| 00:06.15 | p3nguin | ryduh: If you can wait a few minutes, I'll take a look. Just have to finish up some stuff real quick. |
| 00:06.59 | ryduh | Great. thanks. I think it has to do with asterisk matching incoming calls to users/peers |
| 00:07.22 | ryduh | but voip.ms sends the phone number in the From sip line. |
| 00:07.59 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:08.13 | p3nguin | ryduh: I'm looking at you voipms sip context, and it says to use the voipsms-outbound dialplan context. |
| 00:09.21 | ryduh | i should be receiving the calls on ACCT1, which should match with voipms-slo and the [new-order] context |
| 00:09.24 | ryduh | right? |
| 00:09.53 | p3nguin | That's correct. |
| 00:10.13 | p3nguin | What does the [new-order] dialplan context look like? |
| 00:12.09 | ryduh | http://pastebin.com/d6eb0445 |
| 00:12.33 | ryduh | and [voipms-outbound] http://pastebin.com/d106583f0 |
| 00:13.24 | p3nguin | What's will all the includes? |
| 00:13.39 | p3nguin | I don't think that is even valid. What happens when you run dialplan reload? |
| 00:14.25 | CcRnp | hey guys do you know how to use Sphinx4 with asterisk ! |
| 00:14.29 | ryduh | it doesn't show any errors. would you like to see the log? |
| 00:15.21 | ryduh | the includes are for closing during holidays/certain hours |
| 00:16.07 | ryduh | http://pastebin.com/d4110945f |
| 00:16.25 | p3nguin | Just because you include it doesn't mean that it's going to work. |
| 00:16.45 | p3nguin | <PROTECTED> |
| 00:17.08 | p3nguin | Does the [closed-holidays|*|*|1|jan] exist? |
| 00:17.15 | p3nguin | doubtful that it does. |
| 00:18.14 | ryduh | I was going off of this: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
| 00:18.15 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:18.39 | ryduh | but i'm currently trimming my dialplan and getting to basics to figure out the problem |
| 00:19.18 | p3nguin | Do you understand my reasoning, though? |
| 00:19.32 | p3nguin | If the [closed-holidays|*|*|1|jan] does not exist, nothing is going to be used from it. |
| 00:19.43 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 00:20.06 | p3nguin | I just do a bunch of GotoIfs for hours and things. |
| 00:20.22 | ryduh | right but [closed-holidays] exists |
| 00:20.32 | p3nguin | Then fix your includes. |
| 00:20.34 | ryduh | i will do that. for now i've deleted the includes |
| 00:20.48 | p3nguin | Okay, dialplan reload and try again. |
| 00:21.08 | p3nguin | Show me any new problem that occurs. |
| 00:22.36 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:23.14 | ryduh | K so here we are: http://pastebin.com/d68ab38d2 |
| 00:23.23 | ryduh | let me repost sip.conf and the dialplan |
| 00:23.25 | p3nguin | And I'm concerned about your context naming scheme, too. You've used an inbound sip context directing to a dialplan context called voipms-outbound. I would advise you to change that asap to avoid confusion later. |
| 00:25.02 | ryduh | I think that is the problem. I don't want incoming calls going to [voipms-outbound]. They should go to [new-order] |
| 00:25.48 | p3nguin | Change the sip context for that peer, then. |
| 00:26.48 | ryduh | http://pastebin.com/d41d947a |
| 00:27.04 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 00:27.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:27.40 | ryduh | I thought I did. |
| 00:27.42 | p3nguin | I'm still worried about that inbound context going to the outbound dialplan. |
| 00:28.04 | ryduh | That's what I'm trying to fix |
| 00:28.39 | ryduh | when I call, it runs the voipms-outbound and then tries to call the extension, which is the number I'm calling in the first place |
| 00:29.09 | p3nguin | Change the context for that peer, then. |
| 00:29.14 | p3nguin | What's the hold-up? |
| 00:29.48 | ryduh | I have two 'friends' connecting to the same server. asterisk chooses the wrong peer when I get an incoming call |
| 00:30.32 | p3nguin | VoIP.ms's DIDs are routed to accounts in the web portal. Perhaps you need to reconfigure one of them. |
| 00:30.58 | *** join/#asterisk wonderworld (n=w@62.143.22.226) |
| 00:31.13 | p3nguin | How many DIDs do you have? |
| 00:31.15 | ryduh | 1 |
| 00:31.42 | ryduh | i have my DID routed to my sub account: 106629_slo |
| 00:31.42 | p3nguin | Then why do you need to have two peers for voip.ms? |
| 00:32.08 | ryduh | I tried to have outbound calls go through a different subaccount |
| 00:32.34 | p3nguin | You'll want to do that in the dialplan, then. It's not done through the sip context for the ITSP's peer. |
| 00:33.13 | p3nguin | It's absolutely possible to achieve that, but you have attempted it incorrectly. |
| 00:33.32 | p3nguin | Let's concentrate on getting a call inbound first. |
| 00:34.06 | p3nguin | You want calls to come in on acct1? |
| 00:34.16 | ryduh | yeah |
| 00:34.28 | p3nguin | 106629_slo, rather. Didn't notice the name change. |
| 00:35.19 | p3nguin | If you have your DID routed to that account name, it should hit that sip context when it comes in. That sip context says to send the call into context=new-order of extensions.conf. |
| 00:35.43 | ryduh | it doesn't though. it runs [voipms-outbound] because it matches the voipms peer |
| 00:35.53 | p3nguin | [new-order] says to include => order-slo, [order-slo] says to call yourself. :/ |
| 00:37.01 | ryduh | I changed the numbers in the Dial lines in [order-slo]. It actually dials 3 cell phone numbers that are different that my DID |
| 00:37.04 | ryduh | than* |
| 00:37.39 | p3nguin | one moment |
| 00:39.33 | p3nguin | Make sure you reload everything after your changes. |
| 00:40.27 | ryduh | I just did and then I called my DID after that. this is what I got: -- Executing [8057411111@voipms-outbound:1] Dial("SIP/voipms-00dab260", "SIP/18057411111@voipms") in new stack |
| 00:40.51 | p3nguin | Better check the DID routing. Looks like it's wrong. |
| 00:41.46 | ryduh | http://i36.tinypic.com/2uj08ys.png |
| 00:42.04 | ryduh | I've been changing the number to end with 111 to be more anon but wtf now lol. |
| 00:42.59 | p3nguin | Just remember, with Asterisk, you can blacklist numbers that call you. :) |
| 00:42.59 | p3nguin | I wouldn't worry. |
| 00:43.33 | jblack | until they block callerid |
| 00:44.12 | jblack | I'm not trying to undermine you. Sorry. |
| 00:44.17 | p3nguin | Then you reconfigure to not accept anonymous calls. |
| 00:44.23 | ryduh | Here's my sip debug log: http://pastebin.com/dfd91cfe you can see on line 47 it chooses voipms |
| 00:44.44 | p3nguin | jblack: You're being realistic. That's good for a discussion of this type. |
| 00:45.22 | jblack | maybe there's room in this world for a no-callerid-then-do-math-first agi. :) |
| 00:45.42 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:c4f7:1f6d:1dbb:bf1a) |
| 00:47.26 | p3nguin | ryduh: Try changing your voipms sip context to type=user for a minute. |
| 00:48.01 | ryduh | yessssssssss |
| 00:48.08 | p3nguin | and the other context to type=peer |
| 00:48.43 | p3nguin | Save the file, then "sip reload" |
| 00:49.21 | drmessano^ | I just bought a McDonalds hamburger |
| 00:49.25 | drmessano^ | and bit into.. |
| 00:49.27 | drmessano^ | Meat |
| 00:49.29 | drmessano^ | Ugh |
| 00:49.32 | p3nguin | hahaha |
| 00:49.41 | p3nguin | You expected something else? |
| 00:49.46 | jblack | That sounds worthy of a tweet |
| 00:49.48 | drmessano^ | I thought they stopped using meat in their burgers years ago |
| 00:49.54 | p3nguin | oh man |
| 00:49.58 | p3nguin | That's crazy. |
| 00:50.17 | wonderworld | i just had 2 cheeseburgers |
| 00:50.31 | jblack | I had a head of lettuce and two packets of salad dressing. |
| 00:50.31 | drmessano^ | I cant eat cheeseburgers |
| 00:50.35 | wonderworld | mc donalds meat has a plastic flavour |
| 00:50.47 | ChannelZ | thats so you know it's fresh |
| 00:50.55 | ryduh | the DID goes to the right context now [new-order], but can't dial out: [Oct 21 00:50:28] WARNING[30629]: chan_sip.c:3005 create_addr: No such host: voipms |
| 00:51.21 | wonderworld | i don't know if mc donalds changed or if my taste changed. i can't stand it any more. |
| 00:51.58 | ChannelZ | probably both |
| 00:52.14 | ChannelZ | Their fries are still good though |
| 00:52.21 | jblack | nah, they haven't changed. |
| 00:52.34 | jblack | You're probably just become over-sensitive to fat. |
| 00:52.43 | p3nguin | ryduh: That makes sense, because type=user is for people with phones who use your asterisk for calls. |
| 00:52.43 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 00:53.15 | p3nguin | ryduh: Set them both as type=peer and see if the misbehavior comes back. |
| 00:53.32 | ryduh | It does |
| 00:53.40 | jblack | anyone want to do some prompts for me? |
| 00:53.58 | wonderworld | gay line? |
| 00:54.04 | Nivex | \u@\h:\w\$ |
| 00:54.08 | Nivex | there's a prompt for you :) |
| 00:54.09 | jblack | nah. home. |
| 00:54.30 | p3nguin | jblack: If it doesn't take a long time, and is not for a gay chatline, I might consider it. |
| 00:54.50 | p3nguin | jblack: Just give me the script. |
| 00:54.56 | jblack | maybe half a dozen short scripts. |
| 00:55.12 | jblack | Thank you for calling, press 1 for .., 2 for .. etc. |
| 00:55.33 | p3nguin | I'll do it if you want. |
| 00:55.39 | jblack | That would rock. My voice sucks. |
| 00:55.43 | wonderworld | i asume you wouldn't like my charming accent. |
| 00:55.49 | p3nguin | jblack: I feel the same about my own. |
| 00:56.01 | jblack | and it seems funny doing it for my own system, ya know? |
| 00:56.06 | p3nguin | yep |
| 00:56.10 | jblack | that whole first/third person thing |
| 00:56.11 | p3nguin | I had the wife do mine for me. |
| 00:56.24 | jblack | it's kinda like scratching your own back, right? |
| 00:56.32 | p3nguin | More like shaving your own balls. |
| 00:56.41 | jblack | you owe me a new monitor |
| 00:56.44 | p3nguin | lol |
| 00:56.46 | ryduh | So we've confirmed it's the user/peer/friend thing. What's the best SIP type for outbound calls? |
| 00:57.06 | p3nguin | ryduh: peer (or sometimes friend, depending on who you ask) |
| 00:57.10 | p3nguin | ryduh: user is for your phones. |
| 00:57.42 | jblack | I still haven't worked out the logic I want yet. |
| 00:57.46 | ryduh | when I use peer or friend, I get the same problem. asterisk assigns incoming calls to the wrong context |
| 00:58.04 | jblack | every time I start working out what I want, I realize that people would probably be annoyed. |
| 00:58.46 | ryduh | is there a better way to name the sip contexts so they are chosen correctly? |
| 00:59.12 | ryduh | i have voipms-slo and voipms. I register two SIP accounts with voipms on the same server |
| 01:01.02 | wonderworld | ryduh: just set context=context_the_calls_should_go_to for every incoming sip peer |
| 01:02.58 | p3nguin | wonderworld: The problem is that he has two sub accounts on the same box and the ITSP is, for some unknown reason, sending calls into the wrong context/subaccount. |
| 01:03.45 | jblack | p3nguin: http://pastebin.com/m97755ba |
| 01:03.47 | p3nguin | I assume it is because of the whole 'IP address being the same' thing. |
| 01:03.48 | jblack | What do you think |
| 01:03.58 | ryduh | Yeah I think that is the problem |
| 01:04.41 | p3nguin | jblack: Give me a few minutes and I'll record them. Hope .gsm will be sufficient. |
| 01:04.48 | jblack | gsm would be ideal |
| 01:05.01 | jblack | Think they look good? |
| 01:05.15 | p3nguin | jblack: It's safe to assume you want your last name and your first name in the respective places, right? |
| 01:05.23 | jblack | yup. Just didn't want to blurt it on pb |
| 01:05.29 | ryduh | bleh. ill be back tomorrow to see if I can fix it. Thanks p3nguin for helpin out |
| 01:05.31 | p3nguin | I figured |
| 01:05.38 | jblack | what do you tink of the script? Any thoughts? |
| 01:06.21 | p3nguin | jblack: Seems like it could work. |
| 01:07.12 | *** join/#asterisk smash- (n=mssah@c-24-21-182-11.hsd1.or.comcast.net) |
| 01:09.36 | drmessano^ | I just tried to download Debian, and I got a tarball full of 1s and 0s.. Wonder what they expect me to do here |
| 01:09.54 | jblack | count them |
| 01:10.40 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-212-135.dsl.lsan03.sbcglobal.net) |
| 01:11.12 | *** join/#asterisk toddejohnson (n=toddejoh@70.226.215.44) |
| 01:11.58 | drmessano^ | I tried to |
| 01:12.20 | jblack | THen multiply all the individual bits together. |
| 01:12.23 | drmessano^ | But I can only count in Octal |
| 01:12.45 | drmessano^ | OCT31 = DEC25 |
| 01:12.48 | drmessano^ | ^^ WIN |
| 01:13.07 | jblack | E_NOGETIT |
| 01:13.42 | drmessano^ | 31 in Octal is 25 in Decimal |
| 01:14.05 | b14ck | sup guys |
| 01:14.30 | jblack | Ahh, of course |
| 01:14.40 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/x-aqbkvvmhwvrrxskp) |
| 01:15.03 | drmessano^ | So, HAPPY CHRISTOWEEN |
| 01:15.21 | yziquel | say i make a sip call to a regular phone. how can i control the number that will show up on the regular phone? |
| 01:15.40 | yziquel | i mean the calling phone number... |
| 01:15.45 | jblack | i make a sip call to a regular phone. how can i control the number that will show up on the regular phone? |
| 01:16.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 01:17.26 | jblack | I thought was funning. Look at Set(Callerid(num)=XXXX)) |
| 01:18.15 | yziquel | jblack: thanks a lot. |
| 01:18.57 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 01:18.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:21.39 | drmessano^ | HA |
| 01:21.44 | drmessano^ | GNU.org is hilarious |
| 01:21.52 | drmessano^ | Ubuntu provides specific repositories of nonfree software. Even if you don't use them, the default application installer will advertise nonfree software to you. |
| 01:22.02 | drmessano^ | THOSE BASTARDS!!! |
| 01:22.05 | Naikrovek | yeah |
| 01:22.09 | jblack | Did they finally figure that out? |
| 01:22.26 | jblack | remembers when gnu went from debian to gnu over the non-free repo |
| 01:22.32 | jblack | pardon, from debian to ubuntu |
| 01:22.37 | jblack | which shocked me, for just that reason. |
| 01:22.38 | drmessano^ | Debian has repeatedly made tacit or explicit exceptions for specific pieces of nonfree software, such as the blobs included in or accompanying Linux. We're still hopeful that there won't be such exceptions in the future, but we can't turn a blind eye to the situation as it stands today. |
| 01:22.54 | drmessano^ | Panties in a wad? |
| 01:23.14 | jblack | You know richard. |
| 01:23.16 | drmessano^ | WE CANT TURN A BLIND EYE, NO SIR |
| 01:23.16 | Naikrovek | no kidding. they are True Believers Despite Reality |
| 01:23.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:23.46 | Naikrovek | he's too polarized to be correct. |
| 01:24.12 | jblack | It has less to do with facts on the ground, and more to do with satisfying grudges. |
| 01:24.18 | Naikrovek | yes |
| 01:24.25 | drmessano^ | DrMessanoNIX - Uses the words NON and FREE 5 paragraphs apart on the welcome page for the wiki documenting the distro. This is entirely unacceptable. |
| 01:24.29 | jblack | I became less enamored of richard when he lied to my face. |
| 01:24.39 | Naikrovek | what did he say |
| 01:24.53 | jblack | some silly trivial thing of no import. |
| 01:25.02 | Naikrovek | huh |
| 01:25.24 | jblack | a small lie, and of no consequence, that didn't even need to be told. |
| 01:25.27 | drmessano^ | JBlackNIX - Was once give a line of Non-free code, but didn't include it. These sort of shenanigans make us sick, and will not be tolerated. |
| 01:26.03 | jblack | drmessano^: Huh? |
| 01:26.20 | drmessano^ | http://www.gnu.org/philosophy/common-distros.html |
| 01:26.25 | b14ck | richard stallman is an awesome programmer, but a really extreme politician :) |
| 01:26.39 | jblack | Dude. He wrote emacs. There's no atoning for that. |
| 01:27.07 | b14ck | How many people do you know who have written software which has been around that long? Not many =/ |
| 01:27.09 | jblack | There is a special layer in hell, made of ctrl-alt-function keys, for emacs developers. |
| 01:27.27 | jblack | OH, sure, credit where credit is due. We wouldn't be here, for sure. |
| 01:27.41 | *** join/#asterisk adam0563 (n=damagele@c-98-202-116-119.hsd1.ut.comcast.net) |
| 01:27.48 | b14ck | I'm not saying I like emacs, I'm just saying he's a fine coder =p |
| 01:27.52 | b14ck | I'm a vim guy, myself ^^ |
| 01:27.54 | jblack | however, the oldest trees are usually the ones with the most rot. |
| 01:28.12 | jblack | b14ck: Have you actually looked at the code for bash, glibc, gcc, etc ? |
| 01:28.26 | b14ck | Yea, I've done significant systems-level programming. |
| 01:28.38 | jblack | You think that code's fine? |
| 01:28.43 | b14ck | I re-implemented a lot of glibc at my uni for a project. |
| 01:29.00 | drmessano^ | BleuHat Linux - BleuHat Linux is a distro consisting of 72 bytes of code that when burned as an ISO and booted, print the word "Poop" in the middle of the screen. The source was once burned to a CD that rode in a CD case, but not touching or even adjacent to a Windows CD. We cannot tolerate fraternizing with the enemy, so therefore, we cannot endorse this distro. |
| 01:29.00 | b14ck | I think it is well written, and adheres to all POSIX standards. It is a bit bloated now but it has to be. |
| 01:29.02 | jblack | so you're aware of the tangled mess it is. |
| 01:29.05 | b14ck | I don't see how it could be avoidable. |
| 01:30.18 | jblack | A lot of the problems can't be avoided. Aye. but most of those old codebases are as tangled as brambles. They need to institute strong test suites, and refactor things back down to a more spartan state. |
| 01:30.45 | b14ck | Feel free to fix it =p |
| 01:30.50 | jblack | hell. maybe join the 21st century and migrate to c++ (heh, if you're gonna dream....) |
| 01:30.53 | drmessano^ | I prefer nano.. its as close to DOS notepad as this Vista loving, gun toting republican can get using Linux |
| 01:30.58 | b14ck | c++ sucks |
| 01:30.59 | b14ck | =/ |
| 01:31.03 | jblack | better men than I have gone into the brambles never to come back out. |
| 01:31.16 | jblack | That's one of gnu's failing. They only C one way. |
| 01:31.16 | drmessano^ | J# rocks |
| 01:31.38 | b14ck | btw, any python programmers in here? |
| 01:31.56 | drmessano^ | Im a board certified python handler |
| 01:31.56 | b14ck | I wrote a callfile library in python for asterisk, was hoping someone could take a look and make suggestions. |
| 01:32.10 | jblack | They're locked into function oriented design becaue that's what existed in the 80s. |
| 01:32.20 | jblack | Do you have a page for it yet? |
| 01:32.23 | drmessano^ | 10 print "poop" |
| 01:32.27 | b14ck | Yea, sec. |
| 01:32.30 | drmessano^ | 20 goto 10 |
| 01:32.32 | drmessano^ | run |
| 01:32.54 | b14ck | http://github.com/comradeb14ck/pycall |
| 01:33.05 | b14ck | If anyone wants to take a look at that--I'd love suggestions / etc. |
| 01:33.31 | b14ck | I packaged it into a python module so it should be easy to install on any unix-based os |
| 01:33.32 | DavidR2008 | I have question, if you're the last person exit a conference, does the conference exit sound play? and if no one's there to hear it, does it still make sound? |
| 01:33.42 | b14ck | =p |
| 01:34.28 | jblack | a callfile library. There's enough meat there for a library? |
| 01:36.12 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
| 01:36.12 | jblack | heh. callfile = CallFile(trunk_type = 'DAHDI',trunk_name = 'g0',number = '18002223333',application = 'Playback',data = 'hello-world') |
| 01:36.17 | p3nguin | jblack: Festival is being an ass. It's putting too much pause on the front of the recording, so it's taking me some extra time to try to get them to sound the way I want. |
| 01:36.31 | b14ck | I've ended up using callfiles so many times in my apps, I thought it would be nice to make an easy way to use them / schedule them /etc. |
| 01:37.17 | Deeewayne | mog, ping |
| 01:37.27 | jblack | p3nguin: Oh, you're not recording with your voice? |
| 01:37.48 | p3nguin | jblack: Yes, but festival provides me with the prompts. |
| 01:38.01 | jblack | gets confused |
| 01:38.09 | b14ck | jblack, check out the schedule-call.py demo =p |
| 01:38.12 | mog | Deeewayne, pong |
| 01:38.14 | p3nguin | jblack: So it beeps and I start talking, but there is extra pause during the playback. |
| 01:38.42 | p3nguin | jblack: Just trying to get it to sound like I want it to sound. That's why I'm not done yet. |
| 01:39.04 | jblack | I figured you were just using audacity and a microphone. |
| 01:39.29 | jblack | and no worries. I've been putting this off for days. my old system is still running fine |
| 01:40.09 | jblack | b14ck: I didn't mean to belittle you. I'm sure this could be of great use for a tool that needs to generate a lot of call files. |
| 01:40.19 | jblack | And manage ones in flight. |
| 01:40.41 | b14ck | jblack, oh no worries, ya. That's the goal I had it going for (I use them for a lot of stress testing and timing stuff myself) |
| 01:48.24 | jblack | This trying to not be an asshole stuff stucks |
| 01:50.46 | jblack | oh brilliant. "The supreme court agreed tuesday to decide whether guantanamo detainees who are considered no threat can be ordered released in the united states - over the objectiosn of the obama administration and congress - if the prisoners have nowhere else to go" |
| 01:51.23 | *** join/#asterisk chendy (n=chatzill@113.91.39.43) |
| 01:53.21 | wonderworld | well, some of them have been in guantanamo for well over 6 years by now without a trial. just fair to give them a place to live after stealing them years of lifetime. |
| 01:54.40 | jblack | I'd rather off 'em each a million dollars and first class tickets to costa rica |
| 01:54.46 | jblack | offer 'em |
| 01:54.48 | drmessano^ | Of course they have a place to go |
| 01:54.56 | drmessano^ | They're ON a PLACE |
| 01:55.02 | drmessano^ | Its called CUBA |
| 01:55.15 | drmessano^ | Send them out the BACK DOOR |
| 01:55.15 | jblack | actually, they're not in cuba. |
| 01:55.34 | jblack | They're on US soil atm. surrounded by cuba. |
| 01:56.19 | wonderworld | jblack: i think they would prefer your idea |
| 01:56.50 | drmessano^ | [21:49] <jblack> This trying to not be an asshole stuff stucks <--- Sorry, must have missed that |
| 01:57.18 | jblack | didn't say I was successful. |
| 01:57.26 | p3nguin | jblack: Check /notice |
| 01:57.45 | jblack | yeah, it shows in my primary window |
| 01:57.52 | jblack | thanks. |
| 01:57.57 | p3nguin | jblack: If they aren't good enough, I can try again another day on a better system. |
| 01:58.16 | jblack | what plays gsm.... |
| 01:58.20 | p3nguin | asterisk |
| 01:58.23 | drmessano^ | I think my wife stole the sync cable for my MP3 player, of which we both have the same unit |
| 01:58.24 | p3nguin | :) |
| 01:58.26 | jblack | other than 8 |
| 01:58.29 | drmessano^ | How do I correct this? |
| 01:58.31 | jblack | otehr than *, I mean. |
| 01:58.32 | drmessano^ | Do i beat her? |
| 01:58.36 | drmessano^ | Starve her? |
| 01:58.47 | p3nguin | Delete her mp3s. |
| 01:59.04 | jblack | You steal her tampons. |
| 01:59.11 | wonderworld | nah, hide her cell |
| 01:59.14 | jblack | and put cotton balls in their place. |
| 01:59.15 | wonderworld | she will go insane |
| 01:59.58 | drmessano^ | Convert them to ogg, copy them over in Storage mode, and throw a piece of Penguin adorned album art on there for an album called "haxored 2: Danny Boogaloo" |
| 02:00.01 | p3nguin | Make sure it's turned off, otherwise she'll call it trying to locate it. |
| 02:00.30 | drmessano^ | "the fuck is an ogg?" |
| 02:00.37 | drmessano^ | "the fuck is a sync cable?" |
| 02:00.40 | drmessano^ | pwn3d |
| 02:00.46 | p3nguin | Do you mean "what the fuck...?" |
| 02:00.48 | jblack | there we go. "play filename.gsm" |
| 02:00.55 | drmessano^ | No, i do not |
| 02:01.21 | jblack | awesome. I'm gonna have a penguin answer my phone. |
| 02:01.25 | jblack | =) |
| 02:01.27 | p3nguin | haha cool |
| 02:01.43 | p3nguin | I was afraid the delay would ruin them. |
| 02:01.52 | jblack | I don't hear a delay here. |
| 02:02.05 | jblack | in fact, I may end up having to add a gsm shim |
| 02:02.29 | jblack | I think they sound great. |
| 02:02.38 | p3nguin | I try to leave no pause before and none after. |
| 02:02.50 | wonderworld | i like espeak's robotic 80s style voice for prompts. |
| 02:02.53 | jblack | yeah. it's easier to add a shim than to try and take out a pause. |
| 02:02.54 | p3nguin | Works better when adding several files together in sequence that way. |
| 02:03.26 | jblack | yup, and you can always inject another 100ms file in between if the pause is needed. |
| 02:03.37 | wonderworld | sounds like "Joshua" in WarGames |
| 02:03.50 | p3nguin | I'm going to have to create a dialplan so that I can hear your recordings. |
| 02:04.05 | jblack | I need to find a copy of the stephen hawking voice beating off, so tat I can add it to my plan. |
| 02:04.23 | jblack | p3nguin: do you have sox? |
| 02:04.30 | p3nguin | yeah |
| 02:04.31 | jblack | if so, try "play *.gsm" |
| 02:05.20 | p3nguin | oh |
| 02:05.32 | p3nguin | That might work. I tried mplayer and aplay before with no luck. |
| 02:05.34 | drmessano^ | uses shuz |
| 02:05.46 | jblack | yeah, see? the pauses you were worried about don't seem to be a problem at all |
| 02:08.02 | p3nguin | play seems to slow it down just a hair. |
| 02:08.23 | jblack | sure. process calls, etc. |
| 02:08.40 | jblack | perfectly usable |
| 02:11.57 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 02:12.52 | p3nguin | They aren't terrible, I suppose. |
| 02:13.50 | jblack | I want badlands |
| 02:14.46 | p3nguin | Isn't that a place in something Dakota? |
| 02:14.55 | jblack | it's also a computer game |
| 02:15.13 | p3nguin | oh |
| 02:16.01 | p3nguin | ?o?th Dakota |
| 02:25.33 | jblack | gives drmessano^ an odd look |
| 02:26.04 | drmessano^ | ? |
| 02:26.23 | jblack | you tweeted about your cheeseburger? |
| 02:26.38 | drmessano^ | hamburger, yes |
| 02:26.58 | jblack | I love this world. |
| 02:27.39 | jblack | just imagine all the technology, and all the steps and effort, for me to find out that your hamburger had meet on it. via _two_ routes. |
| 02:27.49 | jblack | meat, that is. |
| 02:28.18 | drmessano^ | It was also posted on facebook, thanks to the twitter facebook app |
| 02:28.20 | drmessano^ | and friendfeed |
| 02:28.22 | jblack | Hell. probably satellites got involved, fibre optic cables, mainframes.... all so that in a heartbeat, I could be informed of that. |
| 02:28.44 | drmessano^ | So it was sent to blackberries, tweetdecks, via SMS, and who knows what |
| 02:28.58 | jblack | Used to be that the death of a president took 2 months to spread across the country. |
| 02:29.29 | drmessano^ | Hey, you think thats bad |
| 02:29.48 | drmessano^ | Some people _still_ dont know JFK was killed by UFOs |
| 02:29.49 | jblack | it's not about the burger. :) |
| 02:30.03 | jblack | Good point. Some people still don't know stuff. :) |
| 02:30.45 | ChannelZ | wonders if he'll get a ship notice for Windoze 7 from amazon tonight |
| 02:30.58 | jblack | I guess I just got smacked in the face with how trivial we are with this instant communication shit. |
| 02:31.01 | drmessano^ | Google still doesnt have a twitter clone |
| 02:31.11 | jblack | Not really, no. |
| 02:31.34 | jblack | now that you mention it, why haven't they bought out twitter yet? |
| 02:31.49 | jblack | the social network must be valuable to them. |
| 02:31.59 | drmessano^ | because they haven't figured out how to do it in a way that would fucking piss mchou off |
| 02:32.07 | drmessano^ | but they're working on it |
| 02:32.15 | drmessano^ | Google Labs (TM) |
| 02:32.18 | mchou | say wha? |
| 02:32.25 | drmessano^ | Oh hai |
| 02:32.57 | mchou | drmessano^:: what would piss me off? |
| 02:33.14 | hardwire | erm |
| 02:33.14 | hardwire | so I'm adding SIP headers.. custom X-Header stuff. |
| 02:33.14 | hardwire | and they don't get included in "multi-invite" sessions. |
| 02:33.18 | drmessano^ | Please dont make me explain it.. its not that difficult to put the two lines together |
| 02:33.40 | hardwire | like if a sip phone calls in, then asterisk dials to another sip phone.. first invite from asterisk with header.. second invite from asterisk on behalf of other phone - sans header. |
| 02:34.07 | hardwire | is that the way it should be operating, I understand there could be some concerns about security and race condition management. |
| 02:35.56 | hardwire | ok.. I see now that it adds the header to only the first INVITE |
| 02:36.12 | hardwire | however SipGetHeader gets values from the last INVITE |
| 02:37.12 | hardwire | https://issues.asterisk.org/view.php?id=9516 |
| 02:37.18 | hardwire | goes and finds another tree to bark up |
| 02:37.31 | mchou | drmessano^: so you look into voip.ms? |
| 02:38.30 | hardwire | fixed it |
| 02:38.31 | drmessano^ | Yeah, I like it |
| 02:39.01 | mchou | drmessano^: I tried doing a whois on them. FAIL |
| 02:39.32 | drmessano^ | So? |
| 02:39.39 | mchou | drmessano^: what's the deal with .ms domains? |
| 02:40.10 | mchou | Old sswiss banks by comparison are positively indiscreet :) |
| 02:40.18 | mchou | Swiss* |
| 02:40.18 | drmessano^ | http://en.wikipedia.org/wiki/.ms |
| 02:40.59 | mchou | drmessano^: no no, I mean whois on *.ms |
| 02:41.26 | hardwire | yay for passing asterisk account code correctly through sip headers |
| 02:43.13 | b14ck | i've got a voip.ms account |
| 02:43.16 | b14ck | but they're expensive =/ |
| 02:43.20 | Katty | SO. |
| 02:43.26 | b14ck | much cheaper to just use flowroute, and they've got the same options / nice site / etc |
| 02:43.28 | Katty | i'd like to report that Zombieland....is quite the gigglefest. |
| 02:43.36 | Katty | it comes highly recommended. |
| 02:43.39 | mchou | b14ck: you found somebody better for cheaper? |
| 02:43.43 | b14ck | mchou, yep |
| 02:43.51 | b14ck | mchou, flowroute.com <-- best voip provider imo |
| 02:44.10 | mchou | b14ck: drmessano^ is dissatified :) |
| 02:44.21 | b14ck | of what? |
| 02:44.29 | b14ck | flowroute? |
| 02:44.34 | mchou | yeah |
| 02:44.38 | b14ck | really? |
| 02:44.40 | b14ck | Why? oO |
| 02:44.42 | mchou | for lack of local DIDs |
| 02:44.57 | b14ck | well that's not really a big issue... |
| 02:45.34 | b14ck | flowroute is pretty slick i think, they've got a virtual pri which saves you a ton of money |
| 02:45.38 | drmessano^ | I never said I was dissatisfied |
| 02:46.06 | Katty | well by golly i'm dissastisfied |
| 02:46.13 | drmessano^ | It just doesnt resolve my issue of needing to port friends local numbers when Flowroute hasn't expanded in some time, and doesnt include my area |
| 02:46.26 | Katty | i've not gotten a hug in at least.... 15 minutes. |
| 02:46.29 | Katty | and that is unacceptabuhls. |
| 02:46.31 | mchou | drmessano^: I cant get no satisfaction.... |
| 02:46.34 | Katty | hugs drmessano^ |
| 02:46.42 | drmessano^ | hugs Katty |
| 02:46.52 | Katty | k, all is well in the universe. |
| 02:47.10 | Katty | Dear Universe, thank you for Hugs. Love, Katty |
| 02:47.22 | drmessano^ | Hershey Hugs? Absolutely |
| 02:47.41 | Katty | drmessano^: when you find something that makes you happy, you should always thank the universe. |
| 02:47.51 | Katty | drmessano^: it's just polite. |
| 02:48.07 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:48.34 | drmessano^ | Dear Universe, thank you for making boxer briefs from Fruit of the Loom in at least one color that doesnt leave lint in my buttcrack for excavation in the shower the following day. Love, drme |
| 02:48.40 | drmessano^ | tab fail :( |
| 02:48.56 | russellb | drmessano^: w. t. f. |
| 02:48.58 | Katty | it's okay. the universe understands. |
| 02:49.03 | mchou | drmessano^: TMI |
| 02:49.04 | Katty | hugs russellb |
| 02:49.28 | drmessano^ | Dont buy any of the dark blue varieties, really |
| 02:49.30 | russellb | i kickbaned someone last night for something similar to that :-p |
| 02:49.30 | mchou | lol |
| 02:49.39 | drmessano^ | If you buy a multipack, dont buy Blue |
| 02:49.57 | drmessano^ | If you MUST get a multipack with an obligatory pair in blue, toss them |
| 02:50.03 | drmessano^ | Its not worth it, men |
| 02:50.20 | mchou | wonders how drmessano^ found out about this "issue" |
| 02:50.23 | drmessano^ | Ive written to fruit of the loom about this |
| 02:50.47 | drmessano^ | Its odd.. I have them in 5 different colors.. but the blue ones are like shearing a sheep over the course of a day |
| 02:50.49 | drmessano^ | :( |
| 02:50.55 | russellb | ANYWAY |
| 02:51.19 | drmessano^ | It interferes with my ability to write a good dialplan |
| 02:51.24 | drmessano^ | ^^^ the tie-in |
| 02:52.39 | russellb | I saw probably the worst dialplan I have ever seen today. |
| 02:53.00 | Katty | Dear Universe, thank you for documentation and organized people. |
| 02:53.52 | drmessano^ | 10 Print "SIP/googlevoice@skype.iphone.com" |
| 02:53.55 | drmessano^ | 20 Goto 10 |
| 02:53.59 | drmessano^ | ^^^ Worse than that? |
| 02:54.06 | russellb | much |
| 02:54.19 | Katty | infinate loop is pretty bad :< |
| 02:54.34 | [TK]D-Fender | Katty: Not an infinite loop.... |
| 02:54.43 | Katty | [TK]D-Fender: mister fender. |
| 02:54.46 | Katty | [TK]D-Fender: wherever is my hug? |
| 02:55.06 | [TK]D-Fender | Katty: Skype crashes after 10 iterations due to segfault ;) |
| 02:55.12 | drmessano^ | lol |
| 02:55.15 | [TK]D-Fender | hugz on teh Katty |
| 02:55.19 | Katty | well that explains it! |
| 02:55.23 | Katty | i'd wondered what had happened! |
| 02:55.24 | russellb | skype's fault |
| 02:55.26 | Katty | hugs [TK]D-Fender |
| 02:55.50 | [TK]D-Fender | russellb: He who laughs in the face of adversity .... has a good scape-goat :p |
| 02:56.02 | Katty | speaking of goat. |
| 02:56.10 | Katty | i've read that goat's milk is more healthy for you than cows milk.' |
| 02:56.23 | Katty | tho i kinda doubt the validity of drinking any milk. |
| 02:56.31 | mchou | umm, unless it's got scrapie |
| 02:56.34 | drmessano^ | Goat's milk is low in lactose |
| 02:56.46 | drmessano^ | I can drink gallons of it |
| 02:56.52 | Katty | i suppose that's good for the mostly lactose intollerant |
| 02:56.53 | drmessano^ | Cows milk, not so much |
| 02:56.55 | [TK]D-Fender | Katty: I've heard that neurosing about every litte thing you eat causes brain cancer. |
| 02:56.58 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:57.10 | mchou | [TK]D-Fender: lol |
| 02:57.17 | drmessano^ | One cup of cows milk and you can just call in sick for me, for the next 2 days |
| 02:57.18 | Katty | [TK]D-Fender: that's actually incorrect. |
| 02:57.29 | Katty | [TK]D-Fender: stress triggers a response in the body. |
| 02:57.34 | Katty | [TK]D-Fender: but a healthy liver can combat it. |
| 02:57.48 | Katty | [TK]D-Fender: so, i'm guessing something is wrong with my immune system that makes me spaz out over anxiety ^_- |
| 02:57.56 | Katty | [TK]D-Fender: possibly i just need to chill. |
| 02:58.03 | [TK]D-Fender | Katty: I can see your symptoms have already manifested in a cascading failure of your logical reasoning center..... |
| 02:58.05 | Katty | [TK]D-Fender: but topics of health are very interesting regardless. |
| 02:58.21 | Katty | [TK]D-Fender: well that's okay. we all have our thing. |
| 02:58.30 | p3nguin | neurotic liver failure |
| 02:58.37 | Katty | *hee* |
| 02:58.44 | mchou | [TK]D-Fender: you get your med dregree from drmessano^? :) |
| 02:58.52 | mchou | degree* |
| 02:59.16 | [TK]D-Fender | mchou: rI'm not a doctor..... but I do like to play ;) |
| 02:59.50 | Katty | he also likes to pretend he's my therapist. |
| 03:00.12 | Katty | i hear taking a baseball bat to something is good therapy. |
| 03:00.13 | Katty | hmm. |
| 03:00.23 | Pan3D | :/ |
| 03:00.32 | [TK]D-Fender | Katty: YES! |
| 03:00.41 | Katty | [TK]D-Fender: okay. i'll go get the bat. |
| 03:04.07 | mchou | b14ck: how many DIDs you have with flowroute? |
| 03:04.34 | b14ck | I've got 2 (I use 1 for my home / forwarding number and 1 for dev work). |
| 03:04.45 | b14ck | But I have a lot of clients who I do custom coding for, and they have tons. |
| 03:05.02 | b14ck | I think one larger client spends about ~3,500$ / month on flowroute. |
| 03:05.18 | mchou | b14ck: if you only have 2, VPRI doesnt sound like it's such a great deal |
| 03:05.31 | b14ck | VPRI isn't about DIDs, its about simultaneous calls. |
| 03:05.42 | b14ck | You can have a single VPRI, but 2000000 dids if you want. |
| 03:06.03 | hardwire | anybody know of a good method to store sip traces for all sip calls? |
| 03:06.06 | mchou | b14ck: right, which is what prompted the question in the first place |
| 03:06.07 | b14ck | VPRIs give you UNLIMITED call minutes per month (but only 1 channel at a time) for 17.95$/month |
| 03:06.17 | hardwire | I'm going to send packets to ulogd.. but there has to be a better way |
| 03:07.49 | [TK]D-Fender | brb |
| 03:08.32 | Katty | but...but... bat? :< |
| 03:08.41 | drmessano^ | VPRIs are nice |
| 03:09.09 | hardwire | who? |
| 03:12.39 | mchou | b14ck: I want to make sure I understand this. 1 channel=no mor than one simultaneous call, correct? |
| 03:12.43 | mchou | more* |
| 03:12.54 | b14ck | correct |
| 03:13.04 | b14ck | each *channel* in a vpri costs 17.95$/month |
| 03:13.18 | b14ck | So if you want to allow 2 simultaneous calls, you'd be paying (2*17.95)$/month |
| 03:13.30 | b14ck | But that would be _all_ you're paying. You would not pay for any minute charges. |
| 03:13.41 | b14ck | So businesses (or anything with high-usage) saves a TON of money. |
| 03:14.07 | p3nguin | I would have to talk for 17095 minutes just to break even on my pay-per-minute plan. |
| 03:14.10 | russellb | "businesses" is pretty inclusive :-) |
| 03:14.21 | russellb | i suppose it's even not so bad for a home rate if you use it enough ... |
| 03:14.32 | p3nguin | 17095 minutes is a lot. |
| 03:14.38 | russellb | true. |
| 03:14.50 | russellb | I do not talk that much |
| 03:14.54 | russellb | actually, i hate phones |
| 03:15.10 | mchou | if we did we wouldn't be here, that's for sure :) |
| 03:15.16 | russellb | heh. |
| 03:15.31 | b14ck | Well, most of the people I do work for end up using custom IVR type stuff. |
| 03:15.39 | b14ck | So they continuously have calls coming into their system. |
| 03:15.45 | b14ck | So it's a nice way to save $$ every month. |
| 03:15.48 | b14ck | =p |
| 03:15.51 | russellb | true dat |
| 03:15.58 | p3nguin | I was talking about termination. |
| 03:16.39 | p3nguin | For origination, it's like $7.95 for unlimited (which is probably limited to 3500 minutes). |
| 03:19.49 | mchou | I dunno. for mere mortals voip.ms rates might make more sense |
| 03:20.15 | mchou | for motormouths go with flowroute :) |
| 03:21.58 | b14ck | Still doesn't matter. flowroute's per-minute is the same/lower than voip.ms's |
| 03:22.05 | b14ck | (last time I checked, anyways) |
| 03:24.54 | p3nguin | It's right at the same. |
| 03:25.22 | mchou | p3nguin: how much does voip.ms charge for inbound? |
| 03:26.58 | p3nguin | $1.49 per Month / $0.0149 per minute |
| 03:27.07 | p3nguin | $6.95 per Month / $0.00 per minute / 2 Channels |
| 03:27.18 | nix8n82 | I need to tune my rx and tx gain for a tdm410 card I have..I came across a guide that suggested calling a milliwatt line so far I can't find any numbers in florida and the 800 numbers are diconnect, plus I don't have long distance..is there any other tools that may help me set the proper levels? |
| 03:27.39 | *** join/#asterisk [8none1] (n=[8none1]@67.107.93.2) |
| 03:30.16 | p3nguin | mchou: and toll-free DIDs are $0.99 per Month / $0.029 per minute (for US callers) |
| 03:31.10 | mchou | wow |
| 03:31.10 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 03:31.18 | mchou | just looked them up |
| 03:31.29 | mchou | pac west telecom rate center |
| 03:31.57 | mchou | I'm not so sure that's great news |
| 03:32.54 | drmessano^ | If I am the borat of #asterisk, mchou is monk |
| 03:33.05 | mchou | lol |
| 03:33.28 | mchou | no butt crack jokes pls :) |
| 03:34.20 | mchou | quite frankly I didn't get the gay wrestling reference in Borat :) |
| 03:34.25 | drmessano^ | Goes to a circus and complains about the germs from the animals, the unsanitary face makeup, the dangerously oversized shoes, the pants being asymmetrical to the shirts they wear |
| 03:34.53 | drmessano^ | Oh, and 3 rings being a non-even number |
| 03:35.39 | mchou | drmessano^: what can I say, you got me all figured out |
| 03:37.34 | nix8n82 | <PROTECTED> |
| 03:37.41 | drmessano^ | Google is going to buy the company that makes your pants and serve ads on the zippers, just to piss you off |
| 03:38.24 | drmessano^ | :) |
| 03:38.24 | mchou | what's this fixation with google buying junk all night? |
| 03:38.47 | drmessano^ | Its been referenced twice.. Thats far from a fixation |
| 03:39.13 | jblack | I can't wait for gmarriage. |
| 03:39.30 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-qgzncpdtlxpjxtth) |
| 03:39.42 | jblack | it would be a great relationship. Google already knows everything about me. |
| 03:40.54 | mchou | what's really ironic is thet sun is being swallowed by oracle, and google is being run by ex-sun CEO :) |
| 03:41.44 | jblack | seems to me that all the big companies are own/runby the same group of people. |
| 03:42.11 | mchou | plenty of cheap labor for google in a few, coourtesy of sun/oracle |
| 03:49.26 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 03:57.03 | nix8n82 | can I emulate a milliwatt test line over voip? |
| 04:00.01 | b14ck | brb need to reboot |
| 04:04.37 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 04:07.26 | ChannelZ | make an extension, run Milliwatt() on it.. dial.. |
| 04:08.01 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 04:10.57 | drmessano^ | I cant wait for google to come out with the worlds best database platform |
| 04:11.04 | drmessano^ | They will call it google |
| 04:11.40 | nix8n82 | ChannelZ, thanks..which end of the conversation do I have run the milliwatt app..voip or fxo..does it matter? |
| 04:12.20 | ChannelZ | I don't understand the question. Nor what you're really trying to accomplish |
| 04:13.57 | nix8n82 | I'm trying to tune rxgain and txgain in chan_dahdi.conf for my tdm410p card..and hope the hpec software I purchase can eliminate the echo I have when the fxo port is in use, with my internal extension. |
| 04:14.38 | drmessano^ | Oslec works well for that too, and its free |
| 04:14.45 | ChannelZ | ah.. well without a real external milliwatt source from the telco, it's a little hard to get it right |
| 04:14.59 | nix8n82 | if I go strictly voip or just use the two fxs ports I have no problems with the conversation but when I dail out or in with the fxo port I have bad echo on the fxs or voip ends |
| 04:15.47 | nix8n82 | yeah and I googled and I can't find any numbers in florida or 800 numbers that work |
| 04:15.59 | nix8n82 | they don't have long distance on that line |
| 04:16.44 | p3nguin | numbers to what? |
| 04:16.55 | ChannelZ | a milliwatt test number |
| 04:16.57 | nix8n82 | milliwatt test line |
| 04:17.15 | ChannelZ | I don't know why the telcos are so fucking secret about it |
| 04:17.28 | p3nguin | It's just a regular phone number? What happens when you call it? |
| 04:17.32 | ChannelZ | I had one installer look at me all suspicious and ask "What do you want to know that for?" |
| 04:17.50 | nix8n82 | yeah it's bs.. |
| 04:17.56 | ChannelZ | Another says 'oh we don't use any test signals' (oh good no wonder the service is so crap) |
| 04:18.21 | ChannelZ | p3nguin: makes a 1kHz tone (or 1001 or something wierd I forget exactly) |
| 04:18.35 | nix8n82 | 1004 i think |
| 04:18.44 | nix8n82 | thats what I'm reading anyway |
| 04:18.51 | ChannelZ | yeah |
| 04:18.57 | p3nguin | What does that do for me if I call it? |
| 04:19.30 | nix8n82 | I ran fxotune -d -b1 and it gave me a number of .0514 |
| 04:19.31 | ChannelZ | it's a test signal for tuning the receive gain on a PSTN interface |
| 04:21.23 | nix8n82 | with (234.3/4557.0) |
| 04:21.44 | nix8n82 | do those numbers mean anything to anyone here? |
| 04:21.52 | nix8n82 | cause I don't think I get it |
| 04:21.53 | ChannelZ | not me |
| 04:22.20 | nix8n82 | and thats after I ran the fxotune -i it was like .10something |
| 04:23.27 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 04:25.09 | ChannelZ | I don't think the values it spits out have any real world meaning in comparison to anyone else |
| 04:25.45 | nix8n82 | bummer |
| 04:28.13 | ChannelZ | what's *'s default software echo canceller? MG2? |
| 04:31.01 | nix8n82 | thats what I was using but it seems to be just as affective as hpec |
| 04:34.09 | ChannelZ | I never got a license for their advanced software one |
| 04:34.54 | nix8n82 | we spent the $10 |
| 04:35.13 | ChannelZ | oh yes HPEC |
| 04:35.22 | ChannelZ | (brain was reading HWEC) |
| 04:35.27 | *** part/#asterisk LapTop006 (n=laptop00@bean.studio442.com.au) |
| 04:35.53 | nix8n82 | I don't think it's working even though the debug says echocancellation is enabled |
| 04:36.01 | ChannelZ | I think I can apply for free HPEC license because I have a 'real' digium card but I have to tear the box apart to get the serial number of the card which I forgot to write down before I installed it all |
| 04:36.02 | nix8n82 | when the channel comes up |
| 04:36.35 | nix8n82 | yeah you can..we spent the money cause they didn't want to tear the box open and get the serial #.. |
| 04:36.56 | ChannelZ | although I didn't buy directly FROM Digium |
| 04:37.12 | nix8n82 | I don't think they did either |
| 04:37.24 | ChannelZ | the only echo problems I have seem to be when someone on a cell phone calls in and dials my extension |
| 04:37.51 | nix8n82 | the only time we have echo is when the fxo port is inuse |
| 04:38.04 | ChannelZ | I think because the latency is so high from the cell it takes it a lot longer to figure out |
| 04:38.23 | nix8n82 | do you have echocancel when bridged? |
| 04:39.20 | ChannelZ | yes on my FXS ports |
| 04:41.03 | nix8n82 | I could be retarted..do you have to define echocancel in each channel or can it be before the definition of a channel? |
| 04:41.25 | nix8n82 | in chan_dahdi.conf |
| 04:41.40 | ChannelZ | like most things if it's defined above, it carries through to anything below |
| 04:42.04 | nix8n82 | ok |
| 04:42.12 | nix8n82 | makes me feel a little better |
| 04:52.35 | *** join/#asterisk gardo (n=gardo@110.55.251.124) |
| 04:52.39 | p3nguin | What type of Cisco device would use a 19V DC power adapter? |
| 04:53.45 | hardwire | according to google.. none |
| 04:57.09 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 04:58.56 | TJNII | My SoHo Cisco modem uses about a half dozen voltages ranging from -71 to +12. |
| 05:01.29 | p3nguin | I picked up a 7912G the other day that had a power cube packaged with it. I assumed it was the correct one. I plugged it in and the phone didn't power on, so I checked it and it's only 19V (the phone requires 48V). |
| 05:10.58 | *** join/#asterisk soman (n=somnath@115.119.41.110) |
| 05:31.02 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 05:36.49 | hardwire | sip ladders don't seem that difficult to generate |
| 05:38.35 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 05:48.05 | *** join/#asterisk bmg505 (n=leon@196-209-8-20-rndf-esr-5.dynamic.isadsl.co.za) |
| 05:50.47 | p3nguin | Google seems to think that the Cisco ADP-19FB power cube is for Polycom phones. |
| 05:55.04 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 06:06.53 | *** join/#asterisk masked (i=masked@hpavc/masked) |
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| 06:40.21 | *** join/#asterisk DND (n=arabia@213.132.40.2) |
| 06:40.58 | DND | hi guys, any one from UAE? i wanted to set the tone zone which currently sets to US and want to change it to uae |
| 07:02.29 | ChannelZ | hmm look in indications.conf |
| 07:02.36 | *** join/#asterisk grharry (n=root@ppp-94-65-220-52.home.otenet.gr) |
| 07:03.52 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:06.48 | grharry | hi, need some help with chan_dahdi.conf ... ( have no clue ) drivers are all installed and functional CLI> dahdi show status shows "B4XXP (PCI) Card 0 Span 4 OK 0 0 0", but dahdi show channels is empty ( since no valid conf provided ) ..... Anybody ?? |
| 07:07.39 | ChannelZ | was going to say something politically incorrect |
| 07:07.57 | ChannelZ | did you configure any channels in chan_dahdi.conf? |
| 07:09.13 | ChannelZ | (or should I say 'tried' since obviously it didn't work) |
| 07:09.38 | grharry | ChannelZ: Nothing whatsoever ... ( newbie and clue-less ) .... as far as politically status I am considered "democratic" ... |
| 07:11.06 | grharry | towards the "free" side ! :-) |
| 07:11.08 | ChannelZ | PC comment not for you |
| 07:11.11 | ChannelZ | anyways |
| 07:11.23 | ChannelZ | read chan_dahdi.conf.sample and make one. |
| 07:11.23 | grharry | ok |
| 07:12.52 | ChannelZ | barebones sample: http://pastebin.com/d338edf46 |
| 07:13.30 | ChannelZ | assuming they are FXS ports |
| 07:13.37 | ChannelZ | (I don't know that card) |
| 07:14.35 | *** join/#asterisk |Cybex| (n=John@212.178.82.20) |
| 07:14.39 | grharry | ISDN card |
| 07:15.32 | ChannelZ | oh |
| 07:15.37 | ChannelZ | well I have no idea specifically there |
| 07:16.26 | kaldemar | grharry: have you defined dchan and bchan's in system.conf? |
| 07:16.35 | kaldemar | and the span of course.. |
| 07:17.01 | grharry | you mean /etc/dahdi/system.conf ?? |
| 07:17.32 | kaldemar | yes. you have to configure the span and the channels there, and then in chan_dahdi.conf for asterisk usage. |
| 07:18.11 | DND | ChannelZ there's no UAE in the list but someone from yesterday told me that 2.2.0 has UAE in it |
| 07:18.42 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:19.22 | DND | currently i am setting up for te121 but our aex800 is having problems hanging up the call |
| 07:19.22 | DND | they have to manually hang up |
| 07:19.24 | grharry | kaldemar: http://pastebin.com/m28a3da01 configured by dahdi_genconf |
| 07:20.37 | kaldemar | grharry: that's probably ok, so now you only have to worry about chan_dahdi.conf |
| 07:20.54 | grharry | Currently the only ISDN port connected on the card is port 4 |
| 07:21.01 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:515d:516a:954a:bd37) |
| 07:21.32 | grharry | chan_dahdi.conf ( its rather empty all googled trials failed ) :-[ |
| 07:21.42 | kaldemar | do you use channels 10 and 11. |
| 07:21.46 | DND | why is it there's on one sample in chan_dahdi.conf |
| 07:22.02 | DND | i am also configuring PRI like grharry |
| 07:22.54 | grharry | yep I moved it to chan_dahdi.cong.org ( didnt function by default ) Sorry --- clueless newbie here .... |
| 07:22.56 | kaldemar | i see a load of stuff in the sample config |
| 07:24.33 | kaldemar | grharry: the most important lines for you in chan_dahdi.conf are switchtype, signalling and channel. |
| 07:25.13 | ChannelZ | DND: http://bugs.elastix.org/view.php?id=150 |
| 07:25.47 | ChannelZ | that's indications.. not sure on the actual signalling config though, I don't see anything in 2.2.0.2 specifically |
| 07:26.18 | *** join/#asterisk sergey (n=sergey@91.189.232.114) |
| 07:27.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:27.53 | grharry | kaldemar: http://pastebin.com/m785cbc69 is my initial chan_dahdi ... I've inluded the dahdi_channels.conf also |
| 07:28.23 | kaldemar | are you using a GUI? |
| 07:28.28 | grharry | no |
| 07:28.46 | grharry | meaning ?? FREEPBX or gtk ?? |
| 07:28.51 | kaldemar | what is dahdi_channels.conf ? |
| 07:29.04 | grharry | the one generated |
| 07:29.16 | kaldemar | freepbx or alike. i'm not interested in your desktop usage. :) |
| 07:29.28 | kaldemar | ah, by the conf generating script. |
| 07:29.45 | grharry | no freepbx at this point |
| 07:30.19 | kaldemar | well, where is the line located? |
| 07:30.31 | grharry | port 4 |
| 07:30.38 | kaldemar | coutry wise |
| 07:30.42 | grharry | gr |
| 07:30.46 | grharry | Greece |
| 07:30.52 | grharry | ( greetings ) |
| 07:30.55 | kaldemar | so it's probably euroisdn |
| 07:30.58 | grharry | ye |
| 07:31.00 | grharry | s |
| 07:31.21 | kaldemar | switchtype=euroisdn |
| 07:31.35 | grharry | done |
| 07:32.18 | kaldemar | signaling is then either bri_ptp or bri_ptmp, start with bri_ptmp |
| 07:33.30 | kaldemar | then group=0 and channel => 10-11 under those and you're ready to start trying Dial(DAHDI/g0/${EXTEN}) |
| 07:35.06 | kaldemar | and this all goes under [channels] |
| 07:35.25 | grharry | done !! CLI > dahdi show channels is empty |
| 07:35.39 | kaldemar | did you restart asterisk? |
| 07:35.53 | grharry | reload chan_dahdi ?? |
| 07:36.02 | kaldemar | no, restart asterisk |
| 07:36.09 | grharry | ok |
| 07:36.52 | grharry | module chan_dahdi not loaded !!! |
| 07:37.19 | kaldemar | show the error/warning |
| 07:37.56 | oej | You can always run "module load chan_dahdi.so" with verbosity turned on and you'll see the error message. |
| 07:38.30 | grharry | chan_dahdi.c: Unknown signalling method 'bri_ptmp' |
| 07:38.30 | grharry | chan_dahdi.c: Signalling must be specified before any channels are. |
| 07:38.52 | kaldemar | pastebin your config file |
| 07:39.00 | grharry | chan_dahdi ?? |
| 07:39.18 | grharry | man I feel bad ... and obliged of course !!! |
| 07:39.28 | kaldemar | the file you made those settings in, be that chan_dahdi.conf or dahdi_channels.conf |
| 07:40.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:40.41 | grharry | I hope a greek salad will suffice :-) . http://pastebin.com/m45b34d7b |
| 07:41.25 | kaldemar | you don't have a signaling parameter in there |
| 07:41.52 | kaldemar | ah, my eyes.. you do. |
| 07:43.32 | kaldemar | seems i was wrong on the signalling part, it's not bri_ptmp but bri_cpe_ptmp. it's been a while. :P |
| 07:45.08 | grharry | Unknown signalling method 'bri_cpe_ptmp' |
| 07:45.08 | grharry | <PROTECTED> |
| 07:45.27 | grharry | aterisk 1.4.xx |
| 07:45.49 | grharry | compiled from source |
| 07:46.15 | grharry | mISDN was working :-( |
| 07:47.39 | kaldemar | and you have libpri? |
| 07:48.23 | grharry | yes |
| 07:49.04 | grharry | 344055 2009-10-18 07:04 libpri.so.1.4 |
| 07:49.28 | kaldemar | does lsdahdi print the spans correctly? |
| 07:50.30 | grharry | http://pastebin.com/m5eac2d26 |
| 07:53.12 | kaldemar | interesting. |
| 07:53.27 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:55.05 | grharry | hold hold hold .... |
| 07:55.08 | grharry | <PROTECTED> |
| 07:55.09 | grharry | <PROTECTED> |
| 07:55.09 | grharry | <PROTECTED> |
| 07:55.27 | grharry | dahdi show channels |
| 07:55.36 | grharry | is it ok ????? |
| 07:55.43 | kaldemar | looks so. what did you do? |
| 07:55.57 | grharry | bri_cpe plain |
| 07:56.21 | *** join/#asterisk Tim_Toady (n=moi@adsl119-126.kln.forthnet.gr) |
| 07:58.20 | *** join/#asterisk kissg_hu (n=kissg_hu@BC2492DA.dsl.pool.telekom.hu) |
| 07:58.35 | kissg_hu | hello everyone! |
| 07:59.03 | ChannelZ | ahoy |
| 07:59.05 | kissg_hu | I'd like to ask a configuration-related question. is this the right place to do that? |
| 07:59.15 | ChannelZ | mostly |
| 07:59.48 | kissg_hu | well, I want to create a Hungarian translation for the built-in voice prompts in Asterisk |
| 08:00.10 | kissg_hu | I need to edit the application scripts, because the grammar differences of English and Hungarian |
| 08:00.25 | kissg_hu | which file should I use? how can I ensure, that application codes still can be changed? |
| 08:02.06 | ChannelZ | I'm not sure I understand what you mean by application codes and scripts |
| 08:02.32 | kissg_hu | excuse me, I meant feature codes (like *43 for echo test...) |
| 08:02.52 | ChannelZ | well let me say this |
| 08:02.55 | kissg_hu | the script is the application that runs when dialing a feature code |
| 08:03.09 | ChannelZ | all of the sounds are in /var/lib/asterisk/sounds (usually) |
| 08:04.18 | grharry | kaldemar: thanks for the help where Do I send the ouzo ?? |
| 08:04.32 | ChannelZ | you can change anything in there you want. You could also make a new directory like 'hu' |
| 08:05.00 | ChannelZ | and then in /etc/asterisk/asterisk.conf uncomment languageprefix=yes |
| 08:06.01 | ChannelZ | then change your language for your channels to hu and * will try to play files out of /var/lib/asterisk/sounds/hu/ first |
| 08:06.03 | grharry | kaldemar: Kiitos |
| 08:06.13 | ChannelZ | http://www.voip-info.org/wiki/view/Asterisk+multi-language |
| 08:06.30 | kaldemar | grharry: you're welcome. :) |
| 08:06.38 | kissg_hu | ChannelZ, I've already tried that, and it works okay |
| 08:06.56 | kissg_hu | my problem is, that sometimes it's not possible to translate voice prompts in place |
| 08:07.14 | ChannelZ | why not? |
| 08:07.34 | ChannelZ | You're referring to the voicemail prompts specifically perhaps? |
| 08:07.49 | kissg_hu | I need to concatenate some phrases, because Hungarian grammar is a lot more complex than in English |
| 08:08.21 | kissg_hu | for example, in English, it's completely okay to have separate recordings for words: your + extension + number + is |
| 08:08.37 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:08.44 | kissg_hu | but in Hungarian, I have to record this sentence as a whole like: your-extension-number-is |
| 08:09.06 | ChannelZ | yeah.. well unfortunately the voicemail application is somewhat inflexible in this regard |
| 08:09.56 | kissg_hu | I've created a extensions_custom.conf file and customized the Playback commands in the apps |
| 08:09.57 | gr0mit | <american> Why would anyone not speak English </american> |
| 08:10.22 | kissg_hu | it seems to work, but the disadvantage is, that I have to use static feature codes |
| 08:10.39 | ChannelZ | gr0mit: I wasn't thinking of the piecemeal words particularly in voicemail at first |
| 08:11.03 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 08:12.11 | kissg_hu | using English for voice prompts or instructions is completely okay for me, but unfortunately, a lot of Hungarian people don't speak English very well |
| 08:12.32 | ChannelZ | I'm not really sure what the elegant solution is (or if there even is one). Some things are hardcoded to play certain soundfiles in particular orders (voicemail being the worst offender) |
| 08:13.54 | ChannelZ | short of re-writing the phrase construction logic in the source code.... |
| 08:14.41 | kissg_hu | this means, I would have to modify to source to achieve what I want? |
| 08:14.58 | ChannelZ | actually there's a certain amount you can jack with in say.conf but I've never messed in this myself |
| 08:15.12 | kissg_hu | I see |
| 08:15.17 | kissg_hu | thanks very much |
| 08:15.29 | ChannelZ | it might actually be all you need |
| 08:17.33 | ChannelZ | but since I don't speak hungarian.. :) Way past my bed time, I'm off |
| 08:17.53 | kissg_hu | okay, sweet dreams and thanks again |
| 08:18.24 | ChannelZ | sure good luck |
| 08:18.36 | kissg_hu | thanks |
| 08:23.47 | *** join/#asterisk AlHafoudh (i=0fc3b951@gateway/web/freenode/x-rpznvegndtksbjvi) |
| 08:23.49 | AlHafoudh | hi all |
| 08:24.03 | kissg_hu | hello AlHafoudh |
| 08:24.17 | AlHafoudh | can I use voice modem as FXS? |
| 08:24.25 | kaldemar | no |
| 08:24.38 | AlHafoudh | :( why? |
| 08:25.41 | kaldemar | pretty similar to the reason why you don't use your laptop for a spoon. |
| 08:26.02 | AlHafoudh | sorry, FXO i mean |
| 08:26.06 | AlHafoudh | connection from PSTN |
| 08:26.06 | kissg_hu | I'd like to ask everyone who's reading this and has been using Asterisk for a long time (months, years...) to share the experiences |
| 08:26.31 | AlHafoudh | kissg_hu: just ask |
| 08:26.38 | kissg_hu | is it a good choice for a long term use? |
| 08:26.52 | AlHafoudh | kaldemar: i mean FXO, sorry |
| 08:27.07 | kissg_hu | as far as I know, version 1.4 of Asterisk is production-ready, so I presume it is used in many live installations |
| 08:27.11 | kissg_hu | what are the experiences? |
| 08:27.24 | kissg_hu | (I'm new to VoIP and to Asterisk) |
| 08:27.46 | kaldemar | AlHafoudh: not for that either. if you're looking for a cheap way to connect your asterisk box to a land line, consider an ATA. |
| 08:28.37 | AlHafoudh | kaldemar: i read in wiki: http://en.wikipedia.org/wiki/Foreign_exchange_office "Analog telephone handsets, fax machines and (analogue) modems are FXO devices" |
| 08:28.39 | kaldemar | kissg_hu: it is used in a huge amount of installations. 1.6.0 is also considered stable enough for production by most people. |
| 08:29.20 | kaldemar | AlHafoudh: yes, but to use them as an FXO device in asterisk is a whole another thing. |
| 08:29.43 | gr0mit | kissg_hu, i have used * for approx 4 years. rock solid |
| 08:30.04 | AlHafoudh | kaldemar: is it bad? |
| 08:30.06 | kissg_hu | It's 100% clear to me, that for new installations it is really the best choice, as it's free compared to a commercial PBX, which are said to be very expensive |
| 08:30.45 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:30.45 | kaldemar | AlHafoudh: not bad, but a "forget it". |
| 08:30.47 | kissg_hu | but what's the case with existing installations? when is it really a good choice to switch to Asterisk from a commercail PBX? |
| 08:31.05 | AlHafoudh | kaldemar: i read, codec problem, duplexity, etc. ok |
| 08:31.06 | WinZ | AlHafoudh, I've seen lots of comments on using voice modems with asterisk - the voice quality is bad bad |
| 08:31.18 | AlHafoudh | kaldemar: so what is the really cheap solution, cheap because i want just to learn |
| 08:31.35 | kaldemar | kissg_hu: if you want to cut costs and get flexibility beyond any commercial PBX. |
| 08:32.18 | kaldemar | AlHafoudh: an ATA, like i said earlier. i hear the linksys ones are good. |
| 08:33.04 | kissg_hu | gr0mit: thanks for sharing your experience. do you use a large deployment? |
| 08:33.08 | AlHafoudh | kaldemar: nice, ok, so it makes SIP from POTS and i catch it via ethernet, right? |
| 08:33.20 | kaldemar | AlHafoudh: exactly. |
| 08:33.41 | gr0mit | well, I used to work at Motorola. When I left, I had about 18*servers all over the world |
| 08:33.52 | gr0mit | running scripts to make and receive calls to mobile phones |
| 08:34.16 | gr0mit | I have servers as far afield as Colombia, Buenos Aires, Melbourne, Chicago and Basingstoke |
| 08:34.31 | AlHafoudh | kaldemar: ok, great, thanks, and what about FXS? |
| 08:34.42 | gr0mit | with a mixture of T1, E1/R2 and E1/PRI |
| 08:34.47 | AlHafoudh | kaldemar: its something USBish for FXS ? :) |
| 08:35.21 | kaldemar | AlHafoudh: there are ATA's for FXS too, but if you want to go cheap for testing things out, use a soft phone. |
| 08:35.40 | gr0mit | <PROTECTED> |
| 08:35.47 | gr0mit | also using* |
| 08:36.03 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 08:36.40 | AlHafoudh | kaldemar: yes, i just wanted home to get VoIPified, we have new cordless siemens gigaset, but analog, i want it to be used |
| 08:37.38 | trogs | the linksys pap2t is a reasonable ATA |
| 08:37.46 | kissg_hu | gr0mit: what's your experience regarding FXO cards? I saw some noname cards which are said to be compatible with Asterisk |
| 08:38.01 | kissg_hu | or is it the best to use the cards made by Digium? |
| 08:38.06 | kaldemar | AlHafoudh: then go for an ATA and plug the gigaset in it. |
| 08:38.32 | gr0mit | I have had nothing but trouble with anything analog related. |
| 08:38.56 | gr0mit | I would always avoid any kind of analog card, except possibly for a few analogue extensions i.e. FXS |
| 08:39.02 | kaldemar | kissg_hu: avoid the noname ones at least |
| 08:39.15 | AlHafoudh | kaldemar: ok |
| 08:39.16 | gr0mit | I would strongly recommend an ISDN card with a BRI |
| 08:39.51 | kissg_hu | I see. So, if a company needs to send and receive faxes, it's much better to use an ISDN line than PSTN? |
| 08:40.02 | gr0mit | definitely |
| 08:40.09 | kissg_hu | I've read, that faxing over IP is not a reliable thing |
| 08:40.17 | trogs | faxing can be problematic |
| 08:40.20 | gr0mit | that is also correct |
| 08:40.39 | trogs | you should always use g.711ulaw or alaw codec if you're going to be doing faxing |
| 08:40.48 | trogs | that'll give you the best quality. |
| 08:41.26 | trogs | also, if you're using ATAs to connect your fax machines, you need to turn off the echo cancellation and a few other things. |
| 08:41.26 | Gido-E | kissg_hu reliable enough. |
| 08:41.36 | kissg_hu | yes, that's exactly what I read. using compression can really mess up thing with faxing |
| 08:41.55 | trogs | wish people would just stop using faxes :) |
| 08:42.22 | kissg_hu | trogs: yes, but it's somehow still very popular in some countries (like Hungary) |
| 08:42.37 | trogs | oh, it's popular everywhere still. |
| 08:42.38 | Gido-E | trogs i don't have problems with fax. |
| 08:43.19 | trogs | i had more problems getting alarms to talk properly over voip. |
| 08:43.25 | trogs | for the most part the faxes just work |
| 08:43.38 | AlHafoudh | kaldemar: is this ok for FXS? http://cgi.ebay.co.uk/linksys-PAP2T-PAP2T-NA-SIP-VOIP-adapter-unlocked-New-3O_W0QQitemZ350193046507QQcmdZViewItemQQptZUK_Computing_Networking_SM?hash=item518921d3eb |
| 08:43.58 | kissg_hu | trogs: even when sending through a SIP trunk to a PSTN line? |
| 08:45.10 | *** join/#asterisk tokozedg (n=tokozedg@212.58.115.190) |
| 08:45.18 | kaldemar | AlHafoudh: yes, should be. |
| 08:45.38 | trogs | kissg_hu: yeah, should be okay. provided you get all the settings on the ATA correct. |
| 08:45.59 | trogs | and you need to lock the FAX down to about 9600 speed if possible |
| 08:46.13 | trogs | 14.4k faxing doesn't work terribly well. |
| 08:47.36 | tokozedg | hello, while sending fax with asterisk 1.6.1 with d-link fxs and t.38 turned on on both sides, i get this error:http://pastebin.com/m640ccd62, how can i correct? |
| 08:49.38 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 08:52.45 | kissg_hu | probably a lame question, but as I said I'm new to IP telephony: is it possible to use a "normal" tel. number for a SIP trunk? |
| 08:52.51 | kissg_hu | if yes, how does it work? |
| 08:52.55 | AlHafoudh | kaldemar: can i use SPA3000 for FXS AND FXO with asterisk that asterisk will answer the calls and make them through the SPA3000 ? |
| 08:53.14 | kissg_hu | I mean a normal number like the one is used for PSTN lines |
| 08:53.51 | trogs | works just the same. |
| 08:54.58 | *** join/#asterisk AlHafoudh_ (i=0fc3b951@gateway/web/freenode/x-zxbgdtnflznduibn) |
| 08:54.58 | kissg_hu | so, if I subscribe at a VoIP provider, I can ask for a number (or probably even a group of numbers?) |
| 08:54.59 | *** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com) |
| 08:55.10 | Gido-E | kissg_hu yes |
| 08:55.55 | trogs | yep |
| 08:56.02 | kissg_hu | and technically, how does the provider know, that a number belongs to a SIP trunk? |
| 08:57.09 | trogs | well, they have some kind of softswitch which takes the lines in from either an SS7 interconnect, or an E1/T1 line, and maps it to your sip account. |
| 08:57.35 | trogs | there is nothing differenet about the numbers |
| 08:57.52 | kissg_hu | so that way, I can even have a number with a local prefix (for example, 22 in our county) |
| 08:57.55 | trogs | it's just that they deliver the last mile to you via SIP, rather than via a copper cable. |
| 08:58.10 | trogs | yep |
| 08:58.57 | kissg_hu | I see. but if I ask for that, I have to sign a declaration that I'm using the number at a given geographic location. Is this right? |
| 08:59.22 | trogs | depends on the telco you're getting the numbers from, i guess. |
| 08:59.53 | trogs | there's of course nothing to stop you sending it somewhere else. i have a whole bunch of numbers pointed at my desk phone from all over the country. |
| 09:00.34 | *** join/#asterisk baijum (n=baiju@203.123.174.162) |
| 09:00.47 | *** join/#asterisk AlHafoudh (i=0fc3b951@gateway/web/freenode/x-lsyeqznnfrsfznqc) |
| 09:01.34 | *** join/#asterisk trixadmin (n=sreeraj@122.166.23.169) |
| 09:01.39 | trixadmin | hello team |
| 09:01.48 | kissg_hu | hi, trixadmin |
| 09:02.03 | trixadmin | yes kissg_hu |
| 09:03.14 | kissg_hu | I wonder what distributions are mainly used for live deployments. I've tried Elastix and trixbox, both seems to be quite usable |
| 09:03.32 | *** join/#asterisk Dovid (n=annon@213.8.118.62) |
| 09:03.35 | Dovid | j #asterisk-il |
| 09:03.41 | kissg_hu | both have their pros and contras, but I somehow prefer Elastix |
| 09:04.04 | Dovid | hi |
| 09:04.09 | kissg_hu | hi Dovid |
| 09:04.27 | kaldemar | kissg_hu: people tend to use the linux distribution they're most familiar with. it doesn't really matter which one you use, as long as it's easy to maintain and up to date. |
| 09:04.30 | Dovid | i wrote an agi script in php and I am noticing defunct next to some of my agi's. what causes that (I am sure something inmy code....) |
| 09:04.32 | trogs | i have got a few trixbox instances deployed |
| 09:04.38 | trogs | it's a nice starting point |
| 09:04.47 | trogs | beats the heck out of configuring all that stuff by hand |
| 09:05.07 | kaldemar | kissg_hu: this channel is more for people who don't use a GUI but plain asterisk. there are separate channels for GUI users. |
| 09:05.32 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 09:06.11 | kissg_hu | kaldemar: in fact, it's no problem to me to use a CLI, but in some cases it's quicker to use a GUI |
| 09:06.23 | kissg_hu | for example, if you have to add a large number of extensions |
| 09:07.01 | kissg_hu | of course, it can be done by using scripts, which is just as quick, I think |
| 09:07.03 | kaldemar | kissg_hu: sure. a GUI is fine if you can manage with what the GUI has to offer. |
| 09:07.16 | doolittlework | what do i need to do to not record iax calls in cdr? |
| 09:07.32 | kaldemar | it's actually much quicker with scripts. |
| 09:08.01 | trixadmin | hello dovid do you have idea how to create a script for autodialout @ scheduled time.... |
| 09:08.06 | kaldemar | doolittlework: use app NoCDR in dialplan |
| 09:08.10 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
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| 09:08.33 | doolittlework | before call starts or when it ends kaldemar? |
| 09:09.34 | kaldemar | doolittlework: before it starts. |
| 09:09.38 | trixadmin | hello Team do you have idea how to create a script for autodialout @ scheduled time.. |
| 09:09.51 | *** join/#asterisk soman (n=somnath@118.102.130.5) |
| 09:10.01 | doolittlework | thx |
| 09:10.27 | trixadmin | autodialout can be done with.call file at outgoing folder but what about scheduled time........ |
| 09:10.36 | kaldemar | trixadmin: 1. use a scheduling provided by system and generate a call file 2. generate a call file with it's creation time set to the moment you want to make the call |
| 09:10.54 | kaldemar | trixadmin: first 2 choices that come to my mind. |
| 09:11.39 | Dovid | trixadmin: explain a bit more |
| 09:11.53 | Dovid | you want to have like a dialer ? |
| 09:11.54 | Dovid | !vicidial |
| 09:11.59 | Dovid | ~vicidial |
| 09:12.00 | infobot | [vicidial] a predictive dialer available from http://astguiclient.sourceforge.net/vicidial.html . |
| 09:12.56 | kissg_hu | how do most installations handle emergency calls? I heard that using a separate trunk for that purpose is always a good idea |
| 09:13.16 | kissg_hu | and connecting it to a PSTN line is more than recommended |
| 09:13.30 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 09:13.35 | Dovid | i try to get my clients to use the PSTN |
| 09:13.41 | Dovid | you can always use E911 as a back up |
| 09:13.47 | trogs | best effort. I send all my emergency out one of my e1 lines, rather than my sip trunks. |
| 09:14.00 | trogs | but i mean, everyone calls from their cellphone these days anyway |
| 09:14.01 | trixadmin | no. dovid. Actually i need to intiate calls to external numbers at scheduled time.Time,number will book through web. Ex: Custemer A will book a call through web and needs Asterisk box will take control of the call scheduled time.. |
| 09:14.58 | trogs | of course the problem is the emergency operator sees the location as the location of the e1 trunk delivery, not the callers location |
| 09:15.59 | kissg_hu | trogs, yes but if the caller tells where she/he is, the problem is solved :) |
| 09:16.24 | xrmx__ | ot question, does anyone know how to readd the background of polycom ip650 phones after i've upgraded the sip app from 2.x to 3.x.x? |
| 09:16.30 | kissg_hu | but of course, in some cases it may not be possible to tell the exact location |
| 09:18.04 | kissg_hu | if there really is an emergency some people would say "I don't know where I am or who I am but help!" :) |
| 09:19.03 | kissg_hu | I hope that would not happen to me |
| 09:20.03 | trixadmin | hello dovid hope you got my situation............. |
| 09:20.20 | kissg_hu | alright, I think I'm leaving now. don't want to be too off-topic :) |
| 09:20.29 | trixadmin | and thanks kaldemar ............. |
| 09:20.38 | kissg_hu | thanks for all your help! this is great channel |
| 09:20.41 | kissg_hu | bye! |
| 09:20.44 | trixadmin | is vicidial will enough for my task? |
| 09:24.02 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 09:25.28 | trixadmin | hello dovid are there ............. please help me |
| 09:37.33 | doolittlework | i am trying to setup a remote sip client on my asterisk box |
| 09:37.47 | doolittlework | i have port 5060 open on firewaill |
| 09:38.06 | doolittlework | the remote phone connects and i can se it is dialing but no voice |
| 09:38.19 | doolittlework | is there other ports that needs to be open |
| 09:38.51 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 09:41.00 | kaldemar | ~sipnat |
| 09:41.01 | infobot | [~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 |
| 09:41.05 | kaldemar | doolittlework: ^^ |
| 09:41.18 | doolittlework | yes |
| 09:41.42 | doolittlework | thx |
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| 10:01.35 | galeras | What is the best forum to post some txfax issues? |
| 10:07.15 | *** join/#asterisk BlackSLik (n=jamesmar@64.191.115.29) |
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| 10:26.50 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 10:27.46 | DND | guys i have a te121+echo canceller pci-e card. and when i do dahdi_cfg -v i see "Setting echocan for channel 16 to none" |
| 10:27.49 | DND | how come? |
| 10:28.43 | DND | all other channels have defined mg2 asa echo cancel except this |
| 10:29.43 | kaldemar | channel 16 is for signalling |
| 10:32.07 | manxpower | You don't usually want to echo cancel the data (D) channel. |
| 10:32.20 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 10:32.45 | DND | no, i just installed a PRI card and just looking at the settings. |
| 10:33.01 | DND | this is new to me since im using an AEX800 before |
| 10:33.46 | DND | if i want to have 32 channels, i will need to join the 2 jumper pins right? for E1 line? |
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| 11:06.22 | waa | I'm getting an error when compiling dahdi-tools (trunk) in ubuntu box http://pastebin.com/m794d2251 |
| 11:08.32 | *** join/#asterisk AlHafoudh (i=0fc3b953@gateway/web/freenode/x-vbmuquykxwkzdmfu) |
| 11:21.22 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:24.06 | tzafrir_laptop | waa, that's interesting |
| 11:24.14 | tzafrir_laptop | what version of ubuntu? |
| 11:24.22 | tzafrir_laptop | What kernel? |
| 11:25.34 | garymc | Hey anyone know anything about cisco PWR switches and can i use it to power polycom phones? |
| 11:25.42 | waa | tzafrir_laptop, ubuntu 9.04 kernel 2.6.28-15-server |
| 11:26.17 | tzafrir_laptop | waa, also, what is the output of: ls -l /sys/bus/usb/devices/3-1/serial |
| 11:27.47 | galeras | I'm Using TxFAX to send faxes via Zaptel PRI. I have 2 PSTN PRI Providers, with the first provider, all faxes are trasmited fine. With the second provider, faxes can't be sent, we suspect about the setting of this PRI provider, perhaps is doing some compression somewhere |
| 11:27.48 | galeras | For detailed logs please take a look of http://www.pastebin.ca/1634790 |
| 11:28.18 | waa | tzafrir_laptop, ls: cannot access /sys/bus/usb/devices/3-1/serial: No such file or directory |
| 11:28.59 | waa | tzafrir_laptop, ls -l /sys/bus/usb/devices/ |
| 11:29.00 | waa | 1-0:1.0/ 2-0:1.0/ 3-0:1.0/ 3-1/ 3-1:1.0/ 4-0:1.0/ 5-0:1.0/ 6-0:1.0/ 7-0:1.0... |
| 11:29.38 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 11:29.49 | tzafrir_laptop | what files do you have under 3-1 ? |
| 11:30.40 | tzafrir_laptop | and this big question is: why is that code actually run on a mere 'perl -c' (compile test)? |
| 11:30.40 | waa | tzafrir_laptop, http://pastebin.com/m1ae81a1 |
| 11:30.59 | Gido-E | In the netherlands when connecting to ISDN PRI, is it normally T1 or E1 ? |
| 11:31.32 | tzafrir_laptop | waa, anyway, a quick fix would be to touch the "offending" .depend file which would claim your system has done this test |
| 11:33.43 | tzafrir_laptop | Gido-E, E1 |
| 11:35.04 | Gido-E | Thnx! tzafrir_laptop |
| 11:45.22 | DMeloUK | garymc what model do you have? |
| 11:45.40 | garymc | WS-C3524-PWR-XL-EN |
| 11:45.44 | garymc | Cisco |
| 11:46.20 | garymc | Cant seem to find out if my polycom ip330's and ip650's will work with this. |
| 11:46.27 | garymc | I havnt bought it yet, but have been given a good price for it |
| 11:46.36 | garymc | well if its suitable |
| 11:46.39 | DMeloUK | not usually - that one is pre-standard |
| 11:46.59 | DMeloUK | you can force it by messing around with pin connections |
| 11:47.38 | DMeloUK | hang on let me find you a link |
| 11:49.14 | waa | tzafrir_laptop, You know what should I alter in dahdi-tools/xpp/.depend? |
| 11:49.53 | garymc | ok |
| 11:49.58 | tzafrir_laptop | waa, the code is there to test that those modules "compile" fine. That is, that the local perl likes them |
| 11:50.19 | tzafrir_laptop | aparantly it tries to also run a part of them and fails with some point |
| 11:51.14 | tzafrir_laptop | it will bite you later on, but for now you can fake a positive succefful test by touching those .depend files |
| 11:51.19 | DMeloUK | wait - that's the other way around - forcing an old cisco phone to use a 802.af switch |
| 11:53.19 | waa | tzafrir_laptop, I will try |
| 11:54.08 | DMeloUK | you might want to look at a linksys srw224p if you just need a cheap switch to run phones |
| 11:54.12 | DMeloUK | or a cisco 3 |
| 11:54.15 | DMeloUK | 3560 |
| 11:55.56 | *** join/#asterisk volker- (n=volker@h1311547.stratoserver.net) |
| 11:55.57 | volker- | hi |
| 11:56.01 | DMeloUK | if you still wanted to go with that switch and only had a few polycom's you can always buy the ac adaptors I suppose ;) |
| 11:56.18 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 11:56.33 | volker- | i want that "strangers" can call in my asterisk by user@ip |
| 11:56.40 | volker- | but i just get a forbidden message |
| 11:57.21 | volker- | how can i fix this |
| 11:57.57 | garymc | I want a switch so i dont need the power supplys |
| 11:58.11 | garymc | and I can get this switch for £60 |
| 11:58.29 | garymc | 24 ports of POE, i thought was a good price, but does or will it do the job |
| 11:58.33 | DMeloUK | it won't run the 330 or the 650 |
| 11:58.45 | DMeloUK | it might run older 301 and 601 |
| 11:59.32 | DMeloUK | go with a linksys or later cisco - the trick is to look for something which support 802.af |
| 11:59.54 | garymc | right ok, you 100% on this? |
| 11:59.56 | DMeloUK | and I know for a fact the 3524 does not (bitter experience talking) |
| 12:00.16 | DMeloUK | I made the same mistake a few months back |
| 12:00.36 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 12:00.49 | DND | hi any sites for setting up asterisknow with pri line? |
| 12:01.17 | garymc | ive just done that DND |
| 12:01.19 | kaldemar | DND: someone at #asterisknow might know better. |
| 12:01.49 | DND | gary, can you share it to me? mine is only te121 so its much easier |
| 12:01.58 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 12:02.35 | doolittlework | hi there can one someone explain the differece between dahdi and zaptel |
| 12:03.09 | doolittlework | is zaptel for older hardware? |
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| 12:09.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:09.36 | mbrevda | how can I tell asterisk to always use md5 authentication automaticaly? |
| 12:12.23 | volker- | bye |
| 12:12.27 | *** part/#asterisk volker- (n=volker@h1311547.stratoserver.net) |
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| 12:19.47 | Skeeter- | dahdi-linux wont install on debian |
| 12:24.52 | Chainsaw | Skeeter-: That seems unlikely. |
| 12:25.04 | Chainsaw | Skeeter-: Anyhow, the car doesn't fit in the garage. |
| 12:26.39 | Skeeter- | saying that there is a missing kernel source |
| 12:27.02 | Chainsaw | It would like some kernel sources. Why haven't you granted it that wish? |
| 12:27.11 | Skeeter- | i downloaded it, still get the same things, so i redbuild with the latest kerlnel, still not wkring |
| 12:27.35 | Chainsaw | It wouldn't just like kernel sources. It would like sources for the kernel that you are running *now*. |
| 12:30.06 | Skeeter- | umm |
| 12:30.12 | [TK]D-Fender | And it isn't the sources it wants... its the HEADers |
| 12:30.12 | Skeeter- | formating right now |
| 12:30.27 | [TK]D-Fender | Skeeter-: .... formatting? |
| 12:30.31 | Skeeter- | i will try on a fresh install |
| 12:30.38 | [TK]D-Fender | facepalms |
| 12:31.37 | beek | Morning [TK]D-Fender |
| 12:32.29 | [TK]D-Fender | beek: mornin' |
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| 12:39.27 | afink | morning everyone. |
| 12:39.41 | beek | morning |
| 12:41.44 | afink | I'm having some trouble this morning w/realtime extensions. I setup it up with out any problems on 4 out of 5 * boxes but this one is giving me trouble. * can't connect to it. It keeps giving me an error 1045 (mysql). I've checked permissions on the db, checked my extconfig.conf, loaded the modules in modules.conf and setup my res_mysql.conf. Anybody know anything else I can check? |
| 12:42.22 | beek | can you connect to it from your asterisk box using the mysql CLI? Ignore Asterisk for now. |
| 12:42.34 | afink | yes I can |
| 12:42.49 | beek | Using the same credentials as you want for Asterisk? |
| 12:42.56 | afink | yep |
| 12:43.16 | beek | Anything in the logs? |
| 12:43.45 | afink | which ones should I check? |
| 12:43.55 | beek | CLI and /var/log/asterisk/messages |
| 12:45.02 | afink | [Oct 21 07:22:03] ERROR[11583] res_config_mysql.c: MySQL RealTime: Failed to connect database server asterisk on 192.168.140.190 (err 1045). Check debug for more info. |
| 12:45.24 | beek | Okay, so turn up verbosity and turn debugging on. then try again. |
| 12:46.00 | beek | Err :1045 Access denied for user 'root@localhost'(using password NO) |
| 12:46.39 | beek | 1045 is a perms problem, according to the MySQL site. Ensure that you don't have a typo or weird non-printing character in your config |
| 12:46.51 | afink | ok |
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| 12:52.57 | keeper82 | Hi, I'm trying to enable mysql cdr log: I wrote the right parameters in cdr_mysql.conf but the table is empty |
| 12:53.47 | beek | keeper82: Did you test your database connection from the * box by using the mysql CLI? |
| 12:54.21 | keeper82 | yes, mysql is on the same server as * |
| 12:54.43 | beek | So can you connect to the database from the CLI using the same credentials as what you want to from Asterisk? |
| 12:55.08 | keeper82 | yes |
| 12:55.20 | beek | Did you "logger reload" |
| 12:55.25 | beek | or restart asterisk |
| 12:55.36 | keeper82 | yes :P |
| 12:55.46 | beek | What does the log say? |
| 12:56.03 | keeper82 | you mean /var/log/asterisk/messages? |
| 12:56.07 | beek | yes |
| 12:56.35 | beek | turn up verbosity and debugging and see what's happening. Asterisk is very good about telling you what is going on. |
| 12:56.52 | keeper82 | in the log I see: |
| 12:57.01 | beek | ~pb |
| 12:57.02 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 12:57.25 | keeper82 | uhm it seems that * can't find the settings I wrote in cdr_mysql.conf |
| 12:58.22 | keeper82 | are they loaded automatically or I have to #include them? |
| 13:00.21 | keeper82 | http://pastebin.com/m39bbad64 |
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| 13:08.40 | afink | beek, here is what I have tried so far. In the logs I saw complaints about not being able to find the mysql socket, so I corrected that in res_mysql.conf, in modules.conf I found a typo. I also changed the credentials in res_mysql.conf to be the root user |
| 13:09.39 | keeper82 | ok, found it, i was calling a number without answering, I changed unanswered=yes and then the call was logged |
| 13:09.45 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 13:12.28 | afink | I think I might have found my problem too, there was a blank user in mysql |
| 13:13.21 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:14.29 | Katty | :> |
| 13:14.42 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
| 13:14.47 | d5s | Hi, I'm currently using asterisk version 1.6.0.13. I'm using a audio file (wav format) within calls to 'EXEC Read' function. It does work, but when the audio is finished I hear a noise (like: KRrrrr) then I can press the digit I want. It is important to say that it only happens on my USB B2k device. I know it is not well supported but should work, right? |
| 13:17.16 | Katty | Dear Universe, thank you for people like http://www.youtube.com/watch?v=bgo0CDL6bd0 Love, Katty |
| 13:21.39 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:22.50 | Katty | hi ariel_ |
| 13:23.03 | ariel_ | hello Katty |
| 13:23.08 | ariel_ | morning everyone |
| 13:23.09 | [TK]D-Fender | Katty: Mew. |
| 13:23.53 | Katty | hugs on [TK]D-Fender |
| 13:24.32 | [TK]D-Fender | hugz on the Katty |
| 13:25.49 | d5s | I would like to know if any of you guys has some expertise on USB B2K with asterisk, please. |
| 13:26.34 | ariel_ | what is USB B2K? |
| 13:27.10 | [TK]D-Fender | ariel_: Skype crap |
| 13:27.12 | beek | Hello Katty |
| 13:27.16 | Katty | hi beek :> |
| 13:27.20 | Katty | huggles beekers. |
| 13:27.25 | [TK]D-Fender | d5s: Unsupported dead-end. |
| 13:27.27 | beek | purrs |
| 13:28.16 | Katty | [TK]D-Fender: so i've been tinkering with my google voice account. |
| 13:28.29 | Katty | [TK]D-Fender: it's pretty schnazzy |
| 13:30.23 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 13:32.56 | Lantizia | hey I'm messing with call files.... trying to make that'll run a second from now... |
| 13:32.58 | Lantizia | touch -t $((`date +%y%m%d%H%m.%S`+.01)) /tmp/a.call |
| 13:33.20 | Lantizia | i.e. so it touches it with the current date/time/second and adds .01 (1 second)... but it doesn't work :( |
| 13:33.29 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:33.50 | afink | beek: that blank user was my issue |
| 13:34.01 | afink | thanks for your help! |
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| 13:37.07 | *** join/#asterisk wonderworld (n=w@62.143.22.226) |
| 13:37.44 | [TK]D-Fender | Lantizia: Try asking in ##linux or #(yourdistrohere) |
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| 13:42.11 | d5s | Hey [TK]D-Fender do you a replacement for USB B2K? |
| 13:43.33 | d5s | ariel_: It was originally developed to be used with skype, yes. But We are currently using it as a FXS interface to asterisk. But it is not behaving as expected. |
| 13:44.03 | d5s | ariel_: For me it is been used as a FXS to USB adapter. |
| 13:45.10 | ariel_ | USB to FXS = Majic jack.... |
| 13:45.27 | beek | afink: Glad its fixed |
| 13:45.46 | d5s | We also have one of this. Does not work properlly ;( |
| 13:46.23 | [TK]D-Fender | d5s: thats what "unsupported" means |
| 13:46.28 | wonderworld | hey guys |
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| 13:46.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:48.05 | ariel_ | starts burning his ISO of elastix to cd so he can install it on his xen server. |
| 13:48.52 | d5s | [TK]D-Fender: I understood that it is unsupported. Do you know a better alternative? |
| 13:49.16 | d5s | ariel_: By the way, magic jack isn't supported under linux :( |
| 13:49.25 | ariel_ | d5s: yes I know. |
| 13:49.26 | [TK]D-Fender | d5s: Something that is, like a standards-based ATA |
| 13:51.46 | d5s | I'm really sorry for bothering you guys, but could you point me a expecific model for a standards-based ATA (USB to FXO)? Or could you point me a web site where I can find one to buy? |
| 13:51.54 | d5s | I just need one FXS port |
| 13:52.05 | ariel_ | Linksys 3102 for both |
| 13:52.28 | ariel_ | works great I have also a sipura spa3000 that is still working |
| 13:52.52 | ariel_ | spa1000 for just one fxs is a great unit |
| 13:53.36 | d5s | Thanks ariel_ I will take a look at spa1000 |
| 13:55.52 | ariel_ | d5s: any time. |
| 13:58.35 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 14:00.12 | [TK]D-Fender | d5s: SPA's are NOT USB devices |
| 14:00.32 | [TK]D-Fender | ariel_: And the SPA-1000 was discontinued a long time ago.. |
| 14:00.56 | mbrevda | anyone using Nextiva? |
| 14:01.09 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:01.20 | ManxPower-work | ~answers |
| 14:01.21 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 14:01.25 | [TK]D-Fender | d5s: there is no current supported low density USB FXS for * |
| 14:01.43 | [TK]D-Fender | ~ata |
| 14:01.43 | infobot | ata is probably Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 14:02.11 | mbrevda | [TK]D-Fender: why about xorcoms? |
| 14:02.15 | mbrevda | *what |
| 14:02.29 | [TK]D-Fender | mbrevda: LOW DENSITY <- |
| 14:02.40 | mbrevda | how man do you concider low? |
| 14:02.45 | mbrevda | *many |
| 14:02.59 | [TK]D-Fender | < 2 |
| 14:03.02 | [TK]D-Fender | er... 3 |
| 14:03.13 | mbrevda | concedes |
| 14:03.33 | [TK]D-Fender | mbrWould help if you followed what they guy is looking for :) |
| 14:03.53 | mbrevda | probobly a good idea ;) |
| 14:04.09 | mbrevda | but isnt irc ofr blindly asking dumb questions? |
| 14:07.35 | tzafrir_laptop | is it? |
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| 14:11.11 | WinZ | anyone has formal voicemail greeting records? |
| 14:11.24 | WinZ | in .wav/mp3 or whatever |
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| 14:18.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.53 | ManxPower-work | *grumble* At least my boss's boss waited until after breakfast to ask for something that is impossible. |
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| 14:21.14 | [TK]D-Fender | WinZ: ... huh? "formal"? |
| 14:24.01 | WinZ | [TK]D-Fender, like, "No one is available to take you call at the moment. Please, leave your phone number, name and message after the beep." |
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| 14:24.28 | [TK]D-Fender | WinZ: "core show application voicemail" <- |
| 14:24.38 | WinZ | [TK]D-Fender, the default ones don't really fit, and as I'm not a native speaker, this is a problem |
| 14:24.42 | WinZ | let me check |
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| 14:25.28 | [TK]D-Fender | WinZ: It's voicemail... the world understands these basic instructions. |
| 14:26.08 | [TK]D-Fender | WinZ: "Hi, you don't know me. I jsut called you. Call me back! BYE!!! <click>" |
| 14:26.43 | WinZ | [TK]D-Fender, yes, for personal use it's ok, but for a company, "The number you're calling is not available" - isn't very good |
| 14:27.22 | [TK]D-Fender | WinZ: yes, well "no-one" implies a GROUP whereas most mailboxes are personal to a user. |
| 14:27.38 | *** part/#asterisk errr (n=errr@fedora/errr) |
| 14:27.48 | [TK]D-Fender | WinZ: Feel free to commision a recording you like better |
| 14:28.35 | Chainsaw | You can always ask whoever in the company has the best telephone voice :) |
| 14:28.42 | WinZ | [TK]D-Fender, I'm not blaming the default recordings :) I'm saying, the default recordings do not fit in my case |
| 14:28.49 | WinZ | that's what I should do |
| 14:28.54 | WinZ | have someone record it |
| 14:29.04 | Chainsaw | Indeed, and then you create a directory for your language. |
| 14:29.13 | ManxPower-work | WinZ: You should hire a professional voiceover service |
| 14:29.13 | Chainsaw | Set Asterisk to use it and you're done. |
| 14:29.29 | Chainsaw | As you can see ManxPower is a consultant during the day. |
| 14:29.47 | [TK]D-Fender | By night he's BATMAN! |
| 14:30.03 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 14:30.03 | [TK]D-Fender | na na na na na na na na na na na na na na na na BATMAN! |
| 14:30.20 | ManxPower-work | ~manxpower |
| 14:30.21 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 14:30.31 | ManxPower-work | I no longer do consulting |
| 14:30.51 | Chainsaw | The mindset's still with you though ;) |
| 14:36.06 | ManxPower-work | I'm waiting for the day [TK]D-Fender retires from this channel. Then y'all will be screwed. 8-| |
| 14:36.45 | Chainsaw | What, Fender? I'm sure he was part of the "Freenode support channel" package deal :) |
| 14:37.13 | ManxPower-work | [TK]D-Fender doesn't even work for Digium |
| 14:37.42 | Chainsaw | No, Telephone Kingdom right? |
| 14:37.45 | [TK]D-Fender | If you don't like free support, I'll give you DOUBLE you money back! |
| 14:38.07 | [TK]D-Fender | goes to sell some fridges to the eskimos |
| 14:38.30 | Chainsaw | Ah yes, the food warmer. |
| 14:38.36 | Katty | watches the eskimos use fridge as a sled. |
| 14:38.56 | [TK]D-Fender | Whee!! |
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| 14:56.16 | Naikrovek | prepares to be audited. |
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| 15:22.47 | wcselby | o/ |
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| 15:26.43 | Poincare | my provider offers me an iax2 account on 2 servers as failover |
| 15:27.09 | Poincare | is there a provisioning method in iax.conf to set 2 hosts for an iax2 account? |
| 15:27.15 | [TK]D-Fender | Poincare: No. |
| 15:27.33 | [TK]D-Fender | Poincare: You need to setup 2 peers and probably 2 regiSter's as well |
| 15:27.34 | Poincare | so I have to fix it in the dialplan? |
| 15:27.47 | [TK]D-Fender | Poincare: Yes |
| 15:28.09 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 15:28.20 | Poincare | ok, thats clear |
| 15:28.33 | Poincare | might be an interesting feature though |
| 15:29.37 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 15:32.07 | wcselby | ~sipnat |
| 15:32.08 | infobot | [~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 |
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| 15:35.24 | p3nguin | The way I configured my failover was to configure a sub account and set the failover to route to that sub account. Then on a server on another network, I configured asterisk to answer and pick up voicemail. The way I see it, if my failover is ever used, my primary network with my main * box and my phones will not be able to access the failover system to make/receive calls anyway, so just set up voicemail over there. |
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| 16:14.44 | wcselby | http://www.voip-info.org/wiki/view/How+to+add+information+to+this+wiki <--- I think someone failed to read the rules... |
| 16:18.21 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:19.08 | raden_work | morning Katty |
| 16:19.12 | raden_work | Naikrovek, morning bro |
| 16:19.21 | Naikrovek | hola |
| 16:22.04 | *** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62) |
| 16:23.49 | p3nguin | wcselby: Talking about the dragonsuite spam? |
| 16:23.57 | wcselby | p3nguin - I am |
| 16:24.15 | p3nguin | Maybe I can delete it. |
| 16:24.36 | wcselby | I could delete and restore it to the previous version |
| 16:24.48 | wcselby | I just figured the spamblock thing would kick in soon and do that |
| 16:25.00 | p3nguin | When was it edited? |
| 16:25.41 | wcselby | Wed 21 of Oct, 2009 [11:09] |
| 16:25.50 | wcselby | so like, 15 minutes ago? |
| 16:25.56 | p3nguin | Oh, so it's real fresh. |
| 16:26.01 | p3nguin | I figured it was old or something. |
| 16:26.26 | p3nguin | Is there even a spam bot on that wiki? |
| 16:26.55 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:33.09 | wcselby | think so |
| 16:33.18 | wcselby | the last few updates on that page were from the user "spamblock" |
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| 16:51.14 | *** join/#asterisk [jmc] (n=John@93-45-226-188.ip104.fastwebnet.it) |
| 16:53.09 | ryduh | p3nguin: Other people had the same problem as I did -> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg230946.html |
| 16:55.35 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 16:56.28 | [TK]D-Fender | ryduh: "type=user" |
| 16:58.38 | *** join/#asterisk Mousey (n=wtfisme@sea01-v500-nat.marchex.com) |
| 16:58.41 | [jmc] | hi guys |
| 16:58.41 | Mousey | waves |
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| 17:04.14 | [jmc] | has anyone tried SafiWorkshop? do you believe it's good? |
| 17:04.15 | [jmc] | http://www.safisystems.com/index.cfm?pageMode=productdetails&product_id=8 |
| 17:07.03 | *** join/#asterisk jantypas (n=jantypas@134.sub-75-210-97.myvzw.com) |
| 17:09.22 | ryduh | The problem is I have two SIP channels. I tried user. * chooses the last SIP channel for incoming calls based on IP |
| 17:09.23 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 17:11.15 | [TK]D-Fender | user = auth by username |
| 17:13.46 | ManxPower-work | ryduh: set allowguest=no in sip.conf[general] |
| 17:13.52 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca) |
| 17:17.12 | angryuser | [jmc], the problem is that if you relay on that stuff, if something goes wrong it will be difficult to find the problem |
| 17:18.19 | *** join/#asterisk pirulo (n=andres@65.102.99.5) |
| 17:18.56 | [jmc] | angryuser, yes, that's why I'm not going to use it |
| 17:20.17 | [jmc] | but I was curious to listen to your opinion |
| 17:21.33 | Mousey | its my bot, and it can cry if it wants to. |
| 17:21.37 | Mousey | oops, wrong window |
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| 17:34.10 | dandate2 | anyway to increase gain? |
| 17:34.10 | Gugge | Anyone know why Expire would show -1 when i register an iax peer on my asterisk 1.4.21? http://pastebin.com/m7e113018 |
| 17:34.23 | Gugge | Im using realtime to store the iax peers |
| 17:35.11 | dandate2 | i dont understand how voicemail gain can be increased but not live call? |
| 17:35.37 | dandate2 | im half deaf i cant hear for shit heh |
| 17:36.36 | Qwell | dandate2: 1.6 added a VOLUME() function |
| 17:36.37 | Gugge | you should be able to fix that in your phone. :) |
| 17:37.19 | Qwell | of course, depending on your phone, it may be doing AGC, which would just turn it back down on you |
| 17:39.59 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 17:41.42 | ManxPower-work | dandate2: Because with a live call the audio may not even be going thru Asterisk. |
| 17:45.28 | ryduh | If I have two SIP channels on the same server/IP, how can I tell Dial which one I want to use? |
| 17:46.02 | ManxPower-work | you don't dial by IP, you dial by sip.conf entry. |
| 17:46.17 | Gugge | and if you dial by ip you can Dial(extension@ip) .... |
| 17:46.20 | ManxPower-work | They are "peers" not "channels" |
| 17:46.31 | ManxPower-work | Gugge: it's a bad idea to dial by IP |
| 17:46.41 | Gugge | Sure, but it can be done. :) |
| 17:49.13 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:52.29 | p3nguin | ryduh: Dialing out is easy; making the ITSP choose which peer it uses for incoming calls when both peers on at the same IP address isn't so simple (so it seems). |
| 17:52.59 | *** part/#asterisk DrZeus (n=chatzill@201.226.170.106) |
| 17:53.34 | [TK]D-Fender | ryduh: they have different names. You have to pick the one you want to dial ot of already. |
| 17:53.46 | ryduh | I ended up just disabling the wrong context * chose. |
| 17:53.52 | p3nguin | ryduh: The Dial() just needs to have the context that you want to use to dial out: Dial(SIP/8043322334@voipms-outbound) |
| 17:54.00 | [TK]D-Fender | Ry* doesn't choose which peer to dial OUT of. YOU do. |
| 17:54.11 | [TK]D-Fender | ryduh: * doesn't choose which peer to dial OUT of. YOU do. |
| 17:54.21 | ryduh | Thanks for the clarification on Dial. I think I got that working |
| 17:54.52 | ryduh | wait, so you can use peer OR context? |
| 17:55.34 | [TK]D-Fender | ryduh: No. You dial out a peer |
| 17:55.43 | [TK]D-Fender | RyA context is a section of your dialplan |
| 17:55.48 | [TK]D-Fender | ryduh: A context is a section of your dialplan |
| 17:55.54 | [TK]D-Fender | dangit... |
| 17:56.17 | p3nguin | What's with the tabfail all the time? |
| 17:56.35 | ryduh | [TK]D-Fender: Right. That's what I have. p3nguin was just talking about contexts that's why I asked |
| 17:56.47 | ariel_ | it's called speed typing and not looking at what your figures are doing... |
| 17:56.53 | p3nguin | ryduh: Contexts are those sections where they are labelled like [this]. |
| 17:56.59 | [TK]D-Fender | p3nguin: Whats with the wrong advice all of the time? :p |
| 17:57.14 | [TK]D-Fender | pummels p3nguin with a glossary |
| 17:57.17 | p3nguin | ariel_: You shouldn't be looking at your fingers, anyway. Look at the screen when you type. |
| 17:57.17 | ryduh | right so a peer is defined in a context |
| 17:57.37 | [TK]D-Fender | ryduh: No... a peer POINTS to a context.... |
| 17:58.17 | [TK]D-Fender | ryduh: For where incoming calls landing on it do. |
| 17:58.26 | [TK]D-Fender | ryduh: Which would be an inbound call, not an outbound. |
| 17:58.41 | dandate2 | im using asterisk 1.4 so i guess volume() is out, is it recommended to use AGC then? |
| 17:59.06 | [TK]D-Fender | dandate2: Operate <- |
| 17:59.38 | [TK]D-Fender | dandate2: AGC has nothing to do with making just your calls louder. |
| 18:00.00 | ryduh | [TK]D-Fender: in my sip.conf there is a general context, a [voipms] context and a [voipms-slo] context. What is the relationship between peers and the contexts I define in sip.conf? |
| 18:00.13 | dandate2 | i was just asking about performance i dontu se the phone that much anyway but it is really quiet...i run a call center thats why |
| 18:00.37 | [TK]D-Fender | dandate2: So upgrade, adjust your zaptel/DAHDI configs if thats what they come in on, adjust your phone, or replace your phone. |
| 18:00.49 | dandate2 | ahh cable modem no zaptel heh |
| 18:00.52 | [TK]D-Fender | datOr spend 415 on a handset amplifier |
| 18:01.10 | [TK]D-Fender | dandate2: I jsut gave you 5 options. Pick one |
| 18:01.20 | dandate2 | i got a plantronics amplifier but that made it sound horrrrrible heh |
| 18:01.39 | [TK]D-Fender | ryduh: [voipms] <- this is a PEER, not a CONTEXT |
| 18:01.40 | dandate2 | <PROTECTED> |
| 18:01.58 | [TK]D-Fender | dandate2: Then do something else |
| 18:02.24 | ryduh | [TK]D-Fender: That's what I thought. Again got confused by this line ;) p3nguin: ryduh: Contexts are those sections where they are labelled like [this]. |
| 18:02.44 | [TK]D-Fender | pummels p3nguin with a glossary some more.... |
| 18:03.17 | p3nguin | What's to be confused about? If you have a section that says [my-context] followed by relevant information, that's a context. |
| 18:03.47 | ryduh | p3nguin: in sip.conf is it still a context? |
| 18:03.52 | p3nguin | yes |
| 18:04.23 | p3nguin | user context, peer context, sip context... whatever. |
| 18:04.24 | dandate2 | the network settings in eyebeam/x-lite ask if im on LAN/dsl/dialup etc., does setting this improperly actualy affect anything? lol |
| 18:04.28 | ryduh | p3nguin: in sip.conf how do I define a peer then? with [name] ? |
| 18:04.41 | p3nguin | type=peer within the context makes it a peer. |
| 18:05.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:05.19 | ryduh | p3nguin: ok thanks |
| 18:05.32 | ryduh | p3nguin: sorry for being frustrating |
| 18:05.39 | p3nguin | type=user within the context makes it a... what? That's right, a user. |
| 18:08.24 | *** join/#asterisk Badrobot- (n=badrobot@76.173.229.89) |
| 18:13.06 | *** join/#asterisk sircolin (n=sircolin@83.216.68.241) |
| 18:22.29 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp117-106.adsl.forthnet.gr) |
| 18:22.42 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 18:29.01 | *** join/#asterisk mutex (n=mutex@67-23-13-54.static.slicehost.net) |
| 18:29.33 | mutex | is there a way I can pull the peering status out of asterisk ? |
| 18:29.40 | mutex | I don't mind doing it via bash or perl |
| 18:29.48 | mutex | just not clear where I should look |
| 18:30.02 | Naikrovek | peering status? you mean status of trunks and/or extensiosn? |
| 18:30.07 | mutex | yes |
| 18:30.09 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 18:30.12 | Naikrovek | asterisk -r |
| 18:30.12 | mutex | sip show peers |
| 18:30.13 | Naikrovek | then |
| 18:30.16 | Naikrovek | sip show peers |
| 18:30.19 | mutex | status column |
| 18:30.19 | Naikrovek | iax2 show peers |
| 18:30.35 | mutex | well I can get it out of the command line fine |
| 18:30.41 | Naikrovek | oh i see |
| 18:30.45 | mutex | but I want to get it out programmaicly |
| 18:30.46 | Naikrovek | yeah you can automate that in perl |
| 18:30.48 | *** join/#asterisk scalex000 (n=chatzill@190.80.227.255) |
| 18:30.53 | Naikrovek | i've done that |
| 18:31.11 | scalex000 | hello |
| 18:31.45 | scalex000 | I need help to setup callerId |
| 18:31.46 | mutex | is there an API or do I need to just grab it via 'asterisk -r' |
| 18:31.59 | Naikrovek | mutex: i just do it via asterisk -r & perl magic |
| 18:32.06 | Naikrovek | and some grep i think |
| 18:32.08 | [TK]D-Fender | mutex: Parse CLI or use AMI |
| 18:32.11 | mutex | easy enough |
| 18:32.23 | Joel | mutex, Asterisk::Manager |
| 18:32.52 | mutex | Joel: ah ha, you speak my language |
| 18:33.42 | Naikrovek | yeah that's handy |
| 18:33.45 | Naikrovek | didn't know that |
| 18:33.49 | *** join/#asterisk Knightfal (n=Josh@mailer.1callres.com) |
| 18:33.57 | Joel | I've done perl development with asterisk for almost four years now. |
| 18:34.39 | Naikrovek | nice |
| 18:34.41 | Naikrovek | what have you written |
| 18:34.48 | Naikrovek | 4 years is a long time to make neat stuff |
| 18:36.12 | mutex | some people don't even need 4 years |
| 18:36.38 | Naikrovek | well i'm guessing mutiple neat things |
| 18:36.40 | Joel | Naikrovek, I did considerable work on pbxtra/trixbox pro |
| 18:36.55 | Joel | and I'm now with the Switchvox gang doing whatever they ask |
| 18:37.09 | Naikrovek | neat |
| 18:39.36 | Joel | there was a company inbetween where I had to work on freepbx, but it was a giant joke |
| 18:39.51 | Joel | any company that runs anything based on freepbx I feel extremely sorry for :\ |
| 18:40.55 | Naikrovek | odd that you're bashing freepbx but not trixbox |
| 18:41.01 | Naikrovek | usually it's reversed |
| 18:41.12 | Joel | I've never used ce, I can't bash it. |
| 18:41.35 | ariel_ | Freepbx is running at many locations and company's, doing just fine. |
| 18:41.46 | Naikrovek | ah |
| 18:41.52 | Joel | ariel_, ever done any serious coding for freepbx? |
| 18:41.52 | Naikrovek | yeah i use freepbx it's fine |
| 18:41.58 | Joel | it's a mess. |
| 18:42.00 | ariel_ | Joel: yes |
| 18:42.03 | Joel | written by developers for developers. |
| 18:42.29 | Joel | and the horizontal scrolling in v3 shows just how badly they missed the mark. they took the number one thing you don't do in web design and implemented it |
| 18:43.47 | ariel_ | version 3 is in development and last I knew based on top of FreeSwitch not asterisk |
| 18:43.58 | Naikrovek | developing on freeswitch |
| 18:44.10 | Joel | and? |
| 18:44.12 | Naikrovek | from what i understand it'll be usable on asterisk as well |
| 18:44.16 | Joel | that changes the quality of the product how? |
| 18:44.27 | Naikrovek | Joel: well fpbx3 is a new codebase entirely |
| 18:44.31 | Joel | I built the developer iso for fpbx3. |
| 18:44.36 | Joel | so yes, I know it's a new codebase |
| 18:44.38 | Joel | and it still sucks. |
| 18:44.40 | Naikrovek | k |
| 18:44.41 | Naikrovek | fine |
| 18:44.48 | Naikrovek | i don't know anything about the code so i can't argue that |
| 18:44.50 | Naikrovek | but it works |
| 18:45.07 | Qwell | 2.x works |
| 18:45.11 | Qwell | 1.x worked. |
| 18:45.13 | Naikrovek | yeah |
| 18:45.16 | Joel | 3 doesn't work last I tried. |
| 18:45.23 | Joel | but it's been a while |
| 18:45.24 | Naikrovek | maybe they're not perfect but they work |
| 18:45.44 | ariel_ | Joel if you bashing what your working it's an issue with you. 2.5.X works fine and it can do allot and many biz are using it. And will continue to use it. |
| 18:46.01 | Knightfal | Does anyone have any information on the following errors. We started getting these errors in the log file yesterday. |
| 18:46.03 | Knightfal | ERROR[2912] utils.c: write() returned error: Broken pipe |
| 18:46.03 | Knightfal | NOTICE[15679] utils.c: Timed out trying to write |
| 18:46.18 | jblack | [TK]D-Fender: Ping |
| 18:46.23 | mutex | also: dahdi 2.2.0 doesn't compile on 2.6.31.4 :-( |
| 18:46.26 | Joel | ariel_, huh? |
| 18:46.28 | [TK]D-Fender | jblack: Pong |
| 18:46.37 | Joel | ariel_, sorry, I don't understand your first sentence. |
| 18:46.51 | jblack | I have a really, really badly behaving sip client. Got some time to give some suggestions? |
| 18:47.38 | jblack | by really bad, I mean the client reports it's address as 127.0.0.1, it ignores sip OPTION (thus breaking qualify), etc. |
| 18:48.01 | [TK]D-Fender | jblack: "nat=yes" |
| 18:48.16 | jblack | Have that, and "can" route calls. But |
| 18:48.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:48.48 | jblack | well, the client also silently goes away (i.e. doesn't seem to deregister, just stops talking), so * gets blocked up trying to reach it if it's not there. |
| 18:49.45 | jblack | qualify=yes normally fixes that, but because of no OPTION, that's not a choice. |
| 18:50.40 | scalex000 | hi, does chan_dahdi have privacy command |
| 18:51.59 | ManxPower-work | you woShould set nat=yes for the idiot peer that says it's IP is 127.0.0.1 |
| 18:52.11 | jblack | Like I said, I already have nat=yes. |
| 18:52.17 | ManxPower-work | nat=yes (among other things) totally ignores the IP inside the packet and uses the IP in the packet header. |
| 18:52.29 | jblack | Oddly enough, in sip show peers, it reports Nat N |
| 18:52.41 | [TK]D-Fender | jblack: nat=yes will fix the qualify target |
| 18:53.14 | [TK]D-Fender | scalex000: What kind of channel? |
| 18:53.32 | jblack | No, nat=yes does not. |
| 18:53.35 | scalex000 | FXO |
| 18:53.42 | scalex000 | TK,FXO |
| 18:53.43 | [TK]D-Fender | scalex000: No such thing then |
| 18:53.49 | scalex000 | Ok |
| 18:53.51 | jblack | Like I said, I have nat=yes set. I really do. have had for a week. |
| 18:53.55 | wcselby | Knightfal - I've gotten that error when an AGI script that writes to disk is called multiple times at once (i.e mutltiple calls are coming in at the same time that the AGI is trying to do some kind of disk writing activity for) |
| 18:54.16 | jblack | crosses his heart. |
| 18:54.39 | jblack | for the curious, this is a badly broken sip client on my android phone. |
| 18:54.47 | scalex000 | TK, generally what kind of signalling FXO use, this depend from type of company |
| 18:55.36 | [TK]D-Fender | scaFXO = signalling |
| 18:55.56 | [TK]D-Fender | jblack: pastebin SIP debug... |
| 18:56.03 | [TK]D-Fender | scalex000: FXO = signalling |
| 18:56.43 | scalex000 | TK. yes |
| 18:56.53 | *** join/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 18:57.06 | jblack | fine. doing a pastebin bomb |
| 18:58.38 | jblack | bleh. |
| 18:58.48 | jblack | I just gotta call. I gotta go. THanks anyways. |
| 18:58.55 | wcselby | any good voicemail transcription apps out there for asterisk? |
| 19:00.09 | scalex000 | Tk, generally phone provider use FXO_KS |
| 19:00.50 | *** join/#asterisk DavidR2008 (n=chatzill@208.34.240.2) |
| 19:01.36 | Knightfal | wcselby: We are not using any AGI. We do have a remote client using AMI to pull queuestatus for realtime routing purposes. |
| 19:01.55 | wcselby | Knightfal - are you writing to disk at all when this happens? |
| 19:03.15 | Joel | ariel_, if you want to have a serious conversation about this I'm more then happy to. Feel free to message me. |
| 19:03.25 | *** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com) |
| 19:03.36 | Knightfal | They pull the data from AMI every 6 seconds to get queuestatus so ya alot of stuff is writing to disk. queue_log FULL ect |
| 19:03.51 | Knightfal | ^ wcselby |
| 19:04.29 | wcselby | can you correlate the error messages with the times the AMI is pulilng data? |
| 19:04.57 | *** join/#asterisk buttons840 (n=buttons8@63.230.20.246) |
| 19:05.26 | Knightfal | wcselby:Its difficult because they pull so often and we are constantly at about 100 calls |
| 19:05.46 | *** join/#asterisk QaDeS (n=mklaus@p4FC72958.dip0.t-ipconnect.de) |
| 19:06.31 | wcselby | what I did notice was that those error messages didn't appear to break anything, they just were annoying. but that was my experience. yours may vary |
| 19:07.36 | *** join/#asterisk philippel (n=p_lindhe@pool-98-111-69-201.sttlwa.fios.verizon.net) |
| 19:07.38 | Knightfal | wcselby: The main issues was that calls were queueing and then not being delivered to agents for extended periods of time. We noticed those messages in the logs during the problem |
| 19:08.17 | buttons840 | Using the AMI, I originate a call (using originate) and through the progression of the call, there are multiple hangup events. I'm having trouble finding any predictable patern with these, or how I can determine which is a legitimate hangup? Any suggestions? |
| 19:08.19 | DavidR2008 | Can someone take a look at this pastebin http://asterisk.pastebin.com/d3bfddfdf and tell me why asterisk is picking up three DTMF digits in the Bad DTMF Read section?, I suspect it has something to do with the fact that the duration on all the packets is zero, but I don't know enough about RFC2833 to know for sure |
| 19:09.30 | *** join/#asterisk lucasb (n=bussey@24.67.33.55) |
| 19:09.47 | wcselby | Knightfal - what else is going on with the system when these errors occur? Sounds like it may be a load issue on the box, or a disk activity load issue (too many things hitting the disk all at once)....? |
| 19:10.07 | wcselby | simple test would be to move them off the main box onto their own dedicated box and see if the issues persist / follow |
| 19:10.15 | wcselby | well, I say simple, but you know what I mean |
| 19:11.33 | buttons840 | i expect to receive hangup cause 16 (normal clearing) and do most of the time, i receive 2 or 3 normal clearings per call it appears, but then i get a cause 0 hangup (unknown cause) and i'm not sure why. I'm trying to decide if i should ignore these hangups in my script? |
| 19:14.15 | Knightfal | wcselby: perhaps its recording all of those calls at the same time! :( |
| 19:14.37 | Knightfal | I need to get an external monitoring system I think. |
| 19:18.31 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 19:20.17 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:23.17 | *** part/#asterisk McBoingbo (n=mcboingb@mail.hrsg.ca) |
| 19:24.32 | superbeef | i'm looking for a list of IAX hangup cause codes |
| 19:26.23 | scalex000 | Hey, How to use CallerId on with Dahdi I do not understand very well |
| 19:29.10 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:31.14 | DavidR2008 | scalex000, what do you not understand? |
| 19:33.05 | scalex000 | davidR2008, do you speak spanish?, I do not understand, where I need to use callerId |
| 19:33.49 | DavidR2008 | no habla espanol; sorry do you want to try to talk about this in english? |
| 19:35.41 | scalex000 | davidR2008, Ok |
| 19:36.39 | DavidR2008 | do you want to set callerid for a call leaving your asterisk server, or get callerid on an in-comming call? |
| 19:37.27 | scalex000 | davidR2008, Incoming Call, because when call arrive I only see asterisk |
| 19:37.28 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 19:37.37 | *** join/#asterisk endeavormac (n=endeavor@unaffiliated/endeavormac) |
| 19:37.54 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 19:38.01 | endeavormac | i am having issues getting the gui to present me with a login prompt |
| 19:38.22 | DavidR2008 | is this an analog or digital POTS line? |
| 19:38.39 | DavidR2008 | ~freepbx |
| 19:38.40 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:38.53 | DavidR2008 | endeavormac ^^^^^^^^^^^^^^^^^ |
| 19:39.21 | endeavormac | asterisk... |
| 19:39.21 | endeavormac | on ubuntu |
| 19:39.29 | DavidR2008 | gui? |
| 19:39.33 | endeavormac | the web interface |
| 19:39.50 | DavidR2008 | ~gui |
| 19:39.51 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
| 19:40.16 | scalex000 | DavidR2008, Analog Lines |
| 19:40.37 | DavidR2008 | are you set up to receive callerid from the phone company? |
| 19:41.09 | DavidR2008 | that is something that has to be sent by the phone company |
| 19:42.15 | scalex000 | DavidR2008, yes, I think was setup, I will ask to them but, when dahdi files was setup from automatic script not add the command callerid |
| 19:42.37 | scalex000 | DavidR2008, so I not sure where to put on FXO or FXS |
| 19:42.51 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 19:44.43 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 19:45.04 | raden_work | anyone know where i can get a recorder to plug inline with my cellphones 2.5 mm jack |
| 19:45.12 | superbeef | so what are the chances that an implementation of speex from 2003 doesnt work well with a modern speex install |
| 19:45.26 | DavidR2008 | FXO lines would connect to the phone company so you would put it there, FXS would connect to your analog phones, faxes or modems (not a good idea) here's more info about the difference between FXO and FXS http://www.3cx.com/PBX/FXS-FXO.html |
| 19:47.00 | Knightfal | ~realtime |
| 19:47.01 | infobot | i guess realtime is a feature of Asterisk starting with 1.2 which allows you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
| 19:47.38 | Knightfal | wcselby: Thanks for the starting point |
| 19:48.00 | wcselby | Knightfal - np, glad I could try to help :) |
| 19:53.57 | scalex000 | DavidR2008, Ok |
| 19:53.57 | *** join/#asterisk jamesh1 (n=jhenders@66.111.111.254) |
| 19:58.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:58.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:05.01 | Katty | :< |
| 20:05.07 | Katty | i seem to be feeling annoyed and spastic. |
| 20:08.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:08.44 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 20:08.51 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 20:09.06 | wcselby | Katty - sorry to hear that |
| 20:09.35 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:09.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:11.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:11.40 | scalex000 | DavidR2008, I don't know why on my phone appear from asterisk |
| 20:12.23 | *** part/#asterisk _trine (n=psybnc_a@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 20:12.36 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:15.52 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 20:16.01 | [TK]D-Fender | raden_work: Radio Shack |
| 20:16.43 | raden_work | should just be able to T a 2.5 jack into mic in on recorder im thinking |
| 20:16.58 | raden_work | would rather router cell through asterisk and record but to much work at moment |
| 20:16.59 | geneticx | hello everyone.. anyone can please shed some light.. I'm trying to insert a pause in my dialplan so let's say 90 is dial first it waits and then dials a number |
| 20:17.04 | raden_work | freakin Crazy X's |
| 20:17.25 | geneticx | anyone got any ideas on how to insert pauses into asterisk dialplans? |
| 20:17.32 | raden_work | i dont get it ? |
| 20:17.58 | [TK]D-Fender | geneticx: Wait() |
| 20:18.18 | [TK]D-Fender | geneticx: But I'm quite sure this isn't what you're looking for |
| 20:19.22 | raden_work | the question is what are we waiting for |
| 20:20.55 | scalex000 | hey guy, I have IAX between to pbx why not register or connect, what can I do? |
| 20:22.30 | mog | scalex000, you seeing traffic |
| 20:23.21 | [TK]D-Fender | scalex000: I'm sorry, you forgot to phrase that in the form of a questions </trebek> |
| 20:24.27 | scalex000 | mog, I see registration, but when I try to see peers, not show me online |
| 20:24.44 | ryduh | What's the main difference between Macro and Gosub ? |
| 20:25.29 | leifmadsen | ryduh: how it was implemented -- GoSub() is like a fancy Goto() with memory |
| 20:25.41 | [TK]D-Fender | ryduh: Also depends on what * version <- |
| 20:25.50 | leifmadsen | Macro() can only be nest so many times before the stack size is too big -- GoSub() has practically unlimited recursions |
| 20:26.15 | geneticx | [TK]D-Fender: I'm trying to call a local number through another PBX (Nortel BCM400) and the guy in the other end tell me I should dial 9 to access a line and dial the number, how do I program that into my dialplan? should I dial one number first then wait() and then dial the second? |
| 20:26.33 | ryduh | [TK]D-Fender: I'm running 1.4.26 |
| 20:26.40 | [TK]D-Fender | geneticx: No, you add the 9 tot he number you DIAL. |
| 20:26.58 | ryduh | So with gosub, you don't have the same extension problem you do with Macro right? |
| 20:27.03 | [TK]D-Fender | geneticx: You are not adding steps to your dialplan, you are changing the number you dial in the Dial() command you are calling |
| 20:27.26 | [TK]D-Fender | geneticx: You want a 9 in there... put a 9 in there! |
| 20:27.35 | [TK]D-Fender | checkout time, bbiab |
| 20:28.04 | geneticx | [TK]D-Fender: I tried that already and it calls an extension in the BCM but not the number intended.. |
| 20:28.17 | scalex000 | tk, how to reload iax module |
| 20:29.03 | scalex000 | tk, I don't know why I can not use it if I do not change anything |
| 20:32.59 | DavidR2008 | geneticx, use a comma to put a pause in the dialed number (i.e. 9,1234)? |
| 20:33.15 | geneticx | DavidR2008: ok let me try |
| 20:33.27 | Qwell | what? no. |
| 20:33.47 | Qwell | w would be a 250(or 500?)ms wait |
| 20:34.00 | leifmadsen | I think it is 250ms |
| 20:34.01 | Qwell | , would change things rather significantly |
| 20:34.16 | DavidR2008 | sry, brain fart |
| 20:34.27 | DavidR2008 | Qwell is very correct! |
| 20:34.27 | geneticx | humm.. |
| 20:35.44 | DavidR2008 | scalex000, iax2 reload |
| 20:36.55 | geneticx | =( |
| 20:37.51 | *** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 20:39.32 | raden_work | anyone recommend a high quality tape recorder that doesn't sound all distorted ? |
| 20:40.15 | DavidR2008 | why not use a digital recorder? |
| 20:41.13 | scalex000 | DavidR2008, not work |
| 20:41.51 | DavidR2008 | pastebin cli output with verbose level 3 or higher when you try to do a reload |
| 20:43.01 | *** join/#asterisk timeshell_atwork (n=chatzill@142.46.193.194) |
| 20:43.13 | *** join/#asterisk Trepinon (n=nemus@lbuttars.dsl.xmission.com) |
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| 20:50.47 | Katty | sighs. |
| 20:53.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:55.33 | *** join/#asterisk scalex000 (n=chatzill@190.80.227.255) |
| 20:55.56 | *** join/#asterisk Ta^3 (n=tacvbo@189.146.178.177) |
| 20:56.17 | scalex000 | what is this problem: WARNING[2893]: app_dial.c:1499 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 20:56.29 | *** join/#asterisk plasmid (n=noway@98.235.23.118) |
| 20:57.05 | dustybin | hugs Katty |
| 20:57.35 | plasmid | new user for asterisk. When i create a trunk do I select a SIP trunk or Zap Trunk (DAHDI compatibility mode) when I already have a TDM400 installed? Home use only. |
| 20:58.26 | Katty | hugs dustybin, sighs some more. |
| 20:58.28 | Katty | grumbles a bit |
| 20:58.40 | beek | plasmid: You want to make a call out your POTS line? |
| 20:59.01 | plasmid | beek, I do not have a POTS line. |
| 20:59.09 | plasmid | I haven't used one in 6 years. |
| 20:59.22 | beek | plasmid: Then the trunk will be SIP. |
| 20:59.31 | plasmid | Got it. :-) Ty. |
| 20:59.33 | beek | What are you using the TDM400 used for? |
| 20:59.47 | beek | FXS? |
| 21:00.05 | plasmid | it was a gift. There are two FXS ports ..one for my wife and me and the other one for the kids. |
| 21:00.16 | beek | Okay... |
| 21:00.21 | p3nguin | katty: What's the problem today? |
| 21:00.32 | *** join/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 21:00.52 | Katty | p3nguin: oh you know... |
| 21:00.55 | Katty | p3nguin: sales reps. |
| 21:01.12 | p3nguin | katty: Bugging you to fix their stuff? |
| 21:01.21 | Katty | p3nguin: no, trying to sell stuff that makes me nervous. |
| 21:01.21 | *** join/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 21:01.51 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 21:02.32 | p3nguin | katty: I figured you would have a standard procedure for dealing with people that bother you with such things. |
| 21:02.46 | Katty | unfortunately i'm just the little person. |
| 21:03.00 | p3nguin | gets confused |
| 21:03.03 | Katty | i just install it, and fix it, and maintain it. |
| 21:03.15 | Katty | ^- sarcasm. |
| 21:03.17 | Katty | ^- truth. |
| 21:03.46 | beek | Katty -- Ask [TK] to borrow his Clue-by-four (TM) |
| 21:03.54 | Katty | no it's okay |
| 21:03.56 | p3nguin | What kind of stuff makes you nervous? |
| 21:03.59 | Katty | i just need to think positive for awhile. |
| 21:04.00 | beek | Sorry -- he has the Clue Bat (TM) |
| 21:04.11 | Katty | p3nguin: they're selling someone a server with a crappy backup solution. |
| 21:04.21 | p3nguin | ohhhhhhhhh |
| 21:04.21 | Katty | p3nguin: which is giving me severe heart burn. |
| 21:04.27 | beek | What, a travan tape drive? ;-) |
| 21:04.32 | ariel_ | sends a chocolate bar to help Katty |
| 21:04.33 | Katty | p3nguin: specifically, they're selling them software i have never used before...and making promises about it. |
| 21:04.40 | *** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com) |
| 21:04.43 | Katty | p3nguin: promises they can't make. |
| 21:04.49 | Katty | p3nguin: or at least shouldn't. |
| 21:04.57 | Katty | ariel_: thanks. |
| 21:04.58 | beek | Sure they can -- it's just up to you to keep the promises! |
| 21:05.03 | Katty | and now i don't want to think about it anymore. |
| 21:05.07 | Katty | i want to think about happy bunnys. |
| 21:05.12 | Katty | and maybe dolphins. |
| 21:05.42 | Katty | pretty fall leaves, fireplaces, grilled hamburgers, and strawberries. warm socks, fluffy pillows, and hugs. |
| 21:06.27 | ariel_ | rofl, all above stated things are just great. |
| 21:06.57 | ariel_ | but we don't get pretty fall leaves here. |
| 21:07.01 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
| 21:09.23 | scalex000 | hey there how to setup callerid on dahdi chan analog lines, how to remove (asterisk) |
| 21:14.30 | ryduh | Is there a way to catch invalid extensions when running a Gotosub? [Oct 21 14:11:16] WARNING[1329]: pbx.c:1833 pbx_extension_helper: No application 'Gotosub' for extension (record, 9, 1) |
| 21:17.23 | leifmadsen | 'i' extension? |
| 21:17.32 | leifmadsen | can't remember if that works in that situation |
| 21:18.07 | p3nguin | i in invalid during backgrounded playback (while waiting for someone to press a key). |
| 21:18.08 | thehar | waves to all |
| 21:18.08 | ryduh | I have an i extension in the context I'm calling the GotoSub from and also in the GotoSub context |
| 21:18.22 | leifmadsen | GotoSub? |
| 21:18.24 | leifmadsen | that doesn't exist |
| 21:18.26 | leifmadsen | GoSub() |
| 21:18.29 | p3nguin | gosub |
| 21:18.50 | *** join/#asterisk Tim_Toady (n=moi@188.4.44.234) |
| 21:20.20 | scalex000 | what its mean this : WARNING[9213]: chan_iax2.c:10581 build_user: Unable to support trunking on user 'churchill' without DAHDI timing |
| 21:20.30 | [jmc] | ryduh, I don't think so |
| 21:20.33 | ryduh | lol. whoops. now it runs my i extension the gosub context. :) |
| 21:20.41 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:21.04 | [jmc] | scalex000, afaik you need a timing source |
| 21:21.07 | [jmc] | even ztdummy |
| 21:22.05 | scalex000 | jmc, I do not set ztdummy on this |
| 21:22.51 | [jmc] | ? |
| 21:24.25 | ariel_ | it's time to go home... night folks... |
| 21:24.29 | [jmc] | night |
| 21:24.30 | [jmc] | :) |
| 21:24.34 | [jmc] | a quick question folks |
| 21:24.56 | beek | waits with bated breath |
| 21:25.58 | [jmc] | can someone help me distinguish between FXO and FXS? |
| 21:26.19 | [jmc] | I often confuse the two |
| 21:27.03 | [jmc] | FXS is the interface where the analog phone (as an example) is connected? |
| 21:27.27 | [jmc] | and FXO the one that gets the signal from the phone line? |
| 21:27.47 | [jmc] | or the opposite? :D |
| 21:29.45 | beek | [jmc]: FXS points towards the STATION... FXO points towards the central OFFICE |
| 21:30.33 | [jmc] | ok, what you mean with "station"? |
| 21:30.50 | beek | Okay... FXS points toward the deskSet |
| 21:31.07 | [jmc] | (please forgive this, I know this is a stupid question, but I'm not a native English speaker and I might confuse the terms) |
| 21:31.23 | beek | Not at all... it is confusing. |
| 21:32.02 | [jmc] | :) |
| 21:32.12 | [jmc] | so, I was right then? |
| 21:32.22 | beek | [jmc]: yes |
| 21:32.31 | [jmc] | great ;) |
| 21:32.34 | [jmc] | thanks beek |
| 21:32.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:38.28 | *** join/#asterisk coppice (n=chatzill@86.132.26.86) |
| 21:40.07 | DavidR2008 | is there a list of PRI switch types that dahdi support? i.e. a list of entries that are valid for the switchtype= line |
| 21:40.10 | DavidR2008 | *supports |
| 21:40.19 | *** join/#asterisk arpu (n=arpu@62.178.159.144) |
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| 21:43.08 | DavidR2008 | nevermind, it's in the sample chan_dadhi.conf file, thanks |
| 21:48.20 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
| 21:49.02 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
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| 21:49.55 | *** part/#asterisk ZX81 (n=Matt_Rid@121.74.230.252) |
| 22:01.24 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:03.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:06.18 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
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| 22:11.33 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:16.04 | [jmc] | ok last question for today :P |
| 22:16.53 | ChannelZ | 42 |
| 22:16.58 | [jmc] | lol |
| 22:17.00 | [jmc] | thanks :P |
| 22:17.06 | [jmc] | no, seriously |
| 22:17.08 | [jmc] | is there a general rule to follow when choosing on which hardware to run an Asterisk server? |
| 22:17.45 | [jmc] | I ask this because I have to write down some general instructions, not preparing a specific configuration |
| 22:17.47 | ChannelZ | don't under-buy? |
| 22:17.56 | [jmc] | so I should be able to tell how to choose properly |
| 22:18.02 | [jmc] | ChannelZ, yes, clearly |
| 22:18.13 | [jmc] | but how can I tell if I'm under-buying? |
| 22:18.21 | [jmc] | or over-buying, even |
| 22:18.55 | ChannelZ | Asterisk is not very intensive except depending on the number of calls running through it and what the phone interface is, a shit-ton of interupts can be happening on the bus |
| 22:19.39 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 22:22.33 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 22:22.44 | ChannelZ | it's hard to come up with any sort "if you have this many calls, get an X-Ghz box" |
| 22:23.25 | Katty | looks in |
| 22:25.27 | [jmc] | ChannelZ, that's exactly my problem |
| 22:25.43 | Katty | i like how this box says all natural organic on it |
| 22:25.55 | Katty | and the side of the box says BHT added to packaging material to preserve freshness |
| 22:25.59 | ChannelZ | You can read random thoughts here: http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning |
| 22:26.07 | [jmc] | I don't exactly know what devices they'll be choosing |
| 22:26.29 | [jmc] | ok, let me see :) |
| 22:26.36 | [jmc] | thanks for helping ChannelZ :) |
| 22:27.24 | *** join/#asterisk voipmonk (n=voipmonk@67.204.45.155) |
| 22:27.38 | [TK]D-Fender | Katty: LIES |
| 22:29.05 | Katty | [TK]D-Fender: hello dear |
| 22:31.40 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 22:32.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:41.03 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 22:41.22 | manxpower | ~answers |
| 22:41.23 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 22:42.23 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
| 23:03.19 | *** join/#asterisk metfan2007 (n=metfan20@201.103.119.20) |
| 23:03.23 | metfan2007 | hi all! |
| 23:04.41 | metfan2007 | is there a way to recreate the extensions.conf dialplan from the asterisk dialplan in memory? |
| 23:04.43 | metfan2007 | :D |
| 23:05.14 | beek | metfan2007: Let me guess... you inadvertently deleted the existing plan? |
| 23:06.19 | russellb | extensions.conf.sample makes a reference to "dialplan save" |
| 23:06.53 | beek | Hello russellb -- I enjoyed the session that you and Kevin did in Phoenix. |
| 23:08.05 | russellb | beek: thanks! |
| 23:11.59 | Katty | so. i'm going to write myself a report. |
| 23:12.00 | leifmadsen | russellb: as did I! |
| 23:12.04 | Katty | and not a TPS report, either. |
| 23:12.12 | leifmadsen | backspaces... |
| 23:12.14 | beek | leifmadsen: Your's was good as well... |
| 23:12.19 | *** join/#asterisk ZX81 (n=Matt_Rid@121.74.230.252) |
| 23:12.21 | Katty | the report is about Vitamins and Minerals. |
| 23:12.22 | leifmadsen | w00t |
| 23:12.41 | Katty | what the recommended daily ammounts are, and a comparrison of what i eat for a week against what i should be eating |
| 23:13.22 | Katty | and then, in conclusion, an outline for 1 week that would be pretty close to what i should be eating. |
| 23:13.32 | Katty | and i'm doing all this because i feel dumb. |
| 23:13.37 | *** part/#asterisk ZX81 (n=Matt_Rid@121.74.230.252) |
| 23:13.38 | russellb | leifmadsen: <3 |
| 23:13.40 | Katty | i have no idea how to really read nutrition labels. |
| 23:13.53 | Katty | well i know how to read calories and ingredient lists. |
| 23:13.55 | russellb | leifmadsen: too bad we made it all up, we didn't actually do any of that stuff |
| 23:13.56 | leifmadsen | russellb: oh! do you happen to have a link handy describing the currently proposed dates of the start and end of releases? |
| 23:14.01 | russellb | oh wait, that was supposed to be a /msg |
| 23:14.02 | leifmadsen | russellb: :D |
| 23:14.05 | leifmadsen | oh snap! |
| 23:14.10 | leifmadsen | russellb: lies ftw |
| 23:14.11 | metfan2007 | beek: yes :) ups! |
| 23:14.32 | russellb | leifmadsen: http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html <--- BAM! |
| 23:14.34 | leifmadsen | Katty: I try to stay away from stuff where i don't understand what the ingredient is |
| 23:14.40 | Katty | leifmadsen: i do too. |
| 23:14.41 | leifmadsen | russellb: hawt! updating roadmap |
| 23:14.54 | beek | metfan2007: Did you see russellb's comment re: dialplan save? |
| 23:15.03 | Katty | leifmadsen: but i'd still like to better understand the things my body needs, and to really have soething to look at to know where i'm deficient or going way overboard. |
| 23:15.11 | Katty | leifmadsen: for all i know, i'm horribly deficient in Vitamin D |
| 23:15.19 | leifmadsen | Katty: makes sense |
| 23:15.27 | Katty | leifmadsen: tho, in all reality, i'm probably over simplyfing it |
| 23:15.36 | Katty | leifmadsen: some people have greater needs than others, some less... |
| 23:15.38 | leifmadsen | I'm probably vitamin deficient since I'm a northener |
| 23:16.31 | Katty | leifmadsen: http://ods.od.nih.gov/Health_Information/Vitamin_and_Mineral_Supplement_Fact_Sheets.aspx <- this is where i'm starting. |
| 23:16.46 | metfan2007 | beel: I'm checking it right now |
| 23:17.09 | beek | In voicemail.conf -- is the script in 'externnotify=pgm' called when the poll is done? I have a legacy PBX that I need to set/clr the MWI. If I listen to the message via email (IMAP storage) my pgm doesn't seem to be triggered. |
| 23:18.01 | manxpower | beek: I believe it is only triggered on a message being left |
| 23:18.19 | leifmadsen | beek: what version of asterisk? |
| 23:18.27 | beek | 1.6.0.15 |
| 23:18.45 | leifmadsen | beek: ah... if you were using 1.6.2 you could use MinivmMWI() to turn it off |
| 23:19.00 | metfan2007 | rusellb: so, if I execute dialplan save it will write extensions.conf again? |
| 23:19.07 | russellb | in theory |
| 23:19.08 | russellb | i've never used it |
| 23:19.12 | russellb | try it and see :-) |
| 23:19.21 | [TK]D-Fender | BACKUP!!!!!!!!!! |
| 23:19.23 | beek | The docs state: "called when a voicemail is left, delivered, or your voicemailbox is checked" and I didn't know if "voicemailbox is checked" included the poll. |
| 23:19.26 | metfan2007 | rusellb: hehehehe, I can guess why... hehehehe LOL! |
| 23:19.59 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 23:20.50 | leifmadsen | metfan2007: it'll remove all your formatting and such -- including comments, last I checked |
| 23:20.50 | leifmadsen | so it's not really a great solution |
| 23:20.50 | beek | Probably gives static priorities, too. |
| 23:20.50 | manxpower | hugs his dialplan |
| 23:20.51 | beek | hugs his subversion repository. |
| 23:21.18 | beek | leifmadsen: What does MinivmMWI() do? The poll, then the extenrnotify? |
| 23:21.45 | leifmadsen | beek: no, it is a dialplan application you can use to trigger MWI on or off |
| 23:22.05 | beek | Damn this proprietary old legacy system. |
| 23:22.08 | [TK]D-Fender | leifmadsen: No, he needs to set it on the OTHER PBX, not on * |
| 23:22.19 | beek | [TK]D-Fender: Correct. |
| 23:22.23 | manxpower | leifmadsen: he needs to run a script to poke a legacy pbx when someone deletes their IMAP voicemail. |
| 23:22.26 | [TK]D-Fender | leifmadsen: And triggered when the state should change. |
| 23:22.30 | leifmadsen | I get it now, thanks |
| 23:22.34 | [TK]D-Fender | leifmadsen: :) |
| 23:22.37 | leifmadsen | I'm an idiot, just move along |
| 23:22.45 | thehar | leifmadsen: nonsense! |
| 23:23.02 | [TK]D-Fender | leifmadsen: Far from... just missed one litte (crucial) thing :) |
| 23:24.36 | [TK]D-Fender | I can't never not ever see how one boolean value can't not never cause an interpretation to actually be interpreted backwards! |
| 23:24.42 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1176438379.dsl.bell.ca) |
| 23:24.44 | darkdrgn2k | hey guys |
| 23:24.54 | darkdrgn2k | im having problems registering to a sip that has wored before |
| 23:24.58 | darkdrgn2k | *worked |
| 23:25.00 | darkdrgn2k | i keep getting |
| 23:25.06 | darkdrgn2k | Got SIP response 301 "Moved Permanently" back from 205.205.231.22 |
| 23:25.18 | darkdrgn2k | i THINK the provider moved their servers around, what am i looking for |
| 23:28.44 | metfan2007 | russellb: when I execute "dialplan save" it disconnect me from CLI and an empty extensions.conf is created :( |
| 23:29.00 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:29.29 | russellb | oic. |
| 23:29.35 | russellb | I guess it doesn't work? |
| 23:29.37 | russellb | :-) |
| 23:29.49 | metfan2007 | russellb: :P |
| 23:32.04 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:32.06 | jblack | p3nguin: ping |
| 23:34.52 | Katty | what is the opposite of 'overage' |
| 23:34.57 | Katty | and odn't tell me underage |
| 23:34.59 | darkdrgn2k | so any ideas about 301? |
| 23:35.05 | metfan2007 | russellb: well, I found a very old backup, is better than nothing, thnks anyway! |
| 23:35.12 | russellb | np |
| 23:35.34 | Katty | google says shortage :> |
| 23:36.01 | jblack | p3nguin: unping |
| 23:36.23 | leifmadsen | russellb: check it out now: https://issues.asterisk.org/roadmap_page.php |
| 23:37.36 | russellb | leifmadsen: nice work sir |
| 23:37.56 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:37.57 | darkdrgn2k | huh intersintg.. |
| 23:38.01 | darkdrgn2k | asterisk doesnt follow 301s ? |
| 23:38.01 | leifmadsen | russellb: merci! I'm currently going through the closed features and assigning them to 1.8 if there was a commit |
| 23:38.28 | russellb | and if the commit is post 1.6.2 i presume |
| 23:38.41 | leifmadsen | russellb: right |
| 23:43.37 | jblack | can astdatadir take more than one path? |
| 23:44.47 | jblack | (what I really want to do is add an extra local path) |
| 23:45.05 | hardwire | jblack: for why? |
| 23:46.00 | jblack | I always try to seperate localized data. |
| 23:46.21 | hardwire | symlinks? |
| 23:46.34 | russellb | only one path |
| 23:46.42 | jblack | russellb: Thanks. =) |
| 23:46.44 | hardwire | with multiple subdirectories. |
| 23:47.51 | hardwire | jblack: curiousity is killing me |
| 23:47.54 | hardwire | what are you working on? |
| 23:49.03 | eppigy | Katty: shortage |
| 23:49.07 | jblack | hardwire: Just putting together a new pbx. |
| 23:49.30 | beek | Does the data structure printed when I type: 'voicemail show users' get updated during a poll? |
| 23:49.52 | hardwire | jblack: awwww.. ok |
| 23:50.12 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:d25:f425:6fa7:fd7d) |
| 23:50.13 | jblack | getting my primary phone system off of my local dsl line. |
| 23:50.22 | hardwire | :P |
| 23:50.38 | [TK]D-Fender | jblack: After all the fun we had with exacly HOW you got to your "DSL"? ;) |
| 23:50.39 | jblack | now that I have the cell phone stuff working right, I need to put in place the ivr I got p3nguin to record for me. just silly stuff. |
| 23:50.55 | [TK]D-Fender | remembers much netowking craziness |
| 23:50.58 | jblack | [TK]D-Fender: Oh, it'll be there. I'm just offloading the critical sections |
| 23:51.07 | [TK]D-Fender | jblack: :p |
| 23:51.25 | [TK]D-Fender | jbDo share the prompts he made.... I'm told I could do them professionally myself... |
| 23:51.31 | jblack | goddammittheexwifehasmycellphonenumberfuckme |
| 23:51.51 | jblack | wget http://24.171.73.141/jblack-asterisk-sounds.tgz |
| 23:51.59 | jblack | nothing special. |
| 23:52.14 | jblack | just a small handful of fall-through follow-me-like stuff on the cheap. |
| 23:52.28 | jblack | note to self: once I get the new ivr in, change the cell phone #. |
| 23:52.52 | jblack | unfortunately, sipdroid is an utter failure. They're just not ready yet. |
| 23:55.30 | hardwire | p3nguin: you haz a sexy voice? |
| 23:56.13 | ChannelZ | does he say 'baby' at the end of each phrase? |
| 23:56.19 | jblack | ignoring sip OPTION , which breaks qualify. reporting it's ip as localhost, thus requiring NAT evne when on the local wifi network, packet dropping like a drug dealer being chased by cops.... |
| 23:58.51 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.183.228) |
