| 05:39.02 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 05:39.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.0 (2013/07/15), 10.12.2 (2013/03/27), 1.8.23.0 (2013/07/15), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 06:01.15 | floren | yawns |
| 06:01.33 | floren | 2am here time to go to bed and watch a little bit of Sopranos :) |
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| 10:00.54 | FlashDeluxe | hi! i updated asterisk 1.8.x to asterisk 11-current yesterday and i got a problem: two of my sip phones do not hear the caller from an incoming call. If they dial out, everything works fine. I didn`t change anything in the config and just two phones are affected... I made a call with sip debug on, maybe somebody can help me? http://pastebin.com/wHRC9auZ |
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| 10:33.27 | NiugeS | hi all.. |
| 10:33.31 | NiugeS | anyone know of a right click to call extension for chrome? |
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| 11:50.31 | BorjaGVO | Hi. I just uninstalled asterisk 1.8.11 and installed 1.8.23 |
| 11:52.14 | BorjaGVO | When I try to run asterisk (service asterisk start) it starts but with following error: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" /var/run/asterisk/asterisk.ctl exists |
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| 12:21.37 | Gugge | BorjaGVO: sounds like you somehow got the startup script to use -r |
| 12:21.42 | Gugge | dont do that :) |
| 12:22.32 | BorjaGVO | Gugge: why? |
| 12:23.50 | Cuzner | lol |
| 12:24.12 | Cuzner | i only caught the last 3 lines of that conversation, but it made me giggle. |
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| 12:44.25 | Greenlight | Quick question. If Dial() to a SIP peer times out will it automatically retry? |
| 12:45.04 | leifmadsen | no |
| 12:45.11 | leifmadsen | it will continue at the next line of dialplan execution |
| 12:45.16 | Greenlight | Ahh okay cool thanks |
| 12:45.23 | leifmadsen | simple test :) |
| 12:45.31 | Cuzner | gm leifmadsen |
| 12:45.41 | Greenlight | We're getting an ACK to the INVITE then nothing else |
| 12:45.56 | Greenlight | And in the SIP trace I'm then seeing another INVITE going out |
| 12:46.14 | Greenlight | Guess it's something fancy in the dialplan |
| 12:49.03 | [TK]D-Fender | to retry? Nothing so fancy... |
| 12:49.15 | [TK]D-Fender | as for the failure... that is its own thing |
| 12:49.21 | Greenlight | Naa there's a bit block of code which does an SRV lookup in AEL |
| 12:50.16 | Greenlight | The failure is my carrier, but it appears from SIP trace that after every failure we're sending the retries to a single node |
| 12:50.25 | Greenlight | And killing that node ;/ |
| 12:50.44 | Greenlight | I just wasn't 100% sure if Dial by itself retried |
| 12:51.17 | [TK]D-Fender | Get another carrier |
| 12:51.42 | [TK]D-Fender | There is (was?) "RetryDial" .... not sure on the particulars though |
| 12:52.07 | Greenlight | They're generally quite reliable. I think some of their interconnets to BT have went down this morning though |
| 12:52.34 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 12:52.44 | Greenlight | And so, they're seeing my dump 1000+ calls to a single node |
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| 12:54.00 | linocisco | hi |
| 12:54.16 | linocisco | how can make asterisk deliever call to sale workers outside? |
| 12:54.36 | Greenlight | Dial() their mobiles ? |
| 12:55.44 | linocisco | asterisk can transfer calls to telecomuters with pay per call using PSTN ? |
| 12:55.46 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.192) |
| 12:55.54 | linocisco | Greenlight, yes. dial their mobile |
| 12:56.40 | linocisco | Greenlight, for example for tech support guys, they may solve problems on phone while on the road or may know what is important to go as next destination |
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| 13:01.12 | [TK]D-Fender | linocisco: "core show application dial" <---- |
| 13:01.25 | [TK]D-Fender | linocisco: Every call is just a call. |
| 13:01.43 | linocisco | [TK]D-Fender, is it related? system should automatically call itself |
| 13:01.58 | [TK]D-Fender | linocisco: Your dialplan call whatever you tell it to call |
| 13:02.31 | linocisco | [TK]D-Fender, is it related? system should automatically call itself or send sms as voice mail or something like that if they are busy |
| 13:02.32 | [TK]D-Fender | linocisco: Got a SIP phone? Dial it? Got an analog line, dial out it. Got an ISDN circut? Dial out it. |
| 13:02.44 | linocisco | [TK]D-Fender, thanks bro. is there any reference on this how to? |
| 13:02.51 | [TK]D-Fender | linocisco: "core show application dial" <---- |
| 13:03.11 | WIMPy | linocisco: Have you ever made any call with Asterisk? |
| 13:03.13 | [TK]D-Fender | linocisco: Have you even installed Asterisk yet? I've never seen anyone who has... and has not set it to DIAL a device |
| 13:03.39 | [TK]D-Fender | ~book |
| 13:03.39 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:03.41 | [TK]D-Fender | ^^^ |
| 13:03.57 | linocisco | [TK]D-Fender, i want to make a list of Mobile no. of tech guys or telecomuters. and whenever asterisk get a call, it should automatically transfer |
| 13:04.14 | [TK]D-Fender | linocisco: Then go do it. Call comes in, call goes out. |
| 13:05.28 | linocisco | [TK]D-Fender, thanks anyway |
| 13:06.50 | [TK]D-Fender | linocisco: Dial(SIP/provider/thguysnumber) |
| 13:07.02 | [TK]D-Fender | linocisco: Dial(DAHDI/g1/thguysnumber) |
| 13:07.09 | [TK]D-Fender | linocisco: Every call out is just a DIAL |
| 13:08.23 | linocisco | [TK]D-Fender, we have only PSTN lines connected to asterisk, so as long as the conversation long, transfer calls will be --$$$. but no choice |
| 13:09.06 | [TK]D-Fender | linocisco: Asterisk only has the things you give it to work with |
| 13:09.38 | linocisco | [TK]D-Fender, thanks bro. btw, what is the fax server to be used with asterisk? Hylax? |
| 13:09.54 | linocisco | [TK]D-Fender, no other alternatives opensource and free? |
| 13:10.22 | WIMPy | That also depends on where you want to receive them from. |
| 13:11.39 | [TK]D-Fender | linocisco: Depends on how you define "fax server". I'm sure if you google that you'll find others. |
| 13:11.41 | linocisco | WIMPy, i would like to send and receive fax from computer as email attachment |
| 13:11.50 | [TK]D-Fender | linocisco: And those do not have anything to do with Asterisk directly |
| 13:12.08 | linocisco | [TK]D-Fender, what devices we need to interface? |
| 13:12.14 | [TK]D-Fender | linocisco: * has several more native APPS for the basic receive and send function (fvia image files) |
| 13:13.05 | linocisco | thanks all. c u all. gotta go |
| 13:13.06 | [TK]D-Fender | linocisco: receiving a fax is a single dialplan app like ReceiveFax(), etc, and e-mail-ing typically by a singe shell-script command to sendmail, etc |
| 13:13.15 | WIMPy | That was probably as vague as possible. |
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| 13:14.28 | WIMPy | I thing the best solution is hylafax with the capi plugin. But that will obviousely only work if you have a capi. |
| 13:14.48 | [TK]D-Fender | WIMPy: English is way down his language list and is terrible at descriptions and terminally lazy. |
| 13:15.26 | igcewieling | The first item is excusable, the others are not. |
| 13:16.29 | [TK]D-Fender | He's been at this for over a year now and doesn't seem to have the ability to Google worth a damn |
| 13:17.01 | [TK]D-Fender | Actually a sickening number of people coming in here couldn't google up 4 simple words to find the answer to their problems. |
| 13:17.52 | WIMPy | Well, sometimes the hardest part is to find the right terms. Especially with Asterisk. |
| 13:19.19 | [TK]D-Fender | "asterisk receive fax and send by e-mail" <- |
| 13:19.21 | [TK]D-Fender | Oh please |
| 13:19.40 | [TK]D-Fender | Sorry... no slack to cut for this |
| 13:19.40 | WIMPy | didn't say that was one of these cases. |
| 13:20.26 | [TK]D-Fender | "open source fax server" |
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| 13:27.09 | apb1963 | googles "Open Sesame" |
| 13:28.24 | apb1963 | discovers Chicken shawarma pita |
| 13:28.24 | igcewieling | [TK]D-Fender: it is OK to ignore people. 8-| |
| 13:29.23 | [TK]D-Fender | Too often if you do you keep seeing them come back with the same pointless questions. Better to nip it in the bud and hope they learn something... |
| 13:30.42 | jmetro | american cheese w/ mustard on it |
| 13:30.48 | apb1963 | Open SesameĀ® is a high-phosphorus soluble fertilizer that's designed to bring plants into bloom at your command. |
| 13:30.50 | jmetro | bestfast |
| 13:31.03 | apb1963 | wow. A food AND a fertilizer. Who knew? |
| 13:31.30 | igcewieling | Most food eventually becomes fertilize. |
| 13:31.32 | igcewieling | r |
| 13:31.42 | igcewieling | 8-| |
| 13:31.45 | apb1963 | True... but generally it needs to be processed first |
| 13:32.27 | apb1963 | Besides... I thought it was just a way to open hidden caves. |
| 13:32.55 | apb1963 | Now I am empowered in more ways than one. |
| 13:33.22 | FlashDeluxe | hi folks, i got a short question, i want to execute a bash command after a fax is received, in asterisk 1.8 i did it with System(/bash/script), but that doesn`t work anymore, now after googleling, i am trying it with SHELL(/bash/script) but that won`t work either... Am i missing something? My config entry: http://pastebin.com/GmLi5nPz |
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| 13:35.58 | apb1963 | FlashDeluxe: How do you know it doesn't work? |
| 13:36.17 | [TK]D-Fender | [09:30]jmetroamerican cheese w/ mustard on it <- that's right, and not one of those fancy crazy euro mustard like dijon either. No, for this guy only something solidly 'MURICAN like .... FRENCH'S :p |
| 13:36.33 | jmetro | HELL no. Guldens spicy brown mustard, :3 |
| 13:36.37 | apb1963 | I like Gulden's style |
| 13:36.39 | apb1963 | yeah! |
| 13:36.52 | apb1963 | But it doesn't have to be Gulden's |
| 13:37.13 | apb1963 | Although yellow mustard is good from time to time... sometimes on hot dogs |
| 13:37.17 | jmetro | Guldens has a noticeable flavor difference from other spicy browns. Its like the difference from Miracle whip to Hellmans mayonnaise [bring out the best] |
| 13:37.18 | [TK]D-Fender | FlashDeluxe: I guess you're not really looking at what's happening in your call... |
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| 13:37.49 | apb1963 | Really? I can't tell the difference. Now Mayo vs. Miracle whip... yeah, big difference. |
| 13:37.56 | [TK]D-Fender | jmetro: Miracle Whip ... is not even close to being mayonaise... let alone Hellmans |
| 13:38.05 | jmetro | I like the french's yellow on hotdogs, but cheese & mustard has to be guldens |
| 13:38.22 | jmetro | [making me so hungry now] |
| 13:38.26 | apb1963 | Did you know they call Hellmans' "Best Foods" west of the Rockie's? Took me years to find it in the store. |
| 13:38.46 | apb1963 | Found it totally by accident. |
| 13:38.54 | BorjaGVO | What direcories can I back up from 1.8.15.1 to 1.8.23.0? I mean, I have installed 1.8.23.0 and set up following directories from 1.8.15.1: /var/lib/asterisk /var/spool/asterisk /etc/asterisk. Somehoe, it seems that there is some kind of conflict as things are working oddly. |
| 13:39.07 | jmetro | My mother-in-law used my entire jar of mayonnaise once and bought me "Burmans" mayonnaise as a replacement. I'm afraid of getting botulism |
| 13:39.14 | apb1963 | lol |
| 13:39.36 | jmetro | http://tinyurl.com/lxn97xk |
| 13:39.40 | jmetro | look familiar? |
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| 13:39.52 | [TK]D-Fender | BorjaGVO: There is no significant difference between those 2 versions... |
| 13:40.11 | [TK]D-Fender | BorjaGVO: And your description of "oddly" tells us nothing. Show us an actual problem. |
| 13:40.12 | WIMPy | BorjaGVO: They should all work the same. If you installed it the same. |
| 13:40.13 | apb1963 | whoa... look & feel |
| 13:40.42 | jmetro | except the burmans was like 1$ a gallon |
| 13:40.54 | apb1963 | that's why she bought it :) |
| 13:41.02 | apb1963 | I would have done the same.... at least to try it. |
| 13:41.15 | BorjaGVO | [TK]D-Fender: I will, let me diagnose as far as I can. Then I'll come back ;-) |
| 13:41.17 | apb1963 | for a buck hell |
| 13:41.23 | WIMPy | Apparently you're not allowed to say Mayonaise any more except for a certain brand. Its sold as fies sauce here now. |
| 13:41.33 | apb1963 | fly sauce? |
| 13:42.04 | [TK]D-Fender | "fries" I'm betting |
| 13:42.16 | WIMPy | I don't think you can fly on it. By maybe it can attract flies. |
| 13:42.19 | jmetro | fries dipped in mayonnaise....ugh |
| 13:42.23 | apb1963 | ewwww.... mayo on frys? |
| 13:42.25 | WIMPy | Yes. "fries". |
| 13:42.32 | [TK]D-Fender | yup, very european... |
| 13:42.37 | igcewieling | jmetro: that is VERY common all around the world. |
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| 13:42.40 | BorjaGVO | [TK]D-Fender, WIMPy I know it is working worng because yesterday I put it on production and it didn't work as expected (incoming calls from trunk were being hung up, etc) |
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| 13:42.45 | WIMPy | It's very popular here. Don't know why. |
| 13:42.47 | igcewieling | It is great on sweet potato fries |
| 13:42.49 | apb1963 | that's like ketchup on.... feet |
| 13:42.53 | Greenlight | A quick question regarding AEL dialplan. It seems that behind the scenes this gets converted into a *normal* dialplan; is this correct? And if so, is there any way to see what's being generated so that I can step though it more easily? |
| 13:42.59 | [TK]D-Fender | Predominately French/Belgian |
| 13:43.01 | jmetro | Third-world countries also like to eat boiled half-grown duck eggs :3 |
| 13:43.17 | Vann | =/ |
| 13:43.19 | igcewieling | Greenlight: 1) yes. 2) yes |
| 13:43.29 | apb1963 | da hell is a half grown egg? |
| 13:43.30 | Greenlight | Cool |
| 13:43.33 | [TK]D-Fender | Greenlight: "dialplan show" and yes... AEL is nothign magical, it all gets parsed back, so support will suck |
| 13:43.38 | jmetro | apb1963: google "Balut" |
| 13:43.47 | apb1963 | I will not |
| 13:43.47 | Greenlight | Oh, cool, didn't even think of that. Thanks :) |
| 13:43.57 | igcewieling | Greenlight: there are a lot of AEL haters around, don't let them get to you. AEL IS AWESOME! |
| 13:44.11 | apb1963 | googles ael instead |
| 13:44.29 | Katty | Qwell: YOU |
| 13:44.32 | Greenlight | Yea, it does seem really cool. But when debugging its tricky as it's not stepping through the AEL stuff |
| 13:44.43 | Greenlight | If that makes sense |
| 13:44.52 | WIMPy | AEL just like dialplan with one more step to debug. |
| 13:44.54 | Greenlight | Like it's going a load of GoToIf's etc |
| 13:44.55 | igcewieling | Greenlight: indeed, debugging is harder, but in my opinion worth it |
| 13:45.27 | jmetro | debugging is precisely why i wont program in c++ |
| 13:45.28 | igcewieling | Greenlight: with practice you'll start to see the connection between the AEL and the generated dialplan. |
| 13:46.02 | Greenlight | Yea, definetly, especially now that I can see the "missing link" :) |
| 13:46.07 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.37.86) |
| 13:46.55 | igcewieling | Greenlight: also, don't try to do EVERYTHING in AEL. We have a mixture of normal dialplan and AEL macros. |
| 13:48.11 | Greenlight | Yea, I've most my simple dialplan stuff in normal, but I've an AEL dialplan which handles dialling to use SRV records (kindly dontated to me by someone in this channel) |
| 13:48.24 | igcewieling | one of the macros we used to use http://pastebin.com/rsD3qJNX |
| 13:48.40 | igcewieling | Greenlight: That SRV was from me |
| 13:48.47 | *** join/#asterisk b0scage (boscage@unaffiliated/boscage) |
| 13:48.50 | Greenlight | Ahh cool :) |
| 13:49.08 | Greenlight | I did have to make some small alterations, but aside from that it's been working sweet... till today :) |
| 13:49.10 | jmetro | igcewieling: something wrong with the built-in saydigits ? |
| 13:49.24 | igcewieling | jmetro: yes, it doesn't pause when it sees a - |
| 13:49.36 | jmetro | good point |
| 13:50.03 | igcewieling | 212-555-1212 is spoken as 212 because 555 pause 1212 and I like saydigits to say it with pauses because it is easier to "hear" |
| 13:50.20 | igcewieling | s/because/cause/ |
| 13:50.25 | jmetro | and write down numbers |
| 13:50.28 | igcewieling | screw it, I'm getting more coffee |
| 13:50.37 | jmetro | get me a white chocolate mocha |
| 13:50.41 | jmetro | ploxboxrox |
| 13:50.56 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 13:51.01 | igcewieling | jmetro: we say most of our digits from inside an AGI so the macro isn't really used anymore |
| 13:51.04 | apb1963 | what kind of coffee? |
| 13:51.31 | igcewieling | apb1963: 8 O'Clock Dark Roast, fresh ground, brewed by the cup |
| 13:51.34 | apb1963 | any is max and index initialized like a string? |
| 13:51.48 | apb1963 | s/any/and why/ |
| 13:51.58 | apb1963 | s/is/are/ |
| 13:51.59 | FlashDeluxe | [TK]D-Fender: What do you mean exactly? |
| 13:52.01 | igcewieling | apb1963: because there is only strings in Asterisk. 8-) |
| 13:52.01 | Kobaz | haha jeez |
| 13:52.15 | Kobaz | this customer says "we've been trying to get into this voicemail box for days, we have tried everything" |
| 13:52.17 | apb1963 | s/any/and/ |
| 13:52.23 | Kobaz | the password for 4291 is 4291 |
| 13:52.25 | apb1963 | s/is/why is/ |
| 13:52.27 | Kobaz | how could they have not tried that |
| 13:52.35 | apb1963 | s/any/and/ |
| 13:52.39 | apb1963 | hmm |
| 13:52.43 | apb1963 | is it me? |
| 13:52.43 | jmetro | because having the password be the mailbox is terrible idea |
| 13:52.58 | jmetro | something like 7412369 is way better |
| 13:53.00 | apb1963 | i shouldn't be up this early |
| 13:53.05 | igcewieling | apb1963: the code is pretty old and may not work aymore. I see what you mean, likely a bug |
| 13:53.25 | apb1963 | igcewieling: ever heard of MaCaw brand coffee? |
| 13:53.42 | Kobaz | jmetro: heeey that's my password |
| 13:53.44 | igcewieling | apb1963 nope |
| 13:53.53 | Kobaz | jmetro: but yes that is a terrible idea, but that wasn't my choice |
| 13:54.00 | apb1963 | igcewieling: that's because it's not on the market yet |
| 13:54.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.28) |
| 13:54.51 | apb1963 | how about MaCaw brand chocolate? |
| 13:55.16 | igcewieling | jmetro: there are voicemail.conf settings which kick in and prompt the user to record the greetings and set their password if the password=the mailbox |
| 13:55.29 | [TK]D-Fender | [09:51]FlashDeluxe[TK]D-Fender: What do you mean exactly? <- by what? |
| 13:55.35 | jmetro | igcewieling: oh really? Which one makes them change the password off default? |
| 13:55.47 | apb1963 | where'd that come from? I thought he disappeared. |
| 13:56.05 | apb1963 | oh, there he is |
| 13:56.08 | apb1963 | missed that |
| 13:56.26 | igcewieling | jmetro:grep "force" /usr/src/asterisk/configs/sip.conf.sample |
| 13:56.40 | FlashDeluxe | [TK]D-Fender: "[15:37] <[TK]D-Fender> FlashDeluxe: I guess you're not really looking at what's happening in your call..." |
| 13:56.52 | jmetro | igcewieling: ah its in sip.conf? |
| 13:57.07 | igcewieling | voicemail.conf |
| 13:57.19 | igcewieling | goes back to doing something unimportant |
| 13:57.20 | apb1963 | FlashDeluxe: He means, how do you know it's not working? |
| 13:58.28 | apb1963 | tries to keep his eyes open |
| 13:58.39 | [TK]D-Fender | FlashDeluxe: You are staring at dialplan... and not the actual result |
| 13:59.00 | FlashDeluxe | apb1963: ah ok^^ because i do not see anything in the CLI, i see that my fax is being received and i see the hangup, but i do not see that there is my script executed |
| 13:59.40 | [TK]D-Fender | FlashDeluxe: You are seeing a hanup? Well I guess that tells you how the call REACTS... so don't be surprised at the outcome. Go think about where you are PUTTING your "followup code" |
| 14:00.18 | [TK]D-Fender | FlashDeluxe: And this exact thing is something you'll see dozens of code samples for already... which will tell it to you rather blatantly |
| 14:00.44 | [TK]D-Fender | FlashDeluxe: "Asterisk Standard Extensions" <--------------- |
| 14:02.08 | igcewieling | suspects source .vs. dest channel hangup and "g" option to Dial() .vs. the "h" extension |
| 14:03.06 | apb1963 | thinks http://www.voip-info.org/wiki/view/Asterisk+cmd+System looks like an interesting page |
| 14:03.15 | igcewieling | FlashDeluxe: your issue has nothing to do with the System command and everything to do with the (some would say Kafkaesque) way Asterisk handles hangups. (better in Asterisk 11+, but also totally different from the previous versions) |
| 14:07.53 | [TK]D-Fender | igcewieling: What changed/added in 11+? |
| 14:08.00 | FlashDeluxe | ok ok i got it... i know that my code is shitty, i was just wondering why it worked with 1.8 and not with 11 |
| 14:09.58 | igcewieling | [TK]D-Fender: Hanguphandlers |
| 14:10.03 | igcewieling | (really asterisk 10, but we don't talk about Asterisk 10) |
| 14:10.03 | igcewieling | FlashDeluxe: did you read the UPGRADE*.txt files in Asterisk 11 |
| 14:10.10 | jmetro | a10 wasnt that bad, i worked in it before 11 came out |
| 14:15.55 | *** join/#asterisk Ionic (~ionic@2001:41d0:1:6276:fa1e:dfff:fedd:1f7) |
| 14:18.09 | FlashDeluxe | igcewieling: jep i read it |
| 14:18.10 | [TK]D-Fender | igcewieling: Ah I think I did see this once.. didn't fully set-in |
| 14:18.11 | igcewieling | FlashDeluxe: Let me give you a little quiz to see if you understand what you need to understand. 1) When the caller hangs up, what happens in the dialplan? |
| 14:18.11 | igcewieling | not the callEE, i'm only talking about the callER |
| 14:18.13 | FlashDeluxe | igcewieling: mhh the hangup extension is called? |
| 14:18.13 | igcewieling | FlashDeluxe: WRONG! |
| 14:18.14 | igcewieling | FlashDeluxe: with the "g" option to dial, the call continues to the next prority in the dialplan, without "g" the dialplan ends. |
| 14:18.14 | apb1963 | executes the next priority?? |
| 14:18.15 | apb1963 | yes! |
| 14:18.15 | igcewieling | OK, next question, what happens when the callER hangs up first? |
| 14:18.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:18.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.15 | igcewieling | s/callER/callEE |
| 14:18.16 | apb1963 | I'd guess the same thing but I suppose I'm wrong. |
| 14:18.16 | igcewieling | screw it, I don't have time for this. best of luck, likely you have an issue with which side of the call hangs up first and what happens in the dialplan differently depending on which side hangs up first. |
| 14:18.17 | FlashDeluxe | igcewieling: i would say the hangup extension is called |
| 14:18.19 | igcewieling | FlashDeluxe: so now you have two paths the call can take after once side hangs up Have you made sure both paths end up where you want them to? |
| 14:18.21 | apb1963 | well that's certainly kafkaesque... I'd even go so far as to call it wlatsome |
| 14:18.21 | FlashDeluxe | igcewieling: i guess, that the callee hangs up and thats the reason why my script isn`t called :) |
| 14:18.23 | [TK]D-Fender | [09:33]FlashDeluxehi folks, i got a short question, i want to execute a bash command after a fax is received, in asterisk 1.8 i did it with System(/bash/script), but that doesn`t work anymore, now after googleling, i am trying it with SHELL(/bash/script) but that won`t work either... Am i missing something? My config entry: http://pastebin.com/GmLi5nPz |
| 14:18.23 | [TK]D-Fender | Helps when you read the original post & pastebin |
| 14:18.23 | [TK]D-Fender | there is no DIAL |
| 14:18.24 | [TK]D-Fender | This is a FAX receieve and that app HANGS UP when it's done. |
| 14:18.25 | [TK]D-Fender | There is no "g", there is only you "hangup handler", whatever you define |
| 14:18.26 | *** join/#asterisk serafie (~erin@nat/digium/x-gwoudzezodfjqdih) |
| 14:18.28 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
| 14:19.09 | apb1963 | <igcewieling> FlashDeluxe: with the "g" option to dial, the call continues to the next prority in the dialplan, without "g" the dialplan ends. |
| 14:19.32 | Ionic | what's the asterisk core lib? |
| 14:19.53 | Ionic | or maybe there's none "/ |
| 14:19.55 | Ionic | :/ |
| 14:20.52 | [TK]D-Fender | Ionic: What is your actual goal? |
| 14:21.04 | [TK]D-Fender | Ionic: Because that is a dangerously vague start... |
| 14:21.11 | Ionic | [TK]D-Fender: being able to use ast_get_version(_num)() |
| 14:23.23 | [TK]D-Fender | Ionic: http://tinyurl.com/m44bdxq <-- seems to find it pretty quick |
| 14:23.42 | igcewieling | apb1963: sometimes I'm an idiot. |
| 14:24.43 | Ionic | [TK]D-Fender: not really. the problem is that I can't find a library exporting this symbol... and maybe there's none whatsoever |
| 14:24.51 | igcewieling | [TK]D-Fender: send Ionic to #asterisk-dev, they can be good about slaughtering the weak 8-) |
| 14:25.06 | igcewieling | And it really is a dev question |
| 14:25.12 | [TK]D-Fender | "ast_version.h" |
| 14:25.14 | Ionic | it is, yes |
| 14:25.38 | [TK]D-Fender | Easily grep-able |
| 14:26.35 | Ionic | [TK]D-Fender: finding the header file is trivial, but I haven't found a library exporting (implementing) this symbol yet |
| 14:26.56 | igcewieling | Ionic what makes you think there is one? |
| 14:26.58 | apb1963 | igcewieling: meh |
| 14:27.02 | Ionic | [TK]D-Fender: that is, I ouldn use ast_get_version() just fine, but linking will fail |
| 14:28.03 | Ionic | igcewieling: if there's none, ast_get_version(_num)() is pointless for everything but core asterisk :/ |
| 14:28.20 | Ionic | so I hoped it was usable for outside code, too |
| 14:28.21 | igcewieling | Ionic: I believe app_skel.c should show you how to get that informtion for a module running inside of Asterisk |
| 14:30.16 | file | what are you trying to use it for? |
| 14:30.34 | *** join/#asterisk cneb3000 (~cneb3000@2.221.241.85) |
| 14:30.51 | Ionic | file: determining asterisk's version for an external module |
| 14:31.16 | file | why? |
| 14:31.45 | Ionic | because the module is supporting multiple asterisk APIs and must be able to configure itself based on the actually used version |
| 14:31.56 | WIMPy | ionic: Hmm. Wasn't it you who said using it was just as easy as using the old define |
| 14:32.01 | file | a runtime check won't work, structure definitions change as do the APIs |
| 14:32.08 | Ionic | WIMPy: gotcha :p |
| 14:33.27 | Ionic | file: checking once before compiling the module is good enough (until upgrading asterisk, which will inevitably need a module recompilation anyway) |
| 14:34.47 | Katty | wonders where Qwell is |
| 14:35.37 | carrar | I'm your huckleberry |
| 14:35.38 | Katty | i don't suppose anyone knows his whereabouts today? |
| 14:36.14 | igcewieling | Katty: hungover? |
| 14:36.19 | file | Katty, ummmmm |
| 14:36.25 | file | lemme see |
| 14:36.32 | Katty | file: ty |
| 14:36.33 | file | I don't think he's out |
| 14:37.08 | Katty | good. |
| 14:37.12 | Katty | that means he's somewhere. |
| 14:37.36 | *** join/#asterisk serafie (~erin@nat/digium/x-hnkkoilulmdxnyus) |
| 14:37.39 | Katty | seems to be idle for awhile |
| 14:39.12 | apb1963 | notes that the ast_get_version function is in ../main/version.c |
| 14:39.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:39.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:41.08 | *** join/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 14:41.27 | igcewieling1 | Cox Communications: Making your meth head uncle look reliable. |
| 14:41.33 | Ionic | apb1963: right... and not exported anywhere else, making it useless for external modules :( |
| 14:42.01 | apb1963 | also notes that /etc/asterisk/version contains the version number... or is that just something I did for me? |
| 14:42.21 | Ionic | not on my system |
| 14:42.38 | Ionic | that's probably put there by your distro |
| 14:42.40 | apb1963 | exported?? What does "exported" mean in the context of a c program? |
| 14:42.45 | apb1963 | I have no distro |
| 14:43.13 | apb1963 | If yo're compiling a module, just link in the c file.... done and done. |
| 14:43.27 | Ionic | apb1963: being able to use the function by including the header file and linking a library actually implementing the function |
| 14:43.41 | igcewieling1 | Ionic: depending on how you define "external module", see app_skel.c |
| 14:44.14 | igcewieling1 | You could also look at how Sangoma's transcoding codec_sangoma.c drivers determine the Asterisk version |
| 14:45.27 | apb1963 | just compile the silly little version.c file |
| 14:45.41 | apb1963 | include it in your makefile |
| 14:45.52 | apb1963 | then you create your own .o to link with |
| 14:45.55 | apb1963 | all done |
| 14:46.02 | igcewieling1 | apb1963: I suspect that is how other "external" modules do it. |
| 14:46.08 | apb1963 | dunno |
| 14:46.28 | *** join/#asterisk cstk421 (~cstk421@99-8-81-227.lightspeed.livnmi.sbcglobal.net) |
| 14:46.48 | Ionic | the silly little version.c file is autogenerated during asterisk builds *sigh* |
| 14:46.50 | apb1963 | if you're really ambitious you can even stick the version.o file in a library |
| 14:47.03 | cstk421 | anyone here have a working sip fax setup ? im using elastix. |
| 14:47.09 | apb1963 | is it? |
| 14:47.17 | apb1963 | that would be interesting |
| 14:47.20 | Ionic | apb1963: try to handle asterisk as a black box |
| 14:47.22 | Ionic | yes, it is |
| 14:47.43 | igcewieling1 | Ionic: comments from codec_sangoma.c http://pastebin.com/0NHjC4xb |
| 14:47.56 | igcewieling1 | ~elastix |
| 14:47.56 | infobot | i heard elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 14:48.19 | Ionic | igcewieling1: that's a hardcoded 11.0... yeah, great :/ |
| 14:49.04 | igcewieling1 | Ionic: their code works with more than just 11 |
| 14:49.09 | Ionic | igcewieling1: that's the current solution for chan_capi, too, but I thought it was possible to actually fetch the "correct" version by using ast_get_version()... turns out I'm wrong |
| 14:49.17 | cstk421 | thx infobot. been trying over there. just wondered if anyone had ANY type of sip fax setup working that I could build. any suggestions ? |
| 14:49.37 | apb1963 | is that what they're calling an API these days? |
| 14:49.38 | Ionic | igcewieling1: previous releases defined an ASTERISK_VERSION_NUM macro in the header file containing the version number |
| 14:49.46 | igcewieling1 | cstk421: T.38 is the only thing which gives you hope of reliable fax |
| 14:50.31 | [TK]D-Fender | cstk421: So set your system up properly for T.38 and whatever receive app you want |
| 14:50.41 | *** join/#asterisk DND (5ec8071a@gateway/web/freenode/ip.94.200.7.26) |
| 14:51.03 | DND | hi guys i need help. whenever i call, the other party rings then drops the call. here is the call log: http://pastebin.com/raw.php?i=k332iJWT the box has 2 pri cars and one telco each |
| 14:51.05 | Ionic | igcewieling1: this was removed, because it forced a rebuild of asterisk whenever the svn revision changed (as the svn revision number was included in this version string and the header file autogenerated once again) |
| 14:51.21 | cstk421 | by setup my system for t.38 properly how do you mean ? my sip provider supports t.38 and i have the did ready. the application side is my issue. |
| 14:51.46 | Ionic | so it was decided to switch to not touching the header file and implementing a function instead... which is good for code design |
| 14:52.03 | apb1963 | You CAN get the version by using ast_get_version()... it's just that you need to compile the source code to do it.... or find the .o already compiled. Not sure why it matters if it's in a library or a .o |
| 14:52.28 | igcewieling1 | cstk421: EXACTLY what is the issue. |
| 14:52.37 | WIMPy | apb1963: because you need it before compiling. |
| 14:52.40 | Ionic | apb1963: there's no object file once you've installed asterisk |
| 14:53.02 | Ionic | even less with binary packages provided by your distribution |
| 14:53.16 | apb1963 | I have a version.o on my system |
| 14:53.32 | [TK]D-Fender | DND: responded in #freepbx |
| 14:53.40 | Ionic | apb1963: that's because you've been building asterisk from source on your system |
| 14:53.45 | apb1963 | yes |
| 14:53.49 | apb1963 | isn't that what you're doing? |
| 14:54.03 | Ionic | apb1963: no, as I said, treat asterisk as a black box |
| 14:54.09 | Ionic | like installed by apt-get install asterisk on debian |
| 14:54.16 | jmetro | eww |
| 14:54.17 | Katty | meep. |
| 14:54.19 | apb1963 | oh you mean you want to build your module before building asterisk? |
| 14:54.27 | igcewieling1 | Asterisk IS a black box from the standpoint of external stuff. |
| 14:54.29 | jmetro | Katty: Moop |
| 14:54.33 | Ionic | apb1963: independently of building asterisk, yes |
| 14:54.35 | Katty | jmetro: Mooper. |
| 14:54.51 | Ionic | actually, more like AFTER having installed asterisk |
| 14:55.00 | Ionic | with no access to the original source build whatsoever |
| 14:55.05 | apb1963 | oh no access |
| 14:55.20 | jmetro | YipYipYip http://www.youtube.com/watch?v=uarh5lNtpa8 |
| 14:55.24 | Greenlight | If using an SRV record to place SIP calls, should all entries at the same priority be attempted, *before* going to the next priority? |
| 14:55.35 | apb1963 | mmm.... does asterisk get stripped of symbols after installing? |
| 14:55.56 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:56.01 | igcewieling1 | Greenlight: Asteirsk's SRV support for Dialing is totally seperate from the SRV AEL stuff you are using |
| 14:56.06 | [TK]D-Fender | [10:54]Ionicwith no access to the original source build whatsoever <- asterisk -rx "core show version" > omgversion.txt |
| 14:56.11 | Greenlight | I know that, it's a general SRV question |
| 14:56.18 | apb1963 | oh yah |
| 14:56.19 | Greenlight | I know that natively Asterisk doesn't support SRV failover |
| 14:56.22 | apb1963 | what he said |
| 14:56.34 | cstk421 | i dont know exactly what the issue is. basically after setup it will answer and pickup a fax calls but errors out |
| 14:56.43 | igcewieling1 | Greenlight: Dial will dial the single srv record it selects. It won't try another server, and honestly won't do anything useful. |
| 14:56.43 | Greenlight | I'm just trying to work out what the logic *should* be to failover |
| 14:56.44 | Ionic | [TK]D-Fender: that's requiring a running asterisk instance |
| 14:57.18 | Greenlight | igcewieling1: I know that is what will normally happen. Guess this is more a general SIP/SRV question. |
| 14:57.27 | [TK]D-Fender | [10:51]cstk421by setup my system for t.38 properly how do you mean ? my sip provider supports t.38 and i have the did ready. the application side is my issue. <- did you set your trunk up properly for it? What are you using to receive the fax? |
| 14:57.29 | igcewieling1 | Greenlight: read the SRV RFC. I tried to make my AEL work like the RFC. |
| 14:57.45 | apb1963 | does asterisk get stripped of symbols after installing? |
| 14:57.59 | Greenlight | The AEL doesn't attempt other servers at the *same* priority, that's what im trying to work out |
| 14:58.08 | Ionic | apb1963: yes and no, I figure that's configurable |
| 14:58.43 | igcewieling1 | Greenlight: I could go read the RFC to get your answer, but you could do that too. |
| 14:58.58 | igcewieling1 | SRV is so useless in Asterisk nobody actually uses it. 8-| |
| 14:58.58 | Greenlight | Yea am googling it now dw :) |
| 14:59.01 | cstk421 | Fender- i assume i setup the trunk properly. I setup an IAX trunk and a new fax extension for it to use. I was using the elastix fax. |
| 14:59.19 | Greenlight | Yea, my carrier is insisting with the volumes of traffic i'm putting out |
| 14:59.20 | igcewieling1 | cstk421: I beleive T.38 does not work with IAX |
| 14:59.22 | [TK]D-Fender | cstk421: Your assumption is wrong. |
| 14:59.25 | apb1963 | even if it doesn't, it's too hard to find unless you grep for specific version numbers |
| 14:59.31 | [TK]D-Fender | cstk421: and ther is no T.38 with IAX |
| 14:59.39 | [TK]D-Fender | cstk421: That idea is dead in the water |
| 15:00.00 | [TK]D-Fender | cstk421: Whatever failure rate you're getting.... is what you're going to get |
| 15:00.01 | apb1963 | if yo're willing to grep version by version you could use "strings" |
| 15:00.10 | igcewieling1 | cstk421: the fact you think you can do T.38 with IAX shows a fundamental misunderstanding of, well, pretty much everything. |
| 15:00.13 | apb1963 | but that's really ugly |
| 15:00.22 | apb1963 | wlatsome in fact |
| 15:00.37 | Greenlight | Yup, it's supposed to failover to the *same* priority first: Clients always use the SRV record with the lowest-numbered priority value first, and fallback to other records of equal or higher value if the connection to the host fails. |
| 15:01.05 | igcewieling1 | Greenlight: that is not how I read it. |
| 15:01.19 | igcewieling1 | remember priority and weight are different |
| 15:01.21 | Greenlight | I know |
| 15:02.14 | Greenlight | It certainly makes sense to attempt all at current priority before failover |
| 15:02.23 | Greenlight | In my case my carrier has all nodes at same priority |
| 15:02.39 | Greenlight | But I'd like to honor the RFC to futureproof things |
| 15:03.37 | igcewieling1 | Greenlight: see the "Usage rules" part of the RFC |
| 15:03.39 | apb1963 | but then... if "previous releases defined an ASTERISK_VERSION_NUM macro in the header file containing the version number", but there was no source code... how would that have helped anyway? |
| 15:04.49 | Ionic | apb1963: because a string literal macro is (generally) useable without linking |
| 15:05.13 | apb1963 | but... you need the source code to use it. |
| 15:05.20 | Ionic | apb1963: asterisk -V seems to be the best way of determing asterisk's version so far |
| 15:05.25 | WIMPy | Only the headers. |
| 15:05.26 | Ionic | apb1963: no, you don't |
| 15:05.30 | Greenlight | " While there are still elements left at this priority |
| 15:05.31 | Greenlight | level |
| 15:05.42 | jmetro | "core show version" ? |
| 15:06.20 | igcewieling1 | jmetro: requires Asterisk to be running. asterisk -V only requires Asterisk be installed |
| 15:06.27 | Ionic | jmetro: rather asterisk -V then asterisk -r (because that would require a running asterisk instance and may NOT represent the currently ninstalled version) |
| 15:06.42 | cstk421 | thx fender i will do some more reading and see |
| 15:07.05 | [TK]D-Fender | cstk421: There is no more reading to do for IAX. |
| 15:07.15 | Ionic | anyway, using ast_get_version() is not possible for external modules, that's clear now... and the workaround is to use asterisk -V, ok |
| 15:07.19 | [TK]D-Fender | cstk421: There are no more options to take for it |
| 15:07.30 | Greenlight | Yea, that RFC clears things up. So I'll adjust to attempt all at the current priority before trying next. Thanks ;) |
| 15:07.49 | igcewieling1 | Greenlight: patches welcome 8-) |
| 15:08.10 | Greenlight | Heh just don't laugh - it'll be my first AEL script :) |
| 15:08.29 | igcewieling1 | Greenlight: that's OK, I'm not a programmer |
| 15:09.12 | cstk421 | Fender- so am i looking for a t.38 sip trunk ? Is that verbage correct |
| 15:09.16 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:09.26 | [TK]D-Fender | cstk421: a SIP provider that supports T.38 |
| 15:09.32 | cstk421 | ok that i have |
| 15:09.51 | [TK]D-Fender | cstk421: And that you're running * 1.8+ |
| 15:09.54 | cstk421 | so the elastix fax that requires iax is useless based on what your telling me |
| 15:09.59 | cstk421 | ah |
| 15:10.37 | [TK]D-Fender | [11:09]cstk421so the elastix fax that requires iax is useless based on what your telling me <- "requires IAx"? What are you talking about? |
| 15:10.52 | igcewieling1 | [TK]D-Fender: iaxmodem ? |
| 15:10.58 | [TK]D-Fender | cstk421: Your description of what you are using on both ends is horribly vague |
| 15:12.21 | cstk421 | im sorry im trying. elastix fax setup has me setup an iax extension. |
| 15:12.37 | cstk421 | and you were saying iax will not work |
| 15:12.42 | [TK]D-Fender | cstk421: BOTH ends. Describe EXACTLY what is sitting on each |
| 15:12.54 | cstk421 | k |
| 15:13.09 | [TK]D-Fender | cstk421: what is your call coming IN on? What precise tool are you using to RECEIVE the fax? |
| 15:13.12 | igcewieling1 | cstk421: so when you set up a fax extension it magically makes it IAX or do you mean the documentation instructs you do to this. If so, cite your source. |
| 15:13.47 | cstk421 | this is the source |
| 15:13.47 | cstk421 | http://elastix.wikia.com/wiki/FAX_Setup_@_Elastix |
| 15:13.58 | [TK]D-Fender | cstk421: I don't care about that link/. |
| 15:14.07 | [TK]D-Fender | cstk421: I asked what YOU are getting the callIN from in the first place |
| 15:14.48 | cstk421 | im getting a call in from a fax machine. is that the answer your looking for ? |
| 15:14.55 | [TK]D-Fender | NO |
| 15:15.02 | [TK]D-Fender | cstk421: How. Does. The. Call. Arrive. At. Your. Server. In. The. First. Place? |
| 15:15.21 | cstk421 | sip provider . comcast. elastix/asterisk box |
| 15:15.45 | [TK]D-Fender | cstk421: And what precise app are you using to recieve the fax once it arrives to Asterisk? |
| 15:15.56 | cstk421 | elastix fax uses HylaFax |
| 15:16.09 | *** join/#asterisk _-Jon-_ (~jon@2607:f2c0:f00f:2100::2004) |
| 15:16.26 | _-Jon-_ | Hello all |
| 15:16.39 | [TK]D-Fender | cstk421: You need to be running Asterisk 10+ in order to act as a T.38 GATEWAY and you provider has to support and be setup for T.38. |
| 15:16.49 | [TK]D-Fender | cstk421: Go prove you've done that part right |
| 15:16.53 | cstk421 | ok |
| 15:16.54 | cstk421 | thx |
| 15:16.56 | *** part/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 15:17.22 | [TK]D-Fender | cstk421: And then Elastix needs to have the gateway feature even IMPLENETED in their dialplan. We do not support that. It is their job. |
| 15:17.37 | cstk421 | got it |
| 15:19.25 | Katty | pokes Qwell again |
| 15:19.32 | Qwell | wat |
| 15:19.36 | Katty | YOU"RE AROUND |
| 15:19.40 | Katty | Qwell: HIPPO BIRDIE TWO EWE! |
| 15:19.41 | Qwell | LIES AND SLANDER |
| 15:19.45 | Katty | Qwell: HIPPO BIRDIE TWO EWE! |
| 15:19.50 | Katty | Qwell: HIPPO BIRDIE DEER EWE!! |
| 15:19.51 | Qwell | Katty: I had a coffee mug that said that, growing up. |
| 15:19.56 | Katty | Qwell: HIPPO BIRDIE TWO EWE!!!!!!!!!!!!!! |
| 15:20.09 | Katty | hugs the birthday boy, Qwell |
| 15:20.14 | cstk421 | D-Fender here is the asterisk version = Asterisk 11.4.0 built by palosanto @ rpmbuild64-2.elastix.palosanto.com on a x86_64 running Linux on 2013-05-20 16:19:37 UTC |
| 15:20.22 | Katty | Qwell: so you're *ahem* 30 again this year, i presume? |
| 15:20.25 | cstk421 | dont know how to prove the provider supports t.38 |
| 15:20.30 | Qwell | Katty: 29 |
| 15:20.35 | Qwell | I go backwards now. |
| 15:20.38 | Katty | Qwell: oh, right. 29 again. my mistake. |
| 15:20.44 | cstk421 | i know i chose that provider and when picking a did it had to support t.38 |
| 15:20.59 | Katty | Qwell: have any plans? |
| 15:21.05 | Katty | Qwell: cake, i hope? and candles? |
| 15:21.21 | Qwell | cake tonight. german chocolate. :D |
| 15:21.26 | Katty | mmmm. |
| 15:21.29 | Katty | is your ladyfriend making it? |
| 15:22.16 | jmetro | my ladyfriend makes shepards pie. Mmm |
| 15:22.21 | Ionic | the cake is a lie |
| 15:22.35 | [TK]D-Fender | [11:20]cstk421dont know how to prove the provider supports t.38 <- check that your trunk was set up properly for it. Check your vendors page proving they are indeed supposed to be offering it. Look at the actual CALL. This will prove T.38 negotiation and whether or not they call the gateway setting that is required. |
| 15:22.39 | [TK]D-Fender | cstk421: https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway |
| 15:27.00 | _-Jon-_ | I'm running v11.3 and am trying to get SLA to work for the first time, however I'm getting an error about the SLAStation application not being found. Does anyone know if this is yet another app that's been terminated? |
| 15:28.52 | [TK]D-Fender | _-Jon-_: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SLAStation |
| 15:29.20 | [TK]D-Fender | _-Jon-_: It uses MeetMe as a backend... perhaps you failed a pre-req for it... like DAHDI |
| 15:29.28 | *** join/#asterisk cjm_ (~cjm@tclc.org) |
| 15:29.45 | [TK]D-Fender | _-Jon-_: How many lines are you thinking of doing this for, and what phones? |
| 15:30.21 | _-Jon-_ | I don't believe I have DAHDI installed, so that is very likely the case. |
| 15:30.56 | _-Jon-_ | Currently I'm just testing for my own purposes, but our office is planning on switching to Asterisk with possibly Cisco SPA5xx phones. 8 lines total |
| 15:31.22 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.183) |
| 15:33.38 | [TK]D-Fender | _-Jon-_: SLA is a lie in *.... |
| 15:33.40 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 15:33.54 | _-Jon-_ | Oh? |
| 15:33.55 | [TK]D-Fender | _-Jon-_: and that many line will make for a seriously deficient implementation. |
| 15:34.02 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 15:34.14 | [TK]D-Fender | _-Jon-_: You won't have enough line-keys left for multiple simultaneous calls, etc... |
| 15:35.04 | [TK]D-Fender | _jYou are highly advised to forget about any kind of "key system" usage with Asterisk |
| 15:35.53 | _-Jon-_ | That's what I've done in the past with my own setups (for personal use at home). I think the company was hoping to mirror the existing Nortel system as much as possible |
| 15:36.11 | _-Jon-_ | I'm curious, what is the purpose of SIP phones with the multiple line keys? |
| 15:36.16 | [TK]D-Fender | _-Jon-_: Time to let go. |
| 15:36.33 | WIMPy | Confuse the user. |
| 15:36.47 | [TK]D-Fender | _-Jon-_: Speed dials/BLF, multiple simultaneous calls, multuiple registrations to different servers, etc |
| 15:37.27 | _-Jon-_ | Makes sense. Certainly increases the choice in handsets then |
| 15:37.28 | [TK]D-Fender | _-Jon-_: And SIP-B is the proper RFC for using a SIP system for SLA behaviour, which is something Asterisk does not support at all |
| 15:39.11 | _-Jon-_ | Alright, I guess like you said.. Time to let go! |
| 15:40.00 | [TK]D-Fender | _-Jon-_: My company used to have a Norstar 8X24. I was very happy to have gotten to replace it with Asterisk many years ago |
| 15:41.07 | _-Jon-_ | We've been holding on to this ancient equipment for far too long as well. It was only after our voicemail module died that they're moving on. I've come to love Asterisk over the years I've spent with it :) |
| 15:44.31 | _-Jon-_ | Thank you for your help D-Fender! |
| 15:45.45 | [TK]D-Fender | _-Jon-_: More like "advice for a fuller head of hair" |
| 15:45.59 | _-Jon-_ | Haha |
| 15:49.01 | *** join/#asterisk Nombrandue (~Satan@ip174-71-93-190.om.om.cox.net) |
| 15:51.53 | *** join/#asterisk dtx (~dtx@cdf-imaging.com) |
| 15:52.17 | dtx | Hello everyone! Is there a build of DPMA available for ARM Linux? |
| 15:52.24 | dtx | ARM architecture Linux |
| 15:53.45 | dtx | Is this the right room to ask in? |
| 15:53.59 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:54.12 | pabelanger | dtx: I don't believe so |
| 15:54.18 | pabelanger | might have to email digium direct |
| 15:54.27 | pabelanger | malcolmd: ^ |
| 15:55.07 | malcolmd | dtx: we've had a couple of inquiries about it, but we're not doing anything official at this point. you running fedora core remix or raspbian? |
| 15:57.50 | dtx | malcolmd: I'm not really running embedded anything yet (at least not for the PBX yet). I just have Asterisk Now setup and running on a virtual box, with a bunch of phones scattered throughout the building. I'd much rather run this on a small embedded system that can stay powered for days. Like Beaglebone Black on http://www.raspberry-asterisk.org/ |
| 15:59.48 | *** join/#asterisk psanders (~psanders@ks-184-2-115-125.dhcp.embarqhsd.net) |
| 15:59.53 | dtx | Honestly, I really wish DPMA was not a binary blob. That makes something like this kind of a hassle. |
| 16:01.28 | malcolmd | dtx: i've supplied the occasional build for rPi's for fedora core remix, but i don't think i want to stray any further than that, especially since it's not actually supported (we don't take asterisk support contracts on non-x86 architectures, for example). |
| 16:05.14 | *** join/#asterisk ThothCastel (~chatzilla@37-110-251-66.g3ns.net) |
| 16:05.19 | dtx | malcolmd: I understand the support issue. Is there a location to download that binary? I'm looking at that or possibly thunking the x86 binary through user mode qemu. I'd much rather have a native binary and work from that. |
| 16:05.22 | dtx | :P |
| 16:16.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.28) |
| 16:18.14 | Nombrandue | upgraded to Asterisk 11.4 from 1.6, finding that in doing so, I am being forced to re-write my dialplan finally. Any good resources on translating an extensions.conf dialplan to a realtime dial plan? |
| 16:18.42 | [TK]D-Fender | realtime spec hasn't changed |
| 16:19.06 | [TK]D-Fender | context, exten, priority, app, data |
| 16:19.31 | *** join/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:22.58 | Nombrandue | I didn't read anything about upgrading the DB schema, so I was figuring that the realtime spec was the same. I am looking at translating portions of the flatfile over, beyond the context/extension level. some include statements, if that would work |
| 16:23.48 | [TK]D-Fender | Nombrandue: You should read the Book chapeters on realtime for the basics |
| 16:23.53 | [TK]D-Fender | ~book |
| 16:23.53 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:24.19 | *** join/#asterisk cmendes0101| (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 16:34.30 | Nombrandue | So understanding how to set up realtime isn't what I am after, it is seeing if anyone had any good programatic references to automate this from an existing non-realtime setup. I will reference the book (It is newer than the physical one I have, so that is a plus for me) and manually run a migration, instead of rebuilding the plan in the flatfile and then importing it to the db |
| 16:37.02 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 16:38.04 | [404] | I am looking at changing the default voicemail menu. From my research, it is looking like editing the actual source code and recompiling it is the best way of doing that. Am I correct on this? |
| 16:38.14 | Qwell | correct |
| 16:38.21 | Qwell | s/best/only/ |
| 16:38.57 | [404] | little sed i see |
| 16:39.54 | WIMPy | Or do it in your dialplan instead of using the app. |
| 16:42.09 | *** join/#asterisk nickfennell_ (~nickfenne@unaffiliated/nickfennell) |
| 16:44.16 | jmetro | Run D.M.C |
| 16:48.34 | Maliuta | De La Soul |
| 16:48.55 | [404] | RunĀD.M.C. was an American hip hop group from Hollis, Queens, New York, founded in 1981 by Joseph "Run" Simmons, Darryl "D.M.C." McDaniels, and Jason "Jam Master Jay" Mizell. |
| 16:49.14 | Maliuta | jmetro: we are just listing '90s hip-hop groups right? ... that is what this channel is about ;) |
| 16:49.26 | [404] | De La Soul is an American hip hop trio formed in 1987 on Long Island, New York. |
| 16:49.37 | Maliuta | KLF |
| 16:49.45 | jmetro | Beastie Boys. |
| 16:49.56 | [404] | The KLF (also known as The Justified Ancients of Mu Mu, The Timelords and other names) were one of the seminal bands of the British acid house movement during the late 1980s and early 1990s. |
| 16:50.12 | [404] | Beastie Boys are an American hip hop band from New York City, formed in 1981. |
| 16:50.17 | Maliuta | Intercooler |
| 16:50.29 | jmetro | Blackstreet |
| 16:50.37 | Maliuta | jmetro: are you finding this fun? |
| 16:50.52 | jmetro | => |
| 16:50.55 | jmetro | :> even |
| 16:50.58 | [404] | intercooler isn't hip hop |
| 16:51.28 | [404] | blackstreet is r&b |
| 16:51.34 | [404] | not hip hop |
| 16:51.35 | jmetro | aww |
| 16:51.39 | Maliuta | [404]: no, but I have a cool story about meeting the lead guitarist/singer at a Ben Folds gig before they made it |
| 16:51.51 | dtx | Kraftwerk |
| 16:51.58 | dtx | lol |
| 16:52.02 | Maliuta | word up dtx |
| 16:52.07 | jmetro | Aaliyah |
| 16:52.21 | jmetro | crap is that R&B again |
| 16:52.26 | Maliuta | don't 'dis the Werk ... without them none of the electronica scene would exist |
| 16:52.39 | jmetro | Kraftwerk is pretty great. |
| 16:52.46 | dtx | Hell yeah it is |
| 16:52.54 | Maliuta | Jungle/Drum&Bass is better any how |
| 16:53.10 | jmetro | Herbie Hancock. |
| 16:53.19 | Maliuta | jmetro: word up! |
| 16:53.28 | Maliuta | <- jazz freak too |
| 16:53.36 | jmetro | Dont like jazz myself |
| 16:53.48 | Maliuta | slaps jmetro with some taste |
| 16:53.49 | jmetro | I'm in love with the new Robin Thicke song |
| 16:53.50 | Maliuta | ;) |
| 16:54.20 | Maliuta | my guilty pleasure is Robbie Williams "Man Machine" from the Lock Stock soundtrack |
| 16:54.32 | *** join/#asterisk peetaur2 (~peter@x2f056e4.dyn.telefonica.de) |
| 16:55.02 | jmetro | my headphones-only-secret-music is some rap music |
| 16:55.31 | Maliuta | I went to a VJ gig a few months back, guys name was "DJ Woody" from London he did a '90s hip-hop mix tape ... in chronological order |
| 16:55.40 | jmetro | like "Im On It" by Big Boi & the Purple Ribbon Allstars |
| 16:56.48 | dtx | I'm in love with the women in the new Robin Thicke song, but I don't particularly agree with the song's message. |
| 16:56.54 | Maliuta | jmetro: I listen to all sorts of stuff, (http://www.last.fm/user/Maliuta) but right now I'm basically on a '70s rock kick. Creeedence, Billy Joel, etc. Mostly on vinyl |
| 16:57.11 | jmetro | dtx: Oh, its an awful song lyric-wise but its extremely dancey and catchy |
| 16:57.20 | Maliuta | Who the feck is this "Thicke" person ... I only know of Alan Thicke |
| 16:57.28 | jmetro | Robin Thicke - Blurred Lines |
| 16:57.52 | jmetro | its a resurgance of Good Disco which im also a fan of [think Born to Be Alive] |
| 16:57.55 | Maliuta | try "Washington" |
| 16:58.08 | Maliuta | She has some really good stuff |
| 16:58.20 | *** join/#asterisk kannan (~chatzilla@122.164.106.94) |
| 16:58.52 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:00.23 | Maliuta | People need to release more music on 180GSM vinyl, it really is the only way to listen to music |
| 17:00.58 | jmetro | eh, im okay with high-res digital |
| 17:01.08 | jmetro | i work in phones, g722 sounds pretty darn good |
| 17:01.29 | *** join/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 17:02.20 | digitalirony | sup, seeing this issue where when phone goes to voicemail it starts to play the greeting, but it cuts off, something to the effect of "The per*BEEP*" vs "The person at extension ###...." |
| 17:02.26 | digitalirony | anyone have any ideas? |
| 17:03.03 | Maliuta | jmetro: you looking for another slapping? FLAC is the only digital form I can take (I also have quality kit that you can hear the difference on) |
| 17:03.17 | dtx | digitalirony: your pbx likes to cuss |
| 17:03.38 | digitalirony | dtx lol, newp |
| 17:04.05 | jmetro | Maliuta: Problem is i roll Full DRM-free and for the most part listen on 20$ headphones or 10$ earbuds or very good car stereo :3 |
| 17:04.11 | file | mmm FLAC |
| 17:04.24 | dtx | Maliuta: what is the point of vinyl? It's all digitally mastered anyway. I can write you a plugin that can add hiss and pops if you want. |
| 17:05.10 | Maliuta | dtx: if you have hiss and pop your vinyl is fecked |
| 17:05.38 | file | my downfall was listening to music on my roommate's headphones... |
| 17:05.59 | Maliuta | dtx: and off of vinyl you actually get a better frequency range. In the same way that you can feel lower frequencies you feell higher freqs too ... better listening experience |
| 17:06.23 | Maliuta | file: I have done that to people, although I need to replace my HD595s |
| 17:06.56 | file | Maliuta, my goto headphones are my Beyerdynamic DT770 Pros |
| 17:06.59 | dtx | Maliuta: I guess I don't understand the point of vinyl though. If it's digitally mastered, then you are losing information by the analog transfer to vinyl. If it's digitally mastered, and you get it in FLAC, you lose nothing besides the inconvenience of using vinyl. |
| 17:07.04 | jmetro | Maliuta: we totally forgot about the most important 80's hip hopper |
| 17:07.06 | jmetro | MC HAMMER |
| 17:07.52 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:07.53 | jmetro | & Will Smith kinda. |
| 17:08.11 | Maliuta | jmetro: Vanilla Ice? |
| 17:08.15 | igcewieling | <rant time=10m" class="cox cable" /> |
| 17:08.18 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 17:08.46 | Maliuta | file: I like my Sennheiser, must the the German in me |
| 17:09.13 | jmetro | ars technica is my favorite sounding tech name, personally |
| 17:09.29 | Maliuta | dtx: actually they only digitally master to go to CD. Many studios still use tube amps in production |
| 17:10.42 | dtx | Maliuta: I don't know the "they" you are talking about. I would think that would be the exception not the norm. |
| 17:10.58 | dtx | I could be wrong though. |
| 17:10.59 | jmetro | ^ |
| 17:11.27 | Maliuta | dtx: how many bands do you know that have studio recorded in the last, say, 12 months? |
| 17:12.28 | dtx | Is this a recent change or something? |
| 17:12.54 | Maliuta | dtx: nope, continuance of practice |
| 17:14.46 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 17:14.52 | dtx | I dunno. I just follow stuff like Trent Reznor and what not. He seems to have a pretty digital workflow. |
| 17:15.14 | Maliuta | dtx: for the record this particular recording involved some sections coming out of the desk, onto 8-track, through a tube amp, back to 8-track, and back into the desk |
| 17:15.14 | dtx | But industrial music kinda goes for that sound anyway |
| 17:15.35 | Maliuta | Not all bands are electronica |
| 17:15.56 | dtx | Maliuta: right, but not all bands are acoustic, tube amp, purists either |
| 17:17.11 | dtx | I'm not saying you're wrong by any means. Just most stuff I hear is digitally produced because it's far more flexible. |
| 17:17.54 | dtx | But then again, vinyl has had a resurgence in the last 10 years, so I could be way off. |
| 17:17.58 | Maliuta | dtx: and even with digital flow it depends on what cards and gear people are using. Computers these days can produce analogue quality audio ... if you set them up right |
| 17:18.33 | Maliuta | the sample rate on CD is not the limit of what you can do on a computer |
| 17:18.40 | dtx | correct |
| 17:18.53 | dtx | the sample rate on a DVD is not the limit of what you can do on a computer |
| 17:19.41 | Maliuta | dtx: I blew the mind of a long term muso when I pointed this out |
| 17:19.43 | Greenlight | igcewieling: Still interested in that update to the SRV script? Got it working nicely now. |
| 17:19.47 | *** part/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 17:19.47 | WIMPy | By abusing a video interface for an audio interface? |
| 17:19.57 | Qwell | I'll take "Differences the human ear can't hear" for $100, Alex. |
| 17:20.16 | Maliuta | Qwell: it's not all about the ear |
| 17:20.33 | Maliuta | Qwell: your body is more sensitive than that |
| 17:20.42 | igcewieling | Greenlight: thanks! |
| 17:20.46 | dtx | Honestly, the weakest part of the redbook CD audio format is the 16-bit per sample limitation |
| 17:20.53 | dtx | This severely limits dynamic range |
| 17:21.01 | dtx | DVDs use 24-bit audio, if I'm not mistaken |
| 17:21.11 | WIMPy | CAN use. |
| 17:21.16 | Maliuta | there is 32-bit available today |
| 17:21.24 | Qwell | dynamic range, which is already gone, because of the loudness wars. |
| 17:21.27 | dtx | Most mixing software mixes with 32-bit float |
| 17:21.30 | igcewieling | Maliuta: Are you one of those hippy wholistic listeners? |
| 17:21.31 | Qwell | such things are thoroughly moot |
| 17:21.54 | Maliuta | igcewieling: gothy wholistic listener |
| 17:22.01 | coppice | dtx: properly dithered down from a 24 bit studio system, 16 bits is marginal, but adequate for distribution. if you record and mix at 16 bits you soon run into trouble |
| 17:22.03 | Maliuta | igcewieling: Audiophile |
| 17:22.59 | dtx | coppice: Well, right most music doesn't go from barely audible whispering to blasting audio |
| 17:23.35 | Maliuta | dtx: unless it's recorder by Bjork ;) |
| 17:23.38 | dtx | In fact, I kind of hate the dynamic range of DVDs because of that crap. I like to run it through a compressor so I can listen at a reasonable volume level. |
| 17:23.44 | coppice | dtx: even for very dynamic classical music, 16 bits is not a problem if used well |
| 17:23.49 | Maliuta | s/recorder/recorded/ |
| 17:24.09 | dtx | lol |
| 17:24.10 | Maliuta | coppice: I call "bullshit" |
| 17:24.36 | Qwell | Maliuta: I assure you, you don't want to call bullshit on coppice. |
| 17:24.48 | Maliuta | dtx: seriously? have you heard the volume range on "Oh So Quiet" |
| 17:25.13 | [TK]D-Fender | Qwell: agreed |
| 17:25.16 | dtx | Maliuta: no, but if it sounds like what I'm describing then I likely won't want to hear it either |
| 17:25.22 | dtx | lol |
| 17:25.27 | Maliuta | Qwell: if _you_ want the top and bottom freqs chopped off odf your music ... :) |
| 17:25.48 | [TK]D-Fender | MY BOSTON TERRIER DEMANDS THE FULL EXPERIENCE! |
| 17:25.53 | dtx | lol |
| 17:26.38 | igcewieling | in 10 years you won't be able to hear the top and bottom frequencies anyway |
| 17:26.44 | Maliuta | [TK]D-Fender: I don't recall being your boston terrier :P |
| 17:27.00 | dtx | [TK]D-Fender: That is a good point. Without wide dynamic range, how can I play sub-audible sounds of thunder to freak out my neighbor's dogs? |
| 17:27.31 | Maliuta | igcewieling: no, my life expectancy is somewhere between 55 and 60 - baring some major break through in Leukaemia treatment |
| 17:27.39 | [TK]D-Fender | Maliuta: You are ENTIRELY my bitch :p |
| 17:27.55 | Maliuta | [TK]D-Fender: that's a whole other pay-per-view site :P |
| 17:28.31 | igcewieling | Maliuta: I'm too lazy to do the google search, but there are well known patterns to age related hearing loss and they start long before 55 |
| 17:29.11 | igcewieling | Maliuta: I'm sorry to hear that, but there could easily be breakthroughs |
| 17:29.12 | Maliuta | igcewieling: I return to my point that it's not all about the ear ... especially with music |
| 17:29.39 | Maliuta | igcewieling: for example can you _feel_ low range sounds? |
| 17:30.00 | Maliuta | igcewieling: it's something I came to terms with 10years ago |
| 17:30.21 | dtx | Maliuta: Is this a reference to that Howard Stern movie? |
| 17:30.41 | Maliuta | dtx: what movie? |
| 17:30.51 | dtx | Maliuta: Private Parts |
| 17:31.02 | dtx | Wow, I'm old. |
| 17:31.03 | Maliuta | never seen it |
| 17:31.07 | igcewieling | Maliuta: ONLY IF THE MUSIC IS TOO LOUD |
| 17:31.08 | dtx | lol |
| 17:31.09 | igcewieling | 8-) |
| 17:31.27 | Maliuta | igcewieling: go to a rave sometime |
| 17:31.34 | Maliuta | igcewieling: or any live gig |
| 17:31.36 | coppice | the lowest pedal on a pipe organ is less than 17Hz. you feel rather than hear that |
| 17:31.41 | dtx | Maliuta: youtube "private parts" speaker |
| 17:31.42 | Maliuta | I use ear protection |
| 17:31.54 | igcewieling | Maliuta: Heh, live music is for when you can't get a studio version |
| 17:33.04 | *** join/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 17:33.04 | Maliuta | supports his local music scene |
| 17:33.17 | Maliuta | how else do people get record deals? |
| 17:33.43 | igcewieling | I always assumed oral sex, but I guess that is not the case? |
| 17:34.01 | [TK]D-Fender | is his local music scene |
| 17:34.02 | dtx | In my town, it's all about the local music scene. I'm not sure they even try for music deals. |
| 17:34.06 | Maliuta | igcewieling: that may be how _you_ got your deal ... ;P |
| 17:34.23 | digitalirony | so, having this issue, where if you call into the IVR, and then dial an extension, and that person doesn't answer, the vm-theperson.gsm is supposed to play, and it starts to, but then it just gets cut off and doesn't finish playing. However if you dial the voicemail directly, and don't have the IVR transfer you, it plays perfectly fine |
| 17:34.26 | dtx | Everybody is just fine with playing at The Deli. |
| 17:34.27 | igcewieling | Maliuta: I'll never tell! |
| 17:34.28 | digitalirony | anyone have any idea? |
| 17:34.37 | [TK]D-Fender | [13:31]MaliutaI use ear protection <- phone sex causes hearing-aids |
| 17:34.43 | Maliuta | igcewieling: I have the recordings of the webcam footage :P |
| 17:35.24 | Maliuta | [TK]D-Fender: I'll pay that ... but I meant tripple phlanged in-ear noise reduction devices :P |
| 17:36.06 | dtx | Maliuta: please tell me you don't use Monster cables. |
| 17:36.18 | dtx | :P |
| 17:36.39 | Qwell | dtx: No, there are *much* cheaper companies to buy platinum plated cables. |
| 17:37.06 | Maliuta | dtx: no, I have connections with other companies |
| 17:37.13 | Qwell | BOOM, called it |
| 17:44.22 | dtx | I wonder why more portable digital audio recorders don't support 32-bit float. It has the same precision as 24-bit, but with 256 times the dynamic range. |
| 17:45.23 | dtx | actually 512 times since its sign bit is outside of the mantissa |
| 17:45.26 | Maliuta | dtx: because the general public are idiots who only want to pay $0.99 for a 96kbit mp3? |
| 17:46.32 | Maliuta | I hang with alot of DJ's, musicians, and fellow audiophiles ... we care about quality, and those little quirs added by the artist |
| 17:46.40 | coppice | dtx: with thermal noise limiting you at one end and ear damage at the other, why would you need more than 24 bits? |
| 17:47.59 | Maliuta | meh, it's 03:48 ... I'm for bed |
| 17:48.09 | cjm_ | Hi Folks, FreePBX lets me enable a LOT of codecs, but by default only three are enabled. Is there any reason why I shouldn't enable all the codecs asterisk knows? |
| 17:48.30 | Qwell | cjm_: Do you need/use them? |
| 17:48.55 | cjm_ | Qwell, I don't know, since I'm just starting out on this adventure. |
| 17:49.13 | Maliuta | DOESN'T FREEPBX HAVE IT'S OWN CHANNEL? |
| 17:49.27 | ChannelZ | coughs |
| 17:49.36 | cjm_ | Yes. But I think this is an Asterisk question. |
| 17:49.36 | dtx | Maliuta, no need to yell |
| 17:49.58 | Maliuta | dtx: I must be going deaf from all that music :P |
| 17:50.07 | tm1000 | cjm_: just dont say freepbx in here and you would have been fine |
| 17:50.17 | cjm_ | Ha! Sorry, guys. |
| 17:50.20 | dtx | Maliuta: Also, no need for the misuse of "it's." |
| 17:50.25 | dtx | :P |
| 17:50.57 | Maliuta | dtx: did you not see me say the time is 03:50 here? |
| 17:51.10 | jmetro | must be in spain |
| 17:51.36 | Maliuta | jmetro: that's a few timezones west of here |
| 17:51.44 | jmetro | OR..the other end yeah |
| 17:51.44 | Maliuta | GMT+10 |
| 17:51.45 | jmetro | i was thinking PM |
| 17:51.51 | dtx | Speaking of time, it's lunch time here. |
| 17:51.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:52.02 | [TK]D-Fender | [13:48]cjm_Hi Folks, FreePBX lets me enable a LOT of codecs, but by default only three are enabled. Is there any reason why I shouldn't enable all the codecs asterisk knows? <- Asterisk also knows G.729 ... doesn't mean you have the codec file & its associated licenses installed. And when that because the agreed codec and you have to play something back with transcoding, BAM dropped call. |
| 17:52.12 | *** join/#asterisk geeksteve (~geeksteve@195.110.168.123) |
| 17:52.25 | [TK]D-Fender | cjm_: Details count. Please stand over there on that plastic sheet, willya? |
| 17:52.33 | *** part/#asterisk geeksteve (~geeksteve@195.110.168.123) |
| 17:52.37 | jmetro | He's about to get SLIMED! |
| 17:53.16 | dtx | cjm_: Also, the codecs that compress the most use the most resources. |
| 17:53.50 | dtx | Well, the most CPU, but less bandwidth obviously. |
| 17:53.55 | cjm_ | [TK]D-Fender, Thanks. That is the answer to my question and I appreciate the succinct, if undeserved, reprimand. (-: Don't permit the negotiation to arrive at an agreement you can't support. |
| 17:54.25 | dtx | What are the three it has enabled by default? |
| 17:54.45 | [TK]D-Fender | cjm_: Just giving you ana analogy. |
| 17:55.11 | *** join/#asterisk peetaur2 (~peter@x2f056e4.dyn.telefonica.de) |
| 17:55.27 | cjm_ | So, what is a good set of codecs to enable? Let's assume I have no proprietary licenses, because I don't. (-: |
| 17:55.41 | cjm_ | dtx, ulwa, alwa, and gsm. |
| 17:55.53 | Chainsaw | cjm_: ulaw & alaw are pretty much the gold standard. If you can spare the bandwidth, they are good defaults. |
| 17:56.06 | [TK]D-Fender | cjm_: You shouldn't have to care on a global level... because your TRUNKS & EXTENSION eact have their own space for this |
| 17:56.37 | Chainsaw | cjm_: But indeed. Configure per trunk/elephant. |
| 17:57.31 | cjm_ | [TK]D-Fender, My trunks, at least initially will be POTS lines. But I expect more and more "normal" traffic to be VoIP, where this is a detail that counts, to quote ... |
| 17:58.37 | dtx | cjm_: If you are using SIP trunking I think they usually try to get you to use ulaw and alaw |
| 17:59.06 | dtx | Mine only accepts those |
| 17:59.07 | cjm_ | So, the conclusion is don't enable codecs that might cause the negotiation to arrive at a agreement to use something I have but is unlicensed. |
| 17:59.15 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 17:59.29 | cjm_ | dtx, Module Admin let me add a whole pile of codecs. |
| 17:59.37 | Chainsaw | cjm_: Indeed. Or make Digium very happy and buy one of their G729 transcoder cards. |
| 18:00.05 | [TK]D-Fender | cjm_: Asterisk is software. What are you plugging those POTS lines into? |
| 18:00.57 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 18:01.07 | cjm_ | So, the natural question is, how will I discover the case where a known, freely available codec can save me network bandwidth? (I can add more CPU power much more easily and inexpensively than I can add link bandwidth.) |
| 18:01.40 | cjm_ | [TK]D-Fender, The POTS lines will connect to a Rhino card and Asterisk will have them configured. |
| 18:01.40 | [TK]D-Fender | Discover a case? |
| 18:01.58 | [TK]D-Fender | cjm_: The bandwidth of each codec is documented. There is nothing to "discover". |
| 18:05.08 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:05.23 | cjm_ | [TK]D-Fender, Let me see if I understand... I have 15 codecs currently known to Asterisk, and I have three enabled. If my VoIP performance starts to degrade as more and more calls traffic comes through Asterisk, then I need to see how frequently each of my three codecs are being used, investigate the bandwidth they each use and see if one of the other codecs can do the same job trading CPU for link? |
| 18:06.38 | ChannelZ | Chances are your ITSP only supports a few of them anyway |
| 18:06.53 | ChannelZ | ulaw,alaw,gsm, and g729 (which you have to license) |
| 18:06.58 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:07.09 | igcewieling | ChannelZ: many don't support GSM |
| 18:07.25 | igcewieling | both of our upstreams only support ulaw and g729 |
| 18:08.00 | cjm_ | ChannelZ, I think that helps me... I only need these for VoIP. My POTS traffic has NO encoding, so POTS traffic doesn't matter at all. Is this right? |
| 18:08.19 | ChannelZ | well it still depends on the path |
| 18:09.18 | ChannelZ | if you have a SIP device going to your Asterisk which then goes out analog POTS, the SIP device is still encoded |
| 18:09.19 | cjm_ | Will my logs tell me the number of times negotiation would have used a more link-conservative codec and I vetoed it? |
| 18:09.31 | ChannelZ | so you don't want to use some crap sounding codec on the sip phone |
| 18:09.50 | [TK]D-Fender | cjm_: I'm not sure you understand the concept of "negotiation. When you allow * to use 10 ... the other end will also have their list... and what they settle on will ALWAYS be the same. |
| 18:10.26 | *** join/#asterisk transfinite (~aleph_0@unaffiliated/transfinite) |
| 18:10.28 | [TK]D-Fender | cjm_: There is no "rotation", there is no "failover". I prefer 1,2,3, you prefer 3,2,1, we settle on 2 EVERY time |
| 18:10.33 | ChannelZ | I actually do not know, internally, what the D to A on a TDM card is capable of for instance, if through DAHDI it winds up getting converted to normal ulaw 8khz before being converted to analog |
| 18:10.56 | [TK]D-Fender | cjm_: As for what to save on... if You set your codecs, then you already know what is in use. There is no need to "detect" this. You set it. |
| 18:11.22 | cjm_ | [TK]D-Fender, Yes. I get that... But, I have 3 and the other guy has 7, and suggests a codec that is more link conservative but I say I can't, so we agree on one of my 3. I;d like to know how often that happens, because it means I should probably look into more and better codecs. Right? |
| 18:11.24 | ChannelZ | (IE would a wideband codec between a SIP device and astersk sound maybe a little better once it hit analog) |
| 18:13.01 | cjm_ | ChannelZ, Yes. This is a really good point. Can I configure my codecs depending on the route? SIP to POTS can be high qualify because the link-bandwidth is not an issue. |
| 18:13.26 | ChannelZ | you can configure per sip peer yes |
| 18:13.43 | [TK]D-Fender | cjm_: Offering 10 is pointless if it'll always settle on the same. It will never change on the fly. once you know what they support, JUST PICK IT. |
| 18:13.47 | igcewieling | cjm_: Asterisk will NOT and CANNOT dynamically pick a codec based on anything except what the far end offers. |
| 18:13.51 | cjm_ | ChannelZ, Thanks. I'll keep this in mind. |
| 18:13.52 | ChannelZ | but not per "route" per se because that is negotiated. |
| 18:13.58 | [TK]D-Fender | cjm_: This is for each and every device you talk to. |
| 18:14.26 | [TK]D-Fender | cjm_: Stop thinking there is a choice to make during the offer. There isn't. All you'll see is the same result every time. Offer 3 is poitnless. |
| 18:15.03 | cjm_ | [TK]D-Fender, I'm confused. How do we arrive on the best one in common if we don't both know the full list of the other? |
| 18:15.37 | [TK]D-Fender | As soon as you place a call or get one from them they SEND the list |
| 18:15.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:15.48 | ChannelZ | because "the best one" is subjective |
| 18:15.52 | [TK]D-Fender | or there is this littl ething about reading their PAGE as to what they support |
| 18:16.07 | [TK]D-Fender | Providers list this stuff you know |
| 18:16.23 | [TK]D-Fender | but technically "enabling 3" will always net you the same result |
| 18:16.33 | [TK]D-Fender | \It's always settle on the same one |
| 18:16.37 | [TK]D-Fender | It'll* |
| 18:16.49 | ChannelZ | just do allow=!all,ulaw,alaw,gsm and move on with your life |
| 18:17.14 | igcewieling | In my experience allowing more than one codec for a peer is going to cause drama and misery at some point. |
| 18:17.42 | cjm_ | [TK]D-Fender, ChannelZ I can tell you guys are frustrated with me. I think I am failing to pose my question properly... |
| 18:17.53 | [TK]D-Fender | Possibly |
| 18:18.16 | igcewieling | cjm_: I think the issue is more that you don't understand that Asterisk is dumb as a rock when it comes to figuring out which codec to use. |
| 18:18.37 | [TK]D-Fender | There is nothing to be "smart" about |
| 18:19.33 | [TK]D-Fender | When you propose 10 and they propse 10 where they meet in the middle will always be the same... so there is no point is offering 10 |
| 18:19.39 | cjm_ | So, I have codecs A , B, and C. The other side as A, B, C, and D. Let's assume they get "better" from A to D. We will agree on C, but I want to know how often I had the option to use D and didn't because I don't have it. It might be worth it to get D. |
| 18:19.41 | [TK]D-Fender | pick what you want to use and that is all |
| 18:20.10 | [TK]D-Fender | cjm_: Do you think your options are changing? Your phone support whatever they support, same with your provider |
| 18:20.17 | jmetro | allow=g722 and thats it, just stop there |
| 18:20.17 | [TK]D-Fender | cjm_: this isn't miraculously going to change |
| 18:20.32 | [TK]D-Fender | There is no "how often". |
| 18:20.40 | [TK]D-Fender | They don't offer me 3 today and 10 tomorrow |
| 18:20.47 | [TK]D-Fender | And 7 the next day |
| 18:21.21 | ChannelZ | Check your provider. Chances are they're only going to list 2 or 3. |
| 18:21.24 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 18:21.33 | cjm_ | [TK]D-Fender, There is one point of confusion. I have softphones only. |
| 18:22.29 | ChannelZ | Are they all local? |
| 18:22.40 | cjm_ | [TK]D-Fender, So my phones are not imposing any restrictions. I can see how the choice of phones would make all of this discussion irrelevant. |
| 18:22.59 | [TK]D-Fender | cjm_: They're your softphones You telling me you can't SEE what codecs they support just by staring at it? |
| 18:23.22 | jmetro | tell your softphones to support only speex. that is the best sounding codec. |
| 18:23.25 | ChannelZ | The consideration for you as a human being is balancing how much bandwidth you want to spend vs the quality of the audio. |
| 18:23.39 | cjm_ | ChannelZ, They will be local for now. external call will got out on POTS lines, for now. |
| 18:23.51 | Chainsaw | Also, whether the softphones are on WiFi/3G. |
| 18:23.54 | cjm_ | jmetro, Thanks. |
| 18:24.06 | Chainsaw | Because that means they'll barely have any bandwidth. And thus ulaw/alaw are no longer a great idea. |
| 18:24.22 | jmetro | cjm_: that was a horrible idea, never take my advice. |
| 18:24.50 | [TK]D-Fender | cjm_: Local will be practically irrelevant as bandwidth does. |
| 18:25.19 | cjm_ | Chainsaw, ChannelZ At least for the near term, they will all run on desktops and maybe have a USB headset. |
| 18:25.34 | [TK]D-Fender | cjm_: And speex may be a better code than say G711... but that is what the PSTN uses so your calls are all going to get down=mixed anyway |
| 18:25.53 | Chainsaw | cjm_: I would say ulaw/alaw and don't worry about the rest. |
| 18:26.06 | Chainsaw | cjm_: Your provider is going to support those, it is exactly the audio quality that POTS lines will give you... |
| 18:26.27 | ChannelZ | you might throw one wideband codec in your local devices so if you're calling between them on softphones they'll sound a bit better |
| 18:26.28 | Chainsaw | cjm_: And if you don't plan to scale this up to hundreds or thousands of users you can likely spare the bandwidth to do so. |
| 18:26.37 | cjm_ | Chainsaw, I don't think i will have a "provider", unless you mean the POTS lines. |
| 18:26.52 | ChannelZ | otherwise yes, just do ulaw and be done with it. |
| 18:26.54 | [TK]D-Fender | cjm_: then bandwidth is pretty much nothing. |
| 18:26.57 | cjm_ | Chainsaw, Ha! Right now I am talking about four users! |
| 18:27.12 | [TK]D-Fender | cjm_: My watch can do that... and it's ANALOG |
| 18:27.22 | Chainsaw | cjm_: ulaw & alaw will suffice quite nicely I suspect. |
| 18:27.28 | [TK]D-Fender | cjm_: This was a very sad exercise |
| 18:27.41 | Chainsaw | cjm_: I second the ChannelZ suggestion of looking for G722 support in the softphones, for wideband internal calls. |
| 18:27.58 | [TK]D-Fender | Sure, why not.... |
| 18:28.03 | Chainsaw | cjm_: But if it's a licensed extra on the softphone side I wouldn't bother. |
| 18:28.13 | cjm_ | [TK]D-Fender, O.K., Apparently I am thinking about something that doesn't exist owning to my general ignorance of the whole topic. But, I am beginning to understand. |
| 18:28.39 | Chainsaw | cjm_: I think in your mind you could transcode a stream multiple times. |
| 18:28.56 | Chainsaw | cjm_: Which would allow for some quite complicated setups. But in the real world, you can't. There is only one media stream. |
| 18:29.49 | cjm_ | Chainsaw, No... The fact that I can't make my question clear means that I am clearly asking the wrong question. You guys don't even understand my confusion, which means I need to read the book farther to see why I have a question that is apparently non-sense. |
| 18:30.10 | Chainsaw | cjm_: Fender does have this effect on people. |
| 18:30.10 | [TK]D-Fender | cjm_: Problem is you don't understand your confusion :) |
| 18:30.11 | jmetro | cjm_: What is your original question again? What are you trying to do? |
| 18:30.20 | igcewieling | cjm_: the issue is you are assuming capabilites Astrisk and SIP do not have |
| 18:30.48 | cjm_ | [TK]D-Fender, Yes! And neither to you, so don't give me any attitude. (-: |
| 18:31.04 | jmetro | I have a complicated question. |
| 18:31.21 | Chainsaw | jmetro: Will you be asking it later? |
| 18:32.14 | jmetro | I have a call queue with 2 members in it. If one of the members picks up, i want the other member to not be rang if a call comes in while the first member is on the phone |
| 18:32.23 | jmetro | basically i want to tie the two members together so that if one is in use, the other is also in use. |
| 18:32.46 | cjm_ | jmetro, The original question is, How can I tell when the world has produced a better codec than the one(s) I am using, because the other side continues to offer is and I continue to reject it because I dont have it. I want to find out if my logs will tell my, "Chris, you're a moron. You continue to use those COBOL-age codecs, when everybody else is using these slick new ones." |
| 18:32.52 | jmetro | [and therefore do not ring with a new call coming in] |
| 18:33.06 | igcewieling | cjm_: the answer is you can't. |
| 18:33.19 | [TK]D-Fender | jmetro: use a composite hint as their state device <- |
| 18:33.50 | Chainsaw | cjm_: The "slick new one" is currently G729. You'll have to spend money on licenses or transcoder cards. |
| 18:33.51 | cjm_ | igcewieling, That's a fair answer. Asterisk doesn't log the full negotiation? |
| 18:34.01 | Chainsaw | cjm_: With a 4 user setup.... probably a bit too early to invest. |
| 18:34.05 | [TK]D-Fender | cjm_: Forget "world". Doesn't amtter if the world creates a new codec... ASTERISk has to support it |
| 18:34.06 | igcewieling | cjm_: no. |
| 18:34.23 | [TK]D-Fender | cjm_: As for that... that's why you check for NEW versions.,. and changelogs, etc\ |
| 18:34.29 | cjm_ | Chainsaw, Yeah... I bit. |
| 18:34.30 | igcewieling | cjm_: you can do a sip debug to get the actual packets logged. you can run Dumpchan() after the call ends to see what codecs were used. |
| 18:34.41 | [TK]D-Fender | cjm_: And People don't add codecs to asterisk every day.. or year, even |
| 18:34.59 | igcewieling | [TK]D-Fender: and when they do add codecs, they are seldom ones anyone cares about |
| 18:35.06 | Chainsaw | They'll probably leave it to rot in the bug tracker for a good year or two first, yeah. |
| 18:35.07 | [TK]D-Fender | cjm_: Because the world doesn't care about ever guy coding in his basement so newfangled little codec. |
| 18:35.22 | Kobaz | ~itsplist-usa |
| 18:35.37 | Qwell | ~itsplist-us |
| 18:35.38 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:36.59 | [TK]D-Fender | cjm_: When your softphose support something new... they'll tell you. When Asterisk Supports something new... it'll be documented as well. |
| 18:37.29 | cjm_ | [TK]D-Fender, igcewieling Chainsaw ChannelZ jmetro Thanks very much for the help. I beg your pardon if my ignorance frustrated you, but this conversation has been much more of a help than you might imagine, and I appreciate your help. |
| 18:37.48 | [TK]D-Fender | I hope so... |
| 18:38.00 | Chainsaw | cjm_: Soon you'll know more about telephony than you ever wished to know :) |
| 18:38.16 | cjm_ | Chainsaw, I have very big wishes... (-: |
| 18:39.05 | [TK]D-Fender | cjm_: I also live in a perfect world.. it ends daily with annoying buzzing sound at 7am |
| 18:40.04 | ChannelZ | No prob. I think your stumbling block is assuming these systems will want to "pick the best codec" given a list of 50, but the thing to realize is that they have no idea what "the best" is and given that entire list wouldn't make the decision you think they would. |
| 18:40.44 | Chainsaw | It'll do silly things like use G722 towards a desk phone and transcode to ulaw for the ISDN gateway. |
| 18:40.56 | ChannelZ | Particularly because "the best" is subjective. The best for what? Voice quality? Bandwidth? CPU power? If there was a single easy answer, we wouldn't have all these codecs in the first place. |
| 18:43.52 | cjm_ | It's all very confusing right now... There are apparently huge icebergs chunks of confusion that my TItanic understanding keeps hitting... (-: |
| 18:45.12 | ChannelZ | Fear not, it gets worse |
| 19:01.36 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 19:03.33 | *** join/#asterisk module000 (~module000@CEDAR-CREEK.edge9.Dallas1.Level3.net) |
| 19:03.41 | *** part/#asterisk module000 (~module000@CEDAR-CREEK.edge9.Dallas1.Level3.net) |
| 19:03.48 | *** join/#asterisk module000 (~module000@CEDAR-CREEK.edge9.Dallas1.Level3.net) |
| 19:06.57 | apb1963 | so I just got 4 calls within seconds of each other... An apparent hack attack, although I'm not quite sure what I'm seeing here. http://ix.io/6Ig |
| 19:10.21 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 19:11.31 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:11.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:14.59 | ChannelZ | some hacky script kiddie |
| 19:15.11 | ChannelZ | what was that output even from? |
| 19:16.11 | jmetro | [TK]D-Fender: How does composite hint? |
| 19:17.20 | [TK]D-Fender | make a hint that points to both devices |
| 19:19.10 | jmetro | so like exten=> 108,hint,SIP/108&SIP/109 and thats it? |
| 19:23.08 | [TK]D-Fender | yes |
| 19:24.31 | jmetro | doesnt seem like it worked. unless i reloaded wrong |
| 19:24.37 | jmetro | i did a dialtab reltab |
| 19:25.09 | Qwell | jmetro: In your queues.conf, set the state interface to that |
| 19:25.16 | jmetro | ahh |
| 19:25.22 | [TK]D-Fender | So far I'm not seeing you fully implement what I said... |
| 19:25.25 | jmetro | to sip/108&sip109 ? |
| 19:25.28 | Qwell | 3rd or 4th arg, I think |
| 19:25.30 | apb1963 | output is debug from the softphone... phonerlite |
| 19:25.42 | Qwell | jmetro: no, Local/108@whatevercontext |
| 19:26.03 | Qwell | maybe |
| 19:26.42 | apb1963 | what scares me is that it appears to be attempting to call out using my IP address. |
| 19:27.41 | Qwell | apb1963: You have guest=yes in sip.conf, don't you? |
| 19:27.48 | jmetro | Qwell: Hm. i think i might post my extensions |
| 19:27.51 | apb1963 | nope |
| 19:27.57 | apb1963 | but i'll doublecheck |
| 19:28.25 | [TK]D-Fender | jmetro: No need to use local channels. |
| 19:28.30 | apb1963 | allowguest=no |
| 19:28.30 | [TK]D-Fender | (yet) |
| 19:30.04 | jmetro | hm |
| 19:30.13 | jmetro | im trying to avoid using my local routing and have it dial my members via peername |
| 19:30.35 | Qwell | jmetro: state_interface has nothing to do with actually dialing |
| 19:30.42 | [TK]D-Fender | jmetro: then do it. You can still specify the state device to use |
| 19:30.44 | jmetro | Oh yeah. |
| 19:30.58 | jmetro | so set the interface for both to Local/108 |
| 19:31.03 | apb1963 | the script kiddy is using sipcli ... which Is apparently written by some turkey from Turkey. |
| 19:31.04 | jmetro | @thing |
| 19:31.45 | cjm_ | So, I have two extensions configured (100 and 200), and I have a softphone registered with my Asterisk. If I dial a non-existant extension, I expect to get something that says, "Non-existent Extension." and I don't. When I dial the other extension I would expect to be told, "That extension is not on-line right now." and I don't. If I dial myself, I would expect a busy indication, and I don't. So, either I don't understand what I am seeing, or something is misco |
| 19:31.45 | cjm_ | nfigured, and I am pretty sure it is misconfigured. How can I start to figure this out? |
| 19:31.47 | Qwell | jmetro: something like that |
| 19:32.02 | Qwell | ~book |
| 19:32.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:32.04 | Qwell | cjm_: start there |
| 19:32.47 | cjm_ | Qwell, yeah... I'm on page 85. |
| 19:32.53 | [TK]D-Fender | cjm_: it matches what you tell it to match. So you need to specify the ones you do care about and then have a larger CATCH-CALL pattern that will grab everything else. That is still more dialplan... |
| 19:32.57 | jmetro | oh you know what, i have ringinuse enabled but the user doesnt notice it on the phone they are talking on..so this will never work |
| 19:33.21 | cjm_ | [TK]D-Fender, O.K., That is farther in the book. Thanks. |
| 19:33.29 | cjm_ | Qwell, Thanks. |
| 19:34.38 | jmetro | cjm_: btw, you most likely have multiple lines which stops you from getting a busy signal. |
| 19:35.05 | jmetro | cjm_: for the others, you have to check the dial and see if it worked, if extension exists, etc.. it wont do it unless you program it. |
| 19:35.51 | jmetro | hell you could have it dial the number, and if the phone is offline, run a script to rm -rfv * |
| 19:41.29 | *** join/#asterisk gtTuna (~Tuna@d149-67-253-75.try.wideopenwest.com) |
| 19:42.31 | gtTuna | When trying to dial an external extension (on another network) I have setup (ext 200), I get the following: [Jul 17 15:28:36] WARNING[1112]: chan_sip.c:9382 set_destination: Can't find address for host '200' |
| 19:42.32 | gtTuna | <PROTECTED> |
| 19:43.05 | [TK]D-Fender | gtTuna: and do you see one in "sip show peer 200"? |
| 19:43.44 | gtTuna | 200/200 50.x.x.x D N 1039 OK (76 ms) |
| 19:43.52 | gtTuna | obviously the x's aren't actually there |
| 19:44.55 | gtTuna | i have nat=yes in the extension's config |
| 19:45.01 | [TK]D-Fender | gtTuna: show us the actual full call |
| 19:47.14 | gtTuna | i accidently saved my putty log as raw data :( will need to wait until the user gets back |
| 19:49.18 | jmetro | okay, let me expand upon my problem |
| 19:49.21 | *** join/#asterisk random_cat (~random_ca@gateway/tor-sasl/randomcat/x-49498005) |
| 19:49.22 | jmetro | I have a user with two phones on her desk. |
| 19:49.29 | jmetro | Both are in my queue. |
| 19:49.51 | jmetro | When one is active, the other should be silent. |
| 19:50.20 | [TK]D-Fender | jmetro: Yeah. We got that. Now show us your SETUP so we can see where you went wrong. |
| 19:50.22 | [TK]D-Fender | :p |
| 19:50.23 | jmetro | i dont know if its possible. I have ringinuse enabled so even though its "busy" it still rings...the phone she is using also rings but its silent because it is just ringing another line. |
| 19:50.38 | jmetro | on the same physical phone |
| 19:51.29 | jmetro | i think im just gonna say impossible unless i turn ringinuse off, which i cant do because she is the only one in the queue. |
| 19:52.19 | [TK]D-Fender | You already said you don't WANT it to ring while EITHER is in-use |
| 19:52.40 | jmetro | Precisely. |
| 19:52.51 | jmetro | ring as in , make noise. |
| 19:52.59 | jmetro | since shes on the phone, with a ringing phone in her other ear. |
| 19:55.07 | *** join/#asterisk imox (~imox@24-134-17-195-dynip.superkabel.de) |
| 19:55.19 | cjm_ | jmetro, Thanks. I have no POTS lines connected yet. KISS, I am SIP calling from one softphone to the other on my local net. |
| 19:55.52 | jmetro | Register both phones, exten => whatever,1,Dial(SIP/peername) basically. |
| 19:55.55 | cjm_ | jmetro, Ekiga says in the manual that it doesn't support call waiting, or at least I think it said that. |
| 19:56.07 | cjm_ | jmetro, Yeah.. |
| 19:56.07 | jmetro | from there you can pick up the result of that dial in asterisk code and decide what to do |
| 19:56.22 | cjm_ | O.K., Thanks. |
| 19:56.36 | jmetro | same => n,Dial(${ARG1},20,tr) |
| 19:56.36 | jmetro | <PROTECTED> |
| 19:56.39 | jmetro | is what mine looks like |
| 19:56.54 | jmetro | and then you have Exten=> s-BUSY or s-NOANSWER etc. |
| 19:57.18 | [TK]D-Fender | just use LABELS rather than hacked resulst. |
| 19:57.33 | [TK]D-Fender | and I doubt Ekiga does not support "call waiting" |
| 20:02.52 | jmetro | [TK]D-Fender: how would you do what i did with the exten=> s-STATUS ? |
| 20:02.57 | jmetro | goto's? |
| 20:03.08 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-wziorjjhazwisjud) |
| 20:03.33 | [TK]D-Fender | jmSame gog... just to a LABEL within the current exten |
| 20:03.34 | igcewieling | jmetro: see extensions.conf.sample, the stdexten macro shows you |
| 20:03.37 | [TK]D-Fender | goto* |
| 20:03.46 | igcewieling | oh, bnevermind |
| 20:04.15 | jmetro | Thats fine too |
| 20:04.29 | jmetro | I kinda like the extens, they show up nice and purple. visually appealing |
| 20:05.45 | *** join/#asterisk peetaur2 (~peter@x2f056e4.dyn.telefonica.de) |
| 20:07.34 | *** join/#asterisk CeBe (~CeBe@port-92-206-91-157.dynamic.qsc.de) |
| 20:07.37 | *** join/#asterisk jkroon (~jkroon@196.37.70.13) |
| 20:08.07 | jkroon | hi all, does asterisk have the ability to dump the content of udptl frames in a human-readable manner? |
| 20:08.36 | igcewieling | jkroon: I don't know, but you can always use wireshark |
| 20:09.04 | jkroon | I've got two SIP/ channels with a local (non-optimized) channel inbetween and am having issues receiving faxes (ok, from client perspective) |
| 20:09.12 | jkroon | hmm, thanks, i'll make a quick pcap capture. |
| 20:09.56 | jkroon | broadsoft using SIP to asterisk, pass it through a Local/ channel, back to SIP to GS HT502 HW (configured to use t38) => Konica Minolta FAX machine. |
| 20:11.00 | jkroon | if I originate a call directly to SIP/client and use SendFAX to send the fax it works, if the call comes in from broadsoft side, transmission fails. |
| 20:11.17 | igcewieling | jkroon: T.38 is so fragile I'd hesitate to run the calls through Local/ |
| 20:11.41 | igcewieling | jkroon: a sip client which supports T.38? |
| 20:12.01 | gtTuna | [TK]D-Fender, http://pastebin.com/40Lz6Hdn |
| 20:12.02 | jkroon | igcewieling, the Local/ was there long before the T.38 came along, and getting rid of it will be ... hard, to say the least |
| 20:12.42 | jkroon | ok, redirecting the originate to also go via Local/ makes no difference. |
| 20:12.46 | jkroon | still works. |
| 20:13.42 | igcewieling | jkroon: Make SURE it is switching to T.38 from whatever codec you are using up to the CNG |
| 20:14.01 | igcewieling | We have a problem where Asterisk won't switch to T.38 in some situations |
| 20:14.55 | jkroon | udptl set debug shows udptl messages flowing in both directions - is that good enough? |
| 20:15.05 | igcewieling | jkroon: should be |
| 20:15.49 | jkroon | igcewieling, "some situations" ? |
| 20:15.56 | *** join/#asterisk kusznir (~kusznir@76.178.145.28) |
| 20:16.07 | igcewieling | jkroon: "always" unless Asterisk is terminating the call to ReceiveFax |
| 20:16.28 | igcewieling | I'm sure it is something specific to our setup, so I doubt our experience would apply to you. |
| 20:16.33 | jkroon | ... odd, sending the call into ReceiveFAX always "just worked" for me. |
| 20:17.09 | igcewieling | jkroon: same for us, but SendFax wants to use ulaw not T.38 and if we pass the call through Asteirsk to a T.38 endpoint it simply doesn't work at all. |
| 20:17.17 | jkroon | even switches cleanly from G.729 => T.38, but I suspect that's because the broadsoft side does the switch ... |
| 20:17.34 | igcewieling | Like I said, it works SO terribly I'm sure it is something specific to our weird setup. |
| 20:17.44 | jkroon | channel originate SIP/foo application SendFAX /tmp/file.tiff ... uses t.38 for me. |
| 20:18.09 | jkroon | igcewieling, ast version? |
| 20:18.18 | igcewieling | jkroon: if our issue as a general issue LOTS of people would be reporting it. |
| 20:18.33 | igcewieling | jkroon: 1.8.early, 1.8.latest, and 11 |
| 20:18.54 | jkroon | 1.8 I recall some weird things, 11.2 onwards has been mostly good to me. |
| 20:19.08 | jkroon | keyword being mostly ... |
| 20:19.16 | igcewieling | Need to wait another couple of months before trying it again -- we almost lost customers over this issue when we tried routing fax calls via Asterisk |
| 20:19.25 | jmetro | had an odd issue in one of the earlier 11's that would cause sip autodestructs to render VM unusable |
| 20:19.39 | gtTuna | seems like the call hangs up at exactly 5 seconds |
| 20:19.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.218) |
| 20:20.20 | igcewieling | jmetro: early Asterisk 11s had NAT screwed up when you used the recommended settings instead of nat=yes, that might be it |
| 20:21.42 | jmetro | later than that one |
| 20:21.49 | *** join/#asterisk gusto (~gusto@ppp-62-216-205-41.dynamic.mnet-online.de) |
| 20:21.50 | [TK]D-Fender | digium123 |
| 20:21.50 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 20:22.00 | jmetro | r360139 currently |
| 20:22.17 | jmetro | oh wait im on 10..hm |
| 20:22.49 | jmetro | SVN-branch-11-r378219 there you go |
| 20:22.51 | gtTuna | can anyone take a look at that pastebin i posted and see if they have any ideas on why calls would be hanging up after 5 seconds? |
| 20:22.58 | gtTuna | obviously BYE sip:xxxxxx0013@xxx.xxx.60.133 SIP/2.0 is an issue |
| 20:23.00 | jmetro | couple V's before that there was the VM breaker. |
| 20:23.17 | module000 | jmetro: i had that =P |
| 20:24.41 | digitalirony | http://pastebin.com/6V2W8w4w this is running asterisk 1.8.23. It is a sip call. when it reaches the VM, the playback vm-theperson.gsm doesn't comeplete before it plays the beep.gsm. This only happens when you use the IVR to get to the VM, it doesn't happen if you bypass the IVR and go straight to the VM |
| 20:24.55 | digitalirony | can someone offer me some advice, or anything at all? |
| 20:26.44 | jkroon | igcewieling, could it be that the fax machine refuses to receive at a baud of 2400? |
| 20:27.14 | jkroon | if I send using SendFAX the baud seems to negotiate to 14400, but if it comes in from my upstream it seems to want to go down to 2400 ... which SUCKS |
| 20:29.02 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 20:29.42 | jkroon | igcewieling, how well do you know what needs to happen at the low level of the call? |
| 20:30.47 | jkroon | the one that doesn't work kick off with g.729, then I see a RTP EVENT, FAX CNG, followed a few frames later by a re-INVITE (receiving GW => asterisk) asking to switch to T.38 |
| 20:31.09 | jkroon | an almost immediate response with 100 Trying and 200 OK. |
| 20:33.25 | *** join/#asterisk gusto (~gusto@ppp-62-216-205-41.dynamic.mnet-online.de) |
| 20:34.00 | jkroon | directly after I see a LOT of UDPTL frames going gw=>asterisk, ending with a hdlc-fcs-OK-sig-end before no-signal |
| 20:34.14 | jkroon | asterisk then starts responding with what I presume is coming from the other end ... |
| 20:34.39 | jkroon | including t30-data @ 2400 |
| 20:35.38 | igcewieling | jkroon: I consider T.38 to be my Kryptonite. |
| 20:35.59 | jkroon | lol |
| 20:36.22 | jkroon | where is irroot when you need him ... |
| 20:36.52 | WIMPy | Haven't seen him for a long time. |
| 20:38.24 | jmetro | i think theres an infobot function for that |
| 20:38.30 | jmetro | ~seen irroot |
| 20:38.37 | infobot | irroot <~gregory@2002:c5ac:a5d6::1> was last seen on IRC in channel #asterisk, 322d 3h 46m 26s ago, saying: 'yo yo'. |
| 20:38.37 | jmetro | or something |
| 20:39.13 | *** join/#asterisk dash_ (~dash@unaffiliated/dash-/x-7576607) |
| 20:39.59 | jmetro | last seen saying "yo yo". I like it. |
| 20:41.42 | jkroon | igcewieling, fax machine is responding with "failure to train" |
| 20:41.50 | jkroon | guessing that means it doesn't like the baud rate ... :( |
| 20:45.11 | jkroon | POC minolta |
| 20:45.29 | jkroon | curses and swears at email2fax providers. |
| 20:45.40 | jkroon | make your users pay more by slowing down the faxes ... |
| 20:46.04 | *** join/#asterisk peetaur2 (~peter@x2f056e4.dyn.telefonica.de) |
| 20:46.36 | *** join/#asterisk Fuzai (~quassel@c-24-19-227-166.hsd1.wa.comcast.net) |
| 20:47.17 | CeBe | email2fax? So they print all the spam? :-D |
| 20:47.46 | Fuzai | Hi could someone recomend a good cheap ata box that supports basic hd audio for a single phone line a home? I've got an SPA-2102-SF from clearwire and it's locked beyond locked and I need to replace it for my new voip service. |
| 20:49.23 | WIMPy | Fuzai: That doesn't make sense. You can't have `HD' on analog. |
| 20:50.04 | Fuzai | fair enough, I only knew HD as the current standard from my smartphone and it's useage of it. So then, an ATA box worth a crap that isn't going to break my bank :) |
| 20:50.40 | apb1963 | jkroon: Have you tried g.711 ? |
| 20:53.04 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:53.15 | jkroon | apb1963, not really an option since my upstream only provides for g729 |
| 20:53.23 | jkroon | ie, the source where I get the call from. |
| 20:53.34 | jkroon | so if I don't switch to t.38 i stand absolutely no chance. |
| 20:53.48 | gtTuna | Having issues with a specific phone. No other issues with the other ~200 endpoints on the server. When trying to dial an external extension (on another network) I have setup (ext 200), I get the following: [Jul 17 15:28:36] WARNING[1112]: chan_sip.c:9382 set_destination: Can't find address for host '200'. Then the SIP traces show a 'BYE' directly afterwards. Here's a pastebin: http://pastebin.com/40Lz6Hdn |
| 20:53.54 | apb1963 | have you tried a different provider? |
| 20:54.41 | jkroon | apb1963, i've worked with a few, and so far this one is the one with the fewest issues. |
| 20:55.13 | apb1963 | rumor has it t.38 is your best bet |
| 20:55.50 | jkroon | hehe, yea, until the remote end insists on 2400 baud and the minolta refuses to negotiate at that baud and keeps sending "failure to train" |
| 20:55.56 | igcewieling | gtTuna: what does "sip show peer 200" show for the IP address? |
| 20:56.34 | apb1963 | settings? |
| 20:56.39 | jkroon | so i don't think current issue is as much asterisk as in the two endpoints are ... can't find polite/descriptive words ... not in agreement about what needs to happen |
| 20:56.47 | jkroon | apb1963, at what point? |
| 20:56.54 | apb1963 | endpoints |
| 20:56.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:57.26 | jkroon | basically the call switches from g729 => t38, so that's fine, but the fax machine keeps sending hdlc messages contain "failure to train", as in sip show peer? |
| 20:57.29 | igcewieling | gtTuna: almost all "hangs up when answered" issues are NAT related or firewall related |
| 20:57.56 | apb1963 | what are the settings on the fax machine? |
| 20:58.09 | gtTuna | 200/200 50.158.xxx.xxx D N 1039 OK (80 ms), where the x's match |
| 20:58.39 | jkroon | it's a konica minolta, other than the settings through the web interface (which basically I don't follow) I don't know what else I can do? |
| 20:58.55 | jkroon | and I have no idea about the remote end (nor do I have control over it) |
| 20:59.03 | apb1963 | perhaps it's time to follow them? :) |
| 20:59.57 | gtTuna | igcewieling, i believe this remote user just got a brand new comcast connection installed, and is hooked directly up to their 4 port router |
| 21:00.36 | gtTuna | they're not the most technically adept, so i'm still trying to determine their network setup |
| 21:00.36 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:00.58 | jkroon | lol, apb1963 i'll get one of the guys from konica minolta to check it out in the morning. |
| 21:01.42 | jkroon | the setup works against other fax machines, now that I've managed to (thanks to igcewieling) managed to figure out what's being sent back/forth at t.38 level everything is starting to make a lot more sense. |
| 21:02.26 | apb1963 | how is it you don't know what's on the remote end? |
| 21:02.42 | jkroon | it's internet solution's email2fax gateway. |
| 21:03.01 | apb1963 | o i c |
| 21:04.40 | jkroon | so pure and simple: I know who, and conceptually what, but have no details on the actual implementation. |
| 21:04.57 | jkroon | and if they're throttling to 2400 baud on outbound you're going to see me call the CEO tomorrow. |
| 21:05.00 | igcewieling | jkroon: I did very little, maybe pointed you in the right direction |
| 21:05.44 | jkroon | igcewieling, it was you that pointed me at wireshark - without which I would still have been banging my head against the table. |
| 21:06.17 | igcewieling | 8-) |
| 21:06.57 | igcewieling | gtTuna: Are you SURE you have nat=yes set for that peer? I'd have to check, but I think "N" means "no" |
| 21:08.38 | WIMPy | thinks N means NAT. |
| 21:08.56 | gtTuna | igcewieling, http://pastebin.com/g6sCWwSa |
| 21:10.43 | apb1963 | Asterisk PBX 1.6.2.13 |
| 21:10.51 | jmetro | update! |
| 21:11.06 | apb1963 | This just in from our correspondant jmetro: |
| 21:11.30 | jmetro | I'm sorry, but our news correspondant team has been carried away by monkeys. |
| 21:12.00 | apb1963 | what a way to go. |
| 21:12.27 | apb1963 | speaking of going.... |
| 21:12.32 | apb1963 | hasta la vista |
| 21:13.54 | WIMPy | Time to 'make kaputt' |
| 21:13.55 | gtTuna | it does what it needs to do, and isnt immediately fesable to upgrade this server at the moment |
| 21:15.00 | jmetro | kaput means broken..machen kaput would be to make something broken. |
| 21:19.50 | WIMPy | Well, not yet. |
| 21:19.51 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 21:20.00 | WIMPy | just booted a new kernel. |
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| 21:22.39 | ChannelZ | to the CURB! |
| 21:24.42 | igcewieling | gtTuna: does "sip show peer 200" reflect the NAT set to yes? |
| 21:26.09 | gtTuna | if i set NAT=no, then the 'N' goes away |
| 21:26.22 | gtTuna | so N=nat like WIMPy said |
| 21:28.02 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 21:29.44 | igcewieling | gtTuna: maybe nat router for the endpoint is screwing it up. |
| 21:30.40 | gtTuna | but you seem to think its a NAT issue? |
| 21:31.15 | gtTuna | its like the phone just gives up and sends a 'BYE'....but that "set_destination: Can't find address for host" is strange |
| 21:31.35 | gtTuna | maybe i should try nat=no ? |
| 21:32.47 | gtTuna | i had a cisco 9971 that wouldnt register unless i set nat=no |
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| 21:37.01 | gtTuna | nope, that just made it unregister |
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| 21:46.39 | igcewieling | gtTuna: try setting directmedia=no |
| 21:47.38 | jkroon | gtTuna, have you tried SIP/TCP? |
| 21:48.05 | jkroon | i've seen too many IP gateways off late trying to be smart with SIP (UDP) and getting it very wrong. |
| 21:51.49 | gtTuna | sadly, any further testing with the user will need to wait until tomorrow as they're done with work for the day |
| 21:52.18 | gtTuna | jkroon, the endpoint is a Aastra 6731i that we have hundreds out in the field |
| 21:53.11 | jkroon | well, good luck with that one, but heed my warning: I've seen too many cases recently where it was NOT the SIP endpoint that was to blame, but the NAT gateway. |
| 21:53.13 | jmetro | aastra 6731's eh |
| 21:53.33 | jmetro | i always preferred the 37's |
| 21:53.38 | jmetro | 57's * |
| 21:54.02 | gtTuna | yeah, nicer phone...but for users that just want something with a handset and buttons |
| 21:54.08 | gtTuna | the 6731 works fine |
| 21:54.26 | gtTuna | we'll usually get the 'managers' a 55/57 |
| 21:54.35 | jmetro | aastras are great to add functions to as well. got some neat ones here that integrate with our timekeeping system |
| 21:54.52 | gtTuna | yeah, ive done a lot with the XML stuff |
| 21:55.13 | *** join/#asterisk andrewyager (~andrewyag@8-104-141-114.static-dsl.realworld.net.au) |
| 21:55.31 | jmetro | all i know for aastras is, if it is setup with user, auth name, password, and server, and something doesnt work, it might be broke |
| 21:55.43 | jmetro | but then again my network is perfect so it always comes down to phones |
| 21:55.57 | jmetro | what about a call internally |
| 21:56.12 | igcewieling | If I had more confidence in SIP over TCP I might try it |
| 21:56.30 | gtTuna | i mean, i know the phone is fine...i personally set it up last week and shipped it out |
| 21:56.40 | gtTuna | and tested it beforehand |
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| 21:56.56 | jmetro | doesnt mean anything, ive had the same thing but it goes to a restaurant for a week and comes back deformed by salt and flour |
| 21:57.00 | jkroon | igcewieling, recent versions of asterisk it works reasonably well. |
| 21:57.16 | jkroon | meaning: *you* should not try it on 1.8. |
| 21:57.32 | gtTuna | haven't tried an internal call..ive been testing it by calling their DID, which goes through an external SIP provider |
| 21:57.40 | gtTuna | ill test that out tomorrow as well |
| 21:57.45 | igcewieling | jkroon: *nod* the only sites where it might be useful are all on 1.8+ |
| 21:57.52 | gtTuna | yeah, i enabled TCP on my 1.6 server |
| 21:57.52 | jmetro | setup a phone in their context, dial their direct exten |
| 21:57.55 | gtTuna | bad things happened |
| 21:58.25 | igcewieling | our core asterisk "cluster" is all Asterisk 11, but the customer sites are all 1.8+ w/FreePBX |
| 21:58.38 | gtTuna | haha, mine is backwards |
| 21:59.06 | gtTuna | our asterisk routing/billing gateways are all 1.6 |
| 21:59.14 | jmetro | barfmode |
| 21:59.14 | gtTuna | everything else is 1.8/10/11 |
| 21:59.51 | gtTuna | then like this server is 1.6...just used for 'one off' setups like this person |
| 22:00.01 | gtTuna | they just need a DID and a phone with voicemail |
| 22:03.48 | raden | hugs Katty |
| 22:03.50 | igcewieling | gtTuna: the vast majority of our calls are carrier -> core asterisk cluster -> adtran Netvanta/TA9xx (customer site) -> analog or PRI -> customer PBX. The rest are carrier -> core asterisk cluster -> Freepbx (customer site) => polycom phones |
| 22:06.45 | gtTuna | ours either follow your first setup (we only use TA-9xx's), or just have the phones/softphones in the field |
| 22:06.59 | gtTuna | usually dont have onsite server |
| 22:07.52 | gtTuna | a lot of customers like the idea of having it live at our DC anyways, especially if they have frequent power issues or something |
| 22:08.17 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 22:12.28 | jkroon | ok guys, i think i've discovered another issue with asterisk 11.3.0, probably 11.4.0 as well, chan_sip. It seems that sometimes peers goes UNREACHABLE that should still be reachable, but I'll investigate that more fully on another night. |
| 22:12.46 | WIMPy | Hmm. |
| 22:12.55 | WIMPy | For like up to 3 minutes? |
| 22:13.06 | igcewieling | We have some customers with hosted service with us, but honestly it sucks and we can't do much troubleshooting |
| 22:13.34 | igcewieling | jkroon: Ha! You've discovered the "bug" which has existed since the beginning of asterisk time. |
| 22:13.54 | igcewieling | jkroon: set qualify=10000 and qualifyfreq=120 and see if the issue stops happening |
| 22:15.22 | igcewieling | if you use qualify to keep nat open, try setting qualifyfreq=30 or 45 |
| 22:16.15 | *** join/#asterisk andrewyager (~andrewyag@8-104-141-114.static-dsl.realworld.net.au) |
| 22:16.48 | jkroon | igcewieling, well, round trip in this case is 7ms, so why should it ever go over 2000ms? |
| 22:17.59 | igcewieling | jkroon: because RTT is not qualify. Qualify is RTT + time until endpoint gets around replying after it is done with all the important stuff. I don't know why setting qualify high fixed the issue for us, but it did |
| 22:18.51 | jkroon | ok, that might address an issue at one of my clients, but honestly in this case, this is between two asterisk servers that's currently doing nothing else. |
| 22:19.06 | igcewieling | jkroon: can't hurt to try |
| 22:19.43 | jkroon | as I said ... will check another night as to the details. I completely agree with "can't hurt to try" - but I'm of the "don't mess with it unless you understand it" kind :) |
| 22:19.52 | gtTuna | igcewieling, why cant you dfo much troubleshooting with hosted services> |
| 22:20.17 | jkroon | gtTuna, have you ever connected to a busy asterisk box with verbosity above 0? |
| 22:20.34 | gtTuna | every day? |
| 22:20.35 | jkroon | and with busy I mean 100+ concurrent calls, around 30k call setups/day ... |
| 22:20.53 | igcewieling | gtTuna: because there isn't one our switches on site, usually not one of our routers, and often not even our circuit. |
| 22:21.16 | gtTuna | ah |
| 22:21.47 | gtTuna | yeah, in 90% of these cases, its through our network (fiber/T1 PPP), our routers+switches |
| 22:21.55 | igcewieling | We refer to hosted customers with their own broadband as "soon to be former customers" |
| 22:22.20 | gtTuna | weve been doing some TWC stuff, but still with our router/switches behind |
| 22:22.25 | gtTuna | and thats super frustrating |
| 22:22.28 | igcewieling | gtTuna: 90% of our users are on all our own stuff, but there are some which are not |
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