| 00:01.23 | xx8xx | i see file |
| 00:01.24 | xx8xx | and console |
| 00:02.43 | cervi | where can i find users.conf and sip.conf? |
| 00:02.55 | xx8xx | http://www.pastebin.ca/361014 |
| 00:02.59 | xx8xx | thats users.conf |
| 00:03.10 | xx8xx | http://www.pastebin.ca/361025 |
| 00:03.13 | xx8xx | and thats sip.conf |
| 00:04.29 | cervi | which ID is for cisco and which one for softphone? |
| 00:04.44 | xx8xx | 101 is cisco |
| 00:04.48 | xx8xx | 103 is soft |
| 00:08.00 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 00:08.22 | cervi | you have a difference between 101 and 103 |
| 00:08.32 | cervi | in 101 you have set "registersip=yes" |
| 00:08.38 | cervi | try to remove that |
| 00:10.12 | xx8xx | I did and restarted asterisk |
| 00:10.15 | xx8xx | still same problem |
| 00:11.55 | cervi | we could try a dirty UNSECURE hack |
| 00:12.12 | cervi | copy numberplan-custom-1 to context default, to see, if its an extension problem |
| 00:12.44 | xx8xx | they are both in extention.conf right |
| 00:14.00 | cervi | does it work? |
| 00:14.40 | xx8xx | I can't find the section |
| 00:14.49 | xx8xx | context default ? |
| 00:15.44 | cervi | in extensions.conf copy the line "_480..." from [numberplan-custom-1] to [default] |
| 00:16.38 | xx8xx | I did find the numberplan-custom-1 |
| 00:16.43 | xx8xx | but copy it to where ? |
| 00:16.46 | xx8xx | which section |
| 00:16.49 | cervi | <PROTECTED> |
| 00:16.54 | xx8xx | aha |
| 00:16.55 | xx8xx | ok |
| 00:17.54 | xx8xx | it worked |
| 00:17.55 | xx8xx | :P |
| 00:18.22 | cervi | ok, so for some reason, you telephone wants only to work with [default] :-( |
| 00:18.46 | cervi | Please change later you changes, as now EVERYBODY could use your T1 line! |
| 00:18.54 | xx8xx | so is it ok to have two copies of that dialplan ? |
| 00:19.21 | robin_sz | define ok |
| 00:19.27 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:19.28 | robin_sz | ok it will work |
| 00:19.40 | robin_sz | but it will be a nightmare to maintian |
| 00:19.46 | xx8xx | right |
| 00:19.57 | xx8xx | well I guess thats a bug for asterisknow :P |
| 00:20.03 | robin_sz | doubt it |
| 00:20.16 | xx8xx | so those with cisco phones cant use *now ? |
| 00:20.17 | xx8xx | hehe |
| 00:20.29 | mafkees | eh ? |
| 00:20.36 | cervi | cisco can use it |
| 00:20.36 | mafkees | I have cisco phones |
| 00:20.40 | mafkees | try chan_skinny |
| 00:20.41 | cervi | but something is wrong |
| 00:20.44 | mafkees | it rules |
| 00:20.52 | robin_sz | did you try dialing from the console? |
| 00:20.55 | robin_sz | did that work? |
| 00:21.01 | mafkees | sure |
| 00:21.10 | cervi | for some reason, asterisk is looking in [default] instead of [numberplan-custom-1] |
| 00:21.16 | robin_sz | so |
| 00:21.23 | mafkees | that is with sip? |
| 00:21.24 | robin_sz | I ask again |
| 00:21.38 | robin_sz | cervi, so did you try dialing from the console, did that work? |
| 00:21.57 | xx8xx | dialing from a soft phone works perfect |
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| 00:22.06 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:22.11 | cervi | robin_sz: Its not me who has the problem ;-) |
| 00:22.14 | xx8xx | its only the cisco that wants to look in the default |
| 00:22.15 | mafkees | dialing from cisco phones works as good as with the xlite |
| 00:22.16 | robin_sz | ok |
| 00:22.20 | moprilo | hi.. i was looking to create a new context through the asterisk CLI.. is that posible? |
| 00:22.26 | moprilo | v1.4 (btw) |
| 00:22.27 | robin_sz | xx86xx , so did you try dialing from the console, did that work? |
| 00:22.28 | mafkees | xx8xx: cisco with sip or with sccp image ? |
| 00:22.45 | xx8xx | sip |
| 00:22.50 | mafkees | there you go |
| 00:22.53 | xx8xx | robin_sz how do I do that |
| 00:22.59 | mafkees | the cisco phones were made for sccp |
| 00:23.17 | robin_sz | dial 1234@number-plan-custom1 |
| 00:24.30 | mafkees | today the chan_skinny channel driver became 100% usable |
| 00:24.52 | robin_sz | ive never had any luck at all with chan_skinny |
| 00:25.00 | xx8xx | its giving me a repeated warning |
| 00:25.02 | xx8xx | how do I stop it |
| 00:25.02 | mafkees | it was usable before, if you didn't care about callerid on the phone |
| 00:25.06 | robin_sz | hangup |
| 00:25.19 | xx8xx | I did |
| 00:25.24 | robin_sz | type hangup |
| 00:25.25 | mafkees | today that last issue was fixed |
| 00:25.30 | xx8xx | [Feb 17 17:25:25] WARNING[9005]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 00:25.40 | robin_sz | dont worry about that |
| 00:25.48 | xx8xx | yeah but its going and going |
| 00:25.48 | robin_sz | so ... |
| 00:25.55 | mafkees | hhmm |
| 00:25.57 | xx8xx | but the number did go through |
| 00:26.01 | robin_sz | ok |
| 00:26.03 | xx8xx | my outsidee number rang |
| 00:26.05 | mafkees | I never tried chan_oss or chan_alsa |
| 00:26.06 | robin_sz | right |
| 00:26.18 | robin_sz | oh, I have to go ... things to do |
| 00:26.27 | xx8xx | how do I stop that warnning |
| 00:26.36 | xx8xx | it's filling up the screen |
| 00:26.37 | mafkees | my * is not in my 'to be reached' distance |
| 00:26.46 | mafkees | it's tucked away in a datacenter |
| 00:26.55 | xx8xx | oh got it |
| 00:26.56 | xx8xx | :P |
| 00:27.11 | mafkees | cisco phones should not use sip |
| 00:27.25 | mafkees | the skinny firmware is so much better |
| 00:27.37 | mafkees | for 7905, 7910 and 7960 that is |
| 00:27.44 | mafkees | I did not try any other phone |
| 00:27.58 | mafkees | the ATA's also run better on sccp firmware |
| 00:28.16 | mafkees | J4k3: no way |
| 00:28.34 | mafkees | J4k3: their quality is great |
| 00:28.59 | mafkees | I did not find a phone with a good speakerphone function till I tried the 7960 |
| 00:29.24 | mafkees | and their xml browser rules |
| 00:29.42 | mafkees | I really like this phone (cisco 7960 that is) |
| 00:30.16 | mafkees | and the 7940 is even better when you want to save some $$$$ |
| 00:30.24 | mafkees | cheap, good, stable |
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| 00:33.53 | *** join/#asterisk SkramX (n=mark@HERCULES.sentiensystems.net) |
| 00:35.22 | SkramX | any (php)agi gurus around? I am getting the following in the console.. Launched AGI Script /var/lib/asterisk/agi-bin/outgoing.php\n outgoing.php: Failed to execute '/var/lib/asterisk/agi-bin/outgoing.php': No such file or directory... The file is definitely there, I can read it from / using the path outputted by the console |
| 00:35.28 | SkramX | any ideas? |
| 00:35.28 | SkramX | chmodded to 755 |
| 00:36.05 | cervi | SkramX: how do you start it? |
| 00:36.12 | cervi | Param? |
| 00:36.17 | Strom_C | does it execute when you start it from the console? |
| 00:36.25 | SkramX | ;; call script and stuff |
| 00:36.26 | SkramX | exten => web-submitted,1,agi,outgoing.php |
| 00:36.36 | SkramX | Strom_C: I haven't tried that yet |
| 00:36.49 | Strom_C | SkramX: that should be your first step |
| 00:37.57 | Strom_C | s/console/bash prompt/ |
| 00:38.04 | SkramX | right |
| 00:42.37 | cervi | SkramX: Make sure, /var is not mounted with "noexec" |
| 00:43.52 | SkramX | right |
| 00:44.01 | SkramX | i'm debugging this for someone and there were multiple error |
| 00:44.03 | SkramX | working on it |
| 00:44.04 | SkramX | thanks |
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| 01:00.43 | SkramX | how do i halt all retries? |
| 01:01.08 | mafkees | halt -p |
| 01:01.38 | SkramX | just for asterisk calls |
| 01:02.11 | mafkees | asterisk -rx 'stop now' |
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| 01:04.18 | SkramX | weird |
| 01:04.25 | SkramX | anyone here use voicepulse connect? |
| 01:04.41 | AJaymn | Anyone use Shellshark? |
| 01:10.11 | mafkees | not me |
| 01:14.07 | SkramX | all fixed :) |
| 01:21.49 | kanaeda | http://www.networkworld.com/news/2006/080906-asterisk.html?prl |
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| 01:44.20 | MysticOne | hi all ... hopefully a quick question :) Obviously asterisk does SIP, but does it also do anything with SIMPLE? |
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| 01:55.19 | rbd | hey guys, with meetme, can I use the 'b' (run AGI background script) along with the 'r' option (record conference to a file). the meetme voip-info documentation is a bit unclear there |
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| 02:00.02 | kanaeda | what is the best service provider for multiple simultaneous phone calls |
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| 02:50.03 | Swabby-- | Is it possible to use Asterisk at home if i have VoIP service? |
| 02:50.16 | J4k3 | yes |
| 02:50.45 | J4k3 | there are also cards and network boxes to attach phone lines to your * |
| 02:50.49 | J4k3 | ISDN, POTS, whatever. |
| 02:50.50 | Swabby-- | How does Asterisk know how to handle the incoming calls? Do I still need FXO cards? |
| 02:51.06 | J4k3 | if you have VoIP, its pure IP to whatever you want to use for phones |
| 02:51.10 | Swabby-- | I understand the outgoing...i guess you would specify the SIP login/password info and server info |
| 02:52.27 | Swabby-- | does asterisk say logged into the SIP server and "listen" for a call? |
| 02:52.36 | J4k3 | yes |
| 02:52.58 | Swabby-- | Gotcha... |
| 02:53.00 | Swabby-- | now..for the extensions... |
| 02:53.08 | Swabby-- | two questions |
| 02:53.18 | Swabby-- | 1. Can you get phones that work with a wireless network..or do you have to do some "wiring" |
| 02:53.47 | J4k3 | there are some wifi phones out there |
| 02:53.51 | J4k3 | that resemble cellphones |
| 02:54.05 | J4k3 | I have a utstarcom f1000g. it feels like a 1999 digital cellphone |
| 02:54.08 | J4k3 | ie - cheezy |
| 02:54.12 | *** join/#asterisk pbd (n=pbd@c-67-163-20-134.hsd1.il.comcast.net) |
| 02:54.17 | J4k3 | but it works fairly well once I got my wireless network friendly with it. |
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| 02:55.33 | Swabby-- | Where does my VoIP converter that i got from the VoIP supplier come into play in my little network |
| 02:55.40 | Swabby-- | does it become simply a "backup device" |
| 02:55.41 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:55.48 | Swabby-- | if i use all voip phones because they will be talking to the server anyway? |
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| 02:56.38 | moprilo | how do i create a new context through cli? |
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| 02:59.44 | dansmith | [TK]D-Fender: I'm having trouble getting my polycom registered to asterisk |
| 02:59.59 | dansmith | asterisk says: Registration from '<sip:500@192.168.201.88>' failed for '192.168.201.65' - Username/auth name mismatch |
| 03:06.23 | *** part/#asterisk MysticOne (n=mysticon@puddlejumper.foxybanana.com) |
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| 03:07.29 | bkruse_home | dansmith: i think thats prety self-explantory.... |
| 03:07.50 | Swabby-- | dansmit: check your username/password |
| 03:07.53 | dansmith | bkruse_home: yea, you'd think... but I think I've got the right values |
| 03:08.18 | dansmith | Swabby--: it's the username, apparently, because if I use the wrong password from a softphone, I get a different message |
| 03:08.19 | bkruse_home | dansmith: i beg to differ. |
| 03:08.32 | bkruse_home | dansmith: the ip address's are wrong. |
| 03:08.33 | bkruse_home | look at it. |
| 03:09.11 | dansmith | the "from" should be the IP of the phone? |
| 03:09.17 | bkruse_home | correct. |
| 03:09.19 | bkruse_home | is it? |
| 03:09.28 | dansmith | no, 88 is the server, 65 is the phone |
| 03:09.38 | bkruse_home | oh |
| 03:09.42 | bkruse_home | your right |
| 03:09.54 | dansmith | I thought that seemed right |
| 03:10.04 | bkruse_home | it has to be man. |
| 03:10.26 | bkruse_home | username / password, turn sip debug on |
| 03:10.29 | dansmith | I was wondering if there is anything polycom-specific that I need to know for the poly config |
| 03:10.32 | dansmith | how do I do that? |
| 03:10.54 | bkruse_home | sip debug on the * cli |
| 03:12.22 | dansmith | hrm... |
| 03:12.31 | dansmith | I see "Digest username="501"... |
| 03:17.09 | dansmith | that has to be it, because the softphone registers with "Digest username="500"" |
| 03:20.05 | bkruse_home | gotcha |
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| 03:21.57 | dansmith | now if only I knew why it was doing 501 instead of 500... |
| 03:22.10 | bkruse_home | :P |
| 03:22.19 | bkruse_home | look at ur polycom config file |
| 03:22.50 | bkruse_home | or better yet, web interface |
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| 03:33.57 | dansmith | omg, I'm going to f'in kill this thing |
| 03:34.36 | Swabby-- | do'nt kill it |
| 03:35.04 | kanaeda | dont do that |
| 03:35.15 | J4k3^ | no way man |
| 03:35.17 | J4k3^ | smash it up! |
| 03:35.24 | J4k3^ | get the video camera first tho |
| 03:35.31 | dansmith | hehe |
| 03:35.39 | dansmith | I'm making progress at least |
| 03:36.01 | dansmith | I wish the f'in web interface didn't take 5 f'in minutes to become available after boot |
| 03:40.05 | elriah | Polycom's are easy, what's the problem? |
| 03:40.47 | dansmith | well, now the problem is that I've got the password for the extension plugged into the phone directly but asterisk says the password is wrong |
| 03:40.58 | dansmith | it's right in the config and right in the phone itself |
| 03:41.07 | dansmith | and a softphone works |
| 03:41.17 | elriah | What does your sip.conf look like? |
| 03:41.58 | elriah | dansmith: Try #tribox |
| 03:42.07 | dansmith | elriah: I have :) |
| 03:42.15 | dansmith | although not with the password problem I guess |
| 03:42.27 | elriah | dansmith: Switch to asterisk proper? |
| 03:42.41 | dansmith | heh |
| 03:42.43 | dansmith | now, now :) |
| 03:43.06 | dansmith | but, what in my sip.conf were you going to have me look at? |
| 03:43.47 | elriah | I don't know anything about tribox, sorry :( |
| 03:43.53 | dansmith | heh |
| 03:44.15 | dansmith | I think I'd switch to asterisk proper once I get comfortable enough.. only been a week though :) |
| 03:44.27 | elriah | It's really easy. |
| 03:46.35 | dansmith | are there any restrictions on passwords that I don't know about |
| 03:46.36 | dansmith | ? |
| 03:46.47 | dansmith | like have to be alpha or over a certain length? |
| 03:46.57 | dansmith | for sip, I mean, not trixbox or asterisk |
| 03:47.51 | dansmith | heh |
| 03:48.56 | sharp | I SAW YOU!!! WITH A TICKET STUB IN YOUR HAND!!!!! |
| 03:49.11 | sharp | damn. |
| 03:49.13 | sharp | wrong channel. |
| 03:49.17 | sharp | (excuse me.) |
| 03:50.01 | coppice | why? there is no excuse for using capitals on irc |
| 03:50.49 | sharp | there is when its a song lyric meant to be sung... loudly. |
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| 04:10.00 | sharp | apt-get? |
| 04:10.00 | J4k3^ | (he was a bit dull there for a moment...) |
| 04:10.24 | J4k3^ | windowsupdate.microsoft.com |
| 04:10.27 | J4k3^ | burn! |
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| 04:12.17 | quidpro | Has anybody had any build issues with the Zaptel 1.2.13? |
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| 04:35.36 | tonyheb | The following message could not be delivered to all recipients: |
| 04:35.36 | tonyheb | stie de caliss de marde |
| 04:35.45 | tonyheb | sorry |
| 04:35.50 | tonyheb | pad paste |
| 04:36.47 | Strom_C | The following message could not be delivered to all recipients: Bienvenue a merde |
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| 04:53.14 | tonyheb | Strom_C: frenchy? |
| 04:53.28 | Strom_C | no |
| 04:53.37 | Strom_C | I only know perhaps a dozen phrases |
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| 05:01.06 | foobar778 | hello all |
| 05:01.20 | foobar778 | I got it working!!!!!! |
| 05:01.25 | bkruse_home | foobar778: :D |
| 05:01.43 | foobar778 | outbound calls to priovider |
| 05:02.41 | foobar778 | despite the grief earlier and I apalogize for my part but for the information Fender wqas very off on the extensions.conf any way its all working |
| 05:03.02 | tonyheb | We are having a strange sound when a caller direct dials an extension from the ivr, its almost a crunching sound. Sound like a corrupted sound file, any idea? |
| 05:04.28 | foobar778 | For my part if anyone need to know how to make entries in their sip.conf or extensions.conf I would tell them in two sentences rather than have to read all the manuals and more as I have done |
| 05:06.32 | putzz | foobar: you are back again.......if you dont read the manuals you wont learn, so someone telling you to go read is a good advice. now have a nice day! |
| 05:07.33 | bkruse_home | use the gui |
| 05:10.00 | foobar778 | puttz I got it working and Fender was way off I have read them and it works but I will help if anyone asks me I wont be arrogant and say read the manual when someone is just starting out I will help them alomgf I read a ton and what I needed was just a short bit It took hours I would rather encoiurage someone ratherr than have them give up and not force them tp spend hours just because I have!@!!!!! |
| 05:10.35 | bkruse_home | [TK]D-Fender is never off, it was probably because you did not give him sufficent information, and was just trying to help |
| 05:10.51 | foobar778 | sorry he was way off |
| 05:11.36 | bkruse_home | foobar778: that was because you did not give him sufficent information |
| 05:12.05 | Bobthehunter | just gave ALMOST accurate info |
| 05:12.09 | bkruse_home | foobar778: wow, congrats, you solved your first problem, go get a beer, and stop flaming please. |
| 05:12.10 | foobar778 | I gave him more than he needed and since u werernt there u are talking out your ....... |
| 05:12.13 | Bobthehunter | is my last message ;) maybe he applied |
| 05:12.20 | *** join/#asterisk Op3r (n=Op3r@121.97.213.138) |
| 05:12.28 | bkruse_home | foobar778: its called having more than one nick moron, please stop the flaming. |
| 05:12.37 | foobar778 | First problem hahahaha dont make me laugh |
| 05:13.07 | *** mode/#asterisk [+b %foobar778!*@*] by Qwell |
| 05:13.15 | Qwell | We settled this earlier. It's over. |
| 05:13.16 | bkruse_home | Qwell thanks. |
| 05:13.20 | bkruse_home | thanks qwell |
| 05:13.20 | Bobthehunter | omg.. 2 much drama for me here.. have fun and dont burn in flames |
| 05:13.21 | bkruse_home | temp ban? |
| 05:13.30 | Bobthehunter | assholes ruining this channel again |
| 05:13.36 | Bobthehunter | oh ok |
| 05:13.40 | bkruse_home | Bobthehunter: he banned him. |
| 05:13.42 | Bobthehunter | ill stay since someone took measures |
| 05:13.42 | bkruse_home | :] |
| 05:13.44 | Bobthehunter | ;) |
| 05:13.44 | bkruse_home | :] |
| 05:14.04 | Bobthehunter | well still to much chances where given... |
| 05:14.07 | moprilo | is there a way to create a context dynamicly.. with the asterisk java api or with a CLI command? |
| 05:14.16 | Bobthehunter | after 2 times you ban 1 day then another and 1 month |
| 05:14.21 | Qwell | moprilo: no, but you can add an extension to a context |
| 05:14.42 | moprilo | but if i want to add users dynamicly and don-t want them to share contexts.. ? |
| 05:15.06 | moprilo | any work around ..? |
| 05:15.09 | JT | really pisses me off when people rubbish others' genuine efforts, 'specially since we aren't paid to be here :/ |
| 05:16.13 | bkruse_home | you can make them pretty dynamic with writting and reloading your dialplan |
| 05:16.30 | bkruse_home | JT: seriously |
| 05:16.47 | moprilo | true.. bkruse, i'm affraid i'll go with that.. |
| 05:16.50 | bkruse_home | I normally wouldnt have joined in, but all tk fender does is help people |
| 05:17.57 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 05:18.04 | iq | Hi |
| 05:18.54 | bkruse_home | :P |
| 05:18.57 | bkruse_home | sup iq |
| 05:19.24 | *** mode/#asterisk [-b %foobar778!*@*] by Qwell |
| 05:33.16 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
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| 05:38.24 | Bobthehunter | ? |
| 05:40.25 | JT | bkruse_home: true |
| 05:40.32 | bkruse_home | :] |
| 05:55.16 | tonyheb | Any idea on what could cause the weird sound i'm hearing when an caller dials an extension from ivr? |
| 05:55.50 | tonyheb | I dont think it's a corrupted sound file as I dont see any sound beeing played in the CLI |
| 06:09.27 | *** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca) |
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| 06:16.07 | *** part/#asterisk tonyheb (n=dub@modemcable251.58-131-66.mc.videotron.ca) |
| 06:18.27 | *** join/#asterisk jpe-nyc (n=jpe@p77-37.acedsl.com) |
| 06:23.45 | kanaeda | what is a good free softphone for testing purposes? |
| 06:23.54 | kanaeda | i tried sj phone but it is buggy |
| 06:26.02 | AJaymn | X-Lite ? |
| 06:26.59 | kanaeda | does that require a login? |
| 06:27.27 | AJaymn | ya. you need to give it an exten and stuff.. |
| 06:30.53 | kanaeda | k ill try it out thx! |
| 06:43.38 | *** join/#asterisk pnlarsson (n=nick@c83-248-12-187.bredband.comhem.se) |
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| 07:23.00 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:32.48 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 07:35.02 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 07:42.15 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 07:48.25 | putzz | zZzZzZzZ |
| 07:49.17 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:51.35 | kanaeda | what is default asterisk port |
| 07:53.20 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 07:58.46 | dlynes_laptop | kanaeda, 5060 for sip, 4569 for iax2, ... |
| 07:58.59 | pnlarsson | 5038 for ami |
| 07:59.05 | pnlarsson | etc... |
| 08:03.54 | J4k3 | 666 for the FBI/CIA/DHS backdoor |
| 08:03.58 | J4k3 | j/k ;) |
| 08:04.16 | dlynes_laptop | oh |
| 08:04.29 | dlynes_laptop | I was wondering why my system was listening on udp 666 |
| 08:07.21 | coppice | and 42 has all the answers |
| 08:07.57 | dlynes_laptop | i don't have any answers :( |
| 08:08.01 | coppice | I think the daemons listen on 666 |
| 08:09.00 | dlynes_laptop | xing yen kuai le, coppice |
| 08:09.27 | coppice | dlynes_laptop: san nin faai lok |
| 08:09.42 | dlynes_laptop | coppice, is that the same thing in cantonese? |
| 08:10.06 | coppice | what's yen? |
| 08:10.22 | dlynes_laptop | coppice, cantonese almost always looks and sounds completely different from mandarin, so i have no idea |
| 08:10.22 | J4k3 | something I want a big stack of |
| 08:10.30 | dlynes_laptop | hahaha |
| 08:10.33 | coppice | kuai le == fai lok |
| 08:10.44 | dlynes_laptop | yen means year |
| 08:10.50 | dlynes_laptop | xing means new |
| 08:10.55 | dlynes_laptop | kuai le means happy |
| 08:11.04 | coppice | 新年快樂 |
| 08:11.25 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 08:11.33 | coppice | å¹´ is nian, isn't it? |
| 08:12.02 | J4k3 | its a silly looking rectangle on my ignorant XP install. |
| 08:12.25 | J4k3 | but if I copy and paste it into another window |
| 08:12.31 | J4k3 | its a line of text about 10 charectors long |
| 08:12.33 | J4k3 | thats wild. |
| 08:12.41 | coppice | then you don't have the asian fonts installed |
| 08:12.49 | J4k3 | I've been hacked by a foreign language! |
| 08:12.52 | J4k3 | yeah |
| 08:13.13 | putzz | æ–°å¹´å¿« heh |
| 08:13.42 | putzz | = æ°å¹´å¿« |
| 08:14.21 | J4k3 | ¿bueno? |
| 08:14.39 | dlynes_laptop | coppice, ah...maybe...I just know how to say it...never asked what the hanyu pinyin is |
| 08:14.50 | dlynes_laptop | coppice, so maybe i've always pronounced it slightly incorrect |
| 08:15.47 | dlynes_laptop | coppice, and i've looked at five different websites, and they spell it five completely different ways in hanyu pinyin |
| 08:15.58 | dlynes_laptop | coppice, so nobody seems to know what it is |
| 08:17.53 | dlynes_laptop | coppice, I guess it's: æ–°å¹´å¿«ä¹ (xin nian kuai le) |
| 08:17.55 | putzz | que passa hota |
| 08:18.49 | dlynes_laptop | coppice, I guess you were using traditional chinese? |
| 08:19.05 | coppice | 我ä¸è˜æ¼¢èªžæ‹šéŸ³ |
| 08:19.16 | coppice | 是 |
| 08:19.50 | coppice | where do you come from originally? |
| 08:19.55 | dlynes_laptop | ah....è°¢è°¢ä½ ã€‚ã€‚ã€‚å¾ˆå¥½ |
| 08:21.04 | J4k3 | heh, awesome |
| 08:21.22 | J4k3 | I always wanted a nice tower mounted weather station, and now I have an excuse to buy one |
| 08:21.23 | J4k3 | asterisk. |
| 08:21.42 | J4k3 | time/date... and real local weather (we have no government weather station within 40 miles) |
| 08:26.19 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
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| 08:42.57 | coppice | government weather? is that "its fine and sunny, but definitely no warmer than historic averages"? |
| 08:43.34 | mafkees | "It's a beautiful day to start a war" |
| 08:44.35 | coppice | "There will be liught showers of cluster bombs" |
| 08:46.06 | mafkees | something like that yeah |
| 08:47.00 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:51.33 | J4k3 | its a good day to die for god and country! |
| 08:54.46 | coppice | I think a country's consitution should allow its leaders to declare war, but the must forfeit their lives if they do, if the wars are really so necessary, they should be happy to be the first to lay down their lives for them :-) |
| 08:55.24 | mafkees | coppice: good one. write a letter to your goverment |
| 08:58.56 | tzafrir | coppice, normally they have a good excuse: |
| 08:59.11 | tzafrir | "<the other party> started" |
| 09:04.28 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 09:08.39 | J4k3 | the second one flew straight in, the first one was pulling quite hard. |
| 09:25.54 | coppice | they were probably arguing over which tower they should be going for |
| 09:28.34 | coppice | tzafrir: I didn't say the forfeiting thing should only apply when the declaration comes ahead of the one from the other side. the weasels always try to create the impression they are the injured party. even the nazis did that.... damn, Godwin's law has just struck me. |
| 09:30.12 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 09:35.08 | *** join/#asterisk tRSS (n=tRSS@124.29.197.41) |
| 09:37.01 | tRSS | hey everyone |
| 09:37.29 | tRSS | is it possible to use FWD sip services on a different port? my isp is blocking 5060. |
| 09:38.10 | coppice | bomb the ISP, and teach them not to mess with net neutrality |
| 09:38.30 | tRSS | well, i would love too, but I cant sadly |
| 09:38.49 | tRSS | i am able to configure FWD IAX successfully, but that also works sometime |
| 09:39.08 | tRSS | it would stop working for weeks and then start working by itself |
| 09:39.22 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 09:39.41 | J4k3 | have all your sip traffic replicate itself 10x onto their DNS servers... they'll appreciate the random traffic. |
| 09:39.58 | J4k3 | onto being 'toward' |
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| 10:06.20 | suma | how can i make asterisk to use mysql for DBPut ... asterisk database commands ? |
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| 10:36.33 | *** join/#asterisk twostepsback (n=2stepsba@59.183.55.120) |
| 10:37.14 | twostepsback | hi all, i want to send a recorded voice to a mobile phone from a computer, will asterisk help? |
| 10:39.50 | endre | asterisk always halps |
| 10:40.00 | endre | i do it for wakeup alarm |
| 10:42.05 | *** part/#asterisk unimprtnt (n=brian@c-68-36-163-21.hsd1.nj.comcast.net) |
| 10:42.37 | twostepsback | endre: thanks |
| 10:43.13 | kanaeda | kekek |
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| 14:33.41 | suma | anyone in here ? |
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| 14:55.21 | tzafrir | no |
| 14:56.01 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 14:56.17 | robin_sz | BAH ... |
| 14:56.33 | robin_sz | so, these Snoms ... |
| 14:56.46 | robin_sz | it seems that *sometimes* authenticate |
| 14:56.54 | robin_sz | sometimes not |
| 14:57.16 | robin_sz | the ones on the local network seem to be happily authenticating 100% |
| 14:57.40 | robin_sz | the ones out on VPN sites seem to be getting 401 unauthorised about 50% of the time |
| 14:57.58 | robin_sz | seriously weird |
| 15:05.36 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 15:05.43 | robin_sz | now, this is even wierder ... the phone show "unauthorised" .. sip show peers shows that it is not auth'd either ... |
| 15:05.46 | robin_sz | office (Unspecified) D 0 UNKNOWN |
| 15:05.59 | robin_sz | yet ... it can still make calls ... |
| 15:07.06 | robin_sz | both to internal and external numbers ... I hought the general idea was that unless the SIP device managed to authenticate, it couldn't dial out?? |
| 15:11.55 | mafkees | no |
| 15:11.58 | mafkees | when you dialout |
| 15:12.11 | mafkees | asterisk will send a reply: not authenticated |
| 15:12.17 | mafkees | your phone will send username/pass |
| 15:12.22 | mafkees | asterisk will accept that |
| 15:12.26 | mafkees | and setup the call |
| 15:14.42 | *** join/#asterisk websae (n=websae@adsl-75-48-246-49.dsl.milwwi.sbcglobal.net) |
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| 15:20.39 | jserve | Hi all |
| 15:20.40 | suma | how can i make asterisk to use mysql for DBPut ... asterisk database commands ? |
| 15:21.05 | mafkees | suma: you cant |
| 15:21.29 | mafkees | suma: you have to use an agi script or the mysql() dialplan functions |
| 15:21.48 | suma | i c |
| 15:22.16 | suma | mafkees: thanks |
| 15:22.40 | mafkees | good luck |
| 15:23.44 | suma | mafkees, what is the best way to store the data of asterisk (DBPut ) and retrieve through web? |
| 15:23.58 | suma | DBPut I can use in dialplan |
| 15:24.15 | suma | Is there is any API, i can take and use it to retrieve data through php ? |
| 15:29.51 | mafkees | asterisk -rx 'database show' |
| 15:30.04 | mafkees | and maybe the manager |
| 15:30.18 | mafkees | I dont know if the manager can read the astdb |
| 15:30.26 | mafkees | check the docs about the manager interface |
| 15:30.44 | mafkees | I do know you can do it with the command asterisk -rx 'database show' |
| 15:30.57 | mafkees | I use that in my php script to prepare xml for my cisco phones |
| 15:31.22 | suma | i c |
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| 15:34.17 | suma | that is great ! |
| 15:34.39 | suma | mafkees: Looks like DB works with asterisk manager interface, http://www.voip-info.org/tiki-index.php?page=Asterisk+manager+Example%3A+PHP |
| 15:34.40 | suma | thanks |
| 15:40.40 | websae | anyone in here used a grandstream gxw 4104 (4 port FXO gateway)? |
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| 15:48.08 | PakiPenguin | websae: any issue |
| 15:48.14 | websae | well... |
| 15:48.15 | websae | yes |
| 15:48.19 | PakiPenguin | i've used some of the beta ones |
| 15:48.21 | PakiPenguin | shoot |
| 15:48.24 | websae | I can place calls to it |
| 15:48.29 | PakiPenguin | okay |
| 15:48.34 | websae | but call coming in... |
| 15:48.37 | websae | not hitting asterisk |
| 15:48.58 | websae | i see in the syslog log of the grandstream that it's ringing the fxo port |
| 15:49.59 | *** part/#asterisk mega (n=mega@217.201.137.228) |
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| 15:55.01 | ruied | I'm trying to set a voipbuster outgoing account, when I dial the number * reports the error: "Got SIP response 400 "Bad request" back from 'x.y.z.w' -- SIP/voipbuster-081ebb20 is circuit-busy |
| 15:55.53 | ruied | what colud be the problem? * is behind a nat router, do I need to set STUN? |
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| 15:56.52 | quidpro | Anybody able to help me with me Zaptel 1.2.13 compiler error (RHEL 2.6.9.11 kernel)? Pastebin here... http://www.pastebin.ca/362145 |
| 15:56.57 | [TK]D-Fender | ruied : pastebin your entire sip.conf less passwords, and the section of extensions.conf being called to dial out (including set-up. |
| 15:56.58 | [TK]D-Fender | ~pb |
| 15:57.00 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:09.25 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 16:18.59 | ruied | [TK]D-Fender, http://www.pastebin.ca/362179 this is a litle bit confusing, I'm testing yet.... |
| 16:27.54 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:35.44 | *** part/#asterisk robl^ (n=robl@unaffiliated/robl/x-000001) |
| 16:35.45 | ruied | exten => _351xxxxxxxxx,2,Dial(SIP/{EXTEN}@voipbuster), I've missed the {EXTEN}@voipbuster ... now It looks like it makes the call, but the problem now seems to be the codec... how can I define the codec type for voipbuster account? |
| 16:37.41 | [TK]D-Fender | ruied : Do yourself and everyone else a big favour and remove EVERYTHING that is commented out in both files, and everything that you are not using. Sample files suold never be used, only examined... |
| 16:38.26 | [TK]D-Fender | ruied : onces thats done, redo the pastebin and I'll look |
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| 16:39.12 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:39.20 | ruied | ok |
| 16:44.46 | *** join/#asterisk Opperior (n=chatzill@c-75-69-247-108.hsd1.nh.comcast.net) |
| 16:54.25 | *** join/#asterisk gnubien (n=e@169.143.8.67.cfl.res.rr.com) |
| 16:55.16 | gnubien | hi, grandstream 101 has only 1 RJ45 connector, how to connect phone and eth card at the same time? |
| 16:56.37 | *** join/#asterisk ToyMan (n=Stuart@cpe-69-201-220-167.hvc.res.rr.com) |
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| 16:58.38 | Opperior | Best way would be to run another line. If that's not an option, a small switch would do the trick, but I wouldn't recomend it |
| 16:59.28 | gnubien | Opperior: how to run another line from the cable modem which only has 1 RJ45 connector? |
| 17:00.15 | Opperior | ah, oops. Ok, then you'll need a router, like a Linksys |
| 17:01.11 | gnubien | Opperior: ok, no way to use a cable splitter that has 1 rj45 in and 2 rj45 out connectors? |
| 17:01.23 | Opperior | no, ethernet doesn't work that way |
| 17:02.04 | gnubien | Opperior: ok, granstream 102 has 2 rj45 connectors, still need a router? |
| 17:02.36 | Opperior | let me look up some specs on that device to see |
| 17:02.43 | gnubien | ok |
| 17:02.58 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 17:04.03 | Opperior | this would depend on your cable provider. If they allow multiple devices on their network, then no. If they restrict you to a single device, then yes |
| 17:04.14 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:04.48 | gnubien | Opperior: ok, thanks for the info |
| 17:04.49 | Opperior | either eay, I would still recommend a router with a built-in firewall for security |
| 17:04.59 | robin_sz | [TK]D-Fender, right ... you remember that annoying auth problem ? |
| 17:05.22 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 17:06.17 | robin_sz | [TK]D-Fender, strangely it seems to be intermittent ... somtimes it auths, sometimes it fails ... I havent narrowed it down to when exactly it does what, but its not consistent, which is weird. |
| 17:06.48 | [TK]D-Fender | gnubien : Run it behind a router |
| 17:07.12 | [TK]D-Fender | robin_sz : Strange |
| 17:07.21 | robin_sz | yeah, very |
| 17:07.36 | gnubien | need a router using an ATA also? |
| 17:07.36 | robin_sz | and ... right now its not auth'd .. but I can make calls |
| 17:09.10 | ruied | [TK]D-Fender, sip.conf- http://www.pastebin.ca/362218; extensions.conf- http://www.pastebin.ca/362219 |
| 17:09.56 | robin_sz | yeuw! two general sections? |
| 17:10.30 | gnubien | Opperior: need a router using an ATA? |
| 17:10.40 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917CC8.dip.t-dialin.net) |
| 17:10.41 | Opperior | one sec... |
| 17:10.50 | *** part/#asterisk s1gny|wrk (n=s1gny@p54917CC8.dip.t-dialin.net) |
| 17:10.58 | robin_sz | ruied delete one [general] header and decide on which bits you need and remove the duplicates |
| 17:11.26 | robin_sz | I suspect it will work but it looks messy and is confusing |
| 17:14.07 | [TK]D-Fender | gnubien : You'll need a router if you have a bunch of networked devices. |
| 17:14.56 | Opperior | honestly not familiar enough with ATA devices to say |
| 17:15.07 | gnubien | [TK]D-Fender: thanks |
| 17:15.15 | gnubien | Opperior: ok, thanks again |
| 17:15.52 | [TK]D-Fender | ruied : You're behind NAT right? |
| 17:25.13 | ruied | yes |
| 17:25.51 | [TK]D-Fender | ruied : You're missing all the settings to allow it to work. |
| 17:27.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:27.44 | *** part/#asterisk gnubien (n=e@169.143.8.67.cfl.res.rr.com) |
| 17:27.47 | ruied | [TK]D-Fender, what do I need to add? stun? I've checked with my own voipbuster account that * oliveirasnet is not connected... |
| 17:28.17 | [TK]D-Fender | ruied : http://www.pastebin.ca/362234 |
| 17:29.38 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
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| 17:36.40 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
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| 17:48.46 | ruied | [TK]D-Fender, I've made the changes it seems oliveirasnet is not connected (with my personal voipbuster account the * voipbuster account is not registering). When I dial it reports the following error: http://www.pastebin.ca/362252 |
| 17:50.32 | ruied | ah... I forgot one thing... |
| 17:50.58 | [TK]D-Fender | ruied : You meant he "register" line? :) |
| 17:51.39 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:53.39 | ruied | I meant that the 'oliveirasnet' voip account doesn't appear to be connected to the voipbuster server |
| 17:56.19 | [TK]D-Fender | ruied : Why would it be "connected"? |
| 17:56.38 | [TK]D-Fender | ruied : Its a PEER. you use that to dial out w/ auth. |
| 17:56.49 | [TK]D-Fender | ruied : It does not maintain any kind of idle connection. |
| 17:57.57 | ruied | [TK]D-Fender, ah, ok... Do I need to set the "externalip=a.b.c.d" ? cause I don't have fixed ip... |
| 17:58.10 | Qwell | ruied: externhost |
| 17:58.24 | ruied | ok |
| 17:59.44 | Qwell | (make sure you look at externrefresh also) |
| 18:00.19 | mafkees | Qwell: you really made me enjoy the first couple of phonecalls yesterday evening ;) |
| 18:00.24 | Qwell | ;) |
| 18:00.29 | mafkees | "ooooooooooooooooh, callerid !" |
| 18:00.45 | Qwell | thank wedhorn, not me.. I just cleaned it up (and sent him the dumps :p) |
| 18:00.54 | mafkees | yeah |
| 18:00.57 | mafkees | he did a great job |
| 18:01.10 | mafkees | but you put it in trunk |
| 18:01.14 | Qwell | the actual patch ended up being really simple... |
| 18:01.15 | mafkees | no more patches for me |
| 18:01.25 | Qwell | it was literally just moving some of the calls around |
| 18:01.26 | mafkees | I noticed on asterisk-svn list yeah |
| 18:01.47 | mafkees | sometimes things are easy |
| 18:01.54 | mafkees | sometimes not |
| 18:03.18 | Qwell | I looked at speeddials a bit yesterday... and the way it dials is silly |
| 18:03.42 | mafkees | gheh, what else is new with them phones ;) |
| 18:04.11 | Qwell | well, it' |
| 18:04.11 | Qwell | 's my code that's broken :P |
| 18:04.16 | Qwell | I think I'm just gonna do a simpleswitch, and just queue up some DTMF |
| 18:04.48 | mafkees | that should work |
| 18:05.12 | mafkees | well, you know I'm willing to test :) |
| 18:05.27 | mafkees | I'm trying to understand how it all works |
| 18:05.33 | mafkees | reading and reading and reading |
| 18:05.36 | mafkees | man what a code |
| 18:05.41 | Qwell | yeah... |
| 18:05.44 | *** join/#asterisk friedrich| (n=friedric@e177246080.adsl.alicedsl.de) |
| 18:06.14 | mafkees | not easy to become an asterisk programmer |
| 18:06.34 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 18:06.44 | brookshire | sure it is! |
| 18:07.11 | Qwell | unless you want to develop on chan_skinny :P |
| 18:07.15 | brookshire | asterisk is actually rather easy to program for, compaired some other projects in c |
| 18:07.29 | brookshire | compared to |
| 18:07.31 | brookshire | bah |
| 18:13.19 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
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| 18:16.36 | ruied | hehhe "exten => _351xxxxxxxxx,2,Dial(SIP/{EXTEN}@voipbuster)" I've missed the $ in {EXTEN} |
| 18:16.56 | ruied | it's working now... |
| 18:18.01 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 18:18.05 | bkruse_home | LOL |
| 18:18.07 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 18:18.17 | bkruse_home | so did you see it dialing the word {EXTEN{@voipbuster? |
| 18:18.18 | ruied | :) |
| 18:19.08 | ruied | yeap, that was when I've noticed something was wrong... :) |
| 18:21.07 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 18:23.49 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 18:24.38 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 18:24.52 | DocHolliday | hey bkruse_home |
| 18:30.47 | bkruse_home | DocHolliday: wuts up brotha |
| 18:31.07 | bkruse_home | just throwin down a lil code, and yourself? |
| 18:31.29 | [TK]D-Fender | bkruse_home : thanks for the good word late last night, but I did withhold the ansewr from that guy. He was trash talking and lazy expecting us to just do it all for him and showed no dedication to actually learning *. Qwell had kicked him twice prior and he never learned his lesson |
| 18:31.46 | DocHolliday | getting up in the morning is extremely difficult.. especially when its 1:30 in the afternoon |
| 18:32.18 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-45.intrastar.net) |
| 18:32.32 | bkruse_home | [TK]D-Fender: either way, i knew you were right |
| 18:32.38 | DocHolliday | bkruse_home, trying to research existing fax issues with asterisk |
| 18:32.38 | bkruse_home | :) |
| 18:32.39 | [TK]D-Fender | bkruse_home : :) |
| 18:32.48 | bkruse_home | DocHolliday: thats actually a good idea. |
| 18:33.11 | bkruse_home | just dont transcode anywhere along the line and it makes things more probable |
| 18:33.32 | DocHolliday | well if i get an ATA thats T.38 compatible.. does it matter since asterisk doesnt support T.38? |
| 18:34.05 | bkruse_home | it does support T.38 passthru right Qwell? |
| 18:34.06 | bkruse_home | and [TK]D-Fender i knew that either a) You gave him the answer to the best of your abilities from the information or b) he was being a dick and demanded someone ssh into his box and fix it "ZAPTEL IS BROKE!" or something similar |
| 18:34.14 | Qwell | yes |
| 18:35.23 | DocHolliday | Qwell AH! so technically if the ATA supports T.38 then Asterisk can do passthru to the SIP compatible T.38 provider? |
| 18:35.33 | bkruse_home | correct |
| 18:35.35 | Qwell | that's what I said like 2 days ago, yes :) |
| 18:35.41 | bkruse_home | your endpoint must do T.38 |
| 18:35.45 | bkruse_home | Qwell: thats what i thought :] |
| 18:35.53 | bkruse_home | Qwell: hows SMS going in the new channel driver? has the dude committed to doing it? |
| 18:35.59 | bkruse_home | if not, ima take a stab at it today |
| 18:36.01 | DocHolliday | and if the endpoint *does do T.38* am i guaranteed that faxing will work? |
| 18:36.06 | Qwell | bkruse_home: he wants to stabalize first, but yeah |
| 18:36.14 | bkruse_home | oh, awesome |
| 18:36.18 | bkruse_home | thats exciting |
| 18:36.28 | Qwell | tuesday, I should be getting my headset :D |
| 18:36.35 | bkruse_home | nice!!!! |
| 18:36.45 | Qwell | then I'll write "fxs" support for it |
| 18:36.46 | DocHolliday | what kind of handset? |
| 18:36.50 | bkruse_home | ima use USB instead of bluetooth |
| 18:36.51 | bkruse_home | we can have texting wars |
| 18:36.55 | bkruse_home | Qwell: nice!!!! |
| 18:37.08 | *** join/#asterisk [1]J (n=new@adsl-065-006-173-139.sip.mia.bellsouth.net) |
| 18:37.08 | Qwell | bkruse_home: the other day, I still had Keiths adapter, and I put mine in too... |
| 18:37.15 | Qwell | I did a scan with asterisk, and it found... |
| 18:37.16 | Qwell | ... |
| 18:37.18 | Qwell | ...asterisk |
| 18:37.25 | bkruse_home | woah! |
| 18:37.38 | bkruse_home | thats a potenial uh, loopback, major |
| 18:37.42 | Qwell | same box though, so it was useless :p |
| 18:37.47 | Qwell | well, kinda useless anyhow |
| 18:37.48 | Damin | That sounds pretty bad... |
| 18:37.54 | Qwell | I'm sure *somebody* could find a use for it |
| 18:38.09 | DocHolliday | qwell, what kind of headset are you getting? |
| 18:38.11 | bkruse_home | a meaningless use for it |
| 18:38.19 | bkruse_home | i want to make it compatible with 3rd party SMS gateways and what not. eventually |
| 18:38.21 | Qwell | bluetooth |
| 18:38.26 | bkruse_home | alot of them have super simple API's for it |
| 18:38.39 | DocHolliday | nice |
| 18:39.03 | DocHolliday | bkruse_home, does the Cisco SIP firmware support the headset jack? |
| 18:39.05 | Damin | "Does your VoIP service support fax?". Yeah.. about as well as Tin can and String do... |
| 18:39.38 | bkruse_home | DocHolliday: im almost positive |
| 18:39.39 | DocHolliday | thats what i hate.. its impossible to get a straight answer out of this channel :P |
| 18:39.48 | bkruse_home | some rumor its kind of hacked, but its worked fine for me |
| 18:40.01 | bkruse_home | hacked, because they want you to use MGCP and SKINNY |
| 18:40.05 | bkruse_home | but it doesnt matter :] |
| 18:40.09 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:40.23 | DocHolliday | im running 8.6 SIP and the phones are rock stable |
| 18:41.02 | bkruse_home | figured |
| 18:41.14 | bkruse_home | figured cisco's implementation would be pretty rockin |
| 18:41.20 | DocHolliday | i know for example if you want to use Cisco 7914 in conjunction with Cisco 7960 you need SCCP |
| 18:41.46 | DocHolliday | yeah! |
| 18:41.49 | bkruse_home | well |
| 18:41.56 | *** join/#asterisk netsurfer (n=bbjunkie@user-54446ab5.lns4-c10.dsl.pol.co.uk) |
| 18:41.58 | bkruse_home | for phone to phone and presense stuff, ya |
| 18:42.04 | bkruse_home | but for everything SIP related, no |
| 18:42.10 | ruied | what do I need to set the callerid? from the pstn? I have usecallerid=yes... what more do I need |
| 18:42.11 | Qwell | well, luckily, 7960+7914 work in asterisk ;) |
| 18:42.12 | bkruse_home | i see where your coming from though |
| 18:42.14 | bkruse_home | :D |
| 18:42.18 | Qwell | of course, if somebody were to *cough* send me one... |
| 18:42.29 | bkruse_home | ruied: you cannot, alot of telco's block the callerid field on a PSTN |
| 18:42.50 | DocHolliday | qwell, im suprised Digium wouldn't send you the stuff if you are willing to write the modules? |
| 18:42.51 | bkruse_home | callerid = above the user entry |
| 18:43.06 | bkruse_home | hahaha? |
| 18:43.22 | Qwell | no, *Cisco* should be sending me stuff |
| 18:43.23 | ruied | bkruse_home, going to check inserting my wireless phone in the line.... |
| 18:43.52 | Juggie | Qwell, you need a what, 7960? |
| 18:43.59 | bkruse_home | ruied: k |
| 18:44.04 | bkruse_home | oh, gotcha |
| 18:44.11 | Qwell | I actually asked Cisco once. They basically said "We don't give a shit.", then they went on to say that less than 1% of phones sold are used with something besides CCM |
| 18:44.15 | Qwell | which is complete BS |
| 18:44.15 | bkruse_home | Qwell: we could have jacked it from upstairs in the atrium |
| 18:44.19 | Qwell | Juggie: 7914 :) |
| 18:44.26 | Juggie | oh, no can do :) |
| 18:44.26 | Qwell | bkruse_home: heh |
| 18:44.33 | Qwell | Juggie: or 7985 :P |
| 18:44.35 | J4k3 | Qwell: thats how much cisco appreciates your business... find a real vendor. |
| 18:44.56 | J4k3 | cisco's day came and left long ago. |
| 18:45.24 | bkruse_home | J4k3: in VOIP, sure, kinda, but they still own a huge market involving networking as a whole |
| 18:45.30 | DocHolliday | Qwell, but i would think Digium would be greatful to have hardware support? |
| 18:45.42 | ruied | bkruse_home, you are right.... :( |
| 18:45.58 | bkruse_home | ruied: told ya, i had the same problem too, when i first started using asterisk :[ |
| 18:45.59 | Qwell | DocHolliday: Why? The phones support SIP too |
| 18:46.11 | bkruse_home | get a PRI or a provider that wil let u set the callerid field (few and far in between) |
| 18:46.17 | DocHolliday | Qwell, but is Cisco 7914 supported under SIP? |
| 18:46.23 | Qwell | nope |
| 18:46.25 | [TK]D-Fender | DocHolliday : Nope |
| 18:46.27 | J4k3 | Cisco's business plan seems to be wrapped around selling mostly-sleazy or mostly-completely-outdated crap for absolute sucker prices. |
| 18:46.35 | DocHolliday | which is my point.. |
| 18:46.42 | Opperior | It's all about the name now |
| 18:46.49 | [TK]D-Fender | J4k3 : They become significantly less "sucker" once you get a sales rep |
| 18:46.54 | DocHolliday | it would be good for digium's bottom line if products actually worked with asterisk |
| 18:46.56 | Qwell | DocHolliday: well, Digium *is* willing to write the support - or at least, I am |
| 18:47.01 | J4k3 | [TK]D-Fender: doesn't change the fact their stuff is sleazy. |
| 18:47.25 | [TK]D-Fender | Opperior : Lets say I wouldn't know Cisco's manufacturing quality. they ARE nice phones.... SIP could be a fair bit better and more stable though. |
| 18:47.29 | Qwell | DocHolliday: but it's silly to go and buy every piece of hardware out there |
| 18:47.38 | [TK]D-Fender | J4k3 : Just some of their policies |
| 18:47.45 | DocHolliday | Qwell, sure but thats a pretty popular piece of VoIP gear? |
| 18:47.46 | bkruse_home | [TK]D-Fender: agreed, but i think the lack of SIP support is for a reason :] |
| 18:47.54 | Qwell | 7914? not really |
| 18:47.58 | J4k3 | hell, cisco was still selling 68030-based routers for >$3k retail in 1999 (when the routers got EOL'd) |
| 18:48.06 | DocHolliday | atleast companies are guaranteed they can take all their cisco handsets and transition to asterisk |
| 18:48.09 | [TK]D-Fender | bkruse_home : And what... destabilize their private little empire?! oh noes! |
| 18:48.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:48.53 | DocHolliday | Qwell, *most* companies i have been to now use Cisco handsets.. and as a result they have *many* Cisco 7914 |
| 18:49.02 | J4k3 | In this world, one must know their ABCs... *A*nything *B*ut *C*isco. |
| 18:49.03 | J4k3 | ;) |
| 18:49.15 | Qwell | DocHolliday: well, it *IS* proprietary |
| 18:49.48 | DocHolliday | Qwell, i dont dispute that, but the fact of the matter is there are a lot of *but's* with asterisk.. this would be one less (in my mind anyway) |
| 18:49.53 | Juggie | cisco phones suck, NEXT! |
| 18:50.09 | DocHolliday | Juggie, tell that to all my customers who haven't had one dead Cisco IP Phone |
| 18:50.11 | mafkees | dont listen to all those angry people 7960. You are serving me great ;) |
| 18:50.16 | Qwell | DocHolliday: they *do* work with asterisk :) |
| 18:50.21 | Juggie | DocHolliday, the hardware isnt the problem |
| 18:50.28 | Juggie | the software is horrible |
| 18:50.31 | Opperior | Well, then, what would you use if not Cisco? Just Polycom? |
| 18:50.32 | [TK]D-Fender | Polycom > Cisco |
| 18:50.40 | DocHolliday | Juggie, SIP 8.6 is perfect |
| 18:50.43 | [TK]D-Fender | Polycom > ALL |
| 18:50.43 | J4k3 | anything you don't have to hack to make work > Cisco. |
| 18:50.47 | bkruse_home | [TK]D-Fender: agreed! |
| 18:50.56 | [TK]D-Fender | DocHolliday : Perfect? You mean full presence support, and SLA? |
| 18:51.03 | bkruse_home | [TK]D-Fender: i meant that by making their Sip firmware crap, forces you to use SKINNY/MGCP |
| 18:51.14 | DocHolliday | i stand behind whatever solution i offer |
| 18:51.18 | [TK]D-Fender | bkruse_home : Insidious isn't it? ;) |
| 18:51.28 | Juggie | DocHolliday, if you think ciscos sip FW is perfect, then you really need to try some other phones |
| 18:51.30 | bkruse_home | incredibly, yes |
| 18:51.31 | [TK]D-Fender | DocHolliday : All hail the Almighty Dollar! ;) |
| 18:51.36 | Juggie | because i can assure you, it is FAR FAR FAR from perfect. |
| 18:51.44 | Qwell | bkruse_home: they don't want people using mgcp either |
| 18:52.02 | [TK]D-Fender | Qwell : Ummm.. why OULD anybody use MGCP? |
| 18:52.08 | Qwell | [TK]D-Fender: I have *NO* idea |
| 18:52.10 | Opperior | I've looked at Polycoms, and it seems like they don't have enough programmable buttons |
| 18:52.13 | DocHolliday | Juggie, i have heard Polycom is a good phone... i simply use what i have experience with |
| 18:52.22 | Juggie | go go mitel :) |
| 18:52.24 | [TK]D-Fender | Opperior : Programmable to do what? |
| 18:52.30 | brookshire | opperior, how many do you need? |
| 18:52.31 | bkruse_home | Qwell: agreed. |
| 18:52.32 | mafkees | make coffee |
| 18:52.41 | [TK]D-Fender | Makenshi : My * used to do that for me... |
| 18:52.45 | [TK]D-Fender | (no joke) |
| 18:52.45 | DocHolliday | Qwell, hmm if the Cisco 7914 works with asterisk.. but it doesnt work with SIP? |
| 18:52.53 | Qwell | DocHolliday: right |
| 18:52.53 | Opperior | whatever you want. Quick access to system featues, speed dial, extention monitoring.. you name it |
| 18:52.54 | mafkees | [TK]D-Fender: I know. mine too |
| 18:52.57 | [TK]D-Fender | brookshire : I can arrange that :) |
| 18:53.01 | brookshire | if my coffee pot talked ruby :) |
| 18:53.02 | DocHolliday | Qwell, so how *does* it work? |
| 18:53.06 | Qwell | skinny |
| 18:53.15 | brookshire | brandon! |
| 18:53.19 | brookshire | file! |
| 18:53.20 | [TK]D-Fender | brookshire : I can arrange that too :) |
| 18:53.20 | DocHolliday | ahh okay so whatever phone you want to use cisco 7914 with you simply run sccp? |
| 18:53.21 | bkruse_home | :] |
| 18:53.23 | mafkees | oh yeah |
| 18:53.27 | mafkees | skinny runs fine |
| 18:53.28 | Opperior | I use a Snom 360, and sometimes I find 12 buttons aren't enough |
| 18:53.31 | brookshire | [TK]D-Fender: hot.. i need that |
| 18:53.32 | Qwell | DocHolliday: no, you run it with skinny |
| 18:53.34 | Qwell | sccp is dead |
| 18:53.39 | DocHolliday | hehe |
| 18:53.43 | Qwell | ^ my official comment |
| 18:53.44 | bkruse_home | agreed |
| 18:53.48 | [TK]D-Fender | brookshire : I get back with you on that shortly :) |
| 18:53.55 | DocHolliday | well thats not a terrible solution i guess.. but not optimal obviously |
| 18:54.02 | Qwell | You can go ahead and quote me on that too :) |
| 18:54.03 | brookshire | Opperior: the polycom phone i've got, has a webbrowser! |
| 18:54.11 | mafkees | sccp will die on some softbuttons, being member of a queue, and meetme |
| 18:54.12 | Opperior | so does my Snom |
| 18:54.16 | brookshire | so... there are like.. menus of buttons for other buttons |
| 18:54.17 | mafkees | oh, and it wont work with 1.4 |
| 18:54.21 | DocHolliday | Qwell, i mean if i have to run skinny on one phone..i can live with that |
| 18:54.28 | [TK]D-Fender | Opperior : Get sidecars for IP 601/650 |
| 18:54.41 | brookshire | [TK]D-Fender: then you could have like 50 buttons |
| 18:54.46 | Qwell | DocHolliday: of course, skinny doesn't support hints, so your 7914 will be mostly useless, but hey |
| 18:54.50 | Opperior | but I find users like to have fast access to buttons on the phone it'self, not have to dig though a web interface |
| 18:54.58 | mafkees | indeed |
| 18:55.04 | DocHolliday | Qwell LOL useless POS |
| 18:55.04 | mafkees | no hints, no speeddials |
| 18:55.06 | mafkees | lol |
| 18:55.10 | Opperior | yea, sidecars are nice, but an added expense |
| 18:55.12 | Qwell | you could have 30 line keys |
| 18:55.13 | brookshire | Opperior: you can program the buttons.. the web interface is just the method of delivery |
| 18:55.21 | Qwell | or, however many... what is it, 14? |
| 18:55.27 | [TK]D-Fender | Opperior : Good. Fast. Cheap. Pick two.... |
| 18:55.28 | mafkees | 14 it is |
| 18:55.36 | Qwell | yeah, that's right... |
| 18:55.42 | Opperior | I know, I've quoted that to others myself :) |
| 18:56.02 | [TK]D-Fender | Opperior : then add "Physician, heal thyself!" to the list ;) |
| 18:56.07 | mafkees | <PROTECTED> |
| 18:56.07 | mafkees | 1166 (btn++)->buttonDefinition = BT_CUST_LINESPEEDDIAL; |
| 18:56.10 | brookshire | Qwell: 14 * 3 + 6 |
| 18:56.11 | J4k3 | "good fast and cheap..." "define cheap" |
| 18:56.19 | Qwell | you know...I wonder |
| 18:56.35 | Qwell | is 2 7914s a limitation in CCM, or of the hardware itself? |
| 18:57.09 | brookshire | qwell: i blame gentoo |
| 18:57.10 | DocHolliday | Qwell so its back to using one polycom phone i guess |
| 18:57.35 | Qwell | DocHolliday: like I said - if I had one, I'd get it working. Convince Cisco to send me one :P |
| 18:57.47 | *** join/#asterisk razor (i=razor@rapwap.razor.dk) |
| 18:57.48 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 18:57.52 | DocHolliday | Qwell, if i sold enough of them i'd send you one ;) |
| 18:58.37 | razor | Is there a way to check if there is currently an active call on an extension (i need to return Busy in that case) |
| 18:59.21 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:59.35 | DocHolliday | qwell, but as it is its hard to even get an answer if Cisco 7941/61 works as well as 40/60 with asterisk |
| 18:59.55 | Qwell | they should work jsut fine |
| 19:00.19 | DocHolliday | yeah i dont like 'shoulds' in my life :P going to get a friend to lend me one next week hopefully |
| 19:01.02 | brookshire | DocHolliday: if you absolutely must know if it will work, send a phone to qwell :) |
| 19:01.10 | brookshire | problem solved :) |
| 19:01.18 | DocHolliday | brookshire, well i'll just get one for myself and test it out :P |
| 19:01.25 | brookshire | or you could do that |
| 19:01.36 | [TK]D-Fender | razor : What would you like to do the checking? |
| 19:02.49 | razor | [TK]D-Fender, sorry - i dont understand the question. I am relaying the call to another PBX using zaptel. I need to only let one call pass trough per extension. |
| 19:03.36 | [TK]D-Fender | razor : Ok, you need to know within your DIALPLN. in that case "show application chanisavail" |
| 19:04.04 | DocHolliday | qwell, apparently chan_sccp2 supports 7914? |
| 19:04.14 | Qwell | DocHolliday: but not asterisk 1.4 |
| 19:04.15 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 19:04.28 | DocHolliday | just 1.2? |
| 19:05.37 | DocHolliday | qwell, actually it does work on 1.4 according to this article |
| 19:06.01 | bkruse_home | i seriously want a cupcake |
| 19:06.07 | razor | [TK]D-Fender, looks like something i can use - thanks :) |
| 19:06.12 | DocHolliday | with sprinkles? |
| 19:06.16 | bkruse_home | hmm |
| 19:06.22 | bkruse_home | not really digging the sprinkles now |
| 19:06.49 | DocHolliday | it looks like a use rhas posted a diff file to patch 1.4 for sccp2 |
| 19:07.06 | Qwell | DocHolliday: $20 says it won't compile. |
| 19:07.29 | DocHolliday | i dont like betting with a person that knows more than me :) |
| 19:07.39 | Qwell | <Qwell> sccp is dead |
| 19:08.03 | mafkees | and it crashes a lot |
| 19:08.07 | [TK]D-Fender | I only "gamble" when the outcome is guranteed ;) |
| 19:08.09 | mafkees | IF you use it |
| 19:08.31 | Qwell | [TK]D-Fender: oddly enough, so do I ;) |
| 19:08.52 | mafkees | dont hit the gpickup button if you like your asterisk to keep running |
| 19:08.53 | mafkees | ;) |
| 19:08.57 | DocHolliday | im just being hopeful i guess |
| 19:09.17 | DocHolliday | qwell is there a T.38 ATA you recommend? |
| 19:09.30 | Qwell | nope |
| 19:09.31 | brookshire | Qwell: sure does help that you have the "inside knowledge" :) |
| 19:09.36 | Qwell | brookshire: :P |
| 19:09.52 | DocHolliday | heh |
| 19:10.40 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 19:21.25 | mafkees | Qwell: can I overwrite the softkey labels on the 7960 ? |
| 19:21.34 | *** part/#asterisk razor (i=razor@rapwap.razor.dk) |
| 19:21.35 | Qwell | overwrite? |
| 19:21.36 | mafkees | the 4 under the display |
| 19:21.42 | Qwell | oh, yeah, in code... |
| 19:22.25 | mafkees | found it |
| 19:22.26 | mafkees | nm |
| 19:22.33 | mafkees | gonna rename CFwdAll |
| 19:22.59 | mafkees | like the comments: it's the same |
| 19:23.18 | mafkees | actually, it should be handled different |
| 19:23.25 | mafkees | but that's something to do in the future |
| 19:23.39 | mafkees | food |
| 19:25.41 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:25.44 | PakiPenguin | hello everyone |
| 19:25.55 | PakiPenguin | can anyone suggest a good iax based ata? |
| 19:26.28 | bkruse_home | uh, the iaxy? |
| 19:26.39 | PakiPenguin | anything other then that :p |
| 19:26.47 | bkruse_home | ouch. |
| 19:27.22 | bkruse_home | not really |
| 19:27.39 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 19:34.16 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 19:34.24 | Carp1 | Is this a good phone: http://www.ipphone-warehouse.com/products/2200-11531-001.html?gclid=CN2Bz6zUuIoCFRw8gQodLltFPw |
| 19:35.15 | *** join/#asterisk redax (n=redax@r6.hu) |
| 19:35.17 | redax | hi |
| 19:36.18 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
| 19:36.25 | redax | is it possible to have AND/OR in a GotoIf() command |
| 19:37.30 | *** join/#asterisk _MDC_ (n=marcus@c-7cfde255.06-72-6c6b7013.cust.bredbandsbolaget.se) |
| 19:38.02 | [TK]D-Fender | redax : && |
| 19:38.09 | [TK]D-Fender | redax : || |
| 19:38.35 | [TK]D-Fender | Carp1 : Yes, excellent phone |
| 19:38.44 | Carp1 | Ok, thanks |
| 19:38.46 | _MDC_ | is there a way in queues to run a command (eg CURL) when the call is answered? We would like to get statistics on how long the customers are waiting in the que... |
| 19:38.50 | Carp1 | I am going to buy it I think. |
| 19:38.56 | redax | I read some docs where '|' mentioned, but not worked for me |
| 19:39.13 | PakiPenguin | will iaxy work , if i attach a fax machine with it and send faxes from it in this way faxmachine --> iaxy --> * --> zap |
| 19:39.14 | redax | ok, so like in lang C |
| 19:39.18 | *** join/#asterisk AF-Slash (n=AF-Slash@209-181-28-69.hlna.qwest.net) |
| 19:39.27 | [TK]D-Fender | _MDC_ : thats alrady in the queue_log file |
| 19:39.36 | Carp1 | does anyone use SellVoIP? |
| 19:40.13 | [TK]D-Fender | PakiPenguin : Don't get your hopes up. If I were you I'd forget the idea of fax over VoIP |
| 19:40.33 | _MDC_ | <[TK]D-Fender>: I use a homemade system for logging to a databas with additional information, do I have to use the log file? |
| 19:41.01 | [TK]D-Fender | _MDC_ : if its already there, why are you reinventing the wheel? |
| 19:41.12 | PakiPenguin | ah |
| 19:41.15 | [TK]D-Fender | _MDC_ : And you could always add a macro in your Dial tot he agent. |
| 19:42.17 | _MDC_ | <[TK]D-Fender>: using it together with our case system, but the macro will do it! thanks! |
| 19:43.10 | _MDC_ | <[TK]D-Fender>, btw, will it be the same unique id for the call that goes in to the que and the one that will get it answered? |
| 19:44.17 | *** join/#asterisk thekidrio (n=thekidri@24-205-76-13.dhcp.psdn.ca.charter.com) |
| 19:50.19 | [TK]D-Fender | _MDC_ : how to cheat : pass the Unique ID to the macro IN your dial statement. Worst case : mod the callerID before the call gets dumped in queue. |
| 19:50.23 | *** join/#asterisk mega (n=mega@217.201.159.224) |
| 19:51.40 | *** join/#asterisk vishmaker (n=vishmake@125.63.121.41) |
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| 19:54.12 | Carp1 | I am ordering from a Yahoo store and it doesnt tell me how much shipping costs |
| 19:54.14 | Carp1 | wtf |
| 19:54.23 | Carp1 | but it has 3 dofferent shipping options |
| 19:54.51 | _MDC_ | <[TK]D-Fender>: can i set the macro directly in queues.conf like member=SIP/s1|M^answer or do I have to do a LOCAL/bla ? |
| 19:55.32 | *** join/#asterisk ikaRus1 (i=none@80.179.36.48.static.012.net.il) |
| 19:55.37 | redax | [TK]D-Fender: seems like not double '&' and '|' |
| 19:55.41 | redax | just single |
| 19:56.02 | [TK]D-Fender | _MDC_ : Local |
| 19:56.13 | redax | anyway thanks |
| 19:56.16 | [TK]D-Fender | redax : Go read up on expression on the wiki |
| 19:56.19 | ikaRus1 | any pointers for asterisk scripting tutorials? |
| 19:56.24 | _MDC_ | <[TK]D-Fender>: thanks alot, will try |
| 19:56.29 | [TK]D-Fender | ikaRus1 : as in? |
| 19:56.43 | ikaRus1 | as in learning how to do it |
| 19:57.20 | [TK]D-Fender | ikaRus1 : What kind of scripting? |
| 19:57.46 | ikaRus1 | extentions programming for one |
| 19:58.18 | bkruse_home | look at ael2 |
| 19:58.29 | ikaRus1 | ?? |
| 19:58.39 | bkruse_home | let me link u |
| 19:58.47 | ikaRus1 | thanks |
| 19:59.04 | bkruse_home | http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
| 19:59.06 | bkruse_home | :] |
| 19:59.13 | bkruse_home | asterisk extension language i believe |
| 19:59.17 | ikaRus1 | thanks again. |
| 20:00.14 | ruied | bkruse hehe, about the callerid, I've called to my service provider and they will activate the callerid in 48 hours without costs... :) |
| 20:00.31 | DocHolliday | cant you just set it yourself? |
| 20:00.47 | bkruse_home | nice!!!! |
| 20:00.51 | bkruse_home | DocHolliday: no |
| 20:00.54 | bkruse_home | not on a pstn |
| 20:01.08 | bkruse_home | well, you can set it, but its dis-regarded |
| 20:01.12 | DocHolliday | bkruse_home, if you set the caller ID yourself.. on the call recipient's phone bill will it show the caller ID you set or whats been set by your provider? |
| 20:02.21 | bkruse_home | provider |
| 20:02.28 | bkruse_home | depends on what provider also |
| 20:02.38 | bkruse_home | usually, the local bells for residential do not let you set callerid |
| 20:02.53 | bkruse_home | but enterprise/business types, including PRI/t1 let you set callerid |
| 20:03.03 | DocHolliday | fair enough :) |
| 20:03.08 | bkruse_home | :P |
| 20:03.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:03.20 | DocHolliday | what about setting your phone number.. is that that the same deal? |
| 20:03.30 | bkruse_home | yep |
| 20:03.40 | x86_ | PRI lets you modify CID, but CAS does not |
| 20:03.44 | bkruse_home | all the same field |
| 20:03.50 | bkruse_home | x86correct |
| 20:04.23 | DocHolliday | yeah, but if you do set your number to one other than the actual number.. does the recipients bill show the real number? |
| 20:04.42 | bkruse_home | nope |
| 20:04.50 | bkruse_home | its called callerID spoofing |
| 20:05.10 | DocHolliday | right :) and there is no way for the telco to 'track a spoofed #' |
| 20:05.14 | bkruse_home | the reason for being able to do this is say you have 90 phones but only 1 PRI, but you only need 23 lines at all time |
| 20:05.30 | bkruse_home | each person can have a callerID set to THEIR number or exten, whatever, whatever |
| 20:05.39 | DocHolliday | yup! |
| 20:05.45 | bkruse_home | DocHolliday: hmm, probably not, but they could tell where it came from, i bet |
| 20:06.04 | DocHolliday | the exchange you mean? |
| 20:06.29 | bkruse_home | well |
| 20:06.30 | bkruse_home | no |
| 20:06.37 | bkruse_home | because technically, it is coming from that number |
| 20:06.40 | bkruse_home | its just a field thats set |
| 20:06.50 | bkruse_home | so they could say hey, it came from our pri line 1231 or whatever |
| 20:07.18 | DocHolliday | gotcha, for example i will be buying 1 DID but multiple channels so to speak |
| 20:08.15 | DocHolliday | i'll be setting CID to whatever extension is being dialed from |
| 20:08.15 | bkruse_home | you can do that, yes |
| 20:08.43 | DocHolliday | i dont know why more companies dont do that.. usually you call back and they have no idea who called you! |
| 20:09.45 | bkruse_home | true |
| 20:10.45 | DocHolliday | some VoIP providers seem to offer unlimited DIDs incoming, but i cant seem to get them to give me free local termination :P |
| 20:10.45 | *** join/#asterisk ManxPower (n=manxpowe@129.sub-75-200-130.myvzw.com) |
| 20:13.33 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
| 20:13.48 | ruied | How can I add 3 digits to a number, I've tried this: exten => _282xxxxxxxxx,2,Dial(SIP/351${EXTEN}@voipbuster) |
| 20:14.43 | *** join/#asterisk RoyK (n=roy@cEE71BF51.dhcp.bluecom.no) |
| 20:17.23 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 20:18.42 | ruied | forget, It is like I've wrote... I had tree more 'X' than I needed |
| 20:19.17 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:25.04 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 20:25.44 | *** part/#asterisk mega (n=mega@217.201.159.224) |
| 20:29.23 | J4k3 | hrm... |
| 20:29.44 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 20:29.59 | J4k3 | a transcoded call on my * box, after anywhere from 30 seconds to 5 minutes, I start getting regular 'static' in the call |
| 20:30.23 | J4k3 | it starts off slow, just some pops every second... then more pops, then complete dropouts in the audio, then its totally unusable |
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| 20:55.17 | mafkees | ello oej |
| 20:59.28 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 21:05.35 | k84 | How can i have two voipbuster accounts, and call out via the one which isn't busy automatically? |
| 21:06.56 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 21:16.02 | cervi | k84: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail |
| 21:18.23 | nettie | Hi guys, I would like to connect 2x bri isdn to my asterisk box and I'm in the market for a 2+ bri card. I saw the Digigum one and is very expansive, I also found some clones but in my opinion their price is not low enought to push me buy them and maybe deal with possible troubles. I find very interesting the isdn cards based on the HFC Cologne chipset, they cost almost nothing and *IF* they work this will be a terrific deal. Has anyone experiece using |
| 21:18.34 | *** join/#asterisk J4k3^ (i=jsuter@dhcp-12-197-128-45.intrastar.net) |
| 21:21.20 | cervi | nettie: What is the advantage of HFC? |
| 21:21.40 | cervi | nettie: I am using AVM. They are cheap and work |
| 21:21.41 | *** join/#asterisk RoyK (n=roy@ti211310a080-5551.bb.online.no) |
| 21:24.53 | nettie | cervi: well I read about hfc mostly, I wasnt sure about AVM |
| 21:25.04 | nettie | cervi can you tell me more please? |
| 21:25.31 | nettie | cervi are those easy to find as well? |
| 21:25.41 | nettie | ahhh |
| 21:25.49 | nettie | AVM fritzbox producres |
| 21:28.22 | cervi | nettie: AVM is a german company. As I am german, they are easy to buy :-) |
| 21:28.41 | cervi | I don't know if they are easy to buy in other countries |
| 21:28.53 | nettie | cervi: well I'll ebay.de them :) |
| 21:29.00 | cervi | Yes |
| 21:29.04 | cervi | good idea |
| 21:29.11 | nettie | cervi :http://www.avm.de/en/Produkte/FRITZCard/FRITZ_Card_PCI/index.html## |
| 21:29.17 | nettie | thats' the one? |
| 21:30.00 | nettie | is it fully supported and stable with asterisk? |
| 21:30.45 | cervi | you want not fritzcard (passive), you want active (AVM B1 Card) |
| 21:30.46 | cervi | http://www.avm.de/en/Produkte/Server-Produkte/B1_PCI/index.html |
| 21:31.26 | nettie | ahhh |
| 21:31.28 | nettie | ok |
| 21:31.31 | cervi | Yes, AVM is supported (together with chan_capi) |
| 21:31.33 | cervi | http://www.melware.org/ChanCapiHardware |
| 21:33.55 | *** join/#asterisk abrdeco (n=tbmjrf@c9114778.rjo.virtua.com.br) |
| 21:35.58 | nettie | ohh ok, great! thanx. Do you know if there might be some troubles with asterisk using 2 of them at the same time please? |
| 21:37.25 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 21:37.53 | cervi | yes, it should be possible but try to buy a 2 or 4 Port card instead |
| 21:37.57 | cervi | AVM C2 or AVM C4 |
| 21:38.33 | nettie | ohh |
| 21:39.22 | nettie | do you know how much an AVM C2 is please? |
| 21:40.22 | cervi | in germany it's est. 500 Euros |
| 21:40.31 | cervi | maybe better luck on ebay |
| 21:40.46 | nettie | yeah.. a digium 4 bri is 800ish |
| 21:41.24 | k84 | cervi, thanks |
| 21:41.31 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 21:41.58 | cervi | you could also try an other competitor |
| 21:41.59 | cervi | http://www.hstnet.de/english/products/isdn/saphir_3_ml_pci/index.asp |
| 21:42.07 | [[blah]asfd | is it possible, and if so can someone point me in the direction of how to install an asterisk server on a 2GB usb jump drive? |
| 21:42.19 | cervi | Today I bought a 4 port Card for 150 Euro on ebay |
| 21:42.34 | cervi | Unfortunately you have to ask for Linuxdrivers by email |
| 21:43.05 | cervi | [blah]asfd: Could be a problem |
| 21:43.13 | cervi | for example for /dev files |
| 21:43.35 | cervi | do you want to boot or "plug & start" ? |
| 21:43.47 | nettie | cervi hstnet has hfc |
| 21:43.50 | J4k3 | [[blah]asfd: It can be done (linux doesn't care what you install to really, as long as your BIOS can boot from it) |
| 21:43.54 | J4k3 | but the issue is longevity |
| 21:44.03 | J4k3 | I've personally had flash drives fail within 1 month due to excessive writes |
| 21:44.17 | [[blah]asfd | well, plug start would be ideal, but I know that would not be easy. |
| 21:44.33 | cervi | nettie: What is the advantage of HFC? |
| 21:44.54 | nettie | cervi it's just a different chipset |
| 21:45.08 | cervi | ok |
| 21:45.14 | nettie | cervi as far as I know. I just wanted to point it because the single channel hfc costs 15 euros |
| 21:45.15 | nettie | eheh |
| 21:45.42 | cervi | J4k3: You could try again but avoid write (disable log or use remote syslog) |
| 21:46.38 | *** part/#asterisk abrdeco (n=tbmjrf@c9114778.rjo.virtua.com.br) |
| 21:46.38 | nettie | that's the reason I was very amazed by the pricing and I started to worry about stability and compatibility |
| 21:46.40 | [[blah]asfd | what I REALLY would like is to build something like vonages usb phone they just came out wiht |
| 21:46.40 | [[blah]asfd | with |
| 21:47.04 | J4k3 | cervi: yeah. it *can* be done, if you avoid writes... but you start to lose usability. |
| 21:47.04 | [[blah]asfd | http://www.vonage.com/device.php?type=VPHONE |
| 21:47.17 | J4k3 | you can log to ram... but you lose the log upon reboot |
| 21:47.21 | J4k3 | same for voicemail, etc. |
| 21:47.40 | redax | is it possible to suppress the Manager messages from console like: ` == Manager 'admin' logged on from 127.0.0.1' |
| 21:47.57 | redax | only the manager messages that I dont want... |
| 21:48.16 | cervi | J4k3: You could set up an File/Storage Server, where you could write |
| 21:48.31 | cervi | the application itself still can run on other machines |
| 21:48.38 | cervi | you could use NFS or iSCSI |
| 21:49.11 | *** join/#asterisk Style-Z (n=xizm@adsl-75-41-220-102.dsl.irvnca.sbcglobal.net) |
| 21:49.29 | Style-Z | hello. i was wondering if anyone knew how to DROP calls from a certain area code? |
| 21:49.54 | Style-Z | i just want my pbx to hang up if you call from a certain area code.. not sure how to do this |
| 21:50.10 | Style-Z | im using asterisk 1.07 |
| 21:50.12 | [[blah]asfd | use gotoif and redirect those calls to a hangup() extension |
| 21:50.16 | Style-Z | thanks in advance |
| 21:50.46 | Style-Z | [[blah] i actually tried this and i must be doing something wrong |
| 21:50.53 | Style-Z | let me pastebin what i have |
| 21:50.57 | [[blah]asfd | ok |
| 21:51.52 | Style-Z | http://pastebin.ca/362525 |
| 21:53.09 | [[blah]asfd | try making 1000 an extension instead of part of the macro |
| 21:53.31 | Style-Z | oh ok... like hangup_on_cunts,s,1 |
| 21:53.41 | Style-Z | then have that just be a hangup within its own extenstion |
| 21:53.44 | Style-Z | heh ok |
| 21:53.55 | [[blah]asfd | yeh |
| 21:55.44 | *** join/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 21:56.52 | Style-Z | wierd. its like its not picking up the caller id |
| 21:57.19 | [[blah]asfd | what does your NoOp show? |
| 21:57.23 | Style-Z | i just tried calling in again (i listed my area code as one to hangup on) |
| 21:57.32 | Style-Z | how do i find that out? |
| 21:57.40 | IguanaNed | Asterisk n00b here... Just wondering if anyone has a good solution for Installing a Chat Server on my Asterisk box.. to enable people to chat while in conference call |
| 21:58.26 | [[blah]asfd | watch the cli for the noop output |
| 21:58.50 | Style-Z | yeah im in the CLI its not showing anything when i call in |
| 21:58.56 | Style-Z | which is wierd because it used to |
| 21:59.06 | Style-Z | maybe i got logging to display to cli turned off somehow |
| 22:01.50 | [[blah]asfd | set verbose 5 |
| 22:02.51 | Style-Z | aha |
| 22:02.53 | Style-Z | Verbosity was 0 and is now 5 |
| 22:03.34 | *** join/#asterisk mega (n=mega@217.201.132.73) |
| 22:03.52 | Style-Z | oh interesting |
| 22:04.02 | Style-Z | calling number shows with the 1 |
| 22:04.12 | Style-Z | calling name shows as the number without the 1 prefix |
| 22:04.23 | [[blah]asfd | dont use name |
| 22:04.27 | [TK]D-Fender | IguanaNed : "show application meetme" |
| 22:05.01 | Style-Z | [[blah can i msg you? |
| 22:05.07 | [[blah]asfd | sure |
| 22:07.40 | *** part/#asterisk mega (n=mega@217.201.132.73) |
| 22:08.58 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 22:11.17 | *** join/#asterisk foobar778 (i=johhny@ip68-100-41-120.dc.dc.cox.net) |
| 22:17.38 | IguanaNed | Fender... does metme have chat capabilities? |
| 22:17.59 | [TK]D-Fender | IguanaNed : it is a conference room. Go read up on it, |
| 22:18.47 | IguanaNed | Fender... "Your application(s) is (are) not registereF" |
| 22:19.05 | [TK]D-Fender | IguanaNed : You may only get that app if you installed Zaptel before * |
| 22:19.09 | IguanaNed | I have read up on meetme but it loks to only do voice |
| 22:19.23 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 22:19.28 | IguanaNed | does it do chat too? |
| 22:19.34 | JT | this is asterisk |
| 22:19.38 | JT | it's meant to do voice |
| 22:19.58 | IguanaNed | I got the voice part down I just want to be able to chat at the same time |
| 22:20.19 | [TK]D-Fender | IguanaNed : * does not do IM of any kind |
| 22:21.25 | ManxPower | IguanaNed: IF you want text chat then use a different system |
| 22:21.57 | Style-Z | install an ircd |
| 22:21.58 | Style-Z | haha |
| 22:22.00 | Style-Z | :P |
| 22:22.35 | IguanaNed | I amplanning on installeing ircd |
| 22:22.44 | *** join/#asterisk MarkWD (n=mwulf@cpe-66-25-236-7.gt.res.rr.com) |
| 22:22.53 | IguanaNed | i heard ircu is probably the most secure? |
| 22:23.01 | Style-Z | no idea/ |
| 22:23.09 | Style-Z | google will help with that. |
| 22:23.10 | mafkees | I use ratbox |
| 22:24.22 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917CC8.dip.t-dialin.net) |
| 22:25.13 | *** part/#asterisk s1gny (n=s1gny@p54917CC8.dip.t-dialin.net) |
| 22:28.01 | MarkWD | im having problems getting make to work with redhat es3 does anyone know a howto for that version ? |
| 22:35.25 | mmlj4 | MarkWD: are the tools installed? |
| 22:35.33 | mmlj4 | rpm -qa | grep gcc |
| 22:35.35 | mmlj4 | rpm -qa | grep make |
| 22:35.53 | *** part/#asterisk Style-Z (n=xizm@adsl-75-41-220-102.dsl.irvnca.sbcglobal.net) |
| 22:35.57 | mmlj4 | ...etc.? |
| 22:38.07 | MarkWD | yes |
| 22:38.14 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:38.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:41.20 | MarkWD | gcc-3.2.3-20 |
| 22:42.00 | MarkWD | and make-3.79.1-17 |
| 22:43.45 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:43.50 | *** join/#asterisk fireshade (n=cpalm@c-75-72-127-191.hsd1.mn.comcast.net) |
| 22:44.30 | MarkWD | have you heard of problems with that version of make ? |
| 22:46.21 | fireshade | Greetz: question - I have an iaxy S101i connected to my asterisk backend, the backend has a digium TDM400P card in it. Now, everything works great when the iaxy is provisioned for ULAW, however when provisioned for ADPCM the iaxy registers with the backend, but incomming FXO calls to the IAXy just ring and ring - while from the IAXY, I HEAR the FXO party... any ideas? |
| 22:47.10 | fireshade | iaxy firmware is version 23 |
| 22:48.46 | fireshade | I'm anxious to solve this problem, since the ADPCM codec saves me approximately 40 kb/s vs ULAW (G.711) |
| 22:49.07 | JT | i assume it's going over the Internet :) |
| 22:49.22 | fireshade | JT: correct. |
| 22:49.46 | JT | not many people use the iaxy, not sure how much of a response you'll get |
| 22:50.12 | fireshade | JT: It's so strange, I dial into the FXO line from my cell phone, the IAXy connected phone rings, I pick it up, I HEAR the party on the cell phone.. but the cell phone just keeps ringing |
| 22:50.35 | fetcher | fireshade: maybe try specifying G.726 instead of ADPCM? |
| 22:50.48 | fetcher | they're basically the same codec. Not sure why Asterisk has both... |
| 22:51.00 | fireshade | fetcher: you mean in the iax.conf file? |
| 22:51.37 | fetcher | fireshade: yup. And on the IAXy if it has its own preferred-codec setting |
| 22:52.26 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:52.33 | fireshade | fetcher: I'm currently provisioning the iaxy from the windows tool.. mabey that's a problem, I'll try from the shell tool in linux instead.. thanks. |
| 22:59.30 | *** join/#asterisk pifiu (n=someone@c-65-34-152-249.hsd1.fl.comcast.net) |
| 23:00.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:06.26 | *** join/#asterisk kfiction (n=fiction@ppp4752.dsl.pacific.net.au) |
| 23:06.47 | kfiction | hi there - just wondering what people think of Snom? |
| 23:14.52 | fireshade | Unfortunately, the allow=g726 command didn't work at all for the iaxy.. appears to need allow=adpcm, still can't seem to get the incomming FXO line to stop ringing on dial to the iaxy's extension.. it's so odd that I can hear the caller through the iaxy tho, tho whole time they hear ringing.. is there any zapata.conf settings for "call pickup detection"??? |
| 23:17.01 | JT | that's pretty freaky |
| 23:17.05 | JT | analogue line? |
| 23:17.50 | kfiction | anyone here has had any experience with Snom phones? |
| 23:18.59 | fireshade | JT: yep.. it's an analog line connected to one of two FXO ports on the TDM400P card in the asterisk backend. Things appear to work just fine when iaxy is provisioned for ULAW. |
| 23:19.22 | JT | so where's the ringing again? |
| 23:21.05 | fireshade | JT: I hear continuous ringing from a call into the analog line (FXO) when the extension of the iaxy is dialed.. but from the iaxy, the connected phone rings, I pick it up, and I can hear anything said from the FXO incomming phone, while they hear only ringing... |
| 23:21.58 | JT | ok, so you get a ringing tone in your earpiece, but no phone bells are still ringing or anything? |
| 23:22.17 | *** join/#asterisk ronfox (n=rfoxirc@rrcs-66-91-128-34.west.biz.rr.com) |
| 23:23.05 | fireshade | JT: right... the phone connected to the iaxy initially rings, I answer it.. and can hear anything said from the outside phone, but the outside phone just keeps hearing ringing (through the phone) |
| 23:23.44 | fireshade | JT: It's like the zap channel doesn't recognize that the iaxy answered |
| 23:23.57 | JT | you receive a ringing indication, is a nice way to put it :) |
| 23:24.02 | JT | does the console show much? |
| 23:24.25 | fireshade | JT: Let me try again with the console up... |
| 23:24.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:29.29 | fireshade | jt: here's what I see... |
| 23:29.31 | fireshade | <PROTECTED> |
| 23:29.31 | fireshade | <PROTECTED> |
| 23:29.31 | fireshade | <PROTECTED> |
| 23:29.31 | fireshade | <PROTECTED> |
| 23:29.31 | fireshade | <PROTECTED> |
| 23:29.32 | fireshade | <PROTECTED> |
| 23:29.34 | JT | ARGH |
| 23:29.35 | JT | NO |
| 23:29.36 | JT | stop |
| 23:29.40 | JT | ~pb |
| 23:29.41 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:29.58 | JT | there are at least 300 users here, it is not fair to flood them all |
| 23:30.04 | fireshade | JT: oops... sorry |
| 23:30.31 | JT | well at first glance that looks ok |
| 23:30.59 | JT | i'd be guessing the ringing indication is provided by asterisk |
| 23:31.47 | putzz | anyone have the pdf with PRI trouble codes? |
| 23:32.11 | JT | no, but you can google them |
| 23:32.12 | fireshade | JT: right... it shows that IAX2/1/3 answered the zap channel. I'd agree with that too.. that the ringing indication is being provided by the backend |
| 23:32.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:32.50 | JT | fireshade: when you ring with your cellphone, does asterisk answer and then provide a different ringtone, can you hear when it switches over? |
| 23:33.12 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 23:33.59 | *** join/#asterisk ruied (n=ruied@bl7-214-130.dsl.telepac.pt) |
| 23:36.14 | fireshade | JT: Yes, the asterisk server answers and provides my recorded dialplan.. when I press "1" and the backend routes to the iaxy - the iaxy rings, I pickup and can hear the cell phone but the ringing indication continues to the cell phone |
| 23:37.09 | fireshade | That's odd, considering that the cli shows IAX2/1/3 answered Zap/3-1 |
| 23:37.16 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 23:37.57 | JT | ok, umm pastebin your extensions.comf |
| 23:40.35 | fireshade | JT: ok - pastebin refernence 362658 |
| 23:41.31 | fireshade | hang one sec.. that was iax.conf.. pastebin of extensions.conf comming |
| 23:42.05 | JT | be nice to send the url next time :P |
| 23:42.41 | fireshade | JT: http://pastebin.ca/362660 |
| 23:43.38 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 23:44.13 | JT | iax_additional what the hell, don't tell me it's freepbx |
| 23:44.53 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 23:44.53 | fireshade | JT: worse than that - it's asterisk@home v1.5 |
| 23:44.57 | JT | arrgh |
| 23:45.08 | JT | so it's like 5 centuries old too |
| 23:45.13 | JT | you should upgrade :P |
| 23:45.37 | fireshade | JT: you're absolutely right.. :) I think I'm probably fighting old code |
| 23:46.18 | JT | i don't even know if the revant stuff is in extensions.conf or if it's included |
| 23:46.21 | JT | it's just a mess |
| 23:46.27 | JT | the latest Asterisk 1.2.x is the go |
| 23:46.58 | fireshade | JT: right-o.. I'll read up, switch over, and give a try from that distribution.. thanks for the patience up to this point JT |
| 23:48.28 | *** join/#asterisk tydelCA (i=tydel@216.19.189.153.novuscom.net) |