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| 00:52.53 | jksM | anyone knows the NEC AP200/AP300 SIP devices cost? (or where they can be bought?) |
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| 01:02.15 | carrar | jksM, no, but we can tell you what Asterisk SIP costs |
| 01:02.27 | p3nguin | $$$$$$$$$$$$$0 |
| 01:02.31 | jksM | carrar, I know... blood, sweat and tears |
| 01:02.43 | carrar | Thousands of $$$ |
| 01:02.43 | carrar | in time |
| 01:02.46 | carrar | heh |
| 01:03.02 | jksM | I intend to connect the NEC phones to * btw... |
| 01:03.10 | carrar | do it via a PRI |
| 01:03.14 | carrar | make it easy |
| 01:03.41 | jksM | well, not really possible for me |
| 01:04.01 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 01:04.05 | jksM | I cannot deploy local asteriskes where the phones are |
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| 01:35.42 | tw56 | Hi People. Does anyone have any experience with the quadBri cards and bristuff. I'm trying to get mine working on elastix. I initially thought it was the zaptel -> dahdi issue so i installed that but now i'm thinking it something else. For some reason now i can't get asterisk to start. One of the modules is complaining about dahdi |
| 01:36.05 | tw56 | Error loading module 'app_flash.so': /usr/lib64/asterisk/modules/app_flash.so: undefined symbol: dahdi_chan_name |
| 01:36.21 | tw56 | can't get it to recompile without looking |
| 01:48.42 | *** part/#asterisk alrs (n=lars@74.113.130.2) |
| 02:07.35 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 02:08.54 | drmessano | Wear can i bye a SIP truck? |
| 02:09.05 | voipmonk | lol |
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| 02:14.38 | *** join/#asterisk etfonhomey1 (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 02:15.09 | etfonhomey1 | [TK]D-Fender, do you use imap storage for voicemail? |
| 02:20.42 | luke-jr | etfonhomey1: IMAP is a protocol, not storage |
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| 02:27.29 | drmessano | luke-jr: Guess you never heard of IMAP voicemail storage in Asterisk then |
| 02:34.04 | etfonhomey1 | luke-jr, no shit. But when you use imap as your "protocol" with Asterisk voicemail. The messages are no longer stored on the Asterisk server file system. |
| 02:34.55 | ManxPower-work | Usually they are still stored in a filesystem somewhere. |
| 02:35.39 | p3nguin | You pretty much have two choices: on the asterisk server or on a remote server. |
| 02:38.34 | etfonhomey1 | OK. Here's my problem. "voicemail show users" shows the correct number of voicemails for each user. If a user deletes a message from their "imapfolder" on their IMAP client, "voicemail show users" shows the updated number correctly. However, my MWI indicator on my Polycom's does not turn off. |
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| 02:49.30 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 02:49.37 | shmaltz | hi everyone |
| 02:49.39 | *** join/#asterisk moy (n=moy@74.12.123.169) |
| 02:50.14 | shmaltz | ~helo |
| 02:50.15 | infobot | it has been said that helo is the first command issued during smtp |
| 02:50.29 | shmaltz | ~smtp |
| 02:50.30 | infobot | well, smtp is simple mail transport protocol. rfc2821 |
| 02:50.32 | shmaltz | ~play |
| 02:50.33 | infobot | and now, for something completely different |
| 02:50.41 | shmaltz | ~time |
| 02:50.42 | infobot | 2010.01.27 2:50:42 GMT |
| 02:50.49 | shmaltz | ~date |
| 02:50.49 | infobot | It is now 2010.01.27 2:50:49 GMT |
| 02:50.58 | shmaltz | 7 seconds not bad |
| 02:51.09 | shmaltz | anyhow I'm bored |
| 02:51.16 | shmaltz | ~are you? |
| 02:51.17 | infobot | yes |
| 02:56.46 | nightrid3r | :) |
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| 03:09.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 03:18.30 | Katty | hi |
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| 03:25.29 | ChannelZ | yo |
| 03:26.28 | rycar | sup |
| 03:27.42 | ChannelZ | mah weiner |
| 03:28.25 | Katty | eww/ |
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| 03:29.23 | ChannelZ | nature is ancient |
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| 04:17.55 | random_mike | greetings |
| 04:19.42 | af_ | hey! |
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| 04:32.19 | ChannelZ | nanu nanu |
| 04:32.28 | random_mike | mork n mindy right? |
| 04:32.38 | ChannelZ | yah |
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| 04:46.18 | *** join/#asterisk devdvd (n=twister1@173-31-167-213.client.mchsi.com) |
| 04:46.37 | devdvd | hi all :) |
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| 04:53.39 | ChannelZ | waves |
| 04:53.59 | devdvd | whats up ChannelZ |
| 04:54.32 | ChannelZ | not much |
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| 05:04.32 | helloritesh | Folks, I am trying to use ChanSpy to record a conversation. Here is my code http://pastebin.ca/1767627 and its getting stuck in the ChanSpy |
| 05:05.18 | helloritesh | If I called ChanSpy from a Macro (Dial M()) command, then the cahnnel doesn't fall through so the calls are never bridged. I am sure there is something that I am missing |
| 05:05.41 | helloritesh | The code snippet is here. http://pastebin.ca/1767627 |
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| 05:15.12 | helloritesh | I could use MixMonitor with exten => _901.,4,MixMonitor(${FILENAME}.gsm,b). Is there a way to limit the minitor/recording duration? |
| 05:15.51 | helloritesh | This would be the best approach by the way.. Wish there is a way to limit call-recording length |
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| 05:16.05 | random_mike | http://unf.net/2009/12/asterisk-rick-roll.php |
| 05:16.09 | random_mike | lol ^^ |
| 05:19.16 | drfreeze | Hey |
| 05:20.35 | drfreeze | I activated call monitoring on a system and the files are being combined, but soxmix is not making a joint (merged) file from in/out, but it is concatenating them |
| 05:20.43 | drfreeze | Anyone ever seen this happen? |
| 05:24.54 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 05:32.20 | ChannelZ | Did Comacst ever stop with it's bit torrent traffic interference? |
| 05:33.10 | devdvd | they did for a while but im not sure if they started back as i remember reading an article where the judge might overturn the FCC's penaltys |
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| 05:40.32 | ChannelZ | hmph. Pondering going back to them because DSL is just too slow, but grrr |
| 05:40.56 | p3nguin | channelz: http://www.wsu.edu/~brians/errors/its.html |
| 05:42.03 | ChannelZ | p3nguin: http://www.urbandictionary.com/define.php?term=Grammar%20Nazi |
| 05:42.55 | p3nguin | channelz: http://www.urbandictionary.com/define.php?term=drop+out |
| 05:43.44 | ChannelZ | I masturbate a TON, too. |
| 05:43.57 | p3nguin | It seriously can't be THAT hard to understand that it's means IT IS. |
| 05:44.13 | ChannelZ | I do it to annoy you at this point |
| 05:44.22 | p3nguin | "Did Comacst ever stop with IT IS bit torrent traffic interference?" Yep, makes PERFECT sense. |
| 05:44.39 | devdvd | p3nguin: i understood him just fine. |
| 05:44.46 | devdvd | dont know what your problem is :) |
| 05:44.47 | p3nguin | Of course you did. |
| 05:44.47 | luke-jr | p3nguin: surprisingly, most people seem to not notice the difference |
| 05:45.04 | luke-jr | it's like they read what they see aloud and listen to it |
| 05:45.25 | luke-jr | one time someone was asking about 'hey coo' on another channel |
| 05:45.30 | luke-jr | turned out he meant haiku |
| 05:45.31 | ChannelZ | It's a typing reflex |
| 05:45.45 | p3nguin | hey coo = haiku? |
| 05:46.14 | luke-jr | ChannelZ: it makes you very difficult to understand, whatever the cause may be |
| 05:46.21 | devdvd | well, not really that luke-jr. The human mind can read the first word of a sentence and the last word of a sentence then everything in between can be slightly garbled and we can still understand it. |
| 05:46.52 | luke-jr | devdvd: maybe for short sentences, or something that might be expected to be said |
| 05:47.34 | luke-jr | "it's" seems to be an exception though; I get stuck on that one a bit |
| 05:47.40 | p3nguin | luke-jr: In this particular case, it wasn't so much that the meaning could not be understood, but more the fact that people need to know the difference and the correct usage of each word. It is becoming more and more of a problem everywhere. |
| 05:47.46 | luke-jr | have to re-read it a few times before I pick up what is meant |
| 05:48.42 | p3nguin | luke-jr: It's very simple. "its" is a possessive pronoun, just like his and hers. There is no apostrophe in his, so there isn't one in "its." |
| 05:48.42 | devdvd | well. not really luke-jr. Im trying to find that test i seen one time where literally the first word in the paragraph was spelled correctly and so was the last word and everything in between was messed up. |
| 05:49.31 | luke-jr | p3nguin: I'm well aware. |
| 05:49.54 | luke-jr | devdvd: IIRC, it's the first and last letter of the word, not f/l word of the sentence |
| 05:50.16 | p3nguin | sounds familiar |
| 05:52.22 | devdvd | luke, your right. my bad |
| 05:53.11 | p3nguin | his right, as opposed to his left? |
| 05:53.24 | devdvd | exactly :) |
| 05:54.36 | luke-jr | p3nguin: he meant: "Luke, you're right. My mistake." |
| 05:54.37 | luke-jr | :p |
| 05:54.45 | p3nguin | lol |
| 05:56.02 | hardwire | meh |
| 05:56.54 | hardwire | There's a 10% chance that I'll see this conversation whenever I switch focus to this tab. |
| 05:57.25 | luke-jr | press Ctrl-L |
| 05:57.28 | luke-jr | then it will be an even 0% |
| 05:57.42 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 05:57.44 | p3nguin | Hopefully the issue will be rectified eventually. |
| 05:57.59 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 05:59.38 | hardwire | Hopefully the issue will eventually be rectified. |
| 05:59.55 | drmessano | p3nguin: The E in your nickname is reversed and rounded. Just sayin |
| 06:00.23 | p3nguin | drmessano: Yeah, there was someone else using the 2, so I had to use a 3. Damn NickServ, anyway. |
| 06:00.43 | hardwire | bets it was his original nick and he lost the pw |
| 06:00.51 | *** part/#asterisk etfonhomey1 (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 06:01.00 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 06:01.03 | hardwire | drmessano: I need to say "just sayin" more. |
| 06:01.10 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 06:01.47 | p3nguin | I always find it amusing when people can't find anything else to pick on me about. |
| 06:02.37 | drmessano | I bet his password was "grammernazi" and after realizing his error, he was unable to auth to nickserv due to hypocritical and moral objections |
| 06:03.24 | drmessano | p3nguin: You're short, walk funny, have goofy arms, look like you're wearing a tuxedo, and I hope you get eaten by a Polar Bear. How is that? |
| 06:03.43 | p3nguin | This saddens me. |
| 06:03.48 | luke-jr | drmessano: grammarnazi* |
| 06:04.18 | drmessano | luke-jr: Re-read what I typed |
| 06:04.21 | p3nguin | I think luke-jr missed the joke. |
| 06:04.25 | drmessano | Indeed |
| 06:04.34 | drfreeze | I activated call monitoring on a system and the files are being combined, but soxmix is not making a joint (merged) file from in/out, but it is concatenating them |
| 06:04.36 | luke-jr | I think you missed the spelling error |
| 06:04.37 | drfreeze | Anyone ever seen this happen? |
| 06:04.45 | p3nguin | luke-jr: It was part of the joke. |
| 06:04.54 | drmessano | luke-jr: The spelling error was the fukin joke, dude |
| 06:04.59 | luke-jr | spelling is not grammar |
| 06:05.21 | p3nguin | I try to tell people that all the time, but they just want to argue it. |
| 06:05.27 | drmessano | Nor was the joke referring to grammar |
| 06:05.29 | luke-jr | lol |
| 06:05.30 | p3nguin | It's still part of the joke. |
| 06:07.07 | drmessano | p3nguin: I apologize for my interruption. Please continue with the roasting of luke-jr.. I see now it is well deserved, and I wish you the best of luck with it. |
| 06:07.19 | hardwire | drfreeze: so one channel plays until the call is done then the other plays? |
| 06:07.47 | p3nguin | drfreeze: You're using the Monitor() app and joining the files after the call has ended? |
| 06:07.53 | luke-jr | roasting of me is off-topic here and may only occur in #anime kthx |
| 06:08.02 | drmessano | #anime? |
| 06:08.09 | luke-jr | yes? |
| 06:08.10 | p3nguin | drfreeze: May I suggest using MixMonitor() instead? |
| 06:08.22 | drmessano | Dear <diety> |
| 06:08.29 | drmessano | or deity |
| 06:09.18 | drfreeze | p3nguin: yes |
| 06:09.51 | hardwire | p3nguin: I believe he wants you to suggest it. |
| 06:10.06 | p3nguin | drfreeze: I prefer MixMonitor(); it has never given me any trouble. Try it. |
| 06:10.10 | p3nguin | hardwire: I was getting there. |
| 06:10.12 | hardwire | :P |
| 06:10.13 | drfreeze | exten => 9888,n,Monitor(wav,${EQCALLFILENAME},m) |
| 06:10.22 | drmessano | If you like anime enough that you hang out in its dedicated IRC channel, you should probably go upstairs right now, hand mom the keys to the basement door, and move out immediately before you die a virgin |
| 06:10.25 | drmessano | just sayin' |
| 06:10.39 | hardwire | JUST SAYIN'! |
| 06:10.46 | drfreeze | p3nguin: do I just replace Monitor with MixMonitor? |
| 06:10.50 | hardwire | does it count if I say it after you say it? |
| 06:10.59 | luke-jr | drmessano: I'm married with 3 children, thanks. Anime is not for kids. |
| 06:10.59 | p3nguin | drfreeze: Yes, but watch for changes in the options. |
| 06:11.46 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-iznipaeybipjjqij) |
| 06:11.52 | luke-jr | and #anime is more of a social channel than a strict on-topic channel |
| 06:11.59 | p3nguin | drfreeze: MixMonitor(${EQCALLFILENAME}.wav) |
| 06:13.00 | drmessano | Yes, and comic books are not childrens books, they are serialized illustrated literature |
| 06:13.53 | luke-jr | drmessano: your brain fails |
| 06:15.36 | drmessano | At completing sentences? |
| 06:15.57 | p3nguin | I should just stop talking right in the m |
| 06:16.10 | hardwire | luke-jr: if you're trying to mess with drmessano, then you're bored. |
| 06:16.34 | drfreeze | p3nguin: does mixmonitor use soxmix? |
| 06:16.51 | luke-jr | hardwire: you hit the nail on the head. I am dead bored. |
| 06:16.52 | p3nguin | drfreeze: I don't think so. |
| 06:16.59 | hardwire | drfreeze: depends on what version of asterisk you're using |
| 06:17.05 | drfreeze | 1.4.24.1 |
| 06:18.06 | hardwire | drfreeze: http://svnview.digium.com/svn/asterisk/tags/1.4.24/apps/app_mixmonitor.c?view=markup |
| 06:18.12 | hardwire | looks like you're clean |
| 06:18.21 | hardwire | just search that if you have any functionality questions |
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| 06:33.27 | jayrod422 | i have a queue setup and the customer wants the moh to be ringing - any idea on how to do this or where i can find a ringing gsm file to put as the moh? |
| 06:34.00 | p3nguin | <PROTECTED> |
| 06:34.16 | jayrod422 | the moh is in the queue |
| 06:34.17 | p3nguin | core show application Queue |
| 06:34.22 | jayrod422 | k |
| 06:34.53 | p3nguin | Add the r option, and there will be ringing instead of music while you wait. |
| 06:35.03 | jayrod422 | sweet |
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| 06:52.10 | *** join/#asterisk kruemeltee (n=Maddin@port-92-198-62-82.static.qsc.de) |
| 06:52.26 | kruemeltee | good mornin' :-) |
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| 06:57.56 | ChannelZ | Its a beautiful day in the neighborhood |
| 06:58.04 | ChannelZ | snickers |
| 07:06.34 | kruemeltee | is there any option defining the bandwith on which * talk to SIP telephones? |
| 07:06.58 | ChannelZ | yes.. you can allow/disallow codecs |
| 07:07.09 | ChannelZ | (yes-ish I guess) |
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| 07:07.31 | kruemeltee | so you mean the only way is by selecting the right codec? |
| 07:07.59 | ChannelZ | yes, the codecs themselves are of a static rate so it's not like you get to tweak compression settings |
| 07:09.10 | kruemeltee | okay ... thanks :-) |
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| 07:31.14 | voipmonk | j/oin #asterisknow |
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| 08:22.52 | phix | hmmm, my computer got a surge, I have a TDM400p in it, I thought the surge came from the landline although the FXO module is fine, the FXS module smelt burnt and was not being picked up by the TDM |
| 08:23.13 | phix | The other components in the computer seem fine so I am confused how this happened |
| 08:26.07 | ChannelZ | well the FXS ports require power from the computer, so perhaps the surge toaster your host computer's power supply and thus the FXS module of your TDM card |
| 08:26.39 | phix | ChannelZ: the PSU is working fine though, mobo is fine, RAM is fine, HDDs are fine |
| 08:27.18 | ChannelZ | ....or they were fried by whatever was connected to your FXS port(s) - a FAX machine for instance |
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| 08:37.28 | Heretic | surges and lightning strikes are funny things |
| 08:37.36 | Heretic | had a lightning strike here that seemed to come through the telephone line |
| 08:37.48 | Heretic | server wasn't booting up (clicking noise from PSU) |
| 08:37.56 | Heretic | then 1 day later all is well except for blown network port |
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| 09:28.49 | dandre | Hello, |
| 09:29.46 | dandre | Is there any way to limit the number of waiting calls in a queue to the number of free agents available? |
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| 10:24.10 | vlt | Hello. One of my peers is connected over wireless network. Can I use TCP for SIP to make sure no SIP packets get lost? |
| 10:25.06 | Chainsaw | vlt: If you run Asterisk 1.6 and not an older version, yes, then you can have SIP run over TCP. |
| 10:25.19 | Chainsaw | vlt: Note that this is only call setup, the audio will stil flow over RTP, which is a UDP-based protocol. |
| 10:28.02 | vlt | Chainsaw: Yes, I know that RDP is like UDP. The problem is that sometimes calls seem to be announced a few seconds delayed. Maybe the first attempt didn't made it over wifi ... |
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| 10:29.09 | vlt | hmmm, using Asterisk 1.4.21, so I do have to upgrade, right? |
| 10:29.24 | Chainsaw | vlt: Yes, I would not try it on anything that's older then 1.6.1.X |
| 10:30.16 | Chainsaw | vlt: If there is high loss on the connection though, your audio may still suffer. |
| 10:31.40 | vlt | Chainsaw: There are only few audio drop outs once the connection is established. |
| 10:32.26 | Chainsaw | vlt: Right, you may want to look at your WiFi channel. Perhaps other channels are less busy. |
| 10:32.43 | vlt | Chainsaw: I thought it may be a problem with the very first SIP packet(s) of the call ... |
| 10:32.54 | vlt | Chainsaw: I'll check the wifi channel, thankls |
| 10:32.56 | vlt | thanks |
| 10:33.18 | Chainsaw | vlt: Well, since SIP & RTP are both UDP and you do get audio dropouts... I'd say the connection is just a bit marginal. |
| 10:33.32 | Chainsaw | vlt: I'd see if you can address that first. |
| 10:34.54 | dandre | <PROTECTED> |
| 10:36.22 | fiddur | sounds more like a call group than a queue... |
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| 10:40.21 | Acro | Any detailed information about the CDR data format available somewhere? The documentation on asterisk.org and voip-info.org is rather limited. |
| 10:40.24 | creativx | kinda defeats the purpose of a queue dandre |
| 10:40.40 | creativx | queue = to house more callers than you have available agents |
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| 10:43.30 | dandre | creativx: ok but I want something like ringing the extension which has least calls first, if unavailable then second in that priority order. If all agents are online return the queue application and let the dialplan handle this request |
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| 10:54.17 | creativx | look up strategy |
| 10:54.35 | creativx | you can make the queue reject calls when its full |
| 10:54.38 | creativx | or empty |
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| 10:54.53 | creativx | maybe not full.. i cant recall |
| 11:06.21 | dandre | ok |
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| 12:51.06 | roni | hi everybody .. |
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| 12:58.51 | roni | can somebody give me a little help ? i got my asterisk server working with pstn , and softphones clients ..all is working fine , but the softphone dosent shows who is calling , or what line is calling .. i put a name XXX in the line , but my softphone just shows "Account ID = ", this is happend with all the softphones .. the only one that works was zoiper .. sorry for my bad english |
| 13:05.29 | tzafrir_laptop | roni, how is it connected to the PSTN? analog? digital? SIP? |
| 13:05.42 | tzafrir_laptop | From which country are you? |
| 13:05.53 | roni | Analog , Digium board .. |
| 13:07.17 | tzafrir_laptop | roni, do you have: callerid = asrecieved ; in chan_dahdi.conf? |
| 13:07.27 | roni | i m from Chile, if anyone speaks spanish or portugues (Borned in Brazil) ... :) |
| 13:07.34 | roni | well let me see .. |
| 13:09.52 | roni | i got callerid = asreceived in any of the lines listed in dahdi-channels.conf |
| 13:13.13 | roni | <tzafrir_laptop> : Thank you very much |
| 13:13.48 | tzafrir_laptop | roni, it work now? |
| 13:14.02 | roni | yes man .. |
| 13:14.10 | roni | thank you very much |
| 13:14.13 | roni | was so simples |
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| 13:14.51 | roni | simple .. that i m feeling like a stupid |
| 13:15.03 | roni | hahahahahaaha , thanks... |
| 13:25.56 | dandre | creativx: I have looked at strategy by that doen't help |
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| 13:36.32 | helloritesh | Folks, I want to know if there is a way to record first few minutes of the call. Using MixMonitor forces to record the entire conversation and some calls could last long creating stress on the storage system. I tried a few things (ChanSpy, Macro etc) but none of that is working out. Any suggestions would be helpful. Here is a pastebin link for some of the things I have tried. http://pastebin.ca/1767944 |
| 13:37.34 | [TK]D-Fender | helloritesh: I answered this yesterday |
| 13:37.43 | helloritesh | yeah.. tried but doesn't work |
| 13:37.49 | helloritesh | Thanks for the help. |
| 13:37.52 | helloritesh | I must be doing something wrong |
| 13:37.57 | helloritesh | Please check http://pastebin.ca/1767944 |
| 13:37.59 | fiddur | Yikes. I had added domain=mydomain in sip.conf, as described in http://www.voip-info.org/wiki/view/Asterisk+sip+domain, to get sip calls to a domain into specific context. Nothing on voip-info told med what I should have read in the sample config: "REGISTER to non-local domains will be automatically denied".... ! |
| 13:38.00 | [TK]D-Fender | helloritesh: I said ORIGINATE A NEW CALL.New damn call. |
| 13:38.21 | [TK]D-Fender | helloritesh: ORIGINATE. You following? Spawn an entire new CHANNEL to do the spying |
| 13:38.24 | helloritesh | How do I originate a call from the dailplan/Macro? |
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| 13:38.33 | [TK]D-Fender | helloritesh: lNot jsut call Monitor from withing a macro |
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| 13:39.19 | helloritesh | You mean trigger an AMI or something? |
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| 13:39.19 | [TK]D-Fender | helloritesh: AMI originate, CLI originate (called via System(), or 1.6+'s Originate() app. Take your pick. This is all in the BOOK and in the app docs |
| 13:39.29 | helloritesh | Cool. I will check that out |
| 13:39.32 | helloritesh | thanks again |
| 13:39.41 | helloritesh | I am on 1.4 so I will have to go through System call |
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| 13:39.43 | helloritesh | but that's fine |
| 13:40.28 | [TK]D-Fender | helloritesh: Or AGI. Or send an AMI custom message that a listenng process will use to issue it. |
| 13:40.42 | helloritesh | Yup.. got it.. |
| 13:40.49 | [TK]D-Fender | helloritesh: Or CURL call, or :whatever" |
| 13:41.11 | helloritesh | Yup... I will go with a simple System call to a script calling AMI |
| 13:41.11 | [TK]D-Fender | helloritesh: Probably about a dozen different ways at hand |
| 13:41.32 | helloritesh | true and thanks a bunch.. |
| 13:41.38 | [TK]D-Fender | helloritesh: Actually... i'd use a Call file instead |
| 13:41.56 | [TK]D-Fender | helloritesh: You can pass the channel as a parameter more easily |
| 13:41.57 | helloritesh | brilliant |
| 13:42.05 | helloritesh | that would be super simple |
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| 13:42.27 | helloritesh | and I know for a fact that it would work pretty sweet |
| 13:42.45 | helloritesh | Thanks a million |
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| 13:54.31 | TinoSchwarze | hi there |
| 13:54.45 | danj1980 | Hi |
| 13:54.55 | danj1980 | Does anyone know if this is a problem? |
| 13:54.55 | danj1980 | channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) |
| 13:55.31 | [TK]D-Fender | danj1980: Does something not work? |
| 13:56.01 | danj1980 | We keep losing calls but i dont think that its got anything to do with that. |
| 13:56.11 | danj1980 | I just spotted it and wondered if it was a problem. |
| 13:56.33 | danj1980 | I think the lost calls issue is a problem with the provider. |
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| 14:04.50 | makafre | morning guys, I was wondering if any of you knows about https://issues.asterisk.org/view.php?id=8030 ; I am using 1.4.26.1 and I think it's not included and I am wondering why. This is for MeetMe along with L(..) as in the Dial command |
| 14:05.21 | danj1980 | We keep losing incoming calls at exactly 600 seconds into the call. The asterisk logs show that the DDI provider sends the first BYE message so it seems that they are disconnecting the call. But they insist that it is something on our side. I've done tests with another DDI from the same provider and it worked fine. |
| 14:06.26 | TinoSchwarze | I'm currently struggling with a strange SIP issue. I'm running 1.2.7.1 with SNOM phones (of different types) and twinkle as client. Maybe I'm just missing some point, but when I get an external call, there is no way to call back using the phone. |
| 14:07.39 | TinoSchwarze | For example, if my internal softphone is called as SIP/164, I get From and To reversed. It shows me: "From: Tino Schwarze <sip:164@phones.community4you.de>" |
| 14:07.58 | TinoSchwarze | So pressing the "call" button will make it call itself. |
| 14:08.03 | ManxPower-work | TinoSchwarze: You have something seriously wrong. |
| 14:08.18 | TinoSchwarze | That's what I suspected. |
| 14:08.20 | voipmonk | WOW |
| 14:08.43 | TinoSchwarze | The strange thing is, it's been working like that for over 5 or so years now. |
| 14:09.02 | TinoSchwarze | I'm not sure where to look... |
| 14:09.05 | anonymouz666 | danj1980: if the provider sends the first BYE, then the provider should be able to tell you why this BYE is generated. |
| 14:10.27 | voipmonk | send them your proof that they are sending it |
| 14:10.42 | TinoSchwarze | I'll try to configure a minimum extensions.conf example without any macros or things involved... |
| 14:10.44 | voipmonk | backup your claim and maybe they will be more forthcoming |
| 14:10.55 | ManxPower-work | TinoSchwarze: that is the right way to diagnose the issue |
| 14:11.06 | danj1980 | i've sent over the logs, and they've told me that theres nothing on their side that would cause this. |
| 14:11.42 | danj1980 | Is there something at the start of the call that would cause asterisk to instruct the sip trunk to disconnect after 600 seconds? |
| 14:11.46 | ManxPower-work | danj1980: you didn't do something stupid like set callprogress=yes or busydetect=yes have you? |
| 14:12.11 | ManxPower-work | danj1980: also make sure you have canreinvite=no |
| 14:12.15 | danj1980 | ManxPower-work, dont know what those two are, but either way i dont use them. |
| 14:12.29 | ManxPower-work | danj1980: is there NAT involved? |
| 14:12.30 | danj1980 | canreinvite=no is set on the sip trunk |
| 14:12.36 | ManxPower-work | ~trunk |
| 14:12.37 | infobot | well, 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. |
| 14:12.41 | danj1980 | NAT on the sip phone side |
| 14:12.54 | anonymouz666 | it looks like some kind of timer.... if it is always 600 seconds |
| 14:12.57 | ManxPower-work | you have all the NAT stuff disabled on the phone, right? |
| 14:13.32 | ManxPower-work | danj1980: pastebin the output of a failed call, including dialplan leading up to the Dial |
| 14:13.49 | danj1980 | It doesnt seem like a NAT issue because I've run tcpdump and recorded the problem. All RTP packets are being sent and received, right up to the second the call disconnects |
| 14:13.57 | anonymouz666 | danj1980: very hard to help you without the whole SIP debug for this call. |
| 14:14.24 | danj1980 | i can put it in paste bin, but the logs contain RTP debug and its 18MB in total. |
| 14:14.39 | danj1980 | Can you paste something that big into pastebin? |
| 14:14.44 | ManxPower-work | I was not looking for the debug logs, I was looking for the verbose 3 logs |
| 14:15.02 | ManxPower-work | I just want to see if you are doing something weird in your dialplan. |
| 14:15.05 | TinoSchwarze | ManxPower-work: I enabled SIP debugging and get the following when calling in: http://pastebin.com/d1d3aeeed |
| 14:15.12 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:15.12 | ManxPower-work | You're not using an Asterisk GUI are you? |
| 14:15.21 | danj1980 | I'll try to remove the rtp and debug log and just pastebin the verbose and dialplan |
| 14:15.24 | danj1980 | nope |
| 14:15.29 | danj1980 | manully coded |
| 14:15.44 | ManxPower-work | TinoSchwarze: that is not a sip debug. that is a single sip packet. |
| 14:16.19 | TinoSchwarze | ManxPower-work: It's showing the problem - weird "From" address. |
| 14:16.25 | *** join/#asterisk _zen_ (n=_zen_@cpe-74-66-140-78.nyc.res.rr.com) |
| 14:16.27 | ManxPower-work | TinoSchwarze: do you have Asterisk correctly set up for your NAT stuff? |
| 14:16.37 | ManxPower-work | if you don't, that could explain it. |
| 14:16.38 | danj1980 | Sorry, its got the RTP verbose and I cant edit it to remove it. |
| 14:16.55 | danj1980 | i thought I could do a find and replace, but each line is slightly different |
| 14:16.56 | ManxPower-work | danj1980: I'm not spending 2 hours reading debug info for free. |
| 14:17.09 | TinoSchwarze | ManxPower-work: there is no NAT involved, just a plain 10.2.0.0/16 network. |
| 14:17.21 | ManxPower-work | TinoSchwarze: pastebin.ca your sip.conf |
| 14:17.31 | danj1980 | ManxPower-work, i wouldnt want you to. |
| 14:18.08 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 14:19.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.11 | TinoSchwarze | http://pastebin.ca/1767983 |
| 14:21.36 | ManxPower-work | TinoSchwarze: I thought you were calling between two local phones? |
| 14:22.00 | TinoSchwarze | it's the same if I'm calling between two local phones... |
| 14:22.02 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 14:22.22 | ManxPower-work | TinoSchwarze: your pastebin only has 1 phone. Where is the other phone setup? |
| 14:23.13 | ManxPower-work | TinoSchwarze: where are you located? |
| 14:23.13 | TinoSchwarze | Here it is: http://pastebin.ca/1767986 |
| 14:23.20 | TinoSchwarze | Germany. |
| 14:23.55 | ManxPower-work | now show a pastebin of a failed call between those two phones |
| 14:24.23 | TinoSchwarze | ok... a second... |
| 14:25.03 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:26.48 | TinoSchwarze | ManxPower-work: http://pastebin.ca/1767991 - I called from 129 to 164 and rejected the call. |
| 14:27.28 | ManxPower-work | TinoSchwarze: I see no dialplan stuff in that pastebin |
| 14:27.30 | TinoSchwarze | Things are actually working - I can talk etc. Just calling back a missed call won't work because of the strange From line. |
| 14:28.28 | *** join/#asterisk garymc (n=chatzill@host86-158-86-203.range86-158.btcentralplus.com) |
| 14:28.37 | ManxPower-work | It looks like your PHONE is sending the wrong info. |
| 14:28.46 | ManxPower-work | From: "Kernentwickler" <sip:164@10.2.1.2>;tag=as150ffa84 To: <sip:164@10.2.10.103> |
| 14:29.41 | Katty | well.. |
| 14:29.50 | [TK]D-Fender | TinoSchwarze: remove the FROMUSER from your sip.conf |
| 14:29.55 | [TK]D-Fender | TinoSchwarze: it is bad. |
| 14:30.00 | Katty | i got TNH bird bath broadcasting in 720p this morning |
| 14:30.14 | [TK]D-Fender | TinoSchwarze: and "username" as well |
| 14:30.28 | Katty | but it seems as tho my lil server can't handle two 720/1080 streams at th esame time |
| 14:31.03 | Katty | never to fear tho! there's room for another lil pc behind the couch upstairs :P |
| 14:33.23 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-slokkrwawtdprjaw) |
| 14:33.34 | TinoSchwarze | [TK]D-Fender: removing fromuser and username did the trick for the internal call! :-) |
| 14:33.59 | TinoSchwarze | yeah, I may call back. |
| 14:34.04 | [TK]D-Fender | TinoSchwarze: And stop calling it an "internal call". Every call is jsut a call. |
| 14:34.04 | TinoSchwarze | Now checking external call-in... |
| 14:35.02 | TinoSchwarze | ManxPower-work, [TK]D-Fender: Thanks a lot, that did the trick. Now I'm off to correcting my sip.conf! Whoever decided to put that stuff in there 5 years ago... |
| 14:35.09 | Katty | there's a pretty red bellied woodpecker ont he light house feeder this morning :> |
| 14:35.58 | *** part/#asterisk makafre (n=makafre@64.86.141.133) |
| 14:36.19 | Katty | now he's on the tray |
| 14:36.23 | Katty | gobbling up peanuts. |
| 14:36.35 | [TK]D-Fender | TinoSchwarze: You're welcome. |
| 14:36.57 | Katty | oh! and i finally had a visitor to the hanging bird bath....but it wasn't a bird ;) |
| 14:37.10 | Katty | suspecting a squirrely thought it was another feeding station cleverly hidden in a tree. |
| 14:39.26 | *** join/#asterisk jozza (n=chatzill@BSN-77-186-224.dsl.siol.net) |
| 14:39.42 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:40.05 | *** part/#asterisk cadmium (i=mike@217.194.139.22) |
| 14:41.04 | jozza | hi, i'm making an automatic script for installing asterisk and i want to add the flags to make command to make install extra sound packages and stuff. Where can i find the flags? |
| 14:41.14 | *** part/#asterisk _zen_ (n=_zen_@cpe-74-66-140-78.nyc.res.rr.com) |
| 14:41.52 | *** join/#asterisk elred_ (n=elred_@unaffiliated/elred-/x-5010831) |
| 14:42.26 | jozza | i mean the flags that one would usualy select through menuconfig |
| 14:42.53 | *** join/#asterisk Han (n=han@unaffiliated/han) |
| 14:43.17 | Han | Is it possible to switch the dnd mode off from the serverside? |
| 14:43.30 | ManxPower-work | Han: Are you using a GUI? |
| 14:43.43 | Han | yes. |
| 14:43.48 | ManxPower-work | ~freepbx |
| 14:43.49 | 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 |
| 14:43.58 | ManxPower-work | Asterisk does not have a "dnd mode" |
| 14:43.59 | elred_ | Hello. I am looking for a clean way to use music on hold using shoutcast ? On voip-info.org it only show some application=bashscript.sh with some ogg123 piped to sox. Do you guys know if a proper way do exist ? |
| 14:44.10 | Han | ah, xcuse me |
| 14:44.31 | elred_ | does* |
| 14:44.51 | *** part/#asterisk Han (n=han@unaffiliated/han) |
| 14:48.07 | *** join/#asterisk Deeewayne (n=dwayne@adsl-070-148-053-233.sip.bhm.bellsouth.net) |
| 14:48.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:51.42 | *** part/#asterisk TinoSchwarze (n=tisc@community4you193-el-CHE.eastlink.de) |
| 14:53.17 | *** join/#asterisk yoshx (n=yoshx@38.115.119-80.rev.gaoland.net) |
| 14:53.49 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:54.09 | *** join/#asterisk yoshx (n=yoshx@38.115.119-80.rev.gaoland.net) |
| 14:55.13 | *** join/#asterisk ktwilight_ (n=ktwiligh@91.179.88.138) |
| 14:59.41 | *** join/#asterisk jkroon (n=jkroon@dsl-244-22-201.telkomadsl.co.za) |
| 15:00.27 | jkroon | hi all, i'm strugling to grasp the whole earlymedia change that got made a couple weeks ago in the 1.6.1 branch, would it be possible for someone to explain to me exactly what changed and why? |
| 15:01.44 | ManxPower-work | jkroon: try #asterisk-dev |
| 15:02.28 | jkroon | ta ManxPower-work |
| 15:02.37 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:03.26 | Katty | i'm in the mood for love |
| 15:03.30 | Katty | simply because you're near me! |
| 15:04.06 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:04.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:07.22 | beek | hugs Katty |
| 15:08.34 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:09.38 | Katty | hugs beek |
| 15:10.13 | Katty | beek: have you seen the new broadcast yet? (= |
| 15:11.13 | beek | Katty: No... same URL? |
| 15:11.21 | Katty | beek: nope, it's a different url. |
| 15:11.30 | beek | what is it? |
| 15:11.37 | Katty | beek: http://ustre.am/bEBU |
| 15:12.21 | beek | Wow! That's beautiful. Looks like you have some critters there enjoying the food. |
| 15:12.46 | *** join/#asterisk lordmortis (n=lordmort@203-59-87-141.dyn.iinet.net.au) |
| 15:12.48 | beek | Katty: Looks like better resolution than before. |
| 15:12.49 | Katty | beek: yeah and that's just the right side of the yard. |
| 15:13.05 | Katty | yep. i figured out how to broadcast in high quality |
| 15:13.19 | Katty | the right webcam is capabible of 720, the main one is capable of 180 |
| 15:13.21 | Katty | err 1080 |
| 15:13.26 | beek | It looks really great. |
| 15:13.32 | fenrus | Katty, neat webcam! |
| 15:13.43 | Katty | but the workstation can only handle one high quality stream at a time...so i'm gonna stick another workstation behind the couch to handle the second stream |
| 15:13.45 | fenrus | and i like the idea of feeding squirrels :) |
| 15:14.03 | Katty | infobot: crittercam |
| 15:14.04 | infobot | [crittercam] Katty's broadcast of The Nut House Critter Cam @ http://ustre.am/8H5d and The Nut House Bird Bath @ http://ustre.am/bEBU |
| 15:14.04 | beek | Katty: Nicely done. |
| 15:14.21 | Katty | fenrus: i'll have crittercam is high quality tomorrow |
| 15:14.59 | Katty | fenrus: for the first time :> |
| 15:15.06 | fenrus | hm, a webcam in my countryhouse perhaps... :) |
| 15:15.12 | fenrus | Katty, nice =) |
| 15:15.17 | Katty | fenrus: it has several uses. |
| 15:15.17 | fenrus | will have a look then |
| 15:15.23 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 15:15.24 | Katty | fenrus: we frequently watch for the pizza guy to show up |
| 15:15.36 | fenrus | :D :D |
| 15:16.04 | Katty | it also doubles as a form of home security |
| 15:16.25 | fiddur | Hi. Using res_jabber, I can't seem to get asterisk to send an authorization request. I can send a message from asterisk to another jabber client, but that is sent without authorization. I can ask for authorization FROM asterisk, and get it. But asterisk can't track my presence unless it gets MY authorization. According to Leif Madsen, it should ask for authorization on a jabbersend, but it doesn't... any ideas? |
| 15:16.33 | fenrus | we usually have deer at the place where we feed birds :) |
| 15:16.39 | Katty | wow nice. |
| 15:16.43 | Katty | that'd be amazing to watch |
| 15:16.43 | fenrus | eating the leftovers from the ground |
| 15:17.03 | fenrus | hm, is it called roedeers in english perhaps |
| 15:17.55 | fenrus | I'll try to remember to bring the webcam next time =) |
| 15:18.07 | Katty | horay! |
| 15:19.40 | benngard | talking about deers, i "met" 2 roe deers when i went home (took a shortcut through the forest) yesterday, we where looking at each other for like 1 minute but after that, the "flew" away |
| 15:19.52 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 15:19.55 | fiddur | leifmadsen: Your blog claims res_jabber should send authorization request on jabbersend... it doesn't, for me; could that be up to the jabber daemon or client requiring authorization before accepting a message? |
| 15:19.59 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 15:20.08 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 15:20.36 | Katty | fenrus: ustream accounts are free, btw (= |
| 15:20.47 | Katty | fenrus: they might have a paid version where you can skip the ads, not sure. |
| 15:21.20 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 15:22.24 | ManxPower-work | Quick! Notify CNN! Then everyone can see it! |
| 15:22.51 | fenrus | benngard, i like encountering wild life like that |
| 15:23.04 | fenrus | Katty, neat |
| 15:24.22 | leifmadsen | fiddur: not sure... I was probably using pidgin at the time, and that seems to be what happened for me. I haven't used res_jabber in quite some time now. |
| 15:24.51 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:25.52 | fiddur | leifmadsen: I'm using pidgin, and I get no authorization request... thus, adding asterisk gives a one way subscription :( |
| 15:26.32 | Chainsaw | fiddur: It's a setting isn't it? |
| 15:26.40 | Chainsaw | fiddur: Having it autorespond to authorisation requests? |
| 15:27.25 | Chainsaw | fiddur: (I've done away with the whole buddying system locally as it's company jabber. I have an automatic group "staff" that everyone is buddied to and everyone is a member of) |
| 15:27.54 | fiddur | Chainsaw: Asterisk is responding to my authorization request; but the problem is it doesn't SEND any authorization request... |
| 15:28.41 | fiddur | Chainsaw: the only setting close to that would be autoregister, and that's on... |
| 15:28.58 | Chainsaw | fiddur: Right, okay. And this is 1.6.2.1? |
| 15:29.48 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 15:29.55 | fiddur | Chainsaw: 1.6.1.4 and 1.6.1.6... haven't tried 1.6.2... |
| 15:30.08 | Chainsaw | fiddur: At least try 1.6.1.13 then please; you're well behind. |
| 15:30.22 | *** join/#asterisk grandpapadot (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 15:30.25 | fiddur | Chainsaw: I'll see if there are relevant diffs to res_jabber... |
| 15:31.08 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 15:31.22 | grandpapadot | Hey all, anyone else seen an issue with 1.4.29 where app_followme.so places calls, but then doesn't play prompts and won't accept input to the "accept call" prompts? This issue survives a restart. I thought it was file permissions on the promtps but no joy and there isn't any issues in mantis that I can find. |
| 15:31.35 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
| 15:31.43 | grandpapadot | ... It's affecting 8 of our switches recently updated from 1.4.28 |
| 15:32.41 | *** join/#asterisk Guest_536 (n=IceChat7@80.172.228.33) |
| 15:32.57 | Guest_536 | Hi everyone... |
| 15:33.05 | Chainsaw | Hello numbered Guest. |
| 15:33.17 | Guest_536 | I'm trying to compile a new module, app_alarmtransmitter |
| 15:33.33 | Guest_536 | Hi Chainsaw |
| 15:34.01 | Guest_536 | i'm stuck in detecting the ack tones and kissoff tones... |
| 15:34.25 | Chainsaw | Guest_536: Sorry, I don't use that application myself. I'll leave it to others. |
| 15:34.26 | Guest_536 | i can't seem to find an example of detecting tones... |
| 15:35.12 | Guest_536 | anyone tried app_mwanalyze? |
| 15:35.54 | grandpapadot | Guest_536: Does it require app_ky.so? |
| 15:36.13 | Guest_536 | not that i know of.... |
| 15:37.04 | Guest_536 | u see.. asterisk, now, brings an app_alarmreceiver for receiving alarm events in a protocol called ademco contact_id.. |
| 15:37.31 | Guest_536 | it's working fine, except, i want to be able to send these events thru asterisk to another central |
| 15:37.52 | Guest_536 | so asterisk has a receiver but not a transmitter... |
| 15:37.58 | Guest_536 | need this... |
| 15:39.10 | Guest_536 | it's quite simple, because the transmitter uses dtmf to comunicate an event... BUT.. the protocol requires an ACK and Kissoff, and THAT is done thru tone detection... |
| 15:39.50 | Guest_536 | like 1400Mhz during 50-60ms followed by 50-60ms silence followed by 2300Mhz 50-60ms ........ |
| 15:40.11 | Guest_536 | so it has to be compiled into a new module/function |
| 15:40.24 | Guest_536 | never done it... thus my dificulty |
| 15:40.40 | Guest_536 | anyone!? |
| 15:42.17 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
| 15:43.18 | *** join/#asterisk luckyaba (n=lucky@ip72-194-218-169.sb.sd.cox.net) |
| 15:43.19 | grandpapadot | Hey all, anyone else seen an issue with 1.4.29 where app_followme.so places calls, but then doesn't play prompts and won't accept input to the "accept call" prompts? This issue survives a restart. I thought it was file permissions on the promtps but no joy and there isn't any issues in mantis that I can find, it's affecting multiple installs, looking for guidance. |
| 15:43.33 | voipmonk | Guest_536: reading... stdby |
| 15:44.02 | voipmonk | you want to send event sto another "central" what is that? |
| 15:44.13 | [TK]D-Fender | grandpapadot: No more than 15 minutes and less than a full screen ago.... |
| 15:44.55 | Guest_536 | voipmonk: using asterisk to fire alarm events to alarm central companies |
| 15:45.18 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:45.20 | grandpapadot | [TK]D-Fender: I have small screen? lol, k, sorry, impatient, it's affecting about 100 of our customers, about to roll back to 1.4.28. |
| 15:46.10 | [TK]D-Fender | grandpapadot: You also come in bringing no debug or configs.Some people really don't know how to do themselves any favours |
| 15:46.40 | fenrus | rollback |
| 15:46.47 | fenrus | fault-trace in a calm enviroment |
| 15:47.24 | grandpapadot | .. The followme config is starighth forward, nothing there .. Way too much traffic to debug .. was looking for an easy way out of the issue |
| 15:48.12 | grandpapadot | .. and unfortunately there is no errors |
| 15:48.23 | *** join/#asterisk Znuff (n=ibm86@2001:0:53aa:64c:2c26:2f82:a65a:7c98) |
| 15:49.10 | [TK]D-Fender | grandpapadot: Ok, this kind of feedback tends to get one lumped in the "Crack-pot conspiracy theorist" category... |
| 15:49.38 | [TK]D-Fender | grandpapadot: In a "please someone corroborate so I don't look COMPLETE crazy" kinda way |
| 15:50.15 | *** join/#asterisk slidesinger (n=slidesin@c-68-44-99-50.hsd1.nj.comcast.net) |
| 15:50.35 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:50.58 | fenrus | "too much traffic to debug" ? |
| 15:51.05 | fenrus | wtf, take one call and trace it. |
| 15:51.25 | fenrus | how do you think a large provider debugs stuff? |
| 15:53.07 | grandpapadot | TK/fenrus - sorry guys, was looking for a fast path on this one, going to roll it back, thanks for the help, cheers! |
| 15:53.22 | Guest_536 | anyone compiled app_mwanalyze ? |
| 15:54.57 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 15:56.41 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:57.40 | vader-- | what is a DSX-1 port? |
| 16:00.05 | Katty | t1 |
| 16:00.07 | Katty | or e1 |
| 16:00.09 | [TK]D-Fender | vader--: http://en.wikipedia.org/wiki/T-carrier |
| 16:00.24 | [TK]D-Fender | vader--: Google-able, and wikipediable :) |
| 16:01.30 | vader-- | im trying to figure out if this AdTran TA924 will do what i need it says it has 1 DSX-1 port, but the guy at adtran said it was just made to connect to a user's PBX |
| 16:01.45 | vader-- | here is what the datasheet says |
| 16:01.45 | vader-- | The TA 900 series only has 1 WAN T1 port for data termination and one DSX-1 for T1/PRI emulation toward the customer PBX. These ports are in a fixed configuration and |
| 16:01.45 | vader-- | cannot be programmed to operate in any other fashion. The Ethernet port may be used as a LAN or WAN interface depending on customer requirements. |
| 16:02.00 | vader-- | we have a PRI coming to our building |
| 16:04.31 | [TK]D-Fender | vader--: Doesn't sounds like the right product then. |
| 16:04.45 | [TK]D-Fender | vader--: this seems to take in Data T1 and spit out PRI (IAD) |
| 16:05.09 | [TK]D-Fender | vader--: Perhaps you could clarify your actual needs. |
| 16:05.39 | *** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:07.06 | ManxPower-work | A DSX-1 port is a T-1 port with internal crossover so you don't need a crossover cable to talk to the telco. |
| 16:07.25 | vader-- | im looking for a VOIP Gateway that will take our PRI (23 Channel trunk) line in and present it to the asterisk server but also have 24 ports of FXS available for internal phones that will be presented to asterisk server as well |
| 16:07.32 | ManxPower-work | vader--: Why are you not trying to just bring the line directly into Asterisk? |
| 16:07.47 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 16:07.52 | vader-- | the guy at Adtran said i would need the 924e which is like double the money |
| 16:08.03 | [TK]D-Fender | vader--: So a super-all-in-on-plus-kitchen-sink-appliance? |
| 16:08.09 | Katty | sounds like a splitter, of sorts |
| 16:08.17 | ManxPower-work | I still don't see what you are going to use the Adtran for. |
| 16:08.27 | vader-- | i don't want to use PCI cards |
| 16:08.28 | [TK]D-Fender | vader--: So it has to be a PRI to VoIP + FXS gateway? |
| 16:08.33 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 16:08.38 | Katty | but i've never heard of an adtran box functioning as fxs for phones |
| 16:08.39 | vader-- | im looking at virtualizing our asterisk box |
| 16:08.41 | ManxPower-work | vader--: Then you'll have to get the more expensive one. |
| 16:08.50 | *** join/#asterisk ketema (n=ketema@ketema.net) |
| 16:08.59 | Katty | but hey! i haven't worked with adtrans much (= |
| 16:09.03 | ManxPower-work | vader--: "virtualize your asterisk box"? Oh. Sorry. I thought you were sane. |
| 16:09.08 | vader-- | :-D |
| 16:09.13 | [TK]D-Fender | vader--: sounds like you need 2 units, or one large & undoubtably exensive combo unit should one even exist |
| 16:09.14 | vader-- | just toying with the idea |
| 16:09.31 | vader-- | here is one of the data sheets |
| 16:09.32 | vader-- | http://kb.adtran.com/display/2n/kb/article.asp?aid=3050 |
| 16:09.34 | [TK]D-Fender | expensive* |
| 16:09.41 | ManxPower-work | vader--: you are going to need something to convert the PRI into FXS. You're not going to find that in an Adtran |
| 16:09.43 | vader-- | http://kb.adtran.com/display/2n/kb/article.asp?aid=1371 |
| 16:09.48 | jkroon | is there anybody here that would be able to assist me with decoding and understanding a RTCP packet? |
| 16:09.51 | [TK]D-Fender | vader--: Bad link |
| 16:10.12 | bmoraca_work | vader--: you do not need the 924e to do what you want. trust me :) |
| 16:10.34 | ManxPower-work | bmoraca_work: it looks like no product will do what he wants. He wants to use this product without plugging it into an Asterisk server. |
| 16:10.46 | vader-- | im not looking for the adtran to directly switch between the PRI and the FXS |
| 16:10.52 | bmoraca_work | ManxPower-work: he can do that...he just uses it in the opposite way i use it. it'll work just fine |
| 16:10.54 | ManxPower-work | vader--: Yes, you are. |
| 16:10.57 | vader-- | no i want to use an asterisk server |
| 16:11.09 | vader-- | PRI <--> SIP <--> Asterisk |
| 16:11.16 | vader-- | well more like this |
| 16:11.32 | vader-- | PRI <--> Adtran TA 924 <--SIP--> Asterisk |
| 16:11.40 | [TK]D-Fender | vader--: AudioCodes MP-124 + Mediant 1000 |
| 16:11.40 | jkroon | i've got a packet (pcap) capture of a packet trace that I think is a Counterpath problem, and they say must be a fault on asterisk's side - as far as I can see it's them closing the call down. Would someone mind taking a look? |
| 16:11.53 | vader-- | FXS <--> Adtran TA 924 <---SIP---> Asterisk |
| 16:12.29 | [TK]D-Fender | vader--: 2 boxes |
| 16:12.31 | bmoraca_work | [TK]D-Fender: the TA924 is the cost of the MP-124 and the adtran alone will do what he needs |
| 16:13.37 | bmoraca_work | vader--: all you need is a T1 crossover cable (or to roll the T1 jack your telco extended) and you'll be able to connect the TA924 and use it like you want |
| 16:14.48 | bmoraca_work | lol @ Spyker buying Saab |
| 16:14.55 | [TK]D-Fender | bmoraca_work: So far that spits out T1 to *. That doesn't sound like what he wants |
| 16:15.05 | bmoraca_work | fail buys fail |
| 16:15.32 | bmoraca_work | [TK]D-Fender: that's only one use of the TA924 (how I use it)...but there is no technical reason (real or artificial) why he wouldn't be able to use it the other way |
| 16:15.40 | ManxPower-work | bmoraca_work: So basically he would need a box with 1 pstn port and two CPE ports? |
| 16:15.53 | ManxPower-work | Plus the 24 FXS, of course. |
| 16:15.58 | [TK]D-Fender | bmoraca_work: the is takes in 24 FXS, and a PRI, and spits both back to * via SIP? |
| 16:16.04 | [TK]D-Fender | that it* |
| 16:16.16 | *** part/#asterisk Guest_536 (n=IceChat7@80.172.228.33) |
| 16:16.17 | bmoraca_work | [TK]D-Fender: yep, no reason why the TA924 shouldn't be able to do that |
| 16:16.17 | Katty | the squirrel is just sitting there. |
| 16:16.19 | Katty | staring. |
| 16:16.30 | [TK]D-Fender | bmoraca_work: Doesnt' fit the data sheet |
| 16:16.33 | vader-- | ManxPower right now i have two PCI cards in my asterisk box, one PRI which handles my trunk and one 24 port FXS card which handles my analog phones/faxes. The other 60 phones i have are cisco IP phones |
| 16:16.53 | bmoraca_work | [TK]D-Fender: data sheets aren't always correct...i've been using these things for a while now |
| 16:17.00 | vader-- | i want to eliminate the PCI cards and try and use the adtran ta924 device to take the place of those two cards |
| 16:17.15 | [TK]D-Fender | bmoraca_work: bmoraca_work If you say so... but I'd want better stuff in writing before I'd spend money |
| 16:17.17 | ManxPower-work | vader--: so you want to take a working system and make it more complicated? |
| 16:17.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.41 | bmoraca_work | ManxPower-work: it's not more complicated...it's less complicated...one peice of equipment is replacing three peices of equipment |
| 16:17.48 | ManxPower-work | Send me enough money and I'll drive the 6 miles to the Adtran HQ and ask them direct. |
| 16:17.58 | Qwell | Send me more, and I'll drive the .5 |
| 16:18.07 | bmoraca_work | lol |
| 16:18.11 | vader-- | well if i move asterisk to our virtual enviroment i will be able to perform upgrades easier, backups easier, better redundancy, i can eliminate a physical server (cooling/electrical) |
| 16:18.20 | vader-- | and our current server is out of warranty |
| 16:18.32 | vader-- | gonna cost couple grand to renew the service contract or buy a new server |
| 16:18.33 | Qwell | For the right price, I'll go over to our "friends" at Raytheon and do whatever you'd like too. |
| 16:18.43 | Qwell | "friends", "neighbors", whatever |
| 16:19.06 | Qwell | (really, they're our arch-enemy) |
| 16:19.09 | Katty | service contract? |
| 16:19.09 | [TK]D-Fender | [11:18]<Qwell>For the right price, I'll go over to our "friends" at Raytheon and do whatever you'd like too. <- sounds dirty |
| 16:19.13 | Katty | on a ....server? |
| 16:19.15 | vader-- | our asterisk server really only sees like 5-6 active concurrent calls at a time |
| 16:19.16 | Katty | for what? ^_- |
| 16:19.22 | Qwell | [TK]D-Fender: for the right price... |
| 16:19.23 | Qwell | :p |
| 16:19.40 | vader-- | max when we have like a call-a-thon type deal it's 8-10 active calls |
| 16:19.41 | ManxPower-work | vader--: I agree with trying to get rid of the analog card. |
| 16:19.43 | bmoraca_work | vader--: give me a few hours and i'll test this out for you. i've got some other things i need to look at, so it may take me a few minutes |
| 16:19.46 | [TK]D-Fender | Qwell: thats just haggling :p |
| 16:19.59 | vader-- | mostly from IP phone <---> Asterisk <---> PRI Line |
| 16:20.51 | Qwell | no, but really.. we'd probably start a war with Raytheon if they didn't have like...you know...real weaponry. |
| 16:21.19 | vader-- | our VMware Cluster has High Availability, 2hr 24x7 service contracts, i can manage everything remotly |
| 16:21.19 | Katty | wonders what Raytheon is now |
| 16:21.32 | vader-- | can restore a server in less than 10 minutes |
| 16:21.33 | vader-- | full metal |
| 16:21.39 | Qwell | Katty: like 90% of other companies in Huntsville, they're a defense contractor |
| 16:21.40 | vader-- | bare metal |
| 16:21.47 | Katty | Qwell: oh. |
| 16:21.49 | Katty | Qwell: i see. |
| 16:21.49 | vader-- | so i figured i would toy with the idea |
| 16:21.55 | Katty | Qwell: i can haz defense? |
| 16:22.05 | Katty | actually i'd like to hire a landscaper. |
| 16:22.25 | Katty | maybe they can put a small crater.....errr pond in the back yard. |
| 16:22.27 | Qwell | http://www.raytheon.com/businesses/rms/ |
| 16:22.28 | Qwell | http://www.raytheon.com/businesses/rids/about/index.html |
| 16:22.53 | bmoraca_work | vader--: like i said...i've got a virtualized asterisk box that does 10 concurrent g729 calls to remote phones and it works flawlessly (well, now it does since i ditched the asterisk-based media gateway and moved to an AS5400, but that's besides the point) |
| 16:22.59 | Katty | why are they your arch-enemy, Qwell? |
| 16:23.11 | Qwell | Katty: they're next door. who else would be our arch-enemy? |
| 16:23.17 | Qwell | the farm? pfft |
| 16:23.28 | Katty | why don't you build a nice little coffee shop next door. |
| 16:23.29 | Qwell | OR! The high school! |
| 16:23.36 | Qwell | we have a coffee shop |
| 16:23.38 | ManxPower-work | They are a defense contractor, aren't they? They would be everyone's arch enemy then. |
| 16:24.14 | bmoraca_work | shhh...i'll be making 1k per month off a defense contractor here pretty quick...they're not MY enemy :P |
| 16:24.38 | Katty | only if there's a civil war. |
| 16:24.47 | Katty | or an accident. |
| 16:25.06 | ManxPower-work | bmoraca_work: Are you as proud to provide service to telemarketers? |
| 16:25.21 | Katty | there some telemarketers down the hallway here. |
| 16:25.25 | Katty | they're all fruitcakes. |
| 16:25.25 | bmoraca_work | ManxPower-work: i don't provide service to telemarketers...it's strictly against my ToS |
| 16:25.45 | ManxPower-work | bmoraca_work: Excelent. |
| 16:25.56 | p3nguin | I have people asking me every day to set up "bulk" email servers. I have to decline. |
| 16:26.36 | Katty | p3nguin: did you see the new broadcast? (= |
| 16:27.02 | ManxPower-work | I hate html forms |
| 16:27.04 | Katty | p3nguin: it's a mostly sunny day down here, but there is talk of ice and snow come thursday and friday |
| 16:27.05 | bmoraca_work | ManxPower-work: actually, their contract for maintaining the local ammo plant is expiring in a year and they're downsizing while the city prepares to take it over...they've got a gigantic AT&T Definity system right now that they don't want to bother moving to their new location...they have 5 100-pair lines coming in from the telco... |
| 16:27.08 | p3nguin | katty: I don't know that I saw the old one... but I'll look at the new one if you'll give me the link. |
| 16:27.32 | Katty | infobot: crittercam |
| 16:27.33 | infobot | i heard crittercam is Katty's broadcast of The Nut House Critter Cam @ http://ustre.am/8H5d and The Nut House Bird Bath @ http://ustre.am/bEBU |
| 16:27.37 | Katty | p3nguin: any talk of ice/snow up north? |
| 16:27.39 | ManxPower-work | "If I don't sell drugs to kids, someone more evil will!" |
| 16:28.03 | Katty | we seem to have the opposite policy here |
| 16:28.13 | Katty | if i don't sell these guys a phone system /somehow/ i might lose a quarter |
| 16:28.17 | p3nguin | It killed my browser. |
| 16:28.18 | Katty | which is why we sell not only asterisk... |
| 16:28.22 | dandre | I have setup a queue with ringinuse=no and joinempty=strict. If all my agents are online the caller stays in queue. I'd like to have this call handled by next step in my dialplan. How can I do? |
| 16:28.24 | Katty | but shortel, and talkswitch, and samsung, and toshiba... |
| 16:28.27 | ManxPower-work | bmoraca_work: one of my biggest issues with my last job was that we sold to telemarketers |
| 16:28.37 | bmoraca_work | ManxPower-work: yuk |
| 16:28.42 | Katty | do you know how difficult it is to keep up with all those phone systems? |
| 16:29.09 | p3nguin | katty: I think we might see some snow later tonight and possibly on Friday. |
| 16:29.09 | Katty | i need to borrow your neighbors, Qwell, and take out a few of our sales reps. |
| 16:29.17 | Katty | p3nguin: nothing much i take it? |
| 16:29.34 | leifmadsen | we have flurries right now |
| 16:29.39 | p3nguin | katty: They're just calling for "snow showers." |
| 16:29.42 | Katty | in...huntsville, leifmadsen? |
| 16:29.50 | leifmadsen | toronto |
| 16:30.01 | Katty | i always think you're in huntsville with twisted, for some reason |
| 16:30.17 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:30.21 | p3nguin | isn't manxpower-work in huntsville? |
| 16:30.26 | ManxPower-work | ~manxpower |
| 16:30.27 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 16:30.27 | Katty | i think so. |
| 16:30.36 | ManxPower-work | oh! That doesn't say where I am. |
| 16:30.36 | Katty | i guess not. |
| 16:30.38 | anonymouz666 | 2001 ! |
| 16:30.41 | ManxPower-work | Yes, I'm in Huntsville. |
| 16:30.46 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 16:30.47 | bmoraca_work | ManxPower-work: do you serve Fort Worth, TX? |
| 16:30.56 | Katty | hi Naikrovek |
| 16:31.01 | anonymouz666 | ManxPower-work: how stable was asterisk in 2001? :D |
| 16:31.08 | Katty | Naikrovek: i have HD broadcasting capabilities now! :> |
| 16:31.09 | Naikrovek | bonjour |
| 16:31.14 | Naikrovek | link? |
| 16:31.23 | ManxPower-work | bmoraca_work: TX and FL are not our primary territories, but I can ask. What sorts of service? |
| 16:31.38 | Katty | Naikrovek: http://ustre.am/bEBU |
| 16:31.50 | ManxPower-work | anonymouz666: as long as you didn't do anything stupid, it was reasonably stable most of the time. |
| 16:32.07 | Katty | Naikrovek: that one's actually at 720, the other broadcast, slightly to the left is the 1080...workstation can only seem to broadcast one at HD at a time... |
| 16:32.09 | Naikrovek | Katty: nice sharp pic |
| 16:32.15 | anonymouz666 | as long as you didn't start the asterisk then |
| 16:32.20 | Katty | Naikrovek: but that will come soon enough when i get another puter to stick behind the couch (= |
| 16:32.29 | bmoraca_work | ManxPower-work: wholesale termination/origination and e911 provided over SIP...i've got a potential customer who has an office there that he wants to integrate...VOIP.ms and globalpops both serve it, but i'm always looking for better prices |
| 16:32.42 | Katty | Naikrovek: i know, isn't it? the other camera's gonna be so awesome. i think i'll broadcast the left one in HD tomorrow |
| 16:32.49 | bmoraca_work | my primary service provider doesn't serve Ft. Worth |
| 16:32.51 | Naikrovek | :) |
| 16:33.00 | Naikrovek | lookit that squirrel |
| 16:33.03 | Katty | i know |
| 16:33.11 | Katty | he was all, IS SOMEONE THERE AT THE WINDOW |
| 16:33.12 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:33.16 | Naikrovek | i know |
| 16:33.45 | Katty | Naikrovek: i had visitors to the bird bath this morning! |
| 16:33.51 | Katty | Naikrovek: but...they weren't birds :/ |
| 16:33.58 | Naikrovek | what were they then |
| 16:34.03 | Katty | squirrels, of course. |
| 16:34.07 | Naikrovek | ah |
| 16:34.14 | Naikrovek | well they gotta drink too i suppose |
| 16:34.16 | ManxPower-work | bmoraca_work: I could give you a sales rep contact information, but we are mainly standard business service and not wholesale. |
| 16:34.17 | Katty | my guess is they thought it was another feeding station, cleverly hidden in a tree |
| 16:34.43 | Katty | Naikrovek: of course. i'm not upset about it or anything. it was the original intent to put water in both reach of the birds and the fuzzballs. |
| 16:34.57 | ManxPower-work | secretly puts vodka in the waterer of those bushy tail'd rats. |
| 16:35.10 | Naikrovek | i'd like to see a drunk squirrel |
| 16:35.12 | Katty | i bet that'd be fun to watch. |
| 16:35.14 | bmoraca_work | ManxPower-work: that's what i thought. we'll probably just use globalpops...we've had a pretty good relationship with them in the past |
| 16:35.21 | Naikrovek | go to make a jump, miss, and basically become an airball |
| 16:35.31 | Katty | hey i've seen that before |
| 16:35.36 | Katty | they jump to get on the light house...and WOOOSH |
| 16:35.55 | Katty | usually it's when they don't realize another squirrel is already on it, and it messes them up midflight ;) |
| 16:36.48 | Naikrovek | is that apple tablet reveal today |
| 16:36.52 | Naikrovek | i think it is.. |
| 16:37.05 | Katty | i think i saw that on reddit not too long ago |
| 16:38.32 | bmoraca_work | who cares about the apple tablet? people are hailing it like it's the best invention since the toaster-oven when tablets have been around for over a decade. |
| 16:39.00 | Naikrovek | guy in my office here is salivating |
| 16:39.09 | Naikrovek | he'll buy anything with an apple logo on it |
| 16:39.24 | Naikrovek | doesn't understand that he's being led by the nose |
| 16:39.30 | Naikrovek | soooo funny to watch |
| 16:40.21 | bmoraca_work | SUCCESS! woot |
| 16:40.55 | bmoraca_work | my 20 parameter function and query work properly |
| 16:41.06 | Naikrovek | 20 parameters? |
| 16:41.08 | Naikrovek | wow |
| 16:41.29 | Naikrovek | my spidey sense tingles when i see param lists that long. maybe it's normal in asteriskland |
| 16:41.41 | bmoraca_work | oh, wasn't an asterisk app |
| 16:41.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:42.18 | bmoraca_work | i'm developing an app for a local municipality for maintaining their police department citations...going to integrate with asterisk later for a telephone service...but for now, i'm still building the main app |
| 16:42.31 | ManxPower-work | bmoraca_work: Apple has some very cool, and over priced products. |
| 16:43.09 | bmoraca_work | ManxPower-work: i agree that multitouch is pretty neet...but none of their products are priced appropriately and their customers are like zealots |
| 16:43.36 | Naikrovek | bmoraca_work: what language |
| 16:43.44 | bmoraca_work | Naikrovek: ColdFusion |
| 16:43.52 | Naikrovek | ah i know nothing about CF |
| 16:44.10 | Naikrovek | in java you'd be a fool to pass that many parameters. create an object and pass the object |
| 16:44.40 | coppice | someone complained to me today about the price of apple's fancy new mouse, until I reminded him the logitech one he is using cost the same. apple stuff has such a reputation for high prices people complain about the price before comparing |
| 16:45.42 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:45.46 | Katty | it can't be tha tmuch surely |
| 16:46.00 | Katty | i have a super duper buttony mouse for my WoW macros that costed upwards of 80 bucks. |
| 16:46.14 | Katty | is it more expensive than that? |
| 16:46.22 | [TK]D-Fender | coppice: And unlike all of Apple's previous mice it has a property they lacked : not sucking |
| 16:46.42 | [TK]D-Fender | Katty: 70$ CAD (~USD) |
| 16:47.56 | Katty | that's about what i'd expect a good quality mouse to cost. |
| 16:47.56 | Katty | but it's probably above the typical user's $30 microsoft laser mousey |
| 16:47.56 | Katty | or the el cheapo $10 rosewill mouses |
| 16:48.01 | [TK]D-Fender | Katty: its got a full-top surface capacitive trackpad instead of a wheel |
| 16:48.06 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:48.13 | Katty | very neat! |
| 16:48.39 | [TK]D-Fender | Katty: I can say that it is a very nice mouse |
| 16:49.00 | ManxPower-work | coppice: I was referring to "in general", not "compared to other products" |
| 16:49.25 | ManxPower-work | Of course, I'm one of those people that uses a $30 prepay cell phone from Walmart. |
| 16:49.36 | paulc | looks stunned |
| 16:49.39 | paulc | by WHY? |
| 16:49.49 | paulc | and by "by WHY?" I obviously mean "but WHY?" |
| 16:49.49 | ManxPower-work | paulc: because it's cheap and it makes and receives calls. |
| 16:50.01 | paulc | rubs eyes |
| 16:50.02 | paulc | sorry |
| 16:50.05 | paulc | I'm still waking up/getting into work |
| 16:50.16 | paulc | I thought I read 'prepaid phone card' not 'prepay cell phone' |
| 16:50.17 | ManxPower-work | My service is $40/month unlimited talk/text |
| 16:50.20 | paulc | kicks self |
| 16:50.33 | Katty | prepaid p hone cards have purposes. |
| 16:50.33 | ManxPower-work | paulc: I'm on a "monthly pre-pay" plan |
| 16:50.51 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-cssgooeiplcyvqrl) |
| 16:50.52 | paulc | Nothing wrong with prepaid. I used one for a while, built up some nice Fido Dollars, then combined with my regular/post paid account to get a sweet deal on my iPhone |
| 16:51.18 | Katty | are minors allowed to carry cellphones at school? |
| 16:51.26 | paulc | Interestingly, they just launched a prepaid City Fido plan here.. $40 or $45 a month for unlimited SMS and 1000 minutes a month |
| 16:51.34 | ManxPower-work | *nod* Pre pay is SO much more expensive than contract plans on most carriers, at least I found one where that was not the case. |
| 16:51.39 | paulc | Katty: generally not? |
| 16:51.44 | ManxPower-work | tries to remember the name |
| 16:51.45 | coppice | Katty: its pointless. cellphones won't work in mines |
| 16:51.51 | Katty | coppice: ehehe |
| 16:51.53 | Katty | coppice: oh that was good |
| 16:51.55 | Katty | coppice: <3 |
| 16:52.00 | paulc | LOL nice one! :-) |
| 16:52.18 | [TK]D-Fender | loves his 250 DT, EW @ 6PM, CW, CID, And unlimited data @ 40$ |
| 16:52.22 | Katty | i don't think there were any /real/ cellphones when i was in school |
| 16:52.34 | gr0mit | groans belatedly at coppice 's joke |
| 16:53.08 | ManxPower-work | pagepluscellular.com They are an MVNO using Verizon's network |
| 16:53.15 | paulc | hates how Canadian carriers gouge the customers |
| 16:53.31 | ManxPower-work | Which is important since I'm frequently in an area where ONLY Verizon has service. |
| 16:53.37 | *** join/#asterisk FinboySlick (n=shark@74.117.40.10) |
| 16:53.40 | bmoraca_work | i get my cell free from work...my wife uses MetroPCS: $30/mo unlimited talk/text |
| 16:53.41 | jan__ | can anyone help with skype for asterisk? i got every thing installed on asterisknow, but asterisk won't login to tthe skype account... |
| 16:53.45 | Katty | my mobile plan is free. |
| 16:53.48 | Katty | it's unlimited everything. |
| 16:53.49 | ManxPower-work | ~asterisknow |
| 16:53.50 | infobot | asterisknow is probably based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 16:54.03 | [TK]D-Fender | ManxPower-work: Bad referral |
| 16:54.10 | Katty | well, i don't think it has gps turned on |
| 16:54.35 | ManxPower-work | [TK]D-Fender: pagepluscellular.com works for me |
| 16:54.42 | [TK]D-Fender | Katty: Check your phone. Mine has a hardware GPS. that + unlimited Google Maps = win |
| 16:55.11 | Katty | checks |
| 16:55.15 | Naikrovek | all phones have hardware gps now whether they advertise it or not |
| 16:55.20 | Naikrovek | for 911 purposes |
| 16:55.34 | Naikrovek | erm, alll MOBILE phones |
| 16:55.40 | [TK]D-Fender | Naikrovek: And Big Brother |
| 16:55.47 | Naikrovek | yes, very much |
| 16:56.02 | Katty | well yeah |
| 16:56.06 | Katty | so if you report it stole |
| 16:56.07 | Naikrovek | set your phones to e911 gps only |
| 16:56.09 | FinboySlick | Anyone around here using GS-486 analog voip adaptors? I don't expect wonders out of them, but I'm curious if they'd be suitable to our needs. I'd like something that could use two distinct vlans. Basically be a voip client on one vlan and bridge ethernet on another. |
| 16:56.14 | Katty | but i don't think our company pays for just general gps |
| 16:56.17 | p3nguin | katty: I had a real cell phone when I was in high school... I thought you were younger than me. |
| 16:56.34 | Katty | my dad had a Bag Phone in his car |
| 16:56.48 | paulc | FinboySlick: I wouldn't hold your breath on that front - GS = kinda basic mostly |
| 16:57.00 | Katty | p3nguin: how old ar eyou? |
| 16:57.06 | p3nguin | 31 |
| 16:57.12 | Katty | well i am younger than you |
| 16:57.31 | p3nguin | I had it in my head that you're 27. |
| 16:57.36 | Katty | i'm not 27 |
| 16:57.41 | Katty | i don't think |
| 16:57.44 | Katty | gets calculator |
| 16:57.47 | Naikrovek | lol |
| 16:57.47 | ManxPower-work | <-- *sob* 40 *sob* |
| 16:57.48 | FinboySlick | paulc: Well, their cheap GS-2000 phones do exactly what I need so maybe the 486 would be an analog adapter equivalent. |
| 16:58.01 | p3nguin | katty: Not sure why I would think that, then. |
| 16:58.02 | Katty | no, i'm not 27 lol, i'm 25 |
| 16:58.21 | p3nguin | ;) |
| 16:58.21 | Katty | that's pretty bad when you can't remember your own age. |
| 16:58.28 | p3nguin | it happens |
| 16:58.28 | Naikrovek | not really |
| 16:58.34 | Naikrovek | just means it's not important to you |
| 16:58.44 | paulc | or you're blind drnk |
| 16:58.46 | paulc | drunk |
| 16:58.46 | Katty | well, no, it's not very important |
| 16:58.55 | Naikrovek | i don't know my shoe size, and it hasn't changed in a decade |
| 16:58.58 | Naikrovek | i have to check |
| 16:59.02 | paulc | And so long as you're not breaking the cougar rule - it's all good, right? ;-) |
| 16:59.03 | Naikrovek | why? because it's not important to me |
| 16:59.04 | Katty | i do know my shoe size |
| 16:59.09 | Katty | in both sneakers and dress, and flats. |
| 16:59.13 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 16:59.17 | Katty | HAI SEANBRIGHT |
| 16:59.22 | seanbright | yes |
| 16:59.28 | *** join/#asterisk Cadey (n=x@mail.officebroker.com) |
| 16:59.43 | Katty | Naikrovek: you probably don't have 20 pairs of shoes either |
| 16:59.47 | Cadey | Hi peeps, does astrisk support DPNSS |
| 16:59.47 | [TK]D-Fender | has traditionally been cougar-bait |
| 17:00.05 | Naikrovek | Katty: correct. i have 2 pairs of shoes |
| 17:00.06 | Katty | mrawrr. |
| 17:00.15 | Katty | Naikrovek: dress and casual, i'm guessing |
| 17:00.17 | p3nguin | Anyway, I had a Motorola Populus. It was a neat phone... took a battery pack or regular AA alkalines. |
| 17:00.24 | Naikrovek | katty: work and other |
| 17:00.25 | Katty | Naikrovek: or casual and flipflops? |
| 17:00.53 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:00.56 | Katty | or sandals. i guess all the cool kids call them sandals now. |
| 17:01.58 | p3nguin | Google image search doesn't even know what a Populus is. |
| 17:02.10 | p3nguin | I need to see if I can find mine and give them some images of it. |
| 17:02.13 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:02.24 | Katty | p3nguin: i'd be interested in seeing it too |
| 17:02.48 | p3nguin | It's kind of boring to look at, but the idea of running on 4 AA batteries was really cool to me. |
| 17:03.21 | Katty | it'd be nice for emergency situations |
| 17:03.29 | p3nguin | I had service with Ameritech, before Verizon took over their wireless stuff. |
| 17:04.21 | *** join/#asterisk profxavier (n=chatzill@unaffiliated/neverblue) |
| 17:04.27 | p3nguin | They have a pic of the battery pack for it. http://www.akkushop.de/images/Motorola-D160.jpg |
| 17:04.56 | ManxPower-work | It looks like Verizon staffed up it's "randomly screw up your wholesaler's lines department" again |
| 17:05.03 | profxavier | I have a Polycom IP400 phone, using PoE. When I plug it in, I have a static noise exiting the speaker |
| 17:05.13 | profxavier | where do I start this challenge? |
| 17:05.18 | ManxPower-work | profxavier: your phone is defective. return it. |
| 17:05.25 | Naikrovek | ^^^^^^ |
| 17:05.29 | profxavier | hmm, thats quick |
| 17:05.32 | profxavier | and easy :) |
| 17:05.36 | ManxPower-work | profxavier: polycoms should never have "static" on the speaker when you plug them in. |
| 17:05.37 | profxavier | thanks ManxPower-work |
| 17:05.43 | p3nguin | manxpower-work: http://www.wsu.edu/~brians/errors/its.html |
| 17:05.48 | Naikrovek | suppose it could be a bad handset cord |
| 17:05.57 | profxavier | its a speaker phone |
| 17:06.01 | profxavier | no handset :) |
| 17:06.02 | Naikrovek | ah heh |
| 17:06.04 | Naikrovek | lol |
| 17:06.08 | Naikrovek | i'm so stupid sometimes |
| 17:06.10 | Naikrovek | yes bad phone |
| 17:06.15 | profxavier | *hiccup* |
| 17:06.54 | ManxPower-work | p3nguin: http://encyclopediadramatica.com/Grammar_Nazi |
| 17:07.31 | p3nguin | manxpower-work: Don't deflect. |
| 17:07.39 | p3nguin | I was just trying to help you see your error, but you're trying to be insulting. |
| 17:07.54 | Naikrovek | he takes correction about as well as i do |
| 17:08.07 | p3nguin | I wasn't even mean about it. |
| 17:08.16 | Naikrovek | i know |
| 17:08.35 | Naikrovek | don't worry about it |
| 17:08.38 | p3nguin | I just think everyone should know the difference. |
| 17:08.49 | Naikrovek | for me, i'll admit it's a personality flaw |
| 17:08.51 | ManxPower-work | p3nguin: Everyone should know the difference. I was just not paying attention. |
| 17:10.10 | Katty | are we really going to have a conversation about its its and it's |
| 17:10.17 | Katty | srsly. |
| 17:10.29 | leifmadsen | yes |
| 17:10.34 | Katty | :< |
| 17:10.45 | leifmadsen | people can't resist a good grammar rodeo |
| 17:11.09 | Katty | we should throw marshmallows at them |
| 17:11.18 | leifmadsen | agreed |
| 17:11.25 | leifmadsen | I'm not a big fan of marshmallows |
| 17:11.28 | Katty | WHAT |
| 17:11.34 | leifmadsen | unless they are made into a s'more |
| 17:11.58 | Chainsaw | But marshmallows are awesome. Especially in hot chocolate. |
| 17:12.01 | Katty | i like mine in hot chocolate. |
| 17:12.10 | Katty | it gets all melty |
| 17:12.11 | *** join/#asterisk uluatu (n=uluatu@187.59.72.153) |
| 17:12.12 | Katty | and gooey |
| 17:12.17 | Katty | and swirly :> |
| 17:12.21 | Chainsaw | Katty: It's fantastic. They used to make them in Creative Labs. |
| 17:12.25 | leifmadsen | ok, going to make lunch and have some hot-chocolate :) |
| 17:12.27 | Chainsaw | Katty: I miss that :/ |
| 17:12.35 | Katty | i miss the mini marshmallows |
| 17:12.41 | Katty | they still make the small ones |
| 17:12.43 | Katty | but not the minis |
| 17:12.57 | Katty | the minis were really really tiny...sometimes found in the instant packets of hot chocolate. |
| 17:13.07 | p3nguin | If you buy hot chocolate powered ... nevermind. |
| 17:13.22 | p3nguin | powdered, even |
| 17:13.32 | Katty | p3nguin: now do you seriously think i use instant hot chocolate? |
| 17:13.48 | p3nguin | No, but I was just sayin' where you can find them. |
| 17:14.05 | Chainsaw | Powdered?! We're talking about the good stuff here. |
| 17:14.11 | Katty | http://42ndrecipestreet.blogspot.com/2009/11/mexican-hot-chocolate.html |
| 17:14.16 | Katty | ^- that's the good stuff. |
| 17:14.17 | p3nguin | I don't mind the powdered mix, but I do require the mini marshmallows. |
| 17:14.18 | uluatu | What should I expect from progressinband=yes on sip.onf? |
| 17:14.34 | ManxPower-work | uluatu: pain and misery |
| 17:14.38 | [TK]D-Fender | uluatu: ridicule? |
| 17:14.41 | uluatu | im calling from xlite through asterisk to a phone niumber that gives me 183 |
| 17:15.22 | Chainsaw | Katty: Oh yes. I wish a store nearby sold it. |
| 17:16.07 | ManxPower-work | uluatu: you should check the value of HANGUPCAUSE and/or DIALSTATUS in your dialplan and play the correct audio message. |
| 17:16.21 | *** join/#asterisk albertoandrade (n=albertoa@187.59.72.153) |
| 17:17.28 | uluatu | ManxPower-work: hmmm. What I need is to get 180 ringing when the 183 sdp tells me that the phone is ringing... |
| 17:17.37 | uluatu | ManxPower-work: my app need the 180 always.... |
| 17:18.25 | ManxPower-work | uluatu: I wish you the BEST of luck. |
| 17:18.38 | [TK]D-Fender | uluatu: "vi chan_sip.c" |
| 17:19.07 | uluatu | [TK]D-Fender: Im already there, but it seems a little bit confuse... |
| 17:19.11 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:19.34 | Katty | 1/4 c of cocoa powder contains 50mg of caffeine. |
| 17:19.44 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 17:19.48 | p3nguin | yum |
| 17:19.55 | Katty | 1/4 c cocoa powder makes about 2 cups of hot chocolate. |
| 17:20.03 | *** join/#asterisk voxter (n=voxter@S010600090f53ea17.vc.shawcable.net) |
| 17:20.32 | p3nguin | How much caffeine is in a regular cup of dark roast coffee? |
| 17:20.36 | uluatu | [TK]D-Fender: My thoughts is that asterisk doesn't interpret the sdp from 183 messages |
| 17:20.50 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 17:21.07 | Katty | 2 cups of coke has 45mg of caffeine |
| 17:21.34 | p3nguin | That's what I need to get. I haven't had any cola for a while. |
| 17:21.40 | Katty | 1 cup of coffee is about 65-100 |
| 17:21.48 | ManxPower-work | uluatu: with progressinband=yes I would expect that. |
| 17:21.55 | Katty | depending on how strong you make it, obviously |
| 17:22.18 | uluatu | ManxPower-work: ok, ill try that and get back the results. |
| 17:22.23 | uluatu | thanks for now. |
| 17:22.23 | Katty | so hot chocolate, made from actual cocoa powder, is going to give you more of a jolt cup for cup of the caffeine |
| 17:22.36 | ManxPower-work | uluatu: check #asterisk-dev for code related questions |
| 17:22.37 | Katty | of course 1 cup of dr pepper or coke is going to run you 100 calories |
| 17:22.51 | Katty | and 1 cup of hot cocoa (made with sugar) is going to be a lot higher |
| 17:22.52 | uluatu | ManxPower-work: ok, thanks. |
| 17:23.13 | Katty | guess you could make your hot chocolate with splenda or somethin tho |
| 17:23.16 | Katty | and fat free milk |
| 17:23.20 | Katty | no cream, no marshmallows |
| 17:23.26 | Katty | where's the fun in that tho:< |
| 17:23.36 | p3nguin | I have some Mexican Coca-Cola, which is made with sugar instead of corn syrup. |
| 17:23.53 | p3nguin | splenda is bad. |
| 17:23.55 | Katty | there's a uhh... |
| 17:23.57 | Katty | it's for passover |
| 17:24.00 | Katty | kosher! |
| 17:24.06 | Katty | there's a kosher coke...it has the writing on the bottle |
| 17:24.11 | Katty | that's one which has sugar. |
| 17:24.13 | bmoraca_work | splenda is undigestable |
| 17:24.22 | Katty | bmoraca_work: splenda makes me ravenously hungry |
| 17:24.25 | ManxPower-work | Many companies release limited editions of their drinks with sugar for the lent season |
| 17:24.35 | *** join/#asterisk Tim_Toady (n=moi@77.49.165.135.dsl.dyn.forthnet.gr) |
| 17:25.10 | Katty | in fact i am ravenously hungry right now |
| 17:26.50 | ariel_ | Hello everyone |
| 17:26.56 | Katty | hi ariel_ |
| 17:27.32 | ariel_ | like Pepsi Throw back it is with sugar |
| 17:27.42 | Katty | i can't find the throw backs anymore |
| 17:27.44 | Katty | been lookin :< |
| 17:28.18 | ariel_ | Katty: hope your doing well. |
| 17:28.23 | Katty | am :> |
| 17:28.25 | Katty | you? |
| 17:29.41 | ariel_ | I am doing fine. Just a little cold, but that will not last more then today. |
| 17:30.10 | Katty | glad to hea ryou're doing well (= |
| 17:32.07 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:3c8b:2f33:a077:acde) |
| 17:32.08 | cusco | hello! |
| 17:32.31 | cusco | codecs can be defined for iax users individually, right? |
| 17:32.35 | cusco | allow; disallow |
| 17:33.24 | ariel_ | yes |
| 17:35.53 | *** join/#asterisk adam0563 (n=adam0563@173-14-238-70-Utah.hfc.comcastbusiness.net) |
| 17:35.54 | cusco | woa, I just disallowed all andonlya llowed ulaw |
| 17:36.00 | cusco | couldn't create channel |
| 17:36.06 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:36.34 | ManxPower-work | cusco: does the client have ulaw allowed? if so, pastebin your iax.conf |
| 17:36.35 | *** join/#asterisk elred_ (n=elred_@unaffiliated/elred-/x-5010831) |
| 17:37.07 | cusco | [Jan 27 17:35:15] WARNING[1993]: chan_iax2.c:10272 socket_process: Call rejected by 192.168.2.5: Unable to negotiate codec |
| 17:37.21 | cusco | these are two asterisk boxes |
| 17:37.34 | cusco | well I just allowed alaw |
| 17:37.41 | cusco | it works now |
| 17:38.07 | ManxPower-work | looks like the other box only had alaw enabled |
| 17:38.13 | cusco | since these asterisk boxes are on the same network, and bandwidth isn't a problem, what is the best codec? |
| 17:38.25 | cusco | ManxPower-work: weird because the allow configurations were pasted on both... |
| 17:38.31 | ManxPower-work | cusco: what country are you in? |
| 17:38.38 | cusco | ManxPower-work: portugal, europe |
| 17:38.42 | ariel_ | ulaw and alaw are basic same, just alaw is used with E1's |
| 17:38.48 | ManxPower-work | cusco: alaw would be best for you |
| 17:38.52 | cusco | thanks |
| 17:39.09 | cusco | now, soon we are to setup aother asterisk geographically far away |
| 17:39.14 | cusco | a iax trunk |
| 17:39.29 | cusco | what codec am I to use? bandwith is a problem |
| 17:39.47 | cusco | We were just using gsm, everybody complained about quality |
| 17:40.02 | ManxPower-work | cusco: GSM unless you want to buy the G729 codec. |
| 17:40.09 | cusco | :/ |
| 17:40.15 | ManxPower-work | oh! Asterisk <-> Asterisk? |
| 17:40.21 | cusco | i tried the free implementation of g729 in the past |
| 17:40.25 | ariel_ | if you have b/w issue gsm is 1/2 the size of ulaw, alaw, g729 is even smaller and a very good option is also ilbc |
| 17:40.27 | cusco | quality was weird |
| 17:40.29 | ManxPower-work | cusco: those don't work very well |
| 17:40.33 | cusco | ManxPower-work: yes asterisk to asterisk |
| 17:40.59 | cusco | g729 is free in asterisk to asterisk, right? |
| 17:41.01 | ManxPower-work | cusco: speex, iLIBC or G726 |
| 17:41.13 | *** join/#asterisk Heretic (n=fallen@dsl-246-66-176.telkomadsl.co.za) |
| 17:41.16 | cusco | (since it needs no encoding) |
| 17:41.17 | coppice | or LPC10 |
| 17:41.20 | ManxPower-work | cusco: no. transcoding g729 is not supported without a license |
| 17:41.31 | cusco | ok |
| 17:41.48 | cusco | speex has a metalic voice (so was i told) |
| 17:42.13 | cusco | iLIBC and G726 and LPC10 left to try |
| 17:42.14 | ManxPower-work | if you want to use g729 without a license then EVERYTHING will have to use G729, your phones, your asterisk, your asterisk audio files, everything. MeetMe also won't work. |
| 17:42.22 | cusco | ok |
| 17:42.26 | cusco | nevermind that then |
| 17:42.27 | coppice | speex gives excellent quality |
| 17:42.33 | cusco | are you sure? |
| 17:42.46 | cusco | I was told that speex would give you metalic robotic voices |
| 17:42.59 | ManxPower-work | cusco: why don't you try it and see? |
| 17:43.02 | coppice | by a G.729 salesman? |
| 17:43.07 | cusco | hehehe |
| 17:43.29 | cusco | hmm... I should |
| 17:43.37 | cusco | I will, not yet tho |
| 17:43.38 | cusco | thanks then |
| 17:43.45 | cusco | :-) |
| 17:44.10 | cusco | oh another question |
| 17:45.00 | cusco | we get calls from PRI, queuing... then we have a AEL that catches _XXX and sets some vars and dials the extention registered with that queue |
| 17:45.06 | cusco | all our extentions have 3 numbers |
| 17:45.09 | Qwell | cusco: pretty sure coppice was joking about LPC10 |
| 17:45.14 | Qwell | if not... *smack* |
| 17:45.23 | cusco | so it will always go trough the ael... |
| 17:45.30 | cusco | Qwell: noted |
| 17:46.00 | cusco | now... all the extensions that ... wow |
| 17:46.04 | cusco | nevermind it just hitme |
| 17:46.08 | cusco | :( |
| 17:46.09 | cusco | thanks |
| 17:48.32 | coppice | I was, but LPC10 is not as bad as most people think. most of the implementation people have heard are broken (e.g. the one in Asterisk). A correct one (e.g. spandsp) is not ridiculously bad |
| 17:48.55 | *** join/#asterisk lowtek (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 17:49.03 | Qwell | speak-n-spell ftw? |
| 17:49.29 | coppice | A correct implementation is nothing like speak and spell |
| 17:49.53 | Qwell | coppice: it used LPC10 |
| 17:50.05 | lowtek | Hey guys, is there a list somewhere of what zaptel is required for besides zap/dahdi hardware and a timing source for meetme? |
| 17:50.11 | coppice | no it didn't |
| 17:50.18 | Qwell | wikipedia says it did |
| 17:50.28 | Qwell | http://en.wikipedia.org/wiki/Speak_&_Spell_(toy)#Electronics |
| 17:51.07 | coppice | Qwell: it used 10th order LPC. that's quite different |
| 17:51.21 | Qwell | oh |
| 17:51.25 | Qwell | well then |
| 17:51.50 | [TK]D-Fender | lowtek: IAX2 Trunk Mode, MeetMe, and helps with MoH timing (but not required) |
| 17:51.59 | *** join/#asterisk doctorray (n=ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 17:52.06 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 17:52.18 | lowtek | TK: Thanks. Nothing to do with Local Channels? |
| 17:52.33 | [TK]D-Fender | lowtek: No |
| 17:52.45 | lowtek | TK: Thanks again! |
| 17:55.10 | tzafrir_laptop | lowtek, what version of asterisk is it? |
| 17:55.13 | Katty | yay, the other video feed is in high quality now |
| 17:55.44 | doctorray | my Asterisk has been randomly hanging up on people while they're leaving voicemail as of late, about 25% of the time, kind of annoying. Any idea on where to start troubleshooting? |
| 17:56.22 | doctorray | currently on 1.6.1.10. |
| 17:56.34 | tzafrir_laptop | coppice, so it should be possible to rewrite codec_lpc10 (or whatever this is called) using spandsp? |
| 17:56.46 | coppice | you could |
| 17:57.16 | *** join/#asterisk high-freq (n=hfreq@adsl-99-41-141-195.dsl.ksc2mo.sbcglobal.net) |
| 17:57.56 | *** join/#asterisk AeroCloud (n=aero@ip72-222-149-220.ph.ph.cox.net) |
| 17:58.21 | ManxPower-work | doctorray: remove callprogress=yes and busydetect=yes |
| 17:59.08 | doctorray | ManxPower-work: so it's a DAHDI thing? figured that out from a quick grep |
| 17:59.08 | *** join/#asterisk albertoandrade (n=albertoa@187.59.72.153) |
| 17:59.38 | ManxPower-work | doctorray: you didn't bother to mention if you are using SIP, DAHDI or carrier pidgen over IP. |
| 18:00.45 | doctorray | ManxPower-work: True. :) It was all happening on DAHDI channels. |
| 18:01.26 | ManxPower-work | callprogress=yes should be renamed randomlydisconnectmycalls=yes |
| 18:01.30 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 18:01.34 | doctorray | hah |
| 18:02.11 | doctorray | I'm moving to all SIP in the next few weeks anyway, so it shouldn't be an issue for me in the future |
| 18:02.20 | doctorray | wish I had asked months ago though |
| 18:02.53 | doctorray | better to change to callprogress=no or remove the line? |
| 18:05.07 | ManxPower-work | doctorray: it defaults to "no" if not set |
| 18:08.38 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 18:09.17 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 18:09.25 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:10.19 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 18:10.44 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 18:11.29 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 18:11.53 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 18:12.13 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 18:15.07 | doctorray | ManxPower-work: thanks for the help |
| 18:16.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:19.30 | *** join/#asterisk GameGamer43|Mac (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 18:20.04 | *** join/#asterisk helloritesh (n=hellorit@121.243.172.188) |
| 18:21.06 | helloritesh | [TK]D-Fender: If my channel name ${CHANNEL} is something like "Channel: SIP/12345-ed8f", could I originate a call directly to ${CHANNEL}? |
| 18:22.01 | [TK]D-Fender | helloritesh: No. |
| 18:22.22 | [TK]D-Fender | helloritesh: Originate / Call files "Channel" is the DEVICE to call, precisely like yuo would as with Dial() |
| 18:22.30 | helloritesh | So for MixMonitor, I need to trace the exact channel (mutliple parallel calls to be recorded) |
| 18:22.40 | helloritesh | ah.. |
| 18:22.40 | helloritesh | ok |
| 18:23.13 | helloritesh | no, monitoring a particular (active and bridged) CHANNEL would be interesting then.. |
| 18:23.18 | [TK]D-Fender | helloritesh: you pass the ${CHANNEL} as a VAR in the call-file so taht you can use it in calling ChanSpy <- |
| 18:23.27 | helloritesh | ChanSpy... |
| 18:23.28 | helloritesh | sure.. |
| 18:23.36 | helloritesh | I was trying with MixMonitor.. |
| 18:23.38 | [TK]D-Fender | helloritesh: You keep forgetting there are TWO ENDS to this |
| 18:23.40 | helloritesh | Sure.. That works. |
| 18:23.44 | helloritesh | yeah.. |
| 18:23.46 | *** part/#asterisk GameGamer43|Mac (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 18:23.53 | helloritesh | I got it.. I was just getting carried away with MixMonitor |
| 18:24.04 | [TK]D-Fender | helloritesh: I've told you about a dozen times now. |
| 18:24.09 | helloritesh | I have it all coded .. Just changin MixMonitor to ChanSpy |
| 18:25.50 | [TK]D-Fender | helloritesh: ...... you ......... need .............***BOTH*** |
| 18:26.08 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 18:26.19 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 18:27.08 | paulc | backs away slowly |
| 18:27.45 | p3nguin | Those apps do completely different things, so how do you swap it out like it is interchangeable? |
| 18:28.36 | FinboySlick | [TK]D-Fender is so intimidating. Second only to the guys in #bind ;) |
| 18:28.55 | [TK]D-Fender | will have to see about "removing" the competition |
| 18:28.56 | helloritesh | hehe.. no issues |
| 18:30.48 | FinboySlick | [TK]D-Fender: You help quite a bit though so I guess you're excusable. |
| 18:31.12 | [TK]D-Fender | FinboySlick: Providing 2 days of help on the same topic that should ahve gotten solved yesterday... I know... |
| 18:31.30 | *** join/#asterisk jdoe (i=jdoe@falseprophet.ca) |
| 18:32.17 | FinboySlick | [TK]D-Fender: Hehe, you bullied me into helping myself quite a few times. I'm just trying to be subtle about my gratefulness. |
| 18:32.54 | FinboySlick | [TK]D-Fender: Aren't you Canadian? |
| 18:33.14 | FinboySlick | probably meant gratitude. |
| 18:34.18 | [TK]D-Fender | finYup. |
| 18:34.22 | [TK]D-Fender | FinboySlick: Yup |
| 18:34.45 | FinboySlick | [TK]D-Fender: See? You're even serving your country. Dismantling the myth that we're all nice. |
| 18:35.11 | paulc | LOL nothing wrong with being Canadian (says the bloke with the British accent) (no one realises I was born here!) |
| 18:35.17 | p3nguin | haha |
| 18:35.35 | FinboySlick | paulc: Technically, we still belong to britain anyway. |
| 18:35.50 | paulc | Part of the commonwealth and all that.. minus the slaves? ;) |
| 18:36.25 | FinboySlick | paulc: Governor General having (unused) authority to strike down any law that the queen doesn't like, etc. |
| 18:37.06 | [TK]D-Fender | FinboySlick: "Belong"? Hell no. |
| 18:37.27 | FinboySlick | [TK]D-Fender: Not in practice, but in law, yes. |
| 18:38.12 | [TK]D-Fender | FinboySlick: We are a PART of the "Brittish Commonwealth". Which has about as much certainty attached as my already a poential winnar of the Publisher's Clearinghouse Sweepstakes |
| 18:38.26 | [TK]D-Fender | winner* |
| 18:40.02 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
| 18:41.39 | Katty | herroes. |
| 18:41.42 | Katty | hugs Ad-Hoc |
| 18:42.31 | Ad-Hoc | hug me hug me |
| 18:43.47 | *** join/#asterisk pfn (i=pfnguyen@socal.hanhuy.com) |
| 18:46.19 | *** join/#asterisk freepbxuser (n=staticfi@c-71-230-124-196.hsd1.pa.comcast.net) |
| 18:46.36 | freepbxuser | hi gus. |
| 18:46.38 | freepbxuser | guys^ |
| 18:47.16 | freepbxuser | i'm having big problems. wondering if someone could help debug |
| 18:47.31 | fenrus | try explaining your problem a bit more |
| 18:47.37 | [TK]D-Fender | Like.... at all |
| 18:47.38 | FinboySlick | freepbxuser: 'core set verbose 10' |
| 18:47.44 | freepbxuser | Okay |
| 18:47.57 | [TK]D-Fender | FinboySlick: conclusion[-1] |
| 18:48.06 | freepbxuser | I can recieve incoming calls when asterisk is in verbose mode, but not normal backgrund |
| 18:48.35 | [TK]D-Fender | freepbxuser: huh? |
| 18:49.16 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.8.223) |
| 18:49.36 | FinboySlick | I think he confuses verbose mode with no-fork mode |
| 18:49.41 | freepbxuser | i'm just looking at the error |
| 18:50.06 | [TK]D-Fender | freepbxuser: We aren't. Change that. |
| 18:50.08 | freepbxuser | when I run asterisk with "asterisk -vvvvvvvc" the call is transfered to my extention |
| 18:50.19 | [TK]D-Fender | freepbxuser: What call? |
| 18:50.21 | freepbxuser | Sorry, just pulling up the full log |
| 18:50.37 | [TK]D-Fender | freepbxuser: Screw logs. CLI output ONLY |
| 18:50.52 | FinboySlick | freepbxuser: My guess from what info you've provided: the user you run asterisk with isn't the same as the user your init script starts asterisk as. |
| 18:51.37 | freepbxuser | It happens right off the bat after reboot, or ran as root |
| 18:51.40 | freepbxuser | Channel 'SIP/2941671e0-001b2610' sent into invalid extension 'return' in context 'from-did-direct', but no invalid handler |
| 18:51.50 | freepbxuser | That's the error I get when run in the background |
| 18:51.55 | [TK]D-Fender | freepbxuser: Unrelated |
| 18:52.08 | freepbxuser | Hmm |
| 18:52.09 | [TK]D-Fender | freepbxuser: that is a clear dialplan error |
| 18:52.14 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:52.22 | freepbxuser | Sorry, I'm really new at this |
| 18:52.23 | [TK]D-Fender | Channel 'SIP/2941671e0-001b2610' sent into invalid extension 'return' in context 'from-did-direct', but no invalid handler <--- means exactly what it says |
| 18:52.34 | freepbxuser | I searched google extensively |
| 18:52.47 | freepbxuser | It connects fine the the verbose and 'c' flags though |
| 18:52.54 | [TK]D-Fender | freepbxuser: Search extensions.conf extensively and show us where you think it should match that |
| 18:53.05 | [TK]D-Fender | freepbxuser: where is the CALL for use to examine? |
| 18:53.30 | freepbxuser | I'm sorry, I don't understand |
| 18:53.43 | freepbxuser | It's a a sipgate trunk |
| 18:53.48 | freepbxuser | With one incoming rule |
| 18:54.21 | [TK]D-Fender | freepbxuser: Go to * CLI and show us the CALLS |
| 18:54.46 | freepbxuser | Umm right |
| 18:55.22 | freepbxuser | Paste the log from the CLI when a call is initiated? |
| 18:55.24 | freepbxuser | I'm sorry.... |
| 18:55.55 | [TK]D-Fender | freepbxuser: the entire output from beginning to end. Show us the GOOD call in one PASTEBIN, and a BAD on in another <- |
| 18:55.57 | [TK]D-Fender | ~pb |
| 18:55.58 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 18:56.02 | [TK]D-Fender | ^^^^^^^^^ |
| 18:56.17 | freepbxuser | OKies |
| 18:56.48 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:56.51 | freepbxuser | I'll do the good call first |
| 18:56.59 | freepbxuser | but for the bad call, I get very little info |
| 18:57.08 | freepbxuser | (in the log) |
| 18:57.18 | freepbxuser | When I run in verbose mode, everything is cool |
| 18:57.36 | [TK]D-Fender | freepbxuser: I said ASTERISK CLI OULY. Logs = worthless garbage |
| 18:57.42 | [TK]D-Fender | ONLY* |
| 18:59.51 | *** join/#asterisk rare1980 (n=rare@119.152.82.186) |
| 19:00.08 | freepbxuser | This is the good call |
| 19:00.09 | freepbxuser | http://pastebin.com/d772c052d |
| 19:01.03 | *** join/#asterisk BillyCrook (n=BillyCro@mars.advancedclustering.com) |
| 19:01.04 | *** join/#asterisk Greek-Boy (n=Monching@41.188.154.137) |
| 19:01.12 | freepbxuser | How do I get the CLI output of the application when in backgorund mode without launching it like asterisk -vvvvvvvc |
| 19:01.58 | FinboySlick | freepbxuser: run asterisk -r to connect to your background asterisk. |
| 19:02.06 | freepbxuser | Thanks |
| 19:02.44 | BillyCrook | can anyone recommend a voip provider who would host an asterisk vps (that I would have root on) so I could have maybe one sipphone at home, but design my own call schema, and forward to my cell reasonably fastly, and maybe even use ringback tones to tell people do not hang up, I am being contacted, or something like that? |
| 19:04.14 | freepbxuser | This is the bad output http://pastebin.com/d268cba14 |
| 19:05.13 | freepbxuser | Hmm, weird |
| 19:05.27 | freepbxuser | What's the difference between running it in -vvvvvc and in the background |
| 19:06.37 | p3nguin | You don't know the difference between background and foreground processes? |
| 19:07.34 | freepbxuser | No. Off to google I go. |
| 19:07.40 | [TK]D-Fender | freepbxuser: Whats the "probelm" with this 2nd call? |
| 19:07.57 | freepbxuser | I get no rings on my sip device |
| 19:08.01 | freepbxuser | It just doesn't connect |
| 19:08.11 | freepbxuser | It rings fine in the first example |
| 19:08.39 | [TK]D-Fender | == recordingcheck|20100127-113648|asterisk-1264621008.0: Failed to execute '/usr/share/asterisk/agi-bin/recordingcheck': Permission denied |
| 19:08.42 | [TK]D-Fender | this is your BAD call. |
| 19:08.49 | [TK]D-Fender | PERMISSION DENIED on that script |
| 19:08.56 | vader-- | asterisk 1.2 doesn't have snmp support right? |
| 19:09.05 | freepbxuser | Hmm |
| 19:09.11 | [TK]D-Fender | freepbxuser: Indeed * is running as the wrong user to access scripts on your system |
| 19:09.21 | [TK]D-Fender | freepbxuser: do "amportal chown" from Linux CLI |
| 19:09.41 | freepbxuser | I wonder why It's running as a different user when I run it with verbose mode in the foreground? |
| 19:09.52 | freepbxuser | Okay |
| 19:09.52 | [TK]D-Fender | freepbxuser: then restart * via amportal |
| 19:10.10 | [TK]D-Fender | freepbxuser: what USER are you when you ran it directly? |
| 19:10.18 | freepbxuser | I was root |
| 19:10.23 | freepbxuser | That makes sense |
| 19:10.27 | [TK]D-Fender | freepbxuser: well DUH root can do whatever it wants |
| 19:10.40 | [TK]D-Fender | freepbxuser: So run "amportal chown" and retest |
| 19:10.45 | freepbxuser | But, I was also root when I started it using amportal start |
| 19:11.03 | [TK]D-Fender | freepbxuser: yes, but amportal start tells it to run as ASTERiSK |
| 19:11.12 | freepbxuser | Goddit |
| 19:11.21 | freepbxuser | Thank you so much |
| 19:12.04 | freepbxuser | When running amportal chown, is says setting permissions |
| 19:12.16 | freepbxuser | When I usually start using ampportal start, it goes through the same process |
| 19:12.22 | freepbxuser | So why would they be off? |
| 19:13.24 | [TK]D-Fender | freepbxuser: is it working as it should now? |
| 19:13.40 | freepbxuser | Just testing |
| 19:14.04 | freepbxuser | Still getting permission denied errors |
| 19:15.03 | [TK]D-Fender | freepbxuser: Go to #freepbx for continued support. This is already outside of what is supported in here. |
| 19:15.13 | [TK]D-Fender | freepbxuser: they will advise you on the rest |
| 19:15.21 | freepbxuser | Okay |
| 19:15.24 | freepbxuser | I really appreciate it |
| 19:15.28 | freepbxuser | You've helped a lot |
| 19:16.11 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:21.20 | Skeeter- | iPad ONLY 499$ |
| 19:21.22 | Skeeter- | WOW!!! |
| 19:21.30 | *** join/#asterisk QubeZ (n=qube@64.128.254.34) |
| 19:21.32 | QubeZ | hello all |
| 19:21.34 | thehar | $629 if you want 3g |
| 19:21.47 | QubeZ | once the voicemail password is setup in the cfg file and the user changes, where is it then stored? |
| 19:21.53 | Skeeter- | So $499 for 16GB of iPad. ThatÂ’s our base model. 32GB is $599, 64GB is $799. 3G models cost an extra $130. $629, 729, and 829 with 3G |
| 19:22.42 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:22.53 | leifmadsen | yuck |
| 19:23.14 | leifmadsen | and inability to update it via external memory I presume |
| 19:24.05 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 19:24.08 | [TK]D-Fender | leifmadsen: Of course!@ |
| 19:25.49 | *** join/#asterisk Znuff (n=ibm86@89.165.131.103) |
| 19:26.34 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:27.28 | freepbxuser | And many thanks to you, [TK]D-Fender |
| 19:27.42 | leifmadsen | don't worry, they'll sell you another one in a couple of years when they fix all the problems you find in the first version |
| 19:27.54 | leifmadsen | iWantYourMoney |
| 19:28.55 | *** part/#asterisk freepbxuser (n=staticfi@c-71-230-124-196.hsd1.pa.comcast.net) |
| 19:29.00 | ManxPower-work | QubeZ: It's stored in the config file. |
| 19:31.02 | *** join/#asterisk daidoji (n=daidoji7@99.48.50.198) |
| 19:31.21 | QubeZ | ManxPower-work so when the user changes it, Asterisk updates the config file? |
| 19:31.34 | ManxPower-work | QubeZ: yes |
| 19:33.39 | QubeZ | ManxPower-work guess it didnt, i just changed my pass and the voicemail.cfg is still 1234 |
| 19:34.01 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 19:34.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:36.36 | *** join/#asterisk RobH__ (n=robh@rob.tech.wikimedia.org) |
| 19:40.30 | *** join/#asterisk toppe (n=root@cs181001225.pp.htv.fi) |
| 19:40.43 | ManxPower-work | maybe you have a permission problem. Or maybe you are the one person in the universe where it doesn't work. |
| 19:41.33 | QubeZ | lol, well the new password works but its just not stored in the voicemail.cfg |
| 19:42.49 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:46.41 | Naikrovek | maybe when you reload asterisk it'll get written |
| 19:46.57 | ManxPower-work | QubeZ: are you using some kind of Asterisk GUI? |
| 19:47.15 | Naikrovek | i've written apps that hold changes in memory for a time, then write them to disk every so often. during shut down or reload i also flush the preferences to disk |
| 19:47.20 | ManxPower-work | and it's voicemail.conf not voicemail.cfg |
| 19:53.55 | QubeZ | ManxPower-work no gui, ya sorry voicemail.conf |
| 19:54.25 | ManxPower-work | sounds like you are running something like Realtime |
| 19:55.43 | *** join/#asterisk fizz0029_ (n=fizz0029@96.245.72.14) |
| 19:56.25 | fizz0029_ | Hi All. Have a Q. i want to add the ability to let callers dial a second extension during my voicemail message. is that a PBX functionality or asterisk? |
| 19:56.46 | p3nguin | fizz0029_: yes |
| 19:57.35 | p3nguin | fizz0029_: Have a look at exitcontext= in voicemail.conf |
| 19:57.53 | fizz0029_ | ok thank you |
| 20:05.54 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 20:12.40 | steliosk | Anyone got any info on the pricing (per port and server/site licence) for at&t natural voices ? |
| 20:14.21 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 20:20.04 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:20.39 | Skeeter- | How to bill every call made by Asterisk? |
| 20:21.05 | bmoraca_work | analyze CDRs and bill? |
| 20:21.30 | Chainsaw | If you don't feel like writing the software for it in-house, take a look at something like a2billing. |
| 20:23.39 | *** join/#asterisk smooth_penguin (n=smoove@59.95.6.22) |
| 20:24.26 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.26.249) |
| 20:31.17 | voipmonk | how many asterisk systems, Skeeter- ? |
| 20:31.25 | voipmonk | and how many simultaneous calls? |
| 20:40.08 | Skeeter- | voipmonk, 1 asterisk |
| 20:40.28 | Skeeter- | 30+ simul. calls |
| 20:40.39 | Skeeter- | voipmonk, how does that influence anything |
| 20:41.51 | voipmonk | if you said 3000 it would influence it a bit |
| 20:42.00 | voipmonk | if u said 500 it would influence it as well |
| 20:42.10 | voipmonk | but 30 u can do on one system with no problem |
| 20:42.31 | Skeeter- | and how would that be accomplish |
| 20:42.47 | voipmonk | what else do you need? |
| 20:42.49 | voipmonk | do u need LCR? |
| 20:42.55 | voipmonk | customer interface? |
| 20:43.06 | voipmonk | portal for members to check accounts , cdrs, and pay you? |
| 20:44.11 | voipmonk | do you want to track accounts , destinations, minutes used, and cost... |
| 20:44.17 | Skeeter- | ya |
| 20:44.31 | FinboySlick | is 'emailbody' the only thing one can use to format emails in when using imap storage? Or is there a way to have more control on the message content? |
| 20:44.45 | Skeeter- | reports of every call made by all exten. by which trunk, etc |
| 20:45.10 | voipmonk | Skeeter-: a2billing is free -, jbilling is about $50 for the documentation but is free to install.... if u need something custom - you're going to need some time for development |
| 20:45.18 | FinboySlick | Sort of like what's achieved with mailcmd. |
| 20:45.42 | voipmonk | FinboySlick: look inside voicemail.conf |
| 20:45.54 | voipmonk | and the source for the voicemail app |
| 20:45.56 | Skeeter- | i lookedup a2billing, but it seems like a pain to configure |
| 20:45.59 | voipmonk | there's all kinds of functions in there |
| 20:46.12 | voipmonk | it takes about 15 minutes to configure if u have rates and know what you want |
| 20:47.52 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 20:48.09 | Skeeter- | voipmonk, they customer will bill long distance calls, thats it. But i want to have a full reports of all calls made to have some statistic |
| 20:50.01 | *** join/#asterisk opticy (i=ycitpo@that.violates.us) |
| 20:51.00 | opticy | is there a way to detect if an outbound POTS line is currently in use ? |
| 20:55.18 | Poincare | hmmm |
| 20:55.37 | Poincare | my provider claims that IAX is totaly insecure and a risk for all of us |
| 20:56.17 | Poincare | it says so "on the asterisk site" claims... |
| 20:56.42 | russellb | Poincare: they may be referring to a previous security advisory that was fixed many releases ago now. |
| 20:56.43 | Poincare | Might it be that I'm looking at the wrong asterisk site? Don't realy find scary things there |
| 20:57.31 | Poincare | Well, that was the answer I got when I asked why we got so many time-outs on one of their servers... |
| 20:57.56 | russellb | maybe they are using a really old version? |
| 20:58.24 | Poincare | any way to verify that? |
| 20:58.29 | russellb | http://lists.digium.com/pipermail/asterisk-dev/2009-September/039601.html ..... http://lists.digium.com/pipermail/asterisk-announce/2009-September/000205.html |
| 20:58.30 | russellb | you could ask, heh |
| 20:58.58 | Poincare | hmmm, asking technical information from a telco-helpdesker... |
| 20:59.08 | Poincare | "how to ruin your day 101" |
| 20:59.39 | FinboySlick | voipmonk: Well, I looked at all of voicemail.conf but I hadn't considered the need to dig through source yet. I should assume that the features I'm looking for aren't documented? |
| 20:59.42 | russellb | anyway, they're full of it, based on that initial comment. |
| 21:00.03 | thehar | russellb: i linked to you on my blog |
| 21:00.03 | thehar | hehe |
| 21:00.29 | russellb | orly! neat. |
| 21:00.33 | thehar | my play blog |
| 21:00.37 | russellb | hawt |
| 21:00.40 | thehar | mmhm |
| 21:00.46 | thehar | that i'm testing out on our new prouct |
| 21:00.49 | thehar | wanna seeeeeeee |
| 21:00.51 | thehar | stackable.com |
| 21:01.04 | Poincare | so he wants me to use 'the SIP industrial standard' because it's debugable and more secure |
| 21:01.36 | russellb | Poincare: "more secure" is actually completely false. |
| 21:01.51 | Poincare | oh, maybe he meant 'more stable' :-) |
| 21:02.52 | Poincare | i still don't understand why the many disconnects are related to his customers using old versions |
| 21:03.10 | Poincare | he can't upgrade untill all his customers use 'a recent iax version' |
| 21:06.33 | *** join/#asterisk catojo (n=catojo@189.24.24.35) |
| 21:07.37 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 21:08.31 | Poincare | won't an IAX2 from version 1.x.99 cummunicate properly with a system using IAX2 1.1.1? |
| 21:08.32 | *** join/#asterisk maximCH (n=maximCH@adsl-ecom-4-c15-p038.vtx.ch) |
| 21:10.08 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
| 21:12.57 | russellb | those versions mean nothing to me |
| 21:14.33 | Poincare | russellb: that are hypothetical versions |
| 21:14.51 | russellb | yes, all versions are compatible |
| 21:15.03 | russellb | and the configuration allows disabling security features and setting resource limits per account |
| 21:15.09 | russellb | we designed all of it with service providers in mind |
| 21:16.04 | Poincare | actually our operator is telling me that the disconnects I experience to HIS servers are related to him using an older version of IAX because he can't upgrade his IAX version as long as his customers don't upgrade |
| 21:17.25 | Poincare | russellb: you say 'we', you got an active part in IAX development? :-) |
| 21:18.32 | bmoraca_work | Poincare: he didn't get to be an op in this room by being cute, that's for sure |
| 21:18.50 | Poincare | he, didn't notice the op :-) |
| 21:18.53 | russellb | Poincare: I'm the engineering manger for open source development at Digium. |
| 21:18.54 | *** join/#asterisk christian_lappy (n=christia@pD9E33080.dip.t-dialin.net) |
| 21:18.59 | Poincare | cool :-) |
| 21:19.02 | ManxPower-work | Poincare: Russel is the maintainer forAsterisk 1.2, maybe 1,4 as well |
| 21:19.09 | seanbright | who is russel? |
| 21:19.16 | seanbright | trolls |
| 21:19.35 | Poincare | so, I'll give you our operator's number and I'll have him explain to you why I should migrate from IAX to SIP? :-) |
| 21:19.42 | christian_lappy | guys,please don't throw stones at me...but... |
| 21:19.58 | christian_lappy | is asterisk as powerfull as cisco callmanager ? |
| 21:20.12 | russellb | Poincare: heh, I'd rather not. |
| 21:20.14 | seanbright | no |
| 21:20.18 | seanbright | it is not |
| 21:20.24 | russellb | migrating from IAX to SIP is not necessarily a bad idea. |
| 21:20.25 | bmoraca_work | christian_lappy: depends on what you want it to do...but i'd say that in general it is not |
| 21:20.31 | russellb | but what they told you so far is rubbish for reasoning. |
| 21:20.53 | Poincare | russellb: that's what I guessed... just wanted some confirmation |
| 21:21.21 | christian_lappy | bmoraca_work: mhhh...ok |
| 21:21.29 | russellb | if I were you, I would avoid the most likely unuseful technical detail battle and just migrate to whatever it is they are willing to support |
| 21:21.53 | russellb | (or change providers if you're unhappy with their service for other reasons) |
| 21:22.09 | paulc | Ok, back from lunch - I want something exciting to work on.. is that too much to ask? |
| 21:22.39 | bmoraca_work | christian_lappy: Asterisk has the benefit of being free...whereas CallManager has the benefit of backing from what was once the most valuable company in the world. callmanager has some features that are really cool and most can be developed for asterisk...but as far as pure call throughput, CallManager is superior |
| 21:23.22 | christian_lappy | is there something comparbale to srst availible ? |
| 21:23.46 | [TK]D-Fender | christian_lappy: you have failed to specify your exact goals |
| 21:24.09 | christian_lappy | [TK]D-Fender: that isn't so easy yet |
| 21:24.11 | [TK]D-Fender | christian_lappy: For instance CCM makes a very poor paddle-boat |
| 21:24.32 | [TK]D-Fender | christian_lappy: How can we tell you whats better for your goals if you can't even say what you want? |
| 21:24.32 | christian_lappy | [TK]D-Fender: basically voip+dec integration |
| 21:24.40 | [TK]D-Fender | christian_lappy: "dec"? |
| 21:24.54 | christian_lappy | [TK]D-Fender: dect |
| 21:24.57 | [TK]D-Fender | christian_lappy: and what does "integration" imply? |
| 21:25.10 | bmoraca_work | christian_lappy: not with asterisk alone, no. i believe there might be some telephony hardware vendors that might support something similar...then again, it depends on how much you want to spend on infrastructure, because I could probably implement something very similar in Asterisk |
| 21:26.50 | [TK]D-Fender | checkout time, BBL |
| 21:26.51 | Poincare | russellb: don't tell me... complaining to a telco is like cutting al your communication lines |
| 21:27.25 | christian_lappy | bmoraca_work: working for digium ? |
| 21:30.48 | bmoraca_work | christian_lappy: if all you're trying to do is maintain connectivity to a remote site when their WAN link goes down (basic SRST), that can be configured in Asterisk |
| 21:31.12 | christian_lappy | bmoraca_work: yep, thats what i meant :-) |
| 21:31.17 | bmoraca_work | christian_lappy: if you want to be able to not only call between sites but also call the PSTN with all WAN and power connections down, that's a bit tougher but still potentially possible using various hardware |
| 21:31.51 | christian_lappy | power up, wan line down |
| 21:32.19 | christian_lappy | so, if wan is down it still should be ablw to use the pstn at thw remote site |
| 21:33.05 | bmoraca_work | christian_lappy: depending on your architecture, if the wan link is down, all you would need is point-to-point connections between the sites |
| 21:33.35 | bmoraca_work | if all sites have their own PSTN gateway and origination/termination, then you really don't much need to worry about it |
| 21:34.10 | christian_lappy | yup, pstn (pri) on all sites |
| 21:35.12 | bmoraca_work | all you have to do then is have some basic number manipulation for your local extension dials if the primary IP link between PBXes is not available...that's barely SRST :P |
| 21:35.26 | vader-- | hey bmoraca ever get a chance to play with that adtran unit? |
| 21:35.46 | bmoraca_work | vader--: i haven't yet, no. give me a few minutes and i'll get something hooked up real quick |
| 21:35.53 | vader-- | cool thanks man |
| 21:36.17 | christian_lappy | bmoraca_work: what else is srst then ? |
| 21:36.42 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-252-235.telstraclear.net) |
| 21:37.13 | christian_lappy | bmoraca_work: but ib your setup a asterisk is placed at every remote site ? |
| 21:37.52 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-252-235.telstraclear.net) |
| 21:38.27 | christian_lappy | bmoraca_work: i only know the "cisco setup way" :-( |
| 21:40.12 | bmoraca_work | christian_lappy: that's the only way to do it with Asterisk. with the cisco way, call control is handled at the main location with offload to CME routers at remote sites if WAN connectivity is lost. that's not really possible here...but if you're terminating/originating each of the remote sites locally to that site, you need (well, not necessarily, but it helps) a PBX at each spot anyway |
| 21:40.37 | bmoraca_work | i need to clean up my work bench...my stack of routers and tangle of cables is a bit unruly right now |
| 21:40.50 | christian_lappy | lol |
| 21:41.55 | fenrus | bmoraca_work, you can send any left-over-routers to me :) |
| 21:42.03 | *** join/#asterisk hudys (n=hudys1@62.148.88.41) |
| 21:42.20 | bmoraca_work | 4 cisco routers (including a 7204), 4 cisco switches, a PIX 515, an ASA5505, and two Adtran TAs...plus a barracuda web filter and 4 phones...ick |
| 21:42.29 | christian_lappy | if cisco ones, i could use them also :-) |
| 21:42.35 | fenrus | hoho, the 7200 is a beast ;) |
| 21:42.55 | christian_lappy | yeah, powers the 65xx series also :-) |
| 21:42.55 | fenrus | npe-g1 ? |
| 21:42.58 | ManxPower-work | hugs his Catalyst 5505 |
| 21:43.07 | fenrus | catos *shrugs* |
| 21:43.13 | christian_lappy | catalyst 55xx ? |
| 21:43.19 | christian_lappy | never heard of |
| 21:43.25 | *** part/#asterisk hudys (n=hudys1@62.148.88.41) |
| 21:43.25 | fenrus | google it :) |
| 21:43.31 | christian_lappy | only 4xxx and 6xxx(which wee use) |
| 21:43.37 | christian_lappy | fenrus: :) |
| 21:43.50 | fenrus | L3-switch with CatOS.. ;) |
| 21:44.22 | christian_lappy | catos..ouch.thats ages ago :-) |
| 21:44.31 | bmoraca_work | fenrus: nope...think it's an NPE-150 |
| 21:44.32 | ManxPower-work | The 55xx series are old (and very cheap) switches. Was top of the line for it's time, and still does a great job if you don't expect too much from it's featureset. |
| 21:44.34 | fenrus | yea, and it's dirty too. |
| 21:44.36 | christian_lappy | ditched that about 5years ago or so |
| 21:45.03 | christian_lappy | bmoraca_work: so your experienced with asa also ? |
| 21:45.03 | fenrus | a really port-intensive switch |
| 21:45.15 | fenrus | asa > pix.. |
| 21:45.36 | *** join/#asterisk frocostrocos (n=IceChat7@80.172.228.33) |
| 21:45.45 | christian_lappy | fenrus: i know :) |
| 21:46.07 | christian_lappy | fenrus: but that was on hist list :-) |
| 21:46.17 | fenrus | (: |
| 21:46.20 | *** part/#asterisk frocostrocos (n=IceChat7@80.172.228.33) |
| 21:46.23 | christian_lappy | ha.so.here we go.. |
| 21:46.43 | bmoraca_work | christian_lappy: yes, i am |
| 21:47.04 | christian_lappy | does on of you guys know wether a ststi route has a higher prio then a vpn "learned" one bmoraca_work ? |
| 21:47.08 | Naikrovek | has an ASA and loves it |
| 21:47.13 | christian_lappy | too |
| 21:47.16 | Naikrovek | static? |
| 21:47.24 | christian_lappy | yup |
| 21:47.35 | christian_lappy | static>reverse route injection ? |
| 21:47.38 | Naikrovek | you may be better asking this in #cisco |
| 21:47.44 | fenrus | christian_lappy, how is it learned from the vpn, via some routing protocol? |
| 21:47.47 | Naikrovek | but i think the static route is preferred, yes |
| 21:47.58 | christian_lappy | fenrus: reverse route injection |
| 21:48.04 | fenrus | ah, okay. |
| 21:48.31 | christian_lappy | fenrus: i want t backup a mpls line with a vpn tunnel |
| 21:49.06 | christian_lappy | fenrus: so, either do it with ospf or with a tracked route + vpn |
| 21:49.21 | fenrus | i'd go with bgp :) |
| 21:49.29 | christian_lappy | not with an asa :P |
| 21:49.31 | Naikrovek | well the ASA can do ISP failover easily |
| 21:49.50 | Naikrovek | ah, not with an ASA |
| 21:49.51 | fenrus | i'd throw out the asa, and use some real router :) |
| 21:50.03 | Naikrovek | that's ignorance talkin' |
| 21:50.06 | Naikrovek | asa's rule |
| 21:50.17 | christian_lappy | and they are pretty cheap |
| 21:50.17 | Naikrovek | if you know how to configure them correctly |
| 21:50.18 | fenrus | well, i'd skip the vpn part at least |
| 21:50.28 | Naikrovek | why skip vpn |
| 21:50.41 | Naikrovek | my asa has had a point to point vpn to my india office for 9 months straight |
| 21:50.48 | christian_lappy | works fine |
| 21:51.03 | Naikrovek | if it get disconnected, it automatically reconnects and everything |
| 21:51.08 | Naikrovek | eh |
| 21:51.11 | Naikrovek | headache |
| 21:51.12 | christian_lappy | still have some issueswith a asa<->checkpoint ng though :-( |
| 21:51.13 | Naikrovek | goes back to lurking |
| 21:51.28 | fenrus | if you order some site to site vpn from an ISP that delivers with mpls it's already "safe" from the "internet" |
| 21:51.43 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:52.33 | bmoraca_work | vader--: here's my setup: (SIP PSTN GATEWAY) <=> Adtran 908e <=> (PRI) <=> Adtran 904 <=> (SIP TO ASTERISK)...the 904 is in the place where your 924 would be, and if this works, your application will be fine |
| 21:53.09 | christian_lappy | bmoraca_work: is there someclikc to call features for asterisk ? |
| 21:53.31 | christian_lappy | click to call |
| 21:54.11 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 21:54.28 | bmoraca_work | christian_lappy: not built in, but there are various third-party applications which use AMI to perform that function (snap-a-number for outlook integration, isymphony for operator console) |
| 21:54.50 | christian_lappy | ahhh |
| 21:55.28 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 21:55.50 | [TK]D-Fender | or you could write your own in about 5 minutes |
| 21:56.20 | fenrus | is there some good open source operator console ? |
| 21:57.03 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 21:57.31 | bmoraca_work | fenrus: no. isymphony is the best, and it's free up to 5 users (limited functionality). |
| 21:57.42 | bmoraca_work | is in no way associated with i9 technologies |
| 21:58.28 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 21:58.40 | [TK]D-Fender | bmoraca_work uses Asterisk. iSymphony connects to asterisk. iSymphony is owned by i9 Technologies Association found. |
| 21:58.55 | [TK]D-Fender | bmoraca_work: And I didn't even need Kevin Bacon! |
| 21:59.03 | christian_lappy | thanks for the infos guys..ill be back tomorrow |
| 21:59.10 | bmoraca_work | yeah yeah...should have said "affiliated" |
| 22:00.37 | *** join/#asterisk alberthfmn (n=alberthf@189.61.66.14) |
| 22:00.52 | *** part/#asterisk christian_lappy (n=christia@pD9E33080.dip.t-dialin.net) |
| 22:00.55 | *** join/#asterisk pietro (n=pietro@88-149-224-77.dynamic.ngi.it) |
| 22:01.02 | pietro | hi |
| 22:01.07 | alberthfmn | g'night |
| 22:01.24 | Naikrovek | ot: anyone in here have any railroad experience |
| 22:01.37 | alberthfmn | i've 2 |
| 22:01.58 | pietro | is possibile using a different (to peer name) username for digest autentication in a sip peer ? |
| 22:02.00 | vader-- | bmoraca any fxs stuff in place? |
| 22:02.15 | Naikrovek | i wonder how much a fully loaded freight train weighs |
| 22:02.18 | [TK]D-Fender | Naikrovek: I've had a Monopoly on them..... |
| 22:02.25 | Naikrovek | hehe |
| 22:03.00 | alberthfmn | i've 2 dvg-5004s with 4 FXS ports, i've an asterisk server too. what can i do with this? |
| 22:03.45 | alberthfmn | dvg supports SIP. and its already logged in asterisk server. |
| 22:03.49 | [TK]D-Fender | alberthfmn: I dunno... maybe let the DVG's pass calls to SIP peer entries you create in sip.conf. That's a novel idea |
| 22:04.04 | alberthfmn | yah, this is already done. |
| 22:04.07 | alberthfmn | but |
| 22:04.54 | alberthfmn | i want somethin' like... from my pbx(non-ast), wanna call to the other DVG |
| 22:05.15 | alberthfmn | sip.conf already done. |
| 22:05.48 | alberthfmn | the pbx has 4 fxo ports... |
| 22:06.13 | [TK]D-Fender | alberthfmn: Everything is dialplan <- |
| 22:06.45 | alberthfmn | uhm... |
| 22:06.56 | alberthfmn | agree.. haha |
| 22:06.57 | *** join/#asterisk p3nguin (i=gpz5GvdF@9xf6gbZ.a2infotech.com) |
| 22:07.14 | simplydrew | Is there a way to configure asterisk to keep dialing a number that is busy, and then ring the extension you're calling from when the line you're dialing is free? Kind of like what interrupts the busy signal on most landlines and says "This line is busy...we will continue to dial the number for you, for the next 20 minutes, blah blah"? |
| 22:07.34 | [TK]D-Fender | simplydrew: Also only dialplan. |
| 22:07.45 | Naikrovek | yeah that can be done |
| 22:07.48 | simplydrew | Ah okay. How would that be done in the dialplan? |
| 22:07.52 | alberthfmn | simplydrew, yeah, only dialplan |
| 22:08.00 | alberthfmn | simplydrew, draw |
| 22:08.21 | simplydrew | draw? |
| 22:08.39 | alberthfmn | yeah |
| 22:08.46 | alberthfmn | it can be done. |
| 22:08.52 | alberthfmn | easily |
| 22:08.56 | [TK]D-Fender | simplydrew: Originate a Local channel extension. Call target #. If not answered, wait. Check how many times you've called so far. If to many, stop. If not increment counter. Loop & call again. |
| 22:09.10 | [TK]D-Fender | simplydrew: On the other end point to exten that calls your phone |
| 22:09.37 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 22:09.51 | simplydrew | Oh nice. Alright, just wanted to check |
| 22:10.25 | alberthfmn | nice, fender. |
| 22:10.30 | alberthfmn | do u play guitar? ahah |
| 22:11.15 | alberthfmn | uhm... |
| 22:12.11 | *** part/#asterisk FinboySlick (n=shark@74.117.40.10) |
| 22:15.43 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:17.19 | p3nguin | Oh, the snow is really flying now. |
| 22:25.42 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 22:26.26 | salparadise | so noob here, I am trying to create a .call file and place a call file looks like this: http://dev.inetpro.org/pastebin/54 |
| 22:26.38 | salparadise | but all I ever get is this [Jan 26 09:39:53] NOTICE[23009] pbx_spool.c: Call failed to go through, reason (3) Remote end Ringing |
| 22:26.39 | [TK]D-Fender | alberthfmn: Yes, amongst several other instruments, but my nic has nothing to do with music |
| 22:27.01 | Linuturk | salparadise: are you moving or copying the call file to the spool? |
| 22:27.24 | salparadise | Linuturk: mv |
| 22:28.10 | [TK]D-Fender | salparadise: exten => _XXXXXXXXXX,100,Dial(SIP/l3vector/81${EXTEN},10,gS(1)) <- you have no priority 1 |
| 22:28.41 | *** part/#asterisk BillyCrook (n=BillyCro@mars.advancedclustering.com) |
| 22:28.53 | salparadise | [TK]D-Fender: hmm ok, where does that go exactly? |
| 22:28.57 | [TK]D-Fender | salparadise: Also, your 1st channel does nothing except answer (triggering the exten part), and then dies in 15 seconds tops |
| 22:29.11 | [TK]D-Fender | salparadise: ... this is extensions.conf basics.. you have no STEP 1 |
| 22:29.16 | [TK]D-Fender | salparadise: Chapter 5 <- |
| 22:29.17 | [TK]D-Fender | ~book |
| 22:29.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:30.01 | salparadise | [TK]D-Fender: ok, thanks. Will read the docs. |
| 22:30.58 | alberthfmn | [TK]D-Fender, i c. |
| 22:31.11 | alberthfmn | nor the pkts till here? |
| 22:31.20 | alberthfmn | ok. |
| 22:31.38 | alberthfmn | freud. |
| 22:36.59 | bmoraca_work | vader--: success. it worked just as I thought it would. |
| 22:37.07 | *** join/#asterisk timholum (n=chatzill@64-91-67-5.stat.centurytel.net) |
| 22:37.41 | *** join/#asterisk Skeeter- (n=Skeeter@c66.203.215-71.clta.globetrotter.net) |
| 22:37.55 | Skeeter- | The iPad will be so great with asterisk |
| 22:39.07 | Skeeter- | 30$ for unlimited data plan |
| 22:39.28 | Skeeter- | plug that with a softphone for the appstore et voila |
| 22:40.55 | *** join/#asterisk obsd123 (n=obsd123@ip70-179-167-175.fv.ks.cox.net) |
| 22:41.23 | obsd123 | I purchased a g729 license for testing, but all of my outgoing calls are still using the ulaw codec |
| 22:41.37 | obsd123 | I disallowed ulaw in sip.conf, along with everything else except for g729 |
| 22:41.43 | obsd123 | what am I missing? |
| 22:42.01 | cusco | reload ? |
| 22:44.58 | obsd123 | I've reloaded |
| 22:45.14 | cusco | what does cli show? |
| 22:45.19 | cusco | paste configs as well |
| 22:45.28 | obsd123 | okay, will do |
| 22:46.21 | *** join/#asterisk catojo (n=catojo@189.24.24.35) |
| 22:51.25 | [TK]D-Fender | Skeeter-: Any device with a plan will do the same. iPag however will NOT fit into your pocket, costs a lot more, etc |
| 22:54.11 | timholum | does anyone know of a way to play music through the paging system?, I would like to play music through the onboard soundcard, then when I page it pauses the music and does the page |
| 22:54.39 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:54.43 | bmoraca_work | timholum: what kind of paging system? what music source? |
| 22:54.55 | timholum | I was going to use mpg123 |
| 22:55.17 | timholum | the paging is done through asterisk, |
| 22:55.24 | bmoraca_work | let me repeat: what kind of paging system, what is the music source? |
| 22:56.06 | timholum | Music, Preferably mpg123, Paging, Dial an extention Dial(console/dsp) |
| 22:56.13 | Skeeter- | u cant just have a data plan on an iphone cant u? |
| 22:56.20 | timholum | I am up for eather alsa or oss |
| 22:56.26 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 22:57.10 | [TK]D-Fender | Skeeter-:Sure. |
| 22:57.18 | *** join/#asterisk ChrisWi (n=admin@mx2.wwserver.net) |
| 22:57.31 | Skeeter- | i wish i could live in the US |
| 22:58.02 | bmoraca_work | timholum: you're not answering my questions. but, FYI, your paging system itself will need to do this and it will ahve to support two inputs. ASTERISK is not your paging system...it is but one source of audio for your paging system. ASTERISK cannot easily jump back and forth between audio sources, thus having two audio sources would be preferable. |
| 22:58.05 | vader-- | so bmoraca asterisk was able to utilize both the pri and fxs ports on the adtran unit? |
| 22:58.13 | [TK]D-Fender | Skeeter-: thats what becoming an ex-pat is all about |
| 22:58.15 | vader-- | can asterisk bridge calls from the pri to the fxs? |
| 22:58.21 | vader-- | i.e. a fax coming in |
| 22:59.16 | [TK]D-Fender | timholum: Plenty of easy options |
| 22:59.28 | [TK]D-Fender | TimAnd no you don't need specialized hardware |
| 22:59.42 | Skeeter- | [TK]D-Fender, not sure wat u mean there |
| 23:00.00 | [TK]D-Fender | Skeeter-: Emmigrate. |
| 23:00.15 | b14ck | http://www.nooooooooooooooo.com/ |
| 23:00.23 | Skeeter- | nned visa and a job... |
| 23:00.24 | timholum | bmoraca_work: I have speekers plugged into my asterisk boxes sound card. asterisk dials console/dsp the page comes accrost. that is my paging system :) |
| 23:00.27 | *** join/#asterisk ktwilight (n=keliew@123.55-240-81.adsl-dyn.isp.belgacom.be) |
| 23:01.16 | voipmonk | ciscos with autoanswer and call files seems kinda old school now a days |
| 23:01.39 | voipmonk | but u sound like a god with 50 phones in the same room |
| 23:01.48 | voipmonk | ... i was bored... |
| 23:02.21 | voipmonk | then with voicechangedialer.... well... ill go back under my rock.. |
| 23:02.57 | jaytee | he gets a rock? I'm so jealous! |
| 23:04.47 | bmoraca_work | timholum: you're probably not going to be able to do it then. you might be able to get away with an AGI script that searches for your MPG123 instance and kills it before paging and then starts it up again, but that's hokey as hell. |
| 23:05.39 | timholum | ok. :( I was hoping there was an easy way to do it |
| 23:06.00 | [TK]D-Fender | bmoraca_work: Use any player app with CLI control. Tell it to mute at start, unmute at end. Easy. |
| 23:06.07 | [TK]D-Fender | bmoraca_work: And completely worth it |
| 23:06.12 | Kobaz | what's the pci interupt threshold for a t1 card dropping frames... i'm seeing about 500 interupts/sec and i get spikes up to 580 |
| 23:06.58 | bmoraca_work | [TK]D-Fender: can asterisk access the sound card device at the same time as a music player app? none of the systems i've used asterisk on have sound cards so it's never been an issue for me |
| 23:08.26 | [TK]D-Fender | bmoraca_work: between Chan OSS & ALSA, there are plenty of OTHER local options as well.... lets see what else you can think of... |
| 23:09.16 | bmoraca_work | [TK]D-Fender: this isn't for me...i was giving ideas to someone else. like i said, i've never needed to do local audio on an asterisk box, so i've got 0 experience with that |
| 23:10.02 | [TK]D-Fender | bmoraca_work: I[m just testing your imagination... |
| 23:10.39 | *** join/#asterisk Chinorro (n=Chino@202.219.27.77.dynamic.mundo-r.com) |
| 23:11.02 | bmoraca_work | if it were me, i'd use a real paging system and a cheap MP3 player or CD player so that the customer could modify the music at will, and then interface with that with a SIP-based device from Cyberdata |
| 23:12.29 | vader-- | so bmoraca asterisk was able to utilize both the pri and fxs ports on the adtran unit? |
| 23:12.30 | vader-- | can asterisk bridge calls from the pri to the fxs? |
| 23:12.33 | vader-- | i.e. a fax coming in |
| 23:12.42 | [TK]D-Fender | bmoraca_work: Or you can run a audio player of your choice on your server alongside a local soft-phone on -auto-answer <- |
| 23:13.28 | Corydon76-dig | vader--: yes, but if you have a TDM400 or equivalent (as opposed to a TDM410 or equivalent), then faxes may not be reliable |
| 23:13.35 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:14.01 | bmoraca_work | vader--: i didn't test the FXS at this particular time, but i know it to work based on previous experience (I use them quite often) |
| 23:14.07 | vader-- | corydon sorry bmoraca was testing a adtran device for me |
| 23:14.18 | Corydon76-dig | Ah |
| 23:14.26 | vader-- | bmoraca i can't get ahold of one sales person at adtran to talk about getting a demo unit |
| 23:14.33 | vader-- | i keep getting voicemails or out of office |
| 23:14.51 | bmoraca_work | vader--: like i said, i don't know if they'd be willing to send one out to someone who's not a reseller |
| 23:15.03 | Corydon76-dig | vader--: when did you start calling? |
| 23:15.21 | bmoraca_work | vader--: also, i'm not familiar with their eastern-division sales people, either...i've only ever dealt with the western guys |
| 23:15.51 | Corydon76-dig | Right now it's after 5, Adtran time, and I suspect after 4, you'll starting having trouble getting ahold of ANY salespeople |
| 23:16.52 | Corydon76-dig | Tech support is a different matter |
| 23:18.14 | bmoraca_work | vader--: the way the devices work is pretty cool...you set up a SIP "trunk" (their words, not mine) to your server and you just point the FXS port at the SIP trunk and give it its own credentials and it registers. very easy to set up and very cool. |
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| 23:20.39 | bmoraca_work | vader--: it also supports modem detection and will reinvite from g729 to g711 if detected. faxing over them works quite well |
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| 23:24.14 | bmoraca_work | vader--: i found what that presales guy was talking about yesterday when he said you can't use the PRI and the FXS at the same time: http://kb.adtran.com/display/2/kb/article.asp?aid=1371 |
| 23:24.52 | bmoraca_work | vader--: what that means is that the 924 has a 24 simultaneous TDM-to-SIP conversion limit. you can't have 24 calls in via the PRI and expect to terminate all 24 to FXS. |
| 23:25.07 | bmoraca_work | vader--: but, from what you've said about your calling patterns, that shouldn't be an issue for you |
| 23:25.26 | [TK]D-Fender | bmoraca_work: Nasty limit... only half the device you think it is |
| 23:26.01 | [TK]D-Fender | vader--: Jut get 2 sane separate devices |
| 23:26.10 | bmoraca_work | [TK]D-Fender: not really. in the typical application (SIP to the Network and PRI to the PBX), you'll almost never hit it. |
| 23:26.50 | bmoraca_work | [TK]D-Fender: in vader--'s application, yes, it does potentially halve his capacity...but that's assuming the bulk of his calls are to his FXS ports. if they were, then a second device might be a better option. |
| 23:26.53 | [TK]D-Fender | bmoraca_work: Depends if the only point of FXS is for FAX and not analog sets for users. |
| 23:26.55 | [TK]D-Fender | bmoraca_work: And if he doesn't need 24 channels.... well... |
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| 23:28.51 | bmoraca_work | and the 900e series support 42 simultaneous...so, the sales guy was "technically" right...but not very accurate |
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| 23:33.04 | vader-- | we don't really have too many analog phones |
| 23:34.18 | vader-- | we have 21 FXS ports configured |
| 23:34.41 | bmoraca_work | vader--: how often are they used? |
| 23:35.37 | vader-- | 5 are faxes, 1 postage machine, 2 firealarm, 1 elevator, 1 security system, 10 analog phonesm and one used for telephone conference system |
| 23:36.07 | vader-- | the 924e i think has the capacity to do 23 pri channels and 24 fxs at the same time |
| 23:36.43 | bmoraca_work | vader--: no, 42 TDM-to-SIP conversions at the same time...so 23 PRI channels and 19 FXS |
| 23:36.54 | bmoraca_work | vader--: but that's irrelevant if you'll never get to that point |
| 23:37.52 | vader-- | TA924e 2nd gen can do 126 Simultaneous Calls |
| 23:38.16 | vader-- | 48 TDM to SIP conversions |
| 23:38.22 | bmoraca_work | vader--: right, but that's including SIP-to-SIP calls (b2b user agent). read the document i linked |
| 23:38.35 | vader-- | B2B? |
| 23:38.41 | bmoraca_work | back to back |
| 23:39.00 | bmoraca_work | d'oh, that table is weirdly laid out. you're right, 48 TDM-to-SIP |
| 23:39.17 | bmoraca_work | plus 84 SIP-to-SIP |
| 23:39.33 | vader-- | i need to get a unit to test with |
| 23:39.34 | vader-- | hehe |
| 23:39.43 | vader-- | configure it and then in off hours try and max it out |
| 23:40.14 | bmoraca_work | vader--: if capacity is an issue, you could always grab a Cisco AS5350... |
| 23:40.28 | vader-- | trying to keep this around 1500$ |
| 23:40.29 | bmoraca_work | you'd still need your channel bank, but it'd take care of everything else |
| 23:40.50 | bmoraca_work | and could even switch between PRI and FXS directly |
| 23:42.25 | bmoraca_work | vader--: little more than the TA924, but less than the 924e and would do what you need: http://cgi.ebay.com/Cisco-AS535-2T1-48-Universal-VOIP-Gateway-AS5350-Voice_W0QQitemZ330391723295QQcmdZViewItemQQptZCOMP_EN_Routers?hash=item4cece19d1f |
| 23:42.42 | bmoraca_work | although you wouldn't be able to register your FXS devices directly to asterisk |
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| 23:45.25 | vader-- | bmoraca know any sales guys at adtran i could get a demo from? |
| 23:46.18 | bmoraca_work | vader--: i don't, no. but, you could always buy it from a place that'd take it back if it didn't work |
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| 23:59.11 | Katty | bored. |
