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| 00:19.47 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:19.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:56.30 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
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| 02:11.48 | *** join/#asterisk ctcx (~user@unaffiliated/ctcx) |
| 02:14.51 | ctcx | Thinking about enabling SIP text message, for use for circa 20 users. Thinking about using a client such as jitsi for handling all text messaging over SIP, like a plain chat. |
| 02:15.53 | ctcx | Do you think this can be viable? Would voice calls themselves not be so negatively affected? |
| 02:17.56 | *** join/#asterisk matrix1233 (~matrix123@197.1.28.109) |
| 02:18.44 | ctcx | [TK]D-Fender: excuse me... hello? |
| 02:20.17 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 02:23.01 | wyoung | ctcx: It's nap time for [TK]D-Fender |
| 02:23.56 | ctcx | Ouch sorryu |
| 02:24.02 | ctcx | *sorry |
| 02:24.32 | wyoung | I have been interested in setting up chat via SIP however I have never done it before. |
| 02:24.38 | wyoung | Let me know how it goes :) |
| 02:25.49 | ctcx | Your emoticon is appreciated, yet it's never good thing to be alone in some predicament |
| 02:26.26 | wyoung | You are not alone! I am sure someone else in here has done that before. Or even someone who has used the asterisk mailinglists |
| 02:27.23 | ctcx | Wish there was someone online now with that experience... |
| 02:28.21 | wyoung | They are probably in America. It is America's nap time now. |
| 02:29.25 | ctcx | goes to check time zone tables |
| 02:30.35 | ctcx | ..... if I were an american worker right now I'd be certainly resting from work, but not asleep yet! |
| 02:30.52 | Samot | What do you mean "chat"? |
| 02:31.26 | ctcx | But well, that would be me at least. Maybe they're actually more sane... |
| 02:31.57 | Samot | Define "chat" |
| 02:34.25 | Samot | ctcx: ^^ |
| 02:37.16 | ctcx | Samot: well... like irc, skype, openfire... real time text messaging, 10 or 20 SIP users actively talking to each other through this way... |
| 02:37.29 | Samot | That's not SIP. |
| 02:38.46 | ctcx | Then I guess SIP's message sending/receiving functions work in different way? |
| 02:39.05 | lorsungcu | ctcx: you should use IRC |
| 02:39.11 | lorsungcu | i hear it's pretty swell |
| 02:40.02 | ctcx | Well, we're using IRC right now, aren't we? |
| 02:40.18 | wyoung | :) |
| 02:40.20 | ctcx | But then, guess SIP is not for that? https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend |
| 02:40.31 | Samot | I didn't say there wasn't a way to do it. |
| 02:40.38 | wyoung | Samot: Asterisk supports instance messaging type services right? |
| 02:40.41 | Samot | I said the examples you gave were not SIP. |
| 02:40.49 | Samot | Yes. |
| 02:40.57 | wyoung | \o/ |
| 02:41.06 | Samot | The question isn't about what Asterisk can do. |
| 02:41.11 | ctcx | wyoung: did you mean "instant messaging"? |
| 02:41.16 | wyoung | [TK]D-Fender: BTW, I am in the #samba channel too if you still need assist with it :) |
| 02:41.18 | Samot | You can use the MESSAGE method. |
| 02:41.35 | wyoung | ctcx: yes that is what I meant |
| 02:42.15 | ctcx | Then I'm the one being confused (even after having previously read docs). |
| 02:42.25 | wyoung | ok |
| 02:42.26 | ctcx | Ok. MESSAGE is certainly part of SIP. |
| 02:42.34 | Samot | The real question is: Is SIP really the best way? |
| 02:42.39 | wyoung | you can also sent SIP messages too via it's header I think |
| 02:42.53 | Samot | You have to be carefule. |
| 02:42.56 | Samot | You have to be careful. |
| 02:43.10 | ctcx | Then, hoping it's a correct question, what would be the difference between chat per se and SIP's MESSAGE? |
| 02:43.10 | Samot | First there are no contact headers in a MESSAGE. |
| 02:43.33 | Samot | ctcx: The method in which the messages are sent. |
| 02:43.52 | Samot | SIP MESSAGEs are not meant to be long. |
| 02:44.04 | Samot | The clients need to support it. |
| 02:44.04 | ctcx | Samot: aahh. And also the lack of contact header as you just say |
| 02:44.12 | ctcx | Mm... |
| 02:44.51 | ctcx | Samot: https://jitsi.org/Documentation/RegistrarlessSIPAccount https://jitsi.org/Documentation/FAQ#sip-force-message would this count as "support it"? |
| 02:45.34 | Samot | Sure. |
| 02:46.17 | ctcx | Samot: "Not meant to be long" Good point... What's the maximum number of characters SIP MESSAGE can support? Is it in * wiki? |
| 02:46.43 | Samot | You know when your device sends an OPTIONS keep-alive to Asterisk or vice versa? |
| 02:46.55 | Samot | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE |
| 02:47.26 | Samot | ^^^ You get something like that in return. Which are the _options_ the client/server supports. |
| 02:47.31 | ctcx | No, nothing about that... |
| 02:47.41 | ctcx | I'm a total noob in * |
| 02:47.56 | Samot | So yeah all that does is force the client to use MESSAGE when the other side says MESSAGE isn't an option. |
| 02:47.56 | ctcx | I was just about to try enabling messaging in the dialplan for first time |
| 02:48.21 | Samot | But why do you need to do messages over SIP? |
| 02:48.55 | Samot | This isn't going to be like Skype. |
| 02:49.10 | Samot | People can't "join" rooms, etc to chat with each other. |
| 02:49.43 | ctcx | Samot: truth to be told, in order to avoid setting up an apart chat server for this place, and squeeze at maximum what's already here, which is an * server |
| 02:50.11 | Samot | ctcx: There is no chat. |
| 02:50.11 | ctcx | And I don't seek anything near skype at all |
| 02:50.23 | Samot | There is _messaging_ |
| 02:50.35 | Samot | Ie: I send you a message...you get it. |
| 02:50.38 | Samot | You reply |
| 02:50.47 | Samot | This is basically SMS |
| 02:51.11 | Samot | It will not be a "chat" system like IRC, Skype or other chat/IM programs. |
| 02:52.11 | Samot | So it sounds like you need a chat server for what you want to do. |
| 02:53.29 | lorsungcu | ok but will it be like snapchart |
| 02:53.48 | Samot | Maybe? |
| 02:54.07 | Samot | I just know SIP is not meant for being used for chat rooms. |
| 02:54.19 | Samot | That is what XMPP is for. |
| 02:55.09 | ctcx | Aaahhh, now it's clearer. |
| 02:55.46 | Samot | SIP MESSAGE = text message/SMS |
| 02:55.53 | ctcx | Though, probably SMS-like functionality could be enough for what they need. |
| 02:56.46 | Samot | Are you planning on using Asterisk just for this |
| 02:56.47 | Samot | ? |
| 02:56.59 | Samot | because, again, Asterisk is not really a chat server. |
| 02:57.20 | ctcx | Of course not. |
| 02:57.34 | ctcx | Just as an "extra" function. |
| 02:57.48 | ctcx | The primary function is voice calling of course. |
| 02:57.54 | Samot | OK. |
| 03:00.01 | ctcx | But, SIP MESSAGE can be used for multiple sendings/receivings at same time, right? I.e., say bob is sending alice, and at same time sam is receiving from susan. And so on. |
| 03:00.12 | Samot | Yes. |
| 03:00.43 | ctcx | Since I have no way to test this in massive way yet, I'm a bit afraid. |
| 03:00.54 | lorsungcu | you should be really scared |
| 03:01.28 | ctcx | It's never the same doing tests with 3 users and phones and doing with 100 users |
| 03:01.36 | ctcx | ^ Hey, why? |
| 03:01.54 | ctcx | Samot: is he right or is he just trying to scare away? |
| 03:02.06 | lorsungcu | im scared for you |
| 03:02.12 | *** join/#asterisk Samael28 (~Samael28@79.110.128.128) |
| 03:02.37 | ctcx | WTH |
| 03:02.47 | ctcx | Some mod help please! |
| 03:04.02 | Samot | ctcx: If you're just learning then you shouldn't be touching a production server for any of this. |
| 03:04.11 | Samot | So you're going to need a test server regardless. |
| 03:05.56 | *** join/#asterisk boris_t (~boris_t@363103629.convex.ru) |
| 03:05.57 | ctcx | Samot: er.... I DO have a test server |
| 03:06.29 | ctcx | I'm not stupid enough to mess in production environemnts |
| 03:15.54 | [TK]D-Fender | <ctcx> Some mod help please! <- with what? |
| 03:16.16 | [TK]D-Fender | I'm not seeing an actual question in all of this. |
| 03:17.45 | ctcx | [TK]D-Fender: no, nothing. A guy was scaring out. That's all. |
| 03:17.56 | ctcx | thanks and sorry |
| 03:20.03 | ctcx | [TK]D-Fender: we were discussing about how viable can be using SIP MESSAGE a bit "massively". |
| 03:20.07 | *** join/#asterisk matrix1233 (~matrix123@197.1.28.109) |
| 03:20.34 | ctcx | Like, 50 users messaging among them. |
| 03:21.06 | ctcx | Using something like this https://jitsi.org/Documentation/RegistrarlessSIPAccount https://jitsi.org/Documentation/FAQ#sip-force-message |
| 03:21.19 | Samot | As far as Asterisk is concerned, it is a call. |
| 03:21.26 | Samot | End of story. |
| 03:21.52 | Samot | The SIP MESSAGE still needs to be processed via dialplan like any other call. |
| 03:22.54 | Samot | ctcx: That option is not for what you think. |
| 03:23.01 | Samot | In Jitsi. |
| 03:23.46 | Samot | Jitsi is going to send an OPTIONS to Asterisk and when it gets this: |
| 03:23.54 | Samot | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE |
| 03:24.40 | Samot | ^^ *IF* MESSAGE is missing from that list (ie Asterisk doesn't tell you it supports MESSAGE) that option still will treat the other side as _if it does support it_ |
| 03:27.09 | ctcx | Samot: for which very first thing I'm trying to do is certainly enabling MESSAGE in *'s dialplan just like wiki says. |
| 03:27.47 | ctcx | After that, I expect jitsi to indeed be able to support text messages via SIP MESSAGE, even if as in SMS |
| 03:28.44 | Samot | outofcall_message_context=<context> |
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| 03:29.12 | ctcx | Yes, <context> should be "message" |
| 03:29.22 | Samot | In the sample, sure. |
| 03:29.30 | Samot | It the context you name it. |
| 03:30.22 | Samot | So you could have to SIP peers with that setting.. |
| 03:30.31 | Samot | And send their MESSSAGEs to different contexts. |
| 03:30.40 | Samot | s/to/two/ |
| 03:30.48 | Samot | grrr.. |
| 03:43.17 | ctcx | And just out of curiosity, would SIP MESSAGE be able to keep kind of a "conversation history"? Like in cell phones (ok, smartphones nowadays...) |
| 03:43.33 | Samot | That's up to the client. |
| 03:43.47 | Samot | Sipjs, the IP phone, etc. |
| 03:44.02 | ctcx | Samot: no, wait... do you mean this https://jitsi.org/Documentation/RegistrarlessSIPAccount won't actually enable send/receive SIP MESSAGEs on jitsi? |
| 03:44.06 | Samot | Just like keeping a history of the messages on the server would be up to you. |
| 03:44.47 | ctcx | Samot: no, wait... do you mean this https://jitsi.org/Documentation/RegistrarlessSIPAccount won't actually enable send/receive SIP MESSAGEs on jitsi? |
| 03:44.53 | ctcx | (OUCH sorry for that) |
| 03:45.03 | Samot | ctcx: You are looking at JITSI SPECIFIC stuff. |
| 03:45.12 | Samot | I can't comment on how Jitsi does things. |
| 03:46.29 | ctcx | Ok. thanks for everything. |
| 03:46.36 | ctcx | Needto go. |
| 03:46.41 | ctcx | *need to |
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| 08:08.38 | hollabackatcha | hey. so i have 2 trunks, and i can route incoming calls nicely to different extensions.. (i have 2 numbers) |
| 08:09.19 | hollabackatcha | but no way i can do the same for outgoing calls.. it's always going through the second line it seems (caller id when i test on my phone is from the second line) |
| 08:09.38 | hollabackatcha | although in the logs i see the calls are routed through correct trunks |
| 08:10.07 | *** join/#asterisk Samael28 (~Samael28@79.110.128.128) |
| 08:10.19 | hollabackatcha | i have a grandstream pstn gateway.. and 2 accounts for 2 channels configured on it. |
| 08:10.41 | hollabackatcha | anybody have any idea what to do ? i've been trying things out for whole day now, can't figure it out |
| 08:11.38 | Haris | hello all |
| 08:11.51 | Haris | I upgraded my fpbx based asterisk version from 11 to 13 |
| 08:12.08 | Haris | yesterday cdr seemed fine. today it can't find the lengthy file-name configured in it |
| 08:17.07 | Haris | this ( http://pastebin.ca/3799704 ) is being repeated for most calls |
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| 08:17.59 | Haris | fpbx re-generated the dialplan when we upgraded from 11 to 13 via the asterisk-version-switch script |
| 08:18.10 | Haris | actually, I do have a backup copy of the config files |
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| 08:26.18 | sekil | who knows what fpbx does |
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| 09:17.04 | Haris | ah well. one can always see the dialplan |
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| 10:52.20 | Haris | hello all |
| 10:53.08 | Haris | I have a Q, I have 10 extensions who are member of it. but only 5 out of those 10 are registered with asterisk. will asterisk try to route calls to those non-registered members of the Q as well ? |
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| 11:13.28 | Samot | FFS. |
| 11:13.35 | Samot | Are they static member of the queue? |
| 11:24.18 | Haris | yes |
| 11:24.26 | Samot | Sigh. |
| 11:24.30 | Samot | What does STATIC mean? |
| 11:24.53 | Haris | I know |
| 11:24.53 | Haris | ok |
| 11:25.03 | Haris | I'll make them dynamic |
| 11:25.11 | Samot | So if they are not registered, it doesn't mean they aren't still in the queue. |
| 11:25.12 | Haris | no. I'll keep them static for now |
| 11:25.25 | Haris | work for dynamic will have to wait a bit |
| 11:25.27 | Samot | You change them to dynamic, they have to log in and out of the queue. |
| 11:25.36 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 11:26.10 | Haris | noted |
| 11:26.51 | Samot | Also, STOP COMING IN HERE FOR FREEPBX STUFF |
| 11:27.47 | Haris | *sigh* |
| 11:27.47 | Haris | ok |
| 11:28.10 | Samot | FreePBX OWNS what happens on the server. |
| 11:28.17 | Samot | It OWNS what happens in the dialplan. |
| 11:28.37 | Samot | It is not how a standard Asterisk setup is done. |
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| 11:58.36 | sekil | freepbx get's owned too :) |
| 11:59.35 | Samot | You can be a FreePBX user and ask questions in here about Asterisk based stuff. How it can work or function. |
| 11:59.39 | Samot | No issue with that.. |
| 12:00.01 | Samot | But for help on configuring those things, it's not the right place. |
| 12:00.07 | sekil | Samot: you're with Digium? |
| 12:00.11 | Samot | No. |
| 12:01.24 | Haris | not to discount for my way of getting info, I'v rarely encountered a community where there were three or less guys active each day |
| 12:01.51 | Haris | I do understand I was ask for quite alot of info |
| 12:01.59 | Haris | %s/was// |
| 12:02.49 | Samot | 6:53:08 AMÂ <Haris>Â I have a Q, I have 10 extensions who are member of it. but only 5 out of those 10 are registered with asterisk. will asterisk try to route calls to those non-registered members of the Q as well ? |
| 12:02.56 | Samot | ^^^ That question right there.. |
| 12:03.06 | Samot | Is ANSWERED IN DOCUMENTATION |
| 12:03.14 | Samot | Various documentation. |
| 12:03.35 | Haris | yes. and I'v also recently seen a video on it |
| 12:03.35 | Samot | Asterisk Wiki, queues.conf.sample (found online easily), FreePBX Wiki |
| 12:03.50 | sekil | omg..videos.. |
| 12:04.00 | Samot | So when you ask questions that are easily answered with 5 minutes of Googling.. |
| 12:04.10 | sekil | what ever happened to reading? |
| 12:04.29 | Samot | Well that's even worse. |
| 12:04.38 | Samot | There's moving pictures and talking involved in videos. |
| 12:04.42 | Haris | I usually don't have enough free time in the day to read regularly |
| 12:04.47 | Samot | BS. |
| 12:04.55 | sekil | oh really? |
| 12:05.01 | sekil | and we do ? |
| 12:05.05 | Samot | You claim to be a System Admin with 17 years experience... |
| 12:05.08 | sekil | gets pissed of |
| 12:05.12 | Samot | Reading is PART OF THE JOB |
| 12:05.18 | Samot | LEARNING IS PART OF THE JOB |
| 12:05.42 | Samot | So what you are saying is you don't have enough time in the day to do was part of your job? |
| 12:06.04 | Haris | I think you people have far less stress in your timings than me |
| 12:06.04 | Samot | s/was/what is/ |
| 12:06.13 | Samot | HAHAHAHA |
| 12:06.28 | Samot | Haris: I'm a single person shop. |
| 12:06.44 | Haris | I'm not going to whine about my conditions here or try to justify stuff. I'v already made alot public |
| 12:06.52 | Samot | Managing voice networks and supporting customers for like a half dozen VoIP Providers.... |
| 12:07.18 | Samot | Somehow I have time to learn this stuff... |
| 12:07.24 | Samot | Even NEW stuff. |
| 12:07.26 | Haris | like a specialized shop, with a volume of business that can sustain your speciality in a specific career path |
| 12:07.40 | Haris | and no one to answer to |
| 12:07.40 | sekil | Haris: so what you're saying you're under stress to read? |
| 12:07.46 | sekil | interesting disease |
| 12:07.48 | Samot | Haris: My job requires me to KEEP UP WITH NEW STUFF |
| 12:07.59 | Samot | In order to do that I have to KEEP LEARNING |
| 12:08.00 | Haris | my timings are not yet managed for sustained or lengthy reading |
| 12:08.14 | Haris | or regular reading |
| 12:08.26 | Samot | Haris: I have people to answer to. |
| 12:08.43 | Samot | Haris: They are called my clients. They would like to know how I spend my time since they pay me a lot for it. |
| 12:09.00 | sekil | but you seem to have time to bother people answering questions on answers easily found... |
| 12:09.08 | Samot | ^^^^^^ |
| 12:09.16 | sekil | that seems like bs |
| 12:09.19 | Samot | Who are going to tell you to READ |
| 12:09.22 | voipmonk | you are reading now cut the bs |
| 12:09.28 | Samot | ^^ |
| 12:09.46 | Samot | You come in here at the start of your day... |
| 12:09.48 | Haris | yes, I suppose that's a good point |
| 12:09.56 | Samot | And ask questions ALL DAY |
| 12:10.05 | Haris | I'v been here 24x7 in the past |
| 12:10.10 | Haris | and asked Qs |
| 12:10.16 | Haris | just not from the same guy |
| 12:10.20 | Samot | So how can you say you don't have time to read.. |
| 12:10.37 | Samot | But you can spend 3 hours asking about something you could have READ ABOUT IN 20 MINUTES |
| 12:11.50 | Samot | Zero ownership or responsibility. |
| 12:12.45 | Haris | not that I don't appreciate it .. with permission, I may point out that you could also have pointed me to that doc or wiki page/article, rather than give me a lengthy description or extended info on it |
| 12:12.49 | Haris | I do read. I did upgrade that box |
| 12:12.53 | Haris | and I do the work on my own |
| 12:13.13 | Samot | Haris: I have been point you to the Wikis for a YEAR |
| 12:13.18 | Samot | A YEAR |
| 12:13.45 | Samot | And you are still surprised when you're told "This is in the Wikis" |
| 12:16.56 | *** join/#asterisk sibiria (~sibiria@unaffiliated/sibiria) |
| 12:19.39 | Samot | I take serious offense to people that come in here and say they are just "too busy" to actually learn this stuff, even though it's part of their job, and then suck off the experience others spent time putting in hard work to learn. |
| 12:20.22 | Samot | Because now it's not about "helping out a fellow Asterisk/FreePBX user" |
| 12:21.01 | Samot | It becomes "Please help my do my job" |
| 12:21.08 | Samot | It becomes "Please help me do my job" |
| 12:21.15 | Samot | Which is what this has become. |
| 12:21.54 | sekil | not please.. |
| 12:21.57 | sekil | "do my job" |
| 12:22.22 | Samot | Exactly. |
| 12:22.52 | Haris | I never asked anyone to do my job. I asked for info. I do the work myself. today I asked about error after an upgrade. I asked about a functionality in the Q |
| 12:23.17 | Samot | Functionality that is well documented. |
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| 12:23.17 | Haris | that's not asking anyone to do my job |
| 12:23.25 | Samot | You asked a question that reading would have answered. |
| 12:23.33 | Samot | And then claim to be "too busy to read" |
| 12:24.30 | sibiria | hello. we're currently (still...) running the old asterisk 1.8, and we're trying to find a way to mix callee and receiver as stereo |
| 12:24.33 | sibiria | one in left, the other in right |
| 12:24.38 | sibiria | is this doable? |
| 12:24.50 | sibiria | we're currently using Monitor(), rendering a mixed mono-file |
| 12:24.57 | Samot | MixMonitor |
| 12:25.37 | voipmonk | add " L/R " to your search, sibiria |
| 12:25.37 | Samot | That does what you are looking for. Records the channels into a single recording |
| 12:27.53 | voipmonk | do you care about post processing files, sibiria ? |
| 12:28.41 | voipmonk | are the calls already recorded ? |
| 12:29.04 | sibiria | no this is for a future change. i only need them in stereo instead of mixed together as mono |
| 12:29.08 | sibiria | post-processing not needed |
| 12:29.12 | voipmonk | okie dokie |
| 12:29.26 | voipmonk | less than 500 concurrent calls? |
| 12:29.29 | sibiria | from what i understand, MONITOR_EXEC is being called? |
| 12:29.36 | sibiria | yes, but we have the processing power to handle that amount too :P |
| 12:30.15 | voipmonk | I wonder how you tested that. :) |
| 12:30.21 | sibiria | and MONITOR_EXEC is called with three args, first being in, second out, third argument being the destination file name? |
| 12:30.41 | sibiria | sox/lame isn't heavy, and we have very ample machines |
| 12:32.26 | sibiria | and in fact we're only looking to mix and keep as wav. we only encode later, when archiving |
| 12:33.23 | voipmonk | mixmonitor on the live box or using a mirror port? |
| 12:35.22 | sibiria | on the live box |
| 12:35.47 | Haris | t |
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| 12:41.30 | sibiria | mm yeah MixMonitor seems a better solution than calling a script with MONITOR_EXEC |
| 12:42.01 | voipmonk | try it at scale with sippycup |
| 12:43.17 | voipmonk | watch for any MOS score changes in your voice monitoring |
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| 14:24.57 | sibiria | voipmonk: working ok now. it was a bit of a hassle because sox behaves oddly in regards to paths, but using MixMonitor gets the job done |
| 14:25.25 | sibiria | for some reason, calling sox with two input files without any paths, causes sox to find the first input file but not the second one |
| 14:25.32 | sibiria | really dodgy |
| 14:25.44 | sibiria | had to supply full paths for everything |
| 14:25.55 | sibiria | thanks for the help |
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| 14:35.48 | voipmonk | thats all |
| 14:35.53 | voipmonk | Samot. |
| 14:35.54 | voipmonk | :) |
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| 14:45.44 | Samot | ? |
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| 18:31.48 | nickgaw | Hi, What sound files are by default selected in a new asterisk installation 14.4.0 as I am trying to use make menuselect but am totally blind and that menu system is not the best readible with any screen reader are there ways to manually edit the menu files to configure asterisk before it is compiled as make install does not do much in telling me what features are configured in the menu system and if I try to enter the kernel co |
| 18:32.35 | nickgaw | so how can I suggest improovments to the menu system for accessibility? |
| 18:34.18 | nickgaw | Is there a way to run it from the command line to set options manually? |
| 18:34.59 | r1ppa | I got some code to apply to polycom phones to change default admin password, where do I wedge in the <device> snippet? |
| 18:43.04 | syadnom_ | r1ppa, anywhere |
| 18:45.30 | syadnom_ | r1ppa, I put mine here <phone><device device.set="1"><device.auth device.auth.localAdminPassword="12345" device.auth.localAdminPassword.set="1"/></device></phone> |
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| 18:49.02 | r1ppa | syadnom_, yeah worked, I just assumed it would need its own or there was more to it lol |
| 18:49.04 | r1ppa | thanks |
| 18:49.34 | syadnom_ | the polycom xml requirements are pretty loose. |
| 18:49.42 | r1ppa | syadnom_, hmm I didnt close it off with " device.auth.localAdminPassword.set="1"/></device>" |
| 18:49.46 | syadnom_ | as long as it's properly formatter... |
| 18:50.40 | syadnom_ | r1ppa, I just set that because it might be set to 0 by some previous provisioning or user. |
| 18:50.47 | syadnom_ | so this ensures it will be set. |
| 18:50.55 | nickgaw | Do the Polycom phones have any linux distribution installed on them or how do you edit the xml data on the phone? |
| 18:51.13 | syadnom_ | you don't. only the GUI configs are available |
| 18:51.57 | nickgaw | So there is no way to bypass the GUI and manually configure the phone or change the web interface to add your own features to the firmware? |
| 18:52.26 | syadnom_ | nickgaw, no |
| 18:52.50 | nickgaw | Is the GUI configured by the web or on menus on the phone itself |
| 18:53.05 | voipmonk | to add features to firmware you can call polycom and have them do that for you $ |
| 18:53.12 | syadnom_ | I don't know of any SIP phone that allows custom software to be run except the android based units like grandstream. |
| 18:53.39 | syadnom_ | nickgaw, web primarily. some basic configs on the phone's screen. |
| 18:54.00 | syadnom_ | full configuration access is via xml provisioning |
| 18:54.44 | nickgaw | Does Polycom or other companies have services where once your asterisk system is running you can buy a phone presetup for the extentions you want so all you do is connect it to your network and you are in business? |
| 18:55.23 | voipmonk | yes, I can send you info via email. |
| 18:56.28 | syadnom_ | nickgaw, it's a little more complicated than that but yes. really, your system should have a provisioner and you can get on polycom's ztp service so boxes will connect up and get provisioning without manual configuration. |
| 18:56.49 | syadnom_ | polycom's system is a little rough though. yealink's is a ton simpler. |
| 18:58.40 | voipmonk | you can build your own |
| 18:58.49 | voipmonk | there is documentation to help you |
| 18:59.12 | nickgaw | Do any of the Polycom phones have real buttons or are they all touch screens as I am totally blind and would prefer real buttons like a normal land line phone has? |
| 18:59.32 | syadnom_ | nickgaw, many of their phones are physical buttons. |
| 19:00.00 | syadnom_ | but, there are items that are displayed on the screen and I don't know that there are any accessibility options available... |
| 19:00.19 | syadnom_ | nickgaw, what is your use case here? |
| 19:01.13 | nickgaw | That is why I was asking about preconfigured phones and ways to manually edit the xml files as I am not sure how accessible the web interface is with a screen reader does Polycom ever update the firmware and how are updates delivered and are they automatic? |
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| 19:01.24 | igcewieling1 | not all polycom VVX phones are touch screen. |
| 19:02.00 | igcewieling1 | I *HATE* touch screens. and use VVX410 which has buttons and a decent display |
| 19:02.09 | syadnom_ | updates are available in the web ui or via xml configuration. |
| 19:02.28 | syadnom_ | again, what is your use case? There might be better options that you are not exploring. |
| 19:03.25 | nickgaw | a small home based asterisk system with around 4 extentions and possibley a method with physical buttons for switching between lines or extentions with physical buttons. |
| 19:03.29 | voipmonk | nickgaw, are you still utilizing screen readers? |
| 19:04.07 | nickgaw | yes I use screen readers all of the time as I am totally blind so have no choice. |
| 19:04.21 | syadnom_ | nickgaw, you could consider using an ATA and plug in a traditional phone. Grandstream's HT812 has 4 ports so you could have 4 extensions . |
| 19:05.06 | voipmonk | do you require braille overlay for your ip phones? |
| 19:05.32 | voipmonk | there are phones with raised dimples on the keypad |
| 19:05.39 | nickgaw | and so I could just physically move the normal land line phone between the ports for the extentions but I was thinking of something where I could hear the phone ring like you can hear in office buildings with physical lines ring where the attendent answers the phone on each line. |
| 19:06.18 | voipmonk | nickgaw, do you require a different audible ring for each extension? |
| 19:06.59 | nickgaw | yes so I know what extention is ringing and no I don't need braille as long as the keypad is like a normal number pad on a phone. |
| 19:08.26 | igcewieling1 | So, if you had 100 phones, you'd need 100 different ringtones? |
| 19:08.38 | voipmonk | hehe - just 4 for now |
| 19:09.12 | nickgaw | never thought of that probably not if the phones were in different rooms but I would like one phone to be able to anser all extentions. |
| 19:09.33 | voipmonk | nickgaw, incoming... |
| 19:09.55 | syadnom_ | on a full sip phone, it's not difficult to have distinctive ring on each extension. |
| 19:10.37 | nickgaw | but can you answer any extention from any phone if the phone supports this feature? |
| 19:10.41 | voipmonk | nickgaw, ring |
| 19:11.12 | voipmonk | nickgaw, your mailbox is full. |
| 19:12.00 | nickgaw | yes I am in the process of cleaning it out now can you email me your contact information and I can give you a call when I am finished? |
| 19:12.14 | syadnom_ | nickgaw, yes. you can register 4 separate extensions on the phone and dedicate a button to each. |
| 19:13.56 | syadnom_ | I do this for customers now. Often, they have multiple departments. I set each phone with 10 ringtones and then I alter the sip header for each department. Now they can tell who's phones are ringing even if multiple calls are coming in. |
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| 19:22.32 | r1ppa | syadnom_, hey do you know what the parameters for configuring the look of the background, for instance ensuring their extension is showing , etc |
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| 19:36.28 | Bhakimi | does anybody here know of a good webrtc to stip gateway? |
| 19:36.32 | Bhakimi | *sip |
| 19:43.00 | nickgaw | Can most sip phones automatically update firmware if configured to do so so the end user does not need to do anything to keep the firmware updated? |
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| 19:48.19 | [TK]D-Fender | no |
| 19:48.44 | [TK]D-Fender | firmware always requires you to drop the new files in the proper place, most have to be told to go pick them up |
| 19:49.40 | nickgaw | For my small home setup with 4 extentions what hardware sip phone can you suggest I get? |
| 19:50.26 | [TK]D-Fender | clarify what you maen by 4 extensions |
| 19:50.29 | [TK]D-Fender | mean |
| 19:51.42 | nickgaw | like extention 100, 101, 102 and 103 each phone extention can be called from the automated attendent and the extention rings and each one has it's own mailbox for voicemails. |
| 19:51.58 | [TK]D-Fender | 4 PHONES |
| 19:52.03 | [TK]D-Fender | don't call them "extensions" |
| 19:52.19 | [TK]D-Fender | given you're blind I would recommend just going analog with ATA's |
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| 19:53.22 | nickgaw | ok phones but I am looking for something that I can answer each call where ever it rings in the house like each dialed number in the automated attendent would produce a ring to that phone but I could answer the phone from a hardware device. |
| 19:53.49 | [TK]D-Fender | huh? |
| 19:54.11 | nickgaw | and with an ATA do they make ones with more then one port so I can connect the two line phone to the device or four line to the device? |
| 19:54.26 | [TK]D-Fender | HUH?! |
| 19:54.32 | [TK]D-Fender | Your description is getting MANGLED |
| 19:55.01 | [TK]D-Fender | How many calls does each PHONE have to be able to support? |
| 19:55.01 | nickgaw | like in an office where an attendent sits at a desk and they can answer any line in the building by pressing line one or two that type of setup. |
| 19:55.17 | [TK]D-Fender | You are mangling DISTRIBUTIOn wiht PHONE CAPABILITIES |
| 19:56.16 | nickgaw | I need the phone to support at least one call at a time but with the ability to join calls together if needed or would meet me work best for that. |
| 19:56.41 | k3asd` | is it possible authenticate just one time an extension? |
| 19:58.00 | nickgaw | by support do you mean if another call comes threw would the phone still ring even if I were on another call on the same device? |
| 19:58.33 | k3asd` | I would like that just one user can use the extension 100 and nothing else uses the 100 extension at the same time |
| 19:59.12 | nickgaw | makes sence as this is what I kind of want to do. |
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