| 00:01.13 | dijib | thats no dinner |
| 00:01.41 | ChannelZ | It is if you're a bird |
| 00:02.42 | dijib | or a monarch butterfly |
| 00:02.48 | ChannelZ | Squirrel |
| 00:03.08 | dijib | rodent. |
| 00:03.25 | ChannelZ | Butterflies eat sunflower seeds? |
| 00:04.53 | dijib | only monarchs i think |
| 00:05.07 | dijib | thats what someone who hunts them for photos told me |
| 00:05.38 | p3nguin | Butterflies have no teeth, silly. |
| 00:05.47 | dijib | yeh i dont know dude |
| 00:05.57 | dijib | whats that site for the logs of this channel? |
| 00:06.10 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 00:06.35 | p3nguin | I'd guess your google works the same as mine. |
| 00:07.47 | p3nguin | Butterflies only consume nectar. |
| 00:08.05 | dijib | you consume nectar of the asterisk gods |
| 00:08.14 | p3nguin | If you only had a straw attached to your face, you'd eat less solids, too. |
| 00:08.21 | dijib | lol |
| 00:28.49 | *** join/#asterisk droemel (~droemel@p4FCAD27F.dip.t-dialin.net) |
| 00:29.40 | dijib | Logs from my ringing issue. http://pastebin.com/2dWUudyL whe the SIP/device ringing. the devices ring but caller had no indication of this event |
| 00:30.57 | p3nguin | Did you ever look for the 180 Ringing in the sip debug? |
| 00:31.53 | dijib | oh is that where i was to look. |
| 00:31.55 | dijib | nope. |
| 00:32.13 | p3nguin | If it's SIP, that seems like the obvious place to look. |
| 00:33.24 | dijib | there is no 180 in the sip debug logs |
| 00:33.39 | dijib | using a search |
| 00:34.17 | dijib | ive got 103's |
| 00:39.15 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 00:40.56 | shmaltz | hi every1 |
| 00:42.01 | dijib | ok p3nguin i do have 180 ringing evens afterall |
| 00:45.27 | *** join/#asterisk tts626 (~tim@38.100.208.78) |
| 00:45.37 | *** join/#asterisk tekzilla (~jon@g231181200.adsl.alicedsl.de) |
| 00:46.49 | tts626 | hey all, I'm trying out asterclick and I'm getting an error about a missing agents.conf, which I don't have. Running asterisk 1.6 on AsteriskNow. |
| 00:47.12 | tts626 | anybody have any pointers? |
| 00:47.36 | tekzilla | in my extensions.conf tehre are some contexts included from an autogenerated extensions.ael, i want to add a custom catchall hook. where should i put it and what should it look like |
| 00:48.44 | tyman | Is there asterisk config involved in getting a polycom phone's fwd feature to work? My phones work fine while other phones running latest Polycom UCS software show their fwd'd on the display but don't fwd. |
| 00:49.16 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 00:52.07 | shmaltz | tyman, it's part of the featurex in the polycom xml config files |
| 00:52.14 | dan__t | Asked a few days ago but unfortunately I had to bail - oops. Is it legal to dynamically name a context? Sure I could do it as a macro I guess, but I'm reading that macros can only go 7 levels deep. I don't know if I'd ever get to that limit, but just to be "safe" (I guess?), I'd rather use a real context. |
| 00:52.26 | dan__t | Like [my_context_${CALLERID(num)}] etc etc |
| 00:53.12 | shmaltz | dan__t, should work as long as what ${var} translates to exists |
| 00:53.31 | dan__t | That's kinda neat actually. |
| 00:53.40 | dan__t | I couldn't think of a reason why it wold *not* work |
| 00:54.24 | ChannelZ | Seems a stretch to me, is that documented anywhere? |
| 00:54.35 | *** part/#asterisk tts626 (~tim@38.100.208.78) |
| 00:55.05 | tyman | shmaltz: i'll google that thenâ¦looking for where⦠thanks |
| 00:59.13 | shmaltz | ChannelZ, what? the ${var} context thingy? |
| 00:59.24 | ChannelZ | yes |
| 01:00.22 | shmaltz | I'd be surprised if it doesnt work |
| 01:00.33 | shmaltz | is testing it now on 1.2 |
| 01:02.26 | ChannelZ | I mean you could jump to a context name built from variables like Goto(test-${something},1,1) but how does a context with a variable name in it make any sense? |
| 01:03.35 | shmaltz | done it works |
| 01:03.48 | shmaltz | ChannelZ, i can think of some ideas, example |
| 01:03.54 | shmaltz | multi tennanting based on DID |
| 01:04.06 | shmaltz | give each DID a context with the DID as the name of the context |
| 01:04.51 | ChannelZ | I'd be interested to see your test, I suspect you are interpreting the question differently than I |
| 01:06.25 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 01:07.46 | ChannelZ | If I have a [test-123] context and I do Set(something=123) and Goto(test-${something},1,1) of course that'll work |
| 01:16.40 | shmaltz | ChannelZ, thats exactly what I did |
| 01:18.56 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 01:19.07 | BenC[UK] | Evening guys |
| 01:19.14 | BenC[UK] | Any "queue" experts around? |
| 01:20.11 | BenC[UK] | I am trying to use leastrecent strategy, but its not working as I expected, and I am wondering if I can use penalties or something to help with this |
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| 01:40.21 | VoipForces | Hi all, got a strange problem. Calling a known busy number (on PRI or SIP trunk). I get "DAHDI/26-1 is busy" which is file. The dialplan then PlayTones("busy" but I get a translate.c: no samples for ulawtolin error⦠any ideas running asterisk 1.6 |
| 01:42.46 | *** join/#asterisk gxdssoft (~gxdssoft@190.233.196.199) |
| 01:46.58 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 01:47.46 | james_zhu | core show translations? |
| 01:54.29 | *** join/#asterisk penguin (greenwolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 02:01.10 | penguin | anyone familiar with Opensips? or OpenSer? |
| 02:01.37 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 02:01.48 | pdtpatrick | Question .. what does a queue status of 4 mean ? |
| 02:04.19 | pdtpatrick | http://pastebin.com/13F5iHi0 |
| 02:04.27 | pdtpatrick | I'm see that when i run queue show |
| 02:06.53 | pdtpatrick | seeing* |
| 02:07.43 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 02:20.16 | VoipForces | pdtpatrick: I don't see status 4 on your pastebin. |
| 02:21.09 | VoipForces | james_zhu: Thanks, but it seems like a dialplan or playtones app issue. If I do a Playback before the Playtones it works... |
| 02:21.20 | *** join/#asterisk CoffeeIV (~rgr@66-90-216-138.dyn.grandenetworks.net) |
| 02:21.44 | pdtpatrick | VoipForces: would u know how to change the invalid in that paste bin to say something else? |
| 02:22.43 | VoipForces | pdtpatrick: Well it asterisk says it's invalid there most be a reason. What the diff between this agent and the others? |
| 02:23.04 | pdtpatrick | the only different is they are in a queue |
| 02:23.17 | pdtpatrick | when u call it .. it goes to voicemail based on the time of the day |
| 02:23.30 | pdtpatrick | the problem started happening two ago out of no where |
| 02:23.45 | pdtpatrick | Question .. why does status on this page .. shows blank? |
| 02:23.45 | pdtpatrick | 4 - |
| 02:23.51 | pdtpatrick | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueStatus |
| 02:23.57 | pdtpatrick | scroll to the bottom .. #4 is blank |
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| 02:29.20 | voxter | Asterisk 1.8.5.0, it starts up, most things seem to be functioning, yet some commands (like core stop ?) do not register as available commands. No idea where to begin debugging an asterisk startup, in terms of where the problem is being introduced. Ideas? |
| 02:30.50 | voxter | loading with asterisk -vvvvvvvgc the last "module load" output was from chan_bridge.so and specifically IAX2's stuff. |
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| 02:36.57 | voxter | maybe strace will help me. |
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| 06:46.37 | ChannelZ | It's oh so quiet.. shhhh, shhhh.. |
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| 06:52.10 | singler | ... was until you spoken :) |
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| 07:08.55 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:08.57 | Polysics | hello |
| 07:09.05 | Polysics | further on my originate hangup worlk |
| 07:09.18 | Polysics | how do i know the real channel name of an originate? |
| 07:09.50 | Polysics | that is, i originate to SIP/10004, but the channel name needed to Hangup will be SIP/10004-000000f or something |
| 07:10.15 | Polysics | can i hang up using only the peer name somehow? or know that channel in advance? |
| 07:10.47 | Polysics | situation is: caller comes in, i start moh, then start originating out, on answer people get told who is calling and press 1 to accept |
| 07:11.02 | Polysics | all is good and well until the caller hangs up while the remote end is ringing |
| 07:11.14 | Polysics | does a channel even EXIST while it is ringing? |
| 07:11.52 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:12.40 | kaldemar | a channel exists in ringing state. the SoftHangup app enables you to hang up all channels on a specified device. so you really don't even have to know the whole channel name but there might be consequences. |
| 07:13.11 | Polysics | kaldemar, the system only allows one call at a time, so that might work |
| 07:13.22 | Polysics | but why am i getting no events on caller hangup in that case? |
| 07:13.39 | Polysics | just four VarSets |
| 07:13.48 | kaldemar | no idea. |
| 07:14.32 | Polysics | argh, SoftHangup is an AGI command/application |
| 07:14.35 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:14.38 | Polysics | i need to run it via AMI |
| 07:15.49 | Polysics | can I use Command? |
| 07:16.01 | kaldemar | Command is for cli commands |
| 07:16.23 | Polysics | isn't an Application available in both contexts? |
| 07:17.37 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.229.189) |
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| 07:18.04 | kaldemar | Application? no, does not exist in AMI. |
| 07:18.24 | Polysics | aside from the inherent complexity of the issues, am i doing anything wrong? i mean, i must not be the first person to want to hang up an originated call while it rings :-D |
| 07:18.54 | kaldemar | but you can always execute a dialplan app from AMI via Originate. |
| 07:19.28 | Polysics | hmm |
| 07:19.40 | kaldemar | how are you originating the call again? |
| 07:19.53 | kaldemar | AGI? AMI? app Originate? |
| 07:20.09 | kaldemar | CLI command? |
| 07:20.14 | Polysics | AMI command |
| 07:20.39 | Polysics | i probably should have used the Originate app, but it did not exists when work on this was started |
| 07:20.55 | Polysics | i am using AGI on Adhearsion and have a DB to store values if needed |
| 07:21.14 | Polysics | so ican basically do "anything i want", it's a case of not knowing what i want :-) |
| 07:21.15 | kaldemar | and you're not getting any response or event that has the channel? |
| 07:21.46 | Polysics | there's a Newchannel, obviously, but i can't know for which originate it is |
| 07:22.00 | Polysics | in the case of two calls at the same time, whose is which? :-D |
| 07:22.32 | Polysics | OH |
| 07:22.40 | Polysics | oh lawdy |
| 07:22.44 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 07:22.47 | Polysics | you made me go eureka |
| 07:22.55 | kaldemar | i thought there was only one call at the time to a peer. :) |
| 07:23.12 | Polysics | that is also true |
| 07:23.20 | Polysics | but there is an even easier way out, it seems |
| 07:23.37 | Polysics | i set a variable called DESTCHANNEL to allow bridging on answer |
| 07:23.37 | kaldemar | so you take the newchannel event and parse for what you just originated to. |
| 07:23.56 | Polysics | that VarSet has both channels, even more foolproof |
| 07:24.04 | Polysics | let's see if it works |
| 07:25.35 | Polysics | still, having to catch a specific VarSet event to detect when the caller hangs up in that context is pretty strange |
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| 07:25.54 | *** join/#asterisk reber (~reber@212-198-54-60.rev.numericable.fr) |
| 07:27.25 | nicola_pav | hello. I want to use asterisk HA with redfone. I am trying to access the website but no luck. I am trying to find the fonulator package in order to set it up but i cannot. anyone has any idea? |
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| 07:42.19 | Polysics | eureka! it works! |
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| 09:07.35 | joobie | hey guys.. i have an m4a file that i want to conver to alaw / ulaw.. what's the easiest way to do this to retain quality? |
| 09:10.10 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:10.15 | krion | hey guys |
| 09:10.27 | krion | i'm having a strange behaviour relating astdb |
| 09:10.58 | krion | database show only show registry, not the fwd-unc and stuff like that |
| 09:11.08 | krion | but the astdb has fwd-unc and co |
| 09:11.46 | BenC[UK] | Hi, is there anyway to get a queue using the leastrecent strategy to automatically try the next member if theres no answer/ |
| 09:19.15 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
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| 09:23.07 | wdoekes2 | BenC[UK]: afaik, you can always try a next member, regardless of strategy |
| 09:23.29 | BenC[UK] | if I set the timeout to 3s, it tries the same member again |
| 09:23.51 | wdoekes2 | wow |
| 09:24.07 | wdoekes2 | I would consider that a bug |
| 09:24.41 | wdoekes2 | but it's seems perfectly plausible that the "next" function returns the least-called one again |
| 09:26.06 | wdoekes2 | do you see the same behaviour if you use "fewestcalls" ? |
| 09:27.57 | wdoekes2 | .. looking at the code, I suspect the non-answering member should get a penalty |
| 09:29.38 | wdoekes2 | or not |
| 09:30.01 | wdoekes2 | the config mentions penalties as static |
| 09:32.40 | wdoekes2 | autopause.. but then you would have to unpause them |
| 09:32.50 | wdoekes2 | </thinking_out_loud> |
| 09:33.36 | kaldemar | krion: what makes you think that astdb has content that database show does not list? |
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| 09:43.23 | krion | kaldemar: a cat on it show some info |
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| 09:43.51 | krion | kaldemar: and if, for example, i enable fwd-unc on on sip phone, the fwd is working fine, but not displaying when show database is done |
| 09:49.58 | BenC[UK] | wdoekes2: I havent tried fewest calls - I am using rrmemory at the moment - but because our calls are between 10 seconds (answer phone) and 10 minutes long, its not working too well |
| 09:50.03 | BenC[UK] | I will try fewest calls later today |
| 09:50.47 | *** join/#asterisk killown (~geek@unaffiliated/killown) |
| 09:51.07 | killown | I can't make calls http://bpaste.net/show/18347/ do anyone help me? |
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| 09:53.57 | killown | Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE |
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| 09:58.04 | awk | killown well is the trunk up |
| 09:58.07 | awk | is the link up |
| 09:58.14 | awk | what does dahdi state on the line |
| 09:58.19 | awk | have you got a D channel |
| 09:59.01 | killown | awk, http://bpaste.net/show/18348/ |
| 10:01.38 | awk | oh its a sip trunk, get some sip debug info |
| 10:02.11 | killown | Http://bpaste.net/show/18347/ ???? |
| 10:04.03 | killown | Trunk Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks :/ |
| 10:04.16 | awk | sip debug not verbose output of the CLI |
| 10:04.27 | killown | awk, How sip debug it? |
| 10:04.51 | awk | sip set debug |
| 10:04.57 | krion | kaldemar: any clue ? |
| 10:04.58 | awk | either on or ip of trunk you want to capture |
| 10:09.16 | killown | awk, http://pastie.org/2464851 |
| 10:10.50 | kaldemar | krion: forwarding on a phone says nothing at all. it doesn't even have to use the db. |
| 10:11.04 | *** join/#asterisk Bryanstein (~Bryanstei@unaffiliated/geekjuice) |
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| 10:22.22 | krion | kaldemar: but the info is in the db, i tough they use it |
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| 10:44.00 | awk | killown I have no idea why you getting so many registry requests |
| 10:44.11 | awk | have you phoned the VoIP company and asked them to see why its being rejected on their side? |
| 10:49.10 | *** join/#asterisk dr_ (~dr@83.166.214.174) |
| 10:55.02 | kaldemar | killown: ask in #freepbx how to configure NAT settings. |
| 10:56.35 | kaldemar | killown: altough you left the interesting part of sip debug out of your paste, it looks like you're sending a private ip address to intelbras and hence you never get the responses. |
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| 11:53.15 | killown | Please, anyone help me? http://bpaste.net/show/18349/ I can make calls but incoming calls doesnt work |
| 11:57.19 | kaldemar | killown: what happens in CLI and sip debug when you try to make a call? |
| 11:57.57 | kaldemar | first a wild guess, have you forwarded ports to you asterisk box in your NAT router? |
| 11:58.40 | killown | kaldemar Aaaaaaa you right about the router, thank you |
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| 12:04.59 | killown | kaldemar, I had set up dmz to the asterisk server |
| 12:05.10 | killown | Still doesn work to incoming calls |
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| 12:06.54 | dmz | NAT & SIP don't always play well together |
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| 12:42.03 | irroot | jkroon yo there came accross your asterisk / dahdi for za blog nice thx |
| 12:43.55 | jkroon | irroot, pleasure. |
| 12:44.18 | jkroon | it's outdated though, digium added a patch that stops you from putting "global" suff in chan_dahdi.conf |
| 12:44.36 | irroot | yeah i have a automated approach to it |
| 12:44.55 | irroot | oh i was swearing at telkom eariler im sure that is not wrong :P |
| 12:45.05 | jkroon | so in users.conf in my [line](!) section I've just added all of that in, minor changes but it works equally well. |
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| 12:55.28 | jkroon | after picking up on that and moving everything into users.conf everything worked again as expected. |
| 12:55.53 | irroot | jkroon me no like users.conf |
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| 12:56.50 | jkroon | irroot, horses for courses. use whichever suits your needs :) |
| 12:57.00 | *** join/#asterisk Fritz09 (~Adium@pop1-1207.catv.wtnet.de) |
| 12:58.20 | irroot | jkroon indeed all my stuff is in realtime users.conf for new installs and provisioning and the like is a win .... i have all the scripts and bits in apache already |
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| 13:19.18 | Jacke | how do i reverse polarisation on an analog line from within an app, how do i do that? |
| 13:20.50 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:23.10 | Kellta | Hi, anyone awake? I've an issue I would like to resolve, if anyone can help. I have an asterisk 1.6.2.7 that connects three channels in one app_konference conference room. Two channels is outbound cell phones via a SIP trunk and the third is a local channel with Playback application. The problem is that after a minute or so the volume of the users with phone gets lower and lower. If I skip the Playback channel the audio is perfect. Could there be some |
| 13:23.10 | Kellta | noise reduction algorithm interfering or something like that? |
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| 13:34.52 | killown | kaldemar, Still there for help me? |
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| 13:41.02 | Katty | )= |
| 13:45.37 | irroot | Katty hi there happy spring day |
| 13:47.55 | Katty | it's not a happy day i'm afraid |
| 13:50.03 | irroot | what wrong |
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| 13:58.47 | Katty | irroot: single again, as of about...12 hours ago |
| 13:59.16 | beek | waves to Katty |
| 13:59.28 | irroot | eish |
| 13:59.41 | Qwell | Katty: boo. or yay? |
| 14:01.02 | Katty | some boo. mostly relief |
| 14:01.06 | Katty | and a little bit of dread. |
| 14:01.56 | Katty | Qwell: technically we are on A Break |
| 14:02.02 | Qwell | I see |
| 14:02.10 | Katty | Qwell: but i'm pretty sure that's just BS |
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| 14:07.06 | Katty | Qwell: it other news, i found some new hobbies. |
| 14:07.34 | Katty | Qwell: learning to knit, took up writing in a journal, and i found a new tv series to watch called Lost Girl |
| 14:08.07 | beek | wants a nice sweater. |
| 14:08.46 | Katty | ehn, i'm not gonna knit sweaters |
| 14:08.53 | beek | :( |
| 14:08.57 | Katty | i'm going to knit awesome geeky things, like a Tardis, and star trek pot holders |
| 14:09.23 | beek | How much pot will your star trek holders hold? |
| 14:09.31 | beek | nickel bag? dime bag? |
| 14:10.07 | Katty | it will hold pizza pants. |
| 14:10.09 | irroot | i hope mine grows :P |
| 14:10.15 | Katty | and containers full of lovely casseroles |
| 14:10.19 | Katty | oh and bacons. |
| 14:10.37 | Katty | ...i totally just saw i wrote pizza pants |
| 14:10.41 | Qwell | How come I don't have any containers full of bacon? |
| 14:10.48 | Katty | cause you're not at my house. |
| 14:11.05 | Qwell | Are you trying to bribe me?! |
| 14:12.47 | Katty | i uhh |
| 14:13.05 | Katty | not really? |
| 14:13.13 | Katty | you have an open invitation regardless tho |
| 14:13.19 | Katty | just watch out for the dog |
| 14:13.34 | beek | Katty: Still have the critter cam? |
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| 14:16.23 | Katty | i do not )= |
| 14:16.33 | Katty | it is currently offline. mister you-know-who has borrowed the wireless adapter for it |
| 14:16.53 | beek | "borrowed"? |
| 14:18.27 | Katty | i'll get it back. |
| 14:18.39 | Katty | i'm not worried about that |
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| 14:27.20 | eduzimrs | hi there, im trying to edit cdr_custom.conf but doesn`t take effect at Master.csv anyone knows something im missing? |
| 14:29.06 | Qwell | cdr_custom doesn't use Master.csv.. |
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| 14:33.16 | eduzimrs | right but so where is the real cdr format that i see at Master.csv ? |
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| 14:39.16 | eduzimrs | where do i edit ...cdr-csv/Master.csv ?? |
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| 14:40.25 | navaismo | hi, good morning |
| 14:42.21 | Katty | good morning |
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| 14:43.43 | chuckf | great morning, 4 day weekend coming up and get to work the Indy car race:) |
| 14:44.26 | atheos | chuckf you work for a team, or a venue? |
| 14:44.52 | Qwell | maybe he's a racecar driver. |
| 14:45.23 | chuckf | I'm volunteering at the venue |
| 14:45.29 | Qwell | or maybe he sells hotdogs or something |
| 14:45.54 | chuckf | I'll be 'guarding' the cars as they go from the paddock to pit road |
| 14:47.11 | atheos | chuckf - when I retire, I plan to be a yellow shirt at IMS. I love Indycar racing :) |
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| 14:47.36 | chuckf | that should be fun |
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| 14:48.24 | *** join/#asterisk P-NuT (~P-NuT@188-223-86-64.zone14.bethere.co.uk) |
| 14:49.06 | P-NuT | Hi everyone where can I look for details on doing video through asterisk? |
| 14:49.57 | P-NuT | info seems to be a bit thin on the ground. |
| 14:50.15 | Chainsaw | Well it is fairly new functionality. |
| 14:50.24 | Qwell | it really isn't |
| 14:50.28 | P-NuT | how new? |
| 14:50.32 | P-NuT | I dont think it is |
| 14:50.33 | Qwell | SIP has supported video for a very long time. |
| 14:50.40 | Qwell | Probably even 1.0. |
| 14:50.43 | P-NuT | yeah that what I thought |
| 14:50.56 | Qwell | sip.conf, videosupport=yes |
| 14:50.58 | Qwell | done. |
| 14:51.03 | P-NuT | So, would it be h.323 that does the video? |
| 14:51.10 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:51.15 | Qwell | Nobody uses h.323. |
| 14:51.23 | P-NuT | SIP video then? |
| 14:51.29 | chazzam | he just said SIP, and then you throw out H.323 |
| 14:51.30 | Qwell | oh, good, may isn't here. He would've yelled at me. :) |
| 14:51.32 | chazzam | what? |
| 14:52.09 | P-NuT | So what would the best and easiest way of doing it? |
| 14:52.10 | WIMPy | Yes, somebody should :-) |
| 14:52.17 | Qwell | sip.conf, videosupport=yes |
| 14:52.28 | Qwell | It's literally that easy. |
| 14:52.37 | P-NuT | ok.... |
| 14:52.41 | file | Qwell, codec |
| 14:52.45 | P-NuT | so enable sip video suport, |
| 14:52.45 | Qwell | In fact, it's probably been enabled by default for some time. |
| 14:52.47 | leifmadsen | allow=h264 |
| 14:52.50 | Qwell | file: pfft, allow=all |
| 14:52.54 | P-NuT | right. |
| 14:53.12 | P-NuT | so enable video support, allow h264, anything else? |
| 14:53.16 | leifmadsen | nope |
| 14:53.22 | leifmadsen | place a call :) |
| 14:53.25 | P-NuT | how is the client configured? |
| 14:53.28 | Qwell | have a video phone |
| 14:53.29 | anonymouz666 | anyone know what could cause a DTMF to be detect 22 secs after the digit was pressed? background() executes in 10:20:00, 10:20:22 the DTMF was detected _correctly_. |
| 14:53.29 | Qwell | that might help |
| 14:53.41 | P-NuT | Well, |
| 14:53.44 | leifmadsen | P-NuT: make sure the client can understand and use h.264 |
| 14:53.54 | P-NuT | I want to do all this using a softphone |
| 14:53.58 | leifmadsen | anonymouz666: jitter or latency? |
| 14:54.06 | P-NuT | for video conferencing. |
| 14:54.11 | leifmadsen | P-NuT: ok, so get a softphone that supports the appropriate codecs then |
| 14:54.15 | Qwell | P-NuT: people have had success using jitsi |
| 14:54.19 | leifmadsen | aye |
| 14:54.24 | P-NuT | jitsi you say |
| 14:54.39 | P-NuT | ok then, well I think I have enough info to be pointed in the right dorection. |
| 14:54.42 | P-NuT | Thanks guys! |
| 14:54.47 | anonymouz666 | leifmadsen: of audio core? :) |
| 14:54.47 | irroot | anonymouz666 leifmadsen wait app queues frames and delivers them latter |
| 14:55.02 | leifmadsen | anonymouz666: perhaps? I don't know, throwing out random wild ideas |
| 14:55.06 | anonymouz666 | the machine is under heavy load |
| 14:55.25 | leifmadsen | sounds like processing delay perhaps then |
| 14:55.39 | anonymouz666 | the load is about 6.00 |
| 14:55.44 | leifmadsen | ya that'd do it then |
| 14:55.55 | leifmadsen | lower the load, and I bet it works again |
| 14:56.08 | leifmadsen | cpu's can't keep up with that much processing it sounds like |
| 14:56.09 | WIMPy | What kind of load? |
| 14:56.21 | P-NuT | thanks all! |
| 14:56.32 | WIMPy | You can have a load of 6 at 90% idle. That doesn't mean much. |
| 14:56.37 | *** part/#asterisk P-NuT (~P-NuT@188-223-86-64.zone14.bethere.co.uk) |
| 14:56.47 | Chainsaw | anonymouz666: My rule of thumb is... the moment your load average is higher then the amount of cores in your box, you're in trouble. |
| 14:56.57 | Chainsaw | anonymouz666: So unless you have more than 6 CPU cores in that box, it seems overloaded to me. |
| 14:57.27 | anonymouz666 | yes, I have 8 cores |
| 14:57.44 | anonymouz666 | and yes, it happens sometimes |
| 14:57.48 | Chainsaw | anonymouz666: Then it's high but not unreasonably so. |
| 14:57.52 | anonymouz666 | it sounds really a CPU bottleneck |
| 14:57.58 | *** join/#asterisk delki8 (~delki8@189.5.136.31) |
| 14:58.20 | irroot | greets Wimpy having a problem with only one channel per port on mISDN been used so on 4 ports only 4 calls not 8 you aware of this i have patched it <- when using round robin |
| 14:58.20 | WIMPy | Why do you guys think the CPU has much to do? |
| 14:58.47 | anonymouz666 | another bizarre thing... app_queue deliveres two different calls to the same user in the same second ! |
| 14:59.20 | WIMPy | irroot: No, I can not remember having seen such a thing happen, but I've never used groups in the chan_misdn times. |
| 14:59.24 | irroot | anonymouz666 you have ringinuse on ? |
| 14:59.35 | anonymouz666 | irroot: no sir. |
| 14:59.56 | anonymouz666 | it is off. |
| 14:59.59 | irroot | i have seen this when the state change does not "register" |
| 15:00.14 | Katty | dances with Qwell |
| 15:00.25 | Qwell | stumbles about |
| 15:00.34 | irroot | Katty whats on the jukebox |
| 15:00.39 | Katty | mr saxobeat |
| 15:00.46 | anonymouz666 | irroot: do you live with that or fixed somehow? |
| 15:00.52 | Chainsaw | Katty: Good song :) |
| 15:00.55 | Katty | yesh. |
| 15:01.44 | anonymouz666 | irroot: but how that could be state change if two calls are delivered in the same sec? |
| 15:02.01 | anonymouz666 | you mean the microsecs are different |
| 15:02.02 | irroot | anonymouz666 have put a ignorebusy option in you can look at one of the things it does is "double" checks the state |
| 15:02.53 | irroot | anonymouz666 had a customer that wanted only 1 call at time and had to be aggresive with it sometimes more than one would get handled |
| 15:03.39 | anonymouz666 | in that customer, people are using x-lite if two calls are sent to the same agent, I'll have trouble :P |
| 15:04.31 | anonymouz666 | they love AC button, but they love even more the DND button :D |
| 15:05.07 | irroot | all customers love the DND button |
| 15:05.13 | anonymouz666 | put a manager event when ring no answer returns 0ms, and now they all are on my list :P |
| 15:05.42 | anonymouz666 | 1.8 already have that, i put that on 1.4 version |
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| 15:10.26 | Jacke | how do i reverse polarisation on a fxo interface from within a module? |
| 15:15.52 | anonymouz666 | WIMPy: sorry, I read your message now. you are right, the load is 6, but the CPU has a high % idle |
| 15:16.25 | anonymouz666 | I really don't understand this counters, never know when I am hitting a bottleneck |
| 15:17.12 | WIMPy | anonymouz666: That might be an I/O issue then, like the harddisk. |
| 15:17.38 | WIMPy | Do you have excessive logging enabled? |
| 15:17.50 | anonymouz666 | excessive calls being recorded |
| 15:18.47 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 15:18.48 | WIMPy | Is your RAM eaten away by buffers? |
| 15:19.14 | anonymouz666 | Mem: 8173292k total, 8109532k used |
| 15:19.16 | anonymouz666 | it seems |
| 15:19.53 | WIMPy | Maybe you try to write data faster than possible. That can cause quite interesting issues. |
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| 15:20.23 | anonymouz666 | I have seen many interesting issues in this machine |
| 15:20.33 | WIMPy | Including extreme delays in other places. |
| 15:21.05 | *** join/#asterisk gxdssoft (~gxdssoft@201.230.197.18) |
| 15:21.12 | WIMPy | I guess you should rething your storage concept. |
| 15:21.46 | *** join/#asterisk Gugge (~gugge@46.30.120.1) |
| 15:21.47 | anonymouz666 | the bo (io) column from vmstat seems high |
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| 15:39.26 | Qwell | file: and this! |
| 15:40.28 | *** join/#asterisk Gugge (~gugge@46.30.120.1) |
| 15:40.41 | file | Qwell, C++! |
| 15:40.52 | Qwell | no, u |
| 15:40.56 | LemensTS | What cpu do ya think I should run on 6 incoming analog lines with 10 IP Phones. Probably 4 concurrent calls. |
| 15:41.18 | LemensTS | Intel for sure, but they have changed the chips so much lately im confused |
| 15:41.41 | p3nguin | I'd grab an old 1.8GHz P4 out of the discard pile. |
| 15:41.54 | Qwell | LemensTS: anything |
| 15:42.02 | Qwell | a P3 could handle that |
| 15:42.43 | LemensTS | Awesome. I can stick to 775 than. |
| 15:42.53 | WIMPy | P4s are good for heating the office. |
| 15:43.12 | p3nguin | I have a BRAND NEW 13-year-old Gateway with AMD Athlon 800 (Slot A) processor that I have been waiting to use for something. |
| 15:43.55 | p3nguin | I think it's 13 years old, anyway. I'd have to go to the shed to double check the date. |
| 15:45.30 | coppice | 11 to 13 years sounds about the right range |
| 15:45.43 | p3nguin | Maybe it's only 11 or 12 now that I think of it... it came with Windows Me on it. |
| 15:45.50 | Qwell | coppice: context! |
| 15:46.05 | coppice | Qwell: protext! |
| 15:46.10 | p3nguin | hah |
| 15:46.35 | p3nguin | Windows Me was released in '99, wasn't it? |
| 15:47.08 | coppice | well, windows 98 and 98 second edition fitted in somewhere before it |
| 15:47.35 | Qwell | p3nguin: 2000 |
| 15:48.07 | p3nguin | I couldn't believe that computer being that old was in mint condition when I got it just a few years ago. |
| 15:48.15 | coppice | I bought a WinME notebook in early 2000. I put RedHat on it the day I bought it, but it came with WinME |
| 15:48.49 | leifmadsen | my MacBook Pro has been operating with Ubuntu and Windows for the better part of 2 years :) |
| 15:49.35 | p3nguin | I think that's called a Double Barfer. |
| 15:49.58 | p3nguin | coppice: RH 6? |
| 15:50.08 | coppice | 5.1 I think |
| 15:50.50 | coppice | It came on a stack of pressed CDs. that was pre broadband |
| 15:51.34 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-144-115.dsl.stlsmo.sbcglobal.net) |
| 15:51.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:51.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:53.06 | p3nguin | I was thinking I used 6.0 back around 2000-1 or so, but I suppose it could have been 5.x. I remember a guy downloading it on his cable modem for me, and I was really impressed by the speed he got. I also felt like it was piracy because I didn't know it could be downloaded and used for free. |
| 15:56.01 | *** join/#asterisk drzed (~drzed@synflood.homelinux.org) |
| 15:56.06 | drzed | hi there! |
| 15:56.07 | coppice | 6.0 sucked. they kept screwing up anything with a 0 at the end. 6.2 was excellent |
| 15:57.09 | drzed | pickup problem: try to pickup w/ *8 works but not with *830 (extension 30) |
| 15:57.11 | p3nguin | If I looked enough, I could probably find that CD he burned for me. |
| 15:57.51 | drzed | my extensions.conf looks like this: exten => _*8XX,1,Set(nst=${EXTEN:2}) |
| 15:57.52 | drzed | exten => _*8XX,n,Pickup(${nst}) |
| 15:59.12 | p3nguin | I also remember before the guy downloaded and burned whatever version it was, a guy at the computer shop gave me an unopened RH box set of another version, but I'd have to look for that to find what version it was, as well. Those times are kind of blurry for me today. |
| 16:00.50 | p3nguin | drzed: Is it possible that your features *8 is catching the dialed number before it gets to your extensions.conf entry? |
| 16:02.48 | p3nguin | And does extension 30 have someone calling it? |
| 16:02.56 | drzed | if i read the log corrently, yes: -- Executing [*830@default:2] Verbose("SIP/34-084e9ee0", "1|34 will Anruf von 30 holen") in new stack |
| 16:02.58 | p3nguin | And is extension 30 in the same context? |
| 16:03.00 | drzed | <PROTECTED> |
| 16:03.02 | drzed | <PROTECTED> |
| 16:03.05 | drzed | <PROTECTED> |
| 16:03.46 | drzed | yess it is beeing called && it is in the same context |
| 16:03.48 | p3nguin | You have a phone by the name of 34? And its extension is 30? Crazy. |
| 16:04.50 | drzed | i've got 3 phones: 30, 34, 35; im calling 35->30 and try to pickup the call up with 34 |
| 16:05.08 | p3nguin | That's a terrible naming convention, just so you know. |
| 16:05.59 | p3nguin | Nonetheless, if someone is calling extension 30, and you try to Pickup(30), I would expect good results. |
| 16:07.04 | p3nguin | I'm not good with German translation, so I don't know what the message above says, but I know it says you're trying to pickup extension 30. |
| 16:07.38 | drzed | hm entering Pinckup(30) in the * console does not work |
| 16:07.52 | p3nguin | Of course it doesn't. |
| 16:08.03 | p3nguin | Pickup() is a dial plan application, not a console command. |
| 16:08.53 | drzed | ah ok, i tought i should try to enter Pickup(30) somewhere ... |
| 16:09.22 | p3nguin | You've written it correctly in extensions.conf, as far as I can tell. |
| 16:10.07 | drzed | sry about the german: translated "34 will Anruf von 30 holen" => "34 tries to catch call from 30" |
| 16:10.44 | drzed | what about the "No channel found ... " message? |
| 16:10.49 | drzed | is that ok? |
| 16:11.21 | p3nguin | No, that's not a good thing. I believe that you are doing it correctly. Maybe someone else will help determine why the channel is not found. |
| 16:11.33 | drzed | the next log line was: == Spawn extension (default, *830, 3) exited non-zero on 'SIP/34-084e9ee0' |
| 16:12.03 | p3nguin | That line is okay and normal when the extension ends. |
| 16:12.29 | p3nguin | Oh, wait... |
| 16:12.30 | drzed | i would guess that "non-zero" is not ok .. |
| 16:12.39 | p3nguin | You're using context default? |
| 16:13.04 | p3nguin | Bad bad bad. Your extension 30 is in context lokal (local). |
| 16:13.07 | *** part/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 16:14.04 | p3nguin | Seeing the extension exit non-zero is okay when there is nothing else to do. It is normal. |
| 16:14.08 | drzed | how/wherer do i put this extension in the local context |
| 16:14.38 | p3nguin | cut/paste? |
| 16:14.49 | p3nguin | I don't know how you do it. |
| 16:15.05 | p3nguin | I copy, paste, then delete the original. |
| 16:15.24 | drzed | http://nopaste.voric.com/paste.php?f=z4qu45 |
| 16:16.20 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 16:16.21 | p3nguin | Are you using asterisk 1.0? |
| 16:16.26 | drzed | nope 1.4 |
| 16:16.39 | p3nguin | How the heck... |
| 16:17.01 | p3nguin | 1.4 accepts and runs Dial,SIP/${EXTEN}|55|Ttr ? |
| 16:17.19 | p3nguin | Not to mention, those are terrible dial options. |
| 16:17.30 | drzed | guess so ... hm maybe i read an old manual .. |
| 16:17.33 | WIMPy | Yes, it's that old :-) |
| 16:17.50 | drzed | any suggenstion on that? |
| 16:18.07 | p3nguin | Dial(SIP/${EXTEN},55) |
| 16:18.33 | p3nguin | or Dial(SIP/${EXTEN},55,t) if your phone does not have a transfer button on it. |
| 16:19.09 | *** join/#asterisk irroot (~irroot@197.174.36.133) |
| 16:19.28 | drzed | ok, changed it but it did not help |
| 16:19.42 | p3nguin | Did you remember to run "dialplan reload" after you saved the changes? |
| 16:20.04 | drzed | i just ran "reload" is that also ok? |
| 16:20.12 | p3nguin | stabs |
| 16:20.33 | p3nguin | big knife |
| 16:21.22 | p3nguin | Hiebmesser |
| 16:21.30 | drzed | im, would say that my 3X phones are in the local context and not in the default? |
| 16:21.46 | p3nguin | It looks like they are in local, yes. |
| 16:22.32 | p3nguin | When you hear the phone ringing, run "core show channels" right before you dial *830 on your phone. |
| 16:22.51 | p3nguin | I'd like to see what channels are active when you are trying to pickup a call. |
| 16:22.52 | WIMPy | Don;t you need the context of the incomming call? |
| 16:23.20 | p3nguin | I think that might depend on where the call came from as to whether it is wrong or right. |
| 16:23.41 | p3nguin | If the call comes from another phone with context=local, I would expect it to work. |
| 16:24.12 | WIMPy | Yes, but that's most probably not the use case. |
| 16:24.15 | drzed | kChannel Location State Application(Data) |
| 16:24.18 | drzed | SIP/30-084e5ef0 30@default:1 Ringing AppDial((Outgoing Line)) |
| 16:24.21 | drzed | SIP/35-084e4960 30@default:2 Ring Dial(SIP/30|55) |
| 16:24.22 | p3nguin | If it comes from context from-pstn and has an include for local, then I'd guess you'd have to Pickup(30@from-pstn). |
| 16:24.24 | drzed | 2 active channels |
| 16:24.27 | drzed | 1 active cal |
| 16:24.52 | p3nguin | So you need to fix the problem with using the default context, or Pickup(30@default) instead. |
| 16:26.19 | p3nguin | großen Messer |
| 16:27.06 | p3nguin | Bah, I'm ready for lunch. |
| 16:27.11 | drzed | im sounds like i messed something up? |
| 16:28.04 | drzed | changed extension to: exten => _*8XX,n,Pickup(${nst}@default) |
| 16:28.26 | drzed | does not work either. is it ok to include all contexts in default? |
| 16:40.37 | *** join/#asterisk Unbeerable (~vitek@homer.tomgate.net) |
| 16:41.57 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 16:42.16 | Unbeerable | is there any plans to add binary packages for centos6? http://packages.asterisk.org/centos/ still has rpms for 4 and 5 versions only :( |
| 16:43.20 | p3nguin | drzed: No, do not start including contexts all willy-nilly. |
| 16:43.48 | leifmadsen | pabelanger: ^^^ |
| 16:47.50 | anonymouz666 | Unbeerable: good question. I just started using CentOS 6 also |
| 16:48.09 | anonymouz666 | CentOS 5 uses a kernel 5 years old only |
| 16:48.13 | *** join/#asterisk bitbandit (~taggmcd@c-98-202-116-168.hsd1.ut.comcast.net) |
| 16:48.15 | p3nguin | I think I hear some Bagel Bites calling my name. |
| 16:48.17 | anonymouz666 | new devices loves that |
| 16:49.05 | Unbeerable | I know there are another repos with asterisk already built for el6, but I'd prefer to install this package directly from vendor site |
| 16:49.35 | pabelanger | Unbeerable: Unbeerable: nothing yet, I believe Qwell will create them eventually |
| 16:49.48 | pabelanger | patches welcome |
| 16:49.50 | WIMPy | drzed: Preferrably you shouldn't use default. |
| 16:50.36 | Unbeerable | pabelanger, so I may contribute something to move things on? |
| 16:50.58 | Qwell | There's nothing to contribute, really. |
| 16:51.10 | Qwell | It's a matter of finding time to actually setup a build VM. |
| 16:51.41 | *** join/#asterisk pdtpatr1ck (~pdtpatric@mainstwan.farheap.com) |
| 16:51.58 | pabelanger | does redhat have chroot or something similar? |
| 16:54.18 | irroot | pabelanger that is should have indeed used it often and cant remember needing to install it |
| 16:55.21 | pdtpatr1ck | Question .. all of a sudden my queues started showing this |
| 16:55.22 | pdtpatr1ck | <PROTECTED> |
| 16:55.46 | pdtpatr1ck | that's from the CLI when i run "queue show <queuename>" |
| 16:55.49 | *** join/#asterisk Frem_ (~jamesgeck@64.207.3.161) |
| 16:55.57 | pdtpatr1ck | i've looked at this page: |
| 16:55.58 | pdtpatr1ck | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueStatus |
| 16:56.01 | p3nguin | You just started using Local channels as members, didn't you? |
| 16:56.06 | irroot | pdtpatr1ck that seems right |
| 16:56.18 | pdtpatr1ck | irroot, what seems right? |
| 16:56.37 | irroot | channel penalty rt state / state |
| 16:56.39 | pdtpatr1ck | p3nguin, It has been working for two years and just yesterday crapped out |
| 16:56.53 | irroot | what is the problem you having |
| 16:56.55 | p3nguin | In modules.conf, preload pbx_config and then preload chan_local, then restart asterisk. |
| 16:57.14 | Qwell | pabelanger: we actually build on the distro we make the builds for. |
| 16:57.33 | pdtpatr1ck | irroot, i was just trying to figure out why'd it go invalid so i'll understand the root cause. |
| 16:57.54 | irroot | ah the invalid is the extension state |
| 16:57.58 | pdtpatr1ck | p3nguin, an reasons you've learned from experience? so i can know for future as well. Meanwhile I will do as you just mentioned above |
| 16:58.02 | p3nguin | I've experienced it and I solved it with the instructions given. |
| 16:58.59 | irroot | pdtpatr1ck or channel state you can use device states to manage it |
| 16:59.30 | pabelanger | Qwell: ya, with the Debian and Ubuntu packages the base OS is ubuntu lucid, then we setup chroots for each other OS we want packages for. pbuilder kinda kicks ass for that |
| 17:00.52 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 17:00.53 | pdtpatr1ck | penguin, so add: load => pbx_config.so and then below it add load => chan_local.so .. then restart asterisk ? |
| 17:02.17 | Unbeerable | pabelanger, there are a lot of changes between el5 and el6, like between fedora core 6 and fedora core 11 or 12, including changes in rpm itself. So I think the chances to build el6 packages in el5 environment are very small/ |
| 17:02.42 | p3nguin | pdtpatr1ck: preload, not load. |
| 17:02.58 | p3nguin | preload => pbx_config.so |
| 17:02.59 | p3nguin | preload => chan_local.so |
| 17:03.11 | pdtpatr1ck | oh okay .. sorry. Thanks |
| 17:03.29 | p3nguin | pdtpatr1ck: If you don't want to restart asterisk now, you can unload app_queue.so and then load it again. |
| 17:03.54 | pdtpatr1ck | after adding preload of course right ? |
| 17:04.06 | p3nguin | I think that will straiten it out for the currently running instance. |
| 17:04.14 | p3nguin | They are already loaded this time. |
| 17:04.30 | p3nguin | Did I really just write straiten? |
| 17:04.36 | pdtpatr1ck | :) |
| 17:04.42 | p3nguin | wtf is wrong with me? |
| 17:05.44 | pabelanger | Unbeerable: right, but with chroot you can specific any version of a OS to use. So in the case of the Debian packages, the host OS is Ubuntu Lucid, then we build chroot for Debian squeeze and wheezy. We then build Asterisk with the binaries from the chroot. |
| 17:06.07 | pabelanger | So when a new version of the OS does come out, we don't need to create a new VM, but just run chroot from the command line |
| 17:06.17 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:06.48 | Qwell | pabelanger: which gcc does it use to build? The host gcc? |
| 17:07.39 | *** join/#asterisk trumee (~trumee@cpc2-cmbg7-0-0-cust855.5-4.cable.virginmedia.com) |
| 17:07.42 | pabelanger | Qwell: the version from the chroot |
| 17:08.49 | Unbeerable | I suppose creating a VM is much easier than installing all required staff into the chroot :) |
| 17:08.54 | *** join/#asterisk n4n4k1 (~n4n4k1@209.92.35.234) |
| 17:09.00 | Qwell | much |
| 17:09.49 | n4n4k1 | Have a question regarding calls dropping instantly when trasnferring directly to voicemail using I symphony latest and asterisk 1.6.2.13 |
| 17:10.46 | pabelanger | It might take longer, but the build will not be tied to a specific host computer. Automation FTW :D |
| 17:11.41 | Unbeerable | Qwell, well, I'd like to wish you to find a time and make some people happy :) |
| 17:11.49 | Qwell | Unbeerable: noted |
| 17:16.12 | *** join/#asterisk irroot (~irroot@197.171.161.170) |
| 17:16.47 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 17:18.57 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:19.28 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 17:19.40 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279412163.dsl.bell.ca) |
| 17:21.10 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-lsrexpuhcqdarbzr) |
| 17:22.18 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 17:25.33 | dijib | p3nguin, hows your fax solution work? does it auto detect? will it work beside my press 1 for option? |
| 17:26.55 | p3nguin | My fax stuff does not do any detection at all. It just requires a call to be sent to the fax extension. |
| 17:27.09 | dijib | ahhhh |
| 17:27.19 | dijib | then i need to change it to detect... possible? |
| 17:27.32 | p3nguin | You don't need to change anything of mine. |
| 17:27.40 | p3nguin | Detection would be done elsewhere. |
| 17:27.48 | dijib | ok ok i follow |
| 17:27.59 | dijib | so if fax sent to fex extension. |
| 17:28.52 | p3nguin | It doesn't matter how the fax ends up in the fax-in context on the fax extension, as long as it gets there. |
| 17:29.09 | Katty | can someone tell me if https://www.copi-rite.com/ is working? |
| 17:29.14 | dijib | crap ive got to go to the grocery store. |
| 17:29.17 | dijib | back in 30min |
| 17:29.17 | p3nguin | If you have some type of detector that sends it there, or if you use a Goto, or whatever. |
| 17:29.25 | p3nguin | katty: downforme.com |
| 17:29.33 | Katty | ty |
| 17:29.42 | p3nguin | Well, maybe that's wrong. |
| 17:29.52 | p3nguin | downforeveryone.com? |
| 17:29.58 | p3nguin | Yeah, that's it. |
| 17:34.54 | _Corey_ | Katty: I get someone's SugarCRM |
| 17:36.18 | *** join/#asterisk wolf1161 (~wolf@c-67-168-115-132.hsd1.wa.comcast.net) |
| 17:36.31 | p3nguin | DNS problems? |
| 17:36.40 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:37.11 | Katty | yeah |
| 17:37.17 | Katty | just internal dns fu-bared |
| 17:37.23 | *** join/#asterisk tehrabbitt-1 (~ryana@unaffiliated/tehrabbitt) |
| 17:37.25 | Katty | or... fu-barfed |
| 17:37.31 | Katty | i like that, fu-barfed. |
| 17:37.33 | Katty | i'mma keep that one |
| 17:37.35 | p3nguin | The person that wrote the front page needs to learn some English and try writing it again. |
| 17:37.45 | tehrabbitt-1 | hey everyone |
| 17:37.49 | Katty | the front page for sugarcrm? |
| 17:38.06 | p3nguin | No, the front page of theritegroup.com. |
| 17:38.11 | Katty | oh ha |
| 17:38.15 | Katty | yeah that website is a joke |
| 17:38.16 | _Corey_ | Katty: Login page for their sugarcrm |
| 17:38.24 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 17:38.25 | Katty | but whatever, not my problem ;) |
| 17:38.26 | tehrabbitt-1 | got a new job working at a place that gets alot of recycled VoIP stuff... cisco, avaya,etc... if anyone is looking for stuff let me know |
| 17:38.30 | fullstop | wow, am I full. |
| 17:38.33 | p3nguin | Among other mistakes, it says you only have one client. |
| 17:38.54 | p3nguin | will meet or exceed all our client's conditions of satisfaction |
| 17:38.56 | p3nguin | one client |
| 17:39.29 | Katty | lol that's funny |
| 17:39.56 | p3nguin | Okay, maybe they can leave that and I'll adjust my interpretation to the meaning of each and every client's conditions. |
| 17:39.57 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 17:41.10 | p3nguin | I can go either way with it, I suppose. No emails will be sent on the matter. :) |
| 17:42.21 | tehrabbitt-1 | i also came across one of these in the warehouse too that we're lookign to sell if anyone is interested: http://www.datacomtools.com/Manuals/ts22alo-lo.pdf |
| 17:42.54 | p3nguin | Here's another good one: Customized video surveillance and burglary systems to watch your business when your not there. |
| 17:43.24 | p3nguin | That one deserves an email. |
| 17:43.44 | p3nguin | Who is in charge of that site? |
| 17:43.53 | Katty | hehehehe i just found a definition in my CCNT book that i approve of |
| 17:44.09 | Qwell | Katty: is the definition "Qwell: See; awesome"? |
| 17:44.17 | DrDigital | with the T22P is there a way when they pick the hand set up, it answers? auto answer seems to answer with them on speaker phone and never ringing |
| 17:44.19 | Katty | Bursty - network traffic that is not constant, but requires a lot of bandwidth on demand. |
| 17:44.32 | Katty | now that's MY kind of defintiion |
| 17:44.48 | Katty | Qwell: i'll scribble that one down in the back of the book :> |
| 17:44.58 | Qwell | It should already be in the book. :( |
| 17:45.01 | p3nguin | As in, "I must say, you are very bursty today?" |
| 17:45.28 | Katty | YES |
| 17:45.47 | p3nguin | fullstop: What'd you have? |
| 17:49.30 | p3nguin | I guess trcole3 and/or laurencole might be the recipients of my rant. |
| 17:49.47 | p3nguin | (if I get around to sending it) |
| 17:53.46 | fullstop | bbq chicken sandwich |
| 17:54.27 | p3nguin | shredded chicken or whole chicken breast? |
| 17:54.54 | fullstop | shredded |
| 17:55.01 | fullstop | fresh roll |
| 17:55.04 | fullstop | a salad |
| 17:55.16 | fullstop | and a slice of lemon pie. |
| 17:55.20 | p3nguin | Sounds better than my bagel bites. If they hadn't already been ingested, I'd trade. |
| 17:56.02 | fullstop | Due to hot weather one weekend over the summer, I lost my sourdough yeast culture. :-/ |
| 17:56.23 | p3nguin | OH NO! |
| 17:56.41 | p3nguin | Maybe you can find someone else and get a new starter. |
| 17:57.04 | p3nguin | Sourdough is awesome. It's bothersome, but it's so yummy! |
| 17:57.25 | talntid | I'll take some :P |
| 17:58.16 | fullstop | I need to start a new one, yes. |
| 17:58.26 | fullstop | I usually make english muffins with it. |
| 17:59.19 | fullstop | http://a6.sphotos.ak.fbcdn.net/hphotos-ak-snc6/197014_588821454959_3805250_34103988_5204087_n.jpg |
| 17:59.42 | talntid | damnit, you guys are making me hungry |
| 17:59.48 | *** join/#asterisk gmcharlt (~gmchart`@pdpc/support/active/gmcharlt) |
| 18:00.28 | fullstop | talntid: Those have cranberries in them, too. |
| 18:00.51 | fullstop | but my girls like them more when I put chocolate chips in them. |
| 18:02.07 | talntid | not helping..... |
| 18:02.13 | talntid | jerk. |
| 18:02.14 | talntid | :) |
| 18:02.26 | fullstop | O:-) |
| 18:02.43 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:03.17 | leifmadsen | http://leifmadsen.com/sites/default/files/muffin.jpg |
| 18:03.35 | leifmadsen | fullstop: blueberry muffin ^^^ |
| 18:03.45 | *** join/#asterisk navaismo (~navaismo@187.170.1.109) |
| 18:03.54 | fullstop | leifmadsen: :-9 |
| 18:04.11 | fullstop | However, blueberry muffins are quick to make compared to English muffins. |
| 18:04.26 | leifmadsen | that makes them even better! |
| 18:04.35 | p3nguin | Is ilbc included in the latest 1.4s? I don't see the script under contribs anymore. |
| 18:04.39 | leifmadsen | I buy my english muffins, then put egg, ham, and cheese on mine |
| 18:04.51 | talntid | must....resist.....clicking.... picture.... |
| 18:04.53 | fullstop | When you make them with sourdough yeast, you have to let them rise for ~10 hours before you even start working and cutting the dough. |
| 18:04.57 | leifmadsen | p3nguin: thought it was under a subdir of contrib |
| 18:05.17 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:05.23 | p3nguin | I thought it would be under contribs/scripts/codecs/ but the codecs dir does not exist for me. |
| 18:05.37 | leifmadsen | odd |
| 18:05.38 | p3nguin | But I do have /usr/lib/asterisk/modules/format_ilbc.so already installed. |
| 18:05.47 | leifmadsen | format_ilbc will build yes |
| 18:05.48 | p3nguin | So it confused me. |
| 18:05.53 | leifmadsen | it's codec_ilbc that's the trick |
| 18:05.58 | p3nguin | oh |
| 18:06.04 | p3nguin | I'll look a little harder. |
| 18:06.04 | leifmadsen | that's what the script is for |
| 18:06.29 | p3nguin | Got it. contrib/scripts/get_ilbc_source.sh |
| 18:07.36 | p3nguin | Too many different paths got me confused as to what should be where. |
| 18:08.13 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:09.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:11.01 | wolf1161 | hi everyone I am fairly new to asterisk. I am running freepbx 2.9.0.7 and I am having an issue |
| 18:11.10 | p3nguin | ~freepbx |
| 18:11.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 18:11.17 | navaismo | go to #freepbx |
| 18:11.53 | wolf1161 | ok thank you |
| 18:14.36 | *** join/#asterisk CaptWho (~Capt@unaffiliated/captwho) |
| 18:19.21 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-096.mobile.uci.edu) |
| 18:24.55 | Frem_ | I'm using asterisknow, and I have the asterisk16 package installed. Is there an upgrade procedure to the asterisk18 packages? |
| 18:27.31 | Katty | Qwell: guess who's in a good mood :>>>> |
| 18:27.41 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:29.09 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279412163.dsl.bell.ca) |
| 18:32.25 | *** join/#asterisk P-NuT (~P-NuT@5ad48b0d.bb.sky.com) |
| 18:32.55 | P-NuT | Hi all, does anyone know if video conferencing with more than 2 people is possible with asterisk? |
| 18:33.41 | navaismo | in some where i read with asterisk 10 beta it possible. im right? |
| 18:34.39 | P-NuT | asterisk is 1.8. version 10 you are talking about? |
| 18:34.46 | dijib | meetme P-NuT |
| 18:36.13 | p3nguin | MeetMe does video? |
| 18:36.20 | _Corey_ | P-NuT: It's been a while since I've used it myself but I know some people who use AppConference and are pretty happy with it. |
| 18:36.22 | malcolmd | confbridge does video |
| 18:36.42 | malcolmd | meetme does not |
| 18:36.49 | malcolmd | confbridge in asterisk 10, that is |
| 18:37.03 | dijib | im using 1.8.5 P-NuT |
| 18:37.17 | dijib | oh video confrence... nevermind |
| 18:37.24 | dijib | aparenty i dont read too well p3nguin |
| 18:37.29 | dijib | or write. |
| 18:37.32 | dijib | apparently |
| 18:37.54 | p3nguin | I'd still love to know why a new voicemail is named msg0000 and the message that gets emailed to me shows msg0001. All messages being emailed are real number +1. |
| 18:42.42 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 18:43.08 | *** join/#asterisk samandiriel (~samandiri@S0106c03f0eb613f5.ed.shawcable.net) |
| 18:43.15 | *** part/#asterisk n4n4k1 (~n4n4k1@209.92.35.234) |
| 18:45.40 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
| 18:46.59 | Kobaz | what's a good way to check for proper ground |
| 18:48.49 | p3nguin | ground of what? |
| 18:49.00 | carrar | look at it? |
| 18:49.04 | p3nguin | rack, chassis or something? |
| 18:50.17 | Qwell | Katty: is it me? |
| 18:50.26 | Kobaz | telco ground |
| 18:50.40 | Kobaz | like a wire coming out the wall that's supposed to be the ground |
| 18:50.40 | carrar | make sure you have 3 9' 1/2 inch steal poles in the ground about 3' apart in a triangle shape all wired together and then wired to your rack with heavy duity wire |
| 18:50.54 | carrar | that should work |
| 18:50.57 | Kobaz | how do i make sure it's really a ground |
| 18:51.05 | carrar | hire a electrician |
| 18:51.09 | Kobaz | because i keep getting fried ports on my fxos |
| 18:51.15 | fullstop | multimeter? |
| 18:51.21 | Kobaz | yeah i have a multimeter |
| 18:51.28 | Kobaz | i;ve just never tested a ground |
| 18:52.07 | p3nguin | I don't think phone jacks have a ground. |
| 18:52.15 | Kobaz | no they dont |
| 18:52.40 | p3nguin | So what are you trying to find ground for? |
| 18:53.06 | p3nguin | The NID should have a ground, but you said from the wall. |
| 18:53.07 | Kobaz | confirm that my telco secondary voltage protector is actually grounded |
| 18:53.21 | p3nguin | oh, lightning arrestors? |
| 18:53.48 | Kobaz | yeah |
| 18:54.11 | carrar | use both hands and start touching everything, if you feel something or it kills you, something is wrong |
| 18:54.12 | Kobaz | oh, there's ground wires coming into the dmarcs |
| 18:54.20 | Kobaz | but i dont have a tool to open it |
| 18:54.23 | Kobaz | maybe i do |
| 18:54.32 | p3nguin | If I wanted to test if a ground wire is doing its job, I'd use my Ohm meter and test between the device that is to be grouned and a known ground. If there is any resistance, your ground is not good on that device. |
| 18:54.47 | Kobaz | mm, k |
| 18:55.26 | p3nguin | It should show 0.000 to indicate a dead short, which is provided by the ground wire. |
| 18:55.51 | carrar | everything has resistance |
| 18:56.06 | p3nguin | A good ground wire won't have any to be shown on a meter. |
| 18:56.10 | navaismo | P-NuT i say asterisk 10 beta |
| 18:56.19 | carrar | will have a very tiny amount |
| 18:56.20 | p3nguin | It would have to be extremely long. |
| 18:56.34 | carrar | depending on the sensitivyt of the meter |
| 18:57.08 | dijib | you will have the resistance of the wire to the ground water. |
| 18:57.17 | p3nguin | A good ground wire, of, say, 12 awg, which is maybe six feet long... I would expect to see a dead short. |
| 18:57.22 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:58.01 | dijib | now how are you guys with automotive issues, ie gm with a starter problem |
| 18:58.04 | dijib | :) |
| 18:58.05 | p3nguin | If you are measuring in zillionths of an Ohm, you might show some resistance. |
| 18:58.11 | p3nguin | Describe the problem. |
| 18:58.12 | dijib | or am i in the wrong channel |
| 18:58.20 | carrar | no this is the r ight channel |
| 18:58.41 | p3nguin | #asterisk: telephony, food, and cars |
| 18:58.56 | carrar | .. and Katty |
| 18:59.01 | Kobaz | okay, opened the dmarc, testing the ground |
| 18:59.23 | dijib | lol |
| 18:59.38 | carrar | (sigh)... waiting for news of death by accidental electric shock |
| 19:00.03 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 19:00.15 | carrar | http://www.youtube.com/watch?v=BtQtRGI0F2Q |
| 19:00.19 | dijib | new starter installed yesterday. no shims in old one for mounting. new starter trys to start but just clicks as if its not pushing the gear all the way out and its hitting the flywheel |
| 19:00.22 | luke-jr | what's the standard way to run a script at the end of all calls? |
| 19:00.51 | Qwell | dijib: Show us the starter dialplan. |
| 19:00.56 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 19:01.07 | p3nguin | dijib: Did you use the gage provided to check the distance between the starter drive and the ring gear? |
| 19:01.28 | dijib | exten => battery,n,start() |
| 19:01.31 | p3nguin | Does it "whir" or just click? |
| 19:01.32 | dijib | no guage provided? |
| 19:01.35 | dijib | click |
| 19:01.44 | p3nguin | Sounds like a bad connection. |
| 19:02.01 | carrar | try a different codec |
| 19:02.20 | dijib | well could be a low battery. when i turn ignition it tries to start but then headlights go off, |
| 19:02.26 | dijib | im thinking codec issue also |
| 19:02.31 | p3nguin | Check your cables on the battery, and also check the positive where it attaches on the starter. |
| 19:02.41 | Kobaz | .6 ohms |
| 19:02.45 | dijib | p3nguin, what distance guage? |
| 19:03.11 | Kobaz | that seems high |
| 19:03.50 | luke-jr | ⦠|
| 19:03.50 | p3nguin | dijib: It's a little piece of metal rod. It is used to verify the right distance for the starter drive to engage the ring gear correctly. Too little space and it'll grind against the ring gear; too much space and it'll grind the teeth off. |
| 19:04.18 | p3nguin | .6 Ohms... the ground wire had better be 22 gage and 20 feet long. |
| 19:04.33 | p3nguin | If it's a proper ground, there should be much less than .6 Ohms. |
| 19:05.42 | p3nguin | dijib: If it was a spacing problem, you'd hear either grinding or a whir. Just a clicking indicates to me there is a lack of power -- low battery or bad connection. |
| 19:06.00 | Kobaz | It's probably not a proper ground |
| 19:06.19 | dijib | ok p3nguin your in line with my fathers answer... and the battery has been on the charger for 2 hours now |
| 19:06.27 | Kobaz | i was here when the verizon guy put in the dmarcs forever ago |
| 19:06.30 | dijib | good old suburban |
| 19:06.36 | p3nguin | What year? |
| 19:06.38 | dijib | 99 |
| 19:06.43 | Kobaz | he just found some wire in the ceiling and said "I think this is the ground" |
| 19:06.47 | dijib | she's a beast |
| 19:07.20 | p3nguin | 305 Vortec? |
| 19:07.22 | dijib | 350 |
| 19:07.28 | dijib | 5.7L |
| 19:07.28 | p3nguin | even better. |
| 19:07.38 | p3nguin | 4x4? |
| 19:07.43 | dijib | better on gas then my old 98 jimmy 4.3 if you believe that |
| 19:07.47 | dijib | yes 4x4 |
| 19:07.59 | dijib | but the 1:3.43 hear ratio |
| 19:08.02 | dijib | gear |
| 19:08.07 | p3nguin | I believe it. The 4.3 is a gas hog. |
| 19:08.13 | dijib | they have 343, 373, 410 |
| 19:08.18 | dijib | yes it was |
| 19:08.21 | p3nguin | 3.42? |
| 19:08.32 | p3nguin | Should be a 3.42. |
| 19:08.33 | dijib | is that it? maybe... somewhere in or around there |
| 19:08.50 | dijib | only the 1500 |
| 19:09.21 | dijib | love the truck though still... its like a limo/bus |
| 19:09.21 | p3nguin | I have 3.42 in my S-10 and in my Blazer. |
| 19:09.28 | p3nguin | I had 3.73 in my Camaro and my Cutlass. |
| 19:09.29 | dijib | i sold my jimmy for $200cad 5months ago. |
| 19:09.40 | dijib | p3nguin, your pimped out by the sounds of it |
| 19:10.09 | dijib | i think my blazer has a 3.23 |
| 19:10.27 | dijib | so how do i mcgiver a distance guage? |
| 19:10.47 | p3nguin | I had a lot of fun with my Cutlass. I put my 350 in it with a 700-R4 trans and 3.73 rear end. It came off the line like a rocket, and when it shifted into 2nd didn't just get a little scratch, but may as well have been a full burnout. |
| 19:11.20 | dijib | i believe it with the 3.73 |
| 19:11.20 | Kobaz | i'll test the two dmarc grounds against each other |
| 19:11.53 | p3nguin | Just look to make sure there is about 1/8" to 5/32" between the ring gear and the shaft for the starter drive. |
| 19:12.01 | dijib | do they make a dually rear end in a 3.73? |
| 19:12.09 | p3nguin | Maybe. |
| 19:12.22 | dijib | but how can i see that? |
| 19:12.37 | p3nguin | The 14-bolt came in 3.73, I believe. |
| 19:13.02 | dijib | this is now #asterisk/GM |
| 19:13.26 | p3nguin | Does your bell housing cover the nose of the starter so you can't see between the flexplate and the starter drive? |
| 19:14.08 | p3nguin | I thought that would have the 4L60 or 4L65 trans and have a removable cover. |
| 19:14.35 | dijib | it does then.. i didnt know it was removable. |
| 19:14.39 | dijib | looks heavy |
| 19:14.54 | dijib | ill get under her in an hour after i think the batt should be charged. |
| 19:15.12 | p3nguin | Being a 4x4, it may be cast aluminum instead of the plastic or tin used on 4x2. |
| 19:15.49 | p3nguin | If it has a removable cast cover, it'll have bolts holding it on rather than sheet metal screws. |
| 19:16.21 | p3nguin | I don't have a lot of experience with the newer transmissions to know if they have removable covers or not. You'd have to look. |
| 19:16.32 | dijib | yesterday after installing the starter the night before & testing without issue i took the truck to get some gas. ($180 worth (150L)) pulling into the gas station... my truck dies! smell burning plastic... roll into gas lane... lift the hood to find my + battery terminal GONE, melted away. turnes out the + wire on the starter was touching the header and grounded out.. killing my #1 battery of 2 |
| 19:16.41 | p3nguin | I pretty much know a TH-350 inside and out, though. |
| 19:16.43 | Kobaz | i bumped up the sensitivity of my multimeter, changed the range down, i'm seeing .001 to .002 |
| 19:16.47 | Kobaz | sometimes drops to 0 |
| 19:16.52 | *** join/#asterisk DanFromUK (DanFromUK@2.27.40.63) |
| 19:17.19 | dijib | p3nguin, mines the 4l60e |
| 19:17.38 | p3nguin | That's just a newer TH 700R4 |
| 19:17.47 | dijib | yep |
| 19:17.55 | dijib | with electronic shifting? |
| 19:18.02 | p3nguin | It really should have a cover, but I just don't know how they changed after about '95. |
| 19:18.12 | p3nguin | Yes, that's what the e means. |
| 19:18.17 | dijib | ja. |
| 19:18.17 | Kobaz | p3nguin: so that should be good, right? |
| 19:18.29 | dijib | and im sure it has a cover.. i just am lazy. |
| 19:18.43 | dijib | so 1/8th to 5/32's eh |
| 19:18.48 | dijib | ill check her out in a bit |
| 19:19.02 | dijib | Kobaz, sounds good to me. |
| 19:19.08 | p3nguin | kobaz: If you have an accurate reading of 0.002 Ohms, that's a reasonable ground for smaller ground wire. |
| 19:19.24 | p3nguin | If you had like 0/1 or something, I'd expect 0.0000 Ohms. |
| 19:20.19 | Kobaz | actually my port isn't fried |
| 19:20.19 | Kobaz | hmm |
| 19:20.29 | Kobaz | a week ago when i was here there was a horrible humming |
| 19:20.31 | Kobaz | now it's clean |
| 19:21.34 | Kobaz | time to head out |
| 19:24.44 | dijib | ok so fax detect before i head outside to crawl under this burban. |
| 19:24.48 | dijib | how do? |
| 19:24.55 | dijib | * 1.8.5 |
| 19:25.05 | dijib | NVFaxDetect is no longer needed |
| 19:25.05 | *** part/#asterisk tehrabbitt-1 (~ryana@unaffiliated/tehrabbitt) |
| 19:25.12 | dijib | as i understand |
| 19:25.48 | *** join/#asterisk brettnem (~brett@76-216-204-224.lightspeed.austtx.sbcglobal.net) |
| 19:25.53 | dijib | backgrounddetect? |
| 19:25.57 | brettnem | hey all |
| 19:26.18 | luke-jr | what's the standard way to run a script at the end of all calls? |
| 19:26.50 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 19:26.53 | DanFromUK | hi, is it possible to retrieve the extension that triggers a macro feature? is there a variable that stores the extension name? |
| 19:28.23 | p3nguin | luke-jr: Use the h extension, and either System() or SHELL(). |
| 19:28.48 | p3nguin | danfromuk: ${EXTEN} doesn't do it for you? |
| 19:30.12 | p3nguin | dijib: I use a dedicated fax number over SIP, so I don't worry with detection. Let me know if you figure out how to do detection on a voice/fax line. |
| 19:32.17 | dijib | i will |
| 19:33.26 | p3nguin | Pewp. My g722 build failed. |
| 19:34.22 | carrar | People still use g722? |
| 19:34.31 | p3nguin | Of course they do. |
| 19:34.56 | p3nguin | What did you think they'd use in place of it? |
| 19:35.14 | carrar | GSM!! |
| 19:35.22 | drzed | re |
| 19:35.29 | WIMPy | Instead of G.722??? |
| 19:36.22 | p3nguin | Is there no newer g722 patch for 1.4? These things are several years old. |
| 19:36.41 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 19:36.46 | *** join/#asterisk bitbandit (~taggmcd@c-98-202-116-168.hsd1.ut.comcast.net) |
| 19:37.08 | brettnem | hey all.. I've got an analog line (DAHDI FXS) plugged into an old fashioned overhead paging unit.. Everything works fine, but when the person paging hangs up (SIP phone) there is a fast busy disconnect tone that plays overhead for about 15 seconds.. Anyone know what causes this? Before I upgraded from ZAP to DAHDI it didn't do this. |
| 19:37.32 | brettnem | CLI shows that a hangup on DAHDI, THEN we hear the tone overhead |
| 19:38.12 | drzed | i fixed the context stuff, now my phones (30,34,35) are in context lokal, now my log looks like this: - Executing [*830@lokal:2] Verbose("SIP/34-084f4200", "1|34 wants to pickup from extension 30") in new stack |
| 19:38.16 | drzed | <PROTECTED> |
| 19:38.18 | drzed | <PROTECTED> |
| 19:38.21 | drzed | <PROTECTED> |
| 19:38.38 | drzed | however the (direct) pickup still does not work |
| 19:39.14 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 19:40.47 | brettnem | anyone? :/ |
| 19:41.46 | luke-jr | p3nguin: the wiki says the 'h' extension doesn't work right (eg, with Macros) |
| 19:42.14 | Qwell | What wiki? |
| 19:42.23 | WIMPy | brettnem: You have late media enabled? But I can't remember what the parameter is. |
| 19:42.31 | p3nguin | luke-jr: I didn't know anything about any macros. You asked how to run a script, and I told you how. |
| 19:42.41 | brettnem | WIMPy: late media? I've never heard of that... |
| 19:42.54 | luke-jr | p3nguin: the problem is getting it to run no matter how a call ends |
| 19:42.54 | WIMPy | brettnem: Analog is evil. Use a sound card. |
| 19:43.08 | luke-jr | Qwell: the one with the docs for Asterisk⦠voip-info or smth |
| 19:43.13 | brettnem | WIMPy: yeah, this is an old fashioned paging system built for analog lines.. :/ |
| 19:43.15 | p3nguin | luke-jr: How many ways are there? |
| 19:43.21 | brettnem | WIMPy: is it actually called "late media" ? |
| 19:43.21 | WIMPy | brettnem: You said, you hear it for 15s. |
| 19:43.29 | Qwell | That wiki is often wrong. |
| 19:43.34 | p3nguin | voip-info or smith? don't rely on smith nor voip-info for accurate information. |
| 19:43.48 | brettnem | WIMPy: also, the CLI shows it's hung up, so it seems that the dahdi drivers are providing the tone |
| 19:43.54 | WIMPy | brettnem: No. I can;t remember what it's called. |
| 19:44.26 | WIMPy | brettnem: Yes, they are. You need that to be able to listen to announcements. |
| 19:44.33 | luke-jr | p3nguin: well, if the user hanging up during a macro doesn't work⦠|
| 19:45.01 | brettnem | Qwell: actually, in older version of ast, the "h" extension didn't work so well.. I know there were a lot of variables that were missing.. |
| 19:45.06 | brettnem | I don't think that's the case anymore.. |
| 19:45.16 | WIMPy | brettnem: 'inbanddisconnect' |
| 19:45.25 | Qwell | brettnem: Which is why we don't recommend using that wiki. |
| 19:45.50 | p3nguin | I use h all the time without trouble. |
| 19:45.55 | leifmadsen | same |
| 19:46.03 | luke-jr | Qwell: is there actual documentation somewhere then? Google only ever finds the wiki |
| 19:46.21 | brettnem | Yeah, I think I've used "h" quite a bit without issue too.. I think you have to go back to 1.2 or so for it to be broken |
| 19:46.37 | brettnem | WIMPy: ++ you're my hero |
| 19:46.39 | Qwell | 1.2 was released 5 years ago. |
| 19:46.42 | leifmadsen | ofps.oreilly.com has a link to the Asterisk book, and there is wiki.asterisk.org |
| 19:46.42 | fullstop | I'm looking to upgrade a test server from 1.6 to the latest 1.8 revision. I build from source. |
| 19:46.46 | Qwell | maybe even 6. |
| 19:46.55 | fullstop | What's the best way to upgrade and not leave 1.6 cruft around? |
| 19:47.03 | Qwell | fullstop: make uninstall install |
| 19:47.23 | luke-jr | so 'h' should execute in the main context even if the hangup occurs inside a macro? |
| 19:47.29 | fullstop | Qwell: will that remove anything in /var/lib/asterisk ? |
| 19:47.40 | Qwell | It will remove everything that make install installs. |
| 19:48.16 | fullstop | I only ask because I did that once in 1.4 and it removed all of /var/lib/asterisk/sounds... |
| 19:48.24 | fullstop | including the files that I had added. |
| 19:48.27 | Qwell | Which is something that make install installs. |
| 19:48.29 | dijib | whats a charged voltage for a 12v car battery? |
| 19:48.38 | Qwell | You're putting stuff in the wrong place. :) |
| 19:48.38 | p3nguin | 14V |
| 19:48.52 | dijib | k im at like 13.4 now |
| 19:48.53 | p3nguin | 12-14 |
| 19:48.59 | dijib | and going up |
| 19:49.06 | blizzow | I've just received a complaint that asterisk has dropped a call. I went into /var/log/asterisk/full and looked at the timestamp and found my caller. Here is the pastebin: http://pastebin.com/pk1KPHmf Line 3 and 4 are errors, can someone explain what I should be looking for in the log to explain the cause of the dropped call? |
| 19:49.13 | fullstop | Qwell: where should I be putting my sound files, then? |
| 19:49.13 | p3nguin | Anything less than 12 might give you a slow crank-over. |
| 19:49.30 | dijib | i was at 12.4 earlier and it wouldnt crank |
| 19:49.45 | p3nguin | You never checked the cables like I told you, did you? |
| 19:49.54 | p3nguin | at the battery, and at the starter. |
| 19:49.59 | dijib | yes i did |
| 19:50.02 | dijib | earlier. |
| 19:50.19 | p3nguin | Also, there is always a chance that your ground cable has a bad ground at the engine. |
| 19:50.29 | dijib | shouldnt be |
| 19:50.35 | dijib | looks like its intact |
| 19:50.39 | p3nguin | Shouldn't be, but there's a chance. |
| 19:51.16 | WIMPy | Who installed Asterisk on a car? |
| 19:51.32 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:51.58 | fullstop | Qwell: From what I quickly read, most people would be expected to put their audio files in /var/lib/asterisk/sounds |
| 19:52.01 | DanFromUK | p3nguin: sorry for the delay in replying. no ${EXTEN} only returns 's' |
| 19:52.17 | p3nguin | How did your call arrive at extension s? |
| 19:52.54 | p3nguin | or is that s only in the macro? |
| 19:52.56 | DanFromUK | the call was connected between a sip peer and an external caller. then using a feature, a macro is started by dialing *1 |
| 19:53.12 | DanFromUK | s is only in the macro |
| 19:53.13 | p3nguin | So the original extension was *1 every time? |
| 19:53.17 | DanFromUK | i need the sip peer name |
| 19:53.25 | p3nguin | Oh. you said you wanted the extension. Sigh. |
| 19:53.55 | DanFromUK | sorry. old pbx days |
| 19:54.16 | WIMPy | Better days |
| 19:54.37 | p3nguin | How about ${CHANNEL:0:-9} ? |
| 19:54.45 | DanFromUK | 1sec |
| 19:56.12 | DanFromUK | thats almost perfect! |
| 19:56.20 | DanFromUK | how can i remove the SIP/ part |
| 19:56.34 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 19:56.41 | p3nguin | How about ${CHANNEL:4:-9} ? |
| 19:56.52 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:56.56 | DanFromUK | whats the -9 part do? |
| 19:57.14 | p3nguin | removes the last 9 characters from the channel name |
| 19:57.28 | p3nguin | SIP/000011112222-00000001 |
| 19:57.35 | p3nguin | changes to SIP/000011112222 |
| 19:58.04 | DanFromUK | ah, thats great! |
| 19:58.11 | brettnem | WIMPy: inbanddisconnect=no didn't do anything :( |
| 19:58.25 | brettnem | these are analog lines, I'm not sure if that setting affects FXS |
| 19:58.29 | brettnem | any other ideas? :( |
| 19:59.41 | DanFromUK | p3nguin: thanks for your help |
| 20:02.25 | p3nguin | I'm using http://users.netplex.net/~andrew/asterisk/g722-20090218.patch.gz to get g722 on 1.4.42, but it fails. Any idea what's wrong? http://pastebin.com/EyztEXWd |
| 20:04.36 | WIMPy | brettnem: I would have thought it does the same, but unfortunately you never know exactely what happens where. Did you reload chan_dahdi? |
| 20:04.53 | brettnem | I stopped asterisk, restarted dahdi and started asterisk |
| 20:04.55 | brettnem | just to be save |
| 20:04.57 | brettnem | safe |
| 20:05.02 | WIMPy | ok |
| 20:05.10 | brettnem | but from the docs, it appears to be a PRI setting |
| 20:05.25 | brettnem | "; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI" |
| 20:05.46 | brettnem | it's almost like the signaling is wrong |
| 20:06.16 | WIMPy | reads that as 'if the other end is a pri'. which ist certainly wrong as well. |
| 20:07.20 | brettnem | wish I knew if the cadence is coming from the server or the paging device.. could this possibly be something in indicatiosn? |
| 20:19.10 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 20:24.13 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 20:24.14 | otwieracz | Hello. |
| 20:24.32 | otwieracz | I can use queues to pass incoming call to four SIP clients? |
| 20:24.39 | otwieracz | With strategy=ringall |
| 20:25.37 | navaismo | ring all but only one can answer |
| 20:25.43 | otwieracz | BUt exten => s,1,Queue(noc_cell) doesn't seem to work. |
| 20:25.50 | otwieracz | Yes, one answer and other phones stop ringing. |
| 20:26.28 | navaismo | you added the sip peers as agents in that queue? |
| 20:26.58 | otwieracz | member => SIP/00PHONENUMBER |
| 20:27.37 | navaismo | in the cli the command queue show show your agents as not in use |
| 20:27.39 | navaismo | ? |
| 20:28.05 | otwieracz | http://wklej.org/hash/268e2b0486c/ |
| 20:28.15 | otwieracz | Thats my queues.conf part. |
| 20:28.31 | p3nguin | I really don't like that way unless you need to queue the calls. |
| 20:28.41 | p3nguin | If you just want to call several phones, Dial() can do it. |
| 20:29.02 | p3nguin | Dial(SIP/jack&SIP/jill&SIP/jane,36) |
| 20:29.03 | navaismo | yes dial(sip/phione1&Sip/phone2&....) |
| 20:29.03 | otwieracz | Yes, but there will be about 15 numbers. |
| 20:29.04 | navaismo | jeje |
| 20:29.24 | otwieracz | I want to do it more user-friendly. |
| 20:29.42 | p3nguin | The user shouldn't be administering your asterisk. |
| 20:29.43 | citywok | otwieracz: how is a queue more user friendly? |
| 20:30.01 | citywok | it's actually more work in the long run. &SIP/2&SIP/3 isn't very hard to do |
| 20:30.23 | otwieracz | Yes, but with eight-digit numbers it's not easy to read this. |
| 20:30.29 | otwieracz | And manage, delete numbers. |
| 20:30.35 | navaismo | anayway what show the cli when call go into the queue |
| 20:30.58 | otwieracz | <PROTECTED> |
| 20:30.58 | otwieracz | <PROTECTED> |
| 20:31.14 | otwieracz | And that's all. |
| 20:31.23 | citywok | does it not ring anybody? |
| 20:31.23 | otwieracz | No calls are performed. |
| 20:31.27 | p3nguin | How is it going to be any easier to write the peers in queue.conf as opposed to extensions.conf? It sounds like you're just making excuses. |
| 20:31.29 | citywok | queue show noc_cell in the cli |
| 20:31.31 | navaismo | mmm i think tyour agents are not logged in |
| 20:31.57 | otwieracz | They are cellular phones. |
| 20:32.03 | citywok | wait... are you trying to call people's cellphones, or active sip pers? |
| 20:32.06 | citywok | lol... |
| 20:32.22 | navaismo | cell phones with sip client? |
| 20:32.23 | p3nguin | If you don't have cell phones registered to the system, how will you call them on the system? |
| 20:32.28 | citywok | what you are trying to do is a disaster :P |
| 20:32.42 | p3nguin | citywok: Meet brick wall. |
| 20:32.50 | citywok | lol no kidding |
| 20:32.53 | p3nguin | :/ |
| 20:32.55 | citywok | the lack of comprehension is funny |
| 20:33.05 | p3nguin | and sad at the same time. |
| 20:33.17 | citywok | otwieracz: you need to dial the cellphone like you would dial a cellphone in any other dial statement.... |
| 20:33.34 | p3nguin | If you insist on doing it with a queue, at least use local channels as the members. |
| 20:33.42 | citywok | sip/12535551212@outboundpeer |
| 20:33.44 | otwieracz | Yes, I already find the fail. |
| 20:33.49 | otwieracz | And it work now :) |
| 20:33.51 | citywok | yea, or use local/number@outbound-context |
| 20:34.41 | citywok | if this is a notification for an alarm or something we dial all the cellphones in a local context and then join all of them that ansewr to a meetme bridge |
| 20:34.59 | citywok | so all our techs can chat about it and figure out who is near a PC to fix it |
| 20:35.08 | p3nguin | Great idea. |
| 20:35.33 | jaytee | yeah, that's pretty cool |
| 20:35.34 | citywok | p3nguin: just remember cellphones have voicemail, i have to press 1 on my phone to get dumped in to the confbridge |
| 20:35.42 | citywok | otherwise it just leaves a voicemail on my phone |
| 20:36.01 | citywok | instead of an voicemail_max_length voicemail of the conference bridge lol |
| 20:36.06 | citywok | s/an/a/ |
| 20:36.27 | otwieracz | Hmmm. |
| 20:36.41 | otwieracz | Now I have other problem. |
| 20:37.42 | otwieracz | Queue phones rang only for⦠0.5s? |
| 20:39.11 | otwieracz | Oh, nvm. |
| 20:50.37 | *** join/#asterisk avb (~avb@76.76.102.242) |
| 20:50.42 | avb | [Sep 1 16:48:42] WARNING[3380]: sig_ss7.c:904 ss7_linkset: REL on channel (CIC 61) without owner! |
| 20:50.52 | avb | guys what can cause that? |
| 20:51.40 | avb | thats ss7 connection |
| 20:51.48 | WIMPy | A bug? |
| 20:52.10 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 20:52.19 | avb | after its appears, then lines sending 'all cirquits are busy now' |
| 20:52.30 | avb | WIMPy: :) i hope not |
| 21:12.01 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:14.34 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-itnvniqkoumcvuqh) |
| 21:19.42 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 21:29.27 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 21:32.02 | *** join/#asterisk tomaw (tom@gnaa/tomaw) |
| 21:32.42 | *** join/#asterisk tomaw (tom@freenode/staff) |
| 21:37.49 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 21:37.51 | eja | is it possible to reset a cisco sip phone from asterisk like you can with CME? |
| 21:39.49 | _Corey_ | eja: What model phone? |
| 21:40.26 | eja | i believe it's a 7960 |
| 21:41.03 | _Corey_ | I'm pretty sure they don't respond to SIP NOTIFIES, so you'd have to telnet to the phone and tell it to reboot |
| 21:41.05 | Nugget | telnet is eeeeeeevil! |
| 21:41.27 | _Corey_ | There are some scripts out there if you google a bit |
| 21:41.44 | eja | ok thanks corey |
| 21:41.54 | _Corey_ | yeah no problem |
| 21:54.06 | penguin | real voip engineers use openSer |
| 21:54.14 | penguin | with media relay servers |
| 21:54.23 | penguin | not asterisk |
| 21:55.18 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:55.27 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 21:59.51 | p3nguin | Huh? I would think that a real VoIP engineer could use SER and Asterisk together just fine. I mean, we do, after all. |
| 22:00.35 | penguin | p3enguin: yea but i've been hearing alot of ppl trying to run a service off asterisk pbx alone |
| 22:00.39 | *** part/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 22:00.42 | penguin | i was just commenting on that topic |
| 22:01.16 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:02.22 | blizzow | Does anyone have a suggestion for a windows free softphone client (preferably open source)? |
| 22:02.39 | p3nguin | Zoiper is a good soft phone, but I'm pretty sure it isn't open. |
| 22:02.53 | p3nguin | It's free to use, though. |
| 22:03.27 | *** part/#asterisk Frem_ (~jamesgeck@64.207.3.161) |
| 22:06.18 | penguin | p3nguin: do u run a voip service with ur setup? |
| 22:06.27 | penguin | openser and asterisk |
| 22:07.56 | blizzow | p3nguin: I'm having a terrible time with Zoiper. |
| 22:07.59 | blizzow | It keeps crashing. |
| 22:08.20 | blizzow | All of our sales guys want IT's head on a stick. |
| 22:08.44 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mqefqykdngoejcun) |
| 22:08.59 | anonymouz666 | crash is ugly |
| 22:10.35 | p3nguin | I personally only use Asterisk for business phone systems, with a side of home phones. No commercial services from me. |
| 22:12.06 | greenwolf | p3nguin: so do you design these systems for companies or do you just use for your local office |
| 22:12.36 | p3nguin | both. |
| 22:12.50 | greenwolf | nice |
| 22:13.14 | p3nguin | I typically don't do large-scale deployments. I prefer small to medium businesses because it's so easy. |
| 22:13.30 | anonymouz666 | I like large-scale |
| 22:13.47 | dijib | p3nguin, you wont believe what it was |
| 22:13.52 | greenwolf | yea i agree. I just setup a medium system for a local collections agency around here..nice system and easy deployment |
| 22:13.52 | anonymouz666 | it's a challenge |
| 22:14.04 | dijib | know how i said it shorted out on the manifold? |
| 22:14.28 | dijib | well the wire from batt --> starter was fried, wouldnt carry and current |
| 22:14.55 | p3nguin | Yeah, that makes sense. |
| 22:14.58 | dijib | was the issue. luckely i had some 1 guage around, flux, saulder, and ends laying around and fixed er right up |
| 22:15.18 | dijib | starts like a dream now. without any shims |
| 22:15.36 | p3nguin | I guess the distance was good, then. |
| 22:15.43 | p3nguin | Otherwise, you'd get some noise from it. |
| 22:15.43 | dijib | me3 |
| 22:15.53 | dijib | no its starts stong and short. |
| 22:17.39 | dijib | ive got an issue with asterisk making calls. almost all the time now when i make a call, i get dead air. the callee gets dead air, but call goes through. then i remake the call and we can both hear eachother. |
| 22:18.07 | dijib | any idea |
| 22:18.08 | dijib | ? |
| 22:18.19 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:30.19 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-189.dsl.stlsmo.sbcglobal.net) |
| 22:30.30 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-163-189.dsl.stlsmo.sbcglobal.net) |
| 22:33.47 | p3nguin | anonymouz666: I wouldn't mind doing some larger deployments in a single facility or a facility with a few satellite offices. |
| 22:34.06 | p3nguin | The problem seems to be time and cost. People have no patience nor money. |
| 22:36.13 | WIMPy | Shit. I think my printer has died. It spews out solid back paper. :-( |
| 22:36.21 | p3nguin | Oh no! |
| 22:36.29 | p3nguin | laser or ink jet? |
| 22:36.42 | WIMPy | Laser |
| 22:36.56 | WIMPy | How would an inkjet do that? |
| 22:37.15 | p3nguin | It's bad enough wasting toner like that, but that would be a LOT of ink to spray on paper. |
| 22:37.22 | WIMPy | That was a good old one that really could do black. |
| 22:37.28 | p3nguin | I have no idea how any printer would do it. |
| 22:38.08 | WIMPy | NFI. I guess high voltage or laser malfunction. Both seem fatal :-( |
| 22:46.44 | ChannelZ | postscript error? |
| 22:51.00 | *** join/#asterisk beta2k (~Beta2K@d24-36-128-84.home1.cgocable.net) |
| 22:51.04 | beta2k | hello all |
| 22:51.22 | beta2k | Anyone around know how to setup a sip/iax trunk between two pbx's for internal extension calls? |
| 22:51.28 | beta2k | Basically i have site A and site B, each with a trunk to our provider and a trunk between themfor internal calls |
| 22:51.55 | WIMPy | ChannelZ: That would be nice, but the power-up test page looks the same. |
| 22:52.25 | p3nguin | Okay, so if you have an IAX2 trunk between two asterisk systems, where does SIP come into the equation? |
| 22:53.05 | *** join/#asterisk FreezingCold (~Frozen@unaffiliated/freezingcold) |
| 22:53.07 | FreezingCold | Hey guys! |
| 22:53.08 | beta2k | either one, not both :) |
| 22:53.54 | FreezingCold | I'm sorry to ask this, but could anybody spend the time to baby me along with my first asterisk setup? I normally like to take my time but I have a sister leaving the country tomorrow and I need to get it setup for her =( |
| 22:54.29 | beta2k | I'd be happy with a doc, I don't need step by step over IRC :) |
| 22:54.36 | FreezingCold | Haha |
| 22:54.53 | FreezingCold | Well, somewhat on topic, what are some good SIP providers in the UK? |
| 22:55.06 | greenwolf | lots |
| 22:55.17 | p3nguin | ~book |
| 22:55.17 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 22:55.19 | p3nguin | freezingcold: ^^^ |
| 22:55.20 | FreezingCold | she needs a incoming DID. Unlimited is always better |
| 22:55.35 | FreezingCold | ehhhh, as I said I only have until tonight to get everything running |
| 22:55.42 | p3nguin | Read quickly. |
| 22:55.45 | FreezingCold | Not sure I really have time to go through the whole book right now |
| 22:57.07 | p3nguin | beta2k: I'll at least send you in the right direction to get started. Configure a peer entry for system1 on system2; configure a peer entry for system2 on system1. |
| 22:57.21 | FreezingCold | Draytel seems like the cheapest/best |
| 22:57.31 | FreezingCold | Unlimited incoming DID |
| 22:57.40 | FreezingCold | 10 GBP start up top up |
| 22:58.04 | *** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:58.11 | beta2k | p3nguin: No user context? Just peer? |
| 22:58.43 | p3nguin | You could use type=friend if that's what you feel like doing. |
| 23:03.29 | FreezingCold | hmmm |
| 23:03.32 | FreezingCold | any tips? |
| 23:03.35 | *** join/#asterisk Bidik (~bidik@89.205.111.82) |
| 23:03.47 | FreezingCold | How hard is getting asterisk up with two SIP devices and two providers? |
| 23:03.53 | dijib | FreezingCold, i could help |
| 23:03.54 | FreezingCold | Can I get done in an hour or two? |
| 23:04.00 | dijib | im an * nuub aswell |
| 23:04.06 | FreezingCold | Haha, thanks :) |
| 23:04.09 | dijib | SIP? |
| 23:04.24 | FreezingCold | Yep |
| 23:05.49 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 23:06.40 | FreezingCold | How good is 44uk.co.uk? |
| 23:10.28 | p3nguin | codec_ilbc.c:50:30: fatal error: ilbc/iLBC_encode.h: No such file or directory |
| 23:10.28 | p3nguin | compilation terminated. |
| 23:10.34 | p3nguin | Not Good. |
| 23:15.36 | dijib | :D |
| 23:21.04 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:25.23 | carrar | Oh My |
| 23:27.12 | p3nguin | I of course have no way to know how to fix it, since someone else is responsible for writing asterisk's code. |
| 23:28.35 | carrar | because you can't google? |
| 23:28.48 | anonymouz666 | this fix seems easy |
| 23:28.56 | anonymouz666 | you just have to put this header in the correct place |
| 23:29.21 | carrar | Did you run contrib/scripts/get_ilbc.sh |
| 23:31.00 | Maliuta | or you could just use an option to tell gcc where your libs actually live |
| 23:31.11 | Maliuta | or there is even an environment variable |
| 23:34.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 23:47.51 | p3nguin | I haven't done anything to debug it. I was working on something else and just happened to see that it stopped. |
| 23:52.20 | p3nguin | anonymouz666: Where do you propose I get iLBC_encode.h? It does not exist on my system. |
| 23:53.59 | p3nguin | If I hadn't run contrib/scripts/get_ilbc_source.sh, I'm not so sure it would have gotten this far. |
| 23:54.26 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 23:55.18 | anonymouz666 | this script should download the source for you |
| 23:55.42 | anonymouz666 | if doesn't then you don't have the header file needed to compile codec_ilbc |
| 23:56.09 | p3nguin | Yeah, that's the problem. The file is missing. Why is it missing and where am I expected to get it? |
| 23:56.35 | anonymouz666 | it is missing because this script connects somewhere and get the file, you know, remote things can change :) |
| 23:57.07 | p3nguin | I'll go check out the files that get downloaded. |
| 23:57.27 | anonymouz666 | http://ilbcfreeware.org |
| 23:57.57 | carrar | No iLBC 4U!! :) |
| 23:59.09 | anonymouz666 | I don't miss ilbc, but speex I make sure that every new installation has support to |
| 23:59.28 | p3nguin | I guess the site can't be reached. |
| 23:59.30 | blizzow | Is there a good echo test service that just stays up and running for an unlimited amount of time? I want to debug calls that are dropping seemingly at random. |
| 23:59.48 | anonymouz666 | p3nguin: that could explain why the script does not work |
