| 00:05.54 | ryan_turner | Have any of you all interfaced an asterisk system with VHF radio? |
| 00:06.24 | Katty | i've never heard of VHF radio. |
| 00:06.29 | Katty | but maybe someone else has |
| 00:10.52 | ryan_turner | Like literally 2-way radios in the VHF spectrum :) |
| 00:11.11 | Katty | this is me nodding, like i know what that is. |
| 00:11.14 | Katty | nodsnods |
| 00:11.37 | ryan_turner | Even something as simple as little radios you can buy for camping and stuff at department stores. |
| 00:11.51 | ryan_turner | Id like to use a soundcard input/output to make a really basic "dumb" interface |
| 00:12.21 | ryan_turner | I can handle the radio side of stuff, but as for Asterisk handling the input etc is this something better put in a separate SIP client? |
| 00:14.40 | *** join/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
| 00:19.44 | Synx|hm | Anyone know when sips:blah@blah is used vs sip:blah@blah in SIP headers. From what i can tell in the RFC it doesn't matter what you use, and from the limited signaling i have to look at i cant find an example of asterisk using sips:blah@blah |
| 00:24.03 | sweeper | ok asterisk is *repeatedly* dying with no explanation in the logfiles |
| 00:30.57 | newtonr | sweeper: see logger.conf, be sure you have VERBOSE and DEBUG writing to the log files. If its actually crashing, get a backtrace and file an issue reporter on issues.asterisk.org https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 00:31.13 | newtonr | but of course read https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines first! |
| 00:31.34 | sweeper | ok thanks |
| 00:32.32 | newtonr | ryan_turner: googling Asterisk and radio interfaces comes up with various modules and projects. I personally haven't worked with any of them. |
| 00:32.58 | ryan_turner | Yeah, most of them look like very expensive hardware for commercial systems |
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| 00:38.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 00:50.23 | sweeper | ryan_turner: you might want to use freeswitch as the client, and connect it to asterisk |
| 00:50.29 | sweeper | lots of softphones use an fs core |
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| 01:13.38 | igcewieling | ~book |
| 01:13.39 | 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 |
| 01:14.14 | igcewieling | sweeper: "asterisk -cvvv" |
| 01:14.28 | igcewieling | the -c keeps it in the foreground so you can see the unlogged errors. |
| 01:16.11 | sweeper | ah, nice |
| 01:19.02 | transfinite | can variable expansion be used in patterns? like _1${AREACODE}NXXXXXX ? |
| 01:19.42 | WIMPy | yes |
| 01:19.54 | WIMPy | (if they are global) |
| 01:20.52 | transfinite | Thanks. And can a variable expand into a pattern, if its value begins with _ and is used where an extension is expected? |
| 01:21.30 | WIMPy | hasn't tried that. |
| 01:21.49 | transfinite | I guess it's not needed if I can do _${MYPATTERN} |
| 01:22.02 | igcewieling | transfinite: I would expect the global variable to be evaluated only when the dialplan reloads |
| 01:22.37 | WIMPy | So would I. |
| 01:22.47 | WIMPy | But I haven't tried that, either. |
| 01:23.04 | igcewieling | Generally, I think doing this is a Bad Thing, but I'm not sure why. |
| 01:23.20 | WIMPy | Why? |
| 01:23.42 | WIMPy | That's extremely handy. |
| 01:27.03 | navaismo | Hi, for a BRI card does I need a patch or use mISDN or something else I was googling around and all related topics mention mISDN and other stuff like the Digiums BRI cards can't work natively |
| 01:27.25 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 01:27.59 | navaismo | I still have hardlc errors and sometimes not working calls. Telco act like the NSA can't provide any information or help |
| 01:28.21 | WIMPy | Did yu check the cabling? |
| 01:28.45 | navaismo | I cant but the people helping me said is new |
| 01:29.08 | WIMPy | And correctly terminated? |
| 01:29.43 | navaismo | He said, i cant travel to check :( |
| 01:30.42 | WIMPy | Incorrect termination can cause trouble. |
| 01:31.23 | WIMPy | And as I said the other day, multiple vendors have their manuals wrong there. |
| 01:31.46 | igcewieling | What brand of ISDN card? |
| 01:32.06 | igcewieling | navaismo: for some reason I assumed you were in north america |
| 01:32.13 | navaismo | Nope México |
| 01:32.27 | navaismo | card: Digium Wildcard B410P |
| 01:32.32 | igcewieling | that is still north america, but not the part I was thinking of (NANPA) |
| 01:33.11 | navaismo | hehe |
| 01:33.15 | igcewieling | navaismo: DAHDI should support that, shouldn't it? |
| 01:33.30 | WIMPy | Off course it does. |
| 01:33.35 | navaismo | I think so, but all links i was reading mention a patch or misdn so i get confused |
| 01:33.54 | WIMPy | You can use it with mISDN as well. |
| 01:33.59 | igcewieling | navaismo: welcome to blogs without dates. that is all very very old information |
| 01:34.49 | WIMPy | If you want to use the "old and deprecated mISDN" supported by Asterisk, you probably will need patches. |
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| 01:35.29 | navaismo | Ok, I'll keep with latest asterisk & dahdi |
| 01:36.15 | navaismo | whey european telcos act with such secrecy? |
| 01:36.23 | WIMPy | ? |
| 01:37.52 | WIMPy | What is secret? |
| 01:38.42 | navaismo | the far debug the signaling(if applies), all |
| 01:39.06 | WIMPy | Err, what? |
| 01:39.23 | navaismo | no help at all, |
| 01:39.40 | WIMPy | Is that a box that's located in Europe? |
| 01:39.52 | navaismo | yep |
| 01:40.13 | WIMPy | They check the part up to their NT. The rest is your responsibility. |
| 01:42.43 | navaismo | Hmm maybe a bad habit to call the telco and get live debug from both sides, and of course here we use E1's |
| 01:43.25 | WIMPy | I'm sure they would do it, but I'm even more sure you wouldn;t want to pay for that. |
| 01:45.49 | navaismo | well not me, but why they even charge for a service, anyway that doesn't matter my friend always complain about the telco support--->0 support even if he pay so. I guess he is screw HAHA |
| 01:46.26 | WIMPy | Why would you normally need support? |
| 01:47.54 | WIMPy | I thinl the only people in Europe who don't hate their customers are in Ireland and Switzerland. |
| 01:48.35 | WIMPy | Just out of interest: How did you end up with a box in Europe? |
| 01:49.30 | navaismo | I met this guy along time ago, and occasionally he ask for help |
| 01:49.36 | navaismo | via forums |
| 01:53.37 | navaismo | you know, internet has no countries, religions or ethnic just nice guys helping each others |
| 01:53.56 | navaismo | end of sarcasm |
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| 02:14.31 | navaismo | thanks a lot guys |
| 02:14.44 | navaismo | You are like an oasis on bad days |
| 02:15.17 | navaismo | hope some day i can invite you a beer |
| 02:15.42 | navaismo | off |
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| 03:32.43 | sawgood | Hi: have this 'warning' filling my logs ... Asterisk 1.8 does not like some syntax (can someone offer) a suggestion: http://pastebin.com/HdYutQwm |
| 03:33.19 | sawgood | call leaves system to call a cell phone, but one provider gives a 503, so failover to another provider happens (works), but the error comes up |
| 03:33.30 | WIMPy | core show application execif |
| 03:35.53 | sawgood | WIMPy: thank you examining that now ... here is the actual dialplan (sorry) not the error from the CLI: http://pastebin.com/54L1fRba |
| 03:36.05 | sawgood | could you offer advice on what to change? |
| 03:40.44 | WIMPy | Did you read the message you posted? |
| 03:41.03 | sawgood | yes, and I am testing and retesting with different syntax |
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| 05:11.32 | ryan_turner | Im trying to connect a soft client to a siptrunk and Im getting error "Error while registering - message too large". How would I resolve that? |
| 05:16.42 | ChannelZ | a sip trunk being kind of a meaningless term - are you saying you're trying to get a softphone connected to Asterisk? |
| 05:19.33 | ryan_turner | My understanding is that what I use to place and receive calls with numbers... a rented service associated that has a phone # associated with it. |
| 05:19.46 | ryan_turner | No, not to an asterisk install, but instead directly from the softphone to the trunk |
| 05:20.31 | ryan_turner | I setup asterisk on a VPS today and found that remotely hosting it was a pain. I'm waiting for a raspberry pi to arrive tomorrow and plan to install the raspberry asterisk distro on it. |
| 05:21.23 | ryan_turner | But I have a sip user, pass, server, and associated phone number that should be able to place/receive calls, so I was thinking that I could at least play with that by setting up a soft client connecting directly to it. |
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| 05:29.00 | ChannelZ | Well off hand I have no idea, you'd have to take it up with your provider |
| 05:30.18 | ChannelZ | IE it's not really an Asterisk question you're asking per se |
| 05:31.18 | ryan_turner | Very true :) |
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| 05:32.43 | ryan_turner | Ok, lets go a different way then; I plan to install Asterisk and will have a few phones on different extensions. I'd like to connect that to "regular phones" (That seems to be DID? as the right term?) I've been looking around and it seems like around $20 is the going rate, but I see PhonePower (BroadVoice) is selling their cheapest "SIP Trunking" service for $55/month. |
| 05:33.56 | ryan_turner | Im really interested in just getting 1 local number with the ability to place 1 call at a time; doing all domestic calling. I've looked through a look of services that dont seem to be quite what I need. |
| 05:34.06 | ChannelZ | a DID is just a phone number |
| 05:35.02 | ryan_turner | Ok, so what Im trying to find is SIP Trunking tied to an interface with a DID for inbound and outbound calls? |
| 05:36.41 | ChannelZ | There are tons of VoIP providers with service for as low as $1.50/mo plus per-minute charges, or say $15/mo unlimited inbound but per-minute outbound |
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| 05:37.06 | ryan_turner | Ok, can you please share with me one that does just that? I dont know the right terms so I keep getting lost in things :( |
| 05:37.13 | ChannelZ | You don't need "SIP Trunking" unless you want multiple simultaneous channels. |
| 05:37.20 | ChannelZ | ~itsplist-us |
| 05:37.20 | 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 |
| 05:37.29 | ryan_turner | Yeah, I went through all of those |
| 05:37.51 | ChannelZ | like Vitelity calls it a "virtual PRI" and charge per channel, but the basic service I think they let you do up to 2 |
| 05:38.23 | ryan_turner | Ok, and how do I "tie that in" to Asterisk? |
| 05:38.38 | ChannelZ | you just configure a peer for it |
| 05:38.41 | ChannelZ | in sip.conf |
| 05:39.30 | ryan_turner | Ok, so "Vitelity Link" http://vitelity.net/vitelity-link/ is what I could use for instance? |
| 05:40.02 | ChannelZ | sure, that's what I use at home. |
| 05:40.10 | ChannelZ | I don't use the phone a lot so I just pay per minute |
| 05:40.24 | ryan_turner | Ok, cool. Thank you for helping me. |
| 05:41.00 | ChannelZ | the $7.95/mo is unlimited inbound and then you still pay per in for outbound |
| 05:41.32 | ChannelZ | but I don't even use enough to make the $8 worth it :) |
| 05:42.08 | ryan_turner | I've found this entire process very frustrating. Im just a college student who works at home full time. From previous jobs I have a Polycom IP 321 and a linksys spa 941. I need some sort of phone but it needs to be portable with me as I move from university back to home. I dont get good cell reception in either place and dont like talking on a cell phone or giving out my phone number for that. I've found Skype to be a bit troublesome to |
| 05:42.08 | ryan_turner | o. |
| 05:43.17 | ryan_turner | I figure since I already have the hardware I might as well get a 1-man PBX system up and running and have found figuring out just how to do that very frustrating. Not particularly difficult but just a bit confusing. I really appreciate your help even pushing me in the right direction here. |
| 05:43.27 | ChannelZ | Well it all depends on you having decent latency on whatever network you're on, but you can connect the SPA941 directly to Vitelity without having to run Asterisk at all |
| 05:43.45 | ChannelZ | or many other providers for that matter. Dunno why you're having trouble with whatever you have right now though |
| 05:44.45 | ChannelZ | running Asterisk yourself for this purposes lets you do some potentially fun things, but it also adds a layer of issues particularly where firewalls are involved |
| 05:45.03 | ryan_turner | Well, considering that i do have two phones, and a spare raspberry pi to throw the software on, I'd like to do it. My roommates and I at college enjoy gaming but find that moving our setups into one room every few days is a PITA. Figure I'll check out the push-to-talk :) |
| 05:45.43 | ryan_turner | Im not terribly concerned about firewalls, I expect to just run everything behind a tomato router that is a DMZ on the AT&T service. |
| 05:46.34 | ryan_turner | Im comfortable in iptables and playing with the networking side, though I haven't done any work with SIP stuff so there could be a few "gotchas" that I'm not accustomed to considering. |
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| 05:49.00 | ChannelZ | well I suggest going through the Asterisk book |
| 05:49.52 | ChannelZ | I'm trying to finish writing my little Asterisk Crash Course/Primer as a bootstrap for the uninitiated but probably won't get that up until the end of the week at the rate I'm going |
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| 05:57.47 | ryan_turner | Wow that was easy with Vitelity. |
| 05:57.52 | ryan_turner | Already up and running, just made my test calls. |
| 05:58.20 | ryan_turner | So the same credentials that I used with my softclient are what I'd throw in the configuration as a new peer |
| 05:59.34 | ChannelZ | in asterisk you mean? more or less yes |
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| 06:00.37 | ChannelZ | you generally make 2 peers, one for incoming, one for outgoing |
| 06:01.20 | ChannelZ | then in your case you'd also need a register => line to register with them if you have a dynamic ip |
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| 09:30.48 | davlefou | Hi, how i can test the jitter? |
| 09:31.46 | davlefou | Since servals day, i have chopper sound. |
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| 10:01.32 | Tuju | any idea why cisco 7975 looses registration and doesn't aquire it again? |
| 10:01.50 | Tuju | reboot helps but i'd like it to keep trying itself. |
| 10:02.42 | Tuju | it has that Java software inside. I had my doubts about it and unfortunately those feelings have realized more or less. |
| 10:06.07 | Tuju | is there a such thing as a good SIP client for mac and WP8 ? |
| 10:08.37 | davlefou | wp8? |
| 10:09.02 | davlefou | under linux, sflphone. |
| 10:28.55 | Tuju | sflphone, hmmm |
| 10:29.04 | Tuju | davlefou: wp8 is windows phone |
| 10:29.09 | Tuju | some friends have it |
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| 10:53.56 | danfromuk | Hi, how can I view/calculate the average hold time of a queue? |
| 10:54.46 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 10:55.19 | kaldemar | danfromuk: core show function QUEUE_VARIABLES |
| 10:57.48 | *** join/#asterisk firtina (~kamanato@5.46.3.164) |
| 10:58.00 | firtina | hi all |
| 10:58.46 | danfromuk | kaldemar: thanks. can that be retrieved using a CLI command, or only in the dialplan? |
| 10:58.50 | firtina | i have a problem, i wonder if someone could help me? |
| 10:59.26 | firtina | i want to learn device status of a phone linked asterisk over Cisco Call Manager but i can't |
| 10:59.29 | firtina | what should i do? |
| 11:01.44 | firtina | any idea? |
| 11:06.41 | firtina | slaps ChanServ around a bit with a large trout |
| 11:13.54 | kaldemar | danfromuk: it's a dialplan function. |
| 11:20.54 | firtina | kaldemar could you help me? |
| 11:25.25 | danfromuk | Is it possible to transfer a Caller to a specific extension when the called party hangs up? |
| 11:25.49 | danfromuk | I can see the options for the reverse. |
| 11:27.17 | kaldemar | danfromuk: see g option in Dial. |
| 11:27.31 | danfromuk | Got it. Thanks |
| 11:27.58 | danfromuk | And if the dialling application is Queue, I can see that the option is c |
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| 11:37.41 | firtina | how to learn device status of a phone linked asterisk over Cisco Call Manager? any idea? |
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| 12:40.00 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 12:40.05 | Rico29 | hi all |
| 12:40.19 | Rico29 | is there a way to debug DTMF sent by users in voicemail application ? |
| 12:41.08 | [TK]D-Fender | Thre isn't anything special about DTMF there vs anywhere else |
| 12:41.13 | [TK]D-Fender | So what is the issue? |
| 12:43.13 | Rico29 | old messages reappearing |
| 12:43.43 | Rico29 | but it's a bit hard to debug |
| 12:43.54 | Rico29 | and to know "how and why" it reappeared |
| 12:46.14 | [TK]D-Fender | I don't see why this has anything to do with DTMF... |
| 12:46.51 | Rico29 | if the user press two time "7" : delete > undelete |
| 12:47.17 | Rico29 | [TK]D-Fender> is there a way to make voicemail app more verbose ? |
| 12:48.10 | [TK]D-Fender | it isn't Voicemail |
| 12:48.12 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 12:48.25 | [TK]D-Fender | If you SEE it undelete its because it got the request to. |
| 12:48.29 | [TK]D-Fender | CORE debug shows DTMF |
| 12:54.57 | Rico29 | [TK]D-Fender> can't see the DTMF sent to voicemail app |
| 12:55.03 | Rico29 | with core debug 10 |
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| 13:13.15 | Katty | drags in |
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| 13:13.50 | jmetro | helps Katty with her luggage |
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| 13:17.32 | Katty | hugs jmetro |
| 13:17.33 | *** join/#asterisk serafie1 (~erin@nat/digium/x-imtnfohnretnbkse) |
| 13:18.09 | jmetro | :> |
| 13:19.34 | Katty | what's the word |
| 13:21.13 | Cuzner | bird, duh |
| 13:22.18 | jmetro | I think the word today should be Chocolate personally. |
| 13:29.08 | Katty | chocolate is the word every day |
| 13:31.00 | jmetro | yesterday it was Blueberry. |
| 13:34.28 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
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| 13:39.04 | Vann | Would anyone happen to know a possible cause as to why my SIP channels do not terminate after a hangup? |
| 13:40.29 | Vann | Been googlin' for a while and can't seem to find any conclusive causes. |
| 13:42.30 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 13:44.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:49.38 | jmetro | Vann, depends on the channel. i had a pretty bad case of it with my voicemails until i updated asterisk |
| 13:50.39 | Vann | Hmm. Running a "fresh" source install of the latest Asterisk (11.4.0). |
| 13:50.47 | Vann | I wonder if an SVN build might help. |
| 13:51.44 | jmetro | asterisk 11 r378219 is mine, it might have been the later one giving me issues actually |
| 13:51.52 | Vann | Basically I call an internal extension to test the call. After I hangup, the channel stays open indefinitely. |
| 13:51.55 | Vann | Ahh. |
| 13:52.31 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 13:54.01 | Vann | So far what I've searched seems to indicate a "bug" with asterisk. Not so sure that's really the case. |
| 13:55.03 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:55.30 | jmetro | vann add a same -> hangup to the end of everything |
| 13:56.41 | Vann | I'll try that. Thanks. |
| 13:58.43 | jmetro | for my voicemails i had to have a local channel added and that alleviated some |
| 13:59.23 | jmetro | did a dial(local/${EXTEN}@AutodestructsLol) where autodestructslol was just exten => .X,Voicemail(${EXTEN}) |
| 14:01.54 | *** join/#asterisk DennisG (~DennisG@198-81-ftth.on.nl) |
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| 14:07.58 | *** join/#asterisk ttpearso (~ttpearso@gw.teamgleim.com) |
| 14:09.49 | ttpearso | I've been having some issues with Asterisk + CEL, it seems when I have CEL configuration active, asterisk will segfault upon starting, using 11.4.0, anyone else run across similiar issue? |
| 14:11.32 | Vann | Hmm. That's odd. Source install or binary? |
| 14:11.58 | ttpearso | source |
| 14:12.18 | ttpearso | Recompiled after checking my menuconfig a few times, but no change |
| 14:12.29 | ttpearso | mostly just for sanity check |
| 14:12.49 | Vann | Your CDR/CEL local? ODBC? |
| 14:13.05 | ttpearso | ODBC to local MySQL |
| 14:14.18 | Vann | I take it you can connect fine via isql -v YourDSN ? |
| 14:15.35 | ttpearso | correct |
| 14:15.53 | ttpearso | fyi: FreePBX running on top, but don't think that should matter |
| 14:16.05 | ttpearso | I've had to configure all the CEL manually anyway |
| 14:16.52 | Vann | Do you have "CDR Reports" module installed in FreePBX? |
| 14:17.03 | Vann | That will attempt to autoconfigure CEL which may cause some issues |
| 14:17.18 | Vann | I know it did for me as my CDR is located remotely. |
| 14:17.43 | ttpearso | Possibly installed, let me check |
| 14:18.59 | ttpearso | It is, I'll remove and test |
| 14:19.21 | Vann | Cool. You may have to double-check your cel config after uninstall. |
| 14:20.08 | ttpearso | Yes, still failing to load, poking through config real quick |
| 14:20.56 | Vann | How do you disable CEL to have asterisk load. From FreePBX advanced settings? |
| 14:25.38 | jmetro | that might be a #freepbx question |
| 14:25.40 | ttpearso | I did previously, now was just working with configs directly |
| 14:26.13 | ttpearso | It's in the "advanced settings" iirc |
| 14:26.41 | Vann | jmetro, sorry for the confusion lol. I was asking if that's the method he was using. There's a "Enable CEL Reporting" setting under advanced settings. |
| 14:27.22 | Vann | So if you disable CEL reporting in freepbx asterisk loads up fine? |
| 14:27.43 | ttpearso | Yes |
| 14:27.50 | Vann | hmm |
| 14:27.52 | ttpearso | Loaded just now after disabling it all |
| 14:29.10 | Vann | What happens if you enable it via 'enable=yes' manually in the cel.conf? |
| 14:29.18 | Vann | Same thing?] |
| 14:29.57 | Katty | whatever happened to eppigy? |
| 14:30.24 | Katty | infobot: seen eppigy |
| 14:30.29 | infobot | eppigy <~Dave@snugglenets.com> was last seen on IRC in channel #asterisk, 592d 20h 33m 49s ago, saying: 'oh you fancy huh'. |
| 14:30.50 | Katty | oh. well. that's been awhile. |
| 14:31.53 | ttpearso | Vann: ... it appears it's loading, but not working |
| 14:32.28 | ttpearso | modules are loaded though |
| 14:32.34 | ttpearso | looking at log |
| 14:32.35 | Vann | Asterisk is not working? |
| 14:32.43 | ttpearso | Asterisk is loaded |
| 14:33.16 | jmetro | 592 days |
| 14:33.20 | jmetro | thats pretty impressive. |
| 14:33.46 | Vann | Yeah that's definitely been a while. |
| 14:34.35 | Vann | ttpearso, do you have 'cel_custom.so cel_manager.so cel_odbc.so' under /usr/lib/asterisk/modules/ ? |
| 14:34.40 | ttpearso | Vann: There we go, now back to normal - had un-done 1 too many times in vim, now that the odbc is specified CORRECTLY, it's crashing again |
| 14:34.50 | Vann | ah |
| 14:35.18 | Vann | So it does seem to be possibly ODBC related.. |
| 14:35.41 | ttpearso | Yea, as soon as the connection is correct, it starts bailing |
| 14:35.48 | ttpearso | Yes to all those files btw |
| 14:36.17 | ttpearso | see if I can find interesting log snippet |
| 14:36.30 | ttpearso | Last line before crashing: [2013-07-02 10:34:53] VERBOSE[2251] config.c: == Parsing '/etc/asterisk/cel_odbc.conf': Found |
| 14:36.36 | Vann | Can you post your odbc.ini omitting any sensitive information? |
| 14:36.40 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:37.52 | ttpearso | Vann: sure one moment - test system anyway, nothing sensitive |
| 14:38.37 | Vann | Cool. While you're at it, if you can link to cel_odbc.conf and res_odbc.conf that would be helpful too. |
| 14:38.38 | ttpearso | http://pastebin.com/jR6NtJcG |
| 14:38.52 | ttpearso | Sure, I'll tack those on |
| 14:41.12 | ttpearso | since I'm FreePBX, res_odbc.conf is just includes, I'll paste res_odbc_custom |
| 14:41.12 | Vann | It may have to be res_odbc_*.conf and cel_odbc_*.conf due to freepbx |
| 14:41.17 | ttpearso | :-) |
| 14:41.21 | Vann | :D |
| 14:43.50 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 14:43.54 | ttpearso | http://pastebin.com/nHsDtnmN |
| 14:44.15 | ttpearso | I'm guessing FreePBX had this configed already: /etc/asterisk/cel_odbc.conf |
| 14:44.31 | Vann | yeah |
| 14:44.38 | Vann | that's what "CDR Reports" touches |
| 14:44.43 | Vann | I ran into the same issue |
| 14:44.55 | Vann | I configured it manually, and then FreePBX overwrote it... lol |
| 14:49.01 | Vann | ttpearso, what happens if you run isql -v MySQL-cel |
| 14:49.27 | ttpearso | Provided, I give it credentials, works fine |
| 14:49.30 | igcewieling | Vann: That is what happens when you run FreePBX |
| 14:49.37 | ttpearso | .... Connected! ..... |
| 14:50.13 | ttpearso | Woah. Very interesting |
| 14:50.20 | ttpearso | Doing "help cel" |
| 14:50.24 | ttpearso | SQLRowCount returns 25 |
| 14:50.24 | ttpearso | 25 rows fetched |
| 14:50.24 | ttpearso | Segmentation fault |
| 14:51.33 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 14:51.49 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:52.08 | Vann | hmmm |
| 14:53.17 | ttpearso | Debian Wheezy fyi |
| 14:53.24 | Vann | ttpearso, if you run isql -v MySQL-cel it asks you for username and pass? |
| 14:53.46 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 14:53.50 | *** part/#asterisk fischli (~fischli@data.fischer-ing.de) |
| 14:53.56 | ttpearso | No, I provide them on the command, eg: isql -v MySQL-cel <username> <pass> |
| 14:54.02 | Vann | what happens if you don't? |
| 14:54.03 | ttpearso | I'm logged into the box as root |
| 14:54.16 | ttpearso | So it tried to login as root, and fails |
| 14:54.20 | Vann | if you run the command omitting user and pass |
| 14:54.35 | Vann | Access denied? |
| 14:54.37 | ttpearso | isql -v MySQL-cel |
| 14:54.37 | ttpearso | [S1000][unixODBC][MySQL][ODBC 5.1 Driver]Access denied for user 'root'@'localhost' (using password: YES) |
| 14:54.40 | ttpearso | [ISQL]ERROR: Could not SQLConnect |
| 14:54.45 | Vann | okay |
| 14:54.50 | Vann | good |
| 14:54.51 | Vann | lol |
| 14:54.53 | ttpearso | :-) |
| 14:55.04 | Vann | Not saying this is your issue, but there's an error in your odbc.ini |
| 14:55.17 | Vann | change 'UserName' to 'User' |
| 14:55.28 | Vann | and run isql -v MySQL-cel without username and pass |
| 14:55.36 | Vann | just 'isql -v MySQL-cel' |
| 14:55.59 | ttpearso | works now |
| 14:56.11 | Vann | awesome |
| 14:56.42 | Vann | I don't see how that would give you a segfault, but it's worth trying to see if asterisk loads. |
| 14:56.51 | ttpearso | It does not |
| 14:56.56 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 14:56.58 | Vann | hmm |
| 14:57.08 | ttpearso | fyi: Asterisk ended with exit status 139 |
| 14:57.11 | Vann | what's the last line in the log, same? |
| 14:57.41 | ttpearso | Yea, [2013-07-02 10:57:01] VERBOSE[3804] config.c: == Parsing '/etc/asterisk/cel_odbc.conf': Found |
| 14:58.46 | Vann | do the permissions look for all the files under /etc/asterisk/ ? |
| 14:58.53 | Vann | ownership permissions |
| 14:58.56 | Vann | usually asterisk |
| 15:00.01 | ttpearso | Yea, everything |
| 15:00.05 | ttpearso | asterisk:asterisk |
| 15:01.14 | ttpearso | fyi: same for FreePBX files |
| 15:02.36 | ttpearso | Also just confirmed that installed ODBC was latest rev in wheezy |
| 15:02.43 | Vann | unix-odbc? |
| 15:03.05 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 15:03.13 | Vann | Increasing the log verbosity may help |
| 15:03.16 | Vann | maybe to 5 or something |
| 15:03.28 | ttpearso | Know where that is set offhand? |
| 15:05.38 | Vann | Try this. Kill asterisk - amportal kill |
| 15:05.47 | Vann | then, asterisk -rvvvvvvvvvvvvvvvvvvvvvv |
| 15:05.53 | Vann | curious to see if it tells you anything |
| 15:06.19 | ttpearso | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 15:06.29 | Vann | er, remove the "r" |
| 15:07.01 | ttpearso | Heh, was about to say that |
| 15:07.07 | ttpearso | <PROTECTED> |
| 15:07.07 | Vann | lol |
| 15:07.07 | ttpearso | Segmentation fault |
| 15:07.45 | Vann | you cel_odbc.conf looks exactly as you posted previously? |
| 15:08.37 | ttpearso | Yes, it of course has the "DO NOT EDIT ...." FreePBX disclaimer as the top |
| 15:08.44 | *** part/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:08.48 | Vann | how's cel.conf look? |
| 15:08.55 | *** join/#asterisk stefan0 (~stefano@189.26.70.53.dynamic.adsl.gvt.net.br) |
| 15:08.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:09.10 | ttpearso | Also just confirmed that installed ODBC was latest rev in wheezy[general] |
| 15:09.10 | ttpearso | #include cel_general_additional.conf |
| 15:09.10 | ttpearso | #include cel_general_custom.conf |
| 15:09.10 | ttpearso | #include cel_custom_post.conf |
| 15:09.29 | Vann | er, cel_general_additional.conf |
| 15:09.39 | ttpearso | enable=yes |
| 15:09.39 | ttpearso | apps=all |
| 15:09.39 | ttpearso | events=all |
| 15:09.39 | ttpearso | dateformat=%F %T |
| 15:09.49 | Vann | hmm. everything looks fine |
| 15:09.55 | Vann | crazy =/ |
| 15:10.00 | ttpearso | Indeed :-) |
| 15:10.01 | stefan0 | hi all! is there a way to increase up volume (like dahdi rxgain) inside an IAX2 trunk? |
| 15:10.23 | ttpearso | Why I finally had to pop over here and see if I was just crazy |
| 15:10.34 | Vann | lol, I know the feeling |
| 15:10.55 | igcewieling | stefan0: not really, but you can try the AGC function (requires some extra libraries to build) |
| 15:11.33 | ttpearso | I've got a meeting here shortly, but I'm going to try compiling 1.8, see if it works |
| 15:12.06 | Vann | quick thing to try ttpearso, if you run isql -v MySQL-cel and connect |
| 15:12.15 | Vann | and run 'show tables' |
| 15:12.19 | Vann | what happens? |
| 15:12.47 | ttpearso | seg fault |
| 15:13.06 | Vann | okay, can you post your /etc/odbcinst.ini please ? |
| 15:13.24 | ttpearso | [MySQL] |
| 15:13.24 | ttpearso | Description = ODBC for MySQL |
| 15:13.24 | ttpearso | Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so |
| 15:13.24 | ttpearso | Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so |
| 15:13.24 | ttpearso | FileUsage = 1 |
| 15:13.34 | igcewieling | ~pb |
| 15:13.34 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:13.45 | stefan0 | igcewieling, my scenario is that I have 1 IAX2 trunks between 2 sites and one of these have and E1 link (digium card). At the site that have the digium card everything runs OK, at the site that uses IAX2 the rx volume is too low. any idea? |
| 15:13.47 | ttpearso | Sorry |
| 15:14.08 | igcewieling | stefan0: increase the volume on the dahdi ports |
| 15:14.20 | stefan0 | rxgain=20.0 :o |
| 15:14.29 | stefan0 | @ chan_dahdi.conf |
| 15:14.31 | ttpearso | Vann: fyi files do exist |
| 15:14.41 | Vann | haha, was just about to ask |
| 15:14.43 | igcewieling | pastebin your chan_dahdi.conf |
| 15:15.15 | ttpearso | Might be worth trying i386 version, but that's a hassle |
| 15:15.35 | stefan0 | http://pastebin.ca/2414842 |
| 15:15.45 | stefan0 | igcewieling, http://pastebin.ca/2414842 |
| 15:16.40 | Vann | Yeah I am pretty sure it is an ODBC and not necessarily an asterisk one. |
| 15:16.45 | Vann | issue* |
| 15:16.55 | ttpearso | I'd have to agree |
| 15:16.58 | igcewieling | stefan0: I don't see anything really stupid in the file. |
| 15:17.06 | Vann | You using unixODBC package |
| 15:17.08 | Vann | ? |
| 15:17.11 | igcewieling | now, where are you applying that, the E1 or the Analog ports on the 2nd server |
| 15:17.42 | igcewieling | sorry, I misunderstood, I thought you had dahdi cards at both location. |
| 15:18.07 | igcewieling | stefan0: try removing the .0 and try rxgain=4 and txgain=4 |
| 15:18.20 | igcewieling | you seldom want to go above a gain of 8 |
| 15:18.37 | stefan0 | at the site that has the E1 link. Using rxgain=20 that works fine at 2nd site - but the calls at site #1 are impossible to (very loud) |
| 15:19.08 | stefan0 | s/impossible/'impossible to use' |
| 15:19.12 | igcewieling | stefan0: this is all digital, asterisk doesn't modify the audio by default. |
| 15:19.34 | igcewieling | maybe your SIP endpoints need to have their gain increased too? |
| 15:19.52 | stefan0 | the IAX trunk direct the calls from site #2 to use the dahdi at site #1 |
| 15:19.59 | ttpearso | Vann: yea |
| 15:20.04 | ttpearso | Version: 2.2.14p2-5 |
| 15:20.09 | igcewieling | people have IAX phones? |
| 15:20.33 | stefan0 | yeah, I was wondering something like that. I'm using g729 to passthru but my endpoint are Microsoft Lync |
| 15:20.47 | stefan0 | (and Lync doesn't support g729) |
| 15:20.58 | Vann | ;/ |
| 15:21.13 | igcewieling | stefan0: Ah. Too complicated for me to support. passthru almost never works as expected and Microsoft products are made by Satan himself. |
| 15:21.30 | ttpearso | ... guess I'll compile it real quick |
| 15:21.35 | igcewieling | Though I think if lync doesn't support g729 then maybe trying to use g729 passthru isn't such a good idea? |
| 15:21.58 | Vann | ttpearso, that is definetely worth a try. Either that or attempt to use an older binary. |
| 15:22.04 | stefan0 | Dahdi E1 --- Asterisk #1 --- IAX2 ---- Asterisk #2 ---- SIP Trunk ---- MS Lync Server ---- Endpoints |
| 15:22.33 | jmetro | why would something not support g729? 722 and 729 are the only 2 worth using <.< |
| 15:22.35 | igcewieling | WHAT ARE THE ENDPOINTS? |
| 15:22.41 | ttpearso | Vann: Debian is normally lagged behind a few versions, I'll try latest |
| 15:22.48 | Vann | Even better. |
| 15:23.12 | stefan0 | yeah, I use g711 between Asterisk-Lync-Endpoints (MS Lync clients) since they are in LAN no problem at that |
| 15:23.19 | Vann | jmetro, g729 typically has licensing issues associated with it. |
| 15:23.53 | ttpearso | my my how compiling had sped up over the years |
| 15:23.57 | [TK]D-Fender | Those gain parameters are in db, and 20 is insane. |
| 15:24.32 | igcewieling | [TK]D-Fender: you always seem to help the imposible people |
| 15:24.42 | [TK]D-Fender | [11:18]stefan0at the site that has the E1 link. Using rxgain=20 that works fine at 2nd site - but the calls at site #1 are impossible to (very loud) <- don't pump the gain on server #1 then, do it on #2 |
| 15:25.06 | Vann | ttpearso, it almost makes Gentoo a viable option... ;) |
| 15:25.30 | jmetro | debain4lyfe |
| 15:25.34 | ttpearso | Vann: funny, I used Gentoo when Athlon XP were the latest craze |
| 15:25.34 | [TK]D-Fender | igcewieling: This one is far from impossible. And yesterday's guy came back a few hours later saying he "solved" it... with no details and left. |
| 15:25.47 | stefan0 | [TK]D-Fender, OK, that was my idea, so I asked if there's something like dahdi rxgain to use at IAX2 trunks (maybe SIP) :) |
| 15:26.13 | [TK]D-Fender | stefan0: "core show function VOLUME" <---------- |
| 15:26.48 | igcewieling | [TK]D-Fender: he can't even tell us what brand of SIP phone he is using |
| 15:27.04 | Vann | ttpearso, What a trooper. Took me literally a couple days just to compile the basics back then. Although a slow internet connection did not help. |
| 15:27.07 | [TK]D-Fender | igcewieling: I can tell you how little it matters to us :) |
| 15:27.23 | jmetro | I was already on dsl when the athlons came out. DSL WAS SO GOOD |
| 15:27.26 | [TK]D-Fender | igcewieling: I've just offered the solution for this (as sad as it may be) |
| 15:28.09 | stefan0 | [TK]D-Fender great great great :) |
| 15:28.30 | igcewieling | [TK]D-Fender: I should read "core show functions" more often |
| 15:28.40 | ttpearso | I think I still had 1-way cable back then, super fast down, 56K up. Took many a hours to compile X, iirc it was ~1 day to get a gnome environment from ground up |
| 15:28.50 | stefan0 | I don't use SIP phones! My endpoints are the MS Lync connected to asterisk as a trunk |
| 15:29.07 | [TK]D-Fender | igcewieling: Of all people, yes you certainly should.... |
| 15:29.36 | Vann | My experiences exactly ttpearso . It was fun, but not very practical in my opinion. |
| 15:29.47 | igcewieling | [TK]D-Fender: I usually read it every 2 or 3 months |
| 15:30.12 | Vann | Modern day CPUs can compile the entire OS in probably an hour or so. |
| 15:31.08 | ttpearso | Certainly not, especially in business, the previous sysadmins thought it would be fun. We *just* retired one of our last Gentoo boxes |
| 15:31.10 | jmetro | a full hour to compile something? |
| 15:31.16 | jmetro | i feel like that is too high. |
| 15:31.49 | Vann | lol jmetro |
| 15:32.07 | Vann | An hour to compile an entire gentoo box is not bad at all compared to how it used to be. |
| 15:33.00 | Vann | It still has it's advantages in business for those who want to squeeze every ounce of performance they can. |
| 15:33.09 | Vann | its |
| 15:34.03 | igcewieling | Gentoo is just Linux wanting to be BSD |
| 15:34.16 | [TK]D-Fender | http://funroll-loops.info/ <-------- |
| 15:34.17 | ttpearso | I just don't think it's very cost effective. All the testing that has to go on after you recompile, PHP or anything that your business depends on, bleh. Maybe in some cases, but few and far between imho :-) |
| 15:34.33 | Vann | Agreed. |
| 15:34.36 | igcewieling | We have one Gentoo box because it was required for a specific app |
| 15:35.30 | ttpearso | [TK]D-Fender: lol |
| 15:36.06 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 15:36.07 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:38.29 | ttpearso | Vann: well compiled & tested unixODBC, no joy there either, same results |
| 15:38.48 | Vann | =/ |
| 15:38.58 | Vann | here's something |
| 15:39.16 | Vann | try connecting to mysql locally via 'mysql' |
| 15:39.19 | Vann | and run a command |
| 15:39.25 | Vann | curious to see if it crashes |
| 15:39.30 | Vann | it may infact be a mysql issue |
| 15:39.34 | ttpearso | It doesn't but I'll confirm ;-) |
| 15:40.10 | jmetro | you should be able to run "show all dbs when convenient" if it doesnt spit out 30 gold coins your mysql is borked |
| 15:40.46 | ttpearso | Nah, no problems there, can query asteriskcdrdb without issue |
| 15:40.55 | Vann | Interesting. |
| 15:40.57 | ttpearso | show tables, desc, etc. |
| 15:41.31 | Vann | Basic, but the odbc user has all the required privileges for the cdr? |
| 15:42.02 | ttpearso | yup |
| 15:42.16 | ttpearso | I think maybe I should poke libmyodbc |
| 15:42.30 | ttpearso | It provides the libs in odbcinst |
| 15:43.41 | ttpearso | Maybe I can get away with just installing it from Jessie repos :-p |
| 15:43.44 | Vann | My unixODBC version is also 2.2.13 |
| 15:43.51 | Vann | 2.2.14* |
| 15:43.57 | Vann | on CentOS 6 |
| 15:44.04 | ttpearso | Interesting |
| 15:44.09 | ttpearso | Version: 2.2.14p2-5 |
| 15:51.00 | stefan0 | [TK]D-Fender igcewieling, thanks for your help. I used the volume function at dialplan and will request a test to the users after lunch |
| 15:53.15 | igcewieling | stefan0: remember you can use VOLUME on a g729 call unless you have g729 license |
| 15:53.51 | igcewieling | As asterisk will have to convert the audio to sln, increase the volume, then convert it back to g729 |
| 15:56.53 | [TK]D-Fender | igcewieling: one of his server's has it... and it barely matters which.. |
| 15:57.09 | [TK]D-Fender | igcewieling: He's coming from DAHDI on #1 |
| 15:57.15 | [TK]D-Fender | igcewieling: So either #1 has it or #2 |
| 15:57.49 | [TK]D-Fender | igcewieling: And the lewst point is doing it before going out the peer to #2. Remember that #1 is the one that it is too high for... |
| 15:57.53 | [TK]D-Fender | lowest* |
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| 16:03.10 | stefan0 | hm.. I raised up the volume at #2 box.. like |
| 16:03.25 | stefan0 | exten => _0XX[2-4].,1,Set(VOLUME(RX)=12) \ exten => _0XX[2-4].,2,Set(IAX_CODEC=g729) \ exten => _0XX[2-4].,3,Dial(IAX2/AMTI/${EXTEN},60,TtR) |
| 16:07.44 | stefan0 | well, sounds like that worked. After that let's check the users =] ty u all |
| 16:09.21 | Katty | pops in over lunch |
| 16:12.18 | *** join/#asterisk vedic (~V@183.82.84.84) |
| 16:13.04 | vedic | What is the difference between .wav and .sln formats? |
| 16:13.45 | Katty | hi vedic |
| 16:14.48 | jmetro | https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats |
| 16:16.01 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ecevwmvtfwrlukhl) |
| 16:16.06 | vedic | Katty: hi |
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| 16:18.33 | ttpearso | Vann: think I found my problem |
| 16:18.41 | ttpearso | 0x00007ffff5c14e13 in list_delete () from /usr/lib/libmysqlclient.so.18 |
| 16:19.16 | ttpearso | I use percona, which is slightly different than usual MySQL libs |
| 16:19.40 | igcewieling | that might have something to do with it |
| 16:20.22 | ttpearso | Never run up against anything that had an issue before, but that's the output from gdb after running a command in iSQL |
| 16:20.45 | vedic | jmetro: I am still unclear. What I get from this is sln is 16bit 8Khz signed linear format same like pcm. What is then .wav ? Is the windows wav same as sln? I am recording speech from phone and doing a bit of speech processing activity |
| 16:21.46 | jmetro | wav i beleive can be many many things |
| 16:21.59 | jmetro | i actually convert vox to wav and save it as a sln16 |
| 16:22.16 | igcewieling | vedic: you mean 16khz for sln |
| 16:22.30 | jmetro | load vox in audacity, save as 16hz signed wav, rename it to sln16 in linux to have it play |
| 16:23.25 | igcewieling | sorry, that is sln16. Been doing too much sln16 lately. |
| 16:23.44 | igcewieling | vedic: I suspect the difference wav has an RIFF header and .sln does not |
| 16:23.54 | vedic | igcewieling: I mean 8Khz as I am using PSTN line. As per https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats it sln should be 8Khz I think |
| 16:24.05 | vedic | igcewieling: I see |
| 16:24.25 | igcewieling | BTW, would anyone find a website which lets you upload an audio file and then hands you back the audio file converted into your requested Asterisk format? |
| 16:24.28 | vedic | igcewieling: So if I save a recording in .wav, it will contain the header? |
| 16:24.36 | igcewieling | sorry, would anyone find that useful. |
| 16:24.49 | igcewieling | vedic: yes, a Microsoft RIFF header |
| 16:24.52 | jmetro | igcewieling: please god yes. |
| 16:25.24 | igcewieling | jmetro: If it works I might add g729 transcoding -- we have a couple of old licenses laying around |
| 16:25.39 | jmetro | igcewieling: i know nothing about sound garbage and it angers me when im trying to "CONVERT Mp3 TO ASTERISK" and i get things saying "just use X-khz signed PCM monkey jabbers" |
| 16:26.05 | igcewieling | jmetro: *nod* In theory it should not be too hard to do. |
| 16:26.05 | vedic | igcewieling: yes that site will be helpful for people not familiar with * or sox etc. But in that case why they would need it for asterisk format (seems like the user has technical knowledge) |
| 16:26.24 | vedic | igcewieling: But sure for licensed codecs |
| 16:26.47 | vedic | which people won't like to buy for small work or work in chunks |
| 16:26.54 | igcewieling | vedic: My original idea for this is to help out our support people convert files customers send us for uploading to the customer's FreePBX box. |
| 16:27.02 | jmetro | im great at everything except finding out codec khz signatures headers..whatever the crap is. i just want sound @.@ |
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| 16:29.58 | igcewieling | jmetro: I'm thinking in my script convert everything to .sln16, then hand the file off to "asterisk -rx convert oldfile netfile" so the output file will always work with Asterisk |
| 16:30.20 | jmetro | probably yeah |
| 16:30.38 | jmetro | though personally -rx scripts never work for me |
| 16:34.53 | vlad_starkov | Question: Anyone know is it possible to playback multiple announce sound files with A(x&y&z) Dial option? |
| 16:36.41 | jmetro | vlad are you announcing to the caller or the callee |
| 16:36.57 | jmetro | callee - use the macro option and you can even run code to dynamically select what files you want to play. |
| 16:44.06 | vlad_starkov | jmetro: oh, you mean that it is better to use Macro or Sub for playback than using tiny A option? |
| 16:45.09 | *** join/#asterisk geeksteve (~geeksteve@195.110.168.123) |
| 16:46.41 | jmetro | vlad_starkov: if you only want to play one file, you can use A, otherwise, try the macro/sub. Its what i use and it works great, mainly because i only had to build 1 routine to announce voice titles for all 100 users. |
| 16:47.03 | vlad_starkov | jmetro: very nice, thanks! |
| 16:47.04 | jmetro | "Incoming call for - " ${EXTEN}-Title.sln16 |
| 16:47.42 | vlad_starkov | jmetro: I need it to announce the Department that caller have chosen |
| 16:48.13 | [TK]D-Fender | Then go do it... |
| 16:48.14 | jmetro | how many departments? |
| 16:49.24 | vlad_starkov | 3 |
| 16:50.57 | jmetro | up to you how you want to code it, just make sure you always comment and format your code properly. |
| 16:52.30 | vlad_starkov | jmetro: sure |
| 16:52.36 | vlad_starkov | jmetro: thanks for advice |
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| 17:00.06 | jmetro | vlad_starkov: np |
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| 17:13.03 | ttpearso | Vann: appreciate all the help - finally got it working. libmysqlclient was the culprit, I downgraded one minor version and everything works as expected now |
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| 17:27.40 | Katty | i like pie. |
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| 17:38.06 | |PiP| | which g729 codec should i use on an intel atom cpu? |
| 17:38.34 | davlefou | |PiP|, ask the bench mark |
| 17:38.49 | dr0ck | benchg729 breh |
| 17:40.55 | |PiP| | i bought a single g729 license to test with on my freepbx box. do i need one for each trunk? how does licensing work exactly? |
| 17:41.06 | |PiP| | one for each extension? |
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| 17:43.27 | mnewton | Hey guys I want to intergrate speech to text with asterisk with sphynx - what should I know. Most of the articles I see on google are pretty old 2008ish |
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| 17:45.36 | igcewieling | |PiP|: you need one license per simultaneous call |
| 17:45.47 | igcewieling | you sometime might need a 2nd license on a temp basis |
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| 17:52.00 | protocoldoug | mnewton: not sure of your intended implementation, but, if it's not "for work", a solid toy, without any promises is to use google speech api, here's an implementation of it: http://zaf.github.io/asterisk-speech-recog/ |
| 17:53.26 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 17:53.33 | protocoldoug | (otherwise at werk we use lumenvox, but, you gotta pay mucho dinero) |
| 18:03.20 | *** join/#asterisk windback (~windback@190.123.122.18) |
| 18:04.29 | Vann | ttpearso, awesome news! |
| 18:04.43 | Katty | awesome news? dinner is ready?! |
| 18:04.47 | Katty | gets her fork and napkin |
| 18:05.09 | windback | Is there any way to avoid asterisk playing "please leave your message after the tone.." on voicemail. I dont want to use the "s" option in VoiceMail aplication since it doesnt play anything and I want the user can record his own greeting message |
| 18:05.12 | Vann | I wish.. =P |
| 18:05.12 | windback | ? |
| 18:06.40 | Vann | windback, I am sure there is a way. Not quite familiar with it myself. |
| 18:07.17 | jmetro | windback thats what s does.. |
| 18:07.26 | jmetro | it gets rid of the instructions and the user can use their own greeting |
| 18:08.10 | Katty | sshes into jmetro's box and DELETES THE FILE |
| 18:08.31 | Katty | puts it back |
| 18:08.34 | jmetro | ._. |
| 18:08.53 | Katty | fine fine. |
| 18:08.57 | jmetro | Windback use Voicemail(Exten@Context,us) |
| 18:08.59 | Katty | confirms permissions on file |
| 18:09.12 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 18:09.12 | jmetro | they record the (U)navailable message and (S)uppresses the instructions |
| 18:09.30 | Katty | i'll supress your instructions in a minute. |
| 18:09.34 | Katty | right after i pester Qwell |
| 18:09.36 | jmetro | :X |
| 18:09.41 | Qwell | what?! |
| 18:09.41 | Katty | Qwell: HI. i've not talked to you ALL DAY |
| 18:09.45 | Qwell | IKR? |
| 18:09.48 | dr0ck | replace vm-leavemsg with tt-monkeys |
| 18:09.55 | Katty | Qwell: what's up with that? i'm totally slacking |
| 18:10.01 | Katty | irritates Qwell |
| 18:10.24 | Katty | dr0ck: i like the way you think |
| 18:10.48 | [TK]D-Fender | [14:09]jmetrothey record the (U)navailable message and (S)uppresses the instructions <- It will PLAY the "unavailable" message ... and skip instructions |
| 18:10.55 | jmetro | Indeed. |
| 18:11.04 | jmetro | Thats what i said <.< |
| 18:11.20 | Katty | jmetro: he could be playing 1up again |
| 18:11.26 | Katty | jmetro: i hear he likes to do that. |
| 18:11.27 | [TK]D-Fender | jmetro: Funny... I see you saying it records the unavailable message |
| 18:11.44 | jmetro | they (the user) records the Unavailable and it plays with the switch u |
| 18:11.52 | [TK]D-Fender | nope |
| 18:12.04 | Katty | jmetro: i think he needs coffee. |
| 18:12.08 | Katty | jmetro: or maybe sex. |
| 18:12.11 | Katty | jmetro: maybe both |
| 18:12.18 | jmetro | I shall supply neither :< |
| 18:12.33 | [TK]D-Fender | When I leave you a VM... there is no distinction for the reason I leave it. that is the prompt the caller gets as to the reason they hit VM. |
| 18:12.35 | Katty | [TK]D-Fender: did you see that? jmetro just cut you off. |
| 18:12.55 | jmetro | He's never gotten coffee, but i'm definitely cutting off the sex. |
| 18:13.14 | Katty | jmetro: it's ok. i think he likes geeetars better than girls anyway. |
| 18:13.14 | *** join/#asterisk peetaur2 (~peter@x2f07d71.dyn.telefonica.de) |
| 18:13.27 | [TK]D-Fender | Katty: I'm at risk of losing my "Official Caffiene-Based Life-Form" status.. I've switched to beer at night now... |
| 18:13.43 | Katty | [TK]D-Fender: THE HORROR |
| 18:13.51 | [TK]D-Fender | EYE NOES! |
| 18:13.57 | Katty | [TK]D-Fender: are you going to get a pudge, too? |
| 18:14.10 | Katty | [TK]D-Fender: you'd have an adorable pudge, i bet. |
| 18:14.18 | [TK]D-Fender | ? |
| 18:14.19 | jmetro | Katty: I guess thats why hes a D-Fender..not a D-Gibson |
| 18:14.32 | [TK]D-Fender | My Nick actually has nothing to do with music... |
| 18:14.39 | Katty | jmetro: *hee* |
| 18:14.43 | [TK]D-Fender | And I don't play Fender instruments :) |
| 18:14.46 | jmetro | Katty: Though really he's more of an O-Fender. |
| 18:15.02 | Katty | jmetro: he will always be fender bender to me. |
| 18:15.18 | windback | jmetro, thanks |
| 18:15.22 | [TK]D-Fender | Well.... technicallicall FMC owns Jackson... but I don't count that as being by proxy :p |
| 18:15.31 | [TK]D-Fender | technically* |
| 18:15.33 | coppice | I suspect his real name is Gibson |
| 18:15.42 | windback | jmetro, it works perfectly. I dont know you can mix u and s option |
| 18:15.53 | Katty | maybe his middle name |
| 18:15.53 | [TK]D-Fender | coppice: ... in the foyer with a candlestick! |
| 18:16.25 | Katty | [TK]D-Fender: what does that o stand for. |
| 18:16.35 | jmetro | windback: From what i know, you can mix any options ever as long as their description doesnt say "omits [X] option] or some such |
| 18:17.12 | Katty | [TK]D-Fender: OCTAVIUS?! |
| 18:17.21 | Katty | [TK]D-Fender: that'd be a pretty cool middle name. |
| 18:17.38 | windback | jmetro, do you know a way to avoid asterisk playing the person at extension XXXX and put another default unavail greetings? |
| 18:17.57 | [TK]D-Fender | Katty: Yeah... I should continue my research into grafting robotic exo-skeletons now.... |
| 18:18.26 | [TK]D-Fender | windback: When you supply a file to play back, that's what it does |
| 18:18.38 | jmetro | windback: i beleive that was what dr0ck suggested earlier, replacing the sound file. |
| 18:18.39 | Katty | [TK]D-Fender: don't you dare build skynet. i will hunt you down. |
| 18:18.41 | [TK]D-Fender | windback: No file = system generated prompts |
| 18:18.53 | [TK]D-Fender | Katty: no, IT will hunt us down :p |
| 18:19.02 | jmetro | windback: or what d-fender said |
| 18:19.03 | Katty | [TK]D-Fender: that too. so don't you build it! |
| 18:19.07 | Katty | [TK]D-Fender: or i will haunt you in the past. got it?! |
| 18:19.13 | jmetro | Katty: I would rather build SHODAN than Skynet. |
| 18:19.22 | jmetro | now THAT is a machine that will destroy humanity. |
| 18:19.28 | [TK]D-Fender | Katty: Doesn't that mean your threat has already ended and I'm feeling it now... if at all? |
| 18:19.32 | windback | I know I can put in all mailboxes an unavail message.. But perhaps there is a way to set a default unavail message for all mailboxes |
| 18:19.41 | [TK]D-Fender | Katty: 'cause I'm doin' fine :) |
| 18:19.45 | Katty | [TK]D-Fender: maybe i'm just gettin started. |
| 18:19.56 | Katty | jmetro: true story :< |
| 18:20.04 | jmetro | windback: I'm imagining a VERY clever script. |
| 18:20.18 | Katty | [TK]D-Fender: so about that middle name, before you so cleverly changed the topic |
| 18:20.20 | windback | [TK]D-Fender, Yes.. I want to avoid it |
| 18:20.37 | jmetro | windback: Stat their voicemail folder for an unavailable message |
| 18:20.40 | jmetro | if its found, play it |
| 18:21.33 | [TK]D-Fender | Katty: Sorry, that one's SEEK-RAT |
| 18:21.53 | Katty | [TK]D-Fender: s'ok. we'll just go with octavius. |
| 18:22.48 | windback | jmetro, I want to avoid a script to put an unavail default message in all folders. But if there is no way. I will do it |
| 18:23.58 | jmetro | windback: asterisk script |
| 18:24.07 | jmetro | windback: when voicemail is called, stat their folder for an unavail.msg |
| 18:24.15 | jmetro | windback: if its found, play it. otherwise, play a custom one of your own making. |
| 18:24.24 | windback | jmetro, good idea |
| 18:25.44 | windback | jmetro, are you thinking AGI script for example? |
| 18:25.53 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 18:25.57 | jmetro | asterisk commands, or you can call a php script. i dont like agi |
| 18:26.37 | windback | jmetro, haha.. why you don't like agi |
| 18:26.49 | jmetro | i already know PHP and asterisk, why learn another language :3 |
| 18:27.07 | Katty | jmetro: so you can order food in germany |
| 18:27.12 | Katty | jmetro: DONER KEBAB |
| 18:27.14 | raden | Katty, :D :D :D :D |
| 18:27.19 | jmetro | Katty: Ein paar kekse bitte. |
| 18:27.21 | Katty | hugs raden |
| 18:27.30 | Katty | jmetro: ok, nm then. you're good! |
| 18:27.43 | jmetro | <3 |
| 18:27.54 | raden | gives Katty huge hugs :P |
| 18:31.00 | jmetro | http://pastebin.com/vw8rHpCC |
| 18:31.04 | jmetro | that is what i do |
| 18:39.57 | *** join/#asterisk peetaur2 (~peter@x2f07d71.dyn.telefonica.de) |
| 18:48.18 | jmetro | im guessing everyone is in awe of that beautiful script |
| 18:48.25 | jmetro | either that or are shocked by how terrible it is |
| 18:49.49 | *** join/#asterisk dr0ck (~dr0ck@c-75-70-61-20.hsd1.co.comcast.net) |
| 18:55.04 | *** join/#asterisk tech_travis (~Travis@174.46.237.46) |
| 18:56.35 | Hive | [TK]D-Fender, a couple days ago I came in here with a queue issue where one call was being offered to the same extension many times. I couldn't open the full log then but I have managed to get that log file after hours. I'm wondering if you would take one more glance at this to see if you still think it was someone refusing calls; http://pastebin.com/dDs7c1hS |
| 18:59.09 | transfinite | is there way to dial all the analog channels on a DAHDI card more concisely than DIAL(DAHDI/1&DAHDI/2&DAHDI/3&...) ? Some kind of range or wildcard syntax? |
| 18:59.39 | transfinite | s/DIAL/Dial/ |
| 18:59.40 | WIMPy | no |
| 18:59.57 | WIMPy | But you could define a global variable to make it more readable. |
| 19:00.27 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 19:03.46 | [TK]D-Fender | transfinite: What exactly are you calling all at once there? |
| 19:04.44 | [TK]D-Fender | Hive: that's either DND on the phone, or user-rejected |
| 19:05.00 | Hive | [TK]D-Fender, thanks a bunch |
| 19:05.29 | transfinite | [TK]D-Fender: I have a bunch of analog phones in my house that will be connected to TDM cards. For most incoming calls, I just want them all to ring. |
| 19:05.44 | [TK]D-Fender | Hive: Since I'm seeing between 1-3 seconds I'm inclined to believe its more "user rejection" |
| 19:08.12 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 19:11.45 | *** join/#asterisk russum (~russum@94.139.139.17) |
| 19:22.08 | igcewieling | transfinite: I assume you are dialing FXS ports (phones) and not FXO (lines), correct? |
| 19:24.42 | *** part/#asterisk russum (~russum@94.139.139.17) |
| 19:27.37 | transfinite | that's right |
| 19:28.07 | [TK]D-Fender | transfinite: then nope, that's about it... |
| 19:29.25 | igcewieling | transfinite: stop worrying about how long your dialplan is. It is pointless and counterproductive and will just cause you pain and heartache |
| 19:30.28 | transfinite | i'll try, but as a programmer, long repetitive things already cause me pain and heartache |
| 19:30.48 | WIMPy | >>But you could define a global variable to make it more readable. |
| 19:31.02 | igcewieling | transfinite: dialplan is not a program or a language, your programmer instincts don't work for Dialplan |
| 19:31.22 | igcewieling | Interesting. extensions.conf longest line 299 chars, extensons.ael long line is 349 chars |
| 19:31.39 | igcewieling | WIMPy: doesnt make it more readable in the CLI or the LOGS |
| 19:32.13 | igcewieling | transfinite: trying to make your dialplan "simple and short" is a n00b mistake. |
| 19:32.38 | *** join/#asterisk Savemech (~savemech@109.197.79.141) |
| 19:33.30 | transfinite | guilty as charged |
| 19:35.21 | igcewieling | transfinite: telecoms and PBXs work ONLY be cause they are tied together with ugly kludges. Asterisk helps reduce that a little, but it is still simply an ugly universe. |
| 19:35.43 | [TK]D-Fender | transfinite: if you do that dial in several places you could declare a constant for it and just use that around your dialplan. |
| 19:35.46 | WIMPy | Reduce? |
| 19:36.16 | igcewieling | SURVEY: how many total extensions are shown on people's PBX when you do a "dialplan show" |
| 19:36.37 | igcewieling | WIMPy: have you ever seen a nortel meridian dialplan? Make Asterisk look dreamy |
| 19:37.12 | WIMPy | I'm not American, so no. |
| 19:37.13 | igcewieling | -= 45 extensions (437 priorities) in 18 contexts. =- |
| 19:37.39 | WIMPy | All PBXs have horrible configurations, but Asterisk surely beats them in complexity. |
| 19:37.48 | igcewieling | WIMPy: I disagree. |
| 19:37.58 | WIMPy | 790 extensions |
| 19:38.12 | _Corey_ | igcewieling: You trying to see who can give you the most f'd up output? |
| 19:38.31 | WIMPy | It's not more weird, but surely more work. |
| 19:38.36 | igcewieling | _Corey_: just curious. Our dialplan is VERY non-typical |
| 19:38.47 | _Corey_ | igcewieling: :) |
| 19:38.55 | _Corey_ | here's a non-typical one: |
| 19:38.57 | _Corey_ | -= 17819 extensions (75948 priorities) in 3503 contexts. =- |
| 19:39.42 | igcewieling | _Corey_: dude, use a database 8-| |
| 19:39.59 | _Corey_ | lol, not mine... a customer |
| 19:40.21 | igcewieling | FreePBX: -= 497 extensions (2400 priorities) in 141 contexts. =- |
| 19:43.52 | jmetro | 3503 contexts |
| 19:44.58 | jmetro | 519 extensions here on my small tiny box |
| 19:47.41 | robl^ | WIMPy: I wouldn't say ASterisk more complex. I would say in many ways its more comprehensible and more flexible. The thing with Asterisk, they give you the building blocks and you have to assemble it. Nortel CS1000/Meridian requires a fair bit of cryptic codes.. and its more about turning features on or off per DN. It's much more structured and less flexible. |
| 19:48.55 | WIMPy | robl^: Yes, it's more flexible, but you have to build everything yourself and there are many things that you can't build with the provided blocks. |
| 19:51.49 | robl^ | WIMPy: I agree there are some features or facilities that are difficult to implement with Asterisk.. some require a fair bit of hacking, AGI scripts. |
| 19:52.48 | igcewieling | I suspect most of those features don't even exist in many other PBXs |
| 19:52.58 | WIMPy | You sometimes wonder if it would have been easier to write a PBX from scratch. |
| 19:53.10 | WIMPy | Liek the other way round. |
| 19:53.29 | igcewieling | But how exactly is Dial(SIP/1234) easier than impiting 20 cryptic codes to configure a TN to dial a phone? |
| 19:54.19 | *** join/#asterisk serafie (~erin@nat/digium/x-zhllfcxyjxstuhdt) |
| 19:55.05 | WIMPy | That doesn't give you call back, pickup or BLF, yet, while on PBXs you usually only have to assign a number and everythign else just works. |
| 19:56.17 | robl^ | igcewieling: actually adding one in Nortel is easy.. but has a bit of a learning curve. |
| 19:56.54 | igcewieling | robl^ to you agree with " while on PBXs you usually only have to assign a number and everythign else just works." ? |
| 19:57.07 | ChannelZ | The phone system we had at my old company was configged by like a 1200bps serial port |
| 19:57.19 | ChannelZ | The voicemail ran on OS/2 |
| 19:57.20 | igcewieling | ChannelZ: you were lucky |
| 19:57.41 | igcewieling | ChannelZ: many pbxs must be configured using a phone |
| 19:57.49 | ChannelZ | Well I was especially lucky because I didn't deal with it at all :) |
| 19:58.50 | ChannelZ | but I remember we'd go out to dinner and my friend who did deal with it would forward the calls to his cell phone, but it would take 2 or 3 minutes to login and set the thing up to do it (and to put it back again) |
| 20:00.02 | robl^ | igcewieling: basically ;-) on nortel , you build a "TN" (terminal number). You then add keys -- a key can be a line key (shared or not), feature, autodial, or a BLF. you just say "key 01 SCR 2014" and key 1 of that phone has a shared call, ringing appearance of DN 2014. |
| 20:01.11 | robl^ | igcewieling: actually.. I'd disagree slighlty. KSUs are usually configured by phone (set based admin). Most PBXes used old school serial terminals until fairly recently |
| 20:03.50 | WIMPy | Maybe you guys have been sleeping over the past 20 years? |
| 20:06.40 | igcewieling | WIMPy: robl manages a large number of Nortel boxes |
| 20:06.42 | robl^ | the issue really is (and this goes with most opensource VoIP system) SIP wasn't never really designed to be used in the same way as an old school phone system. They've added some features along the way. However in oldschool PBXes, it was phone based. THe systems I use you define a phone as a physical device.. you then you modifiy the keys based on the need. In SIP world, like Asterisk, each "line appearance" is a separate account. Digium's |
| 20:06.43 | robl^ | <PROTECTED> |
| 20:06.48 | _Corey_ | WIMPy: "fairly recently" in a geological perspective ... :) |
| 20:07.08 | igcewieling | telecom moves slowly |
| 20:07.26 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 20:07.46 | WIMPy | SIP is only part of the issue, but a bigger one. |
| 20:07.50 | robl^ | I still have a dial up modem to dial up a London switch/voicemail at 9600 baud while sitting in Texas ;-) |
| 20:07.50 | WIMPy | _Corey_: Yeah. |
| 20:07.54 | igcewieling | I know the Nortel Meridian we had at the real estate company was configured via a physical phone. They never purchased the license (and/or hardware) for serial access |
| 20:08.17 | jmetro | igcewieling: thats most companies unfortunately |
| 20:09.03 | igcewieling | They had a T-1 card, but a license to enable PRI support on the card was $1,200 (IIRC) |
| 20:09.04 | robl^ | igcewieling: Nortel was HORRIBLE at branding. THere were some Norstar systems that were branded as Nortel Meridian Norstar.. but it isn't the same as the Meridian 1 / CS 1000 series. |
| 20:09.24 | WIMPy | Looks liek the ripoff isn't as bad here. I've never seen anything without serial access inclused. On in recent years with LAn access included. |
| 20:09.32 | robl^ | The Norstar was almost always set-based admin. |
| 20:10.02 | WIMPy | bad bad typing. |
| 20:10.55 | igcewieling | Companies don't replace their PBX yearly. Maybe once every 10 years or so |
| 20:11.50 | WIMPy | They usually don't even touch them for years. if at all after initial setup. |
| 20:12.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.142) |
| 20:12.35 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 20:12.58 | robl^ | I'd say ~10 yrs is about right.. even then, most companies won't ript out and start over, they tend to upgrade to the latest iteration of the same platform.. or its successor. They try to re-use as much of the existing investment as possible for as long as possible |
| 20:13.24 | WIMPy | Makes sense to me. |
| 20:15.17 | jmetro | which winds up sinking them into the POTS-line money pit |
| 20:16.05 | WIMPy | do you think Asterisk is cheaper? |
| 20:16.18 | robl^ | my current employer has been on some version of the Nortel platforms for over 20 years..and its possible they will decide to move to the Avaya solution (as Avaya bought out Nortel and incorporated their technology into their products) |
| 20:16.27 | jmetro | raspberry pi + 3 cents a minute = working phone system |
| 20:16.48 | WIMPy | Plus some weeks for configuration. |
| 20:16.51 | jmetro | avaya is sad, please dont do it. |
| 20:17.37 | robl^ | jmetro: depends on the size of the company, and how much it is worth for them to have reliable systems. I'd say we have around ~20,000 phones installed in our offices world wide |
| 20:17.46 | WIMPy | I like the dektop switching style UI on their phones. |
| 20:18.02 | robl^ | I don't think a raspberry pi would handle our needs ;-) |
| 20:18.44 | _Corey_ | robl^: There are some enterprise Asterisk vendors who work with organizations like yours.... (*cough* *cough*) |
| 20:19.36 | robl^ | _Corey_: wrong feature set. Trust me, I had them take a hard look at Asterisk.. distributed through multiple offices.. but it was turned down by management. |
| 20:19.40 | _Corey_ | Not always an easy sale to upper management, but you're in good company these days. |
| 20:20.54 | igcewieling | Asterisk's biggest problem for enterprise and non-enterprise is lack of SLA |
| 20:21.12 | robl^ | _Corey_: shared line appearances are a must for us. Asterisk lacks a bit there. I know there is an implementation, but its not up to snuff yet |
| 20:22.27 | _Corey_ | robl^: Well, at your scale a "pure" Asterisk solution isn't really a great idea. Kamailio+Asterisk can do a great job scaling and can offer a real BLA function. |
| 20:23.03 | _Corey_ | Kamailio+Asterisk is the combination used by all the "huge" Asterisk deployments I'm aware of. (I think there's one somewhere like this with 250k users) |
| 20:23.06 | igcewieling | _Corey_: Have you personally set up SLA on Kamailio? |
| 20:23.10 | _Corey_ | yes |
| 20:23.30 | _Corey_ | (w/Polycom phones...) |
| 20:23.50 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 20:23.55 | igcewieling | _Corey_: I am very impressed |
| 20:24.18 | _Corey_ | Don't be too impressed... there's a howto somewhere on it. :) |
| 20:24.39 | *** join/#asterisk bandroidx (~bandroidx@unaffiliated/bandroid) |
| 20:24.51 | igcewieling | When I played with Kamailio we had to route all calls to Asterisk for routing anyway so it didn't really help that much |
| 20:25.21 | robl^ | _Corey_: yeah. Kamailo was considered as part of the equation. However upper management narrowed it down to 3 possible solutions and Asterisk didn't make the cut. The final 3 are in the formal RFP stage. |
| 20:25.30 | _Corey_ | Well, every implementation of Kamailio is going to be different |
| 20:26.09 | _Corey_ | robl^: Too bad... ;) |
| 20:26.31 | robl^ | The sad thing is.. I'm not going to be here to see the final solution. |
| 20:27.09 | robl^ | I'm leaving my current employer in a few weeks. |
| 20:27.33 | _Corey_ | Do you have another position lined up? |
| 20:28.42 | robl^ | Depends on the perspective. ;-) I have a couple contract/projects that I will be working on. |
| 20:29.15 | jmetro | his first choice is sitting in his basement manufacturing arduino boards to control peoples silent curtains |
| 20:31.09 | robl^ | This wasn't quite a voluntary departure, but I had known it was coming for nearly 2 years. My position is being moved across the country. I was asked to move, but for family related reasons, I elected not to do so. So I am effectively being laid off. |
| 20:33.43 | _Corey_ | robl^: Ahh... I know we're going to be posting a job in the Northeast US. If you're interested, PM me your e-mail and I'll send you the link when it goes up. |
| 20:34.45 | jmetro | Hm? |
| 20:34.51 | robl^ | _Corey_: thanks, but that would be one heck of a commute. I'm in Texas.. unless you need a telecommuter |
| 20:34.56 | jmetro | BLF's seem to work perfectly fine on my system... |
| 20:35.02 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 20:35.27 | _Corey_ | robl^: Probably wouldn't work... ;) |
| 20:36.52 | robl^ | _Corey_: just add a dial up modem and I can work remotely ;) |
| 20:37.29 | _Corey_ | might be slow... ;) |
| 20:38.16 | robl^ | I've been trying convince my employer that where I worked from doesn't matter, but upper management sees it different. But as it is, I manage telecom for 14 offices world wide already. I'm not sure me moving 1000 miles would make much difference |
| 20:38.39 | *** join/#asterisk peetaur2 (~peter@x2f07d71.dyn.telefonica.de) |
| 20:38.57 | robl^ | getting way off topic here. *now returning you to your normally scheduled asterisk chat* |
| 20:39.30 | jmetro | unless you move somewhere that only has *gasp* DSL |
| 21:11.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.173) |
| 21:16.15 | jmetro | Anyone know how to remove a queue agent from a queue that has a space at the end of their name :< |
| 21:18.13 | ChannelZ | Flame thrower. Burn the entire thing to the ground! |
| 21:19.11 | jmetro | plz. |
| 21:24.30 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 21:27.25 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 21:30.01 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 21:47.52 | ChannelZ | Guess it must be a bug that it allowed the space in there in the first place |
| 21:49.00 | jmetro | [my operator panel] |
| 21:49.21 | jmetro | dont ask me how it inserted a space without asterisk seeing it and getting rid of it like it did for all my other attempts to remove it |
| 21:49.56 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 21:50.34 | igcewieling | Heh, my local cable company does not show the cheapest package as an option on their web site, their customer service people won't mention it either. |
| 21:51.02 | igcewieling | jmetro: Using FreePBX? |
| 21:51.07 | jmetro | isymphony |
| 21:51.16 | jmetro | I say no to dru...freepbx |
| 21:51.30 | igcewieling | FreePBX doesn't trim spaces from field imputs either. |
| 21:51.44 | igcewieling | very annoying |
| 21:57.49 | Weezey | how hard is it to add a trim() to your code? |
| 22:00.59 | igcewieling | Weezey: FreePBX's reason is they use a 3rd party library for database access |
| 22:01.54 | jmetro | vim [lib].lib |
| 22:09.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:09.30 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 22:10.55 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
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| 22:27.37 | jeev | anyone here familiar with mediatrix 1124's? or know of any other channel to ask this in? |
| 22:34.18 | *** part/#asterisk tech_travis (~Travis@174.46.237.46) |
| 22:47.59 | *** join/#asterisk sluke (4659b301@gateway/web/freenode/ip.70.89.179.1) |
| 22:48.07 | sluke | hello |
| 22:48.46 | *** join/#asterisk dannymcc (~dannymcc@146.255.111.108) |
| 22:48.53 | sluke | I'm getting the following error when trying to reboot Ubuntu 12.04 WANPIPE1: Warning excessive fifo errors |
| 22:49.07 | *** join/#asterisk teff (~teff@client-86-31-141-131.oxfd.adsl.virginmedia.com) |
| 22:49.12 | sluke | but if I do wanrouter stop ... I'm able to reboot the machine |
| 22:49.32 | sluke | appreciate if you can shed some light |
| 22:51.27 | sluke | anyone? |
| 22:51.51 | WIMPy | Sounds like a good opportunity to try Sangomas support. |
| 22:53.05 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 22:53.38 | sluke | ok... I was hoping that someone can help me pointing to something to look at other than the sangoma support :) |
| 22:54.37 | WIMPy | There's always the chance you get an answer some hours later. |
| 22:57.00 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 23:00.36 | igcewieling | does dmesg show anything interesting? |
| 23:02.55 | igcewieling | sluke: make sure you have the latest sangoma drivers and a reasonably recent (but NOT latest) DAHDI |
| 23:03.23 | igcewieling | we had an issue with old dahdi and old sangoma where the system would simply lock when trying to reboot in some situations |
| 23:04.24 | sluke | igcwieling: I've already verified that |
| 23:04.35 | *** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net) |
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| 23:18.32 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 23:20.39 | Micc | I've got an asterisk server version 1.8.9.2 that has been working fine for a long time, then I had to change the IP address on it today and it is having problems starting now. It keeps telling me theres no rtp engine loaded. Loading it manually gives no output and locks the server. |
| 23:21.04 | Micc | starting the server with -vvvvc shows nothing suspicious except it doesn't even get to loading rtp. |
| 23:21.35 | Micc | I suppose thats the place to start. |
| 23:21.42 | Micc | find it what its hanging on. |
| 23:23.00 | Micc | I figured it out |
| 23:23.06 | Micc | its not able to connect to my database anymore. |
| 23:23.14 | Micc | its hanging loading res_odbc |
| 23:33.01 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 23:35.20 | igcewieling1 | there you go |
| 23:35.27 | igcewieling1 | also update your asterisk to the latest 1.8 |
| 23:35.36 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:35.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:35.44 | igcewieling1 | upgrading within a major version is usually a low risk thing |
| 23:36.26 | ryan_turner | Fresh install of raspberry-asterisk, have got freepbx's end point configuration manager sending configs out to my polycom ip 321 correctly, but when the phone boots I get error "0702193515|sip |4|03|Registration failed User: 101, Error Code:403 Forbidden" on the phone. Any ideas what I might need to tweak? |
| 23:36.45 | ryan_turner | (101 is the extension I made) |
| 23:37.10 | WIMPy | #freepbx is next door left. |
| 23:37.14 | ryan_turner | aha |
| 23:37.21 | ryan_turner | didnt realize, sorry the whole thing is still fuzzy to me |
| 23:37.25 | ryan_turner | thanks for being polite :) |
| 23:43.01 | *** part/#asterisk sluke (4659b301@gateway/web/freenode/ip.70.89.179.1) |
| 23:43.13 | Micc | igcewieling1, I would generally agree with you, but I tried that a few weeks ago and royally screwed things up. |
| 23:44.07 | ryan_turner | Wow, this is really cool. Old iphone + raspberry pi + google voice motif module = free calling and setup as a total noob in about an hour. |
| 23:44.16 | ryan_turner | err ip phone * |
| 23:49.14 | sruffell | That's cool |