| 00:08.29 | *** join/#asterisk bbryant1 (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 00:09.19 | p3nguin | wimpy: Is there any real reason to have more than one group of channels? |
| 00:17.03 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:19.23 | WIMPy | What do you mean? |
| 00:20.13 | WIMPy | One type of group? One group per ....? |
| 00:22.06 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
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| 00:29.55 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 00:38.32 | p3nguin | I mean, like, if I have 100+ channels, is there any reason to have more than one group? I could have group 0 consisting of all the available channels, couldn't I? |
| 00:40.59 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-165.mobile.uci.edu) |
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| 01:05.18 | WIMPy | You probably would have more that one provider if you have 100+ channels, wouldn't you? |
| 01:06.54 | WIMPy | And even if you only have one, you might want to reserve some chnnels for some people/activities. |
| 01:07.07 | p3nguin | No. I was thinking in the case of that guy who was here recently with over 200 channels trying to make 200 concurrent calls. |
| 01:08.28 | WIMPy | Generally no, but as he was too fast for his telco, interleaving the interfaces might make sense. |
| 01:08.58 | p3nguin | He was only able to make one single call at a time with all those channels. |
| 01:09.22 | p3nguin | I don't know if he ever got that fixed or not. |
| 01:09.30 | WIMPy | Nah, he made a lot more before. |
| 01:10.12 | p3nguin | When I asked him about making more than one call at a time from phones, he said only one at a time. |
| 01:10.26 | WIMPy | He was so fast that he got RNR and even REJ frames. |
| 01:10.49 | p3nguin | What's the solution for that? Add more time between Dial()s? |
| 01:10.57 | WIMPy | Yes. |
| 01:11.09 | p3nguin | How much delay is generally required? |
| 01:11.26 | WIMPy | Or my suggestion was not to use the channels sequentially, but interleave the interfaces. |
| 01:11.35 | p3nguin | How is that done? |
| 01:11.47 | WIMPy | That depends on the other end. |
| 01:12.18 | WIMPy | He had one group per interface. So just using the groups sequentially whould have done that. |
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| 01:26.29 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 01:29.11 | andygraybeal | i'd like to get a 66 block for the phone lines coming in.. phone lines coming in - one is our fax, one is our backup credit card (split into 3 for 3 different credit cards) one both our two different dsl lines. i'd like to get a block with modular jacks on it-- what do you guys recommend? maybe something entirely different? should i go with 110 |
| 01:29.42 | andygraybeal | it's not big and i doubt we'd expand much |
| 01:29.51 | p3nguin | Good old digital subscriber line lines. |
| 01:30.07 | andygraybeal | yes, dsl :) why do you say it like that? |
| 01:30.23 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:30.30 | p3nguin | dsl lines == digital subscriber line lines |
| 01:31.06 | andygraybeal | aah yes, i see |
| 01:31.19 | andygraybeal | it's like so many other things i say :) |
| 01:31.38 | p3nguin | atm machines, pin numbers, pcb boards? |
| 01:32.54 | p3nguin | At any rate, I don't see a problem with 66 for your circuits. |
| 01:34.44 | andygraybeal | yea, i guess, i want those little modulear jacks on the sides of the 66 block, i got a plain 66 block.. and i'm wondering if i can add the modular things to it? i'm new to this. |
| 01:35.33 | p3nguin | I've just mounted surface-mount modular jacks next to the block before. I probably have a pic of one to illustrate. |
| 01:35.50 | WIMPy | A patch panel? |
| 01:37.36 | andygraybeal | WIMPy, yea, like turning it into a patch panel |
| 01:37.49 | andygraybeal | p3nguin, i wonder what produc i should purchase |
| 01:38.25 | andygraybeal | i've seen like the modular jacks to the side of the 66 block in some 'how to' video but i haven't found any online yet. |
| 01:38.26 | sunfone | Keep the 66 block and get some surface mount RJ14 jacks |
| 01:38.40 | sunfone | punch down the RJ14 jacks to your block |
| 01:38.43 | sunfone | very cheap, very reliable |
| 01:39.08 | sunfone | (since you only have four or five) |
| 01:39.12 | p3nguin | http://imagebin.org/170344 |
| 01:39.30 | sunfone | exactly |
| 01:39.47 | andygraybeal | aaaah.. badness!! |
| 01:39.54 | andygraybeal | awesome. |
| 01:40.21 | p3nguin | In this image I had to do a hot-cut from an old DSL circuit to a new one, so I simply added a new jack and switched the plug to the other jack when they said "Go!" |
| 01:40.21 | sunfone | Whats the funky RJ45 yellowness penguin? |
| 01:40.54 | sunfone | homemade patch panel? :) |
| 01:40.56 | p3nguin | I didn't put in the yellow cable. My cable was white. |
| 01:41.05 | WIMPy | Why not just use a patch panel? Isn't that what they're there for? |
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| 01:41.28 | p3nguin | I'm not familiar with patch panels with RJ-12s on them. |
| 01:42.00 | p3nguin | They probably do exist, though. |
| 01:42.15 | sunfone | I think any RJ45 patch panel would work for that |
| 01:42.32 | p3nguin | It would, if you don't mind mixing your connectors like that. |
| 01:42.38 | sunfone | sure |
| 01:42.45 | WIMPy | What's wrong with "RJ45"? The other end would be that anyway. |
| 01:43.08 | p3nguin | People don't plug 8P8C plugs into phones. |
| 01:43.09 | sunfone | Not to put down your homemade patch panel... its cute :) |
| 01:43.45 | WIMPy | No, but into patch panel that terminate the outlets around the house. |
| 01:44.11 | p3nguin | I don't get it. |
| 01:44.12 | WIMPy | And if your phone has snother plud, you use a special cable there. |
| 01:44.38 | sunfone | I just left a site where the "IT Guy" made splitters to run two ethernet links through a single CAT5E run - they were pulled apart and finger twisted together, then wrapped in shrink wrap tubing on both sides. |
| 01:45.05 | sunfone | I should have taken a picture |
| 01:45.41 | p3nguin | When you arrive at a site where someone else did things their own way, you sometimes have to get creative on how to do your job. |
| 01:45.53 | sunfone | indeed |
| 01:46.08 | sunfone | I treat these occasions as an opportunity... to rip it all out and start over |
| 01:46.55 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 01:47.36 | p3nguin | That's bizarre that anyone would do that to Cat 5 cable. |
| 01:47.40 | andygraybeal | i just want the next guy not to hate me.. that is all. |
| 01:48.18 | sunfone | andygraybeal: if you do it the way p3nguin suggested, you will be doing it the "standard" way and no one could blame you :) |
| 01:48.38 | andygraybeal | awesome, thank uou for the help. |
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| 01:50.20 | andygraybeal | i'm excited now. |
| 01:50.40 | andygraybeal | this has been bothering me and i have not known how to approach anyone |
| 01:51.09 | p3nguin | I guess if you wanted to use a regular Ethernet patch panel, you could. It seems like a waste to me, though. |
| 01:51.37 | andygraybeal | i got like 5 24 port patch panels.... i think i'[m only using 3 |
| 01:51.38 | p3nguin | Modular jacks are only $5 each. |
| 01:51.41 | andygraybeal | if that |
| 01:51.55 | andygraybeal | bougt'em on ebay for cheap |
| 01:52.11 | andygraybeal | cat5e.. no back brace.. which i didn't realize.. but oh well. |
| 01:52.31 | andygraybeal | well i mean, i knew they were cat5e... i didn't realize they didn't come with a back brace. |
| 01:52.53 | andygraybeal | but, where i got the phones, there's very little room |
| 01:53.55 | andygraybeal | it's between two air exchangers... it's totally lame, but it's the only out of the way place in the area that has space; the spot where this stuff used to live leaks badly in the winter time with water |
| 01:54.13 | andygraybeal | they've replace our phone system before because it's shorted out |
| 01:54.31 | andygraybeal | and.. this is before i got there |
| 01:54.51 | andygraybeal | it's gotta get moved somewhere else for sure; there's plastic covering it for now |
| 01:56.54 | *** join/#asterisk maxagaz (~maxagaz@220.231.37.106) |
| 01:59.22 | maxagaz | hi |
| 01:59.41 | maxagaz | what is the difference between BID and DID lines ? |
| 02:01.45 | p3nguin | ~bid |
| 02:01.47 | p3nguin | ~did |
| 02:01.47 | infobot | did is probably Direct Inward Dialing, or just a phone number |
| 02:03.26 | maxagaz | is it specific to my provider ? |
| 02:05.55 | p3nguin | If BID isn't known by infobot and BID isn't in this telecom dictionary, you'll have to tell me what it is. |
| 02:08.17 | p3nguin | BID - A SID allocated for accounting purposes. BID's are allocated by Cibernet |
| 02:08.25 | p3nguin | Is that what you're talking about? |
| 02:10.43 | dym | mornings. |
| 02:10.50 | maxagaz | p3nguin, someone from china telecom came and told me I had the choice between DID and BID, then nothing was clear, at the beginning, BID was for 30 lines with one number for all, and DID 30 and as many numbers as needed, but now it seems DID is actually 100 lines |
| 02:11.15 | andygraybeal | p3nguin, that back mount for the 66 block on the plywood; i don't know if mine has that ability, i know i don't have the backmount, and i imainged the 66block would be flush against the plywood; can you explain to me if this is an okay thing to do? or should i find out how to use the backmount like you have? |
| 02:11.24 | WIMPy | Lines or numbers? |
| 02:15.21 | p3nguin | andygraybeal: Take a look at your 66 block. How's the back look? Any reason it wouldn't be safe to flush mount it instead of having it raised? |
| 02:15.55 | andygraybeal | not at all; it's straight plastic, and on the top left, and bottom right there are holes in the chasis to screw through |
| 02:16.19 | andygraybeal | to attach |
| 02:16.37 | p3nguin | Sounds like it is meant to be mounted flush against a wall to me. |
| 02:17.05 | andygraybeal | okay cool that's what i imagined. thank you for explaining. |
| 02:17.17 | andygraybeal | it's heavy for what it is :) |
| 02:17.24 | andygraybeal | i was surprised, happily |
| 02:18.41 | p3nguin | http://imagebin.org/170350 |
| 02:19.05 | p3nguin | Take a look at these. You can see how there is a "bracket" and a punch block on top. |
| 02:19.20 | p3nguin | But you can also see that the blocks have their own tabs for screws. |
| 02:20.51 | andygraybeal | p3nguin, okay thank you, looking |
| 02:21.26 | p3nguin | The block actually snaps onto the brackets, which leaves space behind it to run the wires neatly. |
| 02:22.09 | andygraybeal | intense, i see it's a method for organization. |
| 02:22.27 | andygraybeal | okay, thank you for helping me understand. |
| 02:22.37 | andygraybeal | very intense. |
| 02:23.18 | andygraybeal | i hope that shiaz is labeleled... omg. |
| 02:23.27 | p3nguin | With all the blue and white pairs on the outside, it doesn't really show how wiring goes under it. Let me grab another image. |
| 02:24.02 | p3nguin | http://imagebin.org/170351 |
| 02:24.05 | andygraybeal | ye, more picks the better; i wonder where i can find that same mount, beacuse that looks exaclty like the block i have |
| 02:24.17 | p3nguin | This one shows how the wires go under the blocks, behind the brackets. |
| 02:25.05 | andygraybeal | how long does imaginebin hold the data? can i reference it in an email showing my workmates? |
| 02:25.26 | andygraybeal | meb, i should save it to our box |
| 02:26.02 | p3nguin | two weeks |
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| 02:27.24 | andygraybeal | k cool, i'll save right now "66block images" folder :) |
| 02:28.13 | andygraybeal | thank hyou |
| 02:29.15 | p3nguin | I hope you don't have to deal with that many wires. |
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| 02:32.24 | andygraybeal | hahaa, yea, that is a full time job; we got maybe what.. 6 lines... total :) |
| 02:38.07 | andygraybeal | okay, please don't make fun of me... but our two phone lines (4 wires each) from outside, are run on an outside rated cat5 cable; is this okay? |
| 02:38.16 | andygraybeal | or should i have done something different? |
| 02:38.41 | andygraybeal | i mean, run from outside, to the inside ... which will go to the 66 block |
| 02:42.29 | p3nguin | I run phone wires over the pairs in Cat 5... just look at that first image where I have two modular jacks., |
| 02:42.57 | andygraybeal | thanks p3n |
| 02:44.08 | p3nguin | Just use blue and blue/white for the first pair, and orange and orange/white for the second pair. |
| 02:45.08 | p3nguin | That equates to your red green pair, and to the yellow black pair. |
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| 02:48.24 | andygraybeal | awesom,e thank you. |
| 02:48.33 | p3nguin | You can wire two 6p2c onto those pairs in your Cat 5 if you want two separate plugs, or you can wire both pairs into one 6p4c if you want two lines on one phone cord. |
| 02:49.09 | p3nguin | Or wire them into a modular jack or two. |
| 02:49.20 | p3nguin | It's all up to you, really. |
| 02:49.39 | andygraybeal | yea, okay... i'm still digesting your sentance. |
| 02:49.49 | p3nguin | oh |
| 02:50.16 | p3nguin | I don't have any images of two plugs on a Cat 5 for phone wiring. |
| 02:50.35 | andygraybeal | that's okay, i wondered a little, but i understand. |
| 02:51.08 | p3nguin | You know what the 6p2c or 6p4c plugs are, right? ... the plugs that go into your RJ11 and RJ14 jacks. |
| 02:51.18 | andygraybeal | i hate to sound this dense and apologetic |
| 02:51.49 | andygraybeal | okay, i know what rj11 is, atleast i think. i', not sure what rj14.. lemem wikipedia |
| 02:51.49 | p3nguin | Don't worry about it. Just ask. |
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| 02:52.28 | p3nguin | The RJ11 is the 2-wire jack, and the RJ14 is the four-wire jack. |
| 02:52.47 | andygraybeal | well, then this time i've been calling rj14, rj11. |
| 02:53.09 | andygraybeal | *all this time |
| 02:53.30 | p3nguin | Many people use the terms synonymously, when they are really different things. |
| 02:54.25 | p3nguin | If you grab a phone and look in the jack, it probably only has two conductors in it. That's an RJ11 and the correct plug for it is the 6P2C. |
| 02:55.32 | p3nguin | Or, to be more technical, the RJ11 is on the wall side, and the jack in the phone has another name. |
| 02:56.35 | andygraybeal | aah okay |
| 02:57.20 | p3nguin | Oh, cool... the wikipedia page for modular connectors has a chart/diagram that shows about using Cat 5 pairs for phones. |
| 02:57.28 | p3nguin | about half way down the page. |
| 02:57.36 | andygraybeal | loks |
| 02:58.15 | p3nguin | Compare "twisted pair colors" to "old colors." |
| 02:58.55 | p3nguin | That chart's a good reference. |
| 03:00.12 | p3nguin | I should print that on an index card and throw it in my bag. |
| 03:04.02 | _Shadowfax_ | anybody here have dahdi-linux working with RHEL 6.1? i can complie it but not load the module. |
| 03:05.06 | _Shadowfax_ | insmod: error inserting '/lib/modules/2.6.32-131.6.1.el6.x86_64/dahdi/dahdi.ko': -1 Unknown symbol in module |
| 03:09.49 | p3nguin | andygraybeal: While all this wiring may be off the topic of Asterisk, it's something I enjoy doing and don't mind talking about it if it's not causing problems for others... so feel free to stop in if you have more questions when you start getting into the wiring. |
| 03:10.30 | andygraybeal | cool, thanks. i figiure this is the only place to chat about it online. |
| 03:10.40 | andygraybeal | there is no #telephony :) i tried first. |
| 03:11.27 | andygraybeal | i'm wstill reading 'registerest jacks' in wikipedia |
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| 03:13.51 | p3nguin | Hardly anyone sticks to the standards of registered jacks when talking about the stuff. |
| 03:14.17 | p3nguin | Just look at our computers... we call them RJ45 when we use them for Ethernet between a computer and a switch. |
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| 03:33.20 | andygraybeal | p3nguin, :) |
| 03:34.00 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:36.39 | p3nguin | Just sayin'. |
| 03:39.55 | p3nguin | Now if I could just find someone that knows how to sync the time on an iPod touch with a connected computer, I could call it a day. |
| 03:42.44 | andygraybeal | p3nguin, :) |
| 03:42.52 | andygraybeal | ntpclient on ipod sucks? |
| 03:43.09 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:43.10 | p3nguin | There apparently is a huge lack of ntp clients on all Apple products. |
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| 03:43.51 | andygraybeal | like... them hippies... at apple to do that :) |
| 03:43.56 | andygraybeal | eff the time ! |
| 03:47.08 | p3nguin | I see an app that is supposed to do it. |
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| 03:57.05 | *** join/#asterisk tekzilla (~jon@g224128117.adsl.alicedsl.de) |
| 03:57.20 | p3nguin | Even installing an app that uses an internet time service, it still won't change the time on the device. How annoying. |
| 04:01.14 | tekzilla | i want to intercept calls (who i calling which extension or user) programatically, what would be the best way |
| 04:01.22 | tekzilla | *who is |
| 04:03.11 | p3nguin | Do you want to listen, record, or take the call yourself? |
| 04:04.56 | tekzilla | i dont want to interfere with the call, i just want to know when its taking place |
| 04:05.29 | p3nguin | Would printing the call information on the asterisk cli be enough? You can do it with Verbose() if so. |
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| 04:06.33 | tekzilla | ok so i should just parse cli output |
| 04:06.59 | p3nguin | If you want to read it with an app, there's always the CDR. |
| 04:08.12 | tekzilla | right |
| 04:09.18 | tekzilla | as i want to act pretty much immediately i guess i'll have to continuously parse the CLI output |
| 04:11.07 | tekzilla | there wouldnt be any kind of api for plugins/modules ? |
| 04:14.07 | kaldemar | tekzilla: add a line to your dialplan that notifies you somehow. you can run a dialplan application or a system command with app System or func SHELL. |
| 04:15.04 | kaldemar | one choice is to listen to events from the manager interface. |
| 04:16.15 | kaldemar | define how you want the information and you'll get better answers. |
| 04:16.21 | tekzilla | kaldemar: that sounds like a very good solution |
| 04:16.50 | tekzilla | the info i need is: which extension is being called by whom |
| 04:17.17 | tekzilla | it would indeed be great if i could pass that info to some shell command |
| 04:18.49 | kaldemar | System(/path/to/command ${EXTEN} ${CALLERID(all)}) |
| 04:19.23 | tekzilla | thank you very much! |
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| 04:54.52 | tekzilla | kaldemar: is this a proper catchall extension ? "exten => _X.,1,cmd" |
| 04:55.31 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 04:55.36 | p3nguin | That will match any one digit followed by one or more digits or characters. |
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| 04:56.09 | tekzilla | ok |
| 04:56.34 | p3nguin | So *67 would not match it, but 2014991234 would. |
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| 04:57.06 | tekzilla | yes thanks, then it should work for me |
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| 05:30.42 | rue_mohr | can anyone confirm I'm right that if a nortel MICS is powered down for a week you have to pay ~$400 to buy a few PRI enabler keycode?! |
| 05:31.14 | rue_mohr | cause if so, I think I'm into some firmware hacking |
| 05:39.55 | tekzilla | <PROTECTED> |
| 05:40.42 | p3nguin | hmm |
| 05:41.36 | tekzilla | err that was unintentional |
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| 05:44.22 | p3nguin | Worse things have happened. |
| 05:46.36 | tekzilla | :) |
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| 07:23.41 | irroot | ~freepbx |
| 07:23.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 07:23.54 | irroot | ~trixbox |
| 07:23.54 | infobot | methinks trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 07:24.20 | irroot | using the bot to show collegues the comparision |
| 07:24.25 | irroot | ~beee |
| 07:24.27 | irroot | ~beer |
| 07:24.27 | infobot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
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| 07:39.40 | singler | irroot: playing around with bot? :) |
| 07:39.50 | irroot | indeed |
| 07:39.56 | irroot | fondleing his ass |
| 07:40.24 | irroot | was showing our customer manager the diff between fleapbx and [one]trixbox |
| 07:42.52 | singler | :) |
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| 08:49.45 | Polysics | hello |
| 08:50.13 | Polysics | caller calls, i start MoH then Originate calls to other people they have to press 1 to accept |
| 08:50.18 | Polysics | if they do, they get bridged |
| 08:50.45 | Polysics | everything works well until the CALLER hangs up BEFORE speaking with anyone |
| 08:50.57 | Polysics | i would obviously like to terminate everything |
| 08:51.20 | Polysics | and i would just Hangup using AMI, but apparently the hanging up does not generate ANY event |
| 08:51.26 | Polysics | is that just possible? |
| 08:53.22 | Polysics | i do see the MoH stopped message in the console |
| 08:53.27 | Polysics | but no events |
| 09:01.00 | Polysics | please? |
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| 09:05.04 | merlin8282 | HELO |
| 09:05.08 | *** join/#asterisk coppice (~chatzilla@m121-203-216-134.smartone-vodafone.com) |
| 09:06.27 | irroot | coppice morning sir ... may i pick your brain on the DSP stuff CED/V.21 detection the V21 implementation you have commented on as been flawed that asside will running CED+V.21 detect be better than current ? |
| 09:06.30 | merlin8282 | What is nowadays the best way to receive (and optionally send) faxes through the internet with asterisk ? |
| 09:06.55 | irroot | merlin8282 T.38 is the best option for internet faxing |
| 09:07.41 | merlin8282 | irroot: ok, and does it work when I send a fax with our analog fax to the asterisk (which then would do T.38) ? |
| 09:08.15 | irroot | merlin8282 asterisk 10 has T.30/T.38 gateway now still in beta |
| 09:08.50 | irroot | use of a fax adapter with T38 can also be used |
| 09:08.58 | merlin8282 | irroot: and I suppose that our SIP provider has to be able to do T.38 also ? |
| 09:08.59 | ollii | grandstream supports t38 |
| 09:09.01 | coppice | irroot: you just can't rely on hearing 2100Hz. if you don't look for FAX preamble you will have very quirky results. we did before we started looking for preamble :-) |
| 09:09.06 | ollii | or hylafax with t38modem |
| 09:09.13 | Polysics | basically, why am i getting no event on hangup? |
| 09:09.51 | irroot | coppice yeah and not to mention the quirky bits im getting ATM with V21 only |
| 09:10.12 | coppice | irroot: and when looking for 2100Hz, you need to make sure your detector can deal with all the modulated variants reliably |
| 09:10.24 | merlin8282 | mmm, that would mean that every one who wants to send us a fax would have to send it in T.38 ? |
| 09:10.29 | irroot | and asterisk does not do this currently |
| 09:11.01 | coppice | irroot: what are you using to detect the V.21? you really have to look for FAX preamble, rather than any old V.21 |
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| 09:11.46 | coppice | irroot: I don't know how well Asterisk detects 2100Hz these days, but its track record shouldn't make you very hopeful :-) |
| 09:11.51 | irroot | merlin8282 no they will send via most often POTS the sip provider will gateway to T.38 over net and then you need to gateway back to a faxmachine ... using fax to mail is ok for T.38 supported on asterisk |
| 09:12.11 | merlin8282 | ok. Yes the idea is to do fax2mail |
| 09:12.19 | irroot | coppice yeah indeed lol the V.21 as it stands is looking for 1850 |
| 09:13.02 | irroot | merlin8282 use res_fax/res_fax_spandsp modules for fax to mail |
| 09:13.08 | coppice | that's entirely useless. there is no way to get a simple tone detector to do the job |
| 09:13.20 | ollii | merlin8282: or use hylafax with iax/t38modem |
| 09:14.20 | merlin8282 | mmm, I just see that our sip provider does not support T.38 :/ |
| 09:14.21 | irroot | coppice it does sortof sometimes maybe most the time work but i am using 2100+1850 to initiate T.38 handshake |
| 09:14.56 | coppice | a simple 1850 detector will hit on voice far too often |
| 09:15.06 | irroot | merlin8282 change providers there is little no hope you will have of getting it working see coppice page regarding this |
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| 09:15.57 | Polysics | how is it possible that hanging up results in NOTHING? |
| 09:16.20 | coppice | irroot: since you are already using spandsp, why not use the robust detector it contains? |
| 09:16.33 | irroot | coppice yeah granted with proper dialplan and detecting 1100 and only waiting for 1850 after this is less prone to false start |
| 09:16.52 | irroot | coppice im planing to use it and put it in res_fax_spandsp |
| 09:17.04 | irroot | just not had the time yet its on the todo list |
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| 09:28.10 | ixyd_ | iam using queuelog in my dialplan while queue_log is pointing to an odbc connection (in extconfig.conf), if the odbc is not availiable the queue_log application is heavily blocking the execution of the dialplan :( i already tried to set writetimeout and net_retry_count in odbc.ini but it seems that asterisk doest care about this ;) do you have any ideas how to get rid of this dialplan locking? |
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| 09:40.01 | ixyd_ | i tried setting connect_timeout in res_odbc.conf too, but it seems as none of the timeout parameters is behaving as one would expect it?! even with connect_timeout=>1 (res_odbc.conf) and writetimeout=1 (odbc.ini) and net_retry_count=1 (odbc.ini) it takes about 9 seconds for the queue_log appplication to fail :( |
| 09:41.28 | irroot | ixyd_ you logged a bug ?? |
| 09:41.55 | ixyd_ | not until now...i want to make sure the bug is not my head/configs before ;) |
| 09:42.32 | irroot | if it is we will make fun of you and tell all our friends :P |
| 09:43.04 | irroot | most of us are busy at work and the like so if you log a bug it will get seen too |
| 09:43.21 | irroot | someone may check it out but im bit busy at the moment |
| 09:45.41 | ixyd_ | well i will try a bit more and wait if someone else maybe does have an idea....then i will open a bug :) |
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| 10:10.57 | catphish | does the realtime mysql config support hints from the extensions table? |
| 10:14.33 | ixyd_ | i dont think so |
| 10:17.45 | Polysics | how can i terminate an originate that was started from the current call? |
| 10:18.16 | Polysics | ie. caller comes in, i use originate to dial destination because of intermediate steps, caller hangs up BEFORE someone anwers |
| 10:20.41 | Polysics | can i do ANYTHING about that? |
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| 11:36.50 | _naomi | a channel variable with two underscores is not being inherited, do you know why that might be? |
| 11:37.59 | leifmadsen | more information needed |
| 11:38.11 | leifmadsen | it most certainly works, it just sounds like you might be expecting something else |
| 11:38.32 | wdoekes2 | (a) you're reading it with ${__var} instead of ${var} ? |
| 11:38.41 | leifmadsen | wdoekes2: great point |
| 11:39.10 | wdoekes2 | (b) it's a special variable (name) ? |
| 11:40.19 | wdoekes2 | (c) there is no regular inner/local dialing going on ? |
| 11:40.42 | _naomi | i'm using ${var} |
| 11:40.47 | _naomi | its not a special name |
| 11:41.12 | kaldemar | _naomi: show a CLI output with the set and the noop or verbose |
| 11:41.18 | _naomi | wdoekes2, what do you mean by (c)? there is some complex dialling |
| 11:41.22 | leifmadsen | and the dialplan you're using |
| 11:43.40 | irroot | leifmadsen up at it early hey ... ps R1400 is a go |
| 11:44.06 | leifmadsen | irroot: awesome! I think that is assigned to rmudgett to take through to resolution. I'm usually up about 7am or so most days |
| 11:44.51 | irroot | yeah im sure he will commit it have a good day |
| 11:45.17 | _naomi | heres the cli. its pretty complex sorry. variable is Q_ROW_ID |
| 11:45.19 | _naomi | http://pastebin.com/CMV0zbmp |
| 11:46.12 | irroot | with r1400 commited i have no outstanding 1.8 issues and have not seen deadlocks in ages if you have not moved to 1.8 1.8.7 will be a winner |
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| 11:47.48 | _naomi | dialplan is enormous should i send what i consider relevant bits? |
| 11:48.34 | irroot | NoOp("SIP/200-00000087", "Q_ROW_ID: ") ?? could it be ${....} cant see other response |
| 11:49.23 | _naomi | dialplan line there is exten => s,n,NoOp(Q_ROW_ID: ${Q_ROW_ID}) |
| 11:50.29 | _naomi | vars set in macro-agent-do-queue persist but this one is set in macro-do-queue and does not |
| 11:50.39 | _naomi | maybe nested macros issue? |
| 11:52.24 | kaldemar | NoOp("SIP/100-00000086", "__Q_ROW_ID=158") |
| 11:53.21 | kaldemar | i don't see a Set, only a NoOp. |
| 11:53.47 | _naomi | oh god sorry mate |
| 11:54.22 | _naomi | thanks |
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| 12:35.31 | lowtek | Hey guys, anyone running Asterisk 1.8 on 64-bit ubuntu server? Any issues? Should I stick with old reliable 32-bits? |
| 12:36.02 | lowtek | ~64-bit |
| 12:39.58 | VoipForces | lowtek: I'm running 1.6 64bits on CentOS no problem. |
| 12:40.03 | atheos | O |
| 12:40.04 | atheos | , |
| 12:40.09 | atheos | I'm running 1.8 on Debian 64 |
| 12:40.29 | lowtek | Thanks guys, what types of volume are you pushing? |
| 12:40.56 | VoipForces | lowtek: Got one server with 5x PRI/DAL |
| 12:41.03 | atheos | not too much here, maybe 300-400 calls a day, 50 extensions. |
| 12:41.30 | atheos | my high volume asterisk phones are still running 1.2/1.4 |
| 12:41.42 | atheos | phones/phone systems |
| 12:41.51 | lowtek | Yea, that's where we are now, 1.4 ... 96 servers worth |
| 12:42.17 | lowtek | 16, not 96 ... |
| 12:42.20 | lowtek | type-o |
| 12:42.37 | lowtek | Thinking about finally coming up to 1.8 |
| 12:42.45 | atheos | I'm rewriting one dial plan at a time to get everything up to 1.8 |
| 12:42.47 | lowtek | We're running Debian 5 |
| 12:43.04 | lowtek | 32-bit |
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| 12:44.38 | singler | I would advise to wait for 1.8.7 for upgrade, it seems that it will be very good release :) (1.8.5 is having some issues) |
| 12:45.40 | leifmadsen | actually I find 1.8.5 pretty stable in a few of my lower usage situations -- nothing before that I could get to stay up when using transfers |
| 12:47.43 | mtltemplar | Hey all. Back now for the day. Morning. |
| 12:48.13 | mtltemplar | So I am still having the issue with SIP calls I make dropping after 5 minutes if I mute my phone. If I don't mute it, it will last 'forever' |
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| 12:48.31 | mtltemplar | The SIP debug is http://pastebin.com/GurNSzkr |
| 12:48.40 | leifmadsen | mtltemplar: sounds like asterisk is dropping the call because it isn't getting any audio from the device |
| 12:49.24 | mtltemplar | Right, so I set my rtptimeout and rtpholdtimeout both to 0 and it still happens |
| 12:49.43 | leifmadsen | do you have session-timers enabled? |
| 12:49.47 | atheos | mtltemplar - sounds suspiciously similar to an issue I'm having. put callers on hold, and then loosing audio when returning to the call |
| 12:50.11 | leifmadsen | losing* |
| 12:50.26 | atheos | correct, losing |
| 12:50.37 | atheos | I'm a looser when it comes to spelling |
| 12:50.45 | lowtek | @leifmadsen which version are you recommending for stable+high volume deployments? |
| 12:50.55 | atheos | loser even |
| 12:51.29 | lowtek | Thanks, singler! |
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| 12:56.03 | leifmadsen | lowtek: whatever works in your testing is what i recommend :) I use 1.8.5.0 right now |
| 12:57.21 | lowtek | Cool, tnx :) |
| 12:59.07 | irroot | 1.8.6 is to be born in next few days i suspect |
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| 13:00.51 | awk | 1.8.6 is buggy |
| 13:01.09 | awk | iax doesn't work properly, meetme doesn't work properly... |
| 13:01.37 | awk | ill rather wait for 1.8.42 before i even consider using it... right now i've nearlly lost a few big clients because it is soooo unstable |
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| 13:05.26 | leifmadsen | I'm not a fan of blanket statements |
| 13:05.29 | ollii | we only use 1.8.5 to support single,cheap hfc chips ... ;) |
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| 13:08.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:14.10 | anonymouz666 | singler: that's exactly what I planned to do, wait for 1.8.7. |
| 13:16.17 | anonymouz666 | sipstorecause, important pickups fixes, queue deadlock fix and other few things. |
| 13:17.25 | anonymouz666 | awk: you are lucky, if you can wait. there are things that we can't wait so long. everyone that works with big callcenters know that distributed device state IS needed |
| 13:17.38 | mtltemplar | session-timers? |
| 13:18.42 | awk | anonymouz666: ye and the system DND isn't working properly either.. using it for ring back.. |
| 13:18.55 | awk | getting slin to native alaw issues on a sip channel? |
| 13:19.30 | anonymouz666 | what is system DND? |
| 13:19.48 | awk | feature code for DND |
| 13:21.19 | Katty | zgggnnnn |
| 13:21.22 | Katty | ennffff |
| 13:21.26 | Katty | urbbzzz |
| 13:21.36 | Katty | zombies in |
| 13:21.56 | lowtek | lol, awk, what version do you run in production? |
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| 13:30.13 | mtltemplar | i just set session-timers=originate and it still dropped at precisely 5 minutes of being muted |
| 13:30.22 | mtltemplar | do you think my VOIP provider is somehow overriding my settings? |
| 13:30.54 | mtltemplar | if you look at the sip debug trace, it almost seems that the VOIP provider is the one initiating the BYE |
| 13:31.54 | kaldemar | mtltemplar: do you have RTP timers set? |
| 13:32.52 | kaldemar | mtltemplar: sounds like a 300 second rtpholdtimeout. |
| 13:32.58 | treborsux | <PROTECTED> |
| 13:32.58 | treborsux | <treborsux> like set default volume for loudest |
| 13:33.10 | Katty | ATTENTION LOVABUHLS |
| 13:33.13 | Katty | it is hug time. |
| 13:33.17 | Katty | do you know where your hugs are? |
| 13:33.20 | awk | lowtek: production 1.4.42.-3 |
| 13:33.24 | Katty | hugs leifmadsen |
| 13:33.27 | Katty | hugs Qwell |
| 13:33.48 | irroot | katty hey i want too zombie hugs :P |
| 13:33.54 | Katty | zomb-hugs irroot |
| 13:34.08 | Katty | now they are slight-caffeinated-but-still-mostly-zomb-hugs |
| 13:34.27 | irroot | caffeine for the win |
| 13:34.55 | Katty | caffeine for my face |
| 13:35.37 | *** join/#asterisk bchia (~Adium@nat/digium/x-pklcklerdaxmyfiy) |
| 13:36.03 | anonymouz666 | irroot: did your fix to queue deadlock works fine for the reporter? |
| 13:36.46 | Katty | where are my hugs?! )= |
| 13:36.48 | Katty | )= )= )= |
| 13:37.19 | irroot | anonymouz666 i hope for feedback i have seen this lock so infrequently myself i had some time and took the excelent debug and put a proposal in id like someone to look at the logic of it perhaps |
| 13:37.25 | mtltemplar | yes, i have rtptimeout and rtpholdtimeout at 0 |
| 13:37.30 | irroot | {{{{katty}}}}} |
| 13:37.59 | fenrus | oh haithere Katty |
| 13:39.06 | mtltemplar | both globally and at the peer level |
| 13:39.55 | Katty | yay |
| 13:39.57 | Katty | hugs fenrus |
| 13:40.31 | fenrus | hugs Katty |
| 13:42.07 | anonymouz666 | irroot: if i read correctly it happens two times/day for the reporter, so if he applied the fix will see the result soon |
| 13:43.04 | chuckf | hugs Katty |
| 13:43.17 | Katty | hugs on chuckf |
| 13:43.48 | *** join/#asterisk m_tadeu (~quassel@89.180.63.172) |
| 13:44.25 | m_tadeu | hi...is it possible to trigger an event from an agi to receive in ami? |
| 13:44.47 | m_tadeu | a custom event, I mean |
| 13:45.57 | irroot | anonymouz666 not as easy as that the initial report was re pickup that was already in the branch the queue problem was a double log on ticket got leifmadsen to modify jira to reflect this when i had a proposal for fix |
| 13:47.16 | leifmadsen | m_tadeu: you mean have your AGI attach to AMI? Sure |
| 13:48.20 | m_tadeu | leifmadsen: I'm not sure what you mean by that....what I need is to send a custom event from an agi script and receive it in my manager app |
| 13:54.24 | leifmadsen | m_tadeu: then I don't understand your question |
| 13:55.39 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:58.18 | atheos | m_tadeu, are you wanting to execute an event without it being associated to a call? am I reading your question right? |
| 13:59.05 | m_tadeu | leifmadsen,atheos: I need to send some specific notifications from my agi script during the call sequence....and my manager app should get those notifications in order to do some stuff |
| 13:59.27 | *** join/#asterisk dhartman (~dhartman@wilug/newlug/ricko73) |
| 14:00.01 | kaldemar | m_tadeu: core show application UserEvent |
| 14:00.12 | atheos | asterisk to ago to manager? sure, you can do that. |
| 14:00.19 | atheos | ago/agi |
| 14:00.49 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 14:01.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:01.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.59 | irroot | putnopvut greets there |
| 14:02.08 | McBoingBo | We have some users outside the city that use our VPN and X-Lite/Bria as a VOIP client, they complain the sound gets very choppy at times, and I was wondering what solution do you guys use when this happens? |
| 14:02.08 | putnopvut | hi |
| 14:02.40 | irroot | putnopvut dude i know its been a while but surely the queues container should not be locked while calling ring_one ?? |
| 14:02.42 | m_tadeu | thanx guys...I was looking in agi documentation and forgot about the core apps |
| 14:02.46 | irroot | in app_queue |
| 14:03.12 | atheos | m_tadeu - the perl Asterisk::AGI stuff makes it easy |
| 14:03.38 | putnopvut | irroot: um, I have no idea man. That sounds wrong, but I seem to recall that if weights are involved then the container would be locked for some reason. I'm not 100% sure though. |
| 14:04.29 | irroot | ok thats wehre it was its involved in a dead lock ill look into it more thx millions |
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| 14:09.54 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:12.45 | jayson_r | i am trying to use asterisk for voicemail for my avaya s8730 using analog lines |
| 14:13.04 | jayson_r | when asterisk sees the call, it sees he original caller id rather than the called extention |
| 14:13.08 | jayson_r | has anyone ever seen this? |
| 14:13.24 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 14:14.29 | malcolmd | asterisk doesn't see called extension if the call's coming in from an FXO interface....if Asterisk is the FXS side of the equation, and the avaya's the FXO), then asterisk will know what extension was dialed. |
| 14:15.36 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 14:17.08 | pabelanger | m_tadeu: Or if you are using Python try StarPy |
| 14:17.17 | McBoingBo | what can I do to eliminate echo, it seems no matter what I do, the other end of my conversation can always hear some echo |
| 14:17.39 | pabelanger | McBoingBo: find the source of it? |
| 14:17.45 | pabelanger | what channel tech are you using? |
| 14:17.58 | McBoingBo | how do I find the source of it? |
| 14:18.11 | McBoingBo | I thought it was always the device |
| 14:18.22 | pabelanger | what type of channel, SIP, DAHDI, etc |
| 14:18.28 | McBoingBo | SIP |
| 14:19.11 | pabelanger | then try to figure out which leg of the echo it happens on |
| 14:19.39 | McBoingBo | thats why I am here, I have no idea |
| 14:20.34 | pabelanger | well, we need more information about your setup. EG: PhoneA calls PhoneB, there is echo. If PhoneA calls PhoneC, is there echo? If PhoneB calls PhoneC is there echo? Process of elimination |
| 14:20.51 | McBoingBo | there is always echo on the other end |
| 14:21.01 | McBoingBo | volume will control the magnitude |
| 14:21.02 | pabelanger | well, what is the other end? |
| 14:21.21 | McBoingBo | whomeever/whatever you call |
| 14:21.21 | *** join/#asterisk Polysics (~luca@host7-14-static.4-79-b.business.telecomitalia.it) |
| 14:21.26 | Polysics | hello |
| 14:21.37 | Polysics | i will try asking again :-D |
| 14:21.39 | sruffell | McBoingBo: You could try setting up a softphone and use that and see if the other end has echo..if soâ¦you know it's your phone. |
| 14:21.43 | McBoingBo | it could be a cell phone, another VOIP phone in the office, anything |
| 14:21.47 | sruffell | I mean if not. |
| 14:22.06 | pabelanger | McBoingBo: If you change phones, do you get echo still? |
| 14:22.13 | McBoingBo | yes |
| 14:22.16 | Polysics | caller calls in, moh starts, i originate calls to receivers, when they pick up they are told who is calling and press 1 to accept, if they press 1 they get bridged |
| 14:22.23 | Polysics | everything is working |
| 14:22.24 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:22.47 | Polysics | BUT when the caller hangs up while one of the receivers is ringing, the receiver still rings |
| 14:23.02 | Polysics | how can i make that ringing stop? |
| 14:23.03 | pabelanger | McBoingBo: if you can from phoneA to PhoneB on the same LAN, is there echo? |
| 14:23.29 | McBoingBo | we mostly have Polycom 300/400's which would always have echo, and the new 450's the same |
| 14:23.43 | McBoingBo | pabelanger: yes, echo |
| 14:24.22 | pabelanger | Is RTP going through Asterisk? or are you re-inviting media? |
| 14:25.00 | pabelanger | also, check your phone configuration settings for some sort of echo cancelling ability |
| 14:25.38 | McBoingBo | not sure about RTP pabelanger |
| 14:26.36 | pabelanger | McBoingBo: Start there. If RTP is passing thru asterisk then change it so it does not, retest to see if there is echo. If the answer is yes, then the issue is not with Asterisk but your phones |
| 14:27.20 | pabelanger | If echo goes away, then you have an issue with Asterisk and RTP. Either CPU is to high or something else |
| 14:28.54 | McBoingBo | k thanks |
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| 14:39.21 | irroot | echo is almost always a hard issue physical problem almost always on the hybrid |
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| 14:42.40 | Polysics | no ideas about my problem, please? |
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| 14:58.22 | *** join/#asterisk ThomasFrieling (~Adium@91-64-99-108-dynip.superkabel.de) |
| 14:58.43 | *** join/#asterisk navaismo (~navaismo@187.170.1.109) |
| 14:58.47 | ThomasFrieling | hi all! is there a way to add entries to astDB on asterisk startup? |
| 14:58.48 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 14:58.52 | ThomasFrieling | like a bootstrap? |
| 14:58.55 | treborsux | [2-9]11T|0T|011xxx.T|[0-1][2-9]xxxxxxxxxT|[2-9]xxxxxxxxxT|[2-9]xxxxxxT|[*]xxxT|[#]xxxT|[2-9]xxxT|[2-9]xxT |
| 14:58.55 | treborsux | <treborsux> 3|3|3|4|3|3|3|3|3|3 |
| 14:58.55 | treborsux | <treborsux> that is how it is set |
| 14:58.55 | treborsux | <treborsux> but when i pick up the phone and start dialing it hits send by itself in the middle of dialing why???? |
| 14:59.05 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 14:59.09 | navaismo | morning! |
| 14:59.50 | Gugge | ThomasFrieling: not really, but they stay on reboots |
| 15:01.01 | ThomasFrieling | Gugge: i see |
| 15:01.05 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 15:01.23 | p3nguin | You could always screw up the init script to do it. |
| 15:01.25 | ThomasFrieling | the "astdb" in sip.conf is deprecated and wont be fixed, right? |
| 15:01.40 | voipguynumber1 | purpledragon |
| 15:04.04 | p3nguin | What do you mean astdb in sip.conf? |
| 15:04.14 | p3nguin | That must be something from before my time. |
| 15:09.49 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.142.static.cableonline.com.mx) |
| 15:12.22 | Gugge | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf <- astdb is there ... but i never used it :) |
| 15:13.14 | voipguynumber1 | voip-info.org is so outdated. is there any up2date resource for asterisk? |
| 15:13.23 | p3nguin | I'm looking at my most recent sample file, and it isn't in there. |
| 15:13.44 | Gugge | i dont know if it ever worked either, it just see it there :) |
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| 15:16.52 | p3nguin | I'm kind of wishing it would have been more popular when I started using my astdb for stuff. That seems useful. |
| 15:17.05 | McBoingBo | pabelanger: well the Polycomm 450 has an echo cancellation feature, but it only works well if the headset is plugged into the phone, and not the amp system for the headset. |
| 15:17.32 | p3nguin | Within the past couple years is when I related all my devices to extensions in the db. It would have been perfect for that. |
| 15:17.54 | McBoingBo | I have had a hard time finding just a headset with rj9 connector |
| 15:18.25 | p3nguin | It seems like my headset for my Cisco phone might have that connector. |
| 15:18.30 | p3nguin | Let me check. |
| 15:18.33 | *** join/#asterisk stack_ (~stack_@mail.edpaymentsystems.com) |
| 15:19.04 | McBoingBo | It seems almost impossble to get a headset and not a whole system....gah |
| 15:20.50 | p3nguin | Yep, my phone/headset use the 4P4C. |
| 15:22.11 | McBoingBo | p3nguin: 4P4C, is that some kind of connection converter? |
| 15:22.32 | irroot | yeah spring day tommorow |
| 15:22.32 | p3nguin | It's the plug on the end of handset and headset cords. |
| 15:23.25 | McBoingBo | oh, lol, so what headset do you use? And does anyone know of a good place I can pickup just a rj11 headset? |
| 15:23.49 | p3nguin | You mean 6P6C, probably. |
| 15:24.00 | McBoingBo | err sorry yeah I mean rj9 |
| 15:24.08 | p3nguin | That's a 4P4C. |
| 15:24.16 | McBoingBo | no, RJ11 is 4P4C |
| 15:24.27 | p3nguin | negative |
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| 15:24.37 | p3nguin | RJ11 uses 6P2C. |
| 15:24.46 | McBoingBo | then this site is wrong lol http://www.trianglecables.com/telplugenrj4.html |
| 15:24.50 | p3nguin | I have a plantronics headset, which does not directly have the plug on the cord. |
| 15:25.03 | McBoingBo | sorry I go by connector type not the ends |
| 15:25.21 | p3nguin | Yes, that site is wrong. |
| 15:25.37 | p3nguin | I'm talking about the connector, as well. |
| 15:25.53 | p3nguin | It's not an RJ unless it provides telephone service to a phone. The handset connectors do not. |
| 15:26.00 | McBoingBo | yes indeed, I speak male you speak female ;) |
| 15:26.11 | stack_ | Good morning⦠I just converted from a 1.2 system to a 1.8 system. I'm using Ubuntu Hardy and the Asterisk packages from asterisk. I'm getting the following errors: http://pastebin.com/z1PBzmuc Calls ring busy when this pop up. It seems to be intermittent. Any ideas? |
| 15:26.21 | p3nguin | I'm speaking hermaphrodite. |
| 15:26.39 | p3nguin | You're speaking registered sex offender. |
| 15:27.29 | p3nguin | At any rate, my headset cord and phone use a 4P4C (or RJ9 to those who don't know any better). |
| 15:27.38 | McBoingBo | pfft :P |
| 15:27.52 | p3nguin | I have a plantronics headset, which does not directly have the plug on the cord. I could probably find the cord part number if you need it. |
| 15:28.32 | McBoingBo | p3nguin: I bought several S12 systems from Plantreonics and they echo a lot, echo goes away when I bypass the amp system, so I want to get just headsets |
| 15:28.44 | p3nguin | The headset has some special fancy plantronics connector, so you have to have the right cord for your phone. |
| 15:29.07 | McBoingBo | yeah the "easy disconnect" connectoers |
| 15:30.15 | p3nguin | With my cable going to my phone, I can plug in any of those easy disconnect headsets into my cable. |
| 15:30.42 | McBoingBo | p3nguin: so you only have a headset not a fancy amp system that comes with? |
| 15:31.00 | p3nguin | correct. headset, cable, phone. |
| 15:31.35 | McBoingBo | p3nguin: know the model? |
| 15:31.44 | p3nguin | I'm trying to find a number for you. |
| 15:32.05 | McBoingBo | p3nguin: thanks! |
| 15:32.17 | p3nguin | I remember when I was shopping for the cable, I found a few part numbers that were the exact same cable. Maybe the lengths were different or something. |
| 15:38.00 | p3nguin | It looks like it may be the Plantronics 26716-01 quick disconnect Cisco cable. I'd say it won't work on Polycom, though. |
| 15:40.19 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:40.24 | p3nguin | However, it looks like the 27190-01 is for Polycom. |
| 15:40.52 | McBoingBo | hehe thanks p3nguin |
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| 15:42.02 | p3nguin | It seems to be for H and P series headsets. I don't know if that means it works on your other model headset or not. |
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| 15:44.08 | p3nguin | I also do not know if the Polycom phones have a built-in headset amplifier. |
| 15:44.59 | McBoingBo | p3nguin: what cisco phone do you have? |
| 15:45.05 | p3nguin | 7960G |
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| 15:51.45 | gnuday | Hi I'm looking for a site with some examples for py-asterisk. Any suggestions? Thanks |
| 15:56.31 | McBoingBo | jesus, $200 for a RJ9 headset, is this the norm? |
| 15:56.46 | p3nguin | What model? |
| 15:57.00 | McBoingBo | http://www.1800headsets.ca/headset/sennheiser-c510-headset-cstd01-cord/ |
| 15:57.19 | Qwell | you expect a sennheiser to be cheap? O.o |
| 15:57.23 | McBoingBo | it seems the headsets WITH amp systems are cheaper than just the RJ9 headsets themselves |
| 15:57.40 | p3nguin | That's probably a good price for that headset. |
| 15:58.02 | jaytee | sennheiser is like the Lamborghini of headsets |
| 15:58.06 | McBoingBo | hehehe |
| 15:58.09 | stack_ | Good morning⦠I just converted from a 1.2 system to a 1.8 system. I'm using Ubuntu Hardy and the Asterisk packages from asterisk. I'm getting the following errors: http://pastebin.com/z1PBzmuc Calls ring busy when this pop up. It seems to be intermittent. Any ideas? |
| 15:58.17 | McBoingBo | yeah I knew it was higher end, but that much? ok |
| 15:58.48 | p3nguin | It's about $100 for the Plantronics headsets I use. |
| 15:59.20 | McBoingBo | p3nguin: which model? you gave me the quick disconnect PN |
| 15:59.28 | jaytee | that's because the speakers are made with samarium cobalt magnetic cores and the diaphragms are made from gold impregnated unobtanium :-) |
| 15:59.43 | McBoingBo | woooh deep |
| 16:01.20 | p3nguin | I thought that's what you wanted. I like the HW251 for single ear and HW261 for dual ear. Append an 'N' for noise canceling models. |
| 16:02.22 | p3nguin | http://www.voiplink.com/Plantronics_HW261N_Dual_Ear_Noise_Canceling_Headse_p/plantronics-hw261n-cs.htm |
| 16:04.47 | p3nguin | If you don't want/need a wideband model, drop the W from the part number. |
| 16:05.06 | p3nguin | They should be about the same price, though. |
| 16:05.22 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:07.11 | p3nguin | Cheap! http://ogden-ut.geebo.com/merchandise/view/id/569278-plantronics_h261n_supraplus_office/ |
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| 16:34.05 | raden | hugs Katty |
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| 17:10.02 | killown | Do anyone know a soft sip thats support g729a codec ? |
| 17:10.57 | leifmadsen | killown: likely only commercial soft phones as g729 requires a license |
| 17:11.12 | leifmadsen | killown: the phone by EyeBeam would be an example (EyeBeam) |
| 17:11.23 | leifmadsen | s/by EyeBeam/by Counterpath/ |
| 17:11.24 | killown | leifmadsen, For linux? |
| 17:11.33 | leifmadsen | I don't think they make a Linux one anymore |
| 17:12.01 | leifmadsen | I'm not sure how access to the g729 codec would exist for softphones in linux |
| 17:12.48 | killown | :( |
| 17:13.39 | p3nguin | Why do you have to use g729? |
| 17:14.15 | killown | p3nguin, Because my sip provider does use this codec |
| 17:14.24 | p3nguin | You aren't using Asterisk? |
| 17:15.11 | killown | p3nguin, I am using... |
| 17:15.30 | p3nguin | Using... what? |
| 17:15.32 | killown | p3nguin, What is the best codec? |
| 17:15.42 | p3nguin | There is no "best" codec. |
| 17:15.50 | p3nguin | Are you using Asterisk? |
| 17:15.57 | killown | YES |
| 17:16.19 | p3nguin | If you are using Asterisk to communicate with your provider, you don't need g729 on your phones. |
| 17:17.36 | p3nguin | If the only reason you wanted g729 for your phone was because your provider uses g729, then you don't need g729 on your phones. Use a different codec on your phones that does not require a license. |
| 17:18.21 | killown | p3nguin, G722? |
| 17:18.42 | p3nguin | Does your phone support it? Does your asterisk support it? |
| 17:19.21 | p3nguin | I probably would have picked ulaw, but if you support g722, you could use it. |
| 17:19.53 | killown | p3nguin, The qutecom soft phone supports it |
| 17:20.16 | killown | p3nguin, I am looking for a codec that gives the best quality voice |
| 17:20.39 | p3nguin | If your provider is using g729, I'd use ulaw on the phones. |
| 17:21.05 | p3nguin | You won't be able to use the full potential of g722 with the other leg of every call using a lesser codec. |
| 17:22.10 | killown | p3nguin, Ok, I just need some codec that gives a good quality voice, do you recommend a codec? |
| 17:22.20 | p3nguin | (1220.38) <p3nguin> If your provider is using g729, I'd use ulaw on the phones. |
| 17:23.14 | killown | Ops sorry |
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| 17:23.50 | p3nguin | You won't get any better quality on a call than the lowest codec being used between the two end points. |
| 17:24.53 | p3nguin | So if you used g722 between your phone and asterisk, and g729 between asterisk and your ITSP, you'll just be using more bandwidth between your phone and asterisk and not getting any better quality than the g729 leg. |
| 17:26.01 | jaytee | good point |
| 17:26.10 | killown | p3nguin, Do you know a linux softphone that supports ulaw? |
| 17:26.21 | p3nguin | all of them |
| 17:26.39 | p3nguin | I like twinkle, but it requires qt3 (which many people do not have). |
| 17:27.10 | *** join/#asterisk Phr0zen (Phr0zen@blk-224-135-210.eastlink.ca) |
| 17:27.29 | killown | p3nguin, What about this http://icanblink.com/blink-qt-beta.phtml ? |
| 17:28.09 | p3nguin | It supports ulaw/alaw. |
| 17:28.17 | killown | ok |
| 17:28.45 | p3nguin | Although the people who wrote the web site aren't very smart. "MacOSX" |
| 17:29.15 | p3nguin | Or maybe their space bars got broken mid-page, then began working again later. |
| 17:29.24 | *** join/#asterisk xpot-mobile (~james@155-99-194-206.uconnect.utah.edu) |
| 17:29.30 | killown | The soft sip too... I get a lot errors until bring it to work... |
| 17:31.52 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 17:32.13 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:43.18 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-xdousrrpvgsffkdx) |
| 17:47.18 | *** join/#asterisk zamba (marius@flage.org) |
| 17:47.24 | zamba | i need a decent sip client for linux |
| 17:47.32 | zamba | and please don't say ekiga, because that sucks |
| 17:47.52 | p3nguin | I like twinkle, but it requires qt3 (which many people do not have). |
| 17:48.11 | zamba | twinkle i've tried and that sucked a bit as well :p |
| 17:48.24 | p3nguin | Sounds like you're doing it wrong. |
| 17:48.26 | zamba | hehe |
| 17:48.28 | irroot | zamba blink is working for me |
| 17:48.41 | irroot | zamba you could use asterisk .... |
| 17:48.50 | zamba | hehe |
| 17:48.55 | p3nguin | Asterisk has a soft phone? |
| 17:49.04 | irroot | asterisk "cli dial" |
| 17:49.13 | irroot | or write a AMI script |
| 17:49.14 | p3nguin | And then use a mic and speakers? |
| 17:49.20 | zamba | irroot: that's for mac os.. i need something for linux |
| 17:49.23 | irroot | yip chan_console |
| 17:49.32 | p3nguin | Blink works on Linux. |
| 17:49.33 | navaismo | Zoiper |
| 17:49.39 | navaismo | <PROTECTED> |
| 17:49.52 | irroot | not bria blink you fool who cares about iFad |
| 17:50.49 | irroot | ekiga uses opal |
| 17:51.08 | zamba | http://icanblink.com/ |
| 17:51.10 | zamba | not this? |
| 17:51.24 | p3nguin | http://icanblink.com/blink-qt-beta.phtml |
| 17:51.26 | irroot | zamba yip |
| 18:05.31 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:07.55 | Hive | Hey guys, I've got a problem going on with using GoToIf (newb question i know) Here's a pastebin explaining the problem. http://pastebin.com/1UdRAp2d Any insight is greatly appreciated! |
| 18:08.24 | Hive | Basically it's not jumping to the right spot! |
| 18:08.58 | p3nguin | If your group count is greater than 1, it will go to the next line in dial plan. If it is not greater than one, it will go to the busy label. |
| 18:09.04 | p3nguin | What did you want it to do? |
| 18:09.36 | Hive | it it's > 1 then go to (busy) |
| 18:09.37 | kaldemar | Hive: you should have ?:busy |
| 18:09.39 | p3nguin | except you have a syntactical error |
| 18:09.39 | Hive | ahhh |
| 18:09.40 | Hive | damn it |
| 18:09.49 | Hive | sorry for bringing this to the table -_- |
| 18:09.49 | p3nguin | No ? even. |
| 18:09.57 | Hive | thanks guys haha |
| 18:10.34 | p3nguin | But if it is greating than one, it will go to the next line. ?:busy says if it is NOT greater than 1, jump to busy label. |
| 18:10.59 | p3nguin | So you probably want ?busy instead. |
| 18:11.01 | McBoingBo | p3nguin: We are located in Ottawa here, we have several sales type folks out in Toronto that complain about choppy phone calls through xlite&bria, is ther something I can do about that? I want to blame ISP's at this point but apparently Skype calls are flawless |
| 18:12.54 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:18.25 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0 (2011/08/08), dahdi-tools 2.5.0 (2011/08/08), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 18:18.46 | leifmadsen | Asterisk 1.8.6.0 is now available. More information about this release is available in the release announcement at http://www.asterisk.org/node/51672 |
| 18:19.35 | VoipForces | McBoingBo: I have has very good sucess with Zoiper Biz |
| 18:20.14 | VoipForces | McBoingBo: I would try a hard phone see if you have choppy voice, if so then I would check network usage and then internet bandwidth. |
| 18:20.35 | VoipForces | McBoingBo: Latency also on your network and on the internet to your carrier. |
| 18:24.19 | Naikrovek | Polycom add support for the SIP REASON header yet? anyone know? |
| 18:36.31 | voipguynumber1 | purple dragon |
| 18:36.50 | WIMPy | Naikrovek: They don't? |
| 18:37.02 | p3nguin | voipguynumber1: You said that before and I didn't understand it then, either. |
| 18:37.30 | Naikrovek | WIMPy: don't think so. |
| 18:37.55 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:38.20 | Naikrovek | WIMPy: if you have a ring group with two phones in it, call the ring group, both phones ring. One user picks up the phone, the other user will show a missed call. REASON code would prevent that |
| 18:39.22 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-029.mobile.uci.edu) |
| 18:39.51 | Naikrovek | Asterisk supports it, and would tell the phone that didn't answer that the call was answered by another phone, and then the phone would NOT display a missed call. |
| 18:40.01 | voipguynumber1 | p3nguin: purple dragon |
| 18:40.23 | p3nguin | Yes, that. |
| 18:40.33 | p3nguin | Still don't understand it. |
| 18:40.46 | voipguynumber1 | p3nguin: one day you will... |
| 18:41.12 | p3nguin | I wouldn't know why it would know it one day. It doesn't seem to have much relevance. |
| 18:42.16 | jaytee | for a minute I was thinkin Barney but that's a dinosaur.....not a dragon |
| 18:42.22 | voipguynumber1 | ah but what is relevant these days |
| 18:45.17 | jaytee | I googled. I think he was referencing the slang term on urban dictionary.....not the gay travel guide company in Thailand |
| 18:45.29 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:45.49 | Qwell | jayson_r: How can you be sure? |
| 18:45.51 | jaytee | or he possibly could have been referring to a hybrid strain of cannabis sativa |
| 18:45.55 | voipguynumber1 | lol, that's not what i was referencing but that is hilarious |
| 18:45.56 | Qwell | and by that, I mean jaytee |
| 18:46.13 | Qwell | voipguynumber1: Wanna see something else hilarious? |
| 18:46.35 | WIMPy | Naikrovek: I know it used to annoy me quite a lot before Asterisk supported it. Although there had been patched quite some time before. |
| 18:46.51 | *** mode/#asterisk [+b *!*@*98.118.168.221] by Qwell |
| 18:46.52 | *** kick/#asterisk [voipguynumber1!~north@pdpc/sponsor/digium/Qwell] by Qwell (I laughed. IRL. PS, still permanent.) |
| 18:47.30 | WIMPy | Err. "a patch available" |
| 18:48.55 | *** join/#asterisk trumee (~trumee@cpc2-cmbg7-0-0-cust855.5-4.cable.virginmedia.com) |
| 18:49.15 | leifmadsen | Qwell: our favourite troll? |
| 18:49.21 | trumee | is there any ATA which supports TLS/SRTP with asterisk? |
| 18:49.50 | Qwell | leifmadsen: always |
| 18:49.56 | Naikrovek | yeah who was that |
| 18:50.34 | Qwell | I still haven't gotten a harassing message yet. Maybe he's finally getting a clue. |
| 18:53.40 | leifmadsen | Qwell: you're no naive |
| 18:53.45 | leifmadsen | s/no/so/ |
| 18:53.48 | leifmadsen | :) |
| 18:54.03 | Qwell | I don't have to take that from you! |
| 18:54.38 | Qwell | He probably just hasn't realized he's gone yet. |
| 18:54.46 | jayson_r | Qwell: how can i be sure of what? |
| 18:55.01 | Qwell | jayson_r: you can be sure of my tab completion failure rate |
| 18:55.21 | jayson_r | Qwell: gotcha :-) |
| 18:55.32 | *** join/#asterisk samuelsapps (~samuel_sa@fm-ip-118.136.130.159.fast.net.id) |
| 18:56.30 | Qwell | leifmadsen: I am calling it right now. |
| 18:56.35 | Qwell | That was our dear friend. |
| 18:56.49 | leifmadsen | the dear leader?! |
| 18:56.53 | Qwell | WE LOVE THE LEADER |
| 18:57.02 | leifmadsen | :) |
| 18:59.54 | *** join/#asterisk samuelsapps (~samuel_sa@118.136.130.159) |
| 19:07.47 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:09.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:13.07 | pabelanger | Leader, leader, leader, leader, leader, leader, leader, leader, BATMAN! |
| 19:16.23 | jaytee | <@Qwell> and by that, I mean jaytee < huh? |
| 19:16.51 | Qwell | nothing |
| 19:17.22 | jaytee | thought I'd unintentionally said something that was offensive |
| 19:18.17 | jaytee | ok, now I think I get it. autocomplete |
| 19:18.22 | jaytee | :-) |
| 19:19.56 | *** join/#asterisk dhartman (~dhartman@wilug/newlug/ricko73) |
| 19:25.56 | *** part/#asterisk samuelsapps (~samuel_sa@118.136.130.159) |
| 19:26.15 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:26.17 | [sr] | howdy |
| 19:29.13 | jaytee | wow, the feds want to stop the AT&T T-Mobile merger. reminds me of the early 80's |
| 19:31.42 | _Corey_ | jaytee: Yeah, they filed suit to block and everything |
| 19:34.58 | *** join/#asterisk navaismo (~navaismo@187.170.1.109) |
| 19:35.15 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 19:37.13 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 19:37.25 | hetii | Hello :) |
| 19:38.02 | navaismo | hi |
| 19:41.15 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:41.45 | hetii | I need to clear few things about SIP protocol, because i try to build own proxy server. |
| 19:42.39 | hetii | the question is for what many kind of client sip software (softphones) have settings to define a sip proxy server ? |
| 19:43.27 | p3nguin | AT&T - T-Mobile? What happened to Verizon - T-Mobile from a few years ago? |
| 19:44.05 | hetii | as i see based on "Twinkle" softphone sip message sending by it is equal independed if i set proxy settings or not |
| 19:44.28 | hetii | so who and when use those information ? |
| 19:45.39 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 19:45.46 | p3nguin | I think I'd rather use one of the existing SIP proxies rather than learn all that stuff that only needs to be learned to write a new SIP proxy. |
| 19:47.25 | hetii | the point is that i really need build own one :) so will be nice if someone could provide me a useful information |
| 19:47.39 | Qwell | why? |
| 19:48.01 | p3nguin | Wouldn't it be a lot less trouble to hack an existing one? |
| 19:48.04 | hetii | To be able change some headers when it is required |
| 19:48.06 | Qwell | If you can't figure out how to find the information you'd need to write a SIP proxy - you cannot write a SIP proxy. |
| 19:48.57 | *** join/#asterisk CaptWho (~Capt@unaffiliated/captwho) |
| 19:49.23 | CaptWho | what does it take to turn asterisk into a class 5 switch? |
| 19:49.43 | CaptWho | it looks like it's capable... |
| 19:50.09 | wdoekes2 | class 5 is what asterisk excels at |
| 19:51.11 | wdoekes2 | http://xkcd.com/903/ |
| 19:51.26 | CaptWho | <wdoekes2> have you worked with it in that capacity? |
| 19:51.41 | p3nguin | I don't remember wdoekes2 saying that. |
| 19:52.02 | Qwell | p3nguin: I do. It was before your time. |
| 19:52.39 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 19:52.46 | p3nguin | I've been here since captwho joined, and wdoekes2 only said two things, neither of which was the line captwho just quoted. |
| 19:53.38 | jaytee | my all time favorite xkcd comic: http://xkcd.com/418/ |
| 19:53.48 | CaptWho | i was addressing that question to wdoekes2 |
| 19:53.55 | pabelanger | Asterisk PBX, not Asterisk Switch |
| 19:54.13 | CaptWho | just wondering if he had first hand experience with it |
| 19:55.23 | CaptWho | ...or does anyone have first-hand experience using asterisk as a class 5 switch? |
| 19:55.47 | p3nguin | When you wrap someone's nick in angled brackets, it appears if you are quoting that person. |
| 19:56.09 | pabelanger | ~ask |
| 19:56.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:56.12 | p3nguin | If you want to address someone, it is far easier to just type a couple letters of the person's nick and press the tab key to complete it. |
| 19:56.24 | CaptWho | sorry... using xchat and it just did that for me |
| 19:56.30 | p3nguin | Nasty. |
| 19:56.33 | [sr] | jaytee: whats wrong with the AT&T? |
| 19:56.47 | p3nguin | AT&T wireless sucks. |
| 19:57.31 | jaytee | [sr], nothing in particular. just commenting on the feds trying to block their merger with T-Mobile |
| 19:57.50 | [sr] | hum |
| 19:57.58 | jaytee | kind of like when the feds forced them to break up their "monopoly" back in the early 80's. |
| 19:58.09 | [sr] | its usual the ISP's to provide wireless access's in the US? |
| 19:58.10 | p3nguin | And in the '80s, too. |
| 19:58.33 | p3nguin | Ugh. People and their misplaced apostrophes. |
| 19:58.33 | jaytee | nothing stimulates an economy like government intervention :-) |
| 19:58.39 | CaptWho | oh, that's cool, p3nguin. never knew that before |
| 19:58.42 | *** join/#asterisk Syrex (~syrex@dsl-146-17-198.telkomadsl.co.za) |
| 19:59.10 | wdoekes2 | CaptWho: I spoke too soon, see what pabelanger said, it's a pbx, not a switch. however, my wikipedia result for class 5 turned up that it was something that subscribers would connect to directly, which is something that asterisk does better than the other "classes" mentioned |
| 19:59.14 | p3nguin | HERE COMES AN S |
| 19:59.27 | jaytee | I apologize for my blatant misuse of the apostrophe |
| 19:59.45 | jaytee | goes off to find an online punctuation refresher course |
| 19:59.56 | p3nguin | Don't trust anything you find online. |
| 20:00.13 | jaytee | especially in the personal ads |
| 20:00.31 | p3nguin | Online, people think "it's" means it owns something. |
| 20:01.24 | p3nguin | And they try to sell 85-foot cars and stuff. "Selling my 85' Fiero..." |
| 20:01.31 | jaytee | lol |
| 20:01.44 | jaytee | extra long garage not included |
| 20:02.09 | p3nguin | When I was traveling last month, I saw a tractor/trailer that indicated the driver was from the graduating class of 68 feet. |
| 20:02.18 | p3nguin | Class of 68' |
| 20:02.48 | p3nguin | I just shook my head as I went around him. I don't understand how people can be that dense. |
| 20:02.48 | jaytee | summa cum laude and sum cum quietly |
| 20:04.40 | eduzimrs | ANyone here has some example with app AUTHENTICATE() ? |
| 20:04.51 | p3nguin | Yep. One moment. |
| 20:04.56 | Syrex | Asterisk 1.8 ready for serious production? Burnt myself moving to 1.6 too early... ;) |
| 20:05.22 | jaytee | < 2hrs = cold refreshing Three Floyds Pride and Joy Mild Ale |
| 20:06.02 | navaismo | Syrex if you read the changes yes. |
| 20:06.33 | p3nguin | eduzimrs: http://pastebin.com/25nSuvxn |
| 20:06.41 | eduzimrs | p3nguin: tks |
| 20:07.03 | p3nguin | syrex: That's why you should pay attention to branches that say LTS and those that do not. |
| 20:07.32 | p3nguin | Don't call me and enter extension 123; it isn't real. |
| 20:09.12 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:09.20 | pabelanger | Sylnai: personally, I never move anything into production without creating a test environment first |
| 20:10.07 | jaytee | ~book |
| 20:10.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 20:12.56 | eduzimrs | p3nguin: see if its possible what i wanna do. I want a authentication using a password associated to a name. ex: when callerid (BLAH) branch 123 make a call will need the password to jump to the next priority. I need the password associated to the Callerid or a file that contains the pass and name. |
| 20:13.59 | p3nguin | Authenticate() should be able to do that. |
| 20:14.34 | eduzimrs | im having difficults trying to do that |
| 20:14.40 | eduzimrs | its not clear to me |
| 20:16.04 | eduzimrs | its important that ACCOUNTCODE be setted with the password value |
| 20:16.15 | p3nguin | You can use Authenticate(/path/to/file) |
| 20:16.57 | trumee | anybody knows what is the backend of sipgate.co.uk? |
| 20:17.12 | Defraz | I have a little php that uses the management api in asterisk, I hvae it connecting just fine but I can't figure out the command in asterisk to use to have it dial my phone at my desk SIP/1010 for example then dial the number I have have in my form for example. |
| 20:17.27 | Defraz | I am using the fop2 to do it now but I rather have my php script do because it does a customer lookup. |
| 20:17.32 | Defraz | from another db |
| 20:17.41 | p3nguin | eduzimrs: You'll need to look at the m option, I think. |
| 20:17.46 | eduzimrs | p3nguin: exten => 411,1,Authenticate(/etc/asterisk/pass.txt,m) |
| 20:18.40 | eduzimrs | p3nguin: im trying the content of the file is like "bla:1111" |
| 20:20.27 | Defraz | is there a command you send to a sip device that tells it to call out? |
| 20:20.35 | eduzimrs | p3nguin: "bla" is just the name of the person who is going to use the password 1111 |
| 20:20.51 | Defraz | once the sip device is answered? |
| 20:20.54 | _Corey_ | Defraz: manager show command Originate |
| 20:21.09 | p3nguin | eduzimrs: So that's the valid account code? |
| 20:21.12 | Defraz | okay |
| 20:21.14 | Defraz | thanks for that |
| 20:21.26 | _Corey_ | sure |
| 20:21.42 | p3nguin | originate doesn't actually make the phone dial out, but it does dial out and bridge the phone to the call. |
| 20:22.10 | _Corey_ | Well, it should point him in the right direction... |
| 20:22.15 | p3nguin | yep |
| 20:22.20 | eduzimrs | p3nguin: hum, must i set it before authenticate() ? |
| 20:22.26 | p3nguin | I was only clarifying that it didn't make the phone make the call. |
| 20:22.51 | p3nguin | eduzimrs: The account code is set inside Authenticate(). It goes in the file. |
| 20:23.37 | eduzimrs | p3nguin: sry but its hard to me understand account code. |
| 20:23.56 | p3nguin | accountcode is used for accounting or CDR. |
| 20:24.35 | p3nguin | So if your account code is bla, your password is 1111, I guess. |
| 20:24.41 | eduzimrs | yep, but for an example...how should be the file? |
| 20:24.44 | p3nguin | Your devices should be configured with accountcode. |
| 20:25.10 | p3nguin | Wait, I'm mixing two things. |
| 20:25.26 | p3nguin | When you use password 1111, Authenticat() app sets the account code on the channel to bla. |
| 20:27.13 | eduzimrs | yeap but u konw how this file should looks like? |
| 20:27.26 | p3nguin | In the file, I would use something like 762:105245, where my account code is set to 762 if the password I enter is 105245. |
| 20:27.38 | p3nguin | accountcode:password |
| 20:27.40 | p3nguin | one per line. |
| 20:27.50 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:27.50 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 20:28.02 | eduzimrs | this, using the M option right? |
| 20:28.17 | p3nguin | no |
| 20:28.21 | p3nguin | m, not M |
| 20:28.25 | eduzimrs | yeap |
| 20:28.30 | p3nguin | If M even exists, it probably does something else. |
| 20:28.31 | eduzimrs | sry |
| 20:30.53 | eduzimrs | it says "Password Incorrect" |
| 20:31.30 | eduzimrs | p3nguin something is missing |
| 20:32.13 | p3nguin | I'll test it in a minute. |
| 20:32.30 | eduzimrs | ok |
| 20:33.31 | Qwell | I would imagine it's password:acctcode |
| 20:33.44 | eduzimrs | http://pastebin.com/BgXtfknU |
| 20:34.00 | p3nguin | Oh. The help on the app led me to believe otherwise. |
| 20:34.33 | p3nguin | <PROTECTED> |
| 20:34.34 | p3nguin | <PROTECTED> |
| 20:35.11 | Qwell | also it's a hash |
| 20:35.30 | p3nguin | Now that could be the problem he's facing. |
| 20:35.42 | p3nguin | Hash the password before putting it into the file. |
| 20:36.04 | p3nguin | 762:some-long-hash-for-the-password |
| 20:36.11 | p3nguin | where 762 is the account code that will be set. |
| 20:36.34 | eduzimrs | any idea to creat a hash? md5 ? |
| 20:36.38 | Qwell | md5 |
| 20:36.39 | p3nguin | I guess you can hash it with md5 or md5sum, depending on your OS. |
| 20:37.02 | p3nguin | echo -n 1111|md5sum |
| 20:37.52 | eduzimrs | yeap |
| 20:37.54 | eduzimrs | ill try now |
| 20:38.22 | *** join/#asterisk Subdolus (dexterity@creep.bur.st) |
| 20:41.11 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:42.16 | eduzimrs | doesnt work at all |
| 20:42.19 | eduzimrs | ohh |
| 20:42.46 | eduzimrs | just to let u know my * version is 1.4 |
| 20:42.55 | p3nguin | I can't get it to set my account code, but it accepts my password based on the hash. |
| 20:43.00 | eduzimrs | could be a problem? |
| 20:43.19 | eduzimrs | :q |
| 20:44.01 | navaismo | do you use the 'a' option too? |
| 20:44.08 | p3nguin | I didn't. |
| 20:44.32 | p3nguin | I didn't want to set the account code to the password. That seems silly. |
| 20:45.09 | eduzimrs | i didnt |
| 20:46.30 | navaismo | hum, why dont use an agi to search in a DB the pass and set the name attached to that pin? |
| 20:47.08 | *** join/#asterisk Circlefusion (~circlefus@74.142.2.94) |
| 20:47.30 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 20:48.31 | p3nguin | It makes more sense to use the app_authenticate if it will just work right. |
| 20:50.16 | eduzimrs | yeap |
| 20:51.20 | eduzimrs | the "a" option is to set the ACCOUNTCODE var with the password value? |
| 20:51.33 | p3nguin | That's what it indicated to me. |
| 20:51.46 | p3nguin | I'll try it with the m option and see if it behaves differently. |
| 20:52.11 | eduzimrs | ok |
| 20:53.45 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 20:55.19 | p3nguin | options am made it set correctly from the file. |
| 20:55.38 | eduzimrs | hum could u show ? |
| 20:55.48 | p3nguin | Show what? |
| 20:56.11 | eduzimrs | how is you string |
| 20:56.18 | p3nguin | What string? |
| 20:56.32 | navaismo | thats what i say you -- ja-- ignore me np ¬¬ |
| 20:56.38 | eduzimrs | like "exten => 411,2,Authenticate(/etc/asterisk/pass.txt,a)" |
| 20:56.46 | p3nguin | exten => 300,n,Authenticate(/var/lib/asterisk/passwd,am); |
| 20:57.13 | eduzimrs | and you file like "blha:hash" |
| 20:57.43 | p3nguin | 762:81dc9bdb52d04dc20036dbd8313ed055 |
| 20:57.57 | p3nguin | account code 762, hash for passwd 1234 |
| 20:58.01 | eduzimrs | right ill try |
| 20:59.36 | p3nguin | After my authenticate line, I used: exten => 300,n,Verbose(ACCOUNTCODE is ${CDR(accountcode)}); |
| 20:59.51 | ChannelZ | Does anyone else have problems with ChanSpy on DAHDI channels (in my case analog) and using * to switch channels? |
| 21:00.04 | p3nguin | So when I test it, I call 300, enter ins password 1234, then it prints my account code on the cli. |
| 21:00.23 | eduzimrs | humm |
| 21:00.24 | eduzimrs | omg |
| 21:00.29 | eduzimrs | doesnt work for me |
| 21:00.40 | p3nguin | I'd guess you didn't do it like I did. |
| 21:01.11 | eduzimrs | ill post |
| 21:01.58 | p3nguin | It took me two tests. One with option m to see that it didn't work, and one to use options am that did work. |
| 21:03.09 | eduzimrs | http://pastebin.com/hXEaSCNQ |
| 21:03.13 | eduzimrs | i did too |
| 21:03.40 | p3nguin | What do you have on priority 1? |
| 21:03.51 | p3nguin | Just Answer()? |
| 21:03.53 | eduzimrs | yes |
| 21:04.26 | p3nguin | Can asterisk read /etc/asterisk/pass? |
| 21:04.27 | eduzimrs | ur generating 128 bit hash? |
| 21:05.03 | p3nguin | That's what md5 is. |
| 21:05.05 | eduzimrs | root root perm |
| 21:05.13 | p3nguin | But can asterisk READ it? |
| 21:05.37 | eduzimrs | without "am" option and plain text password it works |
| 21:05.40 | eduzimrs | yeap |
| 21:05.48 | eduzimrs | so it reads |
| 21:06.13 | Qwell | What password are you using? The md5 you've used doesn't match any known numeral. |
| 21:06.21 | Qwell | ie; are you creating it properly? |
| 21:06.36 | eduzimrs | yeap: echo -n1 1234|md5sum |
| 21:06.37 | p3nguin | The one I pasted above is passwd 1234. |
| 21:06.42 | p3nguin | -n1? |
| 21:06.51 | Qwell | ... |
| 21:06.59 | p3nguin | I gave you the hash for 1234 above. |
| 21:07.03 | p3nguin | I also told you how to generate the hash. |
| 21:07.10 | Qwell | You realize that the output for `echo -n1 1234` is "-n1 1234" ? |
| 21:07.23 | p3nguin | (1537.01) <p3nguin> echo -n 1111|md5sum |
| 21:07.43 | eduzimrs | yeap |
| 21:07.50 | eduzimrs | i did = |
| 21:07.59 | eduzimrs | supress n1 so? |
| 21:07.59 | p3nguin | So I used echo -n 1234|md5sum and created my passwd hash. |
| 21:08.08 | Qwell | where are you getting the 1 from? |
| 21:08.20 | p3nguin | pulled directly from his ass, I'd imagine. |
| 21:08.48 | eduzimrs | oh |
| 21:09.00 | eduzimrs | sry about that |
| 21:09.04 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:09.38 | p3nguin | It's like speaking Braille to a deaf person. |
| 21:10.19 | eduzimrs | ahaha |
| 21:10.50 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 21:11.07 | eduzimrs | i agree it was awful |
| 21:11.34 | eduzimrs | but in the end worked! |
| 21:11.45 | eduzimrs | and tks for helping |
| 21:11.50 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:12.24 | eja | does an asterisk reload affect current calls? |
| 21:13.56 | leifmadsen | no |
| 21:13.59 | p3nguin | Don't run reload. Reload what you need to reload. dialplan reload, sip reload, moh reload, voicemail reload |
| 21:14.06 | leifmadsen | +1 |
| 21:14.45 | p3nguin | If you run dialplan reload during a call, any changes to the extension of the current calls will be affected as the call progresses in dialplan, though. I think that's pretty cool. |
| 21:17.24 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:22.32 | Hive | p3nguin, you dont like the general reload? |
| 21:22.44 | p3nguin | No, and neither should you. |
| 21:23.13 | Hive | I wasn't even aware that there was anything other than the "reload" command :X |
| 21:23.20 | p3nguin | ~book |
| 21:23.20 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 21:23.33 | *** join/#asterisk senator (lebbeous@nox.esilibrary.com) |
| 21:25.42 | senator | hi all. if i drop two call files in the spool directory for the pbx_spool, and those files specify the same dahdi channel (but different phone numbers), asterisk tries to make two calls at the same time. i would have expected (incorrectly?) that asterisk would try one call after the other. can someone confirm what's supposed to happen? i was fairly sure i'd seen the one-after-the-other behavior |
| 21:25.48 | senator | in the past. |
| 21:26.18 | senator | this is with asterisk 1.8.5.0 |
| 21:29.46 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 21:29.54 | *** join/#asterisk _trine (~trine@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 21:30.31 | _trine | can anyone enlighten me why I see this from Asterisk >> Function MASTER_CHANNEL not registered |
| 21:31.51 | navaismo | senator i guess if one failed and you set the retry it will retry with your parameters |
| 21:32.03 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 21:32.28 | wdoekes2 | _trine: because func_channel is not loaded? module show like channel |
| 21:32.36 | d_preston215 | How does High-Availability/Clustering work in AsteriskNOW? |
| 21:33.19 | _trine | wdoekes2, it does it everytime I dial a number but it still works |
| 21:34.42 | senator | navaismo: retry does work, yes, so the call that failed will still happen later. still, if a lot of files are placed in the spool directory at once, i should think they would eventually run out of retry time before they could all finish. is pbx_spool not meant to line up calls using the same channel one after the other? |
| 21:35.02 | eja | there is no voicemail reload which is what i need reloaded. unless it reads the voicemail.conf file everytime? |
| 21:36.03 | p3nguin | module reload app_voicemail.so |
| 21:36.39 | eja | thanks! |
| 21:36.50 | p3nguin | voicemail reload does exist in 1.8 branch. |
| 21:36.53 | p3nguin | just for info |
| 21:38.09 | navaismo | senator i dont know |
| 21:39.49 | navaismo | if you have your dahdi chans in a group you can use rX where X is the group number |
| 21:41.28 | senator | navaismo: i can put them into groups. can you tell me what the r is about? |
| 21:43.21 | *** join/#asterisk manji (~manjiki@ppp-2-84-11-158.home.otenet.gr) |
| 21:43.25 | *** join/#asterisk talntid (~erict@li93-153.members.linode.com) |
| 21:44.07 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mqvwyqutqtrpfhun) |
| 21:44.12 | navaismo | is for rotate the channels |
| 21:44.17 | navaismo | in a group |
| 21:44.33 | navaismo | the first call use chan 1 the next chan 2 and so on |
| 21:44.48 | talntid | Anyone interested in making a custom dialplan for me? I'll pay fairly for it. Basically, I want a custom speed-dial system.... they press 1-whatever for custom speed dials, * to setup... setup allows them to change the # and speed dial of everything... without changing the dialplan... |
| 21:45.13 | Qwell | talntid: honestly, sounds like just a few minutes with func_odbc |
| 21:45.27 | talntid | i figured it's probably pretty simple |
| 21:45.41 | Qwell | or even astdn |
| 21:45.43 | Qwell | astdb* |
| 21:45.53 | talntid | astdb would likely be pretty useful :) |
| 21:45.59 | talntid | that's how I was thinking of doing it |
| 21:46.00 | Qwell | much less setup required |
| 21:46.56 | senator | navaismo: ok thanks |
| 21:47.24 | Qwell | *.,1,Read(myspeeddial,someprompt) ; *,2,Set(DB(speeddial/${EXTEN:1})=${myspeeddial}) |
| 21:47.59 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-wmvqjxoyuslukkue) |
| 21:48.17 | Qwell | 1.,1,Set(dialnum=${DB(speeddial/${EXTEN:1})} ; 1.,2,Dial(Local/${dialnum}@somecontext) |
| 21:54.03 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
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| 22:53.07 | *** join/#asterisk fauxalliance (~da_lep@142.163.151.207) |
| 22:53.15 | dijib | what is the best fax solution for * including fax to email |
| 22:53.25 | p3nguin | Over IP? |
| 22:53.29 | dijib | yeh |
| 22:53.44 | p3nguin | I'm pretty satisfied with my method, which uses fax for asterisk. |
| 22:53.44 | *** part/#asterisk fauxalliance (~da_lep@142.163.151.207) |
| 22:53.49 | dijib | i want to recieve and have it email me, and i want a sendto email address to send |
| 22:54.02 | dijib | i like out of the box methodes myself |
| 22:54.11 | dijib | thats one thing i want to achieve |
| 22:54.41 | p3nguin | http://pastebin.com/6RQV9nEx |
| 22:54.41 | dijib | how do i make a ringing sound durring a call to indicate the call is ringing somewhere? |
| 22:54.51 | p3nguin | Ringing() |
| 22:55.37 | dijib | thats it? lol |
| 22:55.48 | *** part/#asterisk senator (lebbeous@nox.esilibrary.com) |
| 22:55.51 | p3nguin | Ringing() sends the ringing indication to the other side. |
| 22:56.17 | dijib | remember that press 1 for.... or anything else. thing. after the person presses 1 i want it to ring till voicemail picksup |
| 22:56.30 | dijib | sup |
| 22:56.54 | dijib | other side is the callee or caller? |
| 22:57.11 | p3nguin | always the caller |
| 22:57.23 | dijib | i thought i tried that and it didnt work. |
| 22:57.28 | dijib | let me try again |
| 22:57.29 | p3nguin | Are you wanting to call someone and then play a ringing sound to them? |
| 22:57.40 | dijib | yes. |
| 22:58.00 | dijib | like after you press 1, it is dead air till either i pickup or voicemail does |
| 22:58.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:58.04 | dijib | for like 20sec |
| 22:58.18 | p3nguin | Ringing() sends the ringing indication. If you want to play ringing sound, you can either use Ringing() followed by Wait(x) where x is the amount of seconds to ring, you can use use PlayTones(ring). |
| 22:58.44 | dijib | playtones ring i think might be what i need |
| 22:59.07 | p3nguin | I use Ringing() then Wait(20) to play 20 seconds worth of ringing sound. |
| 22:59.22 | pabelanger | Is this over SIP? |
| 22:59.24 | dijib | what happens if i pickup durring that 20sec? |
| 22:59.27 | dijib | yes |
| 22:59.29 | carrar | dijib, dial can also force the ringing |
| 22:59.41 | dijib | maybe im using goto.... |
| 22:59.42 | p3nguin | I don't know what that means "pickup durring that 20sec." |
| 22:59.49 | carrar | <PROTECTED> |
| 22:59.50 | carrar | <PROTECTED> |
| 22:59.58 | p3nguin | You typically do not call someone and then play ringing sounds to that person. |
| 23:00.08 | dijib | lol |
| 23:00.16 | pabelanger | Honestly, you should avoid using Ringing() for SIP, and figure out why your devices are not ringing |
| 23:00.31 | p3nguin | Both of you didn't read what he said. |
| 23:00.39 | p3nguin | He wants to call someone and then play ringing sounds to them. |
| 23:00.52 | carrar | <dijib> like after you press 1, it is dead air till either i pickup or voicemail does |
| 23:00.52 | dijib | its not the device the call is going to. its the person on the line that is waiting for the device to be answered |
| 23:01.12 | carrar | assuming you press 1 and it dials someone |
| 23:01.18 | p3nguin | You don't need to play ringing sounds for that. |
| 23:01.32 | carrar | obviously |
| 23:01.37 | dijib | well i do as there is nothing till i pickup |
| 23:01.42 | dijib | i want it to sound like ringing |
| 23:01.51 | p3nguin | You can call one person, and then when you dial another phone, it'll ring. |
| 23:02.44 | pabelanger | Well, if he is dialling over SIP, the far end should indicate 180 RINGING rather then asterisk generating it |
| 23:03.04 | p3nguin | I'd rather not have you calling out to ask someone to press a button, though. That really pisses off a lot of people. |
| 23:03.39 | p3nguin | That, and so does, *ring* (I answer) "Please hold for an important message..." |
| 23:03.43 | dijib | ok someone calls me from landline --> ITSP --> *, they are told to press 1, it then dials my devices (they ring) but the caller has no sound indicating the lines ringing or what its doing. until i pickup or voicemail does... make sense? |
| 23:03.56 | p3nguin | That makes sense. |
| 23:04.21 | p3nguin | Now figure out why your devices aren't sending the ringing indicator, like pabelanger said. |
| 23:04.51 | dijib | wait the device sends the line the ring? |
| 23:04.55 | dijib | really? |
| 23:05.10 | p3nguin | If you call me, my phone is supposed to give you the ringing indicator. |
| 23:05.38 | gogasca | that looks like a progress indicator issue |
| 23:06.08 | dijib | so now im wondering if its * or pap2t |
| 23:06.19 | gogasca | i think what happens is that call already in a connected state |
| 23:06.32 | gogasca | so "ringing" should be played from the source |
| 23:06.37 | gogasca | sorry destination |
| 23:06.40 | gogasca | remote device |
| 23:06.40 | dijib | gogasca, i think your right |
| 23:06.46 | p3nguin | I think his left. |
| 23:06.52 | gogasca | haha |
| 23:06.54 | dijib | lol |
| 23:07.10 | dijib | ok i think p3nguin is right in that left case |
| 23:07.11 | p3nguin | his right, his left... it's nearly the same thing. |
| 23:07.21 | pabelanger | no, the actually ring tone is from your local phone. The far end will send a RINGING event, which causes your local phone to generate the tone |
| 23:07.31 | gogasca | not really |
| 23:07.32 | gogasca | in this case |
| 23:07.37 | gogasca | alreayd went to a connect state |
| 23:07.47 | gogasca | when they instruct caller to press 1 |
| 23:07.55 | dijib | yes its already c onnected inb4 the IVR saying press 1 |
| 23:08.12 | gogasca | so keeps in connected state then asterisk should play back a ringing file |
| 23:08.19 | p3nguin | If I call my main number and get a prompt, the line has already been answered. If I then dial an extension which dials a phone, I get ringing sounds again. |
| 23:08.39 | dijib | ok so playtones(ring) ? do i need a timeout? match voicemail timeout? |
| 23:08.51 | pabelanger | app_disa |
| 23:09.02 | dijib | app_disa is installed |
| 23:09.14 | pabelanger | well, maybe not... |
| 23:09.34 | p3nguin | You don't need disa for that. |
| 23:09.47 | p3nguin | You don't need disa in a lot of cases where people insist on using disa. |
| 23:10.30 | p3nguin | Figure out if there's no 180 Ringing indicator when you hit 1 to dial the phone. |
| 23:10.36 | p3nguin | There should be one. |
| 23:10.37 | pabelanger | ya, ignore what I said about disa |
| 23:10.44 | pabelanger | needs food |
| 23:10.45 | gogasca | no pengu3in |
| 23:10.47 | pabelanger | & |
| 23:10.50 | gogasca | rining was way before |
| 23:10.57 | gogasca | call is already connected state |
| 23:11.03 | p3nguin | If you Dial() a phone, you'd better get a 180 Ringing from it. |
| 23:11.06 | gogasca | there is a rining from * to new extension |
| 23:11.09 | p3nguin | If you don't, it's fucked up. |
| 23:11.14 | gogasca | but not from * to remote end |
| 23:11.21 | gogasca | the issue is form * to remote end |
| 23:11.22 | p3nguin | Ringing is Ringing. |
| 23:11.38 | p3nguin | It sounds the same in either case. |
| 23:12.01 | dijib | you two are making me dizzy |
| 23:12.31 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:12.47 | gogasca | yeah i know what the issue, just not sure how to fix it |
| 23:13.05 | gogasca | its a common no rining when transfer |
| 23:13.10 | gogasca | ringing |
| 23:13.22 | p3nguin | You could use the r dial option like carrar mentioned to force a ringing sound, but most sensible people do not like that option. |
| 23:13.24 | dijib | should i try playtones? |
| 23:13.26 | p3nguin | no |
| 23:13.36 | p3nguin | Ringing() right before the Dial() |
| 23:13.40 | p3nguin | See what happens. |
| 23:13.47 | p3nguin | That'll send the 180. |
| 23:14.05 | dijib | im actually using that right now p3nguin and it doesnt do anything |
| 23:14.27 | p3nguin | Did you check for the 180 with and without it? |
| 23:14.35 | dijib | whats a 180? |
| 23:14.37 | dijib | :/ |
| 23:14.41 | p3nguin | Ringing |
| 23:15.42 | gogasca | 180 is the code for SIP for ringing |
| 23:15.46 | gogasca | hey DJ |
| 23:15.59 | gogasca | i have an ivr and works fine, let me see what im invoking |
| 23:16.01 | *** part/#asterisk _trine (~trine@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 23:16.06 | gogasca | when user hears rining |
| 23:16.10 | gogasca | afteri being transfer |
| 23:16.14 | gogasca | it works fine 4 me |
| 23:16.48 | p3nguin | I just Dial() the phone requested. I only sometimes don't get a ringing sound. |
| 23:19.22 | gogasca | DJ I connected to my IVR and then after 1 in asterisk logs I see this: |
| 23:19.24 | gogasca | <PROTECTED> |
| 23:19.24 | gogasca | <PROTECTED> |
| 23:19.35 | gogasca | i use freepbx but concept is the same |
| 23:20.03 | gogasca | after 1, the extension that i need to reack is 101 |
| 23:21.21 | p3nguin | Typical for FreePBX to use the r option. |
| 23:21.59 | p3nguin | exten => 201,n,Dial(${DEVICE}/ringer=${ringer},30,w); |
| 23:22.03 | p3nguin | Notice there's no r. |
| 23:25.05 | gogasca | so DJ what are you using? |
| 23:27.47 | gogasca | her Mr DJ |
| 23:28.34 | dijib | im using CentOS6 *1.8x.x.xsomething |
| 23:29.15 | dijib | and the r option in dial doesnt work, the ringing() before dial doesnt work. and i put playtones after dial but that didnt work, but i think that was after. |
| 23:29.30 | dijib | after where i needed it |
| 23:31.02 | p3nguin | Nothing after the dial ever gets used unless you use the g option. |
| 23:32.02 | dijib | is that for goatsie? |
| 23:32.08 | dijib | or go to? |
| 23:32.10 | ChannelZ | yes. yes it is. |
| 23:34.46 | dijib | ok another thing i was wondering is in MOH can you have mpg123 make a connection to a stream on demand instead of all the time. |
| 23:34.56 | p3nguin | yes |
| 23:35.03 | dijib | hmmm |
| 23:35.16 | dijib | how would one do that? |
| 23:35.26 | p3nguin | well, wait |
| 23:35.43 | dijib | (600) |
| 23:35.55 | p3nguin | Someone was talking about this recently. |
| 23:36.08 | p3nguin | I think they ended up being able to do it. |
| 23:36.11 | dijib | i tried to a couple days back |
| 23:36.41 | p3nguin | I don't worry about it streaming all the time, so I never bothered to try it. |
| 23:38.30 | dijib | i just want to also keep the CPU load down on this p4 laptop im using as a server |
| 23:39.34 | p3nguin | If one mpg123 stream is costing you much CPU, you've got bigger problems. |
| 23:39.56 | dijib | i dont, i just want to keep everything as minimal as i can |
| 23:40.17 | dijib | plus i want multiple available streams |
| 23:40.56 | dijib | and i have had instances where mpg123 crashes and if it was realtime execution of mpg123 it should always have the stream baring the internet is down |
| 23:42.07 | *** join/#asterisk gxdssoft (~gxdssoft@190.235.98.26) |
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| 23:46.11 | dijib | whats the site with the logs from this IRC channel>? |
| 23:46.19 | dijib | im going to try and find this p3nguin |
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| 23:51.07 | dijib | im thinking everyone went for dinner |
| 23:53.28 | talntid | not me |
| 23:53.31 | talntid | sunflower seeds for me :P |
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