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| 06:07.57 | jmls1 | morning |
| 06:08.22 | jmls1 | pah. cross-compliation is _way_ overrated ... |
| 06:08.57 | jmls1 | hasn't got it working yet, obviously ... |
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| 07:16.21 | snuff-work | it was never meant to be easy |
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| 12:26.52 | tzafrir_laptop | AST-2012-007 applies cleanly on 1.6.2 . AST-2012-008 - with minor tweaks |
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| 12:38.05 | Deeewayne | kpfleming, hey, I was planning on asking you a question this morning regarding Asterisk rejecting calls due to media lines in the SDP |
| 12:38.21 | mvanbaak | kpfleming: RUN! |
| 12:38.45 | kpfleming | Deeewayne: there are no questions |
| 12:39.08 | file | he's going to ask about optional SRTP :P |
| 12:39.09 | mvanbaak | there is cake |
| 12:39.18 | Deeewayne | I'm trying to get SIP TLS SRTP working with Freeswith as the server and Asterisk the client http://pastebin.freeswitch.org/19191 |
| 12:39.45 | Deeewayne | file and I were discussing last night |
| 12:40.38 | kpfleming | is there a question in there? |
| 12:41.09 | Deeewayne | well, there was going to be. I was going to ask if you would consider allowing Asterisk to accept this call :-) |
| 12:41.19 | Deeewayne | but in light of your email, I guess the answer is no |
| 12:41.36 | kpfleming | i can't even see that pastebin, it requires authorization |
| 12:42.00 | Deeewayne | the username/password is on the authorization prompt - pastebin/freeswitch |
| 12:42.31 | file | I think if the offer contained a secure and unsecure stream and we were configured for SRTP then accept only the secure stream, but if we aren't configured for SRTP then reject it totally... that's my revised opinion |
| 12:43.07 | Deeewayne | so, its legal to offer RTP/AVP and RTP/SAVP (?) but you would _not_ want asterisk to try to accept this call? |
| 12:43.09 | kpfleming | optional SRTP is just wrong. |
| 12:43.30 | file | you don't want your media stream to *optionally* be secure |
| 12:43.31 | kpfleming | it's not illegal, it's two media streams. you are not offering two versions of the same stream. |
| 12:43.45 | kpfleming | what would you expect to happen if the SDP answer you got accepted *both* streams? |
| 12:44.20 | Deeewayne | ok, good question. I don't know what I'd expect there :-) |
| 12:44.29 | file | brokenness |
| 12:44.35 | kpfleming | oh, that offer is invalid as well, because the two 'm' lines are using the same port number. that's a no-no. |
| 12:45.12 | Deeewayne | thanks. I'll bring this up w/ freeswitch then as a bug |
| 12:45.31 | kpfleming | it's not unique to freeswitch though... there are a number of SIP endpoints that attempt optional SRTP this way. |
| 12:45.56 | kpfleming | there is a proper way to do it now, but nothing supports it yet. SDPCapNeg |
| 12:46.22 | Deeewayne | is that specifically for specifying two stream options to the same port? |
| 12:48.13 | kpfleming | no, it's for capability negotiation. you can't have two streams on the same port, ever. |
| 12:48.34 | kpfleming | but with SDPCapNeg, you'd offer one stream, with multiple variants to choose from. |
| 12:48.50 | Deeewayne | oh ok |
| 12:48.52 | file | would still not want optional SRTP |
| 12:49.24 | kpfleming | at least with that method, it's clearly specified in the offer and answer. but yeah... if the caller wants SRTP, and the callee declines it, then what's the point? |
| 12:49.49 | Deeewayne | ok |
| 12:51.02 | Deeewayne | kpfleming, file, thanks for your help and have a great day |
| 12:51.08 | kpfleming | sure, you too! |
| 12:51.11 | file | if I don't do I get a refund? |
| 12:51.24 | file | is that an informal contract? |
| 12:51.26 | Deeewayne | file, no - just a bowl of soup |
| 12:51.48 | Deeewayne | it was an offer to have a great day :-) |
| 12:51.50 | file | that is acceptable |
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| 13:06.31 | leifmadsen | heh |
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| 13:36.00 | kam187 | is there a way to force audio between asterisk and your channel driver to open, even if the incoming channel (eg SIP) hasn't opened early media? |
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| 14:34.28 | kam187 | hmm whats 'AST_FRIENDLY_OFFSET' |
| 14:35.50 | mvanbaak | a friendly offset |
| 14:36.22 | file | frame payloads can have some extra space at the front, which is useful for constructing headers for other protocols (like RTP) |
| 14:36.26 | snuff-home | infinite wisdom imparted their mvanbaak :P |
| 14:36.33 | snuff-home | *there |
| 14:37.02 | WIMPy | guesses negative offsets are unfriendly |
| 14:37.02 | file | AST_FRIENDLY_OFFSET is the amount of space recommended for that purpose |
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| 14:37.54 | kam187 | ahhh |
| 14:38.11 | kam187 | whats this ip_pipe stuff? |
| 14:38.16 | kam187 | *io_pipe |
| 14:38.29 | kam187 | it seems there' 2 |
| 14:38.35 | file | I have no idea what you are referring to. |
| 14:39.14 | file | if you are more specific your question may be answered |
| 14:39.47 | kam187 | sorry, my brain is just all over the place |
| 14:40.10 | kam187 | so in my channel driver (which i copied alot of the voice frame stuff from another driver).. |
| 14:41.05 | kam187 | i see that the modem_tech.write method passes my driver a frame from asterisk |
| 14:41.34 | kam187 | i pass this over to my hardware.. at the same time, i read a frame from my hardware and put it into pvt->io_pipe[1] |
| 14:42.15 | kpfleming | that's the code you copied, we can't help you with that |
| 14:42.43 | kam187 | in the modem_tech.read, where asterisk asks for a frame, i'm writing to pvt->io_pipe[0] |
| 14:42.51 | kam187 | ok, i thought u might have some idea |
| 14:42.55 | kam187 | nvm i'll figure it out |
| 15:06.02 | russellb | chan_modem was probably a poor choice to copy. |
| 15:06.44 | russellb | that code was removed from asterisk ages ago |
| 15:07.11 | WIMPy | Oh. wow. Where did you dig that out? Or has someone revived it? |
| 15:07.29 | russellb | i was guessing that's what kam187 copied, since he was referencing modem_tech |
| 15:08.51 | russellb | looks like the last version that was in was 1.2 |
| 15:09.03 | WIMPy | That's what I thought. |
| 15:09.20 | WIMPy | Although it would still be usefull. |
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| 15:10.08 | WIMPy | OTOH, I geuess chan_datacard/chan_dongle are the same, just with some added functions. |
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| 15:20.06 | kam187 | russellb: yeah thats what i copied, although i've changed nearly all of it. the only part i dont understand is this pipe stuff for the voice |
| 15:20.32 | kam187 | but i'm running my dsp function on the raw data from the hardware instead now so i know exactly whats going on |
| 15:33.16 | acidfoo | if I place a channel on hold { ast_queue_control(channel, AST_CONTROL_HOLD); } isn't suppose to stop the ast_channel_tech .read callback to be called ? |
| 15:41.57 | russellb | no |
| 15:52.31 | acidfoo | ok :) |
| 16:00.37 | kam187 | actually, that reminds me of another question! |
| 16:01.01 | kam187 | when do the .read and .write callbacks actually fire? Does it depend on the other bridged channel opening some voice? |
| 16:01.23 | kam187 | for example sip -> asterisk -> my channel driver.. if sip doesnt open early media, will .read and/or write still fire? |
| 16:01.28 | kam187 | assuming i sent progress etc? |
| 16:16.00 | kpfleming | your read and write callbacks will only be called to get or give media frames to your channel driver |
| 16:17.16 | kam187 | right, but if the sip side doesnt open audio, will they never fire? even if i send progress to asterisk? |
| 16:18.02 | kam187 | or will they still fire, and asterisk decides what to do with them i.e. pass them onto sip if media open, or throw them away |
| 16:18.02 | kpfleming | i don't know what 'open audio' means, but your write callback will *only* be called to send you media frames. period. |
| 16:18.17 | kpfleming | if you indicate on your channel's file descriptor that you have media available to be read, your read callback will be called |
| 16:20.41 | kam187 | ahh |
| 16:20.49 | kam187 | how do i indicate in on the channel descriptor? |
| 16:20.53 | kam187 | *it |
| 16:21.20 | kpfleming | you placed a file descriptor into one of the elements of the channel's fd[] array, right? |
| 16:22.28 | kam187 | yeah |
| 16:23.01 | kpfleming | then whatever that fd connects to needd to become 'ready for reading' |
| 16:23.14 | kam187 | thats the io_pipe thing i was taking about |
| 16:23.18 | kpfleming | that will cause the poll() in ast_waitfor() to wake up, and ast_read to be called on your channel |
| 16:23.27 | kam187 | ahh |
| 16:23.59 | kam187 | ok thats fine then |
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| 17:22.45 | rgagnon | kpfleming: Hella-FLOP? Is that like the Facebook IPO? |
| 17:47.06 | kpfleming | rgagnon: lol |
| 18:01.04 | acidfoo | erm, what is the opposite function of ast_channel_set_fd ? if there is one. thank you |
| 18:01.22 | acidfoo | im looking into main/channel.c |
| 18:01.59 | kpfleming | you need to look in channel.h, where the function prototypes live. especially in trunk, where many of the ast_channel API functions are generated by macros. |
| 18:16.01 | acidfoo | I don't find anything that look like ast_channel_unset_fd ;/ |
| 18:16.49 | russellb | guess nobody has ever needed one, heh |
| 18:17.13 | file | set it to -1 ^_^ |
| 18:17.17 | russellb | cleanup is overrated |
| 18:17.45 | acidfoo | file, then free(ead) will never be called I guess |
| 18:18.09 | acidfoo | ah, no |
| 18:18.09 | acidfoo | it will |
| 18:18.09 | acidfoo | file, true |
| 18:22.28 | russellb | sets file to -1 |
| 18:22.54 | file | feels so... negative |
| 18:26.25 | russellb | file: how could you, you're unsigned |
| 18:26.30 | file | oh snap |
| 18:31.34 | kpfleming | sets file to UINT_MAX |
| 18:35.11 | file | :( |
| 18:35.50 | russellb | sets file to ULLONG_MAX |
| 18:36.02 | russellb | that's a large number. |
| 18:36.39 | russellb | good talk |
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| 19:18.35 | kam187 | kpfleming: just tested the audio to the channel driver thing. and unless the sip side opens audio, asterisk doesn't open it to the channel driver :/ |
| 19:18.52 | kpfleming | i don't know what 'open' means at all. |
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| 19:19.21 | kam187 | opens rtp audio to the asterisk |
| 19:19.30 | kpfleming | that still does not make any sense |
| 19:19.39 | kpfleming | you need to use terms that actually match what the code does |
| 19:20.20 | kam187 | well when i make a call over sip, to asterisk, it negotiates a rtp port to send the media on, typically after a 183 message or connect they would open an audio channel between each other |
| 19:21.07 | kpfleming | no... you are confusing things. |
| 19:21.27 | kpfleming | each end of a SIP call assigns an RTP port to be used for media transfer. this is not 'negotiated'. |
| 19:21.48 | kpfleming | this information can be carried in an INVITE, and ACK, or a 1xx/2xx response |
| 19:21.50 | kam187 | yeah ok, thats not really the issue |
| 19:22.18 | kpfleming | so you need to describe in SIP protocol terms what 'not opening audio' means |
| 19:22.30 | kam187 | until rtp opens between the sip client and asterisk, asterisk won't open it to my channel driver |
| 19:22.50 | kpfleming | i give up... |
| 19:23.04 | kam187 | i dont really know what u dont get |
| 19:23.20 | kpfleming | the word 'open' is not relevant here, at all. |
| 19:23.27 | kpfleming | asterisk does not 'open' anything to your channel driver. |
| 19:23.49 | kam187 | ok, well it doesn't ask me for any audio, it doesnt fire the .read call back |
| 19:24.09 | kpfleming | it will not call the read call back until your channel driver tells it that there is audio to be read |
| 19:25.28 | kam187 | ok, so how do i tell it? |
| 19:25.30 | kpfleming | this has nothing at all to do with what any other channel is doing or not doing |
| 19:25.32 | kam187 | i thought i had already |
| 19:25.44 | kpfleming | i told you this before. you need to make your file descriptor that it will read audio from be 'ready' for reading. |
| 19:25.50 | kam187 | it is |
| 19:26.11 | kpfleming | then you've done something else wrong in your code. this has nothing to do with the SIP channel at all. |
| 19:26.26 | kam187 | ok |
| 19:28.14 | kpfleming | you can verify this by using the CLI 'originate' command to open a channel into your channel driver and send it into the dialplan where you run Echo(). until you have this working, and audio flowing in both directions, there's no need to complicate things by adding in SIP channels. |
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| 19:42.37 | kam187 | hmm yeah i see the problem |
| 19:43.26 | kam187 | the way the driver is written, when .write is called, i read from the hardware and write it into the pipe (fd). asterisk polls that and on .read grabs the data.. |
| 19:43.41 | kpfleming | well, that won't work if you aren't being sent data from asterisk |
| 19:43.49 | kam187 | essentiall the .write from asterisk is used to time the actual read of frames from my hardware |
| 19:43.57 | kpfleming | that won't be very reliable |
| 19:43.59 | kam187 | yep, and thats the problem |
| 19:44.11 | kpfleming | your hardware needs to tell you when it has data ready, and you need to tell asterisk so it can be read |
| 19:44.38 | kam187 | its sending to me over a serial port |
| 19:46.49 | mjordan | kam187: you need a thread independent of the channel driver callbacks that is responsible for reading data off of the serial port |
| 19:47.06 | kam187 | mjordan: yeah that's what i'm thinking about now :/ |
| 19:47.38 | kpfleming | and it will need to buffer up data until it has a full frame, then indicate to asterisk that a frame is ready. |
| 19:48.24 | file | you could probably use the ast_timer API at a 20ms interval... provided your hardware will work with that... then every 20ms your read function would be called and you could read data in |
| 19:48.48 | kpfleming | file: sneaky |
| 19:48.51 | jmls1 | ok, I'll bite. What's res_http_websocket for ? |
| 19:49.02 | file | kpfleming, *bows* |
| 19:49.04 | WIMPy | Hmm. That reminds me of something. |
| 19:49.12 | file | jmls1, http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ |
| 19:49.13 | WIMPy | Is that a fixed 20ms callback? |
| 19:49.20 | file | WIMPy, it is what you specify |
| 19:49.56 | WIMPy | Because I had massive audio issues when trying to run ConfBridge with 10ms frames. |
| 19:50.07 | jmls1 | hmmmm |
| 19:50.21 | jmls1 | thinks of things to do with that ;) |
| 19:50.47 | file | jmls1, one thought that crossed my mind was a javascript based queue monitoring solution |
| 19:51.11 | jmls1 | is big into stomp with websockets and node.js |
| 19:51.22 | jmls1 | [and * of course .. ] |
| 19:51.54 | jmls1 | I'm using node to read the user events / cel events and convert them into stomp messages |
| 19:52.07 | pabelanger | file: how would you do that? app_queue would send events over websockets? |
| 19:52.25 | file | pabelanger, haven't thought about it that much - was just a thing that came to mind that someone may toy with |
| 19:52.29 | Juggie | svn rm app_queue.c |
| 19:52.55 | jmls1 | pabelanger: use a user event to pass to node to pass to stomp to pass to a jms ;) |
| 19:54.20 | pabelanger | jmls1: how does node connect to asterisk? AMI? |
| 19:54.22 | Juggie | it would be silly to do any work on queues and not consider the clustering aspect |
| 19:54.41 | jmls1 | pabelanger: yup. Just sits there reading the data stream |
| 19:55.51 | pabelanger | Ya, one of the guys at shopify did a prototype called Q. Using their software batman.js. I think it is the same idea and software involved |
| 19:56.05 | pabelanger | not that I want to work with javascript |
| 19:56.13 | Juggie | :) |
| 19:56.18 | jmls1 | and because node.js can also be a stomp client, it can respond to messages and instruct asterisk to "do things" |
| 19:56.22 | Juggie | i'm not sure I buy into the node.js thing |
| 19:56.56 | jmls1 | i didn't until I started really playing with it |
| 19:57.04 | pabelanger | haha, I googled node.js and this was the 3rd hit: Node.js is Cancer |
| 19:57.06 | jmls1 | thinks that sounds .. very .. creepy |
| 19:57.31 | Juggie | the idea of leaving web sockets open indefinitally until the server finds time to reply and then update the website |
| 19:57.33 | Juggie | seems wrong |
| 19:57.57 | kpfleming | that's already being done using long polling, COMET, BOSH, ec. |
| 19:58.06 | kpfleming | there's no other way to push data to a web browser app |
| 19:58.21 | jsmith | Exactly... websockets is just a more formalized way of doing that more efficiently |
| 19:58.28 | kpfleming | far more efficiently |
| 19:58.28 | jsmith | at least, from what little I know |
| 19:59.07 | Juggie | can you actually handle data when its return'ed w/o closing the websocket? |
| 19:59.15 | kpfleming | yes |
| 19:59.17 | file | of course |
| 19:59.18 | Juggie | or once you get your chunk of data do you close process and reopen |
| 19:59.23 | kpfleming | it's a bidirectional message pipe |
| 19:59.45 | jmls1 | doesn't have to be websockets. Node reads the ami data, sends a stomp message, and *any* stomp client can then process it - and there are stomp clients for any language |
| 20:00.00 | kpfleming | not PL/1, or bash :-) |
| 20:00.01 | Juggie | i see... |
| 20:00.07 | pabelanger | so stomp is basically a proxy? |
| 20:00.16 | Juggie | node.js though is that necessairy |
| 20:00.27 | Juggie | why are we running a server in javascript that runs on chrome's js engine |
| 20:00.33 | jmls1 | stomp is the protocol |
| 20:00.36 | Juggie | that is what it is right? |
| 20:00.49 | jmls1 | yeah, a bloody fast js engine |
| 20:01.01 | kpfleming | V8 does not belong to chrome, but chrome uses it |
| 20:01.08 | kpfleming | so do lots of other things (including Digium phones) |
| 20:01.38 | jmls1 | kpfleming: really ... ? |
| 20:01.38 | Juggie | node.js should update their site then as they state they run on Chrome's JS runtime. |
| 20:01.58 | kpfleming | jmls1: yes, the UI and app environment inside Digium phones is JavaScript running on V8 |
| 20:02.01 | russellb | ugh @ Node.js |
| 20:02.11 | jmls1 | cool |
| 20:02.43 | Juggie | i guess its a love or hate thing :) |
| 20:04.25 | Juggie | has anyone been doing any work on queues related to clustering, etc. queues shared over multiple boxes and so on? |
| 20:04.41 | pabelanger | asteriskscf? |
| 20:04.44 | pabelanger | :p |
| 20:05.13 | Juggie | I have zero knowledge on that product. |
| 20:06.30 | Juggie | --The design section of this space is undergoing massive and pervasive updating. Please check with Digium's Asterisk SCF team before making decisions based on the information herein. We expect the updates to be completed before June 2012. Of course, if it is past June 2012 and this note is still here, please feel free to point it out and inform us that we should "hurry up". |
| 20:06.31 | Juggie | lol :) |
| 20:06.49 | kpfleming | yeah, that. |
| 20:07.06 | kpfleming | also, send truckloads of cash. we like that. |
| 20:07.18 | opticron | looks at the calendar |
| 20:07.24 | jmls1 | thought that you guys always wanted Dr Pepper ? |
| 20:07.34 | kpfleming | we can buy it with cash |
| 20:07.35 | Juggie | Dr. Pepper doesnt pay the bills.... |
| 20:07.41 | jmls1 | Cash ? Why didn't you say so. I had loads of that |
| 20:07.49 | jmls1 | before I got married ... |
| 20:08.27 | Qwell | get unmarried then - I want me some cash |
| 20:08.29 | kam187 | wow that was easier than i thought |
| 20:08.38 | kam187 | got a thread ast_log every 1 second (as a test) |
| 20:08.52 | jmls1 | getting unmarried is a sh*t load more expensive than getting married |
| 20:08.56 | jmls1 | trust me |
| 20:08.59 | jmls1 | :( |
| 20:09.09 | Qwell | jmls1: Oh, I know... |
| 20:09.23 | Juggie | Qwell, rob the roof guy's money |
| 20:09.25 | file | I will enjoy my marriage free and kid free life as long as I can |
| 20:09.41 | kpfleming | file: have you started building the addition to hold your bags of money yet? |
| 20:09.46 | jmls1 | file: live life to the full man, live it to the full :) |
| 20:09.54 | file | kpfleming, I outsource that |
| 20:10.09 | russellb | runs in circles |
| 20:10.20 | kpfleming | blast from the past, there |
| 20:10.27 | russellb | iknowright? |
| 20:10.35 | Qwell | getting too old to run in circles, these days |
| 20:10.46 | russellb | i run in far fewer circles now |
| 20:11.52 | Juggie | I'm happy being DINK's for now... 2 incomes no kids is swell :P |
| 20:12.35 | russellb | urban dictionary says that means "Local slang in Vermont. A stupid person, a jerk, an a-hole. A general putdown of one's abilities. |
| 20:12.44 | russellb | glad you're happy to be a dink. |
| 20:12.59 | Juggie | lol |
| 20:13.11 | Juggie | it doesn't have DINK = Dual Income No Kids? |
| 20:13.25 | russellb | it may ... and if it did, i would choose to ignore it, because it's not as funny |
| 20:13.30 | Juggie | ic |
| 20:13.48 | Juggie | i see your point! |
| 20:15.20 | Juggie | lol, its actually #1 and your vermont slang is #2 :) |
| 20:15.51 | Qwell | urban dictionary is editable. I'm calling wikipedia-style shenanigans. |
| 20:16.45 | jmls1 | Qwell: is it you with the large rabbits ? |
| 20:17.29 | *** join/#asterisk-dev tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:17.29 | *** mode/#asterisk-dev [+o tzafrir_laptop] by ChanServ |
| 20:19.16 | Juggie | hmmm Certified Asterisk.... wow I am way behind |
| 20:19.28 | Juggie | is this a replacement for the old business edition? |
| 20:20.33 | jmls1 | was certified a long time ago. |
| 20:20.47 | Qwell | certifiable* |
| 20:20.48 | jmls1 | but I escaped :-D |
| 20:20.50 | file | there's blog posts about it, just read them :P |
| 20:20.59 | file | and wikiness |
| 20:21.25 | Juggie | file, ya thats where i saw it... was reading kevins blog :P |
| 20:22.50 | Juggie | guess I am a little out of touch... thats what happens when you manage a team of web programmers :P |
| 20:23.09 | kpfleming | find an RSS reader, they are really useful :-) |
| 20:23.27 | Juggie | i use google reader actually |
| 20:23.32 | Juggie | will have to subscribe |
| 20:23.44 | kpfleming | Asterisk Daily News is your friend |
| 20:24.17 | kpfleming | even though it's published from New Zealand |
| 20:24.24 | Juggie | haha |
| 20:24.31 | *** join/#asterisk-dev Tim_Toady (~fuzzy@62.1.174.66.dsl.dyn.forthnet.gr) |
| 20:25.19 | tzafrir_laptop | There were some questions in the blog post that were not really answered |
| 20:25.40 | tzafrir_laptop | From what I understand: it includes extra fixes to properly work with the Digium phones? |
| 20:25.56 | Qwell | tzafrir_laptop: features to support DPMA |
| 20:25.58 | kpfleming | it includes extra features to support Digium phones |
| 20:25.59 | tzafrir_laptop | There weren't that many changes in the "cert" branches |
| 20:26.10 | kpfleming | yes there are |
| 20:26.21 | kpfleming | but the first commit to that branch included all of them |
| 20:26.29 | tzafrir_laptop | Sorry: s/features/commits/ |
| 20:26.44 | kpfleming | if you run a diff between the 1.8.11-cert branch and the 1.8.11 tag, you'll see them |
| 20:27.13 | tzafrir_laptop | Why can't those changes be merged into 1.8 proper? |
| 20:27.21 | kpfleming | this is documented on the wiki pages |
| 20:40.20 | kam187 | awesome! it works :) |
| 20:40.37 | kpfleming | cool |
| 20:44.35 | *** join/#asterisk-dev leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:44.35 | *** mode/#asterisk-dev [+o leifmadsen] by ChanServ |
| 20:55.11 | kpfleming | putnopvut: so that ended up being like 8 lines of code. |
| 20:56.14 | *** join/#asterisk-dev wedhorn (~wedhorn@123-243-209-34.static.tpgi.com.au) |
| 21:04.59 | putnopvut | kpfleming: yep :) |
| 21:05.38 | mjordan | otherwiseguy: in review 1873, did you need to update the configure script? |
| 21:14.50 | *** join/#asterisk-dev jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 21:15.05 | *** part/#asterisk-dev mjordan (~mjordan@nat/digium/x-xhdspkzvphsulajq) |
| 23:13.03 | snuff-work | i found an interesting looking kickstarter project around an IDE concept.. |
| 23:13.08 | snuff-work | http://www.kickstarter.com/projects/ibdknox/light-table?ref=discover_pop |
| 23:13.14 | snuff-work | pretty snazzy |
| 23:26.56 | citywok | that looks like it could be pretty slick |
| 23:27.21 | citywok | i recently switched to intelli-j and have been so much happier than i was with eclipse, that looks like it could be even more powerful. |
| 23:34.11 | kam187 | is there a way to dial the same number out of x channels in a group? |
| 23:35.56 | kam187 | i see u can specify multiple channels using & but can you use a group and limit it to say 5 channels in that group? |
| 23:37.04 | WIMPy | Not exactely sure about your question, but dialling a group usually means that the channel tries to find one free channel within that group. |
| 23:37.16 | WIMPy | But that's up to the channel. |
| 23:37.53 | kam187 | yeah, but say i want to dial for example 100, out of 5 channels in that group |
| 23:38.21 | WIMPy | Dial that group five times. |
| 23:38.40 | WIMPy | Or build something in to that channel. |
| 23:41.08 | kam187 | ahh so dial(mychannel/r1/${EXTEN}&mychannel/r1/${EXTEN}&mychannel/r1/${EXTEN}) for example? |
| 23:41.21 | WIMPy | yes |
| 23:41.59 | WIMPy | Or you make your channel accept something like Dial(mychannel/5*r1/${EXTEN}) |
| 23:42.06 | WIMPy | Or whatever. |
| 23:42.15 | kam187 | cool |
| 23:43.47 | WIMPy | I did a patch for chan_lcr to accept a + in front of the port(list) to make it dial all instead of any port. |
| 23:49.01 | kam187 | hmm whats this ' asterisk.c:976 netconsole: read returned 0' |
| 23:49.23 | kam187 | oh cool, is that in the asterisk trunk? |
| 23:53.56 | kam187 | hmm Skipping dialing interface 'mychannel/r1/100' again since it has already been dialed |