you very much, Max, for this valuable and informative answer. > > Offline billing must be quite complex to set up as several telco may be > involved (or origination,transit or termination). > Moving to normal landline fare seems much simpler ! > > Thanks again > > 2017-03-28 21:41 GMT+02:00 Max Grobecker > : > >> Hi, >> >> in Germany, this kind of regulation is in effect for phone numbers which >> cost more than a normal landline call. >> The regulation states, that the waiting time must not be charged to the >> customer. >> >> >> Most companies implemented this by simply switching their telephone >> numbers to those, which are charged per call >> (so there's no difference in price between waiting for someone to pick up >> or being connected to someone) ;-) >> Or they decided to use a normal landline phone number for which this >> regulation does not apply. >> >> The second method was to not answer the call before really connected to a >> person on the queue and using Early Media as you mentioned. >> But: The maximum length of this Early Media stream is in most telephone >> networks limited to somewhat around 90 to 180 seconds, >> then the call gets disconnected by the network. >> >> I'm not very familiar with regulations and numbering plans in France, but >> maybe there's also something called "offline billing". >> Using this, your call is not billed by the caller's telephone company >> until you send them the amount of time that should be billed for a specific >> call. >> >> >> Your best choice will be, that - if you ever get those regulations - you >> should rely on what your telephone number provider tells you to do ;-) >> >> >> Greetings >> Max >> >> >> Am 28.03.2017 um 15:24 schrieb Olivier: >> > Hello, >> > >> > In France, years ago, there was some discussions about a new regulation >> forcing some providers to not charge anything to callers while those are >> waiting for a call center agent to become available. >> > Once caller and agent are on call with each other, nominal charging >> applies. >> > >> > No matter if those discussions ever did or didn't change current >> regulation, I wonder which dialplan statements could technically comply >> this dual billing requirement ? >> > >> > >> > same = n,Progress() >> > same = n,Queue(whatever,...,macro-option, ...) >> > >> > To me, coupling Progress app with Queue's macro or gosub option like >> above, would let a sysadmin answer a queued call. >> > Doing so, time spent before connection with queue agent should not be >> billed to anyone (caller nor callee), while time spent after connection is >> billed normaly. >> > >> > 1. Should this work ? Am I missing something ? >> > >> > 2. Is there an alternative way to implement this ? >> > >> > 3. Comments ? Suggestions ? >> > >> > Regards >> > >> > >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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 -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update option
Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]
Thank you very much, Max, for this valuable and informative answer. Offline billing must be quite complex to set up as several telco may be involved (or origination,transit or termination). Moving to normal landline fare seems much simpler ! Thanks again 2017-03-28 21:41 GMT+02:00 Max Grobecker : > Hi, > > in Germany, this kind of regulation is in effect for phone numbers which > cost more than a normal landline call. > The regulation states, that the waiting time must not be charged to the > customer. > > > Most companies implemented this by simply switching their telephone > numbers to those, which are charged per call > (so there's no difference in price between waiting for someone to pick up > or being connected to someone) ;-) > Or they decided to use a normal landline phone number for which this > regulation does not apply. > > The second method was to not answer the call before really connected to a > person on the queue and using Early Media as you mentioned. > But: The maximum length of this Early Media stream is in most telephone > networks limited to somewhat around 90 to 180 seconds, > then the call gets disconnected by the network. > > I'm not very familiar with regulations and numbering plans in France, but > maybe there's also something called "offline billing". > Using this, your call is not billed by the caller's telephone company > until you send them the amount of time that should be billed for a specific > call. > > > Your best choice will be, that - if you ever get those regulations - you > should rely on what your telephone number provider tells you to do ;-) > > > Greetings > Max > > > Am 28.03.2017 um 15:24 schrieb Olivier: > > Hello, > > > > In France, years ago, there was some discussions about a new regulation > forcing some providers to not charge anything to callers while those are > waiting for a call center agent to become available. > > Once caller and agent are on call with each other, nominal charging > applies. > > > > No matter if those discussions ever did or didn't change current > regulation, I wonder which dialplan statements could technically comply > this dual billing requirement ? > > > > > > same = n,Progress() > > same = n,Queue(whatever,...,macro-option, ...) > > > > To me, coupling Progress app with Queue's macro or gosub option like > above, would let a sysadmin answer a queued call. > > Doing so, time spent before connection with queue agent should not be > billed to anyone (caller nor callee), while time spent after connection is > billed normaly. > > > > 1. Should this work ? Am I missing something ? > > > > 2. Is there an alternative way to implement this ? > > > > 3. Comments ? Suggestions ? > > > > Regards > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
