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| 07:10.49 | Penguin | channelz: Totally random, but... I overhauled that old AR15A2 rifle and it runs like a champ. I put a new full length (rifle) buffer and new spring in it, cleaned the carbon off the bolt and smeared some CLP on it, cleaned off most of the dry Rem Lube and re-lubed it with white grease, and it's as good as new. The action feels like a well-oiled machine, unlike newer stuff that still has the "gritty" feeling from the ... |
| 07:10.55 | Penguin | ... phosphate finish. |
| 07:11.12 | Penguin | I know it's been a while since we talked about it, but I finally got a round tuit. |
| 07:11.35 | ChannelZ | Nice. You were having ejection failures or something before right? |
| 07:11.48 | Penguin | Failures to feed on every single round. |
| 07:12.01 | Penguin | It wasn't picking up new cartridges. |
| 07:12.03 | ChannelZ | Ok couldn't completely remember |
| 07:12.19 | ChannelZ | Glad you got it running |
| 07:12.20 | Penguin | Yeah, it's been like a year... maybe two? since we talked about it. |
| 07:13.33 | Penguin | I didn't even touch the gas tube, so that clearly wasn't the problem. My initial suspect was a dirty gas tube not providing enough gas to "throw" the carrier back far enough to pick up a new round. |
| 07:14.47 | ChannelZ | Sounds like maybe it was just sticky |
| 07:14.49 | Penguin | I think it was just a combination of carbon grit around the bolt slowing down the opening of it, inadequate lube, and possibly a week buffer spring. |
| 07:16.00 | Penguin | She's ready to rock and roll now, though. I was pretty happy when I fed a magazine through it without any failures. |
| 07:16.10 | ChannelZ | Woot! |
| 07:16.27 | Penguin | Slow fire, rapid fire, ran like the first time I took it out. |
| 07:17.34 | Penguin | I'm about to make some changes on it, though. It's a 20" HBAR, so it's a beast to carry around. I'm going to change the A2 upper to a flat top and stick a scope on it. |
| 07:17.47 | ChannelZ | I need to pull everything out of my safe and clean them. Been dealing with some water problems in my basement |
| 07:18.46 | Penguin | I've been taking my M4 and my 9mm carbine to the range lately, so I can tell the HBAR needs a new purpose. |
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| 07:21.33 | Penguin | Sheesh, was that two years ago that we talked about that thing not running right? Maybe three?! |
| 07:22.06 | Penguin | I need to get more personal time. |
| 07:22.07 | ChannelZ | Nah, maybe two.. dunno |
| 07:22.18 | Penguin | It's a long time, nevertheless. |
| 07:22.25 | ChannelZ | Yeah |
| 07:24.02 | Penguin | My very next project is to figure out if I can convert and Oly Arms 9mm to use a more affordable magazine. |
| 07:24.08 | ChannelZ | I don't think i went shooting a single time last year *bawls* |
| 07:24.24 | Penguin | s/and /an / |
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| 07:25.49 | Penguin | I don't think I did. If I did, it was to the pistol range for a box or two between my wife and myself. |
| 07:26.15 | Penguin | slacking |
| 07:26.26 | ChannelZ | Life is a pain in the ass |
| 07:27.07 | Penguin | I'm trying to make Mondays gun days. I want to go shoot for a couple hours every Monday evening. |
| 07:28.21 | ChannelZ | Good deal |
| 07:28.21 | Penguin | Take a maximum of one rifle for myself, one for the wife, spend 2-3 hours, then go home. I think it's a good plan. |
| 07:29.19 | ChannelZ | Do you live near public land or something you can go to? |
| 07:29.35 | Penguin | I bought her a Sight Mark reflex red dot and had to send it back because it was too blurry, so she's on iron sights until they get that replaced. |
| 07:29.53 | Penguin | I've got private land outside of the city where I shoot. |
| 07:30.00 | ChannelZ | nice |
| 07:30.01 | Penguin | It takes about 10-15 minutes to drive there. |
| 07:30.58 | Penguin | I used those green T posts and wood shipping pallets to shoot at. Just staple paper targets on them for now. |
| 07:31.21 | Penguin | I'm wanting to make some steel gongs for 100 or 200 yards, but that's another project. |
| 07:33.40 | Penguin | Do you do any handloading of ammo? |
| 07:34.18 | ChannelZ | No, I have a bunch of 9mm and .45 brass I've saved over the years but it's not something I've pursued yet |
| 07:34.57 | Penguin | After I shoot a few more .223s, I'm going to have to study up to see what people are liking to load with. |
| 07:35.16 | Penguin | I've got a bit of brass, but I'd like to do a whole batch if I do any. |
| 07:36.29 | ChannelZ | yeah |
| 07:37.20 | Penguin | RCBS has a current rebate if you're considering buying any loading equipment or supplies. $10 back on $50 or $50 back on $300. Purchases Jan 1 - June 30. |
| 07:37.48 | Penguin | Dies are around $50 each, so you can hit $300 pretty easily. |
| 07:38.43 | ChannelZ | Yeah I'm nowhere near doing it now. Too much other chaos to deal with for the forseeable future unfortunately |
| 07:39.15 | Penguin | I hear ya. I'm finally slowing down, which is why I'm so anxious about range time now. |
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| 16:21.59 | Prelude2004c | hi everyone |
| 16:22.02 | Prelude2004c | quick question |
| 16:22.21 | WIMPy | That was not a question. |
| 16:22.48 | Prelude2004c | attended transfer.. any way that once i have the caller on hold and talking with the other part to merge calls ( like 3 way ) and say .. so and so is going to help you from here. .and hangup an they can continue talking? |
| 16:22.58 | Prelude2004c | i have it now where i hang up and then the caller talks to the 3rd party |
| 16:23.03 | Prelude2004c | but i want to be able to introduce them |
| 16:23.53 | WIMPy | That's a bit of a PITA. You have to transfer both to a ConfBridge. |
| 16:24.43 | Prelude2004c | so attended transfer can't do that right? |
| 16:27.21 | WIMPy | No |
| 16:32.46 | [TK]D-Fender | Prelude2004c, * DTMF transfers can't do that, but many phones can via SIP |
| 16:32.54 | [TK]D-Fender | Polycom for example |
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| 17:11.00 | Prelude2004c | ya.. problem is they are using softphone |
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| 17:14.22 | liquidamber | ok so i have a "sandbox" context for inbound calls with no ability to call out, to avoid toll fraud. but in sip.conf i have to give that SIP peer a context. So i can't figure out how to make outbound calls |
| 17:15.43 | liquidamber | weird thing is i havent changed the sip.conf or extensions.conf and it just stopped working on outbound calls. |
| 17:16.08 | [TK]D-Fender | Prelude2004c, Then you're going to have to use a conference room of some kind |
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| 17:18.47 | liquidamber | if this is inside Internal context: exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms) it should be able to exit thru the voipms peer which lives in a different context, no? |
| 17:19.48 | WIMPy | Context only matterns for your call, not if you call a peer. |
| 17:20.28 | liquidamber | ok - it says app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent). but the subscriber is there |
| 17:21.12 | WIMPy | Maybe you provider is unreachable or the person you have called. |
| 17:21.39 | liquidamber | hmm it does say status unknown in sip show peers :/ |
| 17:23.07 | WIMPy | Look ath the *CLI to see what's happening. |
| 17:35.28 | liquidamber | thats stupid. i had ; comments in the sip peer and removing them seems to have fixed it |
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| 17:36.36 | WIMPy | Don't expect 'sip reload' to work. If you change something in your sip.conf, always unload and load the chan_sipmodule, or restart Asterisk. |
| 17:36.52 | liquidamber | good to know, thanks for the help |
| 17:37.41 | WIMPy | You can add new peers, but if you change existing ones, it's about luck, or pahse of the moon, or whatever. |
| 17:38.11 | liquidamber | i occasionally remember to fully restart it... usually after 8 hours of troubleshooting some crap |
| 17:38.53 | WIMPy | That should hopefully be a story of the past, with pjsip. |
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| 18:44.56 | drab | what are organizations entirely on voip doing for emergency calls? do they also have one "normal" telephone in the building or something? |
| 18:46.04 | WIMPy | Hope or alternatively pray? |
| 18:46.53 | WIMPy | Have a cellphone in the first aid kit? |
| 18:53.54 | drab | I'll add those to my options, thank you |
| 19:03.12 | [TK]D-Fender | Prayer : a means of feeling good about yourself for having effectively done nothing. |
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| 19:05.48 | benzrf | [TK]D-Fender: b r a v e |
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| 20:53.14 | danielbw | hello. I am having a lot of trouble troubleshooting a voice quality issue. I have used dahdi_monitor on the same channel I am experiencing the bad voice and the recording is crystal clear. |
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| 20:53.54 | danielbw | The problem only appears for incoming calls from a T1 with 24 e&m channels |
| 20:54.51 | danielbw | If anyone is willing to help me, I'd really appreciate it. |
| 20:55.18 | danielbw | I can give a lot more info, but I will wait until someone is interested. |
| 20:55.45 | danielbw | By the way, outgoing calls are crytsal clear |
| 20:55.51 | danielbw | crystal clear |
| 20:57.39 | danielbw | So, I'm not sure why the call quality would be different and undetectable. |
| 20:58.12 | danielbw | This is my very first time using a VOIP platform, so I'm very green |
| 20:58.20 | WIMPy | What direction do you have the issue? |
| 20:58.27 | danielbw | Incoming |
| 20:58.44 | WIMPy | Not the call, the audio. |
| 20:58.51 | danielbw | oh, rx |
| 20:59.18 | danielbw | the calling party can hear me very clearly, but i can;t hear them |
| 20:59.24 | danielbw | beginning and end of sounds are cut off |
| 20:59.27 | WIMPy | There is no such thing as incomming and outgoing calls. All calls come in one way and go out again on another. |
| 20:59.30 | danielbw | like if they say test |
| 20:59.33 | danielbw | all i hear is eh |
| 20:59.48 | danielbw | well from providor through t1 to asterisk |
| 21:00.03 | WIMPy | If your recording is ok, it must be either the outgoing channel or you found some transcoding issue. |
| 21:00.06 | danielbw | is where i experience the issue |
| 21:00.18 | danielbw | hmm |
| 21:00.30 | WIMPy | No. If you did, it would be bad with dahdi_monitor as well. |
| 21:00.59 | danielbw | ? |
| 21:01.54 | WIMPy | If the recording of dahdi_monitor is ok, the dahdi side is not your issue. |
| 21:02.11 | WIMPy | And you haven't told us about the other end, yet. |
| 21:03.02 | danielbw | forgive me, but i'm not sure what you mean by other side |
| 21:03.13 | danielbw | my sip phone or cell phone? |
| 21:03.47 | WIMPy | Whatever you pass that call on to from Asterisk. |
| 21:04.13 | WIMPy | >>All calls come in one way and go out again on another. |
| 21:04.28 | danielbw | when i call out using my sip phone polycom vvx 310 to my cell phone through the dahdi interface, the sound is crystal clear for tx and rx |
| 21:04.41 | danielbw | when i call from cell to polycom, the polycom rx sounds horrible |
| 21:04.42 | WIMPy | Unless they end up in some application, off course. |
| 21:05.00 | danielbw | my dialplan points dahdi to polycom |
| 21:05.15 | danielbw | and i have en extension that uses a trunk group for dialing out |
| 21:05.55 | danielbw | so i figured it was something to do with my dahdi interface |
| 21:06.02 | danielbw | so i found the dahdi_monitor utility |
| 21:06.05 | danielbw | and recorded it |
| 21:06.21 | danielbw | and the recorded file sounds crystal clear, but the same call sounded horrible |
| 21:06.55 | danielbw | while listening on polycom |
| 21:07.13 | danielbw | so i thought it must be something i need to change in asterisk |
| 21:07.18 | WIMPy | So you should take a look at the SIP side of the calls. |
| 21:07.28 | danielbw | but sip sounds great |
| 21:07.41 | danielbw | and the same call to my cell sounds great |
| 21:07.44 | WIMPy | You just said it doesn't. |
| 21:07.50 | danielbw | it only happens when i call in |
| 21:07.57 | danielbw | from cell |
| 21:08.01 | WIMPy | To the SIP device. |
| 21:08.05 | danielbw | yes |
| 21:08.20 | WIMPy | So take a look at what's going on there. |
| 21:08.25 | danielbw | but i also have slfphone and it eperiences the same exac problem |
| 21:08.44 | danielbw | dialing out sounds awesome |
| 21:08.49 | danielbw | incoming from cell awful |
| 21:08.54 | WIMPy | Take a look at the CODECs used. |
| 21:08.58 | danielbw | i did |
| 21:09.04 | danielbw | so outgoing uses g722 |
| 21:09.18 | danielbw | when i say outgoing i mean from asterisk to the world through my t1 |
| 21:09.19 | danielbw | ok |
| 21:09.26 | danielbw | incoming opposite |
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| 21:09.51 | WIMPy | As dahdi doesn't support G.722 you already have a difference there. |
| 21:10.01 | danielbw | e&m only supports one codec, correct? |
| 21:10.23 | danielbw | how does outgoing not use the same codec? |
| 21:10.27 | WIMPy | So does dahdi. |
| 21:10.39 | danielbw | well, it uses dahdi either way |
| 21:10.42 | WIMPy | Asterisk has to transcode the call. |
| 21:11.01 | danielbw | even if both sides use g711 |
| 21:11.03 | WIMPy | While on the "incomming" call it will try to offe G.711 to your phone(s). |
| 21:11.06 | danielbw | tisnt ulaw g711? |
| 21:11.15 | WIMPy | Yes |
| 21:11.54 | WIMPy | Maybe there's a mixup of alaw/ulaw? |
| 21:12.16 | danielbw | can i paste in here |
| 21:12.25 | WIMPy | ~pb |
| 21:12.25 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:12.29 | danielbw | ok |
| 21:13.44 | danielbw | http://pastebin.com/swfGhU5f |
| 21:13.58 | danielbw | that was while i called from cell to polycom |
| 21:14.56 | WIMPy | You should probably take a close look at the SDP. |
| 21:15.07 | danielbw | sdp? |
| 21:15.15 | WIMPy | That little paste looks ok so far. |
| 21:15.25 | WIMPy | Session Description Protocoll (SIP) |
| 21:16.40 | danielbw | sip.conf ? |
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| 21:17.03 | WIMPy | No, sip debug. |
| 21:17.30 | danielbw | oh |
| 21:17.58 | WIMPy | We already know the difference between a working and a non-working call, but we don't know why soem fail. |
| 21:18.03 | danielbw | isnt it weird that it cuts off at the beginning and end of talking? |
| 21:18.21 | danielbw | it's like it's using some kind of silence suppression |
| 21:18.31 | danielbw | or opposite |
| 21:18.34 | WIMPy | But if your calls incolve dahdi anyway, it might be a good ide to use the same CODEC on the SIP side, so Asterisk doesn't have to transcode. |
| 21:18.54 | danielbw | ok, but i thought i was? |
| 21:19.00 | WIMPy | That might also be an issue. |
| 21:19.05 | danielbw | how do i know if it has to transode |
| 21:19.14 | WIMPy | Not if you try to use G.722. |
| 21:19.27 | danielbw | how\ |
| 21:19.45 | WIMPy | Do you have Silence supression / VAD activated on the phone(s)? |
| 21:19.57 | danielbw | i dont know |
| 21:20.05 | danielbw | but it also happens on sflphone |
| 21:20.14 | WIMPy | Make sure it's off. |
| 21:20.15 | danielbw | so im sure its not a difference |
| 21:20.23 | danielbw | is that in sip.conf? |
| 21:20.45 | WIMPy | No, it's ion the phones config. |
| 21:21.13 | WIMPy | chan_sip doesn't support VAD and does bad things if enabled. |
| 21:21.55 | danielbw | i don't think sflphone even supports it |
| 21:22.00 | danielbw | so that rules that out |
| 21:22.15 | WIMPy | I have no idea if it does. |
| 21:22.25 | WIMPy | But most phones do. |
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| 21:29.05 | danielbw | so how do i go about debugging it |
| 21:29.11 | danielbw | i'm really lost here |
| 21:29.18 | WIMPy | I told you |
| 21:29.45 | danielbw | ok, how do i enable g.722 for an e&m channel |
| 21:29.54 | danielbw | i don't know how to do that |
| 21:29.56 | WIMPy | Not possible. |
| 21:30.03 | danielbw | ok, so i can't do that |
| 21:30.13 | WIMPy | Look at the SDP. |
| 21:30.17 | danielbw | we already know that my phones support ulaw, so no transcoding is taking place |
| 21:30.27 | WIMPy | Ensure you don't have VAD. |
| 21:30.28 | danielbw | ok, what utility provides that |
| 21:30.30 | danielbw | or command |
| 21:30.46 | WIMPy | 'sip set debug on' |
| 21:31.10 | WIMPy | Or tools like tcpdump/wireshark/sngrep/... |
| 21:31.31 | WIMPy | wireshark would even be able to analyze the audio. |
| 21:33.46 | danielbw | ok |
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| 21:33.49 | danielbw | ill be right |
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| 21:37.55 | danielbw | back |
| 21:37.58 | danielbw | lol |
| 21:37.59 | danielbw | not yet |
| 21:49.09 | danielbw | ok, on wireshark |
| 21:49.12 | danielbw | what is wring timestamp |
| 21:49.16 | danielbw | could that be causing it |
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| 21:49.32 | danielbw | wrong timestamp |
| 21:49.34 | danielbw | about 2% |
| 21:50.32 | danielbw | i was able to capture it |
| 21:50.34 | danielbw | and play it back |
| 21:50.45 | danielbw | the playback is exactly how it sounds on the phone |
| 21:50.51 | danielbw | shoppy |
| 21:51.34 | danielbw | im going to make a series of files a compressed file that could help debug this |
| 21:52.04 | danielbw | ill make dahdi_monitor wav files, and capture the convo at same rtime |
| 21:52.16 | danielbw | and ill make a copy of the capture |
| 21:52.23 | danielbw | would you be able to look att his info |
| 21:52.50 | danielbw | man my typing skills suck today |
| 21:53.00 | WIMPy | Better past the SIP debug for someone to take a look, or perhaps the meta date of all traffic. |
| 21:53.23 | danielbw | yeah ill make it send the dubg info to a log file too |
| 21:53.27 | danielbw | good idea |
| 21:53.43 | WIMPy | I'm not the SIP expert. I prefer to look in to things that work. |
| 21:54.02 | danielbw | k |
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| 22:03.26 | danielbw | i can see on the rtp lots of complete silence. shouldn't it be a complete audio stream |
| 22:04.03 | WIMPy | It should be the same as you got from dahdi_monitor. |
| 22:04.30 | WIMPy | What about the other direction? I.e. from SIP phone to Asterisk. Is that stream complete? |
| 22:05.01 | danielbw | i have my mic muted on that one |
| 22:05.03 | danielbw | for testing |
| 22:05.13 | danielbw | i can test |
| 22:05.34 | WIMPy | did you make sure that you don't have VAD activated? |
| 22:12.49 | danielbw | i dont know how to do that |
| 22:13.10 | danielbw | but wouldnt it be on all calls not just some |
| 22:13.23 | danielbw | hehe |
| 22:13.25 | WIMPy | Check the phones manual |
| 22:13.25 | danielbw | more testing |
| 22:13.32 | danielbw | the sflphone |
| 22:13.51 | WIMPy | Probably. Not sure how CODEC selection / transcoding affect timing. |
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| 22:18.35 | danielbw | well, the silence is being injected somehow |
| 22:18.38 | danielbw | i dont know |
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