| 00:00.10 | p3nguin | I thought it would be there, but I didn't see what I expected to see. |
| 00:00.11 | WIMPy | "30 * reconnects_per_second" That should be it. |
| 00:00.18 | pc500 | Is anyone aware of a callbackworld-style script? Purpose of the script is to call back your phone # when you dial and hang up or send a SMS to a given number. Typically used for free incoming calls on international sim cards. |
| 00:00.33 | pc500 | where the per minute rates are otherwise gouging |
| 00:01.09 | [TK]D-Fender | There is no "script".. this is maybe 3 lines of dialplan |
| 00:01.16 | WIMPy | pc500: Have you tried to ask Dr. Google? |
| 00:03.54 | pc500 | WIMPy: - SMS bits might be a bit more complicated, but I suppose the call your cell phone and ring whoever was last on the caller id isn't as hard. |
| 00:10.05 | WIMPy | On the lastest versions I get a 20s delay between chan_iax2 and chan_sip. Does someone have a clue where that might come from? |
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| 01:01.44 | SeRi | p3nguin: My choppines with pstn is not me. is voip.ms |
| 01:19.42 | p3nguin | oh? |
| 01:20.01 | p3nguin | But you tested it with call centric, too. |
| 01:21.00 | p3nguin | My ISP has had some serious packet loss today. Calls have sounded pretty horrible... some even failed over due to asterisk being unreachable. |
| 01:24.36 | p3nguin | I don't see anything in the issue tracker regarding your complaint. |
| 01:26.48 | p3nguin | Ugh. Gmail is even throwing errors due to connectivity problems. |
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| 01:53.24 | MrTelephone | does anyone have a problem with sip realtime and pap2t voicemail indicator disappearing after an hour? |
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| 02:47.43 | saisoma | evening folks! can I have two sip trunks between two * servers? I want to use one trunk for LD and one trunk for local calls (separate digium cards for LD and local) |
| 02:48.09 | WIMPy | ~trunk |
| 02:48.09 | infobot | rumour has it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 02:48.31 | WIMPy | Shouldn;t that be done in the dialplan? |
| 02:49.27 | saisoma | WIMPy: it was originally. I had one * box where phones register (virtual) and did the split there. There were two physical * boxes, one for PRI (local calls) and one for LD calls. That was easy. Now the PRI and LD boxes have been combined. |
| 02:50.13 | WIMPy | And why do you want to make the decision concering that box on another box? |
| 02:50.29 | saisoma | I'd rather keep the logic at the main * box where phones register. Trying to be lazy and just use the same code, have the calls land in the appropriate context |
| 02:51.02 | WIMPy | You can create as many accounts as you like. |
| 02:51.20 | WIMPy | And IAX lets you pass a context with only one account. |
| 02:52.53 | saisoma | gotcha. I think I'll sleep on this and tackle it in the morning (20 hour day so far). Thanks guys! |
| 02:54.37 | MrTelephone | is there a failover feature in myodbc unixodbc? |
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| 04:22.31 | SeRi | p3nguin: you in? |
| 04:22.32 | p3nguin | SeRi, Leave a message after the beep. *BEEP* |
| 05:06.28 | p3nguin | This is friggin' ridiculous. http://pastebin.com/nDJ0xD7L |
| 05:06.37 | SeRi | ? |
| 05:07.33 | p3nguin | I don't know what happened, but my internet quality went completely to hell today. |
| 05:07.33 | SeRi | p3nguin: I am in teh same boat with comcast! |
| 05:07.50 | SeRi | Is been like that for me for some time now |
| 05:07.58 | p3nguin | I thought it was the ISP having packet loss, but it isn't. |
| 05:09.22 | SeRi | :/ |
| 05:11.19 | SeRi | damn this is so bad |
| 05:11.30 | SeRi | voip.ms is having issues also |
| 05:12.14 | SeRi | my incoming been choppy is voip.ms issues with the houston server |
| 05:12.21 | SeRi | whcih server do you use? |
| 05:13.58 | p3nguin | chicago |
| 05:15.05 | SeRi | any issues? |
| 05:15.39 | p3nguin | With packet loss like is reported in the pastebin, do you really think I can test voice calls? |
| 05:16.05 | SeRi | :/ |
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| 05:16.39 | p3nguin | I've got 13-25% packet loss at every single hop. |
| 05:17.28 | SeRi | PING OK - Packet loss = 16%, RTA = 53.18 ms |
| 05:17.38 | SeRi | Thats to voip.ms houston and call centric |
| 05:18.37 | SeRi | WTF! 2052ms / 2000ms |
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| 05:19.59 | p3nguin | Okay, I just finished testing two modems in one location, and they have 0% packet loss to chicago. Two modems in the other building have all kinds of loss. Must be something wrong outside. I guess I need to go to other people in the building and see who is using the same ISP. |
| 05:23.37 | SeRi | I see. |
| 05:26.10 | p3nguin | If there is a customer using Charter telephone service, they'll be more likely to fix it right away. |
| 05:26.38 | SeRi | I am sure.... Thats my excuse with comcast |
| 05:29.51 | p3nguin | Was your video service affected by that tap being bad? |
| 05:31.01 | p3nguin | Or was the tap not really bad? |
| 05:32.28 | SeRi | the tap was bad. I am not sure now was goign on..... I dont have comcast tv so I am not sure about video |
| 05:32.44 | p3nguin | oh |
| 05:33.10 | p3nguin | I thought you had the voice/video/data bundle. |
| 05:33.45 | SeRi | voice/data only |
| 05:38.55 | p3nguin | Ping times are still really good. |
| 05:39.10 | SeRi | mhhhhhhhh |
| 05:39.27 | p3nguin | 100 packets transmitted, 86 received, 14% packet loss, time 99150ms |
| 05:39.27 | p3nguin | rtt min/avg/max/mdev = 17.689/19.177/32.938/1.997 ms |
| 05:41.32 | SeRi | i see |
| 05:41.58 | p3nguin | 19ms is fine by my standards. |
| 05:42.20 | SeRi | once it reaches 50 I am worried |
| 05:42.47 | p3nguin | But I can't survive the packet loss. |
| 05:42.53 | p3nguin | It ruins a call. |
| 05:43.43 | SeRi | it does. big time |
| 05:43.48 | SeRi | Thats my issue :( |
| 05:43.53 | p3nguin | I've got one 1 out of 10 peers registering successfully. |
| 05:44.32 | SeRi | :/ |
| 05:44.40 | p3nguin | They keep trying and just time out. |
| 05:44.48 | SeRi | :( |
| 05:44.55 | p3nguin | And now that 1 of 10 dropped. |
| 05:45.42 | SeRi | call centric is going crazy droping on and off |
| 05:47.11 | p3nguin | They either say Request Sent or Unregistered. |
| 05:47.50 | SeRi | :/ |
| 05:51.42 | p3nguin | Ooooo, now a phone managed to become reachable. |
| 05:51.58 | p3nguin | Not much good it will do with no ITSPs available. |
| 05:56.27 | SeRi | :( |
| 06:01.14 | p3nguin | I guess I'm done worrying with it for the night. |
| 06:01.25 | phix | p3nguin: You still having issues? |
| 06:01.30 | phix | Or something else now? |
| 06:03.34 | p3nguin | I just starting having issues today. |
| 06:03.39 | p3nguin | Packet loss is insane high. |
| 06:03.59 | p3nguin | started |
| 06:04.06 | ChannelZ | Is it only one way? |
| 06:05.28 | p3nguin | I didn't try to test that. I first noticed it by ssh being lagged all to hell, then pings an mtr show loss. |
| 06:05.58 | p3nguin | Not to mention phone calls are non-existent now. |
| 06:06.06 | ChannelZ | When it happens to me I have massive packet loss on my outgoing but not os much on my incoming |
| 06:06.29 | p3nguin | I'll do some tests on that theory in the morning. |
| 06:06.47 | ChannelZ | And in my case it's someone else in the neighborhood hooking up something ghetto to their cable which backwashes into everyone on our run |
| 06:06.51 | SeRi | I have a line monitor setup at dslr and I have some insane noise on my lines.... |
| 06:06.58 | SeRi | packet loss* |
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| 07:19.20 | Wil_Syd | What tool can I use to alter a sip message? I need it to sit between my VSP and ATA and inject a rtpmap attribute in the 200 OK message after call setup. |
| 07:34.19 | SeRi | p3nguin: setup a ping monitor and see how much loss you have from one point to you |
| 07:34.31 | SeRi | I what I have been doing for a few days |
| 07:34.57 | SeRi | I have an avarage of 10% to 25% loss to me |
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| 08:17.57 | nafg_ | What's the simplest way to get the phone number of the part called? |
| 08:18.20 | nafg_ | I originate a call with AMI and I want to get the number I called. |
| 08:20.35 | nafg_ | Anyone around? |
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| 08:56.30 | nafg_ | I originate a call with AMI and I want to get the number I called. |
| 08:56.33 | nafg_ | What's the simplest way to get the phone number of the party called? |
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| 12:03.06 | Wil_Syd | is rtpmap required? |
| 12:03.14 | Wil_Syd | in SDP |
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| 12:50.42 | IsUp | hello |
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| 12:59.38 | Dovid | hi |
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| 15:38.10 | hexad0m | hi all |
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| 16:13.14 | hexad0m | hello all!! |
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| 16:21.16 | MrTelephone | I started using sip realtime with rtcache=yes and asterisk isn't polling the voicemail messages and sending notify to peers. Is there something I could be missing? |
| 16:21.48 | MrTelephone | I'm using version 1.8.5 |
| 16:25.11 | p3nguin | seri: All fixed. http://pastebin.com/mYZWUaem |
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| 16:30.20 | SeRi | p3nguin: what was the issue? |
| 16:30.32 | p3nguin | packet loss |
| 16:30.40 | SeRi | caused by? |
| 16:31.01 | p3nguin | No clue. I went to bed because I was tired of worrying over it. Got up, and it was fixed. |
| 16:31.06 | MrTelephone | penguin, you don't know about mwi and realtime do you ? |
| 16:33.06 | SeRi | p3nguin: I whish my issue was magical like that. |
| 16:33.20 | MrTelephone | what is your issue seri? |
| 16:33.20 | SeRi | I went to bed... woke up and the issue is worst... |
| 16:33.33 | p3nguin | I had registration timeouts every minute or so all the way up to 01:56. After that, I just see successful phone calls. Either the ISP went out and fixed it at 2 AM or whoever plugged in some type of junk equipment finally unhooked it. |
| 16:33.35 | SeRi | Insane snr's in my channels |
| 16:33.50 | SeRi | comcast ISP* |
| 16:33.58 | MrTelephone | coax? |
| 16:34.00 | MrTelephone | RF? |
| 16:34.14 | SeRi | yes. but the issues is not on my permisses. |
| 16:34.30 | MrTelephone | what numbers are you seeing? |
| 16:34.30 | SeRi | I actually whish it was so I could fix it |
| 16:34.40 | SeRi | negative numbers |
| 16:34.58 | SeRi | last test my upload channel had -20.7 |
| 16:35.10 | MrTelephone | unheard of |
| 16:35.17 | MrTelephone | you must be losing a lot of data |
| 16:35.22 | SeRi | thats insane in my book |
| 16:35.38 | MrTelephone | my upstream network will become unusable at 20 |
| 16:35.43 | MrTelephone | should be 30 |
| 16:35.45 | SeRi | ha! loosing? I am unable to do anything! |
| 16:35.53 | MrTelephone | yeah exactly |
| 16:35.55 | SeRi | it fluctuates |
| 16:36.09 | SeRi | now is bleeding to my downstream channels |
| 16:36.17 | SeRi | indeed! 30 is what we are looking for |
| 16:36.18 | MrTelephone | i've never seen negatives. There is a cable break or bad amplifier in the system |
| 16:36.44 | MrTelephone | that's pretty serious |
| 16:36.55 | SeRi | MrTelephone: I agree. Though comcast is actually trying to fix it thats why I am not ranting. |
| 16:37.09 | MrTelephone | I mean the NOC should be able to pinpoint the problem if all the modems on your leg are doing the same thing |
| 16:37.11 | SeRi | they came and change several tap's arounf my naighborhood that had issues |
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| 16:37.34 | MrTelephone | I have a similar problem where I work and I don't know where the problem is |
| 16:37.52 | SeRi | You work for an ISP? |
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| 16:38.12 | MrTelephone | yeah I run a small cable system |
| 16:38.16 | SeRi | ping times to callcentric are showing 2059ms |
| 16:38.22 | SeRi | nice! |
| 16:38.36 | SeRi | This is insane :( those are nasty ping times |
| 16:38.40 | MrTelephone | it's difficult to maintain because the weather is really bad here. Rain one day and ice the next |
| 16:38.46 | SeRi | ouch |
| 16:39.08 | MrTelephone | what size packets do you ping with? |
| 16:39.25 | SeRi | default size. this are reports from icinga |
| 16:39.46 | SeRi | I have a monitor set from dslr pinging me and is reporting 10 to 25% packet loss |
| 16:39.54 | SeRi | constant |
| 16:40.04 | MrTelephone | I'm surprised your modem comes online |
| 16:40.05 | SeRi | the issue is both ways |
| 16:40.34 | SeRi | It reboots every so often.... when it gets very bad on the Down stream channels.... if a channel goes belly up it reboots |
| 16:41.30 | MrTelephone | I was talking to another tech one time and they shut down a street to isolate a problem. They said it was a pair of shitty computer speakers near the modem injecting noise. |
| 16:41.37 | MrTelephone | It could be anything |
| 16:41.43 | p3nguin | My packet loss was bad enough last night that once when I power cycled it, it did not go back online. |
| 16:42.08 | SeRi | :( |
| 16:42.26 | MrTelephone | the snr values on the modem are calculated so if you have very little throughput and some of it goes bad you will see very low numbers |
| 16:42.29 | SeRi | I whish I would be fully off line. it would be easier for the to trouble shoot |
| 16:42.42 | MrTelephone | you need a spectrum analyzer to see what is really happening :( |
| 16:42.46 | *** join/#asterisk hexad0m (~greg@host-78-145-211-246.as13285.net) |
| 16:43.29 | MrTelephone | The coax connections are not meant to come apart and are usually sealed with heatshrink so it's a lot of work to take everything apart to see i there is a bad connetion :( |
| 16:43.32 | SeRi | I had one tech and two eng in my home.... one eng check everything around the area and found several tpas with issues including mine. they said that noise was getting in due to bad taps |
| 16:43.59 | SeRi | they changed them. |
| 16:44.09 | SeRi | and the issue was fixed for a few days |
| 16:44.19 | SeRi | now is back |
| 16:44.20 | MrTelephone | yeah they get water in them and corrosion ruins the RF |
| 16:44.42 | SeRi | one eng check my home for all kinds of issue and found nothing. he said my home is clean as it can get |
| 16:44.44 | p3nguin | I guess it was just coincidental for me. I was out in the field trying to correct unwanted behavior (caused by dial plan), so I sshed into the asterisk system... and made a couple phone calls to debug where the issue was. Calls started breaking up, and ssh started freezing. |
| 16:44.44 | MrTelephone | are you in an urban area or suburb? |
| 16:45.17 | MrTelephone | Voip is a good tool for testing network stability isn't it |
| 16:45.25 | SeRi | indeed |
| 16:45.36 | SeRi | youw ill catch the smallest issues quick |
| 16:45.42 | MrTelephone | I had to change all kinds of taps and amps to get voip working where I work. Internet worked ok with a degraded system |
| 16:45.59 | p3nguin | By the time I was done fixing the dial plan, I couldn't make any more phone calls because the packet loss was so bad. |
| 16:46.04 | SeRi | MrTelephone: I am in the suburbs... I am in the FM of texas... way outside the city :) |
| 16:46.06 | MrTelephone | brutal |
| 16:46.20 | SeRi | p3nguin: :( |
| 16:46.25 | MrTelephone | seri, i'm sure they know what they are doing though they'll get it fixed |
| 16:46.49 | MrTelephone | I wasn't trained by professionals and I do have a hard time to find where problems are. Even with the right tools it is difficult :( |
| 16:47.16 | SeRi | MrTelephone: I am sure. I just whish they where quicker about it. |
| 16:47.25 | MrTelephone | Power values are usually more valuable then the SNR |
| 16:47.46 | MrTelephone | downstream should be -5 to 15 and upstream should be 35-45 or something close |
| 16:48.19 | p3nguin | We aim for 0 and 50. |
| 16:48.33 | p3nguin | -10 to 10 is acceptable. |
| 16:49.17 | MrTelephone | yeah it all works, but what kills it is power fluctuations |
| 16:49.42 | MrTelephone | seri, what are your power levels? |
| 16:50.15 | MrTelephone | I wonder if those fiber node screw up alot? |
| 16:50.31 | SeRi | -8.x on down stream and 36 on upload |
| 16:50.43 | MrTelephone | very unporportional |
| 16:50.57 | MrTelephone | if upload is 36 your downstream should be like 10 |
| 16:51.20 | p3nguin | One time I was in Florida trying to troubleshoot a modem service, and I called Bright House to see what they could tell me. When I told them back home we aim for 0 and 50, he just laughed and said they NEVER see signal like that. |
| 16:51.33 | MrTelephone | lol |
| 16:51.39 | SeRi | indeed. the techs are stun about how my numbers "snr's" do not match the system. |
| 16:51.44 | MrTelephone | well i shoot for 5 because it's in the middle |
| 16:51.55 | SeRi | I am not surprise they do not match one anotehr on power |
| 16:52.07 | MrTelephone | One guy told me to shoot for high upstreams to push up the snr but it never worked. I ran at 50-55 for a while |
| 16:52.13 | p3nguin | 0 is in the middle of our -10 to 10 range. |
| 16:52.46 | MrTelephone | docsis2 min ds is -5 i think |
| 16:52.55 | p3nguin | nah |
| 16:53.02 | p3nguin | It'll run far below that. |
| 16:53.24 | MrTelephone | if you run them at 0 you shouldn't see a change that make them dip to -5 |
| 16:53.38 | MrTelephone | unless someone moves their equipment in there house, bastards |
| 16:54.06 | MrTelephone | I don't know why people always call down each others shit |
| 16:54.07 | *** part/#asterisk hexad0m (~greg@host-78-145-211-246.as13285.net) |
| 16:54.21 | MrTelephone | they laughed at you but he probably didn't even know what power levels are |
| 16:55.29 | MrTelephone | I started seeing problems at -7 with arris |
| 16:56.00 | MrTelephone | but if you get that low there may be some other issue like a loose cable connection |
| 16:56.01 | p3nguin | Today, I'm running at -3, -2, -1, and -1 on downstream channels, with SNR of 34, 34, 35, and 36. 45 upstream power. |
| 16:56.35 | MrTelephone | 45 is a good number |
| 16:57.00 | MrTelephone | I cannot run qam16 on my upstream though. I don't know if it's my shitty cmts or line issues ;( |
| 16:57.33 | MrTelephone | I average 29-30 on upstream snr using qpsk 3.2mhz width but if I switch to qam16 everything shuts down |
| 16:57.38 | p3nguin | My modem shows 16QAM and 64QAM on upstream. |
| 16:57.56 | MrTelephone | yeah those modulation profiles are super sensitive to interference |
| 16:58.14 | MrTelephone | you get 10mbit per channel using 16qam |
| 16:58.37 | p3nguin | I'm using 33000000 Hz upstream. |
| 16:58.37 | MrTelephone | 64qam is crazy |
| 16:58.54 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 16:59.02 | MrTelephone | I have this antique cisco 7200 ubr and I blame it on that piece of shit |
| 16:59.14 | MrTelephone | the newer equipment has better noise avoidance |
| 16:59.57 | MrTelephone | 16qam is 6mhz wide so it runs from 30000000 to 3600000 |
| 17:00.12 | p3nguin | I'm going to run a 10 minute mtr for a report. If it is still 0.0% loss, I'd say the problem is gone. |
| 17:00.57 | MrTelephone | too bad environmental problems like that couldn't be prevented :( |
| 17:01.31 | p3nguin | channelz suspected that someone in the complex plugged in some crap that ruined service. |
| 17:01.51 | p3nguin | Since it magically went away at ~2AM, I'm inclined to agree. |
| 17:03.38 | SeRi | I am running one right now... odly enough it shows 0% loss :/ my last ping time of 2959ms was at 850AM |
| 17:05.03 | *** join/#asterisk psharmor (~tekoholic@97-118-255-195.hlrn.qwest.net) |
| 17:06.05 | MrTelephone | i have this reflectomoter tool to measure kinks in the cable or whatever and i plugged it into a live network by accident. I had the spectrum on and when I ran the scan it wiped out everything |
| 17:06.25 | MrTelephone | it sends pulses at every frequency and reads any reflections |
| 17:07.08 | MrTelephone | or something is stealing service from their neighbor by jabbing a pin into their cable and connecting their cable to it |
| 17:07.09 | MrTelephone | haha |
| 17:07.27 | SeRi | lol |
| 17:07.36 | SeRi | MrTelephone: where you @? |
| 17:07.43 | MrTelephone | northern ontario |
| 17:08.19 | MrTelephone | people break the ends off their cable and just jab the copper part in there tv and shit. I'm sure that attracts some rogue signals |
| 17:08.58 | MrTelephone | we have shaw cable here and if they see anyone using those quick push on cables they cut the in half and make them a real patch cord |
| 17:09.47 | MrTelephone | seri, wall plates sometimes go bad too when the contacts inside the f connector lose their springyness |
| 17:10.14 | MrTelephone | did you try a new wall plate? |
| 17:10.15 | p3nguin | Why would someone cut off a connector just to stick the wire in the jack? |
| 17:10.43 | MrTelephone | the connector rips off because they move their tv from the wall to get at it but the cable is barely long enough to reach the tv from the jack |
| 17:11.00 | MrTelephone | so they don't have the tools to put a new end on so they duct tape it to the connector |
| 17:11.22 | SeRi | MrTelephone: I run deirectly from the tap to my network rack. there is one cylinder in the attic and is using weather proof connectrs and they all have been chaged. along with new wiring from the outside in just in case. |
| 17:12.05 | p3nguin | Is that the same tap they just replaced? |
| 17:12.05 | MrTelephone | I had this vision that you were running a few servers in your bedroom :P |
| 17:12.19 | SeRi | p3nguin: yes |
| 17:12.30 | p3nguin | How was the SNR before they changed the tap? |
| 17:12.30 | SeRi | MrTelephone: lol something like that.... hehehehe |
| 17:12.38 | SeRi | about the same |
| 17:12.56 | p3nguin | I've had people change out parts and service gets worse. |
| 17:13.29 | MrTelephone | did your weather shift or something? |
| 17:13.38 | SeRi | well that happen to me. becausemy download channel had no issues before |
| 17:13.46 | SeRi | now I have issues in the DL channels |
| 17:14.05 | MrTelephone | you need to put a meter on it if you have one |
| 17:14.21 | MrTelephone | did the cable guy ever plug in his meter in front of you? |
| 17:14.34 | *** join/#asterisk wesphillips (~wphill04@75.53.136.233) |
| 17:15.00 | SeRi | Yes. he did some readings but I didnt see teh screen. he told me everything inside the house was perferct |
| 17:15.16 | MrTelephone | -8 is perfect? |
| 17:15.36 | p3nguin | The meter doesn't show the same reading that a modem shows. |
| 17:16.08 | MrTelephone | some meters are pretty shitty |
| 17:16.09 | p3nguin | At least the meters that our techs carry don't. |
| 17:16.21 | SeRi | I would not know. I dont know much about how they read it. |
| 17:16.33 | p3nguin | It shows a signal, but it isn't the same as what you see on the modems. |
| 17:16.40 | MrTelephone | if your meter supports qam signals you just point it to your modem downstream frequency |
| 17:17.10 | MrTelephone | and then some meters have a constellation graph so you can see phase shift and aplitude shifting |
| 17:17.38 | p3nguin | http://pastebin.com/5Sh3HnfX |
| 17:17.44 | p3nguin | Not bad. |
| 17:17.50 | MrTelephone | we didn't have enough money to buy a spectrum (rented one once) but I have a Sunrise Telcom CM1000 and that thing is a good all around tool |
| 17:18.43 | p3nguin | With as many techs as there are, there is no way that everyone can have the best all-around meter. |
| 17:18.53 | MrTelephone | is it possible that your packets were dropped due to congestion? |
| 17:19.04 | MrTelephone | penguin, i agree |
| 17:19.14 | MrTelephone | you usually just need to know the power levels right? |
| 17:19.23 | p3nguin | Usually. |
| 17:19.35 | p3nguin | If there is enough signal, things usually work. |
| 17:19.37 | SeRi | MrTelephone: It could be. I was also told by voip.ms that they been having issues with their routes |
| 17:19.49 | p3nguin | If there is no, escalate to a higher level tech who has a better meter. |
| 17:19.56 | p3nguin | s/no/not/ |
| 17:20.35 | MrTelephone | took me an hour once to diagnose a shit connection once. All it was, was 1 strand of sheilding touching the copper core :( |
| 17:20.43 | p3nguin | We do have something that's pretty cool... |
| 17:21.38 | p3nguin | When someone with the need-to-know is on site, they can tune the TV to a certain channel and see the output from the HP 85721A analyzer. |
| 17:22.05 | MrTelephone | that would be handy |
| 17:22.19 | MrTelephone | those things are like 14 thousand though :( |
| 17:22.20 | p3nguin | You just call in and ask the engineer to change it to whatever node you are on, and it'll show you your node's data. |
| 17:22.30 | SeRi | p3nguin: comcast use to have a channel with that info. they took it out :( |
| 17:22.36 | MrTelephone | do you use calan meters? |
| 17:23.01 | MrTelephone | https://tech.picriverisp.net/ |
| 17:24.22 | MrTelephone | I have a lot of uncorrectable errors at certain times of day. It's relational to he SNR but very hard to pinpoint. Caused by an aggregate of noise issus an unknown location. lol |
| 17:24.27 | SeRi | I wish I could get that data woith icinga :( |
| 17:24.38 | MrTelephone | Don't give the cable guys a hard time because those problems are bullshit o solve |
| 17:24.45 | SeRi | lol |
| 17:24.46 | MrTelephone | haha |
| 17:24.51 | MrTelephone | alright talk to you later |
| 17:24.55 | SeRi | cya! |
| 17:26.38 | MrTelephone | 120 people downloading an average of 1 TB a day |
| 17:27.17 | MrTelephone | people in the sticks don't have anything to do but download shit |
| 17:39.46 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 17:40.39 | *** join/#asterisk hehol (~Adium@ip-178-202-243-147.unitymediagroup.de) |
| 17:42.49 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 17:45.07 | *** join/#asterisk zkn (~zkn@82.131.32.65.cable.starman.ee) |
| 17:51.22 | p3nguin | seri: chicago is showing 45% loss now. |
| 17:51.36 | p3nguin | ubiquityservers |
| 17:51.39 | SeRi | I wonder whats going on with them |
| 17:52.07 | SeRi | a tech told me they had issues on the houston server |
| 17:52.43 | SeRi | I am cant troubleshoot properly with this cable issues |
| 17:53.16 | SeRi | my damn modem has snmp diable. I found the admin page and everything s diable. the damn thing uses POTD and cant crack it so far |
| 17:53.34 | p3nguin | I've got over 80% loss on 0.ge1-1.tsr1.ord1.us.voxel.net, which is between me and the newyork pop. |
| 17:53.44 | p3nguin | 88% |
| 17:53.48 | p3nguin | 89% |
| 17:53.51 | p3nguin | Still going up. |
| 17:57.17 | SeRi | going to test |
| 17:57.36 | SeRi | p3nguin: can you call me via pstn? |
| 17:57.57 | p3nguin | Yes. |
| 17:58.00 | SeRi | shoudl be free |
| 17:58.07 | SeRi | number coming up |
| 17:58.13 | p3nguin | What makes it free? |
| 17:58.21 | SeRi | voip.ms voip.ms? |
| 17:58.28 | p3nguin | That's not via PSTN. |
| 17:58.35 | SeRi | oooo damn |
| 17:58.36 | SeRi | lol |
| 17:58.37 | SeRi | yea |
| 17:58.40 | SeRi | ok one |
| 17:58.42 | SeRi | sec |
| 17:59.15 | SeRi | ok thats the number |
| 17:59.43 | SeRi | I can hear people ok but people heare me like shit |
| 18:11.29 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 18:13.41 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 18:14.04 | nny | quick q, as of 1.8 is adaptive odbc still needed to add custom fields to the cdr db? |
| 18:16.41 | *** join/#asterisk irroot (~gregory@41.53.4.138) |
| 18:17.45 | *** join/#asterisk hesco (~hesco@c-24-98-15-67.hsd1.ga.comcast.net) |
| 18:20.08 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 18:21.05 | hesco | I worked with customer support at a new provider the other day to configure a new trunk. we finally got it working to route inbound calls on the new did, but when it was all said and done, all of my phones had lost their registrations. The support desk at this new provider had access to my desktop and through that my credentials to my telephony server. I'm wondering what he might have changed which would have made these polycom's fail t |
| 18:21.07 | hesco | register. Any ideas? |
| 18:23.51 | ChannelZ | fail in what way? Are you seeing failed registrations on Asterisk? |
| 18:24.05 | ChannelZ | In which case what reason is it giving |
| 18:25.54 | hesco | so far, I have only seen that they are no longer registered in sip show peers |
| 18:27.10 | hesco | I've been on the road all weekend and unable to tend to this until today. I'm running out to the car to grab a phone and see what happens in I `sip set debug peer <phone>` |
| 18:27.14 | kolyaflash | Somebody uses a web interface for Asterisk except Asterisk-GUI? |
| 18:28.44 | ChannelZ | hesco: well then I'd check the phones.. reboot them, make sure they are trying to register to the right hostname/IP, and if your Asterisk is or is behind a firewall that something didn't get changed there |
| 18:29.38 | [TK]D-Fender | kolyaflash: I think you've got that backwards..... |
| 18:30.16 | [TK]D-Fender | kolyaflash: You can double AsteriskGUI's userbase by simply installing it. |
| 18:33.00 | p3nguin | haha |
| 18:33.35 | kolyaflash | [TK]D-Fender: The fact that I had the idea to do something similar, but much easier for beginners and user-friendly. |
| 18:33.47 | hesco | ChannelZ: thanks, I'm setting up my phone now. I already had others on the network reboot their phones. I have changed nothing with my pbx (beyond working with support on that new trunk), though perhaps the folks I co-lo it with may have done something. The pbx continues to process inbound calls. Only now the phones do not register. |
| 18:33.59 | p3nguin | ~cluepon |
| 18:33.59 | infobot | [~cluepon] Get yours now! http://www.bored.com/photos/cluepon_4.html |
| 18:34.04 | p3nguin | [tk]d-fender: Remember that?! |
| 18:34.47 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 18:35.00 | kolyaflash | [TK]D-Fender: not bycicle. It is need for the customer. |
| 18:36.23 | [TK]D-Fender | kolyaflash: ...HUH? |
| 18:36.46 | hesco | ok, phone is now booting up, and to start I see: sip set debug peer 21 |
| 18:36.46 | hesco | Unable to get IP address of peer '21' |
| 18:37.34 | kolyaflash | [TK]D-Fender: I'm sorry. I do not know much English, so forget it. |
| 18:37.47 | kolyaflash | hesco: NAT |
| 18:37.48 | [TK]D-Fender | hesco: Because it hasn't registered. Enable GLOBAL SIP debug and don't try to restrict your view |
| 18:38.14 | hesco | ok, trying that now |
| 18:42.39 | hesco | can the *CLI> noise be redirected to a log somehow so I can more easily search it? |
| 18:43.06 | hesco | so far I'm mostly seeing noise related to trunk registrations |
| 18:45.52 | hesco | This phone has registered line 1 to another pbx w/o issue, but is not registering lines 2 and 3 to the pbx in question |
| 18:48.42 | hesco | ok, inspecting /var/log/asterisk/full the last I see of this IP address was Decmber 4th, my last time at this desk. |
| 18:49.00 | hesco | But nothing about a registration attempt today. |
| 18:49.20 | hesco | [Dec 4 02:11:15] VERBOSE[29813] chan_sip.c: -- Registered SIP '21' at 24.98.15.67 port 5060 |
| 18:49.21 | hesco | [Dec 4 02:11:15] NOTICE[29813] chan_sip.c: Peer '21' is now Reachable. (85ms / 2000ms) |
| 18:49.47 | *** join/#asterisk kriegerod (~krieger@79.135.222.22) |
| 18:51.58 | *** join/#asterisk cerberus_za (~coert@8ta-151-22-106.telkomadsl.co.za) |
| 18:52.45 | kolyaflash | hesco, try "asterisk -vvvvv" |
| 18:53.00 | hesco | the browser interface to the phone reports that line 2 is still pointed at my pbx' IP |
| 18:54.37 | hesco | kolyaflash: ok, did that and have now rebooted phone again, watching console |
| 18:57.19 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 18:57.28 | hesco | ok, did that and still see the December 4th as the most recent entry with this IP, when searching backwards from the end of /var/log/asterisk/full |
| 18:58.29 | kolyaflash | hesco, are you see " Registered SIP" in CLI ? |
| 18:59.55 | kolyaflash | try "server*CLI> core set verbose 3" and "server*CLI> core set debug 4" |
| 19:00.20 | kolyaflash | and are you sure that the phone really connected? |
| 19:00.25 | hesco | Most recent entry reads: |
| 19:00.29 | hesco | [Dec 15 11:20:25] VERBOSE[29813] chan_sip.c: -- Registered SIP '21' at 76.109.144.184 port 5060 |
| 19:00.50 | hesco | that was before I travelled and before this issue surfaced |
| 19:01.31 | hesco | its a Polycom 501, three lines. It provisioned, and registered line one to another server, but lines two and three remain un-registered |
| 19:02.17 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 19:03.25 | kolyaflash | Have you changed anything in config after that? |
| 19:04.48 | hesco | not unless my support tech did when I was distracted, but I tried to watch what he was up to during the desktop sharing session |
| 19:05.04 | hesco | should I be able to telnet to port 5060 on the pbx? |
| 19:05.31 | hesco | because I get when I try: telnet: Unable to connect to remote host: Connection refused |
| 19:05.32 | kolyaflash | hesco, yes you should |
| 19:05.52 | kolyaflash | only if asterisk run on port 5060 |
| 19:05.54 | hesco | perhaps a firewall issue then, checking that |
| 19:06.02 | hesco | yes, default ports |
| 19:06.51 | kolyaflash | try set on client different port as 5060. like 5061 |
| 19:07.10 | hesco | -A INPUT -p tcp -m tcp --dport 5060 -j ACCEPT |
| 19:07.17 | hesco | I'd think that would work |
| 19:08.00 | hesco | telnet 66.163.225.30 5060 and 5061 both fail from here with that error. |
| 19:08.01 | kolyaflash | 'nmap yourserver.com -p 5060' from client |
| 19:08.09 | hesco | ok, doing that now |
| 19:08.29 | p3nguin | ~tcpfail |
| 19:08.29 | infobot | VoIP uses UDP, not TCP. If you opened the firewall for TCP only, don't expect it to work. |
| 19:08.35 | kolyaflash | yep |
| 19:08.51 | kolyaflash | infobot, your right |
| 19:08.51 | infobot | You're right, moron! |
| 19:10.14 | hesco | sorry, this is the next line: -A INPUT -p udp -m udp --dport 5160:5161 -j ACCEPT |
| 19:10.56 | p3nguin | and why, exactly, are you using the wrong ports? SIP is UDP 5060, nothing eles. |
| 19:11.04 | hesco | I added the tcp line for this new trunk provider, whose website advised it. Did not know why tcp would be needed, but added it anyway |
| 19:11.35 | hesco | I had one provider at some time tell me they encrypted sip at 5061 |
| 19:11.45 | hesco | I think that was why that got added. |
| 19:11.57 | p3nguin | I think 5061 is used for TLS/SIP. |
| 19:12.15 | kolyaflash | p3nguin, why nothing else? i'm use different port for security reason |
| 19:12.50 | p3nguin | You're using a different port for obscurity reasons. |
| 19:12.58 | p3nguin | obscurity is not security. |
| 19:14.35 | hesco | kolyaflash: you are apparently on to something. Took me a moment to install nmap on this notebook, but then I saw: 5060/tcp closed sip |
| 19:14.39 | hesco | investigating now |
| 19:14.48 | p3nguin | SIP is UDP, not TCP. |
| 19:14.51 | kolyaflash | p3nguin, Perhaps you're right. |
| 19:15.36 | p3nguin | Unless you have SIP phones which use TCP and you have specifically configured asterisk to listen on TCP, forget about TCP. |
| 19:15.52 | kolyaflash | sudo nmap -sU server.com |
| 19:16.03 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 19:16.53 | kolyaflash | flash-Aspire-5100% sudo nmap -sU 10.9.6.27 -p 5060 |
| 19:16.53 | kolyaflash | Starting Nmap 5.21 ( http://nmap.org ) at 2011-12-18 21:16 EET |
| 19:16.53 | kolyaflash | Nmap scan report for office.user.freenet.local (10.9.6.27) |
| 19:16.53 | kolyaflash | Host is up (0.0036s latency). |
| 19:16.53 | kolyaflash | PORT STATE SERVICE |
| 19:16.54 | kolyaflash | 5060/udp open|filtered sip |
| 19:16.56 | kolyaflash | Nmap done: 1 IP address (1 host up) scanned in 0.35 seconds |
| 19:17.00 | p3nguin | Don't flood. |
| 19:17.02 | p3nguin | ~pb |
| 19:17.02 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:17.06 | hesco | ok, may be on to something now. This appears to be a typo: 5160:5161 in my iptables rule set, just reloaded iptables and rebooted phone. |
| 19:17.10 | kolyaflash | sorry... |
| 19:17.28 | kolyaflash | i'm n00b :) |
| 19:17.32 | hesco | already getting more hopeful output from nmap |
| 19:17.48 | p3nguin | (1310.55) <p3nguin> and why, exactly, are you using the wrong ports? SIP is UDP 5060, nothing eles. |
| 19:17.50 | hesco | and now I have pretty solid black phone images next to my lines |
| 19:18.12 | kolyaflash | "pastebinit with yum or aptitude" OMG |
| 19:18.36 | p3nguin | Some people cannot copy and paste. |
| 19:18.49 | p3nguin | So they can use curlpaste or pastebinit. |
| 19:19.20 | p3nguin | I wonder if pastebinit is still compatible. |
| 19:19.31 | kolyaflash | download pastebinit |
| 19:19.45 | p3nguin | or curlpaste |
| 19:20.21 | kolyaflash | p3nguin, thanks |
| 19:25.13 | hesco | ChannelZ, lolyaflash and [TK]D-Fender : Thank you all for your time today. I kept looking right past that typo in iptables. Your help puts our business back online and in business. |
| 19:27.05 | wdoekes2 | kolyaflash: you may want to tweak the pastebin site as well (pastebinit -l / -b), I notice that the ubuntu default disallows raw view without login. fpaste.org looks reasonable |
| 19:27.38 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:28.06 | ponyofdeath | hi, i have an asterisk server and behind it i have an linksys pap wondering about echo. should i disable the echo on the linksys or what do i need to configure on asterisk to do the echo handling? |
| 19:28.14 | *** join/#asterisk kolyaflash (~Flash@nat3.freenet.lg.ua) |
| 19:29.01 | kolyaflash | wdoekes2, thanks! i'm already read manual |
| 19:29.13 | [TK]D-Fender | ponyofdeath: * can't do anything about the PAP. It does it's own. |
| 19:29.30 | [TK]D-Fender | its* |
| 19:30.22 | kolyaflash | wdoekes2, at ubuntu it work without login and paste it to http://paste.ubuntu.com/. |
| 19:30.59 | kolyaflash | hesco, :) |
| 19:31.00 | wdoekes2 | yes.. it pastes to paste.ubuntu.com, but try getting the raw version from that paste |
| 19:32.16 | ponyofdeath | [TK]D-Fender: ok thanks |
| 19:32.23 | kolyaflash | wdoekes2, ok. I understand :) |
| 19:37.17 | kolyaflash | Anybody use Asterisk with GSM gateway? |
| 19:48.54 | hesco | sorry mate, that would not be me. |
| 19:57.02 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:05.20 | SeRi | ok back. |
| 20:05.47 | SeRi | p3nguin: did I drop audio? |
| 20:08.17 | p3nguin | No clue. |
| 20:08.28 | SeRi | I dont hear your moh any more |
| 20:08.35 | SeRi | but I am connected |
| 20:09.32 | WIMPy | I think that was where I left. No moh mo. |
| 20:09.44 | WIMPy | thinks that could make a good song. |
| 20:09.48 | kolyaflash | guys, what codecs are you using for sounds in asterisk? wav? |
| 20:10.10 | [TK]D-Fender | kolyaflash: The same ones that match the codecs of the channels our calls come in with |
| 20:10.14 | kolyaflash | i mean audio-format |
| 20:10.56 | kolyaflash | [TK]D-Fender, sip for example |
| 20:11.08 | [TK]D-Fender | kolyaflash: Same answer |
| 20:13.15 | *** part/#asterisk nny (~Scott@174.107.223.14) |
| 20:13.41 | kolyaflash | [TK]D-Fender, I mean audio format for IVR or MOH. like wav or gsm |
| 20:14.34 | [TK]D-Fender | [15:10][TK]D-Fenderkolyaflash: The same ones that match the codecs of the channels our calls come in with |
| 20:16.08 | kolyaflash | [TK]D-Fender, i'm sorry :) I think I understand. |
| 20:17.52 | kolyaflash | I forgot that there is such formats as ulaw and alaw |
| 20:21.42 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:35.40 | *** join/#asterisk MrTelephone (~MrTelepho@S01060018e7e1c1b7.ls.shawcable.net) |
| 20:36.02 | MrTelephone | anyone have issues with asterisk not polling voicemails when setup for realtime? |
| 20:36.40 | MrTelephone | I can only get asterisk to send after checking voicemails or when a new voicemail comes in. After an hour the phone stops displaying the mwi. |
| 21:06.00 | *** join/#asterisk chasing`Sol (~cS@41.206.153.179) |
| 21:15.58 | hesco | I'm hitting the road now to get back to work Monday morning. Thanks for the help today folks. |
| 21:16.01 | *** part/#asterisk hesco (~hesco@c-24-98-15-67.hsd1.ga.comcast.net) |
| 21:45.18 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 21:51.29 | *** join/#asterisk wesphillips (~wphill04@75.53.136.233) |
| 21:57.33 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 22:22.10 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 22:25.57 | MrTelephone | will asterisk not forward notify replies to peers from the local machine? |
| 22:45.30 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:48.17 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 23:08.00 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:20.40 | lauris | why agi script is not executed - http://pastebin.com/9Ci32kDb ? |
| 23:34.17 | timgws | lauris: where is your agi script on the server? |
| 23:43.28 | lauris | inside /opt/asterisk/var/lib/asterisk/agi-bin |
| 23:43.53 | lauris | i set highest level of debug, verbose and enabled agi debug, nothing happens after dial is executed |
| 23:48.18 | lauris | i tried to put it before Dial, then it works |
| 23:48.50 | p3nguin | After the call ends, AGI will not run. |
| 23:49.09 | p3nguin | Look at DeadAGI() for that. |
| 23:51.45 | lauris | does System() works after hangup? |
| 23:52.18 | lauris | i'm trying to catch missed calls |
| 23:53.56 | p3nguin | You can run System() after Dial() ends or in the h extension. |
| 23:54.35 | lauris | can I pass callerid to System() script as argv ? |
| 23:54.41 | p3nguin | To make it work after Dial ends, you have to use the g option. |
| 23:54.50 | p3nguin | g Dial option |
| 23:55.10 | lauris | it's so complicated; ) |
| 23:55.35 | p3nguin | Maybe I can clear up one thing for you... |
| 23:56.43 | p3nguin | Extension h is what we refer to as the hangup extension. It is known as this because extension h runs when the call hangs up. |
| 23:57.00 | p3nguin | You can use extension h to run certain apps after the call ends. |
| 23:57.13 | p3nguin | System is one of the apps that will run in h. |
| 23:57.13 | lauris | exten => h,priority,action)_ |
| 23:57.15 | lauris | smth like this ? |
| 23:57.17 | lauris | or _h |
| 23:57.24 | p3nguin | extension h |
| 23:57.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:57.39 | p3nguin | The underscore is used for patterns. |
| 23:57.39 | lauris | ok, will google about it |
| 23:57.51 | p3nguin | h is literal, not a pattern. |
| 23:58.39 | lauris | it worked :) |
| 23:58.45 | lauris | AGI Tx >> agi_request: missed.php |
| 23:59.00 | lauris | eh, but there's no disposition in agi_* ;( |
| 23:59.38 | lauris | i was expecting to find NO ANSWER there |