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| 02:29.27 | nickgaw | Hi, I have been looking for a sip hardware based phone that can work with wireless AC and is open firmware where the phone can be configured either threw a web interface or by sending the phone a configuration file as I am totally blind and can't use the built in menus on the device do you have any suggestions or is this not the proper channel to ask this information? |
| 02:30.42 | nickgaw | I have looked at office supplies stores but they don't know what a sip phone is. |
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| 03:14.33 | [TK]D-Fender | You're asking for a unicorn |
| 03:15.33 | nickgaw | Are you saying such a hardware based wireless phone does not exist if so what is the closest thing then? |
| 03:15.45 | [TK]D-Fender | Depending on your use of "open" |
| 03:16.12 | [TK]D-Fender | And obviously office stores wil know nothing of this. |
| 03:16.31 | [TK]D-Fender | go shop around online VoiP hardware retailers |
| 03:18.00 | nickgaw | Where I can modify the phone threw other methods other then the web interface if that is not accessible with a screen reader like can I ssh into it and edit a configuration file in the phone or write a configuration file and then upload it into the phone either threw sftp or ftp or are there just web interfaces and menus on the phone itself? |
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| 03:18.44 | [TK]D-Fender | depends on the model |
| 03:18.55 | [TK]D-Fender | <[TK]D-Fender> go shop around online VoiP hardware retailers <----------- |
| 03:19.10 | [TK]D-Fender | go see what's out there. Go look at their admin guides to soo how you can configure them |
| 03:19.10 | nickgaw | Do some hardware sip phones exist where I can recompile the firmware with the new settings then upload new firmware with the settings built into the phone? |
| 03:19.23 | [TK]D-Fender | you don't recompile firmware |
| 03:19.31 | [TK]D-Fender | a config file is a dumb config file |
| 03:19.37 | nickgaw | true |
| 03:19.51 | [TK]D-Fender | the word you're looking for is PROVISIONING |
| 03:19.58 | nickgaw | Any good brands I should look at and any I should stay away from? |
| 03:20.09 | [TK]D-Fender | so like I said, get shopping. Go find models. Go read their admin guides to see how to configure them |
| 03:20.27 | [TK]D-Fender | there are veryfew WiFi models to begin with |
| 03:20.43 | [TK]D-Fender | They are pricier and their battery life tends to such |
| 03:20.44 | nickgaw | That is what I have been finding. |
| 03:20.44 | [TK]D-Fender | suck |
| 03:22.35 | nickgaw | Would any types of sip phones have a base that I plug into the network port on the router then I can put the handset on another charger and it acts like a land line phone but just sip? I don't want one of those ATA adapters as transfering and other asterisk tasks won't work on those I don't think. |
| 03:24.10 | [TK]D-Fender | You're immediately wrong there |
| 03:24.24 | nickgaw | I guess I could configure feature codes in asterisk like where you push flash then when you hear the dial tone enter in the code is that how most people setup their phones or do you mainly use phones with buttons like transfer right on the phone itself? |
| 03:24.36 | [TK]D-Fender | of course ATA's can do transfers, 3-way conferencing, etc |
| 03:25.27 | nickgaw | Would the phone just do what I described where you enter in a feature code to do things like transfering? |
| 03:25.29 | [TK]D-Fender | And there is no feature in * for you hitting "flash". "Flash" isn't a thing except for whatever physical interface you plug an analog phone directly into. |
| 03:26.38 | [TK]D-Fender | most transfers are things like: Flash. Get 2nd dialtone. Dial destination. Hear them talk. hang up and call gets handed over |
| 03:26.41 | nickgaw | What I mean by flash is a button where it hangs up and picks it up again like you use it to make three way calls. That is called flash button on some analog phones. |
| 03:26.54 | [TK]D-Fender | ASTERISK does not not talk "flash" |
| 03:26.57 | [TK]D-Fender | an ATA would |
| 03:27.31 | nickgaw | ok but are the feature codes coded into the ATA or in asterisk itself? |
| 03:27.43 | [TK]D-Fender | ATA itself has functionality. |
| 03:28.03 | [TK]D-Fender | Asterisk has DTMF-based features, but those should be avoided if your devices offer their own |
| 03:29.20 | nickgaw | ok so I would just have to learn what codes I enter on my analog phone to do the transfering or are these buttons on the ATA itself as I thought the ATA plugs into the router and I don't want to always have to go into the room where the router is to just transfer a call? |
| 03:29.58 | [TK]D-Fender | there are no buttons on the ATA |
| 03:30.09 | nickgaw | ok did not think so. |
| 03:30.10 | [TK]D-Fender | I just described a common process for a transfer |
| 03:30.14 | [TK]D-Fender | and you don't seem to be listening |
| 03:30.54 | nickgaw | yes I got that message. |
| 03:31.02 | [TK]D-Fender | I've had to say it 3 times now |
| 03:31.17 | [TK]D-Fender | I hope we don't need a 4th |
| 03:31.44 | nickgaw | basically you make a three way call then once the other person answers you hang up. I understand that. |
| 03:32.34 | nickgaw | no I understood your points I was typing when your message came across so had to go back to read it. |
| 03:34.08 | nickgaw | Some of the reviews on some sip phones are not the best. When you either buy a sip phone or an ATA do you often look at the reviews before you buy it? |
| 03:34.40 | [TK]D-Fender | Who's opinion am I going to trust? |
| 03:34.50 | nickgaw | I like to do so but don't always go by what the review says. |
| 03:35.12 | [TK]D-Fender | Just Joe Blow at Best Buy who has no idea what he's doing and can't figure things out leaves a bad review saying "I can't get it to work". |
| 03:35.19 | [TK]D-Fender | Am I going to trust him? |
| 03:35.26 | nickgaw | no |
| 03:36.12 | nickgaw | but you don't always know if the person knows what they are talking about or not I like to read good reviews about devices from people who know what they are doing. |
| 03:36.13 | [TK]D-Fender | Ask around in places where people have actually had real experience with a variety of devices. Read the feature sheets. Read their manuals. Watch some videos on youtube. |
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| 03:37.55 | nickgaw | I think the ATA adapters might be the best thing to start out with. |
| 03:38.27 | [TK]D-Fender | That depends on the actual need you're solving, and how you need to operate |
| 03:38.46 | [TK]D-Fender | Which is something you aren't giving a clear explanation of |
| 03:39.02 | [TK]D-Fender | You mention technical things about configuring without any actual reasoning behind it |
| 03:39.18 | nickgaw | Some of these ATA adapters look ok and some even exist with more then one port. |
| 03:41.38 | nickgaw | What I am talking about is entering in the sip information into the phone from what I read on line lots of these devices have web interfaces to set the sip information up. Some web interfaces are not the best to use with a screen reader but I can not find any good reviews of this type would perhaps writing the company who sells the ATA and perhaps asking if they could configure the sip information for me so when I get it I just plug it in and it works? |
| 03:43.19 | nickgaw | I did write one of the sellers about this with one of the ATA adapters sometime ago but they never wrote back so I am going to try another seller. |
| 03:43.40 | nickgaw | ok I remember you |
| 03:44.00 | [TK]D-Fender | Usually you would contact an actual VoIP integrator company if you expect extra service. |
| 03:44.07 | kai[El] | Hi |
| 03:44.20 | [TK]D-Fender | A retailer tends to just sell you a boxed product and that's it |
| 03:44.55 | nickgaw | I was looking on amazon. |
| 03:45.00 | [TK]D-Fender | And yes I am recalling you as well. They don't make these products to be easily maintained by the blind. It just isn't a thing. This is not consumer tech. |
| 03:45.43 | [TK]D-Fender | The only larger place I can think of that might do it would be https://www.voipsupply.com |
| 03:45.53 | nickgaw | Is it best to look on other sites like special sites that just sell voip hardware and I am ok with paying someone to do the configuration. |
| 03:46.13 | nickgaw | I will check there. |
| 03:46.29 | [TK]D-Fender | I know they have a more dedicated service support staff. call them and ask if they can do a pre-configuration of your devices before shipping |
| 03:47.22 | [TK]D-Fender | If this is for your personal use then an ATA would be infinitely better choice than a fancy WiFi type device. |
| 03:47.25 | nickgaw | ok I will do that and see what I can find out. Do you buy from them and have you had good support if you ever need it? |
| 03:48.07 | [TK]D-Fender | ATA's have basic functionality off more basic phones who have simpler buttons and don't require tyou to see a screen to use |
| 03:48.18 | [TK]D-Fender | I've bought from them and I don't need support |
| 03:48.37 | [TK]D-Fender | I can successfully read the manuals of everything I've gotten my hands on. |
| 03:48.59 | [TK]D-Fender | How many phones are you looking to deploy? |
| 03:49.47 | nickgaw | Most manuals are in pdf format which I can read I guess I will find one ATA adapter and find the manual and read it before I buy anything as this is what I do when I buy other products if I can find a manual. |
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| 14:15.11 | martinvw | Hi. I want to add a SIP header Alert-Info:alert-internal to our complete "internal" context, without having to add it manually to each extension. Is this possible? |
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| 14:20.58 | Samot | Yes. |
| 14:21.14 | Samot | Both SIP and PJSIP have dialplan functions to set SIP headers before you dial |
| 14:21.22 | Samot | Or even in one of the dial macros. |
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| 14:49.10 | martinvw | Samot: hmm, do you mean SIPAddHeader() in the dialplan? That is what I've found so far, but I'm at a loss trying to figure out how to use it. I know that I can use it an extensions, and I've successfully tested it with a test extension: https://pastebin.com/Lg6gySdX |
| 14:51.46 | martinvw | Now when I call this extension from a phone, it works as expected, and my phone will use the ringtone indicated by the Alert-Info header. But my problem now is this: there are multiple extensions in my [internal] context (e.g. our internal phones, and some groups), and some other parts of the dialplan will route calls into these extensions using e.g. Goto(internal,martin,1). If I were to add the SIPAddHeader() call to these extensi |
| 14:51.46 | martinvw | <PROTECTED> |
| 14:52.00 | [TK]D-Fender | add it in your peer definition |
| 14:52.04 | [TK]D-Fender | SetVar <--------- |
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| 14:52.58 | Samot | 10:21:22 AM <Samot> Or even in one of the dial macros. |
| 14:53.22 | martinvw | Samot: sorry, haven't quite figured out what dial macros are :/ |
| 14:53.30 | [TK]D-Fender | that dialplan gets called by other stuff he doesn't want it happening for |
| 14:53.37 | Samot | Or a GoSub |
| 14:53.53 | [TK]D-Fender | <[TK]D-Fender> add it in your peer definition |
| 14:53.53 | [TK]D-Fender | <[TK]D-Fender> SetVar <--------- |
| 14:54.17 | Samot | The idea was not to do it individually |
| 14:54.49 | Samot | But yeah, SetVar can be used. |
| 14:54.55 | Samot | A GoSub could be called |
| 14:55.07 | martinvw | [TK]D-Fender: setvar looks like it might work. I assume I'd add setvar=ALERTINFO=alert:internal to my sip.conf, and then I'd use SIPAddHeader() with this variable in the dialplan? |
| 14:55.34 | [TK]D-Fender | There is a function you should be using instead of that app IIRC |
| 14:56.09 | [TK]D-Fender | You could also check the callerID or some other value to determine if you should set it or not in the dialplan |
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| 14:57.57 | martinvw | [TK]D-Fender: hmm, but that brings me back to my original confusion. Can I do that context-wide, e.g. in the beginning of the context, I'll check the caller ID, add the header if I want, and this will then affect all extensions in that context? Or do I need to do that separately in each extension? |
| 14:58.31 | [TK]D-Fender | Depends how you set up your dialplan |
| 14:58.40 | [TK]D-Fender | "beginning of context" doesn't mean anything |
| 14:58.47 | [TK]D-Fender | extens are extens |
| 14:59.32 | [TK]D-Fender | Most people use macros for repetitive standard dialplan for dialing devices, etc which gives you fewer places to add these checks |
| 15:00.30 | igcewieling | I'm shocked at how well G711 works with faxing. |
| 15:01.26 | igcewieling | I never intended to use g711 with outbound faxing from Asterisk, but I can't seem to make t38 works. |
| 15:02.21 | martinvw | So I'd create a macro that evaluates my conditions, conditionally adds my SIP header, and then I'd add a Macro call to each extension? (Or rather, I'd do that with GoSub because the docs say macros are deprecated?) |
| 15:02.55 | [TK]D-Fender | Depends what your dilaplan looks like as a whole |
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| 16:16.52 | igcewieling | In case anyone wants to see a not-all-that-well-written incoming fax AGI: https://pastebin.com/U4GzJSqh It should be usable by others with a little bit of work. |
| 16:24.21 | ntwrk | I dug into my issue a little bit more. It looks like after a sip reload, peers loaded from realtime work normally until they try to put someone on hold. When they put someone on hold, instead of a QueueMemberStatusEvent with status 8 (ONHOLD) an event for status 1 (READY) is issued. This only affects peers loaded from realtime and only happens for the call the peer was on when the sip reload took place. Subsequent calls are able to fun |
| 16:24.21 | ntwrk | ction as expected. Peers are loaded from realtime, queues are loaded from a flat file. callevents, notifyhold, callcounter, rtcachefriends are enabled, rtautoclear is disabled. for queues, eventmember status is enabled. Someone suggested yesterday that I create a queue_members table and map it in extconfig, this didn't seem to have any effect. |
| 16:37.38 | martinvw | [TK]D-Fender, Samot: thanks for your input! I've got it working. I've changed our sip.conf to route into a new context "frominternal", and defined the context like this: https://pastebin.com/JFDgDzMv |
| 16:41.40 | [TK]D-Fender | martinvw, if you can live with it added on every call they place to anywhere and that you will match everything all the time and never 404 or hit any other kind of handler. |
| 16:42.05 | [TK]D-Fender | martinvw, I would sooner place it only where I needed it |
| 16:45.12 | martinvw | [TK]D-Fender: hadn't considered that. back to the drawing board :( |
| 16:50.08 | martinvw | I'm considering the SetVar approach again. Are there best practices considerin variable naming? Is "INTERNAL" ok, or should I prefix the variable e.g. with my company name to ensure it'll never conflict with potential future Asterisk variables? |
| 16:56.32 | igcewieling | Does anyone have ideas on how to prevent SendFax from using audio instead of T38 to send the fax |
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| 17:09.44 | martinvw | OK, new solution. We have a context [user] that contains the Dial command that will actually make the call to the user phones. I've added setvar=MYPREFIX_INTERNAL=1 to the template in our sip-users.conf, and the following line right before the Dial: same => n,ExecIf($[${EXISTS(${MYPREFIX_INTERNAL})]?SIPAddHeader("Alert-Info:alert-internal")) |
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| 17:21.34 | [TK]D-Fender | that looks decent |
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| 17:39.24 | martinvw | [TK]D-Fender: thanks! |
| 17:40.49 | [TK]D-Fender | Now keep in mind that transferred calls of some kinds may pick that up as well. |
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| 19:55.32 | craigify | igcewieling, about that comment of weird caller ids you mentioned the other day..are you talking about SIP trunks, or PRI, or otherwise? |
| 19:56.44 | craigify | I definitely get all kinds of weird stuff on incoming SIP INVITES, but PRI not really. |
| 19:57.52 | igcewieling | craigify: PRI from Earthlink. I'm not sure about the backhaul, it could be SIP. The only thing I can think of is that it has a SIP backhaul and someone hacked into the telco box which does SIP<->PRI. |
| 19:58.09 | craigify | yeah, that's not a bad theory |
| 19:58.24 | igcewieling | but even then, you'd think the dialed number would be the weird digits |
| 19:58.42 | craigify | the dialed number is valid? |
| 19:58.42 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 19:58.58 | igcewieling | craigify: the dialed number is a telephone number on the PRI |
| 19:59.07 | igcewieling | so, yes it is valid. |
| 19:59.23 | igcewieling | we get just the last 4 digits from the telco |
| 19:59.23 | craigify | that's freaking weird |
| 19:59.30 | igcewieling | It is freaking weird. |
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| 20:02.59 | craigify | I'd almost route them to a phone one day and ansswer it and see what happens |
| 20:02.59 | craigify | lol |
| 20:03.37 | igcewieling | I routed them to the system IVR but none ever sent DTMF. Now I do a Hangup(21) for calls with CallerID longer than 11 chars. |
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