| 00:01.58 | Katty | so quiet tonight.. |
| 00:05.39 | *** join/#asterisk ghenry (n=ghenry@85-189-244-101.daisydsl.managedbroadband.co.uk) |
| 00:10.33 | *** join/#asterisk Magicianx (n=chezvous@76.10.173.93) |
| 00:14.10 | bjingles | hm |
| 00:14.51 | bjingles | This may be an oddball question but in the polycom .cfg files where do I find the one to edit that will give me ext.cfg that macaddress.cfg pulls from |
| 00:15.15 | bjingles | it says do not edit phone1.cfg but the top part of it looks like the part I have to edit for the phone to take an extension |
| 00:17.31 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 00:18.14 | grandpapadot | hi all. In sip.conf, after disallow=all, the allow=g729,gsm,ulaw line represents the preferred order. For some reason, in 1.2.24, I have to specify allow=g729 only otherwise it will fall back to ulaw, any ideas? |
| 00:18.24 | Katty | yawn. |
| 00:19.52 | JT | grandpapadot: the line does not represent the preferred order, it represents allowable codecs |
| 00:21.04 | grandpapadot | JT: How does one set the order? |
| 00:21.06 | *** join/#asterisk Yourname` (i=Myztic@unaffiliated/yourname/x-837320) |
| 00:21.12 | grandpapadot | And thanks for the help. |
| 00:21.31 | JT | grandpapadot: you don't |
| 00:21.31 | Yourname` | Happy new year errrbody! |
| 00:21.34 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 00:21.35 | JT | asterisk can |
| 00:21.38 | JT | asterisk can't do it |
| 00:21.41 | grandpapadot | So I can't specify use g729 over ulaw? |
| 00:21.45 | JT | no |
| 00:21.57 | JT | you can specify one instead of the other |
| 00:21.59 | JT | but not both |
| 00:22.01 | grandpapadot | Hrm... |
| 00:22.03 | grandpapadot | Ok, thanks. |
| 00:22.52 | bllzm | what will show up in asterisk callerid if it's unknown or number is blocked, is there one universal way to recognize these or does it vary from operator to operator ? |
| 00:24.13 | ManxPower | bllzm: what shows up in Asterisk and what shows up on the phone are two different things |
| 00:24.31 | ManxPower | Blocked depends on the telco, unknown is an empty callerid |
| 00:25.15 | bllzm | ManxPower, I need to identify ( on asterisk system ) blocked caller id and respond accordingly |
| 00:25.43 | bllzm | not concerned with phone |
| 00:26.14 | ManxPower | bllzm: Are you using a PRI? |
| 00:28.21 | bllzm | ManxPower, yes |
| 00:28.34 | bllzm | ManxPower, IAX trunk, other end is PRI |
| 00:28.47 | bjingles | Anyone here knows anything about polycom config files? |
| 00:29.24 | JT | bllzm: so not really a pri |
| 00:29.24 | Katty | they fit nicely into the shredder. |
| 00:29.28 | Katty | i know that (= |
| 00:30.06 | ManxPower | bllzm: IF you have a PRI you should be able to get the status if the Caller Presentation (see channelvariables.txt in the asterisk source code /doc direcotry) |
| 00:31.26 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:31.40 | bllzm | ManxPower, does this apply even if I don't have access to PRI but only IAX trunk on the other end ? |
| 00:32.14 | ManxPower | bllzm: I have no idea. That is what I asked if you had a PRI. |
| 00:40.25 | bjingles | the whitepaper for setting up the phone1.cfg file and sip.cfg from polycom states that you need to set up macaddress-user.cfg and randomsip.cfg and then it proceeds to give examples of how they "should" look but gives you no clue as to how to make them |
| 00:40.34 | bjingles | how are you people making your .cfg files for provisioning? |
| 00:40.54 | mosty | bjingles, use a text editor |
| 00:41.07 | mosty | or write a script to generate them from a database with your settings, or something |
| 00:41.13 | bjingles | and just rip directly from the examples given in the whitepaper? |
| 00:41.27 | bjingles | IE manually type the text into a txt editor and name the files? |
| 00:41.27 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:42.03 | mosty | bjingles, copy+paste from the examples if you don't know how else to get started |
| 00:42.55 | dacs | quiestion on Unique identifier, in sip.conf e.g. [400], i know i can change it to whatever name i want e.g [DACSIP], but how i would assign it the 400 exten? |
| 00:43.46 | _ShrikE | exten => 400,1,Dial(Sip/DACSIP) |
| 00:43.54 | mosty | dacs: just name it whatever you want in sip.conf, extensions are configured in extensions.conf |
| 00:43.57 | bjingles | Mosty I did that and it fried my phone when I provisioned it |
| 00:44.25 | mosty | bjingles, probably have the wrong config for your firmware version, or incomplete config or something |
| 00:44.35 | bjingles | hm |
| 00:44.35 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3c12c2ebf95024e5) |
| 00:44.35 | dacs | _ShrikE: mosty aha, i was reading the book and i just got a little confused , thank you |
| 00:45.14 | bjingles | I think that the whitepaper misses something |
| 00:45.17 | mosty | bjingles, look in your server's logs to see if the phone is downloading all the files |
| 00:45.30 | bjingles | it did |
| 00:45.33 | bjingles | and it updated |
| 00:45.42 | bjingles | when I put all the stock files in the folder |
| 00:46.03 | bjingles | and now that I want to edit them and provision them to certain extensions based on mac addresses the whitepaper misses a part I think |
| 00:46.39 | mosty | bjingles, there is a config file with the mac address as part of the filename, you edit that with the details specific to that extension |
| 00:46.55 | bjingles | yes |
| 00:46.57 | bjingles | that was created |
| 00:47.15 | bjingles | and then put that in the sip.cfg "CONFIG_FILE" with phone1.cfg and sip.cfg? |
| 00:47.45 | mosty | no. the phone will try to download the file named with its mac address |
| 00:47.54 | dacs | talk to you later guys |
| 00:47.57 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 00:48.30 | bjingles | <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone1.cfg, sip.cfg" MISC_FILES="" |
| 00:48.30 | bjingles | LOG_FILE_DIRECTORY="/log" OVERRIDES_DIRECTORY="/overrides" CONTACTS_DIRECTORY="/contacts"/> |
| 00:48.40 | bjingles | that's what's in the file |
| 00:48.50 | mosty | don't paste here, use pastebin.com if you want to paste the contenfs of your config files |
| 00:49.01 | bjingles | it was two lines so I figured it was alright |
| 00:49.18 | bjingles | so I should put the ext.cfg in CONFIG_FILES |
| 00:49.27 | bjingles | along with phone1.cfg and sip.cfg |
| 00:49.27 | mosty | yes |
| 00:49.32 | bjingles | how is ext.cfg generated |
| 00:49.34 | bjingles | this is where I'm stuck |
| 00:49.44 | bjingles | the whitepaper assumes you already have it |
| 00:50.15 | bjingles | I have the extension already set up on the asterisk server |
| 00:50.29 | mosty | there is no standard program that generates polycom config files for you |
| 00:50.45 | mosty | you have to write them yourself, or write a program/script to do it for you |
| 00:51.16 | bjingles | so to write one myself I'd use a text editor and call it ext420.cfg |
| 00:51.25 | bjingles | and then write what? this is what the whitepaper doesn't explain |
| 00:51.29 | bjingles | it just assumes you already know |
| 00:51.32 | mosty | yes, start by copy+pasting an example |
| 00:51.40 | bjingles | ok |
| 00:51.46 | bjingles | but that fried my phone |
| 00:52.20 | mosty | then you made some sort of error |
| 00:52.28 | bjingles | it does say use an XML editor |
| 00:52.31 | mosty | what does the phone say? |
| 00:52.32 | bjingles | I used pico |
| 00:52.42 | mosty | http://www.voip-info.org/wiki/view/Polycom+Phones |
| 00:52.43 | bjingles | on boot it says Misc fille error |
| 00:52.47 | bjingles | error is 0x20 |
| 00:53.23 | mosty | you obviously have some sort of error in your file. btw that wiki page has a link to a provisioning util that might help you: http://www.wintrisk.com/ppt.html |
| 00:53.29 | bjingles | lol that's the guide I'm going off |
| 00:55.54 | ManxPower | As I said, check your syntaxz |
| 00:56.09 | ManxPower | bjingles: chances are pico wordwrapped |
| 00:56.17 | bjingles | yes |
| 00:56.17 | fujin | ugh |
| 00:56.19 | fujin | apt-get remove pico |
| 00:56.20 | bjingles | this is what I think happened |
| 00:56.22 | fujin | apt-get install vim |
| 00:56.27 | bjingles | apt-get? |
| 00:56.37 | Katty | vim :< |
| 00:56.39 | bjingles | oh yah lol |
| 00:56.43 | ManxPower | Personally I use jedit for stuff like that, where XML or source code is involved |
| 00:56.44 | Katty | emacs :> |
| 00:56.48 | fujin | viiiiiiiiiiiiiiim |
| 00:56.50 | bjingles | I'm using open office right now |
| 00:57.09 | ManxPower | bjingles: I see you are trying to use EVERY tool that you should not use |
| 00:57.20 | bjingles | sure |
| 00:57.23 | bjingles | so vim |
| 00:57.30 | fujin | vim! |
| 00:57.41 | fujin | it's the *only* _real_ editor. |
| 00:57.46 | bjingles | well it'll always be there |
| 00:57.48 | drmessano | I edit XML with MS Word 2000 |
| 00:57.50 | drmessano | Its cool |
| 00:57.54 | drmessano | j/k |
| 00:58.03 | *** part/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 00:58.26 | hmmhesays | I can't find documentation for directory.xml on polycom phones, |
| 00:58.44 | ManxPower | hmmhesays: It's not in the admin guide? |
| 01:00.11 | hmmhesays | Hey Katty: long time no talk |
| 01:00.19 | hmmhesays | No its not in the admin guide |
| 01:00.34 | hmmhesays | I think i figured out all the params from the options displayed on the phone itself |
| 01:02.14 | bjingles | nano isn't xml no |
| 01:02.21 | bjingles | upgraded pico to nano |
| 01:03.33 | hmmhesays | they also make no mention of being able to map the expansion module keys |
| 01:04.39 | ManxPower | hmmhesays: weird, the diretory xml format is documented in section 3.1.17.1 of my admin manual |
| 01:05.26 | ManxPower | 3.1.17.1 Local Contact Directory File Format |
| 01:05.53 | *** join/#asterisk ddunavant (n=David@68-244-143-208.area3.spcsdns.net) |
| 01:06.03 | hmmhesays | hold let me look again |
| 01:06.46 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:06.52 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 01:07.19 | hmmhesays | page 4-10 yeah but it does list what each tag is |
| 01:08.48 | *** join/#asterisk CCFL_Man2 (i=7163b500@pool-70-105-211-208.scr.east.verizon.net) |
| 01:09.10 | hmmhesays | yeah I'm just having problems today |
| 01:09.23 | hmmhesays | attended transfers don't set any transfer variables that I can use in the dialplan |
| 01:09.43 | ManxPower | hmmhesays: What crappy admin guide do you have? Mine lists each option |
| 01:10.03 | ManxPower | fn UTF-8 encoded string of up to |
| 01:10.03 | ManxPower | 40 bytes |
| 01:10.04 | ManxPower | a |
| 01:10.04 | ManxPower | first name |
| 01:10.31 | bjingles | hm ok some progress |
| 01:10.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:10.47 | bjingles | I was able to assign the extension to the phone I was working on but it's still saying "Line not registered" |
| 01:11.12 | bjingles | after pasting the example and going from there, I believe it may be because I didn't put the secret in and I don't know the syntax to doing that |
| 01:11.19 | bjingles | *to do that |
| 01:11.45 | hmmhesays | nevermind I found it |
| 01:11.49 | hmmhesays | I'm just having a bad day here |
| 01:12.12 | hmmhesays | ManxPower: have you ever seen anything that will default speed dial transfers to blind? |
| 01:12.43 | bjingles | <PROTECTED> |
| 01:13.30 | mosty | bjingles, well that is obviously where you put your first pet's name and the name of the first street you lived on, so that your phone knows your porn name |
| 01:14.22 | hmmhesays | reg.1.auth.userId="498" |
| 01:14.22 | hmmhesays | reg.1.auth.password="12345" |
| 01:14.49 | bjingles | ok thanks |
| 01:15.02 | ManxPower | bjingles: See http://www.fnords.org/~eric/polycom-config-examples/ |
| 01:15.04 | bjingles | I had to manually add that let's see if it takes |
| 01:15.09 | bjingles | ok thanks max |
| 01:15.13 | bjingles | *manx |
| 01:15.21 | ManxPower | hmmhesays: There are some options for default transfer type in 2.x and later |
| 01:15.47 | ManxPower | mosty: I thought that was your Drag Name |
| 01:16.17 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 01:16.20 | hmmhesays | Yeah ManxPower: I've been looking for them the one I have found is for ip 320/330's |
| 01:16.24 | franck | Hi all |
| 01:17.07 | franck | I'm looking at my calls using freepbx and the max use of zap is 3 while I have 8 channels. Something limiting that? |
| 01:17.07 | ManxPower | hmmhesays: I don't think that option is phone specific |
| 01:17.17 | ManxPower | ~freepbx |
| 01:17.18 | jbot | [~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 |
| 01:17.22 | ManxPower | ~zeeek |
| 01:17.23 | jbot | i heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 01:17.36 | franck | hehehe |
| 01:18.11 | franck | there is no special stuff in asterisk that limits the number of zap used? |
| 01:18.28 | hmmhesays | call.transfer.blindPreferred <-- is tha the parameter you speak of? |
| 01:18.44 | bjingles | Hm, epic fail, it took the name and secret but still says the line not registered |
| 01:18.50 | ManxPower | hmmhesays: I have no idea what it is, I set it 4 years ago and never dealt with it again |
| 01:19.02 | bjingles | which is of course why I started this provisioning server |
| 01:19.12 | hmmhesays | thats the only parameter I can find in the manual and it says it is specific to ip 320/330's |
| 01:19.37 | hmmhesays | so your default when you hit transfer key then a speed dial key is a blind transfer? |
| 01:20.35 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 01:21.25 | *** join/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:21.46 | bjingles | What I don't understand about these polycom phones is why they have to register to work and what the benefits are and why you can't turn it off |
| 01:22.00 | bjingles | the sipura phones will work no problem with this asterisk server |
| 01:22.26 | ManxPower | bjingles: they do not have to register to work. |
| 01:22.34 | bjingles | no? |
| 01:22.37 | ManxPower | nope. |
| 01:22.41 | bjingles | really? |
| 01:22.54 | ManxPower | just like every other SIP phone out there. |
| 01:23.12 | *** join/#asterisk tengulre (n=tengulre@124.42.50.54) |
| 01:23.13 | bjingles | it's odd because you try calling other extensions and nothing, you try calling out and nothing |
| 01:23.27 | hmmhesays | <PROTECTED> |
| 01:23.31 | bjingles | the sipura phones work with all the same settings |
| 01:23.47 | hmmhesays | ManxPower: when you hit transfer then speed dial button it is a blind transfer on your poly? |
| 01:27.34 | *** part/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:28.32 | *** join/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:28.47 | *** part/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:28.48 | ManxPower | hmmhesays: I am 20 miles from the nearest polycom phone. |
| 01:29.08 | *** join/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:29.26 | hmmhesays | ok |
| 01:29.51 | hmmhesays | I've looked through the admin guide to find a parameter that would produce that behavior |
| 01:30.05 | *** part/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:30.09 | hmmhesays | cause default is to create an attended transfer when you hit transfer-->speedial |
| 01:30.14 | hmmhesays | *speed dial even |
| 01:30.25 | *** join/#asterisk jamincollins (n=jcollins@asgardsrealm.net) |
| 01:30.37 | *** join/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 01:31.48 | jamincollins | is it normal for Asterisk Realtime to query each extensions context and priority 3 times? |
| 01:31.51 | ManxPower | I take it your users are too stupid to press "BLIND" during a transfer? |
| 01:33.37 | hmmhesays | Its not a stupid user problem, it is a trying to get as close to key system functionality as possible |
| 01:33.57 | ManxPower | *shrug* You are destined to fail. Asterisk is not a key system. |
| 01:34.08 | hmmhesays | not to mention when you use an attended transfer asterisk doesn't set any variables that I can use to trigger certain actions |
| 01:34.33 | hmmhesays | You I know you are one of those people that say "Just make your users change" but that is a shitty selling point, period. |
| 01:34.39 | bjingles | destined to fail with Asterisk lol |
| 01:34.43 | ManxPower | hmmhesays: Chances are An attended transfer is really a three-way call where one leg drops off |
| 01:34.43 | bjingles | I should tell that to my boss |
| 01:35.11 | hmmhesays | Asterisk debug says gives some indication there was a transfer, but that is useless in the dialplan |
| 01:35.23 | hmmhesays | If I can make the poly default to blind then all is well |
| 01:35.23 | bjingles | we've put so many man hours into this sytem we could have purchased 3 telco-run systems by now |
| 01:35.31 | ManxPower | bjingles: If you try to make lemonade out of a chicken you are also destined to fail. A chicken is not a lemon. Asterisk is not a key system. |
| 01:35.47 | bjingles | yeah there's some cliche I wish my boss would understand |
| 01:35.56 | hmmhesays | but people want basic key system functionality, of which you can get most of with poly's and asterisk |
| 01:36.15 | hmmhesays | people want blinky lights and line appearances, it makes them feel better |
| 01:36.19 | Qwell | ManxPower: fresh squeezed chicken...sounds delicious |
| 01:36.26 | ManxPower | We have something like 5 Asterisk systems in production, 3 of them for at least a year. Two more will be installed in the next 30 days. |
| 01:36.35 | bjingles | either way we'll keep dumping man hours into this heap and hope when it breaks we have the money for something else |
| 01:37.01 | bjingles | yeah we run 3 systems in 2 offices and 40 phones |
| 01:37.03 | ManxPower | bjingles: we tested Asterisk for about 18 months before putting the first system into production. |
| 01:37.10 | bjingles | wow |
| 01:37.25 | jamincollins | our primary system is pushing a year and half... no problems with it |
| 01:37.40 | bjingles | with the newly discovered incompetence of the new sys admin the asterisk program has been dumped on me |
| 01:37.52 | mosty | bjingles, if you don't know what you're doing you should pay someone who does, or get one of the "toy" systems like trixbox |
| 01:37.56 | hmmhesays | I will be moving away from asterisk when something better comes along |
| 01:37.57 | Qwell | why is your sys admin handling your telephony? |
| 01:38.03 | bjingles | yes agreed |
| 01:38.06 | hmmhesays | something that doesn't have a broken sip stack |
| 01:38.07 | bjingles | we should pay someone |
| 01:38.33 | bjingles | until we realize it's a sinking ship I need to get this figured out |
| 01:38.39 | ManxPower | Strangely, we have the telecom admin manage the Asterisk systems. |
| 01:38.55 | bjingles | and right now I need to understand why these polycoms won't register after I've provisioned them |
| 01:39.08 | bjingles | which is leading me to the sip.cfg |
| 01:39.20 | jamincollins | some companies strangely think that since it runs on PC hardware it's IT's job to manage it... bad idea |
| 01:39.29 | bjingles | jamin - you're correct |
| 01:39.32 | ManxPower | bjingles: you have the link to my working example config files. |
| 01:39.39 | bjingles | manx yeah I've read them |
| 01:39.45 | bjingles | is that bootrom 4.0.0? |
| 01:39.57 | jamincollins | bjingles: I know... I've been doing PC based phone system installation and support for nearly a decade |
| 01:40.11 | ManxPower | bjingles: none of thse boot files are for a bootrom. The job of the bootrom is just to load the SIP stack |
| 01:41.14 | bjingles | ah yes sorry |
| 01:41.35 | Qwell | when I design a phone, I'm going to make the IP stack load the bootrom |
| 01:41.35 | ManxPower | I'm sure the sip.cfg and phone1.cfg files are for older 2.x sip firmware. |
| 01:41.48 | ManxPower | Qwell: PXE |
| 01:42.07 | Qwell | ahh, of course |
| 01:42.40 | Qwell | I need to design a phone that actually runs asterisk |
| 01:42.47 | jamincollins | bjingles: you could check a sip debug of the phone's IP... check to see that it's even attempting a registration... if it is, check the message |
| 01:43.01 | hmmhesays | I use a wrtsl54gs as a asterisk phone for awhile |
| 01:43.03 | jamincollins | asterisk or IAX? |
| 01:43.08 | Qwell | asterisk |
| 01:43.15 | Qwell | iax on a phone is pretty dumb |
| 01:43.23 | hmmhesays | usb port with a usb sound card and chan_oss |
| 01:43.36 | jamincollins | asterisk on a phone seems serious overkill |
| 01:44.46 | jamincollins | so, no takers on whether it's normal for realtime to query each extensions entry three times? |
| 01:45.36 | hmmhesays | it is |
| 01:45.43 | mosty | jamincollins, no idea, realtime is messy code afaik |
| 01:45.59 | ManxPower | Realtime scares the hell out of me. |
| 01:46.17 | jamincollins | seems to be working well enough so far... just hitting the DB a LOT |
| 01:46.32 | ManxPower | Especially when we moved all the e-mail users into an MySQL database that was stored on a harddrive that failed 30 days later. |
| 01:46.34 | Qwell | realtime extensions is pretty pointless imo |
| 01:47.29 | jamincollins | to each their own... customers don't tend to like editing configuration files |
| 01:47.48 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 01:47.59 | Qwell | well, with something like func_odbc, it makes that point moot |
| 01:48.01 | ManxPower | So don't make them. 8-) |
| 01:48.24 | Qwell | besides, you're still editing a config file, just in a different way |
| 01:48.29 | jamincollins | I don't make them... but if they want to pay me to give them a web interface to their call flow... more power to them |
| 01:48.51 | drmessano | Qwell: IAX on a phone is a good idea when you have an 8 line phone with 8 active calls :) |
| 01:48.52 | ManxPower | you can have a web interface to text configs -- as every sissy asterisk gui has proven |
| 01:48.53 | drmessano | Oh, wait |
| 01:48.57 | drmessano | lol |
| 01:49.41 | *** part/#asterisk jamincollins (n=jcollins@asgardsrealm.net) |
| 01:49.58 | drmessano | Its shocking how many people want an IAX phone.. My guess is "IAX = ZOMG No NAT problems" for those who don't get fixing SIP |
| 01:50.16 | drmessano | But non-sensical, of course |
| 01:50.33 | ManxPower | Every time a customer asks me to build a web interface to Asterisk I say "give me a list of things you want it to be able to do", and the customer never does that. |
| 01:51.04 | ManxPower | It's pretty obvious they just want a gui, they don't want it to be able to do anything. |
| 01:51.15 | drmessano | I can code that |
| 01:51.17 | ManxPower | And without a list of requirements, design, etc, any project will fail. |
| 01:51.20 | drmessano | Big Asterisk logo |
| 01:51.31 | drmessano | Lots of orange |
| 01:51.49 | drmessano | A big button that says "ok" |
| 01:52.49 | drmessano | People want to be able to say they have a Web GUI, because unlike us geeks who got over it in 2000, people are astonished by web GUIs for products |
| 01:53.34 | drmessano | You could design a toaster with a web gui that flashes an animated GIF "Toasting" when its ON, and people would buy tens of thousands of them |
| 01:54.30 | drmessano | BTW, thats my idea.. I'll sue if I see that at Walmart |
| 01:55.23 | ManxPower | I want to become a Walmart Secret Shopper |
| 01:56.21 | ManxPower | Every time I go to Walmart I see something incredibly stupid. Today all the networking gear (linksys routers, wifi cards, etc) were locked in a security cabinet and the lock on the cabinet was broken so you could not even BUY anything out of the cabinet. |
| 01:56.32 | drmessano | LOL |
| 01:57.14 | ManxPower | Another time ALL of the brand/size of peanut butter I buy was removed from the area where the peanutbutter is and was moved 2 aisles over. |
| 01:57.17 | drmessano | Id love to manager a walmart if they didnt have overwhelming Orwellian corporate control of each store |
| 01:57.28 | drmessano | manage* |
| 01:57.44 | ManxPower | drmessano: I just want to be a walmart manager's worst nightmare. |
| 01:57.48 | drmessano | Id put the pink USB drives by the tampons.. "Who else would buy these?" |
| 01:58.08 | drmessano | Sales would be up 300% after I was done |
| 01:58.25 | drmessano | Id put the comic books by the computer parts |
| 01:59.06 | drmessano | Toys on the shoe isle.. So every parent that drags a kid down to buy shoes will have to buy a toy too |
| 02:00.10 | drmessano | Flashlights by the christmas lights "Because youre PROBABLY gonna trip a breaker" |
| 02:00.52 | drmessano | Oh |
| 02:01.16 | drmessano | and burned Linux ISOs by the Windows boxes |
| 02:01.22 | *** join/#asterisk karmicthreat (n=aristaqi@pool-71-115-167-101.gdrpmi.dsl-w.verizon.net) |
| 02:01.29 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:01.39 | drmessano | Because people will buy a $9 game and bootleg windows |
| 02:01.47 | drmessano | They'll buy a $5 CentOS Cd |
| 02:02.37 | drmessano | Asterisk CDs by the Skype stuff "Friends dont let friends Skype" |
| 02:03.07 | karmicthreat | Anyone know any cheap fxs to ethernet channel banks? I have a crap load (49) of phones I need to hook up. |
| 02:03.42 | _ShrikE | karmicthreat: that many channels on any channel bank wont be "cheap" |
| 02:04.03 | *** join/#asterisk osas (n=nnnnnnnn@nslu2-linux/osas) |
| 02:04.17 | karmicthreat | Well yea. But I'm wondering if anything beats rhino. Its something like 1500$ for 24 ports. |
| 02:04.51 | _ShrikE | I have had rather good success with audocodes MP devices. |
| 02:04.58 | _ShrikE | They have a 24 port device as well. |
| 02:05.11 | mosty | i have a polycom 550, when i pickup the handset and dial 115, it makes the call as soon as i enter the second 1. is there a dialplan setting somewhere that might cause this? |
| 02:05.44 | _ShrikE | mosty: check your dialplan in sip.cfg |
| 02:06.13 | drmessano | I priced this out once.. and we came up with $1200 for 24 ports, I believe |
| 02:06.27 | drmessano | No |
| 02:06.28 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 02:06.39 | _ShrikE | you can also speciy dialplan per phone config |
| 02:06.44 | drmessano | $1400 |
| 02:07.15 | karmicthreat | Yea, thats about why I was getting as well. |
| 02:08.01 | karmicthreat | Hopefully nobody will tell me to just use 13 ultra cheap 4 port ATAs. |
| 02:08.04 | ManxPower | I just sent walmart an e-makil via their web site 8-) |
| 02:08.12 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d6e71230c020d589) |
| 02:08.19 | drmessano | $58 per port |
| 02:08.26 | drmessano | You can almost get cheap phones for that |
| 02:08.28 | drmessano | lol |
| 02:08.31 | ManxPower | I use Sangoma card + Adtran channel bank |
| 02:09.01 | mosty | shrike: ahh, i see. cai i just delete the dialplan? |
| 02:09.07 | karmicthreat | The problem is this is in a hotel, and all the wiring is in wall. It would cost a crap load to drag ethernet into 49 rooms. |
| 02:09.20 | drmessano | ah |
| 02:09.25 | drmessano | Not 120? |
| 02:09.35 | drmessano | lol |
| 02:09.43 | karmicthreat | Heh. |
| 02:09.49 | _ShrikE | mosty: you should set the digitmap to match your particular needs |
| 02:09.55 | drmessano | Some dude came to me a while ago wanting to do 120 rooms |
| 02:10.22 | drmessano | It was $7000 for the channel banks.. period |
| 02:11.06 | karmicthreat | I'm tempted to see if I can get 10Mbs over the cat 3 to the rooms. Yea, the channel banks are fricking pricey. Are wifi handsets any cheaper yet? |
| 02:11.23 | drmessano | But considering a run of Cat5 is $100 now for labor + the cable, thats not too bad |
| 02:11.26 | mosty | shrike: i'm trying to figure out if i need a dialplan at all. does it do anything besides dial immediately if one of the patterns matches? |
| 02:12.27 | ManxPower | mosty: if you want to press SEND after you are done dialing every time then you don't need a dialplan |
| 02:12.53 | _ShrikE | mosty: Thats right, it sends digits on pattern matches. You can press send most of the time, but things like blind transfer dont have a send button and require the match (I think). |
| 02:13.35 | mosty | is there a timeout? ie if nothing was dialed in the last 3 seconds, then deliver the call? |
| 02:13.37 | _ShrikE | Im wrong..it does |
| 02:13.59 | ManxPower | mosty: you can also do a timeout. |
| 02:14.11 | ManxPower | In my experience users HATE timeouts and/or having to press SEND |
| 02:14.42 | karmicthreat | drm: 100$ isn't to bad. But I always charge hourly for cableruns. Just because people always want me to run it through the most cramped, blind, cat crap infested room. |
| 02:14.58 | _ShrikE | Yeah.. I've hear the send button called too "cell phoneish" |
| 02:16.10 | ManxPower | I don't run wire. that's what the cable people are for. |
| 02:16.42 | mosty | the only problem i have with the dialplan now is that i can't dial anything beginning with 11, i'm trying to figure out why extensions beginning with 11 should be forbidden |
| 02:17.05 | mosty | actually make that anything beginning with 1 |
| 02:17.14 | ManxPower | mosty: it's pretty pointless to ask here, the answer is in YOUR dialplan on YOUR phone. |
| 02:17.14 | _ShrikE | mosty: are you using a config server? |
| 02:17.16 | drmessano | lol |
| 02:17.40 | drmessano | Yeah, pulling cable is never fun |
| 02:17.58 | drmessano | Especially when you get halfway through and dont know what liquid you slid through |
| 02:18.08 | mosty | _ShrikE, yes. i want to be sure i'm not doing something stupid by allowing numbers beginning with 1 in the dialplan before i change this though |
| 02:18.16 | _ShrikE | pb your sip.cfg and phone.cfg |
| 02:18.49 | mosty | _ShrikE, i won't paste the whole thing, but the dialplan is [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT |
| 02:20.02 | ManxPower | you realze that no part of that dialplan allows 11 at the start of the number, right? |
| 02:20.32 | ManxPower | in fact, it does not look like you have ANYTHING the match your internal extensions |
| 02:20.34 | mosty | ManxPower, yes, but this is from a tarball i downloaded from polycom |
| 02:20.46 | ManxPower | mosty: THEN MODIFY IY |
| 02:21.04 | _ShrikE | mosty: like I said, you need to modify it to match your particular setup |
| 02:21.26 | mosty | ManxPower, i am going to, i'm in the process of figuring out if i will break anything in doing so, these phones are in a different state |
| 02:21.45 | ManxPower | Of course you are going to break something. 8-) |
| 02:21.57 | drmessano | lol |
| 02:22.01 | drmessano | Different state? |
| 02:22.06 | drmessano | Oh, yeah.. they're toast |
| 02:23.36 | mosty | is the dialplan setting available in the web interface? i can't seem to find it |
| 02:23.58 | ManxPower | mosty: http://www.fnords.org/~eric/polycom-config-examples/ |
| 02:24.07 | mosty | ahh i see it in the web interface now |
| 02:25.46 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 02:40.36 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 02:46.00 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:47.29 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 02:47.31 | phix | hey |
| 02:47.54 | phix | the asterisk cdr csv format, where can I find this out |
| 02:47.59 | phix | ? |
| 02:48.28 | mosty | voip-info.org |
| 02:48.34 | phix | ok |
| 02:48.54 | phix | mosty: Now where? |
| 02:49.08 | mosty | use the search engine |
| 02:49.55 | *** join/#asterisk __freedom__lover (n=eduardo@200-207-145-39.dsl.telesp.net.br) |
| 02:50.06 | drmessano | mosty |
| 02:51.07 | phix | thnx |
| 02:52.27 | mosty | drmessano |
| 02:54.32 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 02:54.37 | *** join/#asterisk UserReg_CL (n=COB@pc-117-24-120-200.cm.vtr.net) |
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| 03:14.43 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 03:17.44 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 03:24.22 | mosty | how many udp ports should i restrict for rtp for a pbx with about 20 extensions? |
| 03:24.50 | mosty | what i'm trying to say is, how big a range do i really need in rtp.conf? |
| 03:28.43 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 03:29.28 | TJNII | mosty: IIRC, 2 are used per channel. So you need to ficure out how many sip channels you expect to use and go from there. |
| 03:30.21 | mosty | ok, so i can cut the range way down from the 10,000 ports for a tiny pbx like this |
| 03:33.22 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 03:34.44 | *** join/#asterisk d3wayne (n=deeewayn@76.29.245.9) |
| 03:34.44 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 03:37.54 | *** join/#asterisk jblack (n=jblack@pool-71-173-52-237.sctnpa.east.verizon.net) |
| 03:38.37 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:38.49 | bintut | hello all.. |
| 03:39.41 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 03:39.43 | bintut | what is the right term again for the right type of switch to use for a lan voip? |
| 03:40.11 | bintut | i mean, for a voip on the lan? |
| 03:42.43 | *** join/#asterisk PepOSX (n=pepOSX--@190.78.221.19) |
| 03:51.02 | *** join/#asterisk teddy233 (n=dont@207.134.8.33) |
| 03:51.43 | teddy233 | i'm almost embarrassed to ask this |
| 03:51.51 | teddy233 | but can asterisk run on windows? |
| 03:53.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e9ca1b8d88a0e6a2) |
| 03:56.06 | drmessano | teddy233: I LOL'ed |
| 03:56.08 | drmessano | No |
| 03:56.13 | drmessano | There is an asteriskwin package |
| 03:56.17 | drmessano | I tell you this |
| 03:56.22 | drmessano | So you avoid it |
| 03:56.49 | drmessano | Not because i'm anti windows |
| 03:57.11 | drmessano | But because Asterisk runs on Linux and any windows port is going to be lacking |
| 03:57.14 | drmessano | As is that one |
| 03:57.17 | drmessano | So "No" |
| 03:57.34 | teddy233 | ummm |
| 03:57.34 | teddy233 | ok |
| 03:58.04 | teddy233 | does the NOW have some functions as installing linux then asterisk ? |
| 03:58.12 | teddy233 | or is it limited? |
| 03:59.33 | drmessano | AsteriskNOW? |
| 03:59.54 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 04:00.15 | bintut | teddy233: there are lots of asterisk based software appliances |
| 04:00.34 | drmessano | AsteriskNOW won't do all that Asterisk CAN do, but thats a large task for ANY wrapper |
| 04:01.11 | drmessano | Not that its lacking.. but there would be a large expectation to cover ALL facets of Asterisk |
| 04:01.27 | teddy233 | but it can do advanced configuation ? |
| 04:01.36 | teddy233 | like phone to phone calls |
| 04:01.39 | teddy233 | with extensions ? |
| 04:01.44 | drmessano | Thats not advanced lol |
| 04:01.46 | teddy233 | and music on hold |
| 04:01.49 | drmessano | Yeah |
| 04:01.52 | teddy233 | :} |
| 04:01.56 | drmessano | Those are not advanced things |
| 04:02.02 | drmessano | Those are expected |
| 04:02.15 | drmessano | It will probably do all you expect |
| 04:02.25 | teddy233 | what would advance mean to you? |
| 04:02.41 | drmessano | Really crazy custom applications |
| 04:02.53 | teddy233 | for example ? |
| 04:02.56 | drmessano | Which you could do from CLI |
| 04:03.05 | drmessano | Hard to explain |
| 04:03.27 | drmessano | It will do all the average admin would need |
| 04:03.33 | d3wayne | teddy233: http://www.botanicalls.com/ |
| 04:03.34 | drmessano | As will most of the GUIs |
| 04:03.51 | drmessano | ROFL |
| 04:03.54 | drmessano | Yes, thats crazy lol |
| 04:04.11 | teddy233 | and on my 100Mbit link between sites.. i should be ok ? |
| 04:04.37 | drmessano | Yes :) |
| 04:04.48 | teddy233 | is that website about pot ? |
| 04:04.58 | bintut | teddy233: trixbox, asterisknow, switchvox (limited), centpbx, askozia, elastix, etc.. |
| 04:05.11 | drmessano | eww |
| 04:05.14 | drmessano | Dont say the T word |
| 04:05.45 | Juerd_ | Tea? |
| 04:06.07 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 04:06.27 | drmessano | lol |
| 04:06.33 | d3wayne | here's another: http://www.gophoneplay.com/ |
| 04:08.48 | keith4_ | http://pastebin.com/m4134840e |
| 04:08.59 | keith4_ | that's me trying to receive a fax last night |
| 04:09.48 | keith4_ | incoming fax (from a free 800 service) was routed properly to Zap/7, but then it dropped |
| 04:09.48 | teddy233 | OMG that scares me |
| 04:11.02 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 04:13.04 | CrashSys | Anyone know of a linux-based music-on-hold program that will auto-mix commercials into the stream? |
| 04:13.47 | *** join/#asterisk bjweeks (n=bjweeks@unaffiliated/bjweeks) |
| 04:14.48 | drmessano | MOH automation? |
| 04:14.50 | drmessano | Nice |
| 04:15.51 | CrashSys | http://www.nch.com.au/ims/index.html (something like that) |
| 04:16.06 | CrashSys | but that runs through linux... so I dont need a windows box to sit there and be a maintenance nightmare :) |
| 04:16.38 | CrashSys | Would be real nice if it had a web interface too ;) |
| 04:17.35 | kyron | Hey all, I am browsing through http://voipgizmos.com/ 's list of VoIP phones and would need to know what would be considered a decent phone for Business purposes. With capabilities of about 4 lines, call transfer, conference functions ...ie: basic stuff you find on bisuness phones. I guess my question is to also know which brand is know to be reliable (comparable to a real phone system) |
| 04:18.31 | bjweeks | kyron: Polycom, Cisco and Snom are the big 3 |
| 04:19.29 | CrashSys | Polycom IP501+ or Snom 300+ |
| 04:19.38 | CrashSys | Does cisco make a general-purpose SIP phone yet? |
| 04:19.43 | kyron | Ok, so Astra is not really in the game? |
| 04:20.10 | d3wayne | ~phones |
| 04:20.11 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. ... |
| 04:20.16 | kyron | CrashSys, `Cisco CP-7912G` seems like a quite affordable buy... |
| 04:20.25 | kyron | hehe...cool |
| 04:20.29 | CrashSys | The few astra's I've seen are the 480i's and they are severly lacking from what I think a phone should be for asterisk... just my opinion tho... |
| 04:21.01 | CrashSys | Although, the aastra's have backlight LCD's screens :) |
| 04:23.56 | CrashSys | The only polycom with backlight LCD screen is the 650... |
| 04:24.51 | Nugget | Do not buy a cisco phone unless you have an unshakably compelling reason to do so. |
| 04:24.54 | kyron | CrashSys, lacking in the forms of what, quality? |
| 04:25.07 | Nugget | they're a pain in the ass to buy, an even bigger pain in the ass to get working, and a total pain in the ass to use. |
| 04:25.40 | kyron | Nugget, really?...why so, we had Cisco at our U for a few years (now switched to Nortel...must have to do with funding...) |
| 04:25.54 | Nugget | running callmanager, I'm sure, not asterisk. |
| 04:26.15 | kyron | Nugget, oh that's for sure... Open source at the U...pfff...never |
| 04:26.23 | kyron | ok, so rule out Cisco.. |
| 04:26.28 | Nugget | cisco sip is not UNsupported, it's DISsupported. |
| 04:27.04 | bjweeks | does the polycom 320 have the "buddy list" in 2.1? it says it does yet my phone doesn't have any sign of it |
| 04:27.23 | CrashSys | Kyron: the Aastra's? They look like 1982 "new wave" phones... they seemed unstable and didn't support many features... but maybe the firmware got better :) |
| 04:28.30 | bintut | gtg now |
| 04:28.44 | kyron | CrashSys, what I like about the Astar's 9112i is that it LOOK exactly like what the employees are used to (http://voipgizmos.com/shopexd.asp?id=263) |
| 04:30.08 | CrashSys | well get what you feel comfortable with... lots of people use Aastra's... just not me... |
| 04:30.19 | CrashSys | Just like lots of people drive chevy... just not me... same thing... |
| 04:30.29 | J4k3 | I want a decent phone that doesn't cost as much as a workstation PC. |
| 04:30.36 | J4k3 | $100 for a phone? bullshit. |
| 04:30.47 | J4k3 | I can buy a pocketpc for that price. |
| 04:30.51 | kyron | CrashSys, you say you've seen the 480i...the cordless? |
| 04:30.56 | J4k3 | with cell and wifi radio and a real screen |
| 04:31.04 | J4k3 | why the hell do they want so much for a wired voip phone? |
| 04:31.11 | CrashSys | the cordless is 480CT... never worked with one, but known people who did, it's a real glorified POS |
| 04:31.32 | kyron | I would never propose cordless for any serious installation anyways.. |
| 04:31.33 | bjweeks | J4k3: they are made of large corporations with lots of money |
| 04:31.40 | bjweeks | s/of/for |
| 04:31.50 | CrashSys | Kryon: better off with a PAP2 and a regular cordless from walmart |
| 04:32.03 | J4k3 | bjweeks: well, that market is pretty much saturated... end users need phones too. |
| 04:32.14 | kyron | J4k3, cell phones are a POS as far as sound quality is concerned and are worthe 4 to 5X what you pay for them since they are heavily subsidized.. |
| 04:32.26 | J4k3 | and personally I hate the idea of deploying POTS adapters. POTS blows by default, and POTS adapters are generally poor POTS implentations |
| 04:32.26 | bjweeks | J4k3: end users need IP phones that work with a PBX? |
| 04:32.44 | J4k3 | bjweeks: yes. |
| 04:32.47 | Nugget | a wise person once said.... |
| 04:32.49 | Nugget | <J4k3> "its not that everyone else is expensive, its that grandstream is cheap in every sense of the word" |
| 04:33.10 | J4k3 | Nugget: grandsuck sucks, we both know it |
| 04:33.17 | bjweeks | J4k3: if they are using a PBX they aren't end users... |
| 04:33.17 | CrashSys | hahahahaha... grandstream... now there's an exercise in futility... |
| 04:33.25 | J4k3 | but, I will say, at 1/2 to 1/3rd of the price of the cheapest competition model phones, they're quite decent. |
| 04:33.41 | Nugget | *shrug* those are your words. :) |
| 04:33.51 | J4k3 | but, they suck. |
| 04:34.07 | J4k3 | but, theres no reason why a $35 phone has to suck |
| 04:34.09 | keith4_ | anyone want to take a look at my attempt at fax pass-through using Zap channels? |
| 04:34.11 | kyron | Grandstream are a hell of a lot cheaper |
| 04:34.14 | J4k3 | or a non-sucky phone has to cost $100+ |
| 04:34.29 | keith4_ | i like the $100 snom |
| 04:34.36 | J4k3 | kyron: I think I paid $32 for some 101s. they work, thats all I can say positive about them. |
| 04:35.14 | jblack | I'm running cat5e through the home. I'm using blue/brown for phone, and orange green for ethernet. |
| 04:35.28 | kyron | jblack, don't |
| 04:35.50 | drmessano | Budget Ones? |
| 04:35.50 | Nugget | 29-Jan-2007 23:06 <J4k3> is it common for grandstream budgetone 101's to lock up regularly? |
| 04:35.53 | kyron | unless you promise to stay with 100BT ehternet |
| 04:36.08 | J4k3 | Nugget: yeah, the firmware installed |
| 04:36.08 | kyron | Nugget, I love you quote machine |
| 04:36.14 | Nugget | la la la |
| 04:36.16 | J4k3 | when I bought my phones |
| 04:36.16 | J4k3 | sucked |
| 04:36.20 | keith4_ | jblack: sip phones? or are you going to run analog phones in the same cable as ethernet? |
| 04:36.23 | jblack | Yeah, it's a 200 megabit switch. I dont plan on going gigabit. |
| 04:36.25 | J4k3 | big fucking deal, every phone builder has bad firmware. |
| 04:36.35 | jblack | Analog phones, that are going to an spa8000 |
| 04:36.35 | drmessano | 200MB? |
| 04:36.51 | keith4_ | 200mbit = 100mbit full duplex? |
| 04:37.00 | jblack | Yeah, it's a full duplex switch, so 100mb tx, 100mb rx |
| 04:37.00 | J4k3 | and this is #asterisk, not #sip-device, lets move on. |
| 04:37.09 | drmessano | Isn't that a 100Mb then? |
| 04:37.13 | drmessano | lol |
| 04:37.14 | jblack | You're right. This is off topix |
| 04:37.24 | J4k3 | (there needs to be a #sip-device...) |
| 04:37.37 | keith4_ | faxing? anyone? .... please? |
| 04:37.43 | jblack | The question, actually, is a _punchdown_ one. |
| 04:37.56 | drmessano | Since when is devices used with Asterisk off topic? lol |
| 04:38.00 | Nugget | jblack is a type 66 in a type 110 world. |
| 04:38.07 | J4k3 | drmessano: since none of them run asterisk itself? |
| 04:38.12 | kyron | What is more critical for asterisk performance, CPU power or latency? (ie: preferred scheduler clock setting in the kernel) |
| 04:38.21 | J4k3 | drmessano: at least none that advertise such. |
| 04:38.24 | jblack | Kyron: latency, imho. |
| 04:38.44 | jblack | Anyways, what I was going to ask was whether to punch down the ethernet or leave a hanging jack. |
| 04:38.50 | drmessano | Didnt know there was traffic control |
| 04:39.23 | kyron | Ouch..Snom ain't cheap |
| 04:40.00 | keith4_ | ygwypf? |
| 04:40.24 | kyron | that is always questionnable |
| 04:41.09 | keith4_ | well, the way I see it... the $100 snom is more useful than the $120 avaya crap phones we use at work |
| 04:41.09 | BBHoss | you balking at snoms? |
| 04:41.21 | keith4_ | for that matter, the $100 polycom is better, too |
| 04:41.28 | BBHoss | if you want expensive, go cisco :] |
| 04:41.32 | keith4_ | or even the cheapest linksys sip phone |
| 04:42.42 | BBHoss | polycom has some very inexpensive phones as of late though |
| 04:43.02 | BBHoss | which is a nice change |
| 04:43.27 | keith4_ | yah, and their low-end PoE option isn't bad |
| 04:43.39 | keith4_ | you need to spend a lot more to get PoE in snom, for example |
| 04:45.28 | kyron | Well, I am attempting to see how much it would cost to switch over to VoIP without loss of functionality + Quality from a system with 4 lines/phone + conferencing... |
| 04:47.53 | *** join/#asterisk Xen^ (i=L_NUX@unaffiliated/lnux/x-10290) |
| 04:53.04 | keith4_ | kyron: conferencing is easy, and is a function of the pbx, not the phone... |
| 04:53.36 | kyron | keith4_, I know...but the button needs to be there ;) |
| 04:53.52 | keith4_ | well, the button helps.... but you don't *need* it |
| 04:54.06 | kyron | and call transferring is also a ++ |
| 04:54.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:54.35 | keith4_ | so.... programmable buttons then |
| 04:54.56 | kyron | I am wondering if it's possible to have each phone (the ones with multiple lines) connect to the same lines as the present system (ie: each phone has a line 1 to 4 and can be picked up but anyone) |
| 04:55.50 | kyron | I guess I would have to assign a SIP # for each phone for each line and make call groups or something... |
| 04:57.06 | d3wayne | kyron: http://www.asterisk.org/node/48342 |
| 04:57.39 | kyron | d3wayne, cool!...thanks |
| 04:57.41 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 04:58.09 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 05:06.11 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 05:10.44 | tessier | Help get RMS laid: http://boston.craigslist.org/gbs/m4w/533096562.html <- Nominate for BEST OF! |
| 05:11.08 | *** join/#asterisk Slingky (n=Maxime@modemcable111.80-201-24.mc.videotron.ca) |
| 05:11.37 | Slingky | hi guys! could somebody tell me if asterisk gui allows to configure DISA easily (like freepbx) ? |
| 05:11.53 | keith4_ | what's asterisk gui? |
| 05:12.14 | bjweeks | keith4_: digiums go at freepbx |
| 05:12.37 | keith4_ | ohhh, that |
| 05:13.26 | keith4_ | if i had some suggestions on changes to make to vmail.cgi, where should i send them? |
| 05:13.28 | Slingky | it's the one is asterisknow |
| 05:13.54 | bjweeks | DISA is just plain easy to configure without a GUI |
| 05:16.14 | Slingky | maybe, but it's few clicks with freepbx, just wanted to know if it's same in * gui |
| 05:16.19 | ManxPower | Look at the /topic, do what it says |
| 05:16.31 | ManxPower | Slingky: nobody here uses Asterisk GUI. |
| 05:16.47 | ManxPower | If they did they would be on *gasp* #Asterisk-GUO |
| 05:16.51 | Slingky | i was running adminsparadise then i try to comeback to trixbox 2.4 but jitter audio problem even with boot params changed |
| 05:16.52 | ManxPower | #asterisk-gui that is. |
| 05:16.59 | keith4_ | wow, freepbx really *does* create people who don't know how to configure asterisk |
| 05:17.02 | Slingky | so i'm asking if *now can do the job |
| 05:17.24 | ManxPower | Slingky: exten => 666,1,DISA(no-password) |
| 05:17.26 | Slingky | sorry |
| 05:17.26 | ManxPower | there! |
| 05:18.40 | ManxPower | Asking GUI questions here is like asking Windows questions on a Linux channel. It is rude, unproductive, and just plain not nice. |
| 05:18.52 | ManxPower | Expecially when the /topic directs you to the correct channel |
| 05:19.42 | keith4_ | especially when the first result for "asterisk disa" on google tells you the answer |
| 05:19.59 | bjweeks | asking for freepbx help is more funny |
| 05:20.09 | bjweeks | er, funnier :/ |
| 05:20.40 | Slingky | sorry again, what can i tell more ? |
| 05:21.01 | ManxPower | We'll beat you up for a few more mins before we calm down. |
| 05:22.06 | keith4_ | well, it's not like he asked about trixbox |
| 05:22.19 | jblack | manxpower: You sound like you could use a night at the bar or something. |
| 05:23.15 | kyron | Hey!.. I use Trixbox and it's the gratest and the best and the whole 9 yards! |
| 05:23.23 | Slingky | i won't ask for switchvox too since it's written in topic to don't do that! |
| 05:24.15 | jblack | the nice thing about this rewiring is that I'll be able to switch to ip phones soon enough. |
| 05:24.50 | keith4_ | double-heading cat5 isn't the *worst* thing you could do... but it's probably on the list somewhere |
| 05:25.11 | kyron | BTW, Snom seems to be appreciated here...does this mean I could trust all Snom Phones? IE: I see a bunch of em sold on e-bay (Snom 220) |
| 05:25.15 | jblack | I'm not thrilled about it either. |
| 05:25.45 | *** part/#asterisk osas (n=nnnnnnnn@nslu2-linux/osas) |
| 05:25.46 | kyron | jblack, especially with phone signla/power |
| 05:25.58 | jblack | The house is a hundred years old, though. I'd have to put holes in walls to run multiple cables. I'm barely sliding by with using the old phone cord to pull the new cord though. |
| 05:26.57 | jblack | Some day, after I win the lottery, I'll shell the place out, put in new electrical and ethernet done right. |
| 05:27.22 | jblack | Until then <shrug> it'll do. |
| 05:28.12 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 05:28.20 | kyron | jblack, ohhh...old house..cool . I would worry about the 80V ring that comes through for disrupting the ethernet traffix ;) |
| 05:28.46 | jblack | Actually, there's no pots here, just an ATA. |
| 05:29.16 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:29.25 | jblack | nm. Same 80 volts. |
| 05:30.11 | kyron | ;) |
| 05:30.14 | jblack | I had to make compromises in the wiring itself, and that's 115vac |
| 05:30.28 | J4k3 | ack. |
| 05:30.40 | jblack | Pardon, compromises in the running, right next to electrical conduit. |
| 05:31.06 | J4k3 | I'd rather live in something that appears more like a datacenter than a home |
| 05:31.20 | jblack | Aye. There's ups and downs. As an upside, I'll never, ever run out of space. |
| 05:31.24 | J4k3 | a bricked metal building with raised floors in the living space would be *awesome* |
| 05:31.55 | J4k3 | of course, this is an add-on nightmare and the 2000 sqft house wasn't |
| 05:32.12 | J4k3 | when I shake loose of this POS, I'm doing the bricked-up-metal-building routine. |
| 05:32.17 | J4k3 | its getting very popular here |
| 05:32.28 | jblack | I have an office, a workout room, a game room, a nice kitchen. I have my bedroom, and my daugher has her. She also has her own playroom. We have an excercise room and a server room. I even have a library with a more extensive developer library than most public libraries. |
| 05:32.35 | J4k3 | also, no 2nd story. f stairs. |
| 05:32.42 | jblack | 4 stories here. |
| 05:33.09 | J4k3 | 2 here, the ground is too shifty for a basement. |
| 05:33.44 | J4k3 | this house was also originally 1 story, they added the 2nd... its strong enough for it, but the original downstairs layout wasn't done with stairs in mind |
| 05:33.53 | J4k3 | what I really need is use of a large backhoe |
| 05:35.18 | jblack | While running cable tonight, I found a cable run that just.. stops. At least it's capped, but ... |
| 05:36.19 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 05:37.28 | J4k3 | jblack: I ran into a house a few days ago that had the most insane situation possible. |
| 05:37.30 | *** join/#asterisk andrewn (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:37.55 | J4k3 | cloth cord -> 2 bare wire strung above the back yard, to a shed |
| 05:38.05 | J4k3 | the shed end was no more than 2 meters (6') off the ground. |
| 05:38.09 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 05:38.20 | J4k3 | I damn near put my aluminum ladder into it |
| 05:39.09 | *** join/#asterisk katsup (n=katsup@adsl-76-199-60-38.dsl.irvnca.sbcglobal.net) |
| 05:39.43 | *** part/#asterisk Slingky (n=Maxime@modemcable111.80-201-24.mc.videotron.ca) |
| 05:40.32 | jblack | what was on the wire? ac? |
| 05:40.41 | J4k3 | 120VAC, no ground fault |
| 05:40.48 | kyron | LOL |
| 05:40.49 | J4k3 | one of those old light-socket fuses 'protecting' it |
| 05:41.05 | J4k3 | optimally I would have been cursing profusely had I hit it |
| 05:41.05 | jblack | heh |
| 05:41.14 | jblack | Find any dead squirrels underneath? |
| 05:41.27 | J4k3 | nope, luckily the wires ran about a foot apart. |
| 05:41.57 | jblack | Yeah, I found a cloth wire too. |
| 05:42.01 | kyron | I helped out on some renovations of a 1920 house a while back and it had bare running wires held up by ceramic posts in the ceiling of the basement...quite interesting, especialy when you ASSume they are dead... |
| 05:42.02 | jblack | {no} idea what it is. |
| 05:42.17 | jblack | Never assume they're dead in an old house. |
| 05:42.36 | J4k3 | yeah.. NEVER assume wiring is dead |
| 05:42.51 | [TK]D-Fender | ~assume |
| 05:42.52 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 05:42.53 | jblack | Not unless you can hold both ends in your hands, anyways. |
| 05:42.54 | J4k3 | and even if your meter says its dead, it might just be that you're testing against the same leg. |
| 05:43.59 | kyron | The chap I was working with didn't have a "magic pen"...quite handy to have |
| 05:44.01 | jblack | yeah, those puppies may be hot to the same voltage. Hold one in your hand, reach over with your other hand and grab a water pipe on accident.... |
| 05:44.18 | J4k3 | yep |
| 05:44.35 | jblack | Also, you can't assume that old houses have a good ground either. |
| 05:44.38 | J4k3 | or do what I did the other day... working on something live... put my sweaty leg against a metallic flexible AC vent |
| 05:44.42 | jblack | Or any, for that matter. |
| 05:44.48 | J4k3 | bzzzt + damn near falling through customer's roof. |
| 05:45.01 | J4k3 | yeah, and forget the grounds being properly bonded |
| 05:45.29 | jblack | Yeah. I've got a bad ground here. All of the ups' in the house have a nice ugly red light to remind me. |
| 05:45.38 | J4k3 | same here. |
| 05:46.03 | J4k3 | the only stuff thats fully wired is the stuff I've added. |
| 05:46.06 | jblack | Tried strining some moderate guage water to the cold water pipe. No luck. I need an actual electrician to take care of it. |
| 05:46.39 | J4k3 | yeah. houses like that/this are always good for blowing up modems |
| 05:46.42 | J4k3 | and things of the sort |
| 05:46.59 | J4k3 | sure you can bond the power box to the phone box.... but thats not *Really* enough. |
| 05:47.21 | J4k3 | it'd be nice to even possibly think the ground pin on the outlet was wired back up to the box. |
| 05:47.24 | kyron | I redid most of the wiring in mine at the moment and still much needs to be done...old houses accumulate messy workmanship over the years |
| 05:47.24 | J4k3 | but I'm not that foolish :) |
| 05:48.11 | *** join/#asterisk Slingky (n=Maxime@modemcable111.80-201-24.mc.videotron.ca) |
| 05:48.13 | jblack | Yeah. That's why I got that 66block. It's replacing a rat's nest. |
| 05:48.20 | jblack | First time I ever actually used one. ;) |
| 05:48.25 | J4k3 | oh well, hang onto the old houses... I figure within the next few years the USA will try to do some buy-out program for older houses for energy consumption reasons. |
| 05:48.37 | J4k3 | plus its a great way to falsely raise the economy. |
| 05:49.40 | jblack | Yeah, this one wasn't even insulated when I bought it five years back. I had the roof insulated about 2 years ago. I still need to redo the windows and get insulation pumped into the walls though. |
| 05:50.25 | jblack | This is dead center coal country. Energy was free. Huge places, no insulation. |
| 05:51.11 | J4k3 | thats the point where I think houses should get 'totaled out' |
| 05:51.14 | J4k3 | like a beat up old car |
| 05:51.27 | J4k3 | quite simply, after all that, you've spent enough to build a new efficient house |
| 05:51.38 | J4k3 | and it'll never come close |
| 05:56.08 | jblack | I think they should be modernized and preserved. |
| 05:56.15 | J4k3 | its impossible |
| 05:56.34 | J4k3 | preserved maybe, but not as actual lived-in homes. |
| 05:56.42 | J4k3 | too expensive |
| 05:57.04 | J4k3 | eventually they'll be too expensive to live in due to energy costs being so high, because people still insist on living in energy-munching homes |
| 05:57.08 | J4k3 | and driving energy-munching SUVs |
| 06:03.43 | kyron | why do we still use incandescent... what about LEDs! |
| 06:04.38 | Corydon76-dig | Incandescent bulbs will be illegal to sell in a few years |
| 06:05.05 | jblack | and cfc bulbs will be illegal in 30. |
| 06:05.06 | Corydon76-dig | Fluorescent and LEDs will be the only types you can get |
| 06:05.24 | jblack | Just wait and see, since they have mercury in them. |
| 06:05.36 | Corydon76-dig | probably |
| 06:06.36 | J4k3 | LEDs are far more efficient |
| 06:06.40 | J4k3 | just a matter of cost and output |
| 06:06.57 | kyron | energy doesn't cost enough yet.. |
| 06:08.15 | J4k3 | yep, give it time |
| 06:08.46 | J4k3 | hell, when random power outages become a way of life rather than a rare exception, people will likely end up DCing the essentials in their homes |
| 06:08.53 | J4k3 | inverters are simply too damned inefficient |
| 06:09.57 | [TK]D-Fender | jblack : But where else are we supposed to get our RDA of mercury from? There are only so many fish in the sea, and I give them 20 years tops! |
| 06:11.18 | [TK]D-Fender | ok, bed time. later all |
| 06:11.23 | drmessano | What do you mean I can't have nuclear fission in my office, FBI agent? |
| 06:11.35 | kyron | same here, l8rs ;) |
| 06:11.48 | J4k3 | sleep is for the weak |
| 06:12.02 | kyron | yes J4k3, sleep for a week ;) |
| 06:12.15 | J4k3 | haha |
| 06:12.22 | J4k3 | I try ;) |
| 06:12.54 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 06:13.04 | drmessano | cfc bulbs will be illegal in 10 years |
| 06:13.15 | drmessano | LED's FTW! |
| 06:13.28 | gerphimum | leds look terrible. |
| 06:13.39 | drmessano | They're taking over |
| 06:13.54 | J4k3 | the only problem I've got with LEDs so far is screwy light spectrum issues (getting better by the day...) and the fact that all the LEDs out right now are running at 30 or 60 hz. |
| 06:13.56 | drmessano | We're even using LED's for our tower beacons now |
| 06:13.56 | gerphimum | i havent seen an led light bulb - like, to light a room - ever |
| 06:14.05 | J4k3 | like florecent, they need to be cycled MUCH faster to keep me from wanting to cry. |
| 06:14.07 | metfan2007 | Hi all, I'm trying to use func_odbc, I already configured func_odbc.conf, but I'm receiving error messages that ODBC_DSNNAME is not a registered function, any idea? |
| 06:14.25 | J4k3 | LEDs don't fail every 2000 hours |
| 06:14.30 | J4k3 | like light bulbs do |
| 06:14.55 | J4k3 | the longest life tower bulbs I've seen are 8000 hrs |
| 06:15.03 | gerphimum | i dont know what theyre called, but ive got this track light thing which bulbs that came with it have been working for 2 years now |
| 06:15.05 | drmessano | yep |
| 06:15.06 | gerphimum | theyre small little bulbs |
| 06:15.11 | J4k3 | gerphimum: are they hot? |
| 06:15.16 | J4k3 | if so, they might be halogen. |
| 06:15.17 | gerphimum | quite. |
| 06:15.20 | gerphimum | could be. |
| 06:15.22 | J4k3 | yeah, thats halogen |
| 06:15.23 | gerphimum | they look damn good |
| 06:15.25 | gerphimum | theyre bright |
| 06:15.26 | drmessano | I don't know what the life on an LED beacon is |
| 06:15.29 | gerphimum | and last for fuckin ever |
| 06:15.34 | drmessano | 8 years or something |
| 06:15.59 | J4k3 | drmessano: something lik 20k hours of actual 'lit' time, and most of the LED tower lighting is only running like 25% duty cycle |
| 06:16.08 | J4k3 | so... 25% of 50% of the nighttime hours. |
| 06:16.17 | J4k3 | (for a blinking light) |
| 06:16.36 | drmessano | Yeah, but as explained to me, you subtract enviornmental too |
| 06:16.36 | metfan2007 | any help? |
| 06:16.40 | J4k3 | I'm just glad the f'n strobes are going away. |
| 06:16.44 | drmessano | LOL |
| 06:16.46 | drmessano | Hell yes |
| 06:17.01 | J4k3 | cingular tried to erect a tower with strobes here.. the locals kept shooting the light out |
| 06:17.01 | drmessano | I have a strobe lit tower.. NIGHTMARE |
| 06:17.13 | J4k3 | after like 5 service calls in a few weeks, it got re-lit. |
| 06:17.18 | drmessano | lol |
| 06:18.01 | J4k3 | apparently the light wasn't diffused properly, people were complaining the thing would put shadows in their house *with* their miniblinds shut. |
| 06:18.10 | J4k3 | it hurt my eyes to look at, thats for sure. |
| 06:18.12 | gerphimum | lol |
| 06:18.26 | drmessano | or they left it on DAYTIME 24/7 |
| 06:18.52 | J4k3 | well, they actually still have a strobe on it, that runs during the day |
| 06:19.04 | J4k3 | at nightfall the strobe goes off and its red light. |
| 06:19.39 | drmessano | A tower crew left mine on daytime once.. and I almost had to put sunglasses on going there that night for fear of seizure |
| 06:19.40 | J4k3 | personally, I say, if you're flying ghetto-IFR (I Follow Roads) sub-500' |
| 06:19.43 | J4k3 | you're asking for it |
| 06:20.19 | gerphimum | haha |
| 06:20.35 | gerphimum | no business in the air if youre gonna hit a damn tower |
| 06:20.39 | J4k3 | exactly |
| 06:20.43 | drmessano | yeah, birds |
| 06:20.45 | drmessano | ! |
| 06:20.45 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 06:20.46 | J4k3 | I can see lighting >500' towers insanely |
| 06:21.01 | J4k3 | but sub-500, unless theres a damned good reason to be that low I just don't get it. |
| 06:21.07 | gerphimum | cept for in the crazy fog, but in that case you shouldtn be flying at all |
| 06:21.15 | J4k3 | the old excuse was navigation |
| 06:21.19 | drmessano | I helped someone petition the FCC to unlight their 199 foot tower |
| 06:21.23 | J4k3 | I Say... buy a f'n GPS and STFU :) |
| 06:21.25 | drmessano | Cant believe the process |
| 06:21.57 | gerphimum | http://www.dlink.com/products/?pid=530 discuss |
| 06:21.58 | J4k3 | tell the FCC the structure fell down ;) |
| 06:22.04 | drmessano | OH |
| 06:22.07 | drmessano | Let me tell you |
| 06:22.15 | drmessano | One of my towers... burned to the ground.. |
| 06:22.20 | drmessano | No... really |
| 06:22.35 | drmessano | 50 years ago.. The TV station that occupied the building had a fire |
| 06:22.48 | ManxPower | gerphimum: totally useless unless you have an "n" card |
| 06:22.48 | drmessano | They claimed the tower burned down too, and got insurance |
| 06:22.56 | drmessano | Went to get it registered back when ASR started |
| 06:23.08 | drmessano | and the FCC had it marked as having BURNED DOWN |
| 06:23.14 | drmessano | That was awesome |
| 06:23.17 | J4k3 | hahahaha |
| 06:23.19 | J4k3 | wtf |
| 06:23.38 | J4k3 | I'm thinking... "short of a lot of magnesium... I can't see a tower burning down" :) |
| 06:23.44 | drmessano | Yeah |
| 06:23.46 | drmessano | Exactly |
| 06:27.32 | J4k3 | using oxygen in nitrogen-filled feedlines? :) |
| 06:27.35 | J4k3 | *boom!* |
| 06:27.45 | J4k3 | "oh, I think that was the wrong bottle" |
| 06:29.27 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:30.11 | drmessano | lol |
| 06:30.35 | drmessano | Nothing is louder than the sound of a Nitrogen bottle running out |
| 06:30.37 | drmessano | Toofast |
| 06:30.44 | drmessano | I dont mean "loud" |
| 06:31.05 | drmessano | I mean, like the loudest sound in a gunfight being "click", sort of loud |
| 06:31.16 | gerphimum | ManxPower >> doesnt it perform better than your typical routers in regards to wired performance as well ? |
| 06:32.30 | J4k3 | drmessano: like "nothing is louder than the engines shutting down on the passenger jet you're flying in" ;) |
| 06:35.23 | metfan2007 | pls, I need help, I cannot get func_odbc.so after make |
| 06:35.26 | drmessano | lol |
| 06:35.27 | drmessano | yes |
| 06:36.11 | ManxPower | gerphimum: I can't imagine any reason that it would. |
| 06:38.04 | drmessano | Hmm |
| 06:38.20 | drmessano | I think I am gonna print the PDF of the Asterisk book tomorrow |
| 06:38.22 | drmessano | 11 times |
| 06:38.37 | gerphimum | drmessano >> thats some serious business |
| 06:39.07 | drmessano | Yep |
| 06:41.32 | ManxPower | metfan2007: your extensive search of the mailing list archives and wiki was not helpful? |
| 06:44.17 | metfan2007 | ManxPower, yep, I already checked that unixODBC and ltdl are already installed in my CentOS server, the res_odbc.conf file is already configured, I run make menuselect to see if func_odbc is selected, but it shows XXX like dependencies are not installed |
| 06:44.19 | metfan2007 | :S |
| 06:47.15 | metfan2007 | ManxPower, really I don't know what else to do |
| 06:50.46 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 06:57.07 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:59.24 | J4k3 | http://cgi.ebay.com/GSM-Classic-Mobile-Cellular-Retro-Brick-Phone-100-New_W0QQitemZ130187876661QQihZ003QQcategoryZ3312QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 06:59.29 | J4k3 | oh man, a GSM brick phone |
| 06:59.44 | J4k3 | of course, its from china, so its probably 95% empty dead unused space, and totally sucks. |
| 07:00.04 | J4k3 | but, alas, what the world needs is a good modern brick phone. |
| 07:00.28 | *** join/#asterisk harpal (n=Harpal@124.125.79.212) |
| 07:01.28 | hmmhesays | that episode of earth final conflict was a pile of crap |
| 07:03.21 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 07:03.50 | drmessano | I just got a chinese two-way radio |
| 07:03.55 | drmessano | ITs actually kinda slick lol |
| 07:03.59 | J4k3 | uhf or vhf? |
| 07:04.13 | J4k3 | I've been eyeing getting a couple VHF's for MURS use |
| 07:04.20 | J4k3 | eventually 2M use if I ever bother getting my license. |
| 07:04.33 | drmessano | UHF with scrambling |
| 07:04.39 | drmessano | (voice inversion) |
| 07:04.50 | J4k3 | interesting |
| 07:04.56 | drmessano | The PUXING 777s |
| 07:05.05 | drmessano | Cheapie, but cool |
| 07:05.25 | drmessano | Im thinking about getting another one to use for patch with a PAP2 |
| 07:06.26 | J4k3 | the specs look decent |
| 07:06.43 | drmessano | All plastic |
| 07:06.47 | drmessano | Light as hell |
| 07:07.05 | J4k3 | yeah, built like a decent FRS walkie talkie really |
| 07:07.11 | drmessano | yes |
| 07:07.16 | drmessano | Thats how I would rate it |
| 07:08.01 | J4k3 | I've lost one of my uniden frs walkies from 150' onto sandy grass-covered dirt... the battery door didn't even fall off. |
| 07:08.33 | J4k3 | now, I dropped a cobra from the same tower, same location (both accidental, you'd think I'd make a better lanyard for them...) and it hit th concrete base and busted into a dozen pieces. |
| 07:08.43 | drmessano | lol |
| 07:09.06 | J4k3 | the uniden ones work fairly awesome for plane-to-plane |
| 07:09.35 | J4k3 | it'd still break squelch at 30 miles |
| 07:09.44 | drmessano | Im gonna build a phone patch for FRS <> PAP2 for asterisk |
| 07:09.49 | drmessano | nice |
| 07:09.54 | J4k3 | much better than the 11 mile rating, or the half mile I get around here on the ground (trees) |
| 07:13.32 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
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| 07:20.33 | linagee | does anyone know how cox provides their "cox digital telephone" service? is it packetcable / g.729? |
| 07:23.45 | *** join/#asterisk grEvenX (n=even@1elt2pn.ip.hipercom.no) |
| 07:32.57 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
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| 07:34.07 | bjweeks | linagee: some proprietary over-the-cable system |
| 07:34.57 | linagee | bjweeks: are they compressing? it sure sounds that way. or maybe analog telephony just sucks. :( |
| 07:35.17 | bjweeks | linagee: not sure, I dumped it while back |
| 07:36.14 | linagee | bjweeks: does anyone here get ISDN service to their house just for the purpose of a "hard digital line"? heh |
| 07:36.19 | J4k3 | voip over docsis = get knife, stab eyeball |
| 07:36.27 | linagee | J4k3: ew |
| 07:36.30 | J4k3 | oh, forgot the last step |
| 07:36.31 | J4k3 | twist. |
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| 07:52.35 | drmessano | Yeah |
| 07:52.42 | drmessano | Voip on cable is UUUGLY |
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| 08:48.24 | *** join/#asterisk cjk (n=cjk@d90-129-39-85.cust.tele2.lu) |
| 08:48.43 | cjk | hi, why is asterisk doing an asyncgoto from the place it has stopped and not from the users default context? |
| 08:50.33 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 08:54.05 | J4k3 | what the world needs is ilbc-style loss handling with g729-ish sound quality |
| 08:54.23 | J4k3 | cuz this ilbc sounds awful. maybe its the software I'm using. |
| 09:06.25 | sergee | "[12:05] <-- L|NUX has left this server." hmmm... core dumped? :)) |
| 09:19.34 | J4zen | What are your opinions on running Asterisk in Xen? |
| 09:19.45 | J4zen | Do the same performance issues occur? |
| 09:20.02 | J4zen | in relation to other virtualisation methods |
| 09:20.30 | *** join/#asterisk powerkill (n=powerkil@office.annatel.net) |
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| 09:40.53 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.190) |
| 09:42.05 | Al_WinKiller | hi guys, need help, so,, I got asterisk up and running with digum TE220 card |
| 09:42.23 | Al_WinKiller | seems work good ( mean users can call each other ),, |
| 09:42.47 | Al_WinKiller | the problem is e1 is connected to cisco 5350 ---- > meredian |
| 09:43.01 | Al_WinKiller | and I can't establish connection via e1 |
| 09:43.06 | Al_WinKiller | anybody can help me ? |
| 09:43.07 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 09:43.11 | Al_WinKiller | to explain ? |
| 09:43.20 | Al_WinKiller | I am new in Asterisk and VoIP |
| 09:44.42 | ronr | Al_WinKiller: you installed zaptel? libpri? configured zaptel? zapata.conf? |
| 09:55.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:58.41 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:59.12 | Al_WinKiller | yes, all of them |
| 09:59.35 | Al_WinKiller | when I do ztcfg -vvv |
| 09:59.42 | Al_WinKiller | I see the channels |
| 09:59.58 | Al_WinKiller | Channel 01: Clear channel (Default) (Slaves: 01) |
| 09:59.59 | Al_WinKiller | Channel 02: Clear channel (Default) (Slaves: 02) |
| 09:59.59 | Al_WinKiller | Channel 03: Clear channel (Default) (Slaves: 03) |
| 10:00.04 | Al_WinKiller | and more |
| 10:00.23 | Al_WinKiller | and in the end I see |
| 10:00.24 | Al_WinKiller | 31 channels to configure. |
| 10:00.41 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 10:00.52 | Al_WinKiller | I can show you zapata and zaptel conf too |
| 10:01.28 | ronr | Al_WinKiller: you configured your dialplan? something like Dial(Zap/g1/<number>) ? |
| 10:01.30 | tzafrir_laptop | Al_WinKiller, what i the output of: pri show spans |
| 10:01.39 | *** join/#asterisk Schumie (n=Steve@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 10:03.06 | Al_WinKiller | dude, hold on a second, ( chief is calling me ) brb |
| 10:07.48 | *** join/#asterisk harpal (n=Harpal@124.125.79.212) |
| 10:09.53 | *** join/#asterisk BBHoss (n=hoss@proxy-srv.uah.edu) |
| 10:13.12 | *** join/#asterisk saftsack (n=saftsack@s0655.vpn.hrz.tu-darmstadt.de) |
| 10:14.00 | ronr | is there a tool that converts a mp3 to a bunch of different formats (for moh) like: wav ulaw alaw gsm g729 (basically, the list asterisk install with the default moh sounds)? |
| 10:15.23 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 10:16.21 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-13-12.dsl.lsan03.sbcglobal.net) |
| 10:17.19 | J4zen | ronr: check sox |
| 10:18.20 | tzafrir_laptop | not g729, though |
| 10:18.53 | tzafrir_laptop | g729: asteris's convert command. If you have a license |
| 10:18.56 | *** join/#asterisk zeeesh (n=zeeesh@203.215.176.22) |
| 10:20.43 | Al_WinKiller | so dude, I am back, when I do "pri show spans" |
| 10:20.44 | zeeesh | getting problem by dialing some of my peers who are having "UNREACHABLE" status .. how to make them reachable ? |
| 10:20.56 | Al_WinKiller | I get PRI span 1/0: Provisioned, In Alarm, Up, Active |
| 10:21.13 | ronr | J4zen: thx, I'll try sox |
| 10:21.29 | ronr | Al_WinKiller: try zttool |
| 10:21.38 | ronr | (commandline, not asterisk CLI) |
| 10:21.55 | Al_WinKiller | ok |
| 10:22.40 | Al_WinKiller | I have no zttool , got zttest |
| 10:24.09 | ronr | ok, you need to figure out what alarm you got, but you could try enabling / disabling crc4 in zaptel.conf (that was causing an alarm for me when I was installing asterisk, but no doubt there are many more causes for alarms) |
| 10:24.26 | ronr | btw. what color does the led on the digium card burn at |
| 10:24.26 | Al_WinKiller | ok, hold on a second |
| 10:24.50 | Al_WinKiller | i got this in zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow |
| 10:25.34 | Al_WinKiller | green color and from cisco I get something like "multiframe transfer is established" ( sorry for my english ) |
| 10:25.50 | ronr | Al_WinKiller: why did you put yellow in there? |
| 10:26.12 | ronr | Al_WinKiller: if you got access to the cisco, check if it also has crc4 enabled, if it doesn't remove crc4 |
| 10:26.21 | Al_WinKiller | I saw it in manual ( put it last time ) before was without yellow |
| 10:26.34 | Al_WinKiller | ok, hold on |
| 10:27.26 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:27.52 | Al_WinKiller | on cisco I got only |
| 10:27.53 | Al_WinKiller | interface Serial3/1:15 |
| 10:27.54 | Al_WinKiller | <PROTECTED> |
| 10:27.54 | Al_WinKiller | <PROTECTED> |
| 10:27.54 | Al_WinKiller | <PROTECTED> |
| 10:27.54 | Al_WinKiller | <PROTECTED> |
| 10:27.55 | Al_WinKiller | <PROTECTED> |
| 10:29.24 | ronr | Al_WinKiller: I don't have cisco here so can't tell you how you could check if crc4 is ok, but try span=1,0,0,ccs,hdb3 and see what happens |
| 10:29.31 | Al_WinKiller | on on cisco I have framing NO-CRC4 so I removed it |
| 10:30.32 | Al_WinKiller | ok, done it |
| 10:30.55 | Al_WinKiller | and in ztcfg -vvv I still got last line "31 channels to configure" |
| 10:31.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:32.32 | Al_WinKiller | when I am calling from softphone via cisco and meredian to a phone I got this |
| 10:32.34 | Al_WinKiller | <PROTECTED> |
| 10:32.52 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 10:32.52 | Al_WinKiller | but when I call from softphone to cisco ip phone it is ok |
| 10:33.20 | Al_WinKiller | I got both of them ( softphone and cisco phone ) in sip.conf and in extensions.conf |
| 10:35.36 | Al_WinKiller | dude ? |
| 10:35.40 | Al_WinKiller | are you there ? |
| 10:38.26 | *** join/#asterisk gardo (n=gardo@121.97.198.127) |
| 10:40.09 | tzafrir_laptop | Al_WinKiller, hmmm.... you said layer 1 was up. So why mess with it? |
| 10:40.15 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 10:41.15 | Al_WinKiller | don't know dude,, this is my first asterisk and first time I am configuring e1 |
| 10:42.21 | Al_WinKiller | so look what I did in extensions.conf |
| 10:42.39 | Al_WinKiller | exten => 1299044,1,Dial(ZAP/g1/${EXTEN}) |
| 10:42.42 | Al_WinKiller | and I got |
| 10:43.03 | Al_WinKiller | <PROTECTED> |
| 10:43.03 | Al_WinKiller | <PROTECTED> |
| 10:43.03 | Al_WinKiller | <PROTECTED> |
| 10:43.03 | Al_WinKiller | <PROTECTED> |
| 10:43.03 | Al_WinKiller | <PROTECTED> |
| 10:43.03 | Al_WinKiller | <PROTECTED> |
| 10:43.05 | Al_WinKiller | <PROTECTED> |
| 10:43.07 | Al_WinKiller | <PROTECTED> |
| 10:46.24 | Al_WinKiller | I think the problem is with channels ( on e1 ) |
| 10:47.13 | Al_WinKiller | in my zaptel.conf i got |
| 10:47.15 | Al_WinKiller | span=1,0,0,ccs,hdb3 |
| 10:47.15 | Al_WinKiller | bchan = 1-15, 17-31 |
| 10:47.15 | Al_WinKiller | dchan = 16 |
| 10:47.15 | Al_WinKiller | loadzone = ru |
| 10:47.18 | Al_WinKiller | defaultzone = ru |
| 10:50.13 | tzafrir_laptop | Al_WinKiller, first-off, use a pastebin |
| 10:50.16 | tzafrir_laptop | ~pb |
| 10:50.17 | jbot | [~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://www.rafb.net/paste |
| 10:50.38 | tzafrir_laptop | another thing to do: pri debug span 1 |
| 10:51.01 | tzafrir_laptop | And paste that huge cryptic trace :-) |
| 10:52.33 | ronr | Al_WinKiller: was called away for a while, did you get rid of the alarm on the card? |
| 10:55.08 | Al_WinKiller | so I did debug on cisco and linux ( asterisk ) and look what i got |
| 10:55.48 | Al_WinKiller | Jan 9 14:47:54: ISDN Se3/1:15 **ERROR**: call_incoming: Received a call id 0x1DEE with a bad bearercap from 1299401 on b channel 1 |
| 10:55.51 | Al_WinKiller | it is on cisco |
| 10:57.15 | Al_WinKiller | ok, |
| 10:57.19 | Al_WinKiller | thnx |
| 10:57.39 | Al_WinKiller | can you check my zaptel and zapate conf ? I want to be sure |
| 10:58.16 | Al_WinKiller | so this is zaptel.conf output |
| 10:58.18 | Al_WinKiller | [root@asterisk asterisk]# cat /etc/zaptel.conf |
| 10:58.18 | Al_WinKiller | span=1,0,0,ccs,hdb3 |
| 10:58.18 | Al_WinKiller | bchan = 1-15, 17-31 |
| 10:58.18 | Al_WinKiller | dchan = 16 |
| 10:58.18 | Al_WinKiller | loadzone = ru |
| 10:58.19 | Al_WinKiller | defaultzone = ru |
| 10:58.25 | Al_WinKiller | and this iz zapata.conf output |
| 10:58.48 | Al_WinKiller | [root@asterisk asterisk]# cat /etc/asterisk/zapata.conf |
| 10:58.48 | Al_WinKiller | [channels] |
| 10:58.48 | Al_WinKiller | context=zap-in |
| 10:58.48 | Al_WinKiller | switchtype=euroisdn |
| 10:58.48 | Al_WinKiller | pridialplan=national |
| 10:58.48 | Al_WinKiller | signalling=pri_cpe |
| 10:58.50 | Al_WinKiller | usecallerid=yes |
| 10:58.52 | Al_WinKiller | hidecallerid=no |
| 10:58.54 | ronr | ~pb |
| 10:58.54 | jbot | [~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://www.rafb.net/paste |
| 10:58.54 | Al_WinKiller | callwaiting=yes |
| 10:58.56 | Al_WinKiller | callwaitingcallerid=yes |
| 10:58.58 | Al_WinKiller | threewaycalling=yes |
| 10:59.00 | Al_WinKiller | transfer=yes |
| 10:59.02 | Al_WinKiller | cancallforward=yes |
| 10:59.04 | Al_WinKiller | echocancel=yes |
| 10:59.06 | Al_WinKiller | rxgain=0.0 |
| 10:59.08 | Al_WinKiller | txgain=0.0 |
| 10:59.10 | Al_WinKiller | sorry for flooding |
| 10:59.12 | Al_WinKiller | group=1 |
| 10:59.14 | Al_WinKiller | callgroup=1 |
| 10:59.16 | Al_WinKiller | pickupgroup=1 |
| 10:59.17 | ronr | use a pastebin!! |
| 10:59.18 | Al_WinKiller | immediate=no |
| 10:59.20 | Al_WinKiller | callprogress=no |
| 10:59.22 | Al_WinKiller | callerid=asreceived |
| 10:59.24 | Al_WinKiller | group=1 |
| 10:59.26 | Al_WinKiller | signalling=pri_cpe |
| 10:59.28 | Al_WinKiller | channel => 1-15,17-31 |
| 10:59.30 | Al_WinKiller | end of zapata.conf check it please |
| 10:59.32 | Al_WinKiller | ok, that was all |
| 10:59.45 | Al_WinKiller | can you check it ? or I have to use pastbin ? |
| 10:59.58 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 11:00.03 | ronr | you have to use a pastebin |
| 11:00.11 | Al_WinKiller | ok, hold on |
| 11:01.52 | Al_WinKiller | ok, did it, check it please |
| 11:01.54 | Al_WinKiller | http://pastebin.com/m54fe8ed |
| 11:03.48 | *** join/#asterisk _ys (n=kvirc@91.151.196.254) |
| 11:05.18 | ronr | and pri show span 1 still shows In Alarm? |
| 11:07.04 | Al_WinKiller | asterisk*CLI> pri show span 1 |
| 11:07.04 | Al_WinKiller | Primary D-channel: 16 |
| 11:07.04 | Al_WinKiller | Status: Provisioned, Up, Active |
| 11:07.04 | Al_WinKiller | Switchtype: EuroISDN |
| 11:07.04 | Al_WinKiller | Type: CPE |
| 11:07.05 | Al_WinKiller | Window Length: 0/7 |
| 11:07.07 | Al_WinKiller | Sentrej: 0 |
| 11:07.09 | Al_WinKiller | SolicitFbit: 0 |
| 11:07.11 | Al_WinKiller | Retrans: 0 |
| 11:07.13 | Al_WinKiller | Busy: 0 |
| 11:07.15 | Al_WinKiller | Overlap Dial: 0 |
| 11:07.17 | Al_WinKiller | T200 Timer: 1000 |
| 11:07.19 | Al_WinKiller | T203 Timer: 10000 |
| 11:07.21 | Al_WinKiller | T305 Timer: 30000 |
| 11:07.23 | Al_WinKiller | T308 Timer: 4000 |
| 11:07.25 | Al_WinKiller | T309 Timer: -1 |
| 11:07.27 | Al_WinKiller | T313 Timer: 4000 |
| 11:07.29 | Al_WinKiller | N200 Counter: 3 |
| 11:07.31 | Al_WinKiller | sorry :( |
| 11:07.41 | ronr | ok, I'm done... really, learn to use pastebins and try again with someone else / tomorrow |
| 11:08.08 | Al_WinKiller | sorry |
| 11:08.17 | Al_WinKiller | should we try again ? I will use pastbin |
| 11:10.14 | Al_WinKiller | ok dude,, thnx anyway , I will figure it out from here,, sorry for pasting it here |
| 11:11.12 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 11:11.16 | ronr | good luck |
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| 12:05.40 | RedStalker_Mike | hi all |
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| 12:13.02 | *** part/#asterisk kamanashisroy (n=root@202.56.7.135) |
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| 12:31.46 | ronr | what options should I pass to sox to turn a mp3 into .wav suitable for usage in * moh ? |
| 12:33.34 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
| 12:34.34 | fetcher | is there any conceivable way to change codecs on the fly (in mid call)? For either a SIP channel or IAX2? |
| 12:41.20 | RoyK | fetcher: iirc that's done with a reinvite - as with a t.38 call. it starts with an rtp session and later sends a reinvite to switch to udptl |
| 12:42.11 | *** join/#asterisk DeepY0X (n=DeepY0X@201.240.80.101) |
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| 12:59.55 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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| 13:27.08 | dominic1 | Does anybody use crypt with misdn?? |
| 13:28.52 | alinux-lb22 | Hi All i have Asterisk card with four modules all are RED ..are this FXS or FXO modules ? |
| 13:29.22 | *** join/#asterisk duckz (n=duckz@85-204-47-228.etth.opensys.ro) |
| 13:30.30 | [TK]D-Fender | alinux-lb22: FXO |
| 13:30.58 | [TK]D-Fender | alinux-lb22: On the (safe?) assumption your meant a DIGIUm modular analog card of some sort. |
| 13:31.07 | alinux-lb22 | yep |
| 13:32.51 | alinux-lb22 | thanks [TK]D-Fender |
| 13:34.27 | ronr | in musiconhold.conf I set mode to quietmp3, dropped a bunch of mp3 files in the right dir and did moh reload, what did I forget (because, it's not working, I get no moh and the console tells me it is unable to open the file (it shows the file without the .mp3 extension) |
| 13:36.25 | [TK]D-Fender | ronr: Go verify what version of mp123 you're running... |
| 13:36.54 | ronr | [TK]D-Fender: you mean mpg123 I think, that's 0.67 |
| 13:37.23 | [TK]D-Fender | ronr: thats bad. The only version taht is supported by * is 0.59r |
| 13:37.25 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 13:37.34 | [TK]D-Fender | ronr: Yes, mpg123 |
| 13:38.40 | ronr | [TK]D-Fender: ok, I'll look that version up somewhere and install it (I still wonder why sox -c 1 <mp3file> output.was didn't work btw) |
| 13:38.48 | ronr | that should be output.wav |
| 13:38.56 | *** join/#asterisk VijayG (n=vijay@58.68.47.109) |
| 13:39.17 | [TK]D-Fender | ronr: Why aren't you using Native MoH anyways? |
| 13:40.30 | ronr | [TK]D-Fender: dunno, been trying to replace the default music with some mp3's, tried converting them with sox to other formats but it didn't work so I decided to try and let the mp3's play |
| 13:40.31 | tzafrir_laptop | I must point out that 0.59 is quite obsolete by now. mpg123's development has moved on. 1.01 as been released recently |
| 13:41.21 | [TK]D-Fender | ronr: Using MP3 is fine... just let Native do it. |
| 13:41.44 | tzafrir_laptop | but still, using mp3 moh for playing files is quite pointless |
| 13:41.59 | [TK]D-Fender | tzafrir_laptop: True, but for all reports, 0.59r is still the only one that really reliably(?) workds with * |
| 13:42.15 | ronr | [TK]D-Fender: so, just mode=files should do it? (I though I already tried that, but I'll try again) |
| 13:42.37 | [TK]D-Fender | ronr: Correct. And naturally you need asterisk-addons (format_mp3) for that |
| 13:43.45 | [TK]D-Fender | tzafrir_Yes converting to a mono telecom-related codec does make sense, but it saves you mucking around when you can just drop in music in the form its most often encoded in |
| 13:44.24 | ronr | [TK]D-Fender: that's not the agx-ast-addons package nor in asterisk 1.4.16.2 right? |
| 13:45.02 | [TK]D-Fender | ronr: Correct. just the one clearly listed at asterisk.org |
| 13:45.15 | ronr | [TK]D-Fender: what should be the correct options for converting it to the right wav (got plenty of diskspace and never enough cpu speed) |
| 13:45.20 | *** join/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 13:45.46 | [TK]D-Fender | ronr: You can find that on the WIKI, I don't know offhand... I just use MP3 personally... |
| 13:45.48 | RockHound | good day. is it possible to log when a trunk can not be dialed since it is full? |
| 13:46.07 | [TK]D-Fender | RockHound: Sure.... add something in your dialplan for that. |
| 13:46.27 | ronr | [TK]D-Fender: ok, thx, I'll just install the addons and move to wav if stuff gets too slow |
| 13:46.27 | [TK]D-Fender | ronr: In you place I'd just install asterisk-addons... |
| 13:46.41 | [TK]D-Fender | ronr: What CPU & kind of load? |
| 13:46.48 | RockHound | [TK]D-Fender: ok thanks |
| 13:47.32 | ronr | [TK]D-Fender: no problems whatsoever right now, very low load (on a pentium duocore), but the company has been growing quite rapidly the last few years and if that continues.... |
| 13:48.27 | [TK]D-Fender | ronr: I don't think will have to worry about then unless you've got a lot of calls on hold... even then.. |
| 13:49.25 | *** join/#asterisk dundel (n=daniel@200.2.161.172) |
| 13:50.11 | *** join/#asterisk |chodorenko| (n=chodoren@etm005.nl.ded.neolocation.net) |
| 13:50.15 | |chodorenko| | Hi ALL |
| 13:50.24 | |chodorenko| | i Have one problem |
| 13:51.10 | |chodorenko| | in extenshen i check present file or not |
| 13:51.37 | |chodorenko| | if present then play this file, if no then skip |
| 13:52.07 | |chodorenko| | exten => pizza_in,n,Wait(1) |
| 13:52.07 | |chodorenko| | exten => pizza_in,n,Set(prazdnik_YesNo=${IF(STAT(t,pizza/prazdnik)?yes:no)}) |
| 13:52.08 | |chodorenko| | exten => pizza_in,n,ExecIf($[${prazdnik_YesNo} = yes]|Background|pizza/prazdnik) |
| 13:52.41 | |chodorenko| | why prazdnik_YesNo always == yes ? hou its correct |
| 13:53.36 | ronr | [TK]D-Fender: mp3's working fine now, thx |
| 13:54.39 | Al_WinKiller | ronr ? dude ? help ? :) |
| 13:54.42 | [TK]D-Fender | |chodorenko|: because you are not evaluating STAT. You forgot to reference it in ${} |
| 13:54.44 | Al_WinKiller | I got this one |
| 13:55.02 | Al_WinKiller | <PROTECTED> |
| 13:55.16 | Al_WinKiller | when I start asterisk with asterisk -vvvvvvvvvvgc |
| 13:55.20 | Al_WinKiller | any idea ? |
| 13:56.42 | ronr | Al_WinKiller: no, I don't know, but maybe [TK]D-Fender or tzafrir_laptop knows, just make sure you use a pastebin ;) |
| 13:57.00 | Al_WinKiller | ok, I will |
| 13:57.34 | Al_WinKiller | I have done "pri debug span 1 " so I will past it on pastbin ( output ) you check it pls |
| 13:57.52 | |chodorenko| | [TK]D-Fender: please correct my error |
| 13:58.21 | [TK]D-Fender | |chodorenko|: I jsut told you exactly where the error was. How about you try and show me where you think they belong.... |
| 13:58.33 | tzanger | morning tk |
| 13:58.41 | Al_WinKiller | ok, here is it http://pastebin.com/m643ce896 |
| 13:58.46 | |chodorenko| | (prazdnik_YesNo=${IF(${STAT(t,pizza/prazdnik)}?yes:no)}) this? |
| 13:58.59 | [TK]D-Fender | |chodorenko|: See? Not that hard... |
| 13:59.06 | [TK]D-Fender | tzanger: Mornin' |
| 13:59.30 | Al_WinKiller | ronr ? |
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| 14:00.05 | Al_WinKiller | guys ? can someone help me ? |
| 14:00.40 | ronr | Al_WinKiller: this is out of my league (for now anyway), can't help you |
| 14:00.53 | Al_WinKiller | ok :( |
| 14:00.56 | [TK]D-Fender | Al_WinKiller: Ext: 1 Cause: Incompatible destination (88), class = Invalid message (e.g. parameter out of range) (5) <-- not sure about the full meaning of this, but ar you sure the # you dialed is valid? |
| 14:01.54 | *** join/#asterisk Havokmon (n=None@mail.valeoinc.com) |
| 14:01.58 | Havokmon | gm all |
| 14:02.37 | Al_WinKiller | you mean destination ? yes,, it goes like this asterisk (E1)---->(E1)Cisco 5350(E1) --->(E1)Meredian ---- > phone |
| 14:02.41 | *** join/#asterisk d-k-t (n=dt@125.118.34.74) |
| 14:03.00 | Havokmon | Can someone bind up the voip provider list for me? I forget the keyword |
| 14:03.08 | Havokmon | bind/bring |
| 14:03.09 | |chodorenko| | [TK]D-Fender: i not fully understand You , my english is veary bad :( |
| 14:03.12 | [TK]D-Fender | Havokmon: For where? |
| 14:03.15 | Havokmon | US |
| 14:03.24 | [TK]D-Fender | |chodorenko|: Looks like you fixed it fine |
| 14:03.27 | [TK]D-Fender | ~itsplist-us |
| 14:03.27 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com |
| 14:03.30 | Havokmon | Thanks |
| 14:04.01 | [TK]D-Fender | Al_WinKiller: Ok, so not a real PSTN number per-se? |
| 14:04.04 | Al_WinKiller | and from cisco I got this |
| 14:04.26 | Al_WinKiller | Jan 9 17:56:35: ISDN Se3/1:15 **ERROR**: call_incoming: Received a call id 0x2227 with a bad bearercap from 1299402 on b channel 1 |
| 14:04.26 | Al_WinKiller | Jan 9 17:56:35: ISDN Se3/1:15 EVENT: process_rxstate: ces/callid 1/0x2227 calltype 2 CALL_CLEARED |
| 14:04.37 | Al_WinKiller | yes it is a pstn |
| 14:04.58 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:05.35 | Al_WinKiller | and in extensions.conf I got this "exten => 1299044,1,Dial(ZAP/g1/${EXTEN})" |
| 14:06.27 | [TK]D-Fender | Al_WinKiller: Ok is was just a guess and is equally out of my league, sorry.... |
| 14:06.40 | Al_WinKiller | hm,, ok :( |
| 14:08.42 | *** join/#asterisk FOMICIANO (i=FOMICIAN@201.48.5.109) |
| 14:09.36 | ronr | Al_WinKiller: I'd call over the telco company and have them call over the a guy with his own PBX and let them proof to you the cisco is configured ok and works |
| 14:10.43 | ronr | (I've spend days trying to get my E1 to work and finally the telco box was misconfigured, they came over and 5 min. later I was calling out) |
| 14:11.37 | kaldemar | Al_WinKiller: the cisco complains about the bearer capability, you can see the sent bearer capability in the SETUP message. |
| 14:12.35 | Al_WinKiller | yes,, I know,, and I don't get it , cuz in Cisco I got speech,,, bearer I mean,, so it should work |
| 14:13.55 | [TK]D-Fender | chodorenko: And while you're at it, re-read STAT's instructions : show function STAT |
| 14:15.21 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
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| 14:17.54 | defswork | what are my options for providing 90 analog pots ? |
| 14:18.30 | Qwell | defswork: quad T1 card, and a few channel banks, probably |
| 14:18.46 | [TK]D-Fender | defswork: 4 x AudioCodes MP-124 |
| 14:19.07 | [TK]D-Fender | defswork: (or MediaTrix 1124's) |
| 14:19.10 | Qwell | audiocodes a sip gateway or something? |
| 14:19.17 | [TK]D-Fender | Qwell: Yup |
| 14:19.38 | defswork | feck - expensive |
| 14:19.42 | [TK]D-Fender | Qwell: Far less load on *, redundency capabilities, and no Zaptel madness to worry about. |
| 14:20.21 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:20.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:20.51 | [TK]D-Fender | defswork: Or you could make a bigger wiring mess and use 12 x Linksys SPA-8000 |
| 14:21.12 | defswork | :) |
| 14:22.08 | [TK]D-Fender | defswork: Mediatrix 1124 = $1500 ($62.5/port). SPA-8000 = $240 ($30/port) |
| 14:22.47 | [TK]D-Fender | defswork: I've got my first testimonial on the SPA-8000 now and its everything I expected. |
| 14:23.06 | Qwell | [TK]D-Fender: "it doesn't suck that much"? |
| 14:23.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:23.43 | [TK]D-Fender | Qwell: You're sounding more and more like coppice every day ;) |
| 14:24.05 | defswork | [TK]D-Fender: well ? |
| 14:24.13 | [TK]D-Fender | Qwell: The Linksys were always pretty friendly and feature-full. Its just like 8 ATA ports slapped into 1 box. |
| 14:24.47 | _x86_ | [TK]D-Fender: do they make an SPA that has 24 ports? |
| 14:24.49 | [TK]D-Fender | Dodge! Parry! Lunge! THRUST!!!!!!!! |
| 14:24.52 | _x86_ | or better, 48 ports? |
| 14:24.58 | [TK]D-Fender | _x86_: Nope. |
| 14:24.59 | defswork | [TK]D-Fender: do you recall me mentioning hotel - I went to see them - they need to stick with analog handsets |
| 14:25.08 | defswork | but they've only got 20 rooms anyway |
| 14:25.28 | [TK]D-Fender | defswork: Thats fine. |
| 14:25.46 | defswork | but now a guy I know whats to look into his hotel - 90 rooms :o |
| 14:25.51 | defswork | wants* |
| 14:26.25 | *** join/#asterisk mog (n=mog@nat/digium/x-fe918e87d62e5879) |
| 14:26.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:27.21 | defswork | SPA 8000 is 185UKP - but I don't fancy 12 of them |
| 14:27.47 | Qwell | defswork: that's going to be one of the cheapest options |
| 14:28.26 | [TK]D-Fender | Qwell: the SPA-8000 scales at the same price as their 2-port models.... very nice. |
| 14:28.39 | [TK]D-Fender | Qwell: and 8 is a great density for smaller companies. |
| 14:28.40 | defswork | Qwell: yeah but think of the mess :) |
| 14:28.57 | Qwell | well, there's still the other two options... |
| 14:29.16 | defswork | I'm just looking for UK suppliers to see the uk price |
| 14:30.10 | chodorenko | [TK]D-Fender: i reread stat function manual 10 and not understant why if file exist then fungtion return "1" , and if not exist return "" |
| 14:30.20 | chodorenko | why not "0" |
| 14:31.38 | [TK]D-Fender | chodorenko: http://pastebin.com/m6053d69f <-- whree do you see this "t" option for "file exists?". |
| 14:32.51 | chodorenko | [TK]D-Fender: not t , its error my , i can use e and f |
| 14:33.26 | [TK]D-Fender | chodorenko: Good, now go fix your error. |
| 14:33.34 | chodorenko | [TK]D-Fender: yes |
| 14:34.16 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:34.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:34.28 | drmessano | 20 SPA8000s? |
| 14:34.31 | drmessano | Err |
| 14:34.34 | drmessano | 12 I mean |
| 14:34.44 | defswork | http://www.voipon.co.uk/vegastream-vega-5048-48-fxs-2-fxo-p-598.html < what about this ? 48 FXS |
| 14:39.14 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
| 14:40.08 | chodorenko | [TK]D-Fender: http://pastebin.com/m21430627 |
| 14:40.30 | _x86_ | [TK]D-Fender: when there is an ATA with 24 or 48 ports in a single box, I might consider using one |
| 14:40.49 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 14:40.57 | defswork | _x86_: I just pasted one |
| 14:42.58 | [TK]D-Fender | defswork: That'd work fine. Is the cost good/port? |
| 14:43.07 | defswork | 43UKP |
| 14:43.19 | *** join/#asterisk Morrocco (n=ivan@189.182.30.6) |
| 14:43.44 | _x86_ | what's that translate to Real Money(tm)? |
| 14:43.50 | _x86_ | i mean, USD... typo sorry |
| 14:44.01 | [TK]D-Fender | chodorenko: exten => pizza_in,n,Set(prazdnik_YesNo=${IF(STAT(f,pizza/prazdnik)?yes:no)}) <-- you forgot to evaluate STAT again.... |
| 14:44.09 | defswork | well most uk stuff is expensive - might well be 40-50USD |
| 14:44.34 | [TK]D-Fender | _x86_: Um... USD isn't real money... its being printed out of the Milton Bradly:Monopoly factory :p |
| 14:44.43 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:44.58 | _x86_ | wow! they have a USD currency button... i clicked it and the price for one of these 48 port units is $4,167.90 |
| 14:45.02 | Dr-Linux | how can i export/import astdb? |
| 14:45.15 | _x86_ | that's insane |
| 14:45.57 | defswork | _x86_: thats uk price simply converted to us |
| 14:46.14 | Dr-Linux | _x86_: talking to me? |
| 14:46.18 | defswork | _x86_: bought in us I'd bet its about 2200 USD |
| 14:47.17 | _x86_ | defswork: that's a great price, actually... if only i could find a US reseller that carries that brand ;) |
| 14:47.31 | _x86_ | googling for it only shows suppliers in .UK and .RU |
| 14:47.43 | chodorenko | [TK]D-Fender: evaluate ? ${IF((${STAT(f,pizza/prazdnik)})?yes:no)}) ? |
| 14:47.57 | _x86_ | defswork: here we go... http://store.dmsvoip.com/ProductDetails.asp?ProductCode=Vega+5048 |
| 14:48.11 | _x86_ | defswork: $4,000 USD from a US supplier... that's crazy take |
| 14:48.12 | defswork | _x86_: 41750 for the 24 port |
| 14:48.12 | _x86_ | talk* |
| 14:48.17 | defswork | $1750* |
| 14:48.21 | _x86_ | url me |
| 14:48.42 | defswork | http://www.voipsupply.com/product_info.php?products_id=3207&utm_medium=shoppingengine&utm_source=smarter |
| 14:48.49 | defswork | 1584 even there |
| 14:48.55 | [TK]D-Fender | chodorenko: No, but getting warmer. |
| 14:49.09 | _x86_ | hmm |
| 14:49.11 | defswork | 2754 for the 48 port |
| 14:49.17 | _x86_ | I have an account with VoIP Supply too |
| 14:49.23 | defswork | http://www.voipsupply.com/product_info.php?products_id=3208 |
| 14:49.31 | [TK]D-Fender | defswork: Well its definitely better scaled than the Mediatrix 1124 there. |
| 14:49.41 | defswork | thats 1400UKP |
| 14:49.45 | defswork | I might import some :) |
| 14:50.10 | defswork | _x86_: get one and let me know how you get on with it :) |
| 14:50.17 | _x86_ | $57.39/port |
| 14:50.28 | _x86_ | that's the Vega 5000 48-port |
| 14:50.46 | [TK]D-Fender | defswork: Duty + VAT = ouch |
| 14:50.55 | _x86_ | I'm paying $54.16/port now with Rhino analog FXS channel banks |
| 14:51.03 | _x86_ | although, I'm not at all happy with them |
| 14:51.12 | _x86_ | I've had to return 4 so far |
| 14:51.18 | [TK]D-Fender | defswork: Poor over-taxed Brit.... should ship it via Sherwood to avoid "the man" ;) |
| 14:51.26 | defswork | yeah |
| 14:51.45 | defswork | get them to ship it with stated value of $10 |
| 14:52.05 | [TK]D-Fender | _x86_: And you constantly leave off the cost of your T1 card and the zaptel craziness you go through for it and lack of reduncy capabilities |
| 14:52.18 | _x86_ | [TK]D-Fender: yeah I was just thinking that... |
| 14:52.44 | _x86_ | hmm this sounds very feasible actually |
| 14:53.01 | _x86_ | anyone ever use one of these vegastream boxes? |
| 14:53.06 | _x86_ | i've never even heard of them |
| 14:53.18 | defswork | get one on approval |
| 14:53.25 | _x86_ | eh? |
| 14:53.28 | [TK]D-Fender | _x86_: A few of the more experienced people here have and have been happy with them. |
| 14:53.32 | defswork | someone must be willing to provide one to test etc.. |
| 14:53.59 | _x86_ | I do have an account with VoIP Supply ;) |
| 14:54.25 | _x86_ | [TK]D-Fender: who specifically? I'd like to talk to them about their experiences with it |
| 14:55.00 | defswork | I must admit I have little motivation for doing anything analog - I'm still loved up on asterisk and ip phone goodness |
| 14:55.28 | _x86_ | defswork: oh me too... but the company I work for does not wire cat5 to salespeoples' desk, just cat3/rj11 |
| 14:55.31 | [TK]D-Fender | _x86_: Can't rcall offhand |
| 14:55.41 | _x86_ | defswork: (sales people here have no computer... just telephone sales) |
| 14:55.54 | defswork | call center ? |
| 14:55.55 | [TK]D-Fender | _x86_: Stupid people. |
| 14:56.05 | _x86_ | defswork: i'd love to put Polycom IP330's everywhere :) |
| 14:56.12 | _x86_ | [TK]D-Fender: no kidding |
| 14:56.43 | [TK]D-Fender | _x86_: If they have no computers you coud save some $ and go with IP 320's |
| 14:56.50 | defswork | _x86_: I've only used aastra handsets so far |
| 14:56.53 | _x86_ | hmm audiocodes has a 24-port FXS SIP gateway too |
| 14:57.06 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-03338182a1f0f626) |
| 14:57.07 | defswork | someone here recommended them and I've had no problems so far |
| 14:57.20 | _x86_ | [TK]D-Fender: true, but the point is mute... they are not going to spend $50k or more re-wiring all the desks ;) |
| 14:57.35 | _x86_ | defswork: I'm a loyal polycom user :P |
| 14:57.50 | *** join/#asterisk af_ (n=getsmart@88-149-241-230.dynamic.ngi.it) |
| 14:58.04 | defswork | _x86_: do they tftp boot etc.. ? |
| 14:58.11 | [TK]D-Fender | defswork: You're in the UK, pricing changes things a lot. Over there, Linksys is probably your best choice. |
| 14:58.43 | [TK]D-Fender | defswork: Polycoms HTTP,FTP,TFTP and the secure variations of eatch |
| 14:58.45 | Dr-Linux | again: |
| 14:58.54 | Dr-Linux | how can i export/import astdb? |
| 14:58.56 | defswork | [TK]D-Fender: linksys models are the cheapest for sure |
| 14:58.57 | coppice | If you're in the UK, emigration is your best choice |
| 14:59.12 | defswork | coppice: oddly enough I tend to agree |
| 14:59.15 | [TK]D-Fender | coppice: Ex-pat FTW... |
| 14:59.20 | tzafrir_laptop | Dr-Linux, look for berkely db 1.85 utilities maybe? |
| 14:59.22 | chodorenko | [TK]D-Fender: http://pastebin.com/m5ae8af71 |
| 14:59.44 | defswork | coppice: NZ is top of my list |
| 15:00.08 | coppice | looking for sheep country, so you must be Welsh |
| 15:00.09 | Dr-Linux | maybe db1_dump185 tool |
| 15:00.19 | tzafrir_laptop | Dr-Linux, or generally, so dbFOO tools of the ancient berkely db version used in Asterisk |
| 15:00.21 | defswork | nah - plumb bang in the midlands |
| 15:00.47 | coppice | plumb bang in the midlands of nowhere |
| 15:00.49 | chodorenko | [TK]D-Fender: please give me excample hove i can chect present file or not ? |
| 15:01.15 | chodorenko | * check present |
| 15:02.36 | [TK]D-Fender | chodorenko: You already see what it returns.. just deal with that. |
| 15:05.06 | _x86_ | http://www.voipsupply.com/product_info.php?products_id=1901 |
| 15:05.39 | _x86_ | what do you guys think of that one? |
| 15:06.54 | [TK]D-Fender | _x86_: 4x the price of everything else... oh yeah... a bargain for sure |
| 15:07.59 | *** join/#asterisk marl (n=marl@84.13.1.46) |
| 15:08.03 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 15:09.08 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:10.02 | defswork | what poe switches do you guys use ? |
| 15:11.41 | marl | hi folks, can anyone give me any hints asto how to do the following? have a number going to my * box via IAX, want to be able to dial that number, and then imediatly dial a 4 digit extension, and have that then dial other numbers according to the 4 digit extension. eg. dial 01411231234p1234 and have that dial my mobile (using 4 digits to provide some security against non auth use) p=pause |
| 15:12.01 | [TK]D-Fender | defswork: D-Link DES-1536's (now DES-1228's) |
| 15:12.34 | _x86_ | defswork: I use HP ProCurve 3500yl |
| 15:12.51 | [TK]D-Fender | marl: Basic IVR... |
| 15:12.56 | _x86_ | defswork: 48 10/100/1000 PoE ports, 4 10/100/1000/10000 uplink ports |
| 15:13.13 | defswork | _x86_: expensive I bet :) |
| 15:13.19 | *** part/#asterisk harpal (n=Harpal@124.125.79.212) |
| 15:13.51 | _x86_ | defswork: $5000 USD |
| 15:14.05 | defswork | I'm a cheapskate :) |
| 15:16.49 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584434.dsl.bell.ca) |
| 15:18.29 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:20.01 | defswork | thanks VijayG |
| 15:20.03 | [TK]D-Fender | defswork: Get a Linksys PoE switch then |
| 15:20.12 | [TK]D-Fender | defswork: Again best pricing in UK |
| 15:20.40 | _x86_ | [TK]D-Fender: would you say vegastream > mediatrix? |
| 15:20.42 | defswork | [TK]D-Fender: just been looking at those dlink 1228s |
| 15:20.58 | *** part/#asterisk VijayG (n=vijay@58.68.47.109) |
| 15:21.03 | defswork | they aren't poe afaict |
| 15:21.40 | [TK]D-Fender | _x86_: No personal experience with Vegastream, but I haven't heard ill of them |
| 15:22.08 | coppice | streaming all the way to Vega much incur big latencies |
| 15:22.14 | defswork | aah 1228p :) |
| 15:22.15 | [TK]D-Fender | defswork: DES-1228P <- |
| 15:22.29 | *** join/#asterisk VijayG (n=vijay@58.68.47.109) |
| 15:22.30 | [TK]D-Fender | defswork: $357 USD :) |
| 15:23.05 | defswork | yeah abut 300GBP |
| 15:23.19 | *** part/#asterisk VijayG (n=vijay@58.68.47.109) |
| 15:23.24 | *** join/#asterisk jpsharp (n=jsharp@nathost.atl.agiosat.com) |
| 15:26.45 | marl | hi, if im setting up an IVR in *, is there a way to force a call to a certain option or extension according to there CALLID? eg. if a call comes from my mobile im atomaticly sent to extension 4321 and if a call is withheld/unaavailable then it goes to exten 1234 ? |
| 15:27.06 | marl | and any other numbers calling in go to the ivr |
| 15:27.34 | defswork | marl: sounds perfetcly doable to me |
| 15:28.13 | marl | ah, doable i know, any idea how to config the exten lines thow? LOL |
| 15:28.22 | [TK]D-Fender | marl: "show function CALLERID" |
| 15:28.33 | [TK]D-Fender | marl: "show application gotoif" |
| 15:28.42 | marl | ive seen it done before, i even had it running on one machine, but that was trashed and cant find any backups of it now :( |
| 15:28.52 | marl | thanks TK :) |
| 15:32.27 | *** join/#asterisk dundel (n=daniel@200.2.161.172) |
| 15:36.30 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 15:37.00 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 15:38.40 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:39.09 | *** join/#asterisk Delvar (n=Delvar@77.240.56.17) |
| 15:40.18 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 15:45.07 | fiXXXerMet | We're going to have multiple locations. Our own, then we own a few companies in a few locations. So we'd use a separate context for each, right? |
| 15:45.19 | fiXXXerMet | Then in each context, we can use whatever extensions and devices (sip.conf) that we want? |
| 15:45.39 | fiXXXerMet | So SIP/2000@loc1 != SIP/2000@loc2 ? |
| 15:46.00 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 15:46.23 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 15:47.21 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 15:47.50 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 15:47.54 | Havokmon | Do all switchtypes support callerid? |
| 15:48.24 | Havokmon | pri switch types I should say. Not getting any with national2.. not sure if I'm not sending, or not receiving |
| 15:48.54 | jpsharp | Should be part of the call setup message, if your provider is sending it. |
| 15:49.14 | Havokmon | I am the provider ;) |
| 15:49.33 | Havokmon | besides being extremely rusty, I just never worried about caller id :) |
| 15:49.44 | jpsharp | NI-2 does support callerid, so if its not there, something's amiss. |
| 15:49.52 | Havokmon | ok. thanks |
| 15:52.41 | [TK]D-Fender | fiXXXerMet: Are we talking about them all using 1 system or 1 PBX each? |
| 15:53.08 | [TK]D-Fender | Havokmon: AFAIK all PRI supports CID & DID |
| 15:53.27 | mosty | i have a polycom 550 here, i'm trying to get it to display both callerid name and number, currently it only shows the name. is this something that requires a specific firmware version to work? |
| 15:54.08 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 15:54.14 | [TK]D-Fender | mosty: it should display both.... |
| 15:54.23 | methods | is there anyway for me to connect to a phone and reconfigure it over the network ? |
| 15:54.54 | [TK]D-Fender | methods: Depends on the "phone" and the "network". |
| 15:54.55 | mosty | [TK]D-Fender, what firmware version are you running? |
| 15:55.04 | [TK]D-Fender | mosty: All sorts. |
| 15:57.24 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 15:57.36 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 15:59.06 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:59.07 | mosty | [TK]D-Fender, on the first line it shows the callerid name, on the second line it shows the phones extension (which is the sip username). i can append ${CALLERID(num)} to ${CALLERID(name)} and then the correct callerid number appears on the first line, but it's truncated |
| 16:00.39 | mosty | firmware version is 2.1.2.0078 |
| 16:01.16 | mmlj4 | anyone have any problems with teliax tech support? |
| 16:02.40 | *** join/#asterisk Federico2 (n=fede@pdpc/supporter/base/Federico2) |
| 16:02.42 | Federico2 | hi there |
| 16:03.14 | Federico2 | I'm running * behind a firewall/NAT that isn't allowing UDP traffic nor incoming TCP connections |
| 16:03.46 | Federico2 | Is there a way to set up STUN or something else to let me receive calls from the Internet? |
| 16:04.56 | mosty | Federico2, why can't you fix your firewall? |
| 16:04.58 | fiXXXerMet | [TK]D-Fender: We're going to host all of the sites here with 1 asterisk box |
| 16:06.26 | mosty | [TK]D-Fender, where on the display is callerid number displayed on your polycom phones? |
| 16:07.21 | Federico2 | mosty: because it's not mine :) |
| 16:08.03 | jpsharp | You're pretty well stuff-outta-luck. |
| 16:08.13 | Federico2 | me? |
| 16:08.25 | *** join/#asterisk p4c0 (n=dark@200.124.22.34) |
| 16:08.38 | p4c0 | hello, [TK]D-Fender have a minute? |
| 16:08.40 | jpsharp | Yep. |
| 16:09.19 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:09.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:09.40 | ronr | Federico2: do you have port 22 outgoing access? you could create some tunnel to some other server that is allowed to do stuff (dunno how reliable that is though) |
| 16:09.45 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 16:09.59 | [TK]D-Fender | fiXXXerMet: well you can't have multiple SIP accoutns with the same username, but you can have the same extensions, jsut in different contexts, yes |
| 16:10.06 | [TK]D-Fender | f4Just ask your question... |
| 16:10.09 | JayTee52 | mornin *ers |
| 16:10.34 | [TK]D-Fender | Federico2: If it isn't allowing UDP, you're baked... |
| 16:10.37 | Federico2 | ronr: nope... I'm behind a corporate firewall |
| 16:10.40 | *** join/#asterisk nixguy (n=matmoj@fw.packetfront.com) |
| 16:10.42 | [TK]D-Fender | Federico2: Read up : |
| 16:10.44 | [TK]D-Fender | ~sipnat |
| 16:10.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:11.05 | Federico2 | thanks |
| 16:11.36 | nixguy | hi im having some problems getting my pri card working properly |
| 16:11.48 | nixguy | placing calls to the pstn gives me: |
| 16:11.48 | nixguy | <PROTECTED> |
| 16:11.48 | nixguy | <PROTECTED> |
| 16:11.49 | nixguy | <PROTECTED> |
| 16:12.02 | Federico2 | exact... I'm just searching about STUN |
| 16:12.11 | nixguy | running ztcfg -vv shows my 31 channels as being configured :| |
| 16:12.30 | p4c0 | i have a ATA that will like to sniff any ideas? my network is wireless and i have one wired and one wireless interface |
| 16:12.40 | mosty | does anyone have a polycom phone that shows both callerid name and number simultaneously on incoming calls? i want to know where the number is on the display |
| 16:12.45 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:13.27 | ronr | mosty: mine shows name & SIP channel |
| 16:13.28 | [TK]D-Fender | nixguy: your "/r" is not appropriate. Remove |
| 16:13.51 | nixguy | [TK]D-Fender: hm k |
| 16:14.07 | nixguy | any idea what configfile i have it in? |
| 16:14.09 | [TK]D-Fender | p4c0: Whats to sniff for? |
| 16:14.16 | nixguy | ill check the ones under /etc/asterisk |
| 16:14.24 | mosty | ronr: does it show the sip channel underneath the name, or on the same line? |
| 16:14.29 | [TK]D-Fender | nixguy: How is it you're running * and don't know where that line comes from? |
| 16:14.34 | p4c0 | [TK]D-Fender, user agent |
| 16:14.42 | ronr | mosty: underneath |
| 16:14.53 | [TK]D-Fender | p4c0: did you set it up with *? |
| 16:15.12 | nixguy | [TK]D-Fender: im still learning :) |
| 16:15.17 | nixguy | found it |
| 16:15.25 | [TK]D-Fender | nixguy: You set this up? |
| 16:15.44 | mosty | ronr: and the sip channel that is displayed, is that your sip channel, or the caller's? |
| 16:15.53 | *** join/#asterisk MrWorta (n=root@h1210056.stratoserver.net) |
| 16:15.53 | p4c0 | [TK]D-Fender, no, it's from the provider i can't modify it, maybe just to set my ip to match the one of the provider and try it to connect to me... but isn't something esier? |
| 16:16.23 | nixguy | [TK]D-Fender: i got som help with the pri/asterisk stuff, i dont really come from the telephony part of it. |
| 16:16.33 | [TK]D-Fender | p4c0: Wireshark, etc |
| 16:16.37 | nixguy | but he couldnt finish it , so i started looking at it myself |
| 16:18.03 | nixguy | [TK]D-Fender: same msg |
| 16:18.06 | nixguy | <PROTECTED> |
| 16:18.15 | Federico2 | this is not going to be easy |
| 16:18.16 | ronr | mosty: don't know (my sip channels are mac addresses so I don't recognize them and I think by now I have all or near all caller ids overridden in asterisk so it's hard to test) |
| 16:18.17 | nixguy | i reloaded the asterisk config, and could verify that the /r wasnt there |
| 16:18.23 | *** join/#asterisk gm123 (n=vioman@d36-10-149.home1.cgocable.net) |
| 16:18.45 | ronr | there is one phone laying around telling me the sip channel, but I don't know which one |
| 16:19.16 | mosty | ronr, thanks anyway. my phone here displays it's own channel, which is not what i would expect |
| 16:19.17 | *** part/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 16:20.02 | dbtid | out of curiosity, i'm trying to go to thevoice.digium.com but it doesn't seem to exist. |
| 16:20.05 | dbtid | did something happen to it? |
| 16:21.36 | dbtid | i guess it is now theivrvoice.com ? |
| 16:21.58 | file | dbtid: thevoice went away... awhile ago? over a year? you can now order prompts directly from digium.com |
| 16:22.13 | dbtid | i'm reading the pdf book and it's mentioned there |
| 16:22.15 | nixguy | dbtid: i dont knnow about the old name, but there doesent exist such a dns record anyawy so yes probably |
| 16:22.26 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 16:22.39 | mog | dbtid, its now the far easier http://store.digium.com/productview.php?category_id=8&product_code=8IVRPROMPT&main_category_id=8 |
| 16:22.41 | mog | link |
| 16:23.17 | mog | if you go to digium.com, click store and then ivr prompts |
| 16:24.44 | dbtid | thanks |
| 16:24.44 | *** join/#asterisk didz_ (n=voce@201.19.64.193) |
| 16:24.53 | dbtid | i do love the "weasels have eaten our phone system" |
| 16:25.09 | mog | tt-weasels is my favorite prompt |
| 16:25.21 | nixguy | [TK]D-Fender: you still around? i know im close, when i call into my asterisk from the pstn i actually see my phone number and the connection its trying to make.. |
| 16:31.14 | *** join/#asterisk ddunavant (n=David@68-244-231-253.area3.spcsdns.net) |
| 16:32.06 | *** join/#asterisk af_ (n=getsmart@88-149-240-22.dynamic.ngi.it) |
| 16:32.54 | *** join/#asterisk chanko (n=chatzill@77.221.6.79) |
| 16:34.07 | chanko | Please, is there anyone with expirience with both, sangoma and digium cards? |
| 16:34.20 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:34.27 | dandre | Hello |
| 16:34.43 | chanko | Hi |
| 16:34.50 | twisted | this...is...beautiful... |
| 16:34.51 | mosty | chanko, yes |
| 16:34.55 | twisted | http://thedailywtf.com/Articles/I-am-right-and-the-entire-Industry-is-wrong.aspx |
| 16:35.50 | dandre | is it possible from the manager interface to dial an extension and then place it in some context that could be a playback message and then a record application? |
| 16:35.57 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 16:36.06 | chanko | Hi mosty ... I have to decide what card to order for ss7 signalling purposes.. |
| 16:36.41 | *** join/#asterisk kareena (n=k@unaffiliated/kareena) |
| 16:36.48 | kareena | hi |
| 16:37.08 | kareena | is there any programme that encode wav and .au to g723 and G729 format? |
| 16:37.35 | mosty | kareena, asterisk |
| 16:37.49 | kareena | can do it from windows? |
| 16:38.28 | mosty | kareena, you run asterisk on windows? |
| 16:38.34 | mosty | chanko, http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7 |
| 16:38.47 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:39.14 | kareena | yes |
| 16:39.26 | nixguy | dear god |
| 16:39.29 | nixguy | :) |
| 16:39.48 | kareena | like pbx |
| 16:39.54 | kareena | for testing |
| 16:41.28 | chanko | Yes I've seen that. Ppl behind libisup recommend Sangoma ... |
| 16:42.00 | mosty | kareena, do you have g729 licences installed on asterisk in windows? if not, then you can't encode/decode g729 |
| 16:42.11 | dandre | is it possible from the manager interface to dial an extension and then place it in some context that could be a playback message and then a record application? |
| 16:42.39 | mosty | dandre: originate command? |
| 16:43.15 | dandre | I haven't succeed in using it |
| 16:43.29 | chanko | mosty, I'd like to hear from someone with real experience with ss7 and digium cards ... |
| 16:43.51 | exvito | hi all... does anyone know of a SIP hard-phone that, when ringing on an incoming call, can have the "answer" controlled remotely by software ? (note: this is different from autoanswer) |
| 16:44.17 | mosty | dandre: pastebin the context in your dialplan, and the originate command you are trying |
| 16:44.42 | mosty | exvito, for what purpose? |
| 16:44.56 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128685315.dsl.bell.ca) |
| 16:46.01 | mmlj4 | anyone have any problems with teliax tech support? |
| 16:46.03 | exvito | mosty: integration with call centre agent software (agent gets notification of incoming call - both ring + application - and clicks button on app to answer phone) |
| 16:46.08 | jblack | mmlj4: I do. |
| 16:46.26 | mmlj4 | jblack: do tell |
| 16:46.32 | kareena | mosty yes i have two type of license one from digium and one free from intel |
| 16:46.33 | mosty | exvito, sounds like you should be using a softphone instead |
| 16:46.40 | jblack | Nah. |
| 16:46.43 | dandre | I use this command: |
| 16:46.44 | dandre | <PROTECTED> |
| 16:47.05 | mosty | kareena, then it would probably work, but i do not have experience with asterisk in windows |
| 16:47.06 | mmlj4 | np |
| 16:47.13 | dandre | but it shows me the help message of the originate command |
| 16:47.36 | exvito | mosty: ...that would be a possibility but then again, how to have the soft-phone answer when the app button is clicked ? |
| 16:48.14 | kareena | ok |
| 16:48.20 | exvito | mosty: (I'm looking for a way of having the agent interact only with the app and not with the phone interface - whichever it is...) |
| 16:48.24 | mosty | dandre, then you're not calling originate properly |
| 16:48.51 | mosty | exvito, you could turn this app *into* a softphone |
| 16:49.08 | dandre | that what I guessed but I don't know how to call it properly |
| 16:50.52 | exvito | mosty: ...that is a good idea. However it seems a bit out of our direction / skills. How would you go about it ? (developing a soft-phone from scratch is, for me at least, out of the question!) ;) |
| 16:51.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:52.09 | dbtid | ok, i'm not understanding something about dialing one sip phone from another |
| 16:52.12 | *** join/#asterisk gardo (n=gardo@121.97.198.127) |
| 16:52.14 | dbtid | in sip.conf i've defined 1000 and 1001 |
| 16:52.18 | jblack | ok |
| 16:52.20 | dbtid | in extenions.conf |
| 16:52.23 | dbtid | i've configured |
| 16:52.35 | *** join/#asterisk ashert (n=hard@ool-4574b37a.dyn.optonline.net) |
| 16:52.43 | dbtid | exten => 1000,1,Dial(SIP/1000@10.0.24.108) |
| 16:52.43 | dbtid | exten => 1001,1,Dial(SIP/1001@10.0.24.108) |
| 16:52.50 | dbtid | 10.0.24.108 is my * server |
| 16:52.54 | jblack | ~pb |
| 16:52.55 | jbot | [~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://www.rafb.net/paste |
| 16:52.58 | mosty | exvito, you could try integrating an opensource softphone. maybe you could run asterisk on these pc's, and use the manager interface |
| 16:52.59 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 16:53.01 | dbtid | well, it's 2 lines... |
| 16:53.08 | dbtid | unless you want the whole thing... |
| 16:53.15 | dbtid | which i can put them up if you want |
| 16:53.16 | jblack | peharps it would be better to use that to paste more information, rather than being so selective. |
| 16:53.21 | dbtid | ok |
| 16:53.21 | dbtid | brb |
| 16:53.37 | *** part/#asterisk duncanh (n=dhutty@SHANGRILA.net.cmu.edu) |
| 16:53.44 | tzanger | just call it the BSA |
| 16:53.47 | tzanger | Bigass Storage Array |
| 16:54.00 | *** join/#asterisk `paul (n=aldee@125.252.68.68) |
| 16:54.06 | tzanger | haha |
| 16:54.07 | tzanger | wrong channel |
| 16:54.12 | jblack | tzanger: A good name, if I ever heard one. :) |
| 16:54.40 | dbtid | http://rafb.net/p/B08qjr85.html |
| 16:55.00 | outtolunc | BASS big ass storage system <G> |
| 16:55.07 | outtolunc | hehe |
| 16:55.07 | `paul | evrything was running well and then i installed a mailserver on the machine where asterisk was running suddenly i cant register my sip phone... help pls |
| 16:56.29 | exvito | mosty: ok.. asterisk on the PC's would be overkill and not doable as they'll be running windows; the second issue about having the app be a soft-phone as well is very interesting but difficult since we're targetting a lightweight web-based front end... (but still possible with webbased / java/activex softphones) |
| 16:58.29 | dbtid | jblack, did you look at my paste? could you please give me an idea as to what my mistake is? |
| 16:58.58 | Uatec | *sigh* |
| 16:59.05 | Uatec | SER isn't as complicated as i first thought |
| 16:59.13 | Uatec | but getting the bloody thing working with radius is a bummer |
| 17:01.36 | chanko | paul, could you check is your asterisk running "asterisk -rvvv", for the beggining ... |
| 17:02.52 | mosty | exvito, you might be able to use SIP NOTIFY messages to tell a phone to answer, but i don't know which phones if any support that. |
| 17:03.46 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:04.37 | *** part/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 17:05.12 | exvito | mosty: ...that would be something more along my initial thoughts. Now, either via SIP NOTIFY of via some "phone-specific" HTTP service, the question remains (as it would be the easiest to implement): is there any such phone ? I guess I'll post this question to the asterisk-users mailing list... |
| 17:06.17 | *** join/#asterisk gongoputch (n=gongoput@74.95.184.161) |
| 17:07.37 | hmmhesays | I wonder how hard it would be to write a small app to alert that there is mail in your voicemailbox |
| 17:07.48 | Qwell | umm |
| 17:07.51 | Qwell | externnotify |
| 17:08.18 | Qwell | it runs an external script/app when you get new vm |
| 17:09.48 | hmmhesays | Qwell, well I mean some application that is notified the same way say your phone is |
| 17:10.46 | `paul | chanko: iam running it at vvvvvr |
| 17:11.19 | `paul | chanko: the problem occured when i installed a mail server... maybe its blocking some ports or sumthin |
| 17:13.24 | marl | can anyone spot what ive done wrong with his exten line? exten => 01411231234,n,GotoIf($["{CALLERIDNUM}" = "012312345"?emmaoutbound|stephen|1:) |
| 17:13.43 | Qwell | "]? |
| 17:14.00 | putnopvut | And put a $ before {CALLERIDNUM} |
| 17:14.09 | Qwell | and I'm not sure what happens if you have a : there.. |
| 17:14.19 | chanko | paul, ok and with netstat -naup | grep asterisk" |
| 17:14.52 | chanko | ... you can see line with port 4060 |
| 17:15.03 | Qwell | 4060? |
| 17:15.16 | russellb | and don't use CALLERIDNUM, use CALLERID(num) |
| 17:15.24 | dbtid | anyone? http://rafb.net/p/B08qjr85.html |
| 17:15.38 | dbtid | i'm trying to dial from one sip phone @ 1000 to another @ 1001 |
| 17:16.10 | marl | ok, will try that in a few mins, bck soon thanks |
| 17:16.16 | chanko | sorry 5060 |
| 17:16.25 | `paul | udp 0 0 0.0.0.0:5060 |
| 17:16.30 | `paul | its right there |
| 17:18.02 | chanko | paul, and iptables -L shows you there is no blocking on udp ports |
| 17:19.52 | `paul | wait |
| 17:21.22 | `paul | there are few lines with DROP,ACCEPT and REJECT |
| 17:22.14 | *** join/#asterisk cjk (n=cjk@d90-129-39-85.cust.tele2.lu) |
| 17:22.47 | cjk | hi, after a forkcdr asterisk does not fill out duration and billsec so my calls is like it has never been answered. is that normal? |
| 17:22.49 | *** join/#asterisk Kigh (n=kai@213.239.211.111) |
| 17:27.37 | *** join/#asterisk khronos (n=khronos@c-66-229-159-175.hsd1.fl.comcast.net) |
| 17:27.55 | Kigh | hi |
| 17:28.27 | mosty | cjk, forkcdr is a hack |
| 17:28.34 | Kigh | i have two asterisk servers (1.2.19) that are behind a NAT and both connect to the same asterisk on the internet.. |
| 17:28.58 | Kigh | i now realize, that the asterisk on the internet treats connection attempts from both machines the same |
| 17:29.08 | cjk | mosty, its a really important application |
| 17:29.48 | mosty | cjk, it's still a nasty hack |
| 17:30.03 | cjk | hmmmm |
| 17:30.09 | Kigh | what do i need to do so that the asterisk on the internet to treat both peers as different ones? |
| 17:30.54 | Kigh | both machines must register with credential on the internet, but one of the peers is never registered. (asterisk matches the second peer as the first one, based on the public IP i think) |
| 17:31.28 | mosty | cjk, you might be able to use chan_local with the /n option instead, depending on what you're trying to do |
| 17:31.33 | Kigh | i want them to use different contexts, but at the moment they use the same . |
| 17:32.06 | cjk | mosty, well user one calls user two, i need an outgoing cdr for one and an incominv cdr for two |
| 17:32.29 | mosty | cjk, why? |
| 17:33.03 | mosty | cjk, i mean, wouldn't the CDR's be identical? |
| 17:33.36 | cjk | mosty, no, after the cdr i set some different variables |
| 17:33.44 | cjk | i mean after the fork |
| 17:33.48 | cjk | i change the accountcode |
| 17:34.16 | mosty | cjk, so you can bill both parties? |
| 17:34.36 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 17:34.36 | cjk | mosty, yes |
| 17:34.42 | Ritzerisk | i cant for the life of me get iax exts to register i get like output connection refused in the iax debug |
| 17:34.50 | Ritzerisk | i get like an error 29 |
| 17:35.15 | Ritzerisk | and they are registering no 127.0.0.1 localhost |
| 17:35.26 | mosty | cjk, you can probably use userfield to set the callee's account code, and just use a single cdr |
| 17:35.56 | cjk | mosty, thats more of a hack |
| 17:36.09 | cjk | mosty, but the problem is with blind transfers |
| 17:36.21 | cjk | 302 redirects etc... |
| 17:36.45 | khronos | How can I tell what password a certain sip peer is trying to login to my server with? |
| 17:37.18 | khronos | I have a peer that keeps failing nad I'd like to have my server match what the peer is trying to send. |
| 17:37.54 | dbtid | sigh |
| 17:38.23 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:38.56 | mosty | cjk, the asterisk CDR code is messy in current versions of asterisk, there is no good way to deal with all possibilities at the moment that i know of. i think i read that the cdr code is being redesigned for asterisk 1.6 though |
| 17:39.16 | cjk | im on 1.6 |
| 17:39.45 | codefreeze | yeah, I've done some work, but none of it is finished enough to commit. |
| 17:41.27 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:42.31 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 17:43.04 | Ritzerisk | would there be a reason or a limit i can adjust for IAX2 exts to register as the localhost |
| 17:43.16 | *** part/#asterisk chanko (n=chatzill@77.221.6.79) |
| 17:43.30 | dbtid | i don't mean to be a bleating horse, but would someone mind looking at this? |
| 17:43.37 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:43.41 | dbtid | jblack perhaps? |
| 17:43.42 | jblack | back. |
| 17:43.44 | dbtid | anyone? http://rafb.net/p/B08qjr85.html |
| 17:43.49 | jblack | Sorry. I had to go clean the kitchen |
| 17:43.50 | dbtid | i didn't know you'd left |
| 17:43.53 | dbtid | oh, i understand |
| 17:43.59 | dbtid | i do a lot of housework too |
| 17:44.03 | dbtid | (i'm assuming you're male) |
| 17:44.08 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:44.08 | dbtid | nice to work from home. |
| 17:44.20 | jblack | I wasn't expecting it. My daughter does most of the cooking in the house. Went into the kitchen, realized it had to be clean. Yeah, I'm male. |
| 17:44.34 | dbtid | how old is your daughter? |
| 17:44.37 | jblack | 13. |
| 17:44.49 | jblack | She cooks well, but can get forgetful about cleaning up after herself. |
| 17:45.05 | jblack | Can you remind me as to your symptoms? |
| 17:45.33 | Havokmon | Wow. I had to finish the Mac and Cheese my 14 year old started yesterday :/ |
| 17:46.16 | jblack | Yeah. Night before last (she has the flu), she made lemon baked fish. Night before that, parmasean encrusted chicken bread. |
| 17:46.37 | jpsharp | Damn. Can I adopt her? |
| 17:46.40 | jblack | dbtid: Remind me what's wrong for you? |
| 17:46.42 | Havokmon | Wow. That's impressive. |
| 17:47.18 | jblack | jpsharp: If you want me to sell her, then I gotta tell you about all of the features of this model. Full yes/no sir support, straight A student. Also does laundry. |
| 17:48.15 | jblack | Well, considering her skills in coffee making... let's say an even upteen billion dollars? |
| 17:48.40 | jpsharp | Done. Just don't try to cash the check for a day or two. |
| 17:51.34 | jblack | dbtid ran away |
| 17:52.05 | Havokmon | heh 'Make Offer: Amazon-descendant, phone-obsessed, hearing impared and strong willed 14 yr old girl. Can cook Raman noodles, fill dishwasher, and recite PETA propaganda.' |
| 17:52.28 | Ritzerisk | http://pastebin.comm561dbe28 |
| 17:52.28 | jpsharp | So, a typical teenager. |
| 17:52.29 | jblack | you poor man. |
| 17:52.31 | *** join/#asterisk wglenncamp (n=wglennca@c-68-63-251-212.hsd1.ky.comcast.net) |
| 17:52.57 | Havokmon | lol.. hey my wife has "Mom's night", here is as good a place as any ;) |
| 17:53.00 | jblack | Mine cares about animals, but not so much that she won't eat them. ;) |
| 17:53.10 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 17:53.18 | jpsharp | I just have a "very spirited" 5 year old daughter. |
| 17:53.20 | wglenncamp | Need an opinion for a small call center deployment. Interfacing with T1 line... Digium Cards or Mediatrix (or Audiocodes) digital gateway? |
| 17:53.30 | Havokmon | yeah - the vegetarian thing is new this week :/ |
| 17:53.34 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:53.56 | hmmhesays | bah the polycom's are using a refer on attended transfers |
| 17:54.03 | jpsharp | I'd go with a digium card. One less piece of hardware to mess with. |
| 17:54.04 | dbtid | soirry |
| 17:54.15 | `paul | question my machine seems to block incoming ports used by asterisk... how do i edit iptables? |
| 17:54.19 | dbtid | jblack, the problem is that * doesn't like my extensions |
| 17:54.23 | wglenncamp | But quality-wise... Which is better. |
| 17:54.26 | dbtid | [Jan 9 11:50:33] WARNING[24401]: app_dial.c:1112 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 17:54.45 | jblack | Ok, that's useful. Which number were you trying to dial? |
| 17:54.51 | wglenncamp | I am looking for the best quality. And trying to limit the load on the server. |
| 17:54.53 | dbtid | i've got this laptop, a VM, a windows machine, and two irc sessions going :) |
| 17:55.05 | dbtid | well, i'm trying to dial 1000 from 1001 and/or 1001 from 1000 |
| 17:55.12 | dbtid | oh wait |
| 17:55.24 | dbtid | i don't know that i defined what extension 1000 IS |
| 17:55.26 | dbtid | did i? |
| 17:55.28 | dbtid | is that my problem? |
| 17:55.38 | dbtid | in sip.conf [1000] is kind of like a user account isn't it? |
| 17:55.42 | jblack | You do in internal. |
| 17:55.46 | jblack | And include it in phones. |
| 17:55.47 | dbtid | um |
| 17:55.55 | gm123 | can anyone help with Asterisk SLA using SIP trunks?? |
| 17:56.00 | dbtid | well, then i did that |
| 17:56.04 | dbtid | because [phones] |
| 17:56.04 | jblack | Yes, 1000 is a user account. |
| 17:56.04 | dbtid | is |
| 17:56.17 | mosty | wglenncamp, sangoma card with hardware echo cancellation |
| 17:56.18 | dbtid | ok, so then the association is there |
| 17:56.19 | jblack | I notice that they're not requiring auth, btw. |
| 17:56.27 | dbtid | well, right now, this is all a test |
| 17:56.29 | *** join/#asterisk af_ (n=getsmart@88-149-240-22.dynamic.ngi.it) |
| 17:56.32 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:56.32 | dbtid | i want to be able to make these work |
| 17:56.38 | dbtid | in the real environment, there will be auth |
| 17:56.41 | dbtid | i'm just playing now |
| 17:56.41 | jblack | You may want to consder adding "secret PASSWORD" to them. |
| 17:56.46 | jblack | Because here's your problem.... |
| 17:57.02 | jblack | Oh, no, that's not it. |
| 17:57.09 | dbtid | heh |
| 17:57.18 | dbtid | well, doesn't SIP/1000 refer to sip.conf's [1000] entry?? |
| 17:57.19 | wglenncamp | My distributor doesn't offer sangoma. Digium, Audiocodes, or Mediatrix? |
| 17:57.26 | jblack | Yes, it does. |
| 17:57.29 | dbtid | i should say |
| 17:57.29 | Qwell | find a new distributor? |
| 17:57.32 | jblack | * is griping your Dial is wrong, though. |
| 17:57.33 | dbtid | SIP/1000@10.0.24.108 |
| 17:57.35 | Qwell | oh, wait |
| 17:57.37 | jblack | As in syntax. |
| 17:57.38 | Qwell | . != , |
| 17:58.08 | dbtid | exten => 1000,1,Dial(SIP/1000@10.0.24.108) |
| 17:58.15 | dbtid | exten => 1001,1,Dial(SIP/1001@10.0.24.108) |
| 17:58.24 | wglenncamp | I am pretty happy with them.. Anywho.. You think that an internal interface card has better quality than an external gateway? |
| 17:58.35 | dbtid | that connects exten 1000 w/ sip.conf [1000] right? |
| 17:58.47 | dbtid | anthat connects exten 1001 wd / sip.conf [1001] right? |
| 17:58.54 | jblack | Yes, it does. We already covered that. |
| 17:59.08 | Qwell | wglenncamp: I'm biased, but go with Digium |
| 17:59.09 | dbtid | right |
| 17:59.12 | dbtid | sorry |
| 17:59.13 | wglenncamp | HAHA |
| 17:59.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:59.38 | wglenncamp | Hey, the server will be running Asterisk BE though. ;) |
| 17:59.54 | jblack | Hmm. Is this your actual extensions.conf, as is on the hard drive? And did you reload * after you changed it? |
| 17:59.57 | file | yay I can afford dinner |
| 18:00.02 | Qwell | well, you're gonna have a hard time getting support with anything else... |
| 18:00.08 | Qwell | file: O.o |
| 18:00.14 | wglenncamp | I see... |
| 18:00.26 | wglenncamp | Digium it is then... Thanks! |
| 18:00.35 | gm123 | nobody has done it?? |
| 18:01.03 | dbtid | tes |
| 18:01.06 | dbtid | oh |
| 18:01.26 | dbtid | i bet i didn't reload |
| 18:01.51 | dbtid | i just did |
| 18:01.54 | dbtid | it fails |
| 18:02.03 | jblack | What's the error this time? |
| 18:02.14 | dbtid | <PROTECTED> |
| 18:02.14 | dbtid | [Jan 9 13:01:42] NOTICE[25045]: chan_local.c:570 local_alloc: No such extension/context 1000@default creating local channel |
| 18:02.14 | dbtid | [Jan 9 13:01:42] NOTICE[25045]: app_dial.c:508 wait_for_answer: Unable to create local channel for call forward to 'Local/1000@default' (cause = 0) |
| 18:02.23 | jblack | Great. |
| 18:02.32 | jblack | Remember how I said "you want secrets for these phones?" |
| 18:02.38 | dbtid | yes |
| 18:02.42 | dbtid | i sure do |
| 18:02.50 | jblack | Well, the phones aren't logging into their sip contexts, so they're dropping to default. |
| 18:03.01 | dbtid | ok |
| 18:03.06 | jblack | Add secrets to both of them. The user name will be the sip context, the pass will be as you set. |
| 18:03.08 | Ritzerisk | hmmm im having an issure registering more then 2 iax2 exts is there like a force or allow more then 2 |
| 18:03.09 | dbtid | i'll update that in the phones and in sip.conf |
| 18:03.09 | Ritzerisk | <PROTECTED> |
| 18:03.09 | Ritzerisk | <PROTECTED> |
| 18:03.09 | Ritzerisk | <PROTECTED> |
| 18:03.10 | Ritzerisk | <PROTECTED> |
| 18:03.19 | jblack | ritzerisk: Type ~pb |
| 18:03.47 | mosty | Ritzerisk, pastebin your iax.conf |
| 18:04.51 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 18:07.50 | tzafrir_laptop | jbot, tell Ritzerisk about pb |
| 18:08.35 | *** join/#asterisk de4dpixel (n=de4dpixe@unaffiliated/de4dpixel) |
| 18:09.19 | dbtid | jblack, if i do something like secret=1000 |
| 18:09.21 | dbtid | is that ok? |
| 18:09.26 | dbtid | or does it have to be in quotes or something? |
| 18:09.34 | dbtid | because one of the phones doesn't appear to be registering |
| 18:09.38 | jblack | how about secret=unguessable |
| 18:09.39 | dbtid | unless i put the password in wrong... |
| 18:09.48 | dbtid | well, in this little environment, it really doesn't matter |
| 18:09.50 | jblack | but it can be a number. |
| 18:09.54 | dbtid | and one of the phones will only let me use numbers |
| 18:10.20 | jblack | I don't know very much about IP phones. |
| 18:10.33 | dbtid | it's ok |
| 18:10.37 | dbtid | this one is kind of odd |
| 18:10.45 | dbtid | it's got an "Authentication" and a "Password" entry |
| 18:10.50 | dbtid | i must have just used the wrong one |
| 18:11.03 | jblack | Many things have "userid" and "authentication". |
| 18:11.18 | jblack | The authentication one is almost always not the one you want. |
| 18:11.38 | dbtid | sigh |
| 18:11.40 | dbtid | still not working |
| 18:11.43 | jblack | That's used to authenticate as one person, then be addressed as someone else. |
| 18:12.02 | dbtid | well i have both values set at the same thing |
| 18:12.19 | jblack | Well, if you just want to play with things a bit before you actually fit it, put in [default] in your extensions.conf, "include => internal" |
| 18:12.27 | jblack | before you fix it, that is. |
| 18:12.43 | jblack | Oh wait. I see a problem in your sip.conf |
| 18:12.50 | jblack | Never mind. |
| 18:13.06 | dbtid | [Jan 9 13:12:58] NOTICE[25138]: chan_local.c:570 local_alloc: No such extension/context 1001@default creating local channel |
| 18:13.12 | dbtid | why is it going for [default] |
| 18:13.16 | dbtid | when context=phones |
| 18:13.17 | dbtid | in sip.conf? |
| 18:13.30 | dbtid | at least, that looks what it's trying to do |
| 18:13.50 | jblack | Because the phones aren't logging in. :) |
| 18:13.53 | dbtid | lol |
| 18:14.03 | dbtid | i'll shut one of them off |
| 18:14.05 | jblack | run asterisk -r, then run "sip show peers" |
| 18:14.07 | dbtid | and just work with one of them |
| 18:14.11 | dbtid | i have |
| 18:14.13 | dbtid | here's the results |
| 18:14.23 | dbtid | 1001/1001 10.0.24.132 D 5060 Unmonitored |
| 18:14.26 | dbtid | i shut off the one phone |
| 18:14.31 | dbtid | that information is correct |
| 18:14.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:14.35 | jblack | Huh, ok, that one is good, then. |
| 18:14.41 | dbtid | but i'm gonna shut off that phone now and watch the process |
| 18:14.42 | jblack | Did you reload your sip.conf too? |
| 18:14.46 | dbtid | um |
| 18:15.13 | dbtid | yeah i had |
| 18:15.20 | dbtid | but i just reloaded everything and i'm powering up 1000 |
| 18:15.52 | dbtid | how do i know it logged in properly? |
| 18:15.56 | dbtid | i can paste the logs |
| 18:15.57 | jblack | sip show peers. |
| 18:16.18 | dbtid | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline] |
| 18:16.22 | dbtid | it shows that 1000 is there |
| 18:16.26 | dbtid | 1001 is powered off |
| 18:16.40 | dbtid | is there more detailed info in 'sip show peer 1000'? |
| 18:16.44 | jblack | Ok. So, dial 1000 from 1000. See if you get congestion. |
| 18:16.46 | jblack | There is. |
| 18:17.13 | dbtid | [Jan 9 13:16:54] NOTICE[25145]: chan_local.c:570 local_alloc: No such extension/context 1000@default creating local channel |
| 18:17.18 | dbtid | same problem |
| 18:17.24 | jblack | hmm. |
| 18:17.41 | dbtid | ok all the algo say MD5 |
| 18:17.52 | dbtid | that's just the encryption being sent i assume |
| 18:17.56 | jblack | That's another way to authenticate. I haven't used it. |
| 18:18.07 | dbtid | the phones are apparently both doing it :) |
| 18:18.36 | jblack | Offhand, I don't see what's wrong. |
| 18:18.45 | dbtid | ok |
| 18:18.48 | dbtid | thank you for all your time |
| 18:18.52 | dbtid | i'm gonna go get some lunch |
| 18:18.54 | dbtid | bbiab |
| 18:21.54 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.10) |
| 18:22.31 | RipeR-81 | hello.. anyone here has used asterisk with vonage ? |
| 18:24.21 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 18:25.18 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:25.41 | *** join/#asterisk TwoCards (n=ozanblot@88.240.217.32) |
| 18:25.46 | TwoCards | hello |
| 18:26.10 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 18:26.20 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:26.40 | TwoCards | anyone alive for helping me about configuring zaptel.conf and zapata.conf for two cards (d)in one system ? |
| 18:26.54 | TwoCards | anyone alive for helping me about configuring zaptel.conf and zapata.conf for two cards (digium te120p and tdm800p)in one system ? |
| 18:26.56 | *** join/#asterisk Blinkiz (n=niklas@h-89-233-204-231.wholesale.rp80.se) |
| 18:28.07 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 18:28.37 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 18:29.07 | mort_gib | TwoCards: What's your problem?? |
| 18:29.55 | TwoCards | i have successfully installed trixbox... added extension, did some modem configs etc, also IVR and Voicemail is okay |
| 18:30.17 | TwoCards | i can dial out to PSTN and of course i can receive calls from PSTN |
| 18:30.19 | Qwell | ~trixbox |
| 18:30.20 | jbot | extra, extra, read all about it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 18:30.48 | mort_gib | TwoCards: I think trixbox is like a swearing word here :-) -Thanks Qwell |
| 18:30.56 | TwoCards | so TDM800 is working... yeeaah i realise it :) |
| 18:30.58 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 18:31.09 | TwoCards | TDM is configured perfect i believe |
| 18:31.11 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1096745879.dsl.bell.ca) |
| 18:31.14 | TwoCards | hold on i'm sending via private |
| 18:33.22 | iCEBrkr | haha.. trixbox == swear word. |
| 18:35.18 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:36.38 | Nugget | "Hi everyone, I need help. I'm trying to run trixbox with realtime patches on an eMachines server I found in the storage closet. I've got 40 users with a mixture of grandstream phones and x-lite (unregistered). I'm using four clone x100p cards I bought off ebay and I compiled a pirated version of the g729 codec. Can you help me set up fax over sip?" |
| 18:36.47 | *** join/#asterisk dataworm (n=bla@modemcable040.107-81-70.mc.videotron.ca) |
| 18:37.15 | Nugget | "oh, and I'm in a hurry because all the phones aren't working and my boss is mad" |
| 18:37.32 | nDuff | gaaah. |
| 18:37.45 | jpsharp | Hey, I'm having the same problem! |
| 18:40.07 | mosty | Nugget, ask #trixbox - we don't do that here |
| 18:40.27 | mosty | bah |
| 18:40.43 | mosty | i should have realised nobody is quite that stupid |
| 18:40.48 | Nugget | heh |
| 18:40.51 | Qwell | mosty: you'd be surprised |
| 18:40.51 | jpsharp | Hook, line... |
| 18:42.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:42.32 | outtolunc | and stinker |
| 18:43.38 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 18:43.45 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
| 18:43.45 | *** mode/#asterisk [+o twisted] by ChanServ |
| 18:43.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:46.13 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 18:48.39 | _x86_ | Nugget: hahahaha |
| 18:48.55 | *** join/#asterisk Blinkiz (n=niklas@h-89-233-204-231.wholesale.rp80.se) |
| 18:50.05 | *** join/#asterisk orioni (n=orion@92.60.24.44) |
| 18:54.33 | mosty | i'm trying to get Pickup working with asterisk 1.4.17, no matter what context i put in Pickup(123@context), asterisk says "pickup_exec: No target channel found for 123", even though 123 is in the ringing state. what could i have missed? |
| 18:54.36 | *** join/#asterisk NirS (n=chatzill@87.68.59.206.cable.012.net.il) |
| 18:54.42 | NirS | hello all |
| 18:54.51 | *** join/#asterisk pLr (n=plr@unaffiliated/plr) |
| 18:56.42 | *** join/#asterisk TwoCards (n=ozanblot@88.240.217.32) |
| 18:59.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:04.41 | *** join/#asterisk orioni (n=orion@92.60.24.44) |
| 19:05.42 | *** join/#asterisk Blinkiz (n=niklas@h-89-233-204-231.wholesale.rp80.se) |
| 19:11.24 | dbtid | jblack, what context should [general] have in sip.conf? |
| 19:11.36 | dbtid | it WAS context=default |
| 19:11.45 | chodorenko | Hi |
| 19:11.46 | dbtid | i set it to context=phones |
| 19:11.51 | dbtid | but that causes weird things :) |
| 19:11.59 | dbtid | LOTS Of error messages, and CONGESTION messages |
| 19:12.00 | *** part/#asterisk orioni (n=orion@92.60.24.44) |
| 19:14.13 | *** join/#asterisk DaPrivateer (n=matt7229@66.92.79.218) |
| 19:16.23 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
| 19:16.29 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 19:17.00 | dexpdx | what would the correct way to signal a polycom to auto-answer from the dialplan? |
| 19:18.36 | dataworm | If I want to try SRTP witch Asterisk branch should I get? |
| 19:19.37 | *** join/#asterisk angom_w (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:20.32 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 19:24.06 | dexpdx | in otherwords does SipAddHeader work? |
| 19:29.38 | Corydon76-dig | Yes, it does |
| 19:29.47 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:30.05 | *** join/#asterisk Victor_Yure (n=aaa@200.166.132.131) |
| 19:34.38 | tzafrir_laptop | hi NirS |
| 19:36.21 | Havokmon | ugh.. all these itsp's have $25 DID port fees :( |
| 19:43.47 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:44.02 | dbtid | jblack, i fixed it |
| 19:44.04 | lmadsen | Havokmon: ya, because they have to pay the port fees |
| 19:44.17 | dbtid | in extensions.conf i was using SIP/1001@10.0.24.108 |
| 19:44.21 | dbtid | what i needed was |
| 19:44.24 | dbtid | SIP/1001 |
| 19:44.29 | dbtid | it all works now |
| 19:44.36 | lmadsen | ya, rarely do you need the @foo part |
| 19:44.47 | lmadsen | that should be defined by your peer in sip.conf |
| 19:49.20 | hmmhesays | what the hell is tos=0x68 that doesn't add up |
| 19:51.39 | *** join/#asterisk angom_w (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:52.08 | Havokmon | lmadsen: yeah I understand - but paying $2000 just to port numbers so customers don't get a busy kinda sucks |
| 19:52.34 | lmadsen | it sure does :) |
| 19:56.54 | ddunavant | ? |
| 19:56.58 | ddunavant | nvm |
| 19:57.58 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 19:58.23 | *** part/#asterisk angom_w (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 20:04.27 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 20:07.49 | captiancrash | is it possible to make SIP calls from an Astereisk server to an Iwatsu PBX that has IPNET cards? |
| 20:08.05 | Qwell | do the IPNET cards let it do SIP? |
| 20:08.35 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 20:08.48 | captiancrash | Qwell, I suppose that's more the question I should have asked. Do the IPNET cards do SIP... |
| 20:08.51 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 20:08.52 | Qwell | no idea |
| 20:14.34 | Havokmon | captiancrash: I think Iwatsu's are the old Fujitsu's... and if so, no, that's only for FIPN |
| 20:19.00 | kyron | Where should I look for how the calculation of bandwidth per line? I need to know which type of Internet connection I will require for my installations ;) |
| 20:20.24 | jpsharp | 1.21 GIGABITS! |
| 20:20.44 | jpsharp | But seriously, bandwidth depends on what codec you use. |
| 20:20.56 | kyron | rajiv, didn't notice you were here: Assuming that you're the one responsible behind net-misc/asterisk, I wanted to know if it was still active and if it required much work to bring it up to 1.4.1? |
| 20:21.07 | kyron | jpsharp, hehe, conservative figures heh... |
| 20:21.36 | kyron | inter-office codec I control, but my provider only accepts u-ulaw |
| 20:22.06 | jpsharp | 75kbps per call, about. |
| 20:22.35 | jpsharp | ulaw is 64kbps + 10-12Kbps overhead. |
| 20:23.15 | kyron | and we're talking bits not bytes...of course.. |
| 20:23.33 | jpsharp | Right. |
| 20:23.42 | jpsharp | 75 kilobits per second. |
| 20:23.47 | kyron | hmm...so ulaw is quite close to the PRI BW |
| 20:23.58 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:24.20 | kyron | so I'm looking into an ADSL connection and wondering how many calls I could carry over it....and also wondering if I should play with the MTU |
| 20:24.25 | jpsharp | It is the same. The voice on a PRI is digitized using ulaw or alaw, depending on which side of the atlantic you're on. |
| 20:24.46 | jpsharp | nah, the MTU should be fine. Voice packets are pretty small. |
| 20:25.54 | kyron | jpsharp, yeah, I was actually thinking of lowering it..actually, it |
| 20:25.55 | chodorenko | please answer me howeto i can check exist file or now ? and if exist play as background sond ? |
| 20:26.06 | kyron | it's the window size that should be tweaked in this case.. |
| 20:26.31 | jpsharp | No, the window size shouldn't change. This is UDP not TCP. |
| 20:26.32 | kyron | wow, thats almost understandable |
| 20:26.49 | kyron | jpsharp, as I as typing that I also kicked myslef in the head |
| 20:27.21 | jpsharp | Now that takes talent. |
| 20:28.22 | kyron | jpsharp, flexibility ;) |
| 20:28.48 | chodorenko | please answer me howeto i can check exist file or now ? and if exist play as background sond ? |
| 20:29.22 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 20:31.13 | davevg-btwtech | chodorenko, look at the STAT function to see if a file exists |
| 20:34.22 | chodorenko | davevg-btwtech: if file note exist this fungtion return is "" and if a use "IF" fungtion then i return error http://pastebin.com/d77a6d213 |
| 20:36.03 | davevg-btwtech | chodorenko, here is a sample from one of my dialplans GotoIf($["${STAT(e,/var/lib/asterisk/sounds/btwtech/${messageid}.ulaw)}"="1"]?:dial) |
| 20:36.06 | chodorenko | davevg-btwtech: its my error in code ? jr this bug & |
| 20:38.12 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 20:38.51 | chodorenko | davevg-btwtech: Yeh ...... You Supper !!!!! |
| 20:39.20 | chodorenko | davevg-btwtech: "fsdafsdf"="1" its my error , no one manual this no write |
| 20:40.23 | CrazyTux | Hello guys, anyone ever seen this: [Jan 9 14:38:15] WARNING[8007]: chan_sip.c:4852 process_sdp: Unsupported SDP media type in offer: image 18340 udptl t38. I'm trying to receive a fax inbound, however receiving that warning. |
| 20:41.35 | twisted | yes, that's called T.38 |
| 20:42.39 | CrazyTux | twisted, yes, but what do I need to do to support it |
| 20:43.27 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:45.31 | *** part/#asterisk Slingky (n=Maxime@modemcable111.80-201-24.mc.videotron.ca) |
| 20:45.38 | CrazyTux | twisted, 1.4 is suppose to have proper support, no/ |
| 20:48.32 | jpsharp | I'm running 1.4.4 and its still hit & miss whether it works or not. |
| 20:49.23 | puppet | well |
| 20:49.30 | puppet | t38 support in * as i get it isnt complete |
| 20:49.42 | puppet | it cant be endpoint as i get it |
| 20:51.56 | jpsharp | I've been trying to use it as a relay between T.38 capable ATAs and my Quintum CMS960. It sort of works with Grandstream ATAs, doesn't work with the VoipInc ATAs I have, and if I try to use it with the Quintum ASG200s I have, it really irritates *. |
| 20:53.39 | *** join/#asterisk mtryfoss (n=mtryfoss@6.81-166-192.customer.lyse.net) |
| 20:54.38 | mtryfoss | is it possible to pass a variable through the queue application? (from the callling channel to the called channel) |
| 20:57.01 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:57.15 | dbtid | interesting; this UTstarcom F3000 has rlogin WIDE OPEN. |
| 20:58.15 | puppet | cany anyone recomend the cheapest sip-phone there is avaiable or cheapest atabox |
| 20:58.19 | *** join/#asterisk trippss (n=ss@72.20.150.196) |
| 20:58.22 | puppet | that is something to have |
| 20:58.24 | Qwell | ~cheap |
| 20:58.25 | jbot | [cheap] a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:59.18 | puppet | Qwell: not for a big company ;) |
| 20:59.26 | puppet | Qwell: just need something that "works" |
| 20:59.33 | puppet | eveyone dont need a ciscobox at home :) |
| 20:59.36 | trippss | so i'm installing a new test * server from ubuntu packages - i'm going to use it for local sip phones and trunk all activity over iax to remote * server . . . do i need to install zaptel drivers? |
| 20:59.47 | puppet | trippss: yes |
| 20:59.49 | puppet | trippss: for ztdummy |
| 21:00.15 | trippss | puppet: ok thanks |
| 21:00.33 | mtryfoss | puppet: linksys spa-942 is a good and cheap phone |
| 21:01.59 | puppet | MrWorta: i could have one of those at home, but my GF colleges, at her company just need a basic BASIC phone with like answer/hang on function, or best way is a atabox then? |
| 21:04.10 | fujin | trippss: yes and no, some things require a timing source (MeetMe, trunk=yes for IAX2, app_page, some MOH) |
| 21:04.24 | fujin | for example I don't have ztdummy at all, yet have a fully functional pure-voip system |
| 21:04.29 | fujin | (we opted to not use MeetMe) |
| 21:04.40 | fujin | or IAX2, for that matter :0 |
| 21:05.31 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 21:05.45 | jstew | Greetings |
| 21:06.28 | jstew | Can anyone recommend a good traffic shaper? I have a T1 that's saturated alot of the time and need some QoS for our voip calls. |
| 21:06.50 | [hC] | for something as small as a t1, id suggest a wrt54gl running dd-wrt |
| 21:06.56 | [hC] | or a linux box with a traffic shaper on it |
| 21:07.33 | puppet | [hC]: openwrt ftw ;P |
| 21:07.36 | jstew | I have some traffic shaping on our firewall using pf but things get messed up during heavy downloads from our users |
| 21:07.47 | [hC] | puppet: same thing, dd-wrt just doesnt make you use the CLI :) |
| 21:07.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:07.55 | puppet | [hC]: not openwrt niehter |
| 21:08.04 | [hC] | puppet: new UI? |
| 21:08.13 | puppet | OpenWrt Kamikaze - With X-Wrt Extensions 7.07 |
| 21:08.16 | jstew | We have 2 DMZs on our firewall though |
| 21:08.18 | *** join/#asterisk AndyGraybeal (n=andy@node122.34.251.72.1dial.com) |
| 21:08.19 | puppet | webif² |
| 21:08.40 | jstew | So I was thinking of a router that sits between the T1 line and the fw to do nothing but traffic shape :D |
| 21:09.34 | [hC] | puppet: cool, ill have to dig up some screenshots.. but ive been really happy with dd-wrt too |
| 21:09.56 | [hC] | jstew: pf configured properly could do it |
| 21:10.28 | puppet | [hC]: dd-wrt lacked some stuff tho, and its not gpled |
| 21:10.31 | jstew | hmmm you think so? I have 2 dmzs as well as a lan to take into consideration. |
| 21:10.46 | puppet | asit should be |
| 21:10.52 | puppet | [hC]: dd-wrt breaks licensing rules |
| 21:10.59 | puppet | last i checked anyway |
| 21:11.07 | [hC] | puppet: what did it lack? i dont lose sleep over them breaking licensing. |
| 21:11.25 | puppet | was something i needed anyway but iu dont rememebr right now ill come up with it soon :) |
| 21:11.34 | jstew | Downloads are a problem because once the packet gets accepted into the interface, it's already too late to do any QoS because the bandwidth has already been used. |
| 21:12.56 | hmmhesays | blahbittblahblah |
| 21:13.14 | hmmhesays | yeah you can only really traffic shape your uploads |
| 21:13.40 | [hC] | it does also somewhat work for downloads in my experience |
| 21:13.52 | [hC] | because the interface will start queueing and slowing down the download |
| 21:13.55 | tzafrir_laptop | trippss, try: m-a a-i zaptel |
| 21:14.03 | jstew | bleh, I'm just going to get a separate data line and be done with it |
| 21:14.14 | tzafrir_laptop | that is: after installing the package zaptel-source |
| 21:14.18 | jstew | I've spent too many hours on this anyway lol |
| 21:14.23 | hmmhesays | you don't really need to traffic shape your downloads anyway, usually your upload buffer is going to cause the problem |
| 21:14.25 | trippss | fujin: thanks - i didn't think it was absolutely necessary but we're using trunk for IAX2 . . . cool. |
| 21:15.18 | trippss | tzafrir_laptop: sorry for being dense - what do you mean by m-a a-i |
| 21:15.50 | fujin | module-assistant is the tool to get debianized source of kernel modules |
| 21:16.06 | trippss | ok gotcha |
| 21:16.17 | tzafrir_laptop | trippss, this is a command to run in the terminal, as root |
| 21:20.13 | trippss | module assistant is ma what is ai |
| 21:21.52 | trippss | tzafrir_laptop: were you sending me something? i haven't gotten anything. tia |
| 21:22.07 | tzafrir_laptop | trippss, no |
| 21:22.25 | tzafrir_laptop | trippss, a-i stands for auto-install |
| 21:22.52 | tzafrir_laptop | build a -modules deb package and install it |
| 21:23.19 | puppet | http://www.atcom.cn/En_products_AG188.html what do u guys thkn? |
| 21:23.27 | Qwell | ~cheap |
| 21:23.28 | jbot | extra, extra, read all about it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 21:23.28 | errr | if my voicemail volumes are too low when being emailed I see in voicemail.conf you can do volgain= what values should I try to raise it? |
| 21:23.30 | tzafrir_laptop | Alternatively: m-a build zaptel # just build a deb package |
| 21:24.01 | tzafrir_laptop | Qwell, any better IAX+SIP ATA? |
| 21:24.39 | puppet | tzafrir_laptop: and tis one here dont cost a fortune and i need 2-3 |
| 21:25.10 | trippss | tzafrir_laptop: i was going to install zaptel-source, module-assistant prepare, m-a build zaptel, then dpkg -i the deb package. sound right? |
| 21:25.43 | tzafrir_laptop | trippss, m-a a-i saves you the need for dpkg -i |
| 21:25.51 | trippss | gotcha |
| 21:26.16 | tzafrir_laptop | And usually m-a prepare is not really needed |
| 21:29.23 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 21:29.59 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 21:31.11 | MrTelephone | is there any good solution to low voicemail volumes? |
| 21:32.23 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-32-175.lns10.syd7.internode.on.net) |
| 21:34.08 | *** part/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 21:35.05 | *** join/#asterisk darkseer (n=kredinel@201.19.94.89) |
| 21:35.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:37.10 | *** part/#asterisk lirakis (n=lirakis@65.200.189.220) |
| 21:37.31 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 21:37.34 | darkseer | anyone can help me with a little question? what machine i need to build an asterisk with 500 lines without slowdowns? :) (sorry my english) :P |
| 21:38.01 | puppet | darkseer: 500 lines wich type, what codec, etc etc? |
| 21:39.17 | darkseer | humm lemme see |
| 21:39.59 | trippss | mmm maybe I needed to install zaptel before asterisk? i don't see ztdummy . . . |
| 21:40.26 | trippss | and maybe i need libpri or not? |
| 21:40.40 | Havokmon | MrTelephone: advertise a phone sex service? ;) |
| 21:40.41 | JT | yes zaptel needs to be installed before asterisk if you want asterisk to use it |
| 21:40.56 | Havokmon | That'll increase the volume on your vm system ;) |
| 21:43.07 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:46.22 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 21:46.25 | MrTelephone | haha |
| 21:46.37 | MrTelephone | sound volume not call volume |
| 21:47.25 | genz | Using Originate, like the first example in the wiki, the caller-id gets sent to the context, but not the exten |
| 21:47.32 | genz | Any idea on how to get it to send to both? |
| 21:55.28 | tzafrir_laptop | which phones, besides polycoms, can be provisioned over http? |
| 21:56.11 | trippss | lsmod | grep zap shows i've got a crc_ccitt module in use i haven't seen nor do i think i need. how can i remove this from loading? |
| 21:56.15 | tzafrir_laptop | trippss, you don't need libpri. However, if you use asterisk from packages it is built with libpri support |
| 21:56.45 | tzafrir_laptop | trippss, this means that zaptel uses that module |
| 21:56.45 | trippss | tzafrir_laptop: yeah I noticed that the ubuntu pkgs have it in there . . . |
| 21:56.49 | tzafrir_laptop | no problem here |
| 21:57.01 | trippss | never seen it on any other box running ztdummy . . |
| 21:57.47 | JT | ztdummy has nothing to do with libpri |
| 21:57.57 | JT | libpri is for pri signalling |
| 21:58.32 | trippss | mmm i suppose crc_ccitt must be built into the kernel on the other boxes . . . who knew ;) |
| 21:59.30 | genz | Anyone know much about Originate? |
| 21:59.47 | *** join/#asterisk Roa (n=roa@unixmexico/Roa) |
| 22:01.27 | outtolunc | genz: what is wrong with the info on the wiki |
| 22:02.16 | genz | outtolunc: That system only sends caller ID to the context (internal) but the the exten (external). |
| 22:02.17 | *** join/#asterisk G-nerd (n=AskMe@dhcp-077-249-041-129.chello.nl) |
| 22:02.22 | G-nerd | hello guys! |
| 22:03.21 | outtolunc | when you 'originate' the channel TECH type needs to allow the callerid rewrite, which also means in the case of provider/pstn they must also allow it |
| 22:04.00 | genz | outtolunc: Which also means when the provider is giving you control, and you don't send it, you show up as private. Which is the problem I'm living in now. |
| 22:04.34 | outtolunc | genz: note, most 'provider' do the callerid 'name' lookup from their databases regardless, some allow you to rewrite the callerid 'num' |
| 22:05.53 | genz | outtolunc: I have a T1 Pri, and can send/block my CID from all the phones, its the click-to-call functionality of Originate that's being ornery. |
| 22:06.05 | outtolunc | so set it |
| 22:06.43 | Netgeeks | ah, you mean when you originate the channel originate allows you to specify callerid, but the next part - the exten that you connect the originated channel to, doesn't? |
| 22:07.13 | genz | Netgeeks: Thank you. Yes. |
| 22:07.21 | Netgeeks | Yeah, we ran into that too |
| 22:07.28 | genz | Did you find anything? |
| 22:07.42 | Netgeeks | no, I band-aided and bubble gumed it |
| 22:08.01 | *** join/#asterisk RoyK (n=roy@ip-2-14-149-91.dialup.ice.no) |
| 22:08.16 | genz | That sounds better to me than not being able to use it at all for some numbers. Can you point me in the right direction? |
| 22:08.35 | genz | Or, at least, in the direction you followed... |
| 22:09.24 | Netgeeks | I basically created a call request record (set up a db entry in an external db) and called an extension that was actually the request.id, then I loaded that db entry and set up the call. Note that I was already using external db for other thiings and the setup record already existed, so I wasn't adding alot. |
| 22:09.31 | Netgeeks | thats probably not the case for you |
| 22:10.31 | darkseer | anyone have only a little knowledge to tell me usin any type of codec etc, to work with 500 lines without problems? if u already saw anythin like it workin =p u have an ideia! hehehe at moment i need only this info! :) |
| 22:10.52 | genz | outtolunc: its ok, you problably just skipped over some lines of my messages |
| 22:11.23 | genz | Netgeeks: oy, my kingdom for an elegant solution... yours is a pretty creative method |
| 22:11.26 | outtolunc | yeah obviously my fault <G> haha |
| 22:11.38 | *** join/#asterisk Victor_Yure (n=aaa@200.166.132.131) |
| 22:13.12 | lmadsen | darkseer: eh? I tend to use G.711u, or G.729a for high density (as long as I don't have to transcode) |
| 22:14.31 | Netgeeks | I don't know the problem with originate, it could possibly be an easy fix, or both you and I could be not doing something a little simpler like for the channel use a local channel, and have that local channel store the callerid in an asterisk db or maybe a global variable specific to the channel, then in the extension portion of the originate have the extension code pull the data out and clear the db tree if thats what you used. |
| 22:14.31 | Netgeeks | Like I said bandaid and bubble gum tho |
| 22:14.35 | darkseer | and the machine to keep it working? |
| 22:15.43 | Netgeeks | of course the problem with local channels and all that stuff is that your cdr records can get ... um.... interesting to read |
| 22:15.46 | outtolunc | Netgeeks: he most likely just needs to fill other 'Variable's with that callerid info, and set them back to callerid vars in the context on connect |
| 22:15.56 | genz | Netgeeks: Couldn't we use VARIABLE, have the config read it, and then set it over there |
| 22:16.12 | Netgeeks | genz: I think thats just what out recommended |
| 22:16.26 | genz | Netgeeks: So I see |
| 22:18.20 | mocker | Anyone here using Aspect w/ Asterisk? |
| 22:19.50 | darkseer | lmadsen hum, what type of hardware i need to keep these 500 lines workin without slowdowns? |
| 22:20.28 | lmadsen | darkseer: probably 4GB RAM + 2x dual or quad-core CPUs, and ideally a RAMdrive |
| 22:20.43 | mocker | celeron |
| 22:20.56 | darkseer | mocker lol |
| 22:21.00 | lmadsen | see TFoT2 for the noticeable number of additional channels you can run when running in a ramdrive :) |
| 22:21.06 | lmadsen | you can thank file for that test |
| 22:21.17 | G-nerd | lmadsen, you could also buy telephonecard using DSP processors |
| 22:21.18 | JT | running 500lines soley on one machine is a dumb idea though |
| 22:21.19 | lmadsen | darkseer: note NO TRANSCODING :) |
| 22:21.28 | lmadsen | JT: I would agree |
| 22:21.46 | lmadsen | better to spread that over 2-4 boxes |
| 22:21.58 | darkseer | hum |
| 22:21.59 | lmadsen | although it does complicate the logic a bit |
| 22:22.09 | genz | Netgeeks: Variable: TESTVAR=Bob in the c2c should it answer as ${TESTVAR} in a noop, right? |
| 22:22.09 | G-nerd | Cyberthech has telephone cards compatible with Asterisk |
| 22:22.27 | lmadsen | you need to use more tools like DUNDi and replicating data (or storing some of it in the database -- I like to use func_odbc) |
| 22:22.32 | Netgeeks | genz: that is what I understand |
| 22:22.51 | genz | outtolunc: That's what you'd say, too? |
| 22:23.08 | G-nerd | 500 lines, how does one want to connect all those cables to a pc? |
| 22:23.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:23.33 | Havokmon | G-nerd: 110 ;) |
| 22:23.46 | G-nerd | how much lines can a pc handle fysically? |
| 22:23.57 | JT | "fysically"? |
| 22:23.59 | outtolunc | genz: use 'Variable: __PHONENUM=xxxxxxxxxx' (dont' forget \r\n), and in extensions use exten => _X.,n,Set(__PHONENUM=${CALLERID(num)}) |
| 22:23.59 | darkseer | via ip phone :P |
| 22:24.10 | G-nerd | so ther is actually already a limit |
| 22:24.19 | drmessano-LT | 500 lines on one machine |
| 22:24.19 | outtolunc | those __ are important |
| 22:24.19 | G-nerd | yes all those phone cables |
| 22:24.26 | JT | G-nerd: PRI or IP would probably be the best |
| 22:24.33 | drmessano-LT | Can I be your on-call support? $100/hour |
| 22:24.34 | JT | 500 analogue lines would be stupid |
| 22:24.42 | G-nerd | why? |
| 22:24.51 | JT | because analogue friggen sucks arse |
| 22:24.52 | G-nerd | not cost effective? |
| 22:24.53 | JT | it's shit |
| 22:25.03 | JT | poor call control, too many cables |
| 22:25.10 | JT | poor audio |
| 22:25.10 | drmessano-LT | But i'm not punching down 500 lines |
| 22:25.23 | G-nerd | hmm I guess you are right JT |
| 22:25.23 | JT | get with the 90s, get PRIs |
| 22:25.31 | [hC] | anything above 10-12 lines, no sense to use analog lines. |
| 22:25.49 | outtolunc | whoops |
| 22:25.54 | outtolunc | wrong code |
| 22:25.55 | G-nerd | ok, |
| 22:25.57 | JT | the level is closer to 5-6 where i am |
| 22:26.19 | G-nerd | but could an ethernet cable handle 500 telephone lines? |
| 22:26.28 | G-nerd | I mean virtual telephone lines :) |
| 22:26.37 | [hC] | over IP? yeah of course. |
| 22:26.39 | outtolunc | genz: in extensions do something like exten => _X.,n,Set(CALLERID(num)=${PHONENUM}) |
| 22:26.40 | JT | i guess it's possible to put 500 g.711 calls over sip over 100Mbit/s |
| 22:26.43 | G-nerd | really? |
| 22:26.50 | JT | yes, obviously |
| 22:26.51 | [hC] | its absolutely possible to do 500 g729 calls. |
| 22:26.59 | [hC] | and yeah you could probably do g711 too. |
| 22:27.00 | JT | 1000/0.085 |
| 22:27.03 | JT | err |
| 22:27.09 | G-nerd | oooh ok, than we are talking about at least 100MB connection |
| 22:27.12 | JT | 100/0.085 |
| 22:27.17 | JT | 100Mbit/s |
| 22:27.19 | JT | not 100MB |
| 22:28.01 | genz | outtolunc: Ok, but when you set it in the Originate action, how do you access it from the normal method. The noop doesn't return the value as one would expect |
| 22:28.06 | G-nerd | so with a good Linux server with Ateris installed on it could handle 500 telephone lines simultaneously? |
| 22:28.37 | outtolunc | genz: you need to be 'originate'ing from a manager session |
| 22:29.24 | JT | Ateris? |
| 22:29.28 | G-nerd | Asterisk |
| 22:29.48 | JT | i'm sure you can get machines to handle 500 channels |
| 22:29.57 | JT | but it's advisable to have more than one |
| 22:30.00 | JT | as machines fail |
| 22:30.31 | G-nerd | But personally I still don't like the concept to use the pc as a DSP, telephonecards should do the work |
| 22:30.46 | G-nerd | yes I understand JT |
| 22:30.49 | JT | then asterisk is probably the wrong thing for you |
| 22:30.59 | G-nerd | hmmm, why? |
| 22:31.11 | JT | the whole philosophy of zaptel is using the host cpu for processing |
| 22:31.16 | JT | read up about zaptel |
| 22:31.28 | G-nerd | yes you're right |
| 22:31.42 | JT | although some people have now realised that doesn't make much sense for things like transcoding to g.729 |
| 22:31.58 | G-nerd | That's why these cards circuit were actually quite simple |
| 22:32.00 | JT | so digium have even released devices more on the dsp side of things |
| 22:33.47 | G-nerd | JT, the reason why I don't like the concept, is because the CPU is actually intended to do more OS related stuff and not doing number crunching things (well, number crunching in another manner) |
| 22:34.36 | JT | so should we get DSPs for microsoft word? |
| 22:34.44 | JT | another DSP for internet explorer? |
| 22:35.01 | fetcher | after an Asterisk server outage, a lot of Polycom SIP phones (IP 501's) seem to get into a "stuck" state where they can't properly re-register. Manually power cycling the phones fixes it. |
| 22:35.18 | fetcher | Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.96 <--- this is the error message that appears on the * console, over and over, once for each affected phone |
| 22:35.27 | G-nerd | no no, DSP is more for processing signals, like audio, or radiosignals etc.... |
| 22:35.35 | fetcher | and only when * goes down briefly but the phones don't |
| 22:36.06 | G-nerd | like the acronym stands for |
| 22:36.56 | tzafrir_laptop | G-nerd, the problem is that developing hardware takes much more time |
| 22:37.05 | tzafrir_laptop | And hence it costs much more |
| 22:37.37 | G-nerd | maybe it is more an money issue |
| 22:37.59 | tzafrir_laptop | Furthermore, developing software, it is cheaper to fix your mistakes, and hence the price of improving is not as high |
| 22:38.41 | JT | pc cpus are quite powerful now that dsps often aren't needed |
| 22:38.42 | G-nerd | but than Asterisk should run on Linux with no GUI or something, just a very stripped Linux and Asterisk |
| 22:38.56 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:39.25 | tzafrir_laptop | That is not to say that DSPs are of no use. |
| 22:39.44 | JT | sure |
| 22:39.50 | G-nerd | yes, you are right, this is actualy the best way to have very cheap pbx and stable |
| 22:41.03 | G-nerd | well I mean asterisk as PBX as one thing |
| 22:41.19 | G-nerd | I know the possibility is endless |
| 22:42.28 | kyron | Yeah, as a comparison, traditional PBXs ran under Motorola's 68K processors... quite a few leaps back from present multi-core multi-GHz processors ;) |
| 22:42.32 | G-nerd | I have some C/C++ experience, but I have never programmed for Linux OS' s applications, only that windose stuff |
| 22:42.49 | G-nerd | oooh really? |
| 22:43.00 | kyron | G-nerd, don't start programming under Linux, you won't be able to go back to windows |
| 22:43.13 | G-nerd | well, that is actually the point |
| 22:43.36 | G-nerd | I don' t hate windose, but the strategists of windose |
| 22:44.03 | G-nerd | If they can, you pay even a license for everytime you start windose |
| 22:44.03 | kyron | G-nerd, yes, really, traditional PBX are pure hardware (digital + analog) oriented beasts that are robust, complex, and tend to have some cryptic language to configure (hey...quite like *!) |
| 22:44.03 | tzafrir_laptop | G-nerd, but then again, some people take the "DSP" approach too seriously: http://www.rowetel.com/blog/?p=40 |
| 22:44.51 | fetcher | G-nerd: extra money for every reboot, huh? Now there's a guaranteed money maker ;) |
| 22:45.12 | G-nerd | well, DSP' s were maybe more important when there were no powerfull processors |
| 22:46.02 | G-nerd | I work at an telecom comapny and they work with windose systems only |
| 22:46.21 | G-nerd | that is why I want to learn Linux well |
| 22:46.22 | kyron | G-nerd, DSPs are still very important and aren't close to being obsoleted by mainstream processors |
| 22:46.27 | fetcher | yeah, I remember some old 68k-era Apple Macs that had Motorola 56001 DSP coprocessors. Those were dropped when the switched to PowerPC architecture |
| 22:47.01 | G-nerd | no I know that kyron, there are certain field where DSP' s are necessary |
| 22:47.05 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 22:47.16 | G-nerd | like control loopbak |
| 22:47.18 | kyron | Telecom with windows only...yerk!... that's like Fido (Microcell) that had their cell system bogged down to a halt because they were "smart" enough to use Exchange and windows on their infrastructure (Melissa virus days) |
| 22:47.53 | kyron | G-nerd, well...any field where analog meets digital ;) |
| 22:47.58 | G-nerd | yes yes, but I was happy to have a job |
| 22:48.28 | G-nerd | yep kyron, I agree with you |
| 22:48.38 | kyron | G-nerd, Microsoft is employment security, you'all always have to do (repair) stuff around a Microsoft infrastructure. |
| 22:48.52 | Qwell | "always"? |
| 22:48.54 | Qwell | no |
| 22:49.01 | G-nerd | the clients already told us to use Linux, but we don' t have experience on Linux |
| 22:49.11 | kyron | Hey, the'll even make sure you keep learning...let's revamp the GUI! |
| 22:49.18 | *** join/#asterisk mercestes (n=merceste@uslec-66-255-0-96.cust.uslec.net) |
| 22:49.21 | kyron | tsk...get some! |
| 22:49.24 | kyron | :) |
| 22:49.38 | mercestes | if you are on IRC, chances are your not getting any from the opposite sex. |
| 22:49.41 | JT | kyron: he probable means for windows for management |
| 22:49.46 | kyron | Qwell, ok, in a closed isolated environment...maybe not |
| 22:49.55 | JT | kyron: most of telecomms does still happen in dsp |
| 22:49.57 | Qwell | isolated? no |
| 22:49.58 | G-nerd | Now I' m a bit familiar with Linux, the hard thing is still configuring with bash |
| 22:49.59 | mercestes | oh, that's not what he meant by "get some." |
| 22:50.01 | mercestes | nevermind then. |
| 22:50.10 | kyron | LOL |
| 22:50.29 | kyron | did I say something suggestive without my consent? |
| 22:50.31 | kyron | gneheheh |
| 22:50.56 | mercestes | <kyron> tsk...get some! |
| 22:51.02 | G-nerd | but who said that programming in Linux is very addictive? |
| 22:51.15 | kyron | G-nerd, well, leraning _that_ approach is the best since it's the most flexible and independant of a GUI...much more portable and useful ...career wise... |
| 22:51.33 | kyron | mercestes, yeah, didn't think it could be interpreted in that manner.. |
| 22:51.38 | G-nerd | yes kyron, it is the hard way but the best way |
| 22:51.55 | JT | hard is debatable |
| 22:52.12 | JT | considering the wide availability of free development tools |
| 22:52.20 | JT | easy comes to mind |
| 22:52.34 | kyron | G-nerd, anyhoo, if you want to go "500 lines on a machine" get serious redundant hardware, of which you have 2 choices: hyper redundant hardware or redundant server (offline hotswap) |
| 22:53.03 | G-nerd | ooh that a good point, which development tools do Linuxprogrammers use? I mean for writing Linux services etc... |
| 22:53.51 | kyron | JT, it's hard to configure * using vi and a command line. It's easy doing it with FreePBS...but you don't learn much (ironically, this is how I did it for the moment but I do intend to rebuild my * machine under Gentoo) |
| 22:53.59 | G-nerd | kyron, and what about using multiple machines and those machines connected with each other |
| 22:54.03 | *** join/#asterisk angryuser (i=aster@df01t2-213-44-88-241.d4.club-internet.fr) |
| 22:54.19 | *** join/#asterisk Maliuta (i=nikolai@119.11.99.20) |
| 22:54.27 | JT | kyron: i think it's easier in many ways to hand code asterisk |
| 22:54.39 | kyron | G-nerd, keep a clone (and I really mean clone) as a spare and swap it if required |
| 22:54.50 | JT | you have precise control, with freepbx, who knows wtf it will do |
| 22:54.52 | G-nerd | they say a big problem must be cut into smaller problems ;) |
| 22:55.03 | puppet | JT: the code gets cleaner |
| 22:55.05 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 22:55.16 | kyron | JT, yes, once you know it well enough... ie: inserting 500 SIP extensions is a joke to script...don't even think of using a GUI for that |
| 22:55.16 | puppet | in freepbx u get like, 1000 config files |
| 22:55.33 | G-nerd | hahahahaha |
| 22:55.34 | puppet | kyron: realtime ftw |
| 22:55.58 | kyron | puppet, oh...is that why I get lost when attempting to understand the config?... good to know it's not "standard" |
| 22:56.13 | puppet | kyron: the config isnt hard if u have seen a workign config, then read samples >( |
| 22:56.16 | puppet | :) |
| 22:56.44 | kyron | puppet, that's what I wanted to do with FreePBX...but I found the 1k config files a little overwhelming |
| 22:57.13 | puppet | haha yeah ;P |
| 22:57.19 | puppet | u cant do it ther u have to try manually :) |
| 22:57.23 | JT | it does everything in stupid obfuscated ways |
| 22:57.36 | JT | it has random useless AGI |
| 22:57.44 | JT | and horrible dialplans |
| 22:57.46 | G-nerd | Who runs his own biznez selling Asterisk machines? |
| 22:58.20 | puppet | JT: bump |
| 22:58.22 | G-nerd | anyway, that says enough hahaha |
| 22:58.45 | JT | puppet: ? |
| 22:58.47 | puppet | G-nerd: well wich biznez uses freepbx? |
| 22:58.56 | JT | lots... |
| 22:58.57 | puppet | JT: bump as in, true very true about agi and stuff |
| 22:59.00 | G-nerd | why not? |
| 22:59.09 | JT | unfortunately a lot use freepbx |
| 22:59.16 | [hC] | I own a 'biznez' selling asterisk systems |
| 22:59.18 | mercestes | I have several clients who used freepbx. That is why they are my clients now. |
| 22:59.24 | [hC] | although i really refuse to say biznez |
| 22:59.27 | [hC] | Are you 19? heh |
| 22:59.31 | angryuser | gooed evening everybody |
| 22:59.31 | G-nerd | I read on a forum they changed the freepbx skin of the GUI |
| 22:59.40 | JT | [hC]: that's unfair on most 19yos |
| 22:59.43 | G-nerd | and put their won logo on it |
| 22:59.48 | [hC] | JT: you're right im sorry. |
| 22:59.54 | JT | fonality owns it now |
| 23:00.00 | JT | they now sell a proprietary version |
| 23:00.02 | JT | trixbox pro |
| 23:00.05 | JT | own trixbox that is |
| 23:00.13 | [hC] | they still have trixbox CE too |
| 23:00.17 | puppet | [hC]: i said it cause he said it ;P |
| 23:00.37 | tzafrir_laptop | trixbox != freepbx. And they're not the only one distributng freepbx |
| 23:01.03 | G-nerd | hmmm |
| 23:01.06 | angryuser | i am searching a good tutorial for asterisk realtime, like hiw extensions are managed and examples of tables, any ideas besides voipinfo? |
| 23:01.14 | angryuser | *how |
| 23:01.27 | puppet | angryuser: read the wiki |
| 23:01.54 | puppet | voipinfo says it all |
| 23:01.55 | drmessano | You dont need Trixbox to have FreePBX.. thank god |
| 23:02.06 | bkruse | you do not need either, thank God. |
| 23:02.10 | G-nerd | well guys, I got to go the bed, Got to work tomorrow, programming C# for windose, but I' ll dream tonight about programming C/C++ on Linux :) |
| 23:02.22 | bkruse | C#? |
| 23:02.32 | bkruse | just playin :] |
| 23:02.40 | G-nerd | pfew |
| 23:02.46 | G-nerd | ...zzzzzzz |
| 23:02.56 | bkruse | night |
| 23:03.09 | puppet | c# is easy to code ;P |
| 23:03.14 | G-nerd | c u next time guys, thanks for talking |
| 23:03.36 | G-nerd | (well not in C++ and C for ARM microcontroller) |
| 23:03.41 | G-nerd | ok but got to go |
| 23:05.17 | angryuser | i am thinking of writing my manager in windev wish me luck ;) |
| 23:05.34 | hmmhesays | windev? |
| 23:06.01 | angryuser | yes google wil help you |
| 23:06.55 | angryuser | <puppet> wiki was updated recently, more info, thx for pointing me back |
| 23:07.31 | puppet | http://www.voip-info.org/wiki-Asterisk+RealTime |
| 23:07.37 | puppet | says it all, i just did that and it worked splended |
| 23:08.26 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 23:09.00 | *** join/#asterisk mog (n=mog@nat/digium/x-745253e681f537b7) |
| 23:09.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:11.16 | AndyGraybeal | russellb: should i install the asterisk-alsa module? |
| 23:11.37 | AndyGraybeal | wooh, softfax for asterisk :) nice |
| 23:11.57 | russellb | AndyGraybeal: sure, if you want |
| 23:12.10 | russellb | it just lets you make calls from the asterisk CLI using an ALSA sound device |
| 23:12.11 | AndyGraybeal | do i need to it work with jack and asterisk and pd? |
| 23:12.16 | russellb | no |
| 23:12.18 | AndyGraybeal | ah okay |
| 23:12.23 | russellb | and the jack stuff is _brand_ new |
| 23:12.28 | russellb | it's in a developer branch |
| 23:12.39 | russellb | svn co http://svn.digium.com/svn/asterisk/team/russell/jack asterisk-jack |
| 23:12.47 | AndyGraybeal | jack russell :) |
| 23:12.51 | russellb | :) |
| 23:12.53 | AndyGraybeal | okay rock on thank you |
| 23:12.57 | russellb | np |
| 23:13.49 | Qwell | AndyGraybeal: that's funny... |
| 23:14.33 | AndyGraybeal | lemme get this asterisk installed... my distro has packaged the asterisk version 1.2.13-21 ...... this is probably not the version i want eh? |
| 23:15.16 | AndyGraybeal | Qwell: haha.. in another channel .... this guy name 'sprouts' is in.. and 'russellb' was right above him in my nickname list... and i read "brussellspouts" at first glance. |
| 23:15.25 | AndyGraybeal | er.. i mean "brussellsprouts" |
| 23:15.26 | drmessano | How can you download something, install it, and while you get ALL the updates to the app and OS, it's never REALLY updated.... |
| 23:15.33 | Qwell | AndyGraybeal: nice |
| 23:15.33 | russellb | AndyGraybeal: 1.2.13 is pretty old |
| 23:15.37 | russellb | a few years probably |
| 23:15.50 | AndyGraybeal | russellb: tha's what i thought.... most evil... go openSUSE packagemanagement |
| 23:15.52 | russellb | AndyGraybeal: that developer branch is the bleeding edge plus the jack stuff, heh |
| 23:16.10 | drmessano | Furthermore, how can I download say 2.0 of something, upgrade all the pieces to new, and still have 2.0 |
| 23:16.24 | mocker | Hey, I like 1.2.13. :) |
| 23:16.46 | mocker | .17 ftw. |
| 23:17.34 | angryuser | can onyone help me solve the problem withe the message "rtc: lost some interrupts at 1024 hz" ? all last versions of * |
| 23:17.58 | angryuser | it is a non stop flood |
| 23:18.23 | drmessano | What kind of versioning scheme results in "I have 2.0, which has all the code of 2.8 but shows 2.0" |
| 23:18.28 | hmmhesays | looks like a problem with your real time clock |
| 23:19.27 | drmessano | Qwell: I have a question |
| 23:19.28 | angryuser | i have enabled then emulation in kernel recompiled, but nothing happens.... |
| 23:19.33 | angryuser | the |
| 23:19.42 | Qwell | drmessano: I have an answer |
| 23:20.06 | drmessano | If you created an app |
| 23:20.09 | drmessano | Lets call it.. |
| 23:20.12 | Qwell | no |
| 23:20.17 | Qwell | let's email it |
| 23:20.57 | drmessano | Matchbox |
| 23:21.00 | drmessano | Thats a GOOD name |
| 23:21.09 | Qwell | okay, matchbox |
| 23:21.18 | drmessano | and I gave you 1.0 to download |
| 23:21.26 | drmessano | err |
| 23:21.35 | drmessano | you gave me 1.0 to download |
| 23:21.41 | drmessano | and I use it.. la la la la |
| 23:21.41 | Qwell | why would I give you 1.0? |
| 23:21.48 | drmessano | Update it 17 times |
| 23:21.59 | drmessano | Because you cant wait for me to use it |
| 23:22.07 | Qwell | did you buy it? |
| 23:22.10 | drmessano | TO where 1.1.7 is out |
| 23:22.14 | drmessano | No, open source |
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| 23:22.29 | drmessano | and I came back to you for support |
| 23:22.37 | Qwell | I'd charge you $1M. |
| 23:22.40 | drmessano | and you have this half ass numbering scheme |
| 23:22.48 | drmessano | Where I actually am on 1.0 |
| 23:22.48 | Qwell | wait, go back |
| 23:22.56 | drmessano | 1.0 with all updates |
| 23:23.00 | drmessano | No 1.1.7 |
| 23:23.05 | Qwell | I wouldn't use a half-ass numbering scheme |
| 23:23.06 | drmessano | But 1.0* |
| 23:23.10 | drmessano | Say you did |
| 23:23.16 | Qwell | okay, say I did |
| 23:23.17 | drmessano | Just.. for shits and giggles |
| 23:23.22 | drmessano | Ok |
| 23:23.22 | drmessano | So |
| 23:23.37 | drmessano | I have 1.0* *Not 1.1.7 but all the files of 1.1.7 |
| 23:23.54 | drmessano | How the HELL would you keep up with who has what and how to support it? |
| 23:24.10 | Qwell | you don't |
| 23:24.11 | drmessano | Because technically... |
| 23:24.19 | drmessano | I can have 1.0 ALA 1.1.7 |
| 23:24.23 | drmessano | I can have 1.0.1 ALA 1.1.7 |
| 23:24.28 | drmessano | etc etc |
| 23:24.29 | drmessano | Ok |
| 23:24.42 | drmessano | I was beginning to think something liek that would make sense on this planet |
| 23:24.54 | drmessano | That *I* was the crazy one for thinking "WTF" |
| 23:24.57 | Qwell | are you going somewhere with this? O.o |
| 23:25.06 | drmessano | My fav subject |
| 23:25.10 | drmessano | That Matchbox sounding app |
| 23:25.27 | Qwell | I would've called it cocopuffsbox. |
| 23:25.28 | drmessano | Description of the number system |
| 23:25.32 | drmessano | ROFL |
| 23:25.39 | drmessano | FrostedFlakesBox |
| 23:26.08 | drmessano | "Hi, I am using 2.0 not 2.0, but 2.0-ish 2.1.7-like-ish" |
| 23:26.14 | drmessano | "Oh, continue" |
| 23:27.05 | drmessano | I got a Slurpee like Brain Freeze from reading that |
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