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| 06:14.40 | aum | the asterisk.org website displays binary rubbish, I've tried several browsers, same shit |
| 06:15.20 | ChannelZ | apparently it is broken. |
| 06:15.43 | ChannelZ | try asterisk.org/download and navigate from there |
| 06:16.04 | aum | it happened just after I went to display the intro video, and clicked 'cancel' to the signup (I have already signed up) |
| 06:16.27 | aum | same problem - same binary stuff with asterisk.org/download |
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| 06:16.36 | ChannelZ | looks like Drupal fell on its ass |
| 06:16.42 | ChannelZ | hmm weird /download works for me |
| 06:16.56 | aum | for me, /download redirects to /downloads |
| 06:17.09 | ChannelZ | well yes |
| 06:17.19 | ChannelZ | but I still get a mostly normal site |
| 06:17.26 | aum | and /downloads is binary rubbish |
| 06:17.41 | aum | ChannelZ, are you displaying from cache? |
| 06:18.26 | ChannelZ | hmm no I just reloaded and it's fine. |
| 06:18.48 | ChannelZ | Looks like maybe the webserver is screwed, it's spitting out gzipped content but not marking it as such. |
| 06:21.58 | ChannelZ | Parts of the site work, parts don't. |
| 06:23.08 | aum | i have a job interview in 4 days, and need to learn asterisk by then. Luckily I found a path to the video via digium |
| 06:24.49 | ChannelZ | mmyeah..good luck with that.. |
| 06:27.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 06:29.57 | ChannelZ | Oooo and now it's fixeded. |
| 06:30.19 | ChannelZ | Either that or the site is load balanced and this is a different server than before. |
| 06:32.55 | aum | still garbled for me |
| 06:35.38 | ChannelZ | yeah back to that now. |
| 06:35.42 | ChannelZ | I would say they are Having Issues. |
| 06:35.47 | aum | TM |
| 06:36.10 | aum | do the web staff know? |
| 06:43.23 | ChannelZ | at 1am on a Sunday? probably not |
| 06:43.43 | aum | wtf? back up :) |
| 06:45.16 | aum | nice work guys |
| 06:45.42 | ChannelZ | it's random. get-started is still broken here. |
| 06:45.45 | aum | is downloading asterisknow, with the aim of setting up a proof of concept pbx server in time for interview |
| 06:45.55 | ChannelZ | community is broken |
| 06:46.00 | aum | needs to get quick exposure to the python api |
| 06:46.14 | ChannelZ | It's probably a corrupt Drupal cache or something |
| 06:47.13 | aum | is not the greatest fan of drupal |
| 06:47.32 | ChannelZ | me either, it's kind of a bloated mess |
| 06:47.43 | aum | finds it less than endearing how drupal breaks websites with every major release |
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| 13:57.28 | michael_work | so i'll be annoying and i'll ask : is it me or asterisk.org is ill :) |
| 14:17.22 | jameswf | Neat |
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| 15:01.37 | michael_work | and it's back |
| 15:01.38 | michael_work | :) |
| 15:02.59 | michael_work | here is our request to reviewboard of the fix of asterisk not always honor the ringinuse=no option https://reviewboard.asterisk.org/r/3409/ |
| 15:44.12 | pabelanger | michael_work, yup, something is fubar'd |
| 15:44.17 | pabelanger | file, ^ |
| 15:44.35 | michael_work | pabelanger, ? |
| 15:44.42 | pabelanger | asterisk.org |
| 15:44.54 | michael_work | it looks fine last time i checked |
| 15:45.00 | michael_work | i mean it got back |
| 15:45.07 | michael_work | ha |
| 15:45.13 | michael_work | it's gonne crazy again :) |
| 15:45.44 | michael_work | btw it's not only main but other pages too |
| 15:46.14 | pabelanger | yup |
| 15:46.22 | michael_work | http://www.asterisk.org/products for example |
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| 17:56.08 | JamesJRH | pabelanger: Do you know if ringing consumes a line? |
| 17:56.35 | JamesJRH | PSTN that is. |
| 17:56.51 | JamesJRH | (Or anyone else if you know.) |
| 17:57.18 | pabelanger | JamesJRH, what do you mean? |
| 17:57.41 | WIMPy | doesn't understand the question, either. |
| 17:57.56 | [TK]D-Fender | My ears ring all the time... |
| 17:58.00 | JamesJRH | pabelanger: For example, if I only have one line, can I ring out to say 5 phones, but as soon as the first one picks up, hang-up on the other 4? |
| 17:58.00 | [TK]D-Fender | #tinnitus |
| 17:58.06 | JamesJRH | :-) |
| 17:58.19 | WIMPy | no |
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| 17:58.42 | [TK]D-Fender | [13:58]JamesJRHpabelanger: For example, if I only have one line, can I ring out to say 5 phones, but as soon as the first one picks up, hang-up on the other 4? <- you are dangerously vague here |
| 17:58.49 | [TK]D-Fender | This could either a yes, or a no. |
| 17:58.51 | JamesJRH | WIMPy: It was actually out of context because it was a leftover question from the other day. |
| 17:59.02 | [TK]D-Fender | Never talk about lines & phones generically. |
| 17:59.31 | [TK]D-Fender | Be specific about the interfaces everything is one |
| 17:59.34 | [TK]D-Fender | on* |
| 18:04.17 | JamesJRH | [TK]D-Fender: pabelanger told me that one can use an ‘ITSP’, and connect via a trunk, in order to interface with the PSTN. The trunk will have a number of ‘lines’ or ‘channels’ that, as long as the server running Asterisk has enough resources, will be able to carry upto this number of calls simultaneously. |
| 18:04.44 | [TK]D-Fender | JamesJRH: Well your phones are independent of that... |
| 18:04.54 | JamesJRH | I was wondering what happens during the ringing stage, before the call is connected. |
| 18:05.05 | [TK]D-Fender | james you could have 1 call coming in and ring 100 SIP phones if you want... |
| 18:05.15 | [TK]D-Fender | while that line is ringing.. that is taking up 1 channel to your provider. |
| 18:05.16 | JamesJRH | Oh.. |
| 18:05.31 | [TK]D-Fender | When 1 answers, the rest "hangup" to the PBX automatically |
| 18:05.32 | JamesJRH | I'm talking about ringing PSTN phones. |
| 18:05.48 | JamesJRH | Mobiles, landlines. |
| 18:06.01 | JamesJRH | (Sorry.) |
| 18:06.13 | [TK]D-Fender | Fix your description of what you are doing then |
| 18:06.28 | [TK]D-Fender | You want to dial 5 different PSTN numbers via your provider simultaneously? |
| 18:07.10 | JamesJRH | For example, if I only have one channel, can I ring out to say 5 PSTN phones over the ITSP's trunk, but as soon as the first one picks up, hang-up on the other 4? |
| 18:07.36 | JamesJRH | Is that clearer? |
| 18:08.02 | JamesJRH | is not competent in this area. |
| 18:08.23 | WIMPy | no |
| 18:08.34 | WIMPy | One call, one channel. |
| 18:08.58 | [TK]D-Fender | 5 calls out = 5 channels |
| 18:09.04 | WIMPy | The only exception is receiving call on ISDN. That's possible without channel. |
| 18:09.17 | [TK]D-Fender | You don't get to tell them "multi-dial this out on YOUR side.. but to me it's 1 until someone answers" |
| 18:09.29 | JamesJRH | WIMPy: So does a call that isn't yet connected still require a channel? |
| 18:09.38 | WIMPy | yes |
| 18:09.45 | [TK]D-Fender | it is a call.. from you to the provider |
| 18:09.56 | [TK]D-Fender | forget about the FAR end. |
| 18:09.56 | JamesJRH | Okay. |
| 18:09.57 | WIMPy | In all cases but the mentioned one. |
| 18:10.01 | [TK]D-Fender | that doesn't matter |
| 18:15.37 | JamesJRH | So even while it's ringing, the channel is transceiving audio with the provider, right? Consuming resources and utilising the channel. Am I right in saying that this is different to VoIP because the ringing sound is produced by the VoIP client until the call is established? |
| 18:15.59 | WIMPy | Not neccessarily, but that's not what is counted. |
| 18:16.15 | WIMPy | That would only affect the needed bandwidth. |
| 18:16.46 | JamesJRH | WIMPy: For PSTN or VoIP? |
| 18:17.03 | WIMPy | voip |
| 18:17.20 | WIMPy | On PSTN you transfer audio as soon as you start a call. |
| 18:18.10 | WIMPy | For the called party the channel would only be reserved, but not used until answer. |
| 18:18.37 | [TK]D-Fender | JamesJRH: your provider is using a channel to the PSTN to place your call for you... your "VoIP" end to the provider isn't the issue. Youa re paying for yoru provider to hand you a channel to the outside at all |
| 18:18.49 | [TK]D-Fender | VoIP is not magic |
| 18:19.19 | [TK]D-Fender | They have a limited number of channel they can terminate, and that is what you are paying for. |
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| 18:24.53 | JamesJRH | [TK]D-Fender: Oh, I wasn't refering to the VoIP trunk, I was just comparing PSTN to VoIP in general. That in PSTN, the exchange transmits the ringing sound back to the caller, consuming resources, whereas in VoIP, the ringing is only signalled to the caller, consuming minimal resources until the call is connected. |
| 18:25.13 | JamesJRH | Have I understood this correctly for outgoing calls? |
| 18:25.50 | WIMPy | With VOIP it can be both ways. |
| 18:26.07 | JamesJRH | Though when someone calls in to you, on PSTN, surely no audio is transmitted. |
| 18:26.16 | JamesJRH | Surely that's signalling. |
| 18:26.20 | [TK]D-Fender | no |
| 18:26.32 | [TK]D-Fender | To accepts the call you have to RESERVE the channel |
| 18:26.41 | JamesJRH | Okay. |
| 18:26.56 | [TK]D-Fender | Do you think you can say "sure, I'll look around" and then come up short when you don't have the resources to support the audio? |
| 18:27.01 | WIMPy | That doesn't make sense. |
| 18:27.05 | [TK]D-Fender | No empty sales here |
| 18:27.19 | [TK]D-Fender | Call comes... channel has to be freee to arrive |
| 18:27.33 | WIMPy | You don't need a channel to get a call. |
| 18:27.39 | JamesJRH | Oh, yes, I see.. |
| 18:27.45 | [TK]D-Fender | to carry it you do. |
| 18:27.49 | WIMPy | You need a channel to place a call or to accept an incomming call. |
| 18:28.53 | JamesJRH | ..Because I guess there's no way to say that only one of multiple calls will be picked-up. Right? |
| 18:29.07 | WIMPy | And off course you cann free a channel by putting a call on hold. |
| 18:29.13 | [TK]D-Fender | JamesJRH: You are still bneing vague as to how they are placed.. |
| 18:29.31 | WIMPy | Not on outgoing. On incomming it would work. |
| 18:29.50 | JamesJRH | Okay, so I understand outgoing, but for incoming.. |
| 18:30.12 | [TK]D-Fender | [14:28]JamesJRH..Because I guess there's no way to say that only one of multiple calls will be picked-up. Right? <- you can DIAL multiple different calls out.. each taking up 1 channel until one answers,.. and the others get cancelled when 1 answers |
| 18:30.15 | WIMPy | "call waiting" |
| 18:30.19 | [TK]D-Fender | "core show application dial" <- |
| 18:31.32 | [TK]D-Fender | Dial(SIP/provider/12345&SIP/provider/67890,20) |
| 18:32.06 | [TK]D-Fender | dials 2 target #'s on the PSTN via [provider]. When 1 answers, the other ones gets hung up on and freed |
| 18:34.27 | JamesJRH | ..Incoming example: If I only have one channel, can say 5 PSTN phones call me over the ITSP's trunk, but I only pick up the first one and leave the other 4 to wait? (On Android, I can be in a call, and be aware that someone is calling. Then again I do not know that there is only one channel.) |
| 18:34.37 | [TK]D-Fender | no. |
| 18:34.41 | [TK]D-Fender | You hAVE ONE channel to the provider |
| 18:35.19 | WIMPy | Still one call, one channel. On the PSTN it's adifferent story. |
| 18:35.20 | [TK]D-Fender | They refuse the 2nd onward |
| 18:35.34 | [TK]D-Fender | your provider gets the calls... and ditches them, |
| 18:35.42 | [TK]D-Fender | You pay for one channel.. that is all they will pas to you |
| 18:35.55 | [TK]D-Fender | You don't get to see 10 and pick the one you want |
| 18:36.23 | [TK]D-Fender | Whene your limit is reached the surplus gets dropped however they determine |
| 18:36.43 | JamesJRH | So does call waiting like on Android require at least 2 channels? |
| 18:37.05 | WIMPy | Android what? Some VOIP client or GSM or what? |
| 18:37.29 | [TK]D-Fender | Android" is an OS... not a and no, call waiting is a SINGLE channel |
| 18:37.34 | JamesJRH | Mobile calls are PSTN right? |
| 18:37.36 | WIMPy | And off course your ITSP limiting the number of incomming calls seems stupid. |
| 18:37.42 | [TK]D-Fender | because you can't talk to both ends simultaneously. |
| 18:37.53 | WIMPy | That's called PLMN. |
| 18:38.05 | WIMPy | And no, they don't need additional channels. |
| 18:38.08 | [TK]D-Fender | It also represents a single "incursion" into an existing channel |
| 18:38.17 | [TK]D-Fender | You can't have multiple calls attempting to butt in. |
| 18:38.29 | JamesJRH | [TK]D-Fender: I know Android is an OS, but Android supports it, and smartphones with Android have call waiting. |
| 18:38.30 | WIMPy | You could. |
| 18:39.05 | [TK]D-Fender | [14:38]JamesJRH[TK]D-Fender: I know Android is an OS, but Android supports it, and smartphones with Android have call waiting. <- this is a BAD description because you aren't being SPECIFIC as what WHAT you are doing on it. CELL is one thing... VoIP via SIP, etc is a completely different matter |
| 18:39.50 | [TK]D-Fender | when you are talking "cell" say "cell". Androind can run a VoIP client and the rules are NOT the same |
| 18:40.25 | [TK]D-Fender | Anyway, call waiting is a single incursion on a single channel already in progress, and interrupts it and its managed by your telco. |
| 18:40.32 | JamesJRH | PSTN is landlines and mobiles. Mobile call still work in the legacy way, unless you have a VoIP client on your smartphone. |
| 18:40.41 | [TK]D-Fender | This is nothing liek how an ITSP works to send you calls |
| 18:40.58 | JamesJRH | Hmm. |
| 18:41.03 | WIMPy | Only on POTS. On ISDN or GSn it is managed by your terminal. |
| 18:41.32 | WIMPy | And off course you can have multiple calls waiting. |
| 18:46.36 | JamesJRH | So I guess the mobile network has multiple channels to receive from PSTN on some internal system, then when it receives more than one call it signals to the mobile that there is a call waiting, and this signalling has nothing to do with PSTN, i.e. it is GSM-specific. Right? So to implement call waiting on PSTN, you have to have at least 2 channels. There more channels you have, the more calls you could have waiting. |
| 18:47.12 | WIMPy | No. That *IS* PSTN and how it works. |
| 18:47.13 | JamesJRH | s/There more/The more/ |
| 18:47.48 | [TK]D-Fender | Call waiting is tech-specific |
| 18:48.03 | [TK]D-Fender | GSM's handling is one thing, analog's is another. |
| 18:48.15 | WIMPy | Yes. |
| 18:48.16 | [TK]D-Fender | each takes up multilple channels at the provider's side |
| 18:48.23 | [TK]D-Fender | and that is what you are paying for |
| 18:48.32 | WIMPy | For POTS the terminal is the switch, not your phone. |
| 18:48.41 | [TK]D-Fender | You pay for one channel... forget about ever getting that 2nd call |
| 18:48.55 | JamesJRH | infobot? |
| 18:49.12 | JamesJRH | Was my message too long? |
| 18:49.44 | [TK]D-Fender | Bot has some limitations.. we get the idea.. |
| 18:49.54 | JamesJRH | ..i.e. it is GSM-specific. Right? So to implement call waiting on PSTN, you have to have at least 2 channels. There more channels you have, the more calls you could have waiting. |
| 18:50.13 | WIMPy | No. |
| 18:50.18 | JamesJRH | (In case it got cut.) |
| 18:50.34 | WIMPy | The fact that there is no channel is what makes a call a waiting call. |
| 18:53.25 | JamesJRH | WIMPy: Doesn't that contradict: 18:35:19 < [TK]D-Fender> They refuse the 2nd onward |
| 18:53.41 | WIMPy | That is for VOIP. |
| 18:53.42 | JamesJRH | Refuse or wait? |
| 18:53.47 | JamesJRH | Oh. |
| 18:53.54 | [TK]D-Fender | If your cell provider is OFFERING you call waiting on your cell. then you are PAYING for that service |
| 18:54.01 | [TK]D-Fender | When talking about TISP's that is NOT the same |
| 18:54.06 | [TK]D-Fender | You are mixing things up constantly |
| 18:54.12 | WIMPy | Although I don't know why any ITSP would do that. |
| 18:54.41 | WIMPy | Did you ever pay extra for that service? |
| 18:54.47 | [TK]D-Fender | Stop comparing common cell service.. to ITSP VoIP services |
| 18:55.10 | [TK]D-Fender | "Well I get X on cell, why not VoIP" is NOT valid. The tech is not the same, the packages are not the same. |
| 18:55.50 | WIMPy | Well, it makes sense to compare performance when comparing prices. |
| 18:58.18 | [TK]D-Fender | I'm making sure we aren't comparing apples & oranges. |
| 18:58.25 | [TK]D-Fender | Pricing is another thing. |
| 18:59.29 | JamesJRH | [TK]D-Fender: So what was wrong with what I said about the cell network implementing the call waiting requiring at least 2 PSTN channels on some internal system, and then passing that service on using a GSM-specific mechanism, i.e. not PSTN. Have I not got that bit right? There has to be an interface between PSTN and GSM somewhere. |
| 18:59.35 | JamesJRH | [TK]D-Fender: That's not mixing up anymore; I've separated the concepts in my understanding. |
| 19:00.17 | [TK]D-Fender | "There has to be an interface between PSTN and GSM somewhere." <- that is your cell company |
| 19:00.35 | [TK]D-Fender | THEY are the interface that takes the raw PSTN channels and pumps it to your phone the way they do |
| 19:00.36 | JamesJRH | “some internal system”? |
| 19:00.50 | [TK]D-Fender | ^ could you be a bit more vague? |
| 19:01.32 | JamesJRH | Well I don't know the internals of a cell network. |
| 19:01.36 | WIMPy | It's not as if they were really different. GSM just adds the radio layer and authentication and such. But call handling is much the same. |
| 19:01.40 | [TK]D-Fender | Cell company would be taking up 2 channels on your behalf to support call-waiting for you.. and they provide that service... you are also only talking to ONE at a time |
| 19:02.34 | WIMPy | Now you're being vague about channels being used. |
| 19:02.36 | [TK]D-Fender | With VoIP, every channel is independent. I ahve an * server... I ahve have a HUNDRED phones here so that multiple people could handle those calls INDEPENDENTLY. |
| 19:02.38 | [TK]D-Fender | that is what you are paying for. |
| 19:02.59 | [TK]D-Fender | Call-waiting is not a true multi-channel service |
| 19:03.12 | WIMPy | Not at all. |
| 19:03.53 | WIMPy | >>The fact that there is no channel is what makes a call a waiting call. |
| 19:04.03 | JamesJRH | [TK]D-Fender: Is call-waiting GSM-specific? |
| 19:04.17 | WIMPy | No |
| 19:04.29 | [TK]D-Fender | WIMPy: You can signal to flip them... but they aren't simultanous & independent |
| 19:04.38 | [TK]D-Fender | WIMPy: I don't get be on both at once. |
| 19:04.41 | WIMPy | GSM is more ore less just mobile ISDN. |
| 19:05.07 | WIMPy | Yous, you could, by starting a 3-party-call. |
| 19:05.21 | JamesJRH | [TK]D-Fender: That's what I'm asking. |
| 19:07.30 | [TK]D-Fender | WIMPy: ok, on the presise that both are independent streams to your phone, and your phone is doing the mixing, etc., that is still not 2 separate calls like those being handled by separate endpoints. Something to provider is expecting as far as your handling so they can rate you as not being a constant drain on multiple channels |
| 19:07.41 | JamesJRH | The question before I made the cell-network comparison was can I just have one channel, and have 2 *non-simultaneous* calls where one is waiting? |
| 19:08.04 | JamesJRH | Well, 5 actually, but still. |
| 19:08.19 | WIMPy | On PSTN: yes, on VOIP: no. |
| 19:08.55 | WIMPy | At least not with SIP. |
| 19:09.24 | WIMPy | Other protocolls probably can. |
| 19:15.34 | JamesJRH | WIMPy: I didn't even realise that fixed channels are relevant with VoIP, if it isn't being bridged to PSTN (via say an ITSP). I thought VoIP software could have as many calls as the resources can support. |
| 19:15.57 | WIMPy | That would make sen, yes. |
| 19:16.12 | WIMPy | But there seem to be ITSPs out there who like things different. |
| 19:16.56 | WIMPy | sense |
| 19:18.39 | JamesJRH | WIMPy: Okay, so when you say “on VOIP: no”, are you referring to VoIP bridged to PSTN via an ITSP? |
| 19:19.04 | WIMPy | Yes. |
| 19:19.31 | JamesJRH | Oh.. I see the catch.. |
| 19:19.38 | WIMPy | Well, Just via an ITSP. It probably doesn;t matter if thare is PSTN involved or on-net calls. |
| 19:20.23 | WIMPy | But sill SIP doesn't know something like call waiting. |
| 19:27.31 | JamesJRH | ..So the catch I see from reading the above: although PSTN can do call-waiting, and channels are not directly relevant to VoIP, accessing the PSTN *via* a VoIP trunk to an ITSP, may not allow the call-waiting of the PSTN. I.e. having one native PSTN channel can have a 2nd incoming call wait, but having one PSTN channel via a SIP trunk to an ITSP, all calls would just be rejected when there is a call engaging the line. |
| 19:28.20 | JamesJRH | That was quite long. Did that get cut again? |
| 19:29.28 | WIMPy | Almost. |
| 19:30.26 | WIMPy | Call waiting is not related to a channel. That's why you can have multiple calls waiting, no matter how many channels you've got. |
| 19:30.51 | WIMPy | Unless you're using POTS in which case multiple waiting calls just couldn't be signalled. |
| 19:31.11 | JamesJRH | Okay good, that's what I was hoping. |
| 19:31.59 | JamesJRH | So a landline is POTS. That's can't handle the signalling. |
| 19:32.21 | WIMPy | It can be POTS. It can also be ISDN or even somethign bigger. |
| 19:32.30 | [TK]D-Fender | Your inbound provider has to allow it, and Asterisk has not way to flip them. |
| 19:33.10 | [TK]D-Fender | And still has no bearing on how ITSP's work |
| 19:33.11 | JamesJRH | WIMPy: Other than POTS, what else is PSTN? So it is generally ISDN when it's not POTS? |
| 19:33.26 | WIMPy | Well, There seem to be some suport in Asterisk, but I'm not sure how usefull it is. |
| 19:33.27 | JamesJRH | [TK]D-Fender: Okay. |
| 19:33.56 | [TK]D-Fender | WIMPy: Only thing I've seen is FLASH on analog channels... |
| 19:34.03 | [TK]D-Fender | WIMPy: Which is another animal |
| 19:34.33 | WIMPy | It's certainly a rather huge disadvantage of using Asterisk. |
| 19:35.23 | JamesJRH | WIMPy: So ISDN is a type of PSTN. And ISDN has the ability to signal call-waiting. |
| 19:35.56 | WIMPy | yes |
| 19:36.02 | JamesJRH | ITSPs or Asterisk may not support this signalling. |
| 19:36.41 | WIMPy | SIP ertainly doesn't. So that implies a no in many (most) cases. |
| 19:36.50 | JamesJRH | Okay. |
| 19:40.04 | JamesJRH | WIMPy: So assuming Asterisk has *any* support for it, if SIP doesn't, the server running Asterisk would either have to use a direct ISDN interface card, or use a trunk protocol that supports it to an ITSP that supports it. |
| 19:40.27 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 19:40.40 | JamesJRH | Perhaps IAX supports it. |
| 19:41.16 | WIMPy | Forget about it with Asterisk. |
| 19:42.13 | WIMPy | It may be available with Skinny and there is some rudimentary support in DAHDI, but I doubt it's possible to make any sensible use of it other than logging missed calls. |
| 19:42.32 | JamesJRH | Okay. |
| 19:42.59 | JamesJRH | Thanks for helping me understand the situation anyway. |
| 19:44.49 | JamesJRH | (Bbl.) |
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| 20:37.52 | gusto | so |
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| 23:33.05 | JamesJRH | WIMPy: I had to rush off earlier, though I didn't have much more to say. |
| 23:33.05 | JamesJRH | I was asking about incoming calls to help me understand PSTN a bit more (which is a bit of an alien to me). |
| 23:33.08 | JamesJRH | I don't mind if Asterisk only has rudimentary support for call-waiting because, call-waiting is less important to me than the ability to ring out to multiple PSTN phones with the intention of only one picking up. |
| 23:33.12 | JamesJRH | I understand now that to do this, I'll need a PSTN channel per outward ring even before one of the phones picks up. It makes sense due to the legacy of ringing being transmitted as audio from the exchange, unlike in VoIP which doesn't connect audio until the called party picks up. |
| 23:33.20 | JamesJRH | So basically I'll need to have a few more channels than I first thought. |
| 23:33.39 | [TK]D-Fender | Out calls has nothing to do with call waiting |
| 23:33.44 | [TK]D-Fender | those are outbound channels |
| 23:33.57 | [TK]D-Fender | Thre is no "dial multiple over one channel " option |
| 23:34.28 | [TK]D-Fender | And I gave you a precise sample for doing it along with the statemtn that if one of them answers the others will be terminated |
| 23:40.29 | JamesJRH | <PROTECTED> |
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| 23:40.34 | JamesJRH | 23:33:39 < [TK]D-Fender> Out calls has nothing to do with call waiting ← Call-waiting being an incoming feature, I see the outgoing equivalent to be ringing more than one phone that hasn't yet picked up. Though the apparent asymmetry of PSTN utilising the line (for audio) when ringing out, but not when ringing in, means that these 2 concepts also don't appear symmetrical. |
| 23:41.16 | [TK]D-Fender | indeed they aren't |
| 23:49.51 | JamesJRH | [TK]D-Fender: Yes, I see your sample. What is the ‘,20’ for? |
| 23:50.22 | JamesJRH | [TK]D-Fender: Also, what do you mean by “core show application dial”? |
| 23:51.03 | [TK]D-Fender | read Dial's instructions |
| 23:56.42 | JamesJRH | [TK]D-Fender: Well I tried to look for Asterisk documentation the other day and the website was mangled in parts. It appears to still be partially mangled, so please could you link me to a non-mangled documentation page, or just answer directly? |
| 23:57.08 | JamesJRH | I wonder if the 20 is for rings maybe. |
| 23:57.32 | [TK]D-Fender | I just gave you the literal command to copy & paste into * CLI to GET the instructions |
| 23:58.27 | [TK]D-Fender | And that's 20 seconds. |