| 00:00.16 | drmessano | sfire: User/password/proxy is all you need for 99% of clients |
| 00:00.31 | sfire | ooohhh I'm not using the proxy |
| 00:00.34 | sfire | could that be the issue |
| 00:01.14 | drmessano | hang on |
| 00:02.11 | drmessano | Before you go latching at boobs flashing between the grainy bars of your scrambled porn channel of guess at this phone, what server related settings are there? |
| 00:02.32 | sfire | http://www.voip-info.org/wiki/view/Nokia |
| 00:02.38 | sfire | that is the guide I was following |
| 00:03.01 | sfire | sown in the basic phone configuration section |
| 00:04.23 | drmessano | Ok |
| 00:04.29 | sfire | I just threw some proxy info in there .. It still registered however still no audio |
| 00:04.40 | drmessano | Forget proxy |
| 00:04.43 | sfire | ok |
| 00:04.45 | drmessano | it has a registrar |
| 00:04.49 | drmessano | You dont need proxy |
| 00:05.00 | drmessano | gah |
| 00:05.22 | drmessano | Is it registered? |
| 00:05.26 | sfire | yep |
| 00:05.27 | drmessano | Ok yes |
| 00:05.42 | drmessano | So whats the issue? No one to call you? |
| 00:05.44 | sfire | I removed the proxy crap and it re-registered just fine |
| 00:06.08 | sfire | no.. no one is at the office.. I was just trying to use the echo test .. or voicemail or anything to hear audio |
| 00:06.24 | sharp | anybody want to help me test zrtp?? |
| 00:06.45 | drmessano | sfire: Is your box set up to handle nat? |
| 00:07.07 | sfire | yep |
| 00:07.18 | drmessano | and youre sure its correct? |
| 00:07.25 | sfire | lol.. no |
| 00:07.32 | drmessano | Of course not |
| 00:07.49 | sfire | nat=yes |
| 00:07.49 | sfire | externip=173.65.14.10 |
| 00:07.49 | sfire | localhost=192.168.0.0/24 |
| 00:07.54 | drmessano | Rip apart 3 routers, throw bullshit in whatever config box looks good |
| 00:07.55 | sfire | thats all I have in it currently |
| 00:07.57 | drmessano | HAHAH |
| 00:08.00 | drmessano | fail |
| 00:08.08 | drmessano | localnet=192.168.0.0/24 |
| 00:08.39 | drmessano | and in which file do you have those settings? |
| 00:08.44 | sfire | cowers head in shame |
| 00:08.48 | sfire | sip_nat.conf |
| 00:09.00 | drmessano | Fix it and try a call |
| 00:13.37 | sfire | drmessano, you totally rock |
| 00:13.55 | drmessano | Its always far easier that it looks |
| 00:14.28 | sfire | I really really appreciate it |
| 00:15.25 | drmessano | Now I feel bad for the things I called you |
| 00:15.34 | drmessano | and for unplugging C4colo's router |
| 00:16.16 | sfire | yea.. echo test is perfect now :) |
| 00:16.55 | drmessano | Good stuff |
| 00:17.11 | drmessano | Now go fix your DMZ shit |
| 00:17.17 | drmessano | 5060 10001-20000 |
| 00:17.23 | drmessano | before I hack you or whatever or something |
| 00:18.48 | sfire | there.. off |
| 00:18.49 | sfire | hehehe |
| 00:19.17 | sfire | echo test still works.. so you were 100% correct.. not a port problem |
| 00:19.34 | KavanS | lol |
| 00:19.45 | drmessano | You dont need to tell me I am right. Really, its not necessary.. No, please.. stop.. I beg you |
| 00:20.18 | drmessano | People are wildly confused over ports |
| 00:20.44 | jaytee | I'm just in awe of your razor honed troubleshooting skills |
| 00:21.09 | drmessano | and I wont claim to be a MICUSUODJDHCSBNP, but the ports your client negotiates vs the far end are two different things |
| 00:21.33 | drmessano | and it confuses a lot of people.. dont believe the hype |
| 00:22.54 | sfire | now I can hide at home when the boss isn't in the office :p |
| 00:22.56 | sfire | hahahahaha |
| 00:26.23 | drmessano | At my old job, they had put in "modem" lines to each office... Each line was a split of their direct line... one end into the PBX, the other to that jack.. |
| 00:26.29 | drmessano | So I took a SPA-3102 |
| 00:26.41 | drmessano | Set it up to register the FXO back to my PBX at home.. and plugged it in |
| 00:26.53 | drmessano | bam, office phone ringing at home |
| 00:27.32 | sfire | hehehehe |
| 00:27.58 | sfire | this is the first "real" phone system that I've done (if you can't tell) |
| 00:27.59 | sfire | hehehe |
| 00:28.13 | sfire | I did it at home for testing but it was all internal networking |
| 00:28.21 | drmessano | Even better though, office phone out.. So I could make work calls from here |
| 00:29.15 | sfire | on this nokia I just dial 9xxxxxxx@serverip and it uses the work system |
| 00:29.20 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 00:29.24 | sfire | if I just dial normally it uses my home callcentric :) |
| 00:29.31 | sfire | I can be on both SIP networks at the same time |
| 00:29.46 | sfire | I love this phone :) |
| 00:29.50 | telnettech | TK: will you look at this pastebin and tell me if you see anything wrong? http://pastebin.com/d17d96bf0 |
| 00:30.28 | telnettech | or jaytee......http://pastebin.com/d17d96bf0 |
| 00:31.03 | Kevin` | what module(s) are needed for the originate cli command |
| 00:33.04 | *** join/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 00:33.18 | telnettech | i know it is not correct |
| 00:34.19 | PanicMan | Remote VoIP Switch >> Asterisk >> E1 Provider, need someone who can provide me the solution |
| 00:34.22 | jaytee | dunno |
| 00:34.56 | telnettech | i think it should be this after im looking at it.......http://pastebin.com/d100a56d6 |
| 00:35.17 | jaytee | why are you using Background instead of Playback on line 21 and you have playback commented out on line 20? |
| 00:35.42 | PanicMan | anyone care to help ? |
| 00:35.42 | telnettech | the playback is not needed |
| 00:35.52 | jaytee | and quit putting leading ...... crap in front of http in your links |
| 00:36.05 | telnettech | that is a mistake that i just commented out but havent removed |
| 00:36.22 | *** part/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 00:36.31 | drmessano | I hate when people do that |
| 00:36.36 | telnettech | but im thinking that the playback should be in there and not the background |
| 00:36.38 | drmessano | If I have to copy and paste, I am done |
| 00:37.40 | jaytee | unless you're waiting for user input at that point Playback would be preferable |
| 00:37.47 | KyleK | in gnome-terminal its clickable |
| 00:38.04 | jaytee | Background is going to expect digits |
| 00:38.10 | telnettech | jaytee thats what i was thinking but then for some reason i had a brain fart and put the background....it stinks |
| 00:38.31 | jaytee | so does your style of posting pastebin links but we won't go into that now |
| 00:39.08 | KyleK | whats the _Z match? |
| 00:40.00 | drmessano | KyleK: Im sure it works in Lynx too, ROFLCOPTER.. ZOMG LMFAO... |
| 00:40.03 | drmessano | um yeah |
| 00:40.06 | telnettech | _Z is any digit 1 to 9....depends on how many codes the PMS vendor has for room status |
| 00:40.24 | telnettech | that is taken care of by another application running between asterisk and PMS |
| 00:42.02 | telnettech | KyleK: so the housekeeper will enter a code 1 thru 9 followed by a pound key so that we can match it to the correct status in the pms system so that when tehy pull a report on the guest rooms, they can see words and not some number code |
| 00:42.27 | drmessano | Interesting |
| 00:42.28 | jaytee | why not just hire maids that don't suffer from PMS? |
| 00:42.41 | drmessano | Holy crap |
| 00:43.00 | drmessano | and I thought cutting out a skunks spray gland was cool |
| 00:43.03 | telnettech | PMS(property mgmt system) |
| 00:43.49 | jaytee | makes mental note to never stand downwind of drmessano |
| 00:45.04 | KyleK | telnettech: so when they're done with a room they dial up the system and punch in a code like 9->they stole the TV? |
| 00:45.17 | sfire | can anyone direct me to a guide on setting up one of those automated call routing systems? the "press 1 for sales, 2 for support, ect" |
| 00:45.23 | jaytee | I feel like I'm only looking at half of something |
| 00:45.41 | jaytee | sfire, search the WIKI for IVR |
| 00:45.47 | jaytee | lots of stuff there |
| 00:45.49 | sfire | thanks jaytee :) |
| 00:46.02 | drmessano | Arent you using FreePBX? |
| 00:46.11 | jaytee | http://www.voip-info.org |
| 00:46.16 | sfire | drmessano, yes |
| 00:46.26 | drmessano | Its all in the GUi, dude |
| 00:46.27 | sfire | I didn't see it in the menus |
| 00:46.29 | drmessano | Just click |
| 00:46.43 | drmessano | --> #freepbx |
| 00:47.04 | sfire | ok |
| 00:48.01 | telnettech | KyleK: more like money out of your wallet......but yeah, they use this by the supervisor who goes thru and checkes the room and then they clear the room, based on another code they enter using same system, to clear room as ready for guest |
| 00:52.31 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 00:53.40 | jaytee | is the room_status var a global var or local? |
| 00:54.57 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 00:56.21 | telnettech | room_status is just local in the astdb |
| 00:56.47 | jaytee | and PMS reads the status flag from the astdb, ok |
| 00:58.06 | telnettech | no there is a AGI script that takes it from the astDB and passes it to the pms.....that is the send_status part of the dialpan |
| 00:59.04 | telnettech | there is a part of the dialplan that is not showed....i was trying to make the system hangup after a count of 3 tries |
| 00:59.45 | telnettech | the other part is already working...but we are seeing calls that are not hanging up properly on the CLI that we believe is slowing down the service |
| 01:00.07 | telnettech | i shouldnt say we....i should say I |
| 01:01.01 | *** join/#asterisk LakeSolon (n=blake@96-42-127-243.dhcp.roch.mn.charter.com) |
| 01:01.47 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:01.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:02.46 | *** join/#asterisk chendy (n=chatzill@58.251.103.75) |
| 01:06.51 | *** join/#asterisk philippel (n=p_lindhe@pool-98-111-74-92.sttlwa.fios.verizon.net) |
| 01:08.34 | jaytee | <PROTECTED> |
| 01:08.42 | telnettech | yes |
| 01:08.51 | jaytee | ugh |
| 01:09.12 | telnettech | but it should be able to carry to 1.4 though when we are finished with the beta testing |
| 01:09.38 | therealcircut | is there any way to force sip clients to re-register with the pbx? |
| 01:09.41 | *** join/#asterisk saftsack (n=saftsack@87.146.74.142) |
| 01:10.08 | jaytee | not with the way you're using Set to set the value of the room_number "variable" or field. |
| 01:10.34 | telnettech | what do you mean? |
| 01:11.12 | therealcircut | u talkin to me? |
| 01:11.28 | telnettech | no sorry jaytee |
| 01:12.20 | telnettech | BBIAB |
| 01:12.35 | jaytee | in 1.4 the syntax is Set(DB(room_number)=${EXTEN:0:2}) |
| 01:13.08 | jaytee | or room_status rather |
| 01:15.36 | philippel | anyone familiar with the new queuerules.conf? I'm trying to understand the format of the file and it's a bit vague from the documentation? If you have a rule called [myrule] as in the documentation, do you set 'myrule' to the same value as your queue number so that rule applies to your given queue? |
| 01:16.02 | philippel | or, is there a setting in queues.conf that allow you to assciate a named queuerule with a give queue that I did not see? |
| 01:17.01 | philippel | ok I was being dumb, was looking in an old version of the queues.conf sample file, sorry |
| 01:17.36 | jaytee | what version did they add queuerules.conf in? |
| 01:18.00 | philippel | don't know - I've just been looking at trunk scanning various changes and additions |
| 01:19.27 | jaytee | ah, 1.6.0.1 has it |
| 01:19.47 | jaytee | but 1.4.15 doesn't. |
| 01:20.07 | philippel | oh - I could have answered that, sorry, it is not in 1.4.x anywhere |
| 01:20.34 | jaytee | yeah, I don't recall seeing it on 1.4.22 either |
| 01:20.34 | philippel | if you are interested in queue stuff though, I've got a usef patch, well potentially useful to some: |
| 01:21.00 | philippel | M15168 |
| 01:21.04 | philippel | https://issues.asterisk.org/view.php?id=15168 |
| 01:21.09 | jaytee | I only have 1 queue with 2 possible users |
| 01:21.09 | philippel | guess that doesn't work here |
| 01:21.16 | jaytee | not very complicated either |
| 01:21.41 | philippel | jaytee well depending on your situation, the above coudl be interesting or not |
| 01:21.58 | philippel | anyhow brb |
| 01:28.39 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 01:29.57 | telnettech | im back |
| 01:30.03 | carrar | THANKS |
| 01:31.09 | jaytee | :-) |
| 01:31.11 | telnettech | jaytee: yeah we will change that part but Im working on the system that i have for now and want to get it working correctly for this customer |
| 01:31.35 | telnettech | the beta is not done yet.....should be by end of summer |
| 01:32.16 | jaytee | so are you aware that your pastebin has the value of count set to 2 the very first time GotoIf is executed? |
| 01:32.35 | telnettech | jaytee: they are slowly seeing things my way as far as doing the right thing before just putting a noncompleted system on the market |
| 01:33.19 | drmessano | anyone know if the fedora mirrors have a round robin DNS address i can use for downloads? |
| 01:33.33 | telnettech | see that is where i need the help....i dont seem to be able to get that correct |
| 01:33.41 | *** part/#asterisk philippel (n=p_lindhe@pool-98-111-74-92.sttlwa.fios.verizon.net) |
| 01:34.18 | jaytee | it's a loop that's supposed to quit if the value of count exceeds or matches 3? |
| 01:34.36 | jaytee | and by quit I mean Hangup() |
| 01:34.43 | telnettech | It is cause of the Answer function of the dialplan |
| 01:35.23 | jaytee | I don't recall if the h extension existed in 1.2 or not |
| 01:37.02 | telnettech | yeah but that is if someone hangs up the phone |
| 01:37.15 | jaytee | telnettech, if you change the starting value of count from 1 to 0 on line it will run 3 times and then jump to priority 8. |
| 01:37.47 | telnettech | ahahahahahahah |
| 01:37.57 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:37.58 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 01:38.29 | jaytee | otherwise your if there is 1 failed attempt to get the room status the second time the GotoIf runs the way you have it it will see it as 3 instead of two and try to hangup. |
| 01:39.44 | telnettech | jaytee: it will jump to 8 if the count hasnt reached 3 yet, right? |
| 01:40.50 | jaytee | no, if count equals 3 it will jump to priority 8 the way you have it. |
| 01:41.08 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-154-133-57.dsl.lsan03.sbcglobal.net) |
| 01:43.03 | jaytee | but you really only get 2 iterations with count set to 1 at priority 3. the next priority increments the value of count to 2 and the first time it runs if it fails it jumps back to priority 4 and increments to 3 and hangs up so you really only get 1 possible failed attempt. |
| 01:44.02 | voxter | anyone here use web-meetme |
| 01:45.23 | telnettech | i have changed the count to equal 0 but it should jump to 8 only if the count has not reached 3......i thought before the : is true and after : it is false |
| 01:45.48 | telnettech | I dont have anything before the : |
| 01:46.23 | jaytee | you're right, I missed the colon |
| 01:46.37 | telnettech | ok |
| 01:46.58 | telnettech | i was ready to enroll into the basic course if I didnt get that right |
| 01:48.15 | telnettech | and the autofallthrough is yes in the general section of the extensions.conf file so becuse i didnt put anything before the : it will do the NoOp and then hangup |
| 01:49.28 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:50.20 | jaytee | ok, yeah I can see it jumping to 8 as long as count is less than 3 but it's still starting at 2 the first time GotoIf evaluates count |
| 01:51.16 | telnettech | i have changed that part |
| 01:52.11 | telnettech | let me ask, do i even need the Set,(Count=0) as the 3rd line |
| 01:52.32 | telnettech | or can i delete that and the next line take care of setting the variable |
| 01:52.40 | jaytee | so the rest "should" work then assuming that Set(room_status) will actually set the value of the key in the astdb |
| 01:54.04 | jaytee | don't know, from habit I always initialize the value of a variable or counter before I run code that might loop. |
| 01:55.01 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:56.14 | telnettech | ok.....so here is what i am putting into the system for this function.....http://pastebin.com/d6c3255d6 |
| 01:57.10 | jaytee | brian!!! how many times have I told you to leave a damn space between your link and what you typed. enough with all the damn periods!!!............................................. |
| 01:58.00 | jaytee | I can't right click and open a new tab if you type a period right before http: and that's just downright rude, man! |
| 01:58.10 | leifmadsen | .................. |
| 01:58.25 | carrar | jayteeyouaresodamnpickywhatarewegoingtodowithyou? |
| 01:58.26 | leifmadsen | jaytee: get a better IRC client! |
| 01:58.46 | leifmadsen | carrar: you mean .http://jayteeyouaresodamnpickywhatarewegoingtodowithyou.com. |
| 01:58.51 | carrar | haha |
| 01:58.57 | carrar | goes to register |
| 01:59.01 | carrar | woah it's already taken |
| 01:59.15 | drmessano | Who the hell puts ...... between a sentence and a link? |
| 01:59.22 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 01:59.48 | carrar | drmessano, no sain person, thats for sure!! |
| 01:59.54 | kn0x | im trying to write an AGI in PHP, and fgets(STDIN, 4096); is only returning the first character |
| 02:00.08 | telnettech | sorry |
| 02:00.10 | jaytee | I'm old, I'm cranky and you'll get my X-chat when you pry it from my cold dead fingers |
| 02:00.35 | carrar | IRC via CLI, thats how I ROLL |
| 02:00.58 | kn0x | is there a reason fgets stops reading STDIN after the FIRST chacter |
| 02:01.21 | drmessano | It doesnt work with mIRC under Vista - Pretty Pink edition |
| 02:01.31 | telnettech | I have a habit of doing that when i have a few thoughts in a row......see I just did it again wothout thought |
| 02:01.44 | carrar | http://us.php.net/fgets |
| 02:02.56 | telnettech | thats how i also update service support tickets when i have multiple points to add after i work on it |
| 02:03.47 | telnettech | i even do that in emails |
| 02:04.08 | drmessano | Well, I like to wipe by butt on the carpet like the dog does.. That doesnt make it ok. |
| 02:04.17 | jaytee | rofl |
| 02:04.27 | rob0 | tmi |
| 02:04.30 | drmessano | *my* |
| 02:04.41 | jaytee | I'd never take my shoes off in his house |
| 02:04.43 | telnettech | roflmao |
| 02:04.55 | drmessano | Point being, knock off the leading ................... before the links, thoughtboy |
| 02:05.09 | rob0 | s/take my shoes off in/enter/ |
| 02:05.43 | rob0 | So drmessano, which kind of carpet is best for that? |
| 02:05.45 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 02:05.53 | jaytee | just asking man! I rarely get to take advantage of the awesomeness of the right mouse click and I hate copying and pasting links |
| 02:05.55 | rob0 | is thinking of the dual meaning of "pile" |
| 02:06.18 | carrar | kn0x, you should cat your file into a PHP test program |
| 02:06.23 | telnettech | so anyways jaytee this is what i am putting in the system now that we have gone thru it http://pastebin.com/d6c3255d6 |
| 02:06.25 | drmessano | I wasnt prepared for the comment I made about the carpet, so therefore, I am void of followups |
| 02:06.31 | rob0 | :) |
| 02:06.44 | drmessano | Thats how it happens sometimes |
| 02:06.48 | drmessano | I just roll.. |
| 02:06.49 | rob0 | Think of something! QUICK!! |
| 02:07.01 | rob0 | roll ... on the carpet? |
| 02:07.09 | drmessano | lol |
| 02:07.26 | jaytee | telnettech, I'm still not sure about your method of using Set to set the value of room_status if it's an astdb key. the syntax in 1.4 is Set(DB(family/key)= somedamnthing |
| 02:07.51 | telnettech | it has been working jaytee |
| 02:07.53 | jaytee | but I don't recall enough about 1.2 |
| 02:07.57 | drmessano | Well, taking the bender approach from The Breakfast Club, which kind of carpet is best for that? jaytee's |
| 02:08.08 | rob0 | haha |
| 02:08.18 | drmessano | John Hughes style |
| 02:08.21 | drmessano | Booyah |
| 02:08.36 | jaytee | drmessano, might as well! it'll blend in fine with all the ground-in cat puke |
| 02:08.38 | telnettech | i am just adding the count part as I was watching for something else on the CLI and seen calls to the maid service prompt at 9pm |
| 02:08.40 | rob0 | Do you know why a dog licks his/her ... ahem ... genital area? |
| 02:08.41 | kn0x | carrar: i have |
| 02:08.51 | telnettech | I know they dont have housekeeping at this time of night |
| 02:08.57 | kn0x | carrar: fgets is only reading the first character of input |
| 02:09.11 | jaytee | because it can |
| 02:09.36 | jaytee | the better hotels have housekeeping all hours. |
| 02:09.47 | drmessano | jaytee: A-MEN brother.. effin cats.. |
| 02:10.24 | telnettech | well this is a hilton garden inn that is in Beavercreek, Ohio....near dayton |
| 02:10.26 | jaytee | after these feline monsters are gone that's it for me and cats |
| 02:10.34 | drmessano | You know how to keep the cat from puking on the carpet? |
| 02:10.40 | jaytee | don't feed it? |
| 02:10.42 | drmessano | Let the pitbull sleep inside |
| 02:10.49 | telnettech | 2 or 3 star max rating |
| 02:10.59 | drmessano | or |
| 02:11.06 | drmessano | "feed it to the dog" |
| 02:11.15 | drmessano | Which works for most cat jokes |
| 02:15.05 | telnettech | ok im out of here for tonight |
| 02:15.10 | telnettech | good night guys |
| 02:15.45 | jaytee | nite |
| 02:16.30 | drmessano | They dont make em like that anymore, kids |
| 02:16.39 | jaytee | hehe |
| 02:18.55 | drmessano | God I hate Nancy Grace |
| 02:19.28 | drmessano | ** BREAKING NEWS ** Asterisk found with SIP stack hidden under source tree |
| 02:20.16 | jaytee | She'll get all the victims on with live interviews and use phony compassion to exploit their suffering for better ratings. |
| 02:20.32 | drmessano | **SHOCKER** Asterisk contained the word TCP in sip.c 2 YEARS BEFORE 1.6 |
| 02:20.57 | drmessano | Everything is a shocker or breaking news |
| 02:21.12 | jaytee | she was one of the "Top Five People I'd most Like To Punch In The Face" quiz choices I took on Facebook |
| 02:21.16 | florz | "under the source tree"? I suppose that that's outside the part that's getting compiled, then? =:-) |
| 02:21.29 | Qwell | florz: no, it compiles top-down |
| 02:21.35 | jaytee | along with Bill O'Reilly in #1 slot and Dr. Phil at #3 |
| 02:21.55 | florz | but ... how comes asterisk doesn't understand SIP, then? |
| 02:22.57 | jaytee | think of SIP as Spanish then think of the guy that dropped out of Spanish II mid-semester, that guy's Asterisk :-) |
| 02:23.00 | drmessano | ** BREAKING NEWS ** Avaya phone system found dead less than a half mile from Digium Headquarters |
| 02:23.17 | Qwell | drmessano: at Adtran? |
| 02:23.25 | drmessano | heh |
| 02:23.42 | Qwell | wonders how far Adtran actually is. I can see it from my office. Right there *points* |
| 02:23.53 | drmessano | Im jealous |
| 02:24.06 | Qwell | wtf |
| 02:24.18 | drmessano | Office, Window, being able to point.... :( |
| 02:24.20 | Qwell | http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=adtran,huntsville,+al&vps=1&jsv=159e&sll=37.0625,-95.677068&sspn=46.27475,114.257812&ie=UTF8&latlng=34721663,-86682284,8623527788399314309&ei=OKQcSvDIEZrSMKHn6PsD&cd=3 |
| 02:24.22 | Qwell | Click. |
| 02:24.24 | Qwell | now. |
| 02:24.29 | florz | jaytee: shouldnt that suffice for being able to tell different parts of sentences apart reliably? :-> |
| 02:24.32 | jaytee | they make righteous CSU equipment and channel banks but their SIP phones are ugly looking |
| 02:24.35 | drmessano | hahahha |
| 02:24.56 | drmessano | Is that their new corporate entity name? |
| 02:25.36 | drmessano | Thats too cool |
| 02:25.36 | Qwell | anyways - answer, 0.8mi |
| 02:25.49 | drmessano | We need to get "Digium Dr. Pepper stash" on there |
| 02:26.41 | Qwell | I wonder.. |
| 02:26.45 | jaytee | so like, it's 10:28pm and you're still at the office? |
| 02:27.17 | Qwell | jaytee: no |
| 02:27.31 | Qwell | lame. No "Lockheed Martin Baseball Field" |
| 02:28.42 | Qwell | jaytee: You've never virtually pointed over a VPN connection? |
| 02:29.47 | drmessano | http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=digium,huntsville,+al&sll=34.726888,-86.674554&sspn=0.005449,0.008111&ie=UTF8&ll=34.736957,-86.549456&spn=0.010897,0.016222&t=h&z=16&iwloc=B |
| 02:29.51 | jaytee | sure |
| 02:29.51 | drmessano | Thats hardcore |
| 02:29.59 | drmessano | Digium codes in an open field |
| 02:30.12 | drmessano | Taking "open source" to a new level |
| 02:30.35 | Qwell | heh, that isn't an open field |
| 02:30.47 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-7c139644331f79a5) |
| 02:31.11 | Qwell | note the use of the photoshop clone tool |
| 02:31.47 | drmessano | Corporate espionage? |
| 02:32.00 | drmessano | CAMOFLAUGE! |
| 02:32.04 | drmessano | Oh I get it |
| 02:32.42 | drmessano | Scroll in close, and you can see russellb eating a popsicle in the parking lot.. below the mesh |
| 02:33.08 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:33.10 | Qwell | want a good map? |
| 02:33.12 | kn0x | anyone tell me what im doing wrong with my PHP AGI |
| 02:33.20 | kn0x | fgets is only getting 1 char of input |
| 02:33.24 | drmessano | Sure |
| 02:33.36 | kn0x | http://pastebin.ca/1435736 |
| 02:33.59 | drmessano | OMG |
| 02:34.06 | drmessano | I just noticed something |
| 02:34.13 | drmessano | go to that link I posted |
| 02:34.15 | drmessano | and look at the ads |
| 02:34.31 | drmessano | Asterix on Windows? |
| 02:34.32 | drmessano | Try 3CX Phone System for Windows |
| 02:34.32 | drmessano | Easy to install, Free, Download! |
| 02:34.32 | drmessano | www.3CX.com/Phone-System/ |
| 02:34.34 | Qwell | http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&sll=34.726888,-86.674554&sspn=0.005449,0.008111&ie=UTF8&t=h&ll=34.712165,-86.656537&spn=0,359.996513&z=19&iwloc=B&layer=c&cbll=34.711856,-86.656307&panoid=aOG2P3jIeia8Z3Qr5EE8zw&cbp=12,189.09,,0,-17.71 |
| 02:34.44 | drmessano | They cant even SPELL the product they're alluding to |
| 02:35.12 | drmessano | R U ASTERIX BUT, U CAN HAZ WINDOZ? MEH |
| 02:35.33 | drmessano | awesome |
| 02:35.33 | Qwell | best map ever :D |
| 02:36.08 | Qwell | Standing up is a replica Saturn V rocket. In the building is a real one. |
| 02:37.55 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 02:38.59 | jaytee | below the Saturn 5 in the map are more rockets. one looks like another Saturn 5 on it's side and broken apart in stages and just above the 1st stage looks like an Atlas lying on it's side. |
| 02:39.30 | seanbright | i heard that saturn vs burnt a ridiculous amount of fuel per second |
| 02:39.38 | seanbright | on that discovery series... when we left earth? |
| 02:39.39 | Qwell | jaytee: they built the building for it. it's inside now |
| 02:39.51 | jaytee | yeah, it was outrageous what the first stage did |
| 02:40.00 | seanbright | 3 tons per second |
| 02:40.14 | jaytee | kerosene and lox |
| 02:40.33 | mmlj4 | and bagels, don't forget the bagels |
| 02:40.45 | Qwell | jaytee: name the jet just to the east of that |
| 02:42.11 | rob0 | 34.7, -86.6 sounds like HSV |
| 02:42.18 | jaytee | I don't see a jet, I see a shuttle |
| 02:42.28 | rob0 | and therefore the jet would be an SR-71 |
| 02:42.41 | jaytee | to the right of the Atlas? |
| 02:42.45 | seanbright | or "blackbird" |
| 02:42.56 | drmessano | Ah |
| 02:42.57 | Qwell | seanbright: at the street |
| 02:43.01 | Qwell | err, jaytee ^^ |
| 02:43.02 | drmessano | I just scrolled there :( |
| 02:43.10 | jaytee | leaked like crazy till the fuselage heated up and expanded |
| 02:43.20 | seanbright | puts on D.A.R.Y.L. |
| 02:43.35 | rob0 | I saw (on radar) the one at Smithsonian while it was setting its cross-continental speed record. |
| 02:43.39 | jaytee | oh, at the top, yep that's a SR-71 |
| 02:43.48 | Qwell | rob0: O.o |
| 02:43.58 | drmessano | I thought seeing the inside of the shuttle mockup at Cape Canaveral was cool |
| 02:44.04 | jaytee | they had orange drag chutes, more orangey than Digium orange |
| 02:44.08 | Qwell | rob0: how/why did you know that? |
| 02:45.22 | jaytee | if you took a #2 pencil and dragged the point along the fuselage halfway up all along the both sides from tip to tail the aircraft would tear in half after flying a mach 4+ for more than an hour. |
| 02:45.38 | seanbright | jaytee: i totally did that once |
| 02:46.01 | seanbright | then jinx sent me and max into space |
| 02:46.09 | jaytee | if it was operational you'd never have gotten within 1000 yards of it before being challenged and |
| 02:46.10 | drmessano | I did that.. but instead of a pencil, I used menthos |
| 02:46.11 | rob0 | has been through there on I-565 quite a few times |
| 02:46.14 | drmessano | Its on youtube |
| 02:46.20 | jaytee | they'd have shot you at 500 yards |
| 02:46.38 | seanbright | jaytee: i'm quick like a bunny |
| 02:46.39 | rob0 | (my permanent residence is near Florence) |
| 02:46.40 | seanbright | :) |
| 02:46.44 | Qwell | rob0: oh |
| 02:47.02 | jaytee | I liked the spacey blond girl in Spacecamp |
| 02:47.12 | drmessano | ZOMG |
| 02:47.18 | seanbright | kelly preston |
| 02:47.19 | drmessano | Spacecamp rocked |
| 02:47.19 | seanbright | tish |
| 02:47.22 | jaytee | the other one from Red Dawn, not so much |
| 02:47.26 | seanbright | travolta's wife |
| 02:47.28 | Qwell | jaytee: heh, there are a few (innocent enough looking...) streets here that you can't go down |
| 02:47.41 | jaytee | and I'd still do Kate Capshaw |
| 02:47.48 | Qwell | you go down those streets - you better have a good answer |
| 02:47.55 | seanbright | jaytee: i prefer lea thompson |
| 02:48.10 | drmessano | Lea Thompson got HOTTER |
| 02:48.11 | seanbright | the whole loraine from bttf thing... |
| 02:48.20 | drmessano | That doesnt happen often |
| 02:49.17 | seanbright | she was hot in bttf 2 as a cougar with big fake cans |
| 02:49.40 | seanbright | sorry... cans is derogatory |
| 02:49.43 | seanbright | jugs. |
| 02:49.46 | seanbright | there. better. |
| 02:50.03 | MaliutaLap | norks? |
| 02:51.00 | jaytee | there was a period of about 6 months where I desperately wanted to die and be reincarnated as Salma Hayek's bicycle seat |
| 02:53.48 | seanbright | you are not alone. |
| 02:53.57 | seanbright | she doesn't do it for me anymore, though. |
| 02:53.59 | seanbright | not sure what happened. |
| 02:54.40 | jaytee | yeah, it just passed after awhile |
| 02:56.26 | seanbright | oh... so i saw something today and i have to share |
| 02:56.38 | seanbright | feel free to ignore it if you are not a fan of the ladies |
| 02:56.49 | drmessano | wanders off |
| 02:56.54 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 02:57.12 | seanbright | <PROTECTED> |
| 02:57.18 | seanbright | do it now. thank me later. |
| 02:59.43 | drmessano | The turkey is done, apparently |
| 02:59.49 | jaytee | oooh! perky! |
| 03:00.08 | seanbright | zing! |
| 03:00.34 | jaytee | that would have been better as a Powerpoint presentation :-) |
| 03:00.39 | seanbright | heh |
| 03:00.50 | seanbright | too good not to share. |
| 03:05.03 | drmessano | Jon and Hate plus 8 |
| 03:05.49 | seanbright | 'howard the duck' was released on dvd and no one told me? wtf? |
| 03:05.57 | drmessano | Whoa |
| 03:06.13 | drmessano | Maybe Electric Dreams will follow soon |
| 03:06.15 | jaytee | isn't Leah Thompson in that? |
| 03:06.25 | jaytee | Electric Dreams is great! |
| 03:07.02 | drmessano | Lenny Von Dohlen and Virginia Madsen |
| 03:07.40 | drmessano | I actually have it on VCD |
| 03:08.03 | drmessano | and get this.. it was an official release.. in singapore |
| 03:08.09 | drmessano | I thought it was some bootleg |
| 03:08.21 | drmessano | But apparently thats as close to DVD as its gotten.. |
| 03:09.11 | jaytee | Virginia Madsen. mmmm, mmmm, mmmm, mmmmm. mmmmmmmmmmmm! |
| 03:09.29 | drmessano | WHOA |
| 03:09.31 | drmessano | http://www.play.com/DVD/DVD/4-/8852274/Electric-Dreams/Product.html |
| 03:09.40 | drmessano | UK release from Apr |
| 03:10.14 | jaytee | will that play on a US player? or do you own a multi-region player? |
| 03:11.16 | drmessano | I have no idea if mine is region free.. But I guess I can get the DVD and find a player to go with it :) |
| 03:12.29 | jaytee | region 2 PAL format :( |
| 03:12.42 | drmessano | I have the soundtrack on CD |
| 03:12.44 | jaytee | most players sold in the US are region 1 only |
| 03:12.53 | jaytee | NTSC |
| 03:13.03 | drmessano | yeah |
| 03:13.33 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 03:13.48 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 03:13.54 | jaytee | this is the same shit i had to put up with for Rain Man and a couple of odd movies I happened to like that most have probably never heard of |
| 03:14.32 | drmessano | Like? |
| 03:14.49 | jaytee | and they've got people actually paying 300 bucks for used Season 1 sets of Farscape |
| 03:15.16 | drmessano | couple of odd movies I happened to like that most have probably never heard of <--??? |
| 03:15.23 | jaytee | Like? you mean odd movies? A Razor's Edge with Bill Murray, took forever to come out on DVD. |
| 03:15.53 | jaytee | and Farewell to the King starring Nick Nolte. Most people probably hated this flick but i loved it. |
| 03:16.14 | kn0x | what about ob_implicit_flush(true); ?? |
| 03:16.18 | kn0x | oops |
| 03:16.29 | drmessano | Ive never seen ob_implicit_flush(true); |
| 03:16.33 | drmessano | Is it on DVD? |
| 03:17.30 | jaytee | what? |
| 03:17.42 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
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| 03:17.50 | jaytee | I'm not gonna google that |
| 03:17.54 | jaytee | nope, not gonna do it! |
| 03:18.20 | carrar | wuss |
| 03:18.26 | jaytee | still not googling ob_implicit_flush(true): |
| 03:18.35 | carrar | Two girls and a Flush! |
| 03:18.48 | carrar | or, should have used a flush |
| 03:18.57 | jaytee | php |
| 03:19.30 | carrar | OH |
| 03:19.31 | carrar | PHP |
| 03:24.26 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 03:25.32 | drmessano | http://wisedonkey.livejournal.com/98142.html |
| 03:25.36 | drmessano | thats what it got me to |
| 03:27.45 | Qwell | ascii...map? |
| 03:28.49 | drmessano | You expect PHP to have a function for that? |
| 03:29.09 | seanbright | gah... |
| 03:30.09 | jaytee | I feel bad for Mike Tyson and I'm not really a big fan of his |
| 03:42.39 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 03:43.14 | [T]ank | are there companies out there that offer asterisk hosting? Similar to how I can go and get a web host for $9 per month? |
| 03:43.31 | [T]ank | not necessarily that price... but that type of service |
| 03:43.42 | [T]ank | where they just give me a blank config and I can set it up however I need to? |
| 03:51.07 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
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| 04:31.16 | jordanl_ | can i set up a sip provider in sip.conf for incoming calls only and match calls to that peer based on a hostname lookup? |
| 04:32.18 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 04:37.57 | carrar | you can |
| 04:38.43 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 04:41.08 | *** join/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
| 04:41.24 | dshap | is anyone here familiar with Youmail.com ?? |
| 04:41.59 | dshap | i have a quick question about their voicemail service...I am trying to implement something similar on my asterisk box |
| 04:42.41 | dshap | they have you set your cell phone such that it uses call forwarding if you do not answer a call |
| 04:42.52 | dshap | the person who is trying to call your cell phone is forwarded to Youmail's phone number |
| 04:43.07 | dshap | and then that person hears your greeting and can leave a message in your mailbox |
| 04:43.45 | dshap | i tried setting my phone's call-forwarding number to my asterisk server's number and when the call was forwarde, i got the caller ID number of the phone that made the call |
| 04:43.48 | dshap | NOT the cell phone |
| 04:44.05 | dshap | but wouldn't the asterisk server need the callerID of the cell phone in order to know which mailbox someone is trying to reach? |
| 04:44.13 | dshap | does anyone have the answer to this? |
| 04:44.42 | j_kroon | hehe, you'll note the cellular providers make you forward to some shortcode followed by your cell number generally. |
| 04:45.15 | dshap | to forward to Youmail, i just need to dial *004*[youmail's number]# |
| 04:45.18 | dshap | it's a GSM code |
| 04:45.26 | dshap | and it sets my call-forwarding to dial youmail's number |
| 04:45.43 | drmessano | dshap: Why would you expect a forwarded call to show the CID of the phone who forwarded it? |
| 04:46.00 | dshap | i'm not saying that i expect it to |
| 04:46.12 | dshap | i'm just saying that i would like to obtain the CID of the phone who forwarded the call |
| 04:46.20 | dshap | and am asking how i can accomplish this |
| 04:46.26 | j_kroon | drmessano, because it's the phone making the call :p. In ZA I would expect exactly that. |
| 04:46.27 | dshap | it seems to me that Youmail does this |
| 04:46.43 | drmessano | j_kroon: Exactly lol |
| 04:47.03 | dshap | ive been using ${CALLERID(num)} |
| 04:47.08 | dshap | maybe there's some other function? |
| 04:47.16 | drmessano | This isnt an asterisk problem |
| 04:47.23 | j_kroon | dshap, no, you've got the right one. |
| 04:47.48 | j_kroon | is [youmail's number] the same for all their clients or do they have a block and is issueing a new number for every client? |
| 04:47.53 | jordanl_ | carrar: what are the basic settings that i need to use in sip.conf for the trunk? |
| 04:47.53 | dshap | then do you guys have any idea how Youmail would know to connect to my voice mailbox when someone else's call is simply forwarded to their server? |
| 04:48.02 | dshap | same number for everyone i'm almost sure |
| 04:48.10 | dshap | but i'm not 100% sure |
| 04:48.11 | j_kroon | _almost_ |
| 04:48.15 | drmessano | If the CID is the original CID, not your cell phone, you cant expect some dialplan in Asterisk to make it so |
| 04:48.18 | dshap | a different number for everyone? |
| 04:48.20 | dshap | wouldn't that be insane? |
| 04:48.24 | dshap | wait a sec... |
| 04:48.25 | drmessano | Not really, no |
| 04:48.31 | dshap | im gonnacall the number from a different phone |
| 04:48.45 | drmessano | Here comes the "ohhh" |
| 04:49.07 | dshap | "please enter your 10 digit phone number" |
| 04:49.16 | dshap | it's not my own number |
| 04:49.49 | j_kroon | dshap, you note that the forward number is *004*some_number? |
| 04:49.56 | j_kroon | note the stars. |
| 04:50.08 | dshap | im pretty sure the *004* is just the code that says "set my forwarding number" |
| 04:51.27 | j_kroon | is it possible to set a phone to forward to some number and immediately when answered send some dtmf string? |
| 04:51.32 | dshap | i thought call-forwarding counts as minutes on your voice plan |
| 04:51.54 | dshap | so i would think taht u could get the CID of the phone doing the forwarrding |
| 04:51.54 | dshap | um |
| 04:51.58 | j_kroon | it may well, doesn't say they have to use your phone's number as CID. |
| 04:52.39 | dshap | im not sure what you're getting at |
| 04:53.53 | dshap | the setup on my cell phone was as simple as *004*some_number |
| 04:54.07 | dshap | apparently that's how you do all phones on AT&T |
| 04:54.12 | dshap | generic config |
| 04:54.34 | drmessano | dshap: Its also entirely possible that youmail is getting a different CID string from its upstream carrier |
| 04:54.55 | drmessano | In a way you wouldnt forwarding otherwise |
| 04:55.22 | dshap | so it's upstream carried might be able to detect when they are receiving a forwarded call |
| 04:55.33 | dshap | and somehow obtains the CID of the phone doing the forwarding? |
| 04:56.00 | drmessano | No, that information is passed anyway.. its up the carrier what remains in transit |
| 04:56.28 | dshap | how might i determine if i have access to the same information with my own carrier? |
| 04:57.06 | drmessano | It isnt a feature you call up AT&T and pay 3.95 a month for |
| 04:57.42 | dshap | no it's included in regular plans |
| 04:57.56 | dshap | i called AT&T and verified that i wasn't paying extra for the ability to do that |
| 04:58.16 | drmessano | You pay extra to get callerid of the phoen forwarding the call? |
| 04:58.33 | drmessano | .... |
| 04:59.02 | dshap | you say upstream carrier |
| 04:59.10 | dshap | you mean like the VoIP SIP trunk provider |
| 04:59.11 | dshap | right? |
| 04:59.21 | drmessano | What youre not getting here is that youre asking for something that you as a LITTLE END USER are not going have access to |
| 04:59.44 | dshap | okay so you're saying i need to pay more to access it |
| 05:00.06 | dshap | i guess i'm just looking for guidance as far as how i might look into paying to access it |
| 05:00.15 | dshap | if there's a name for what i want to do |
| 05:00.17 | dshap | or something like that |
| 05:00.52 | *** join/#asterisk shinao1 (n=shinao1@41.219.208.157) |
| 05:01.05 | drmessano | [00:57] <drmessano> It isnt a feature you call up AT&T and pay 3.95 a month for |
| 05:01.13 | drmessano | ~cluebat |
| 05:01.14 | infobot | *WHACK* *WHACK* *WHACK* |
| 05:01.29 | dshap | ... |
| 05:01.35 | dshap | AT&T is not my provider though |
| 05:01.40 | drmessano | God damnit |
| 05:01.53 | dshap | okay |
| 05:02.01 | dshap | what feature are you talking about |
| 05:02.03 | dshap | call forwarding? |
| 05:02.29 | dshap | *sigh* |
| 05:03.14 | drmessano | What you are asking for is something that a BUSINESS or CARRIER would be a in a position to request.. you are NOT going to call up as joe end user, offer to pay an extra $10 a month, and have them change how your CID works |
| 05:03.25 | carrar | dshap is a diversion header train wreck! |
| 05:03.34 | drmessano | hence the whole [00:57] <drmessano> It isnt a feature you call up AT&T and pay 3.95 a month for, which I DID NOT MEAN LITERALLY |
| 05:03.40 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 05:03.46 | drmessano | Get a fucking clue! |
| 05:03.52 | carrar | call them and ask |
| 05:04.14 | carrar | tell them you are willing to pay $3.95 |
| 05:04.16 | dshap | i guess i'll call and ask |
| 05:04.21 | drmessano | Go for it |
| 05:04.23 | dshap | maybe i'm trying to start a business |
| 05:04.31 | dshap | and maybe i'm willing to pay more |
| 05:04.32 | carrar | and you want it in 30 mins or less |
| 05:04.44 | drmessano | Cool, get a virtual PRI and they will probably talk to you |
| 05:05.18 | carrar | Is that a PRI in the clouds? |
| 05:05.21 | dshap | when i call up and ask if it's possible to get the CID of the forwarding phone |
| 05:05.29 | dshap | is there a name for that? |
| 05:05.32 | drmessano | Wait til they stop laughing |
| 05:05.34 | dshap | specifically |
| 05:05.34 | drmessano | Then... |
| 05:05.36 | dshap | the feature that i want |
| 05:05.49 | carrar | YES |
| 05:06.09 | carrar | Feature CIDOTFP |
| 05:06.46 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.8) |
| 05:07.12 | dshap | i don't think you guys understand how new i am to all of this asterisk/VoIP/telephony stuff |
| 05:07.13 | dshap | haha |
| 05:07.18 | carrar | oh we do |
| 05:07.21 | dshap | the reason i am here in this IRC channel |
| 05:07.25 | dshap | *IS* to get a fucking clue |
| 05:07.38 | drmessano | http://www.voip-info.org/wiki/view/RDNIS |
| 05:07.39 | dshap | yea but you get so frustrated when i don't understand something |
| 05:07.40 | dshap | hah |
| 05:07.52 | drmessano | No |
| 05:08.23 | drmessano | But when you say stupid shit like [01:02] <dshap> AT&T is not my provider though |
| 05:08.37 | dshap | i honestly thought you misunderstood something i said earlier |
| 05:08.40 | dshap | sorry |
| 05:08.42 | drmessano | When someone is trying to put something in context |
| 05:08.46 | jordanl_ | what are the basic trunk settings in sip.conf for a SIP provider through which i want to receive incoming calls only (not using SIP registration, but domain/IP lookup)? |
| 05:08.47 | drmessano | FAIL |
| 05:09.00 | dshap | alright well thanks for the link |
| 05:11.41 | dshap | so do you think Youmail is likely using RDNIS? |
| 05:11.52 | dshap | and because of that they probably pay a shit ton of money for a PRI line? |
| 05:12.15 | drmessano | *A* PRI line? |
| 05:12.22 | dshap | many PRI lines? |
| 05:12.35 | carrar | Bank of soundcards as modems |
| 05:12.38 | drmessano | Depends how big they are.. could be dozens |
| 05:12.54 | drmessano | or so.. or brought in using different tech |
| 05:12.58 | carrar | soldered in parallel |
| 05:13.11 | drmessano | ss7 over RS232 |
| 05:13.21 | carrar | thats the latest craz |
| 05:13.26 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 05:13.31 | drmessano | DB9, werd |
| 05:13.40 | dshap | all of that just so they can get the CID of the forwarding number? |
| 05:13.42 | dshap | =\ |
| 05:13.47 | drmessano | ROFL |
| 05:13.50 | dshap | well it probably is very fast and support a ton of concurrent calls and stuff |
| 05:14.05 | dshap | but what if someone wanted to run something small with the same functionality? |
| 05:14.07 | dshap | they are SOL? |
| 05:14.13 | drmessano | No dshap, its called ONE OF THE BENEFITS of RUNNING A BUSINESS that involved TELEPHONY |
| 05:14.14 | dshap | *I* am SOL? :-\ |
| 05:14.15 | carrar | SS7 over ICMP, encapsulated into xmodem over serial/rs232 |
| 05:15.15 | drmessano | lol |
| 05:16.45 | drmessano | carrar: I found a project that actually beat me |
| 05:17.52 | dshap | damn |
| 05:18.03 | dshap | so if RDNIS is out of the question |
| 05:18.29 | dshap | are there any other options for implementing something like Youmail on my own on a smaller scale? |
| 05:18.31 | drmessano | Skype may be a good alternative |
| 05:18.49 | *** join/#asterisk apeiron (n=Chris@c-76-124-252-61.hsd1.pa.comcast.net) |
| 05:18.54 | drmessano | or, you can try Youmail.. it sounds a lot like what youre trying to do |
| 05:19.22 | drmessano | http://www.youmail.com/home/index.do |
| 05:19.34 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 05:19.49 | dshap | i just found a forum post where some guy said the SIP headers that bandwidth.com provides have some of the RDNIS info in it |
| 05:19.59 | dshap | u think that's probably bullshit? |
| 05:20.31 | drmessano | Could very well find a SIP provider that offers it |
| 05:21.35 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 05:21.56 | dshap | this blows haha...this basically puts a lid on the whole reason i wanted to learn how to use asterisk in the first place |
| 05:22.19 | dshap | i was so stoked when i got a stupid dialplan to work earlier today |
| 05:22.31 | dshap | where u could call in and dial extensions to hear different audio files |
| 05:22.35 | dshap | it took me so damn long to get that far |
| 05:22.37 | dshap | hah |
| 05:23.55 | drmessano | You know what you could do |
| 05:24.05 | dshap | wat |
| 05:25.08 | drmessano | Well.. Youre thinking of using the CallerID or RDNIS info as your "switch" where the call is directed to.. what if you go sign up for an IPKALL number and route all calls on it direct to VM |
| 05:25.23 | drmessano | and if you want someone else set up, get them to get an IPKALL number |
| 05:27.30 | dshap | not sure how that solves my problem |
| 05:27.38 | drmessano | ..... |
| 05:27.53 | drmessano | Ok, lets go over this again |
| 05:28.12 | drmessano | Youre thinking ONE number.. CID of the phone forwarding ---> Picks the mailbox |
| 05:28.13 | dshap | i assure you i am paying full attention and am not stupid |
| 05:28.14 | dshap | haha |
| 05:28.21 | dshap | right |
| 05:28.21 | drmessano | Right? |
| 05:28.23 | dshap | yes |
| 05:28.23 | drmessano | ok |
| 05:28.35 | dshap | ok |
| 05:28.36 | dshap | im dumb |
| 05:28.37 | dshap | sorry |
| 05:28.37 | dshap | yes |
| 05:28.39 | dshap | got it |
| 05:28.45 | dshap | haha |
| 05:28.48 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-214-193.lns10.mel4.internode.on.net) |
| 05:28.52 | drmessano | BUNCH OF Numbers |
| 05:28.54 | drmessano | Yeah |
| 05:28.57 | dshap | so these IPKALL numbers |
| 05:29.00 | dshap | i can get a ton of them |
| 05:29.01 | dshap | for free? |
| 05:29.44 | drmessano | Just gotta sign up with different addresses.. unless you have some real plan for bulk calls and you can probably email them and work with em |
| 05:30.24 | dshap | and i guess if my plan takes off and gets to the point where i can't keep adding IPKALL numbers |
| 05:30.37 | drmessano | If it takes off, you wont be using IPKALL |
| 05:30.37 | dshap | at that point it's probably worth the investment in the RDNIS service |
| 05:30.40 | dshap | right |
| 05:30.45 | dshap | hm |
| 05:30.46 | drmessano | You'll be buying in bulk |
| 05:31.01 | drmessano | Blocks of 100, 500, 1000 numbers |
| 05:31.08 | drmessano | or RDNIS |
| 05:31.16 | dshap | so i can use these IPKALL numbers with my current SIP termination trunk |
| 05:31.17 | drmessano | However youre gonna do it |
| 05:31.27 | drmessano | No, you direct the calls right to your box |
| 05:31.52 | dshap | aren't the IPKALL numbers PSTN DID's? |
| 05:32.03 | drmessano | They ask you for an "extension" and a proxy address.. which turns into the number@yourbox the call is sent to |
| 05:32.17 | drmessano | Yes, they are PSTN DIDs |
| 05:32.25 | dshap | so the number@mybox |
| 05:32.46 | dshap | wait a sec sorry |
| 05:32.50 | dshap | confused again |
| 05:32.55 | drmessano | gah |
| 05:33.01 | dshap | i thought PSTN DID --> my box is called VoIP trunking |
| 05:33.08 | drmessano | Ok, the call comes in on the telephone wires thingo |
| 05:33.08 | dshap | that's what i'm paying voip.ms to do for me |
| 05:33.22 | dshap | oh |
| 05:33.23 | dshap | orry |
| 05:33.24 | dshap | sorry |
| 05:33.24 | dshap | hahaha |
| 05:33.35 | dshap | i should have mentioned that my box doesn't have an analog input |
| 05:33.38 | dshap | it's pure VoIP |
| 05:33.42 | drmessano | So? |
| 05:34.08 | dshap | is IPKALL essentially a free SIP origination service? |
| 05:34.09 | drmessano | We're not talking about using Analog |
| 05:34.14 | drmessano | Yes |
| 05:34.19 | dshap | with a free DID? |
| 05:34.40 | dshap | why didn't i know about this before |
| 05:34.40 | dshap | wow |
| 05:34.41 | drmessano | No, its origination with no DID.. you call it using app_esp |
| 05:34.44 | dshap | oh |
| 05:34.55 | drmessano | ~cluebat |
| 05:34.56 | infobot | *WHACK* *WHACK* *WHACK* |
| 05:35.05 | dshap | so what # would someone put their cell phone to forward to then? |
| 05:35.12 | drmessano | Origination without a DID is a cake without an oven |
| 05:35.25 | dshap | right |
| 05:35.30 | drmessano | If you tell me you dont like cake, I am gonna slap you |
| 05:35.38 | dshap | okay so you were being sarcastic |
| 05:35.42 | dshap | SIP origination, free DID |
| 05:35.44 | dshap | all free |
| 05:35.53 | dshap | i love cake |
| 05:35.56 | drmessano | Sorry, didnt put enough smileys :) |
| 05:35.59 | drmessano | Me too :) |
| 05:36.01 | drmessano | yep :) |
| 05:36.01 | dshap | haha |
| 05:36.12 | dshap | ugh so why am i paying voip.ms for Origination! |
| 05:36.31 | drmessano | Because you dont want a washington state phone number |
| 05:36.31 | dshap | they do give me unlimited channels... |
| 05:36.39 | dshap | i don't care what the phone number is |
| 05:36.56 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 05:37.03 | dshap | ok here we go |
| 05:37.10 | dshap | a user of my service gets an IPKALL number |
| 05:37.21 | dshap | they forward their missed calls to the IPKALL number which goes to my asterisk box |
| 05:37.28 | dshap | what if 2 ppl call them at the same time |
| 05:37.37 | dshap | does the IPKALL service offer multiple channels? |
| 05:37.44 | drmessano | Nope |
| 05:37.55 | dshap | so people who call popular people are gonna get a busy tone |
| 05:37.55 | drmessano | I dont think youre gonna forward more than 1 call at a time |
| 05:38.09 | dshap | probably not often |
| 05:38.16 | drmessano | No |
| 05:38.20 | dshap | i'm thinkin large-scale here, drmessano |
| 05:38.22 | dshap | large scale |
| 05:38.24 | drmessano | This wasnt a "usage" statement |
| 05:38.39 | drmessano | Listen, youre the newb here |
| 05:38.46 | *** join/#asterisk kapr (n=IceChat7@116.71.222.43) |
| 05:38.48 | dshap | true |
| 05:38.50 | drmessano | [01:38] <drmessano> I dont think youre gonna forward more than 1 call at a time |
| 05:38.53 | drmessano | As in |
| 05:39.21 | drmessano | Your cell provider is not going to forward 5 simultaneous calls from your cell to somewhere |
| 05:39.25 | carrar | large-scale using IPKALL? |
| 05:39.26 | dshap | ohhhh |
| 05:39.27 | carrar | wtf |
| 05:39.29 | dshap | gotcha |
| 05:39.35 | drmessano | Fuckin DUH |
| 05:39.42 | dshap | so right now if you called my cell, and got my voicemail greeting |
| 05:39.47 | dshap | and someone else tried to call my cell |
| 05:39.49 | dshap | and i didnt pick up |
| 05:39.55 | drmessano | Right now |
| 05:39.55 | dshap | they'd probably not get my voicemail greeting? |
| 05:39.58 | drmessano | If I call your cell |
| 05:40.03 | drmessano | and it forwards the call out |
| 05:40.06 | drmessano | Thats probably it |
| 05:40.30 | dshap | gotcha |
| 05:41.03 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-214-193.lns10.mel4.internode.on.net) |
| 05:41.18 | drmessano | Im well aware you want more than 1 concurrent call coming into your little project |
| 05:41.28 | dshap | hah |
| 05:42.00 | dshap | well there wouldn't be any problem having multiple IPKALL numbers all connecting to my box at the same time |
| 05:42.01 | dshap | would there? |
| 05:42.04 | dshap | different IPKALL numbers |
| 05:42.13 | drmessano | No |
| 05:42.18 | dshap | k |
| 05:42.22 | carrar | really now, whats the goal |
| 05:42.34 | dshap | i can't fully say hahah |
| 05:42.36 | drmessano | carrar: Next Vonage |
| 05:42.37 | dshap | it's a secret :-p |
| 05:42.46 | carrar | yeah |
| 05:42.50 | carrar | everything here is |
| 05:43.15 | carrar | voip is 31337 k-r4d s3cr37 |
| 05:43.17 | dshap | it will basically be a lot of work for a feature that i'd love to have with my voicemail system |
| 05:43.21 | dshap | that i'm pretty sure no one else offers |
| 05:43.29 | dshap | which i'm probably wrong about |
| 05:43.32 | drmessano | carrar: Next week after he gets his first call terminated from IPKALL, he'll be asking about billing and how to start a "ISP" "No, I mean a SIP ISP" |
| 05:43.33 | dshap | but at the very least |
| 05:43.41 | dshap | i'll be able to use it personally if it fails |
| 05:43.42 | carrar | heh |
| 05:43.45 | dshap | and i know i'll get good use out of it |
| 05:43.57 | dshap | haha |
| 05:44.02 | dshap | this is for every-day cell phoners |
| 05:44.43 | drmessano | dshap: if you give me 10 mins, I can pull a list of the "you" of every week since Feb 2006 |
| 05:44.52 | dshap | hahahaha |
| 05:44.58 | dshap | i don't doubt it for a second |
| 05:44.59 | drmessano | Their nicks are all a blur |
| 05:45.01 | dshap | but hey |
| 05:45.08 | dshap | you never know if you don't try |
| 05:45.09 | dshap | right? |
| 05:45.17 | drmessano | AH |
| 05:45.21 | drmessano | Thank you |
| 05:45.23 | carrar | you can ask if it's been done before |
| 05:45.24 | dshap | hahaha |
| 05:45.29 | drmessano | Thats the term I needed to search with |
| 05:45.33 | drmessano | They ALL said that one |
| 05:45.34 | drmessano | BRB |
| 05:45.36 | carrar | there are a lot of wheels talked about in this channel |
| 05:45.50 | dshap | my idea is so simple that i'm sure it's been thought of before |
| 05:45.53 | dshap | probably done before |
| 05:45.59 | dshap | but |
| 05:45.59 | carrar | probably |
| 05:46.12 | carrar | probably already part of asterisk |
| 05:46.13 | dshap | i've looked hard online to try to find something i could use personally |
| 05:46.13 | drmessano | I hear YouMail offers a similar service |
| 05:46.17 | dshap | ahhaha |
| 05:46.19 | dshap | okay okay |
| 05:46.27 | dshap | whatever |
| 05:46.37 | dshap | i'm just another every-week dude since feb '06 |
| 05:46.49 | drmessano | [00:42] <dshap> is anyone here familiar with Youmail.com ?? |
| 05:46.49 | drmessano | [00:42] <dshap> i have a quick question about their voicemail service...I am trying to implement something similar on my asterisk box |
| 05:46.57 | drmessano | So yes, its been done |
| 05:47.04 | dshap | ok |
| 05:47.08 | dshap | well whatever |
| 05:47.15 | carrar | What evah! |
| 05:47.19 | carrar | oh no you don't |
| 05:47.23 | dshap | i may be able to get independent study credit for college if i can make a legit project out of this |
| 05:47.26 | drmessano | apparently that dshap guy found some site similar to your idea |
| 05:47.33 | dshap | if it turns into a business then more power to me, if not, then fuckit |
| 05:47.50 | carrar | woah their family guy |
| 05:47.53 | drmessano | Go for it.. Fight the future.. or some line from hackers |
| 05:48.31 | carrar | on roller blades |
| 05:48.35 | carrar | and camo painted laptops |
| 05:48.49 | drmessano | HACK THE GIBSON, MAH FRIEND |
| 05:49.14 | dshap | how do these IPKALL people make money |
| 05:49.15 | dshap | doing what they do |
| 05:50.11 | drmessano | They listen in on calls, sell the good ones to prankcalls.com |
| 05:50.13 | carrar | they got you |
| 05:50.18 | dshap | lol |
| 05:51.14 | drmessano | Reminds me of that one provider I had |
| 05:51.18 | drmessano | The calls sucked |
| 05:51.22 | carrar | back in bandcamp? |
| 05:51.25 | drmessano | Wasnt sure why I stayed with them |
| 05:51.34 | drmessano | Kept telling myself to leave |
| 05:51.41 | drmessano | To go find another provider |
| 05:51.52 | drmessano | But then they started kissing me right behind the ear |
| 05:51.55 | drmessano | ..... |
| 05:52.10 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 05:52.41 | drmessano | So I just loaded up that account with another $50 |
| 05:52.53 | carrar | You could just get 30 magicjacks on your USB port |
| 05:53.00 | *** join/#asterisk MikeJ_ (n=MikeJ@freeswitch/developer/mikej) |
| 05:53.15 | dshap | so the "SIP Phone number" on IPCALL |
| 05:53.18 | dshap | IPKALL* |
| 05:53.29 | dshap | is that the extension? |
| 05:53.45 | dshap | and the proxy is my IP address? |
| 05:53.48 | drmessano | Yeah |
| 05:53.56 | drmessano | Yep |
| 05:54.16 | dshap | so i'd probably set the SIP phone number to my voip.ms DID |
| 05:54.22 | dshap | because that's the extension i use on incoming calls |
| 05:54.53 | drmessano | right on |
| 05:54.57 | dshap | well for my voicemail system it would have to be unique |
| 05:55.17 | drmessano | Yes... at least 20 digits |
| 05:55.17 | dshap | because each IPKALL number needs to ring an extension that corresponds to a certain mailbox |
| 05:56.14 | drmessano | yeah |
| 05:56.19 | drmessano | 25 digits |
| 05:56.25 | dshap | ? |
| 05:56.36 | drmessano | You want every mailbox to be unique |
| 05:56.44 | dshap | oh hahaa |
| 05:56.48 | drmessano | I would use random 25 digit numbers |
| 05:56.57 | drmessano | i'm thinkin large-scale here, dshap |
| 05:56.59 | dshap | right |
| 05:57.01 | drmessano | large-scale |
| 05:57.06 | dshap | i don't think 24 digits provides enough combinations |
| 05:57.10 | dshap | for the scale i'm talking here |
| 05:57.15 | carrar | definately |
| 05:57.32 | carrar | might tack on a additional 3 |
| 05:57.44 | carrar | 3+25 |
| 05:57.46 | drmessano | or use hex |
| 05:57.47 | dshap | i bet the IPKALL people would be sketched out if they got even just hundreds of applications for numbers to my box |
| 05:57.57 | vi390 | hi, can someone tell, what goes wrong, when I get the error. "Call from '12345' to extension '12345' rejected because extension not found." : Where , I have in sip.conf something that catches incoming with context=reff ; and also have a part in extension.conf with [reff]. Does the error mean, that it did not read the "context=reff" part ? |
| 05:58.13 | dshap | oooo |
| 05:58.16 | dshap | let me try to answer this |
| 05:58.19 | dshap | i had this problem earlier today |
| 05:58.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:58.28 | carrar | I'm kinda sketched out now |
| 05:58.37 | dshap | vi390: are you using the "s" extension in your reff context? |
| 05:58.47 | vi390 | dshap: yes |
| 05:58.52 | dshap | and let me guess |
| 05:58.56 | dshap | you're calling in from a PSTN line |
| 05:59.00 | dshap | using a SIP VoIP trunk |
| 05:59.07 | vi390 | dshap: yes :) |
| 05:59.07 | dshap | so like a cell phone number or a land line |
| 05:59.08 | dshap | righ? |
| 05:59.10 | dshap | hah |
| 05:59.11 | dshap | okay |
| 05:59.13 | carrar | match on 12345' rejected |
| 05:59.14 | carrar | <PROTECTED> |
| 05:59.15 | carrar | err |
| 05:59.19 | carrar | match on 12345 |
| 05:59.22 | carrar | in that context |
| 05:59.44 | carrar | exten => 12345,1,answer |
| 05:59.45 | dshap | you need to change "s" to the actual extension that is being ringe |
| 05:59.53 | dshap | what he said ^ |
| 06:00.00 | vi390 | ooh, okey .. thanks |
| 06:00.10 | carrar | exten => 12345,n,playback(stopcallingme) |
| 06:00.13 | carrar | exten => 12345,n,hangup |
| 06:00.25 | vi390 | I thought that context=reff binds it ;) |
| 06:00.26 | drmessano | Not sure why 12345 would be calling 12345.. Sounds like some crazy shit dshap would do with a couple skype phones, a vonage adaper, asterisk, and a sony walkman |
| 06:00.26 | vi390 | ok thanks |
| 06:00.39 | dshap | hahaah |
| 06:00.53 | carrar | maybe |
| 06:00.55 | vi390 | drmessano: ;) |
| 06:00.56 | drmessano | Cant tell you what its for, though |
| 06:00.57 | carrar | exten => 12345/12345,n,playback(stopcallingme) |
| 06:01.02 | drmessano | HA |
| 06:01.05 | carrar | heh |
| 06:01.13 | drmessano | ha |
| 06:01.14 | drmessano | shit |
| 06:01.16 | dshap | drmessano, if you're still in this channel when my service opens up for private beta, you're in |
| 06:01.25 | carrar | OH SWEET |
| 06:01.27 | drmessano | ~savemoney |
| 06:01.27 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 06:01.29 | carrar | I can't wait |
| 06:01.32 | dshap | it will be freeeeeee |
| 06:01.35 | dshap | maybe |
| 06:01.57 | dshap | freemium |
| 06:02.03 | drmessano | I have like no feeling in my left pinky and ring finger, cant find the damn shift key to ~ |
| 06:02.18 | carrar | Almost as good as freeinternet.com |
| 06:02.18 | drmessano | my infobot foo is weakened |
| 06:04.36 | drmessano | I signed up for free internet once |
| 06:04.45 | drmessano | But I couldnt get online to use it |
| 06:04.54 | carrar | heh |
| 06:05.07 | drmessano | breaks out his BlueLight CDs |
| 06:06.15 | drmessano | Netzero with the java banner |
| 06:07.35 | dshap | so i don't get it |
| 06:07.42 | dshap | you guys just chill in the asterisk IRC channel in your spare time? |
| 06:07.49 | dshap | or you're paid to do this? (lol doubt it) |
| 06:07.50 | dshap | or what |
| 06:08.01 | drmessano | Why do you doubt it? |
| 06:08.27 | dshap | haha only because of the way you treat some of the people who seek help in here lol |
| 06:08.29 | dshap | <----------- |
| 06:08.35 | drmessano | Because we have no choice but to bow to your intellectual superiority? |
| 06:08.38 | carrar | we get paid |
| 06:08.46 | carrar | $1,000,000 |
| 06:08.49 | dshap | hahaha |
| 06:09.28 | drmessano | Yeah.. We get paid... and every now and then, we get to beat a newb around for fun.. Tie little pointy things to them and roll em around on the carpet, etc |
| 06:10.02 | dshap | i feel like you'd have an "@" in front of your name if you got paid |
| 06:10.14 | kapr | anyone has a prefered USB ZAP card? |
| 06:10.59 | drmessano | dshap: You're not much of a 5th wall sort of guy, are you? |
| 06:11.14 | dshap | i'm not familiar with that expression |
| 06:12.13 | drmessano | perhaps I can help you a little with it.. |
| 06:12.18 | drmessano | carrar: get the tools |
| 06:12.42 | drmessano | I mean ummm.. |
| 06:13.59 | drmessano | Damn IMAP toolkit |
| 06:14.03 | carrar | heh |
| 06:14.04 | drmessano | Good riddance |
| 06:14.44 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:15.16 | drmessano | Compiling the shared librares for the IMAP toolkit C-client under RHEL/CentOS.. |
| 06:15.18 | drmessano | 6 words |
| 06:15.27 | drmessano | You cant get there from here |
| 06:15.28 | carrar | --shared |
| 06:15.33 | carrar | thats 8 |
| 06:15.37 | carrar | shared |
| 06:15.48 | dshap | if i have CentOS and want to learn about it |
| 06:15.51 | dshap | could i read a book on RHEL? |
| 06:15.56 | dshap | are they that similar? |
| 06:16.00 | carrar | YES YOU CAN |
| 06:16.06 | dshap | sweet |
| 06:17.42 | drmessano | carrar: What I read from the devs of the toolkit was that they dont offer that, although others have patched for it, because it saves little memory and usually makes things unstable and generates bug reports needlessly |
| 06:18.47 | drmessano | I need to generate a bug report.. "Shit shouldnt be this hard" |
| 06:18.47 | carrar | make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC" |
| 06:19.05 | carrar | that was imap-2007a |
| 06:20.00 | carrar | 100 bucks says you can guess what I am googling |
| 06:20.03 | carrar | can't |
| 06:20.10 | drmessano | Yeah, does a quick make and created the static libraries |
| 06:20.22 | carrar | thats all you need |
| 06:21.04 | carrar | then in asterisk |
| 06:21.12 | carrar | ./configure --with-imap=/home/source/imap-2007a |
| 06:21.25 | carrar | or wheere everyou have the source |
| 06:21.35 | drmessano | Yeah, but then you need to symlink or rename that directory to imap-2004g or use the --with-imap every time you configure |
| 06:21.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 06:22.18 | carrar | how often do you recompile? |
| 06:22.25 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 06:22.26 | drmessano | Often |
| 06:22.32 | carrar | everytime you restart? ;) |
| 06:22.36 | carrar | heh |
| 06:22.37 | drmessano | lol |
| 06:22.41 | drmessano | Every few weeks |
| 06:22.41 | carrar | sip reload |
| 06:22.42 | carrar | WAIT |
| 06:22.45 | carrar | I need to recompile! |
| 06:22.53 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 06:23.00 | drmessano | or more often if I am working on something |
| 06:23.12 | carrar | scripts rulZ |
| 06:23.21 | drmessano | Id rather have it be a no brainer.. and install the libraries to some known path |
| 06:23.36 | carrar | create your own wiki, then cut and paste |
| 06:23.51 | drmessano | i do enough of that already |
| 06:23.58 | drmessano | Trying to make things easier |
| 06:24.05 | dshap | do you guys use AGI? |
| 06:24.12 | carrar | HELLZAYEAH |
| 06:24.16 | drmessano | Like installing IMAP right the first time to where I have one less thing |
| 06:24.22 | dshap | with PHP? |
| 06:24.23 | *** part/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 06:24.28 | carrar | PHP is for web |
| 06:24.41 | dshap | you can use it for AGI scripting as well |
| 06:24.48 | carrar | You can use cobol too |
| 06:25.05 | dshap | hm |
| 06:25.08 | drmessano | I found the imap 2007e in EPEL.. |
| 06:25.08 | carrar | But just because you can, doesn't mean you should |
| 06:25.28 | dshap | welli have experience with PHP so that is why i was asking |
| 06:25.39 | dshap | but yes, my experience with PHP is for web apps |
| 06:25.40 | drmessano | I wrote PHP |
| 06:25.56 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 06:26.05 | carrar | drmessano, yeah I install the imap shit back in March 2008 |
| 06:26.20 | carrar | I'm sure things have changed a little |
| 06:27.08 | carrar | I like using perl for AGI personally, and use PHP for my web apps |
| 06:27.11 | drmessano | I much prefer source, but its my understanding theres a good 8 to 10 patches applied to the current RPM in EPEL that work around issues on RH based platforms |
| 06:27.22 | drmessano | So it may be a better way to go |
| 06:27.46 | drmessano | and much easier than this crap.. |
| 06:27.47 | carrar | I don't use RPM's |
| 06:27.51 | carrar | if I can avoid it |
| 06:28.03 | drmessano | Well, same here |
| 06:30.26 | *** join/#asterisk lou_gr (n=lou@static062038221130.dsl.hol.gr) |
| 06:31.16 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 06:31.49 | dshap | let's say Youmail had 1 million users |
| 06:32.06 | dshap | how many channels do you think they need for that? |
| 06:32.13 | dshap | people leaving voicemails/checking voicemails |
| 06:32.23 | dshap | actually |
| 06:32.26 | dshap | what if they're just leaving voicemails |
| 06:32.37 | dshap | and they just check online so they aren't on the channels checking their voicemails |
| 06:32.43 | carrar | Is that 1 million in the US? |
| 06:32.46 | dshap | yes |
| 06:32.54 | carrar | 1 million in Texas? |
| 06:33.03 | dshap | 1 million anywhere in the US |
| 06:33.09 | dshap | just a random sample of mobile users |
| 06:33.18 | dshap | how many people do you think are receiving voicemails |
| 06:33.19 | dshap | at the same time |
| 06:33.21 | dshap | max |
| 06:33.26 | dshap | obviously a crazy guessi |
| 06:33.28 | dshap | guess* |
| 06:33.31 | dshap | i'm just curious what you guys think |
| 06:33.42 | carrar | their datacenter is in once place for the whole country? |
| 06:33.52 | carrar | not spreadout? |
| 06:34.03 | dshap | i don't know |
| 06:34.05 | dshap | but what if it was |
| 06:34.27 | dshap | they give me a 714 number for their voicemail system which is probably because my own cell phone number is based in southern california near 714 |
| 06:34.31 | dshap | but then again.... |
| 06:34.35 | dshap | it says their offices are in southern california |
| 06:34.39 | dshap | so i don't know for sure |
| 06:34.43 | dshap | they may just be running 1 data center |
| 06:34.46 | carrar | I would say if they have 1 million vm customers then they need 123,456 concurrent calls checking vm |
| 06:34.50 | dshap | seems like not that big of a company |
| 06:34.53 | carrar | heh |
| 06:35.03 | carrar | OR |
| 06:35.10 | carrar | I would say if they have 1 million vm customers then they need 12,345 concurrent calls checking vm |
| 06:35.27 | dshap | seriously |
| 06:35.30 | carrar | maybe 54,321 lines |
| 06:35.42 | carrar | thats probably more realistic |
| 06:35.50 | carrar | but even then I think thats high |
| 06:36.00 | drmessano | 32,768 |
| 06:36.17 | carrar | 86,753.09 |
| 06:40.49 | dshap | what's a ballpark cost to get a PRI? |
| 06:41.07 | carrar | depends |
| 06:41.17 | carrar | could be 400 |
| 06:41.19 | carrar | could be 900 |
| 06:41.23 | carrar | or more |
| 06:41.24 | dshap | per month? |
| 06:41.26 | carrar | yes |
| 06:41.30 | dshap | jesus |
| 06:41.37 | dshap | and that's only a handful of concurrent calls, right? |
| 06:41.43 | carrar | 23 |
| 06:42.41 | carrar | Here is washington they are terminating your RateCenter and TOLLFREE calls for free |
| 06:42.47 | carrar | if you have a PRI |
| 06:43.16 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:43.44 | *** join/#asterisk xrmx__ (n=rm@host59-183-dynamic.6-79-r.retail.telecomitalia.it) |
| 06:44.12 | dshap | i gotta find a SIP provider that has RDNIS |
| 06:44.14 | dshap | that is my only hope |
| 06:44.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:44.34 | carrar | Most of the larger SIP carriers accept diversion headers |
| 06:44.56 | dshap | would that mean that ${CALLERID(rdnis)} would contain the phone number that i want? |
| 06:45.17 | carrar | AKA user calls your desk, you forward to your cell using the orignal callers ANI |
| 06:45.39 | carrar | even if the caller is out of state |
| 06:45.57 | carrar | cell phone sees the outof state caller id number |
| 06:46.06 | dshap | that is NOT what i want though |
| 06:46.09 | carrar | oh |
| 06:46.11 | carrar | what do you want |
| 06:46.18 | carrar | rock and roll? |
| 06:46.31 | dshap | i want the cell phone (i.e. my asterisk server) to see the desk's phone number |
| 06:46.41 | carrar | heh |
| 06:46.44 | carrar | thats simple |
| 06:46.51 | carrar | just re0write the call caller id |
| 06:46.54 | carrar | re-write |
| 06:47.42 | carrar | outside call -> * -> cell phone (cell phone sees * did) |
| 06:47.43 | dshap | the specific implementation of this i explained to drmessano earlier and he said it wasn't possible |
| 06:47.58 | dshap | no sry i didn't fully explain to you |
| 06:48.08 | dshap | outside call A --> outsidecall B --> * |
| 06:48.12 | dshap | need * to see B |
| 06:48.23 | carrar | Why are you forwarding? |
| 06:48.29 | carrar | why not go directly to * |
| 06:48.30 | dshap | it's voicemail |
| 06:48.36 | dshap | outside call B is actually a cell phone |
| 06:48.41 | dshap | set to forward missed calls to some number |
| 06:48.47 | carrar | do your own vm |
| 06:49.08 | carrar | then you can |
| 06:49.19 | carrar | Did you know Asterisk can do voicemail! |
| 06:49.23 | dshap | i want to supportmany users |
| 06:49.25 | dshap | yes i'm fully aware |
| 06:49.25 | carrar | It's HIP |
| 06:49.27 | dshap | trust me |
| 06:49.35 | dshap | i know all the features |
| 06:49.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 06:49.46 | carrar | Who isy our vm carrier? |
| 06:49.51 | carrar | you need to ask them |
| 06:49.51 | dshap | i use Youmail.com |
| 06:50.00 | carrar | they will laugh |
| 06:50.19 | dshap | my project is different |
| 06:50.24 | dshap | i need that functionality for it to work |
| 06:50.28 | dshap | the A --> B --> * |
| 06:50.29 | carrar | do your own vm |
| 06:50.33 | dshap | what do you mean by that |
| 06:50.43 | carrar | You make asterisk be the "B" |
| 06:51.07 | carrar | A -> * SPLIT (*->B) (*-> cell) |
| 06:51.42 | dshap | this doesn't work for what i'm trying to do |
| 06:51.48 | dshap | in my case A is anyone i know who is trying to call me |
| 06:51.58 | dshap | they will try to reach me on my cell phone |
| 06:52.00 | carrar | Perhaps #ineedmorecrack has the answers |
| 06:52.00 | dshap | which is B |
| 06:52.21 | carrar | B should be You (*) |
| 06:52.22 | *** part/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 06:52.30 | carrar | Your DID |
| 06:52.34 | carrar | do it right |
| 06:52.49 | dshap | when you want to leave a voicemail for your friend |
| 06:52.55 | dshap | you call your friend's cell phone number |
| 06:52.57 | dshap | he doesn't pick up |
| 06:53.02 | dshap | then you are forwarded to his voicemail system |
| 06:53.11 | dshap | right? |
| 06:53.21 | carrar | I don't leave voicemails like that but sure |
| 06:53.49 | dshap | his voicemail system somehow determines that you are calling for HIM and not someone else on the same system |
| 06:54.03 | carrar | I call his * box, it calls his cell, his home, it work |
| 06:54.07 | carrar | leaves voicemail on * |
| 06:54.19 | carrar | he gets the email |
| 06:54.20 | carrar | page |
| 06:54.24 | carrar | and vm |
| 06:54.32 | dshap | ok that sounds sweet, but it does not work for my project |
| 06:54.37 | dshap | i understand what you are saying completely |
| 06:54.42 | carrar | Your project is in ERROR |
| 06:54.56 | dshap | have you ever thought you had an amazing idea to make something |
| 06:54.58 | dshap | that no1 else has done |
| 06:55.02 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 06:55.07 | dshap | but the problem is you will need help in building it |
| 06:55.11 | dshap | but if u tell other people the iea |
| 06:55.12 | dshap | idea* |
| 06:55.12 | carrar | Ever thought their might be a reason for that |
| 06:55.14 | dshap | they might steal it |
| 06:55.34 | carrar | talk to the VM carrier |
| 06:56.21 | dshap | they're not gonna tell me how their system is built |
| 06:56.33 | carrar | they will tell you if they can do a feature that you need |
| 06:56.38 | carrar | aka forward with original ANI |
| 06:57.09 | carrar | 9% change they will |
| 06:57.09 | dshap | ohhh |
| 06:57.13 | carrar | chance |
| 06:57.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:57.22 | dshap | so you're saying like a different code on my cell phone |
| 06:57.28 | carrar | If they do, they will be using the diversion header with their SIP carrier |
| 06:57.37 | dshap | ok |
| 06:57.51 | dshap | speaking of SIP carriers |
| 06:58.06 | dshap | right now im paying $1.00 per month for a DID |
| 06:58.16 | dshap | and it's $0.01 cents per minute both origination and termination |
| 06:58.23 | dshap | with a 6 second billing cycle |
| 06:58.28 | dshap | err billing increment |
| 06:58.31 | dshap | and unlimited channels |
| 06:58.36 | dshap | does that sound like i am getting a good deal? |
| 06:58.52 | carrar | whats your bill |
| 06:59.02 | carrar | depends on your usage |
| 06:59.09 | dshap | i mean...i just bought it last week |
| 06:59.17 | dshap | i paid $25 into an account and then as i use it they just deduct from it |
| 06:59.54 | carrar | probably fine for then ew occasional user |
| 06:59.55 | dshap | i dont have high usage, but eventually i might |
| 06:59.57 | dshap | right |
| 06:59.57 | carrar | new |
| 07:00.02 | dshap | k |
| 07:01.14 | *** join/#asterisk SebastianS (n=schu@212.5.200.111) |
| 07:03.45 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:08.13 | *** join/#asterisk \void\ (n=void@fer227.internetdsl.tpnet.pl) |
| 07:09.04 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 07:09.51 | dshap | alright im outta here - thanks carrar and drmessano for putting up with my shit |
| 07:10.09 | dshap | i'll likely be back as i attempt the impossible task that is my little project |
| 07:10.29 | dshap | if my little project ever makes it big, i'll come back here and if you guys are still around i'll give you a token of my appreciation |
| 07:10.35 | dshap | peace late. |
| 07:13.06 | *** join/#asterisk quintana (n=sylvain@aghnar.doowan.net) |
| 07:20.53 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 07:28.05 | vi390 | hi again, its weird. still having the "extension not found" problem. Just cant find out WHY the extension can not be found. >sip show users, shows it all correct (have set up a test extension which reads "[waitbeep]exten => 100,n,Answer() ..." and my SIP register gets redirected to "100" now It says "Call from '<SIPID>' to extension 'waitbeep' rejected because extension not found." - Why is the extension not found. Its right there. And >dialplan show, sh |
| 07:28.05 | vi390 | ows "'100' => 103. Answer()" what seems to be correct. But still the error ?? |
| 07:29.02 | Pan3D | vi390: pastebin the sip debug and your config |
| 07:29.11 | vi390 | ok |
| 07:30.21 | *** join/#asterisk mikkel (n=mikkel@130.226.39.220) |
| 07:32.21 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:33.35 | *** join/#asterisk bennykill (n=yeah@212.90.131.28) |
| 07:33.43 | bennykill | hi |
| 07:34.03 | bennykill | was geht? |
| 07:34.50 | bennykill | need help with my fritzcard pci :( |
| 07:35.47 | bennykill | we have to make a school project and want to connect an asterisk server with isdn |
| 07:35.58 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:41.42 | vi390 | extension not found problem => http://pastebin.com/d2a834a17 |
| 07:42.02 | vi390 | .. why is there the error "extension not found" |
| 07:43.47 | vi390 | there is [waitbeep] in extensions.conf, and >sip show users gives out the correct DEf.Context = waitbeep for the incoming user |
| 07:45.47 | kaldemar | vi390: you have no priority 1 for exten 100. |
| 07:46.25 | vi390 | kaldemar: ok! I thought I can start with "n" |
| 07:47.38 | vi390 | kaldemar: thx, thats it |
| 07:48.13 | bennykill | can someone help me with the fcpci driver? |
| 07:51.17 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
| 07:56.54 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:05.29 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 08:08.59 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 08:10.31 | *** join/#asterisk pluesch0r (n=pluesch0@91.186.157.67) |
| 08:11.04 | pluesch0r | hi everybody. how do i reset a voicemail password? setting it to 0000 in voicemail.conf and then reloading asterisk doesn't work. |
| 08:13.15 | styelz | maybe its a dtmf issue instead |
| 08:13.22 | *** join/#asterisk unasi7 (n=unasi7@94.23.3.124) |
| 08:14.00 | arekm | hm, I'm having problem with variable inheritance in asterisk 1.4. I set Set("SIP/arekm-twinkle-082129d0", "__REC_NOTIFIED=1"), later there is few Goto()s and ${__REC_NOTIFIED} ends being "0", what could be the reason? |
| 08:14.26 | pluesch0r | styelz: hm. could be. i recently upgraded asterisk. how to debug that one? |
| 08:14.53 | arekm | afaik __ means "inherit" so it should stay "1" |
| 08:15.07 | styelz | set verbose in console to 3 and watch the input it receives |
| 08:16.48 | kaldemar | arekm: show a cli output |
| 08:16.54 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:17.35 | pluesch0r | styelz: nah. says 'Incorrect password '0000' for user '100' (context = default)' |
| 08:17.38 | arekm | kaldemar: http://pld.pastebin.com/f10f2972c, there is __REC and __REC_NOTIFIED with the same problem |
| 08:18.18 | bennykill | can someone help me with fcpci module? |
| 08:18.59 | arekm | kaldemar: GotoIf("SIP/arekm-twinkle-082129d0", "0?macro-notifyrecording|s|5") is exten => s,1,GotoIf($["${__REC_NOTIFIED}" = "1"]?macro-notifyrecording,${EXTEN},5) |
| 08:20.21 | kaldemar | arekm: don't use ${__REC_NOTIFIED} but ${REC_NOTIFIED} |
| 08:20.53 | kaldemar | use __ in front of the variable name only when setting the variable. |
| 08:22.58 | arekm | kaldemar: that was it, thanks! |
| 08:23.00 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:27.48 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 08:28.24 | pluesch0r | styelz: got any other idea on how to debug this? |
| 08:30.10 | asim- | im using asterisk 1.6.1.0 up from 1.6.0.1, when i use atxfer (attended transfer) it hang ups the call now. |
| 08:30.11 | asim- | wtf happened? |
| 08:31.27 | jordanl_ | what are the basic trunk settings in sip.conf for a SIP provider through which i want to receive incoming calls only (not using SIP registration, but domain/IP lookup)? |
| 08:31.56 | pluesch0r | jordanl_: i've got that .. |
| 08:31.59 | jordanl_ | would you use type=user and domain=sip.theprovider.com? |
| 08:32.19 | pluesch0r | but without using a sip provider. |
| 08:32.23 | jordanl_ | pluesch0r: you use something like that? |
| 08:33.04 | *** join/#asterisk joobie (n=joobie@210-84-22-83.dyn.iinet.net.au) |
| 08:33.26 | pluesch0r | jordanl_: http://pastie.org/491113 |
| 08:33.36 | pluesch0r | that's what i'm using on my asterisk server to be able to get called. |
| 08:33.45 | pluesch0r | with the sip information inside of DNS. |
| 08:35.21 | jordanl_ | pluesch0r: but any source address can send you calls, right? |
| 08:35.50 | pluesch0r | yes. |
| 08:35.52 | pluesch0r | exactly. |
| 08:35.53 | jordanl_ | as long as the RURI has your.domain part |
| 08:35.58 | jordanl_ | user@your.domain |
| 08:36.32 | pluesch0r | jordanl_: this setup is simply to be able to receive calls. |
| 08:36.55 | jordanl_ | pluesch0r: yes, i know, but i'm asking about the realm part |
| 08:36.59 | pluesch0r | to forward them to an extension, you need to set up credentials for your various sip users. |
| 08:37.12 | pluesch0r | mhm |
| 08:37.19 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.21) |
| 08:38.07 | jordanl_ | can you specify a trusted source domain/IP though? |
| 08:38.31 | pluesch0r | i don't understand the question. |
| 08:38.35 | jordanl_ | so you can receive calls from the trusted source, but reject all other attempts |
| 08:38.58 | pluesch0r | dunno. don't have that in my setup. |
| 08:39.19 | jordanl_ | pluesch0r: thanks for the help though |
| 08:40.59 | kaldemar | jordanl_: define a peer with a static ip or hostname and use the insecure parameter for that peer. |
| 08:42.12 | pluesch0r | hm. what do i need to do to reset the whole voicemail stuff alltogether? |
| 08:42.12 | pluesch0r | there seems to be something completely wrong. |
| 08:43.17 | *** join/#asterisk kerx (n=pmessri@adsl-69-105-59-236.dsl.irvnca.pacbell.net) |
| 08:43.22 | jordanl_ | kaldemar: insecure=invite ? |
| 08:43.28 | xnixan | Hi, what would be the most economical solution (PCI / PCIX, other) to use in 2 ISDN lines with asterisk? |
| 08:44.27 | kaldemar | jordanl_: insecure=port,invite |
| 08:44.31 | kerx | hi, i have two separate queue's, with the same agent member's in them. when an agent log's into a specific queue (ex. Queue1). And a user dials in-to Queue2, the call is still sent to Queue1. Anyone know how it's possible to stop this? |
| 08:45.03 | kerx | err, i messed up. I meant to say when they dial into- Queue2, it still call's the logged in agent of Queue1 |
| 08:46.24 | bennykill | can someone help me with fcpci module? |
| 08:47.26 | jordanl_ | kaldemar: thanks |
| 08:47.52 | bennykill | can someone help me with the fcpci driver? |
| 08:51.16 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 08:51.59 | pluesch0r | nvm, solved it. |
| 08:53.09 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 09:01.24 | Kevin` | any recommendations for stores to buy parts? (ata) |
| 09:03.09 | pluesch0r | how do i set the default language to something other than en? |
| 09:05.17 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
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| 09:23.13 | tzafrir_laptop | xnixan, 2 ISDN ports? |
| 09:23.34 | tzafrir_laptop | or two ISDN lines in a single port? |
| 09:28.16 | pluesch0r | anybody? default language to !en? |
| 09:30.03 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
| 09:34.02 | tzafrir_laptop | pluesch0r, what version of Asterisk? |
| 09:34.21 | tzafrir_laptop | Also: you can also set this in the channel driver configuration (e.g.: sip.conf) |
| 09:34.38 | pluesch0r | tzafrir_laptop: i did set it in the channel driver, it's not working for incoming calls. |
| 09:34.52 | tzafrir_laptop | which channel? |
| 09:35.03 | pluesch0r | tzafrir_laptop: this is asterisk 1.4.21 |
| 09:35.04 | tzafrir_laptop | maybe it is overriden elsewhere? |
| 09:35.15 | pluesch0r | the general channel. |
| 09:35.19 | tzafrir_laptop | by an explicit Set? |
| 09:35.32 | pluesch0r | in sip.conf, i did set language=de |
| 09:36.34 | pluesch0r | that setting works if i'm calling my mailbox, for example. |
| 09:36.48 | pluesch0r | but if i'm calling from external, i still get the english 'this line is busy' message, instead of the german one. |
| 09:39.24 | vi390 | what could be a possible cause of, when I place calls (forexample internal, to test) but can not hear anything, when answering the call (neither in one direction, nor in the other) |
| 09:41.34 | pluesch0r | tzafrir_laptop: any idea? |
| 09:42.49 | tzafrir_laptop | pluesch0r, I'm busy right now. But generally show a trace of a call with debug level 3 . I think it should also show the file it tries to access |
| 09:43.08 | pluesch0r | tzafrir_laptop: i'm doing that already. it's playing the en-files. |
| 09:43.36 | pluesch0r | .. and i'd rather not add a Set(CHANNEL(language)=de) to all my extensions. |
| 09:43.47 | tzafrir_laptop | so pastebin it . maybe it will give others here a clue |
| 09:45.02 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
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| 09:47.15 | pluesch0r | ah well. i'll just set it in every extension. sigh. |
| 09:54.14 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 09:54.46 | vi390 | ist there a known mistake somewhere, when users can phone each other, but do not hear anything ? |
| 10:02.25 | *** part/#asterisk pluesch0r (n=pluesch0@91.186.157.67) |
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| 10:15.32 | Curus | vi390: Try canreinvite=no, then if that helps, debug why. |
| 10:20.51 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 10:26.33 | vi390 | Curus: ok, fixed that. Thanks |
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| 10:35.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 10:36.43 | *** join/#asterisk cjk (n=cjk@85.93.200.99) |
| 10:37.47 | cjk | hi, i only see messages in dmesg "Enabling ecan on channel: 12 " for incoming calls and not even for all calls. any idea where the problem could be, i have a digium pri card with a vpm module |
| 10:42.50 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 11:13.10 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 11:13.14 | ThoMe | hello |
| 11:13.25 | ThoMe | i have: [eingehend] |
| 11:13.25 | ThoMe | include => VerboteneNummer |
| 11:13.31 | ThoMe | and : |
| 11:13.32 | ThoMe | [VerboteneNummer] |
| 11:13.32 | ThoMe | exten => _019X.,1,Macro(Verboten) ; 019x |
| 11:13.36 | ThoMe | doesnt works :-( |
| 11:13.50 | ThoMe | when i copy paste exten => _019X.,1,Macro(Verboten) ; 019x directly in the [eingehend] then works it |
| 11:13.53 | ThoMe | but why? |
| 11:16.38 | *** join/#asterisk esaym (n=user@24.174.186.34) |
| 11:17.07 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:17.07 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:19.17 | *** join/#asterisk benrometsch (n=benromet@87-194-120-46.bethere.co.uk) |
| 11:19.48 | benrometsch | Hi - I'm trying to figure out how to modify which of my 4 ISDN channels are used when I make an outbound call - is this located somewhere in the extensions.conf? |
| 11:21.24 | ThoMe | leifmadsen: hi |
| 11:21.39 | ThoMe | i would like test if my number 0900 xxxxx or 00900 |
| 11:21.46 | ThoMe | is it posible with one line? |
| 11:21.51 | ThoMe | have now exten => _019X.,1,Macro(Verboten) ; 019x |
| 11:23.58 | leifmadsen | ThoMe: exten => _0[09][09]X.,1,NoOp() |
| 11:24.12 | ThoMe | oh okay |
| 11:24.21 | ThoMe | :-) thank you very much :-) |
| 11:25.22 | leifmadsen | np |
| 11:25.51 | Curus | They will match rather more than specified though |
| 11:26.07 | Curus | E.g. 00912xxxxx |
| 11:26.18 | *** join/#asterisk r0d3nt|m (i=r0d3nt@foster.stonedcoder.org) |
| 11:26.25 | leifmadsen | right, you'll need to tweak the end part |
| 11:26.38 | leifmadsen | I just showed how to use the [09] part |
| 11:26.49 | leifmadsen | can't give away ALL the secrets :) |
| 11:27.03 | leifmadsen | it's up to the reader to make sure they understand what I told them, and that it isn't a trap! :) |
| 11:27.09 | ThoMe | Curus: hm ? |
| 11:27.23 | ThoMe | exten => _0[09][09]X.,1,NoOp() not good? |
| 11:27.40 | Curus | ThoMe: It's good if you don't mind matching 00012xxx |
| 11:27.58 | ThoMe | 00012 ? in germany? |
| 11:28.33 | Curus | ThoMe: I'm just telling you what the code does |
| 11:28.52 | ThoMe | okay :-) |
| 11:29.32 | Curus | Asterisk "regular" expressions aren't strong enough to handle optional characters in the middle. |
| 11:29.53 | leifmadsen | right, and it would match 099X too |
| 11:30.46 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 11:32.07 | leifmadsen | only other way is to do something like exten => 0X.,1,GotoIf($[${EXTEN:0:4} = 0900 | ${EXTEN:0:4} = 0090]?handle_call:hangup) |
| 11:32.43 | leifmadsen | s/0X/_0X/ |
| 11:33.24 | ThoMe | emm |
| 11:33.28 | ThoMe | and 011 or 0011 ? |
| 11:33.30 | ThoMe | exten => _0[011][011]X. <<wronG? |
| 11:33.42 | ThoMe | have copy paste _0[09][09]X. and change 9 = 11 |
| 11:33.43 | ThoMe | not good? |
| 11:33.51 | Curus | Not good indeed. |
| 11:33.54 | leifmadsen | ThoMe: you're not understanding |
| 11:34.04 | leifmadsen | [09] means match 0 or 9 in that position |
| 11:34.20 | leifmadsen | _[09][09] would match 00, 09, 90, 99 |
| 11:34.24 | Curus | There's probably a good introduction to Asterisk "regular" expressions in the Asterisk book, isn't there? |
| 11:34.31 | leifmadsen | Curus: yes there is |
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| 11:58.25 | jerryeguru | is there any open source call management software to regulate & control outbound calling |
| 12:00.03 | jerryeguru | okay any commercial one? |
| 12:02.21 | *** join/#asterisk saftsack (n=saftsack@p57924A8E.dip.t-dialin.net) |
| 12:02.41 | Aiatek | vicidial |
| 12:03.03 | Aiatek | sorry |
| 12:03.07 | Aiatek | not your case |
| 12:09.01 | jerryeguru | Aiatek: if i already had asterisk installed can i have vicidial installed too on the same host to regulate outpound calls |
| 12:12.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:17.15 | jerryeguru | ok got my answer VICIDIAL Dialer is suite of software is designed to work with an Asterisk system that has Zap (T1/E1/PSTN), IAX or SIP trunks and SIP/IAX/Zap phones. |
| 12:20.15 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
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| 12:20.49 | *** join/#asterisk Silicium (n=Silicium@2001:bf0:c080:200:0:0:0:23) |
| 12:20.51 | Silicium | hi there |
| 12:21.20 | Silicium | any idea how i can use a HFC BRI Card off-asterisk e.g. as a normal Dialup card? |
| 12:21.37 | Silicium | so i want to test zapRAS to my Asterisk with it |
| 12:21.50 | tzafrir_laptop | Silicium, either though misdn or through chan_dahdi |
| 12:22.06 | Silicium | ok i will try misdn |
| 12:22.24 | Silicium | misdn will create a device for me? |
| 12:22.28 | Silicium | <PROTECTED> |
| 12:26.17 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:31.24 | *** join/#asterisk Vec (n=Vector@host-87-74-7-50.dslgb.com) |
| 12:32.33 | Vec | Anyone here perhaps know the default password on an Avaya Desk phone ? I am trying to flash it with the SIP firmware but need to get its IP etc ? |
| 12:35.20 | Silicium | google avaya default passwords... |
| 12:36.06 | Silicium | or RTFM |
| 12:36.33 | xnixan | tzafrir_laptop, sorry for being late what ever that helps reducing cost |
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| 12:37.18 | xnixan | tzafrir_laptop, 2 ports or to devices with one port! |
| 12:37.26 | xnixan | two* |
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| 12:47.43 | ariel_ | Morning |
| 12:49.49 | timeshell_atwork | Happy Wednesday! |
| 12:50.31 | *** join/#asterisk lost_soul (n=shawn@cpe-67-241-67-197.twcny.res.rr.com) |
| 12:53.15 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 12:54.05 | Vec | http://svn.digium.com/svn/asterisk/branches/1.4/ < would this be the latest version of asterisk 1.4 ? i.e. newer than Asterisk 1.4.25 ? |
| 12:55.46 | tzafrir_laptop | it's the latest . Not the latest released. |
| 12:56.35 | *** join/#asterisk s14ck (n=s14ck@ccscliente154.ifxnetworks.net.ve) |
| 13:01.32 | xnixan | tzafrir_laptop, about the ISDN question do you have a solution? |
| 13:01.51 | *** join/#asterisk kinoko (n=kinoko@LNeuilly-152-21-20-55.w193-253.abo.wanadoo.fr) |
| 13:02.14 | tzafrir_laptop | well, there are quite cheap cards. Certainly single-port ones |
| 13:02.25 | tzafrir_laptop | (I was hoping others would respond to that) |
| 13:03.20 | xnixan | tzafrir_laptop, would you mind direct me to a link? |
| 13:05.06 | Silicium | when i modprobing hfcpci and hfcmulti, which devices i must use for dialing? |
| 13:07.37 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:08.15 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:28.32 | [TK]D-Fender | Ah ha... KerryG is at 888voipstore.com now : http://www.voipstore.com/2009/05/using-the-snom-820-ip-phone/ |
| 13:28.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:28.59 | [TK]D-Fender | And the new Snom 820 looks to be ripping off Linksys style now |
| 13:30.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:31.02 | [TK]D-Fender | And as per usual he demos its setup with trixbox CE |
| 13:31.07 | bionoid | tzafrir_laptop: Thanks for the reverse polarity hint the other day, worked like a charm! |
| 13:31.37 | therealcircut | yo |
| 13:31.43 | therealcircut | [TK]D-Fender: i have good one for you |
| 13:32.04 | therealcircut | who did muhammad ali first face after his exile from boxin for 3.5years?! |
| 13:32.06 | *** join/#asterisk Dovid (n=annon@ool-4355e297.dyn.optonline.net) |
| 13:32.37 | Dovid | hi has anyone got "Denying call id=-74 reason=unconditional" on a snom phone ? cant seem to figure out why the phone wont take the call |
| 13:32.51 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:32.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:33.49 | therealcircut | that wasnt my real question, but it was on my cup of ice tea from dunkin today |
| 13:33.59 | therealcircut | and i figure if u know all, why not have u win me a free bagel |
| 13:34.23 | therealcircut | anyways, i was wondering if theres anyway to force phones to re-register with a SIP server using sipsak or some other tool |
| 13:34.32 | [TK]D-Fender | therealcircut : http://www.google.ca/search?hl=en&q=who+did+muhammad+ali+first+face+after+his+exile+&btnG=Google+Search&meta=&aq=f&oq= |
| 13:34.33 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:34.46 | [TK]D-Fender | therealcircut: 2nd bloody link answer visible in the earch itself |
| 13:34.55 | [TK]D-Fender | therealcircut: JFGI <---- |
| 13:35.26 | [TK]D-Fender | therealcircut: Go read your phone's spec, this clearly has nothing to do with * |
| 13:35.34 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096591399.dsl.bell.ca) |
| 13:35.48 | therealcircut | [TK]D-Fender: all you got me was a lousy 9 pc hash browns?! |
| 13:36.09 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 13:36.20 | [TK]D-Fender | therealcircut: NEVER BITE THE HAND THAT FEEDS! |
| 13:36.36 | [TK]D-Fender | schools therealcircut |
| 13:37.03 | therealcircut | your only 1 for 2 today my friend |
| 13:38.16 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 13:38.37 | devyll | can anybody tell me why do I get " Really destroying SIP dialog " ? |
| 13:39.25 | [TK]D-Fender | devyll: Thats just a notice. What is your actual problem? |
| 13:40.46 | devyll | well, it's filling the logs. I need to be sure it's normal to destroy sip dialog , and maybe stop that from logging. |
| 13:40.56 | devyll | don't know if it makes sense. |
| 13:41.42 | devyll | everything seems to work fine besides that notice. |
| 13:42.44 | *** join/#asterisk asim- (n=asim@mx1.beatthatquote.com) |
| 13:42.52 | asim- | what versions of asterisk are people running? |
| 13:43.01 | asim- | i've been using 1.6.1.0 for a day and its not doing so good |
| 13:43.21 | leifmadsen | "not doing so good" is extremely vague |
| 13:45.07 | rhassing_work | I'm using 1.6.1.0 for two weeks now and I'm happy :-) |
| 13:45.27 | asim- | well attended transfer does not work |
| 13:45.33 | Zhad | asim> What's been the problem? |
| 13:45.38 | asim- | it would not play .sln files |
| 13:45.52 | asim- | and now after 4 hours its stopped accepting calls |
| 13:46.43 | Zhad | the weirdest thing with 1.6.1 that I remember was needing to fnd libresample from somewhere. |
| 13:47.10 | Dovid | hi has anyone got "Denying call id=-74 reason=unconditional" on a snom phone ? cant seem to figure out why the phone wont take the call |
| 13:47.35 | Silicium | PAAAIN |
| 13:47.36 | Silicium | <PROTECTED> |
| 13:47.57 | seanbright | Dovid: DND? |
| 13:48.03 | asim- | libresample? |
| 13:48.10 | leifmadsen | Zhad: it's on the svn server |
| 13:48.19 | leifmadsen | under svn/thirdparty/ |
| 13:48.39 | [TK]D-Fender | devyll: then your loggin level is too high. |
| 13:49.11 | [TK]D-Fender | 1.6.1.0 = bleeding edge first release of new branch. |
| 13:51.17 | Zhad | leif> that's probably where I got it from. |
| 13:51.57 | asim- | dahdi isnt running. hmm |
| 13:52.56 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
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| 14:00.37 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
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| 14:01.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.35 | Nunners | I'm about to ask a question using a combination of config and logfiles, and can't remember what the website is where I can post thiem with a url associated.... |
| 14:02.23 | Nunners | Can someone remind me please? :) |
| 14:03.11 | [TK]D-Fender | ~pb |
| 14:03.12 | 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://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 14:03.36 | Nunners | pastebin... that's it... thanks |
| 14:05.22 | Dovid | seanbright: DND is off |
| 14:06.22 | Dovid | seanbright: I get a 486 back from the phone. that error is in the SNOM phones log. |
| 14:09.17 | Dovid | seanbright: I think i got it. the phone was forwarded. i am not on site to see it :( |
| 14:10.34 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:13.21 | esaym | can I have 2 sip voip providers going into the same context? Everything works fine when I only enable one provider. But when I enable both of them I get "chan_sip.c:12679 handle_response_invite: Received response: "Forbidden"" and "Everyone is busy/congested at this time" |
| 14:13.47 | [TK]D-Fender | esaym: context is dialplan and has nothing to do with a 403 |
| 14:14.05 | esaym | yea, this is weird |
| 14:14.13 | Aiatek | you can have as many as you want |
| 14:14.14 | [TK]D-Fender | esaym: Guess you'd better SHOW US |
| 14:14.32 | esaym | unless my provider doesn't allow 2 accounts to do to the same ip address or something |
| 14:19.18 | esaym | See: http://pastie.org/491404 when I uncomment that one account, that is when I start having trouble |
| 14:20.43 | Tegg | hi |
| 14:21.15 | [TK]D-Fender | esaym: Stop setting "insecure, use "type=peer" |
| 14:21.40 | therealcircut | ok heres a question |
| 14:21.41 | Tegg | i have a WARNING[2973]: pbx.c:2966 ast_register_application: Already have an application 'VoiceMail' |
| 14:21.54 | therealcircut | so i can do: "asterisk -rx 'sip notify snom-reboot <peer>" |
| 14:21.55 | Tegg | and don't find the doubleconfiguration |
| 14:21.58 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 14:22.13 | Nunners | I've recently updated from 1.4.20 to 1.4.25 (latest) and since then when we dial out, it doesn't select the correct Zap channel. Did anything change between the two versions - I've checked and cannot see anything in the changelogs.... debug: http://pastebin.com/m47d81a1a |
| 14:22.13 | Nunners | <PROTECTED> |
| 14:22.14 | therealcircut | where can i find more of those notify mesages to load into asterisk |
| 14:22.34 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:22.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:23.46 | esaym | [TK]D-Fender: ok I will try that |
| 14:28.13 | Nunners | Mmm... this gets wierder. Now turned off the macro stuff, just using normal commands, and it's saying ZAP unavailable, even though I know it's loaded. Something has changed between the versions... |
| 14:28.15 | *** join/#asterisk simprix (n=simprix@69.50.82.130) |
| 14:30.15 | Tegg | where is the voicemail configurated in the standard configs ? |
| 14:30.35 | [TK]D-Fender | Tegg: voicemail.conf |
| 14:31.26 | Tegg | and the i deactivated the Voicemail |
| 14:31.30 | Tegg | there |
| 14:31.32 | esaym | [TK]D-Fender: if I do that then I cannot receive calls but only place them? |
| 14:31.43 | Tegg | but i still get the warning |
| 14:32.03 | esaym | (I both send and receive calls from my provider) |
| 14:33.01 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:34.18 | [TK]D-Fender | esaym: ... JUST DO IT |
| 14:35.02 | esaym | [TK]D-Fender: oh is this just for trouble shooting purposes right now? |
| 14:35.13 | [TK]D-Fender | Tegg: Ah, you have MULTIPLE voicemail MODULES loading. There is a base app_voicemail.so, and versions for IMAP, tec. Go look in your MODULES folder. |
| 14:35.20 | *** join/#asterisk Slade- (i=user@255.85.204.68.cfl.res.rr.com) |
| 14:35.26 | [TK]D-Fender | Tegg: edit modules.conf to noload the ones you don't need |
| 14:35.37 | Tegg | ahh noload again :) |
| 14:35.45 | [TK]D-Fender | Tegg: And thats jsut a warning anyway unless you've got some sort of real error to show us |
| 14:36.38 | esaym | [TK]D-Fender: yes still the same issue |
| 14:37.02 | Tegg | i don't know who i can ask else, sorry |
| 14:40.03 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 14:40.45 | telnettech | jaytee. I will try not to use all the dots today |
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| 14:46.36 | [TK]D-Fender | esaym: And I am never seeing the problem |
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| 14:50.01 | esaym | [TK]D-Fender: yes it might be an issue with my provider |
| 14:50.38 | [TK]D-Fender | esaym: No, the problems debugging this are PEBKAC right now... |
| 14:50.51 | [TK]D-Fender | esaym: You aren't showing anything |
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| 15:10.45 | Slade- | hey in asterisknow.. do i have to install a module or something for h323 support? |
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| 15:12.58 | leifmadsen | hard to say if h323 stuff is even available in asterisknow |
| 15:13.13 | Slade- | the website says it is.. the website could be lying tho |
| 15:13.39 | Slade- | i was reading this.. but it talks about a file that isnt there.. http://forums.digium.com/viewtopic.php?p=40763&sid=042aa447fdbfc13dad21948f9ecffe28 |
| 15:16.10 | leifmadsen | Qwell: ^^^ |
| 15:16.15 | leifmadsen | website could be out of date... |
| 15:17.07 | Slade- | trying to interface with cisco silliness. and h323 seems the best |
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| 15:42.55 | therealcircut | ok so i can reboot the snoms and the grandstreams, but no luck with the polycoms ;/ |
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| 15:44.04 | therealcircut | woot |
| 15:44.07 | therealcircut | just got it :D |
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| 15:47.40 | SiLiCiUM | sorry for metaquestions, but, anyone have some expirience with app_zapRAS ? |
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| 15:55.55 | SiLiCiUM | https://nopaste.eof.name/47 <-- any ideas? |
| 15:56.39 | seanbright | invalid cert |
| 15:56.44 | seanbright | use a better pastebin |
| 15:58.06 | vsemenov | does asterisk support voice IVR? |
| 15:58.45 | SiLiCiUM | vsemenov: with plugins yes |
| 15:59.06 | SiLiCiUM | but there afaik there are no stable projects |
| 15:59.08 | vsemenov | what are the names of the plugins i would need? |
| 15:59.19 | SiLiCiUM | vsemenov: google will help you |
| 15:59.23 | vsemenov | ok thanks |
| 16:01.11 | therealcircut | no he wont |
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| 16:01.53 | vi390 | I have a incoming call forward in extensions.conf : exten => 10,n,Dial(SIP/200,60) ; how can I forward incoming calls to all connected sip clients ? |
| 16:02.48 | SiLiCiUM | forward to _? |
| 16:02.50 | SiLiCiUM | :D |
| 16:02.53 | SiLiCiUM | _X |
| 16:03.17 | SiLiCiUM | i solved with a caller group |
| 16:03.34 | vi390 | ok not forward, but "dial" |
| 16:03.43 | Kevin` | vi390: dial(SIP/200&SIP/201) ? |
| 16:03.51 | SiLiCiUM | Kevin`: yes |
| 16:04.19 | vi390 | Kevin`: mhh yes, but still explicitly to give all of them there |
| 16:04.33 | vi390 | caller group sound like wht I want. |
| 16:04.38 | Kevin` | google says http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 16:04.39 | vi390 | gonna check that out |
| 16:04.43 | SiLiCiUM | ueeee, bye just go home |
| 16:05.29 | vi390 | Kevin`: ah this sound like what I want ;) thanks |
| 16:09.25 | jaytee | "SERENITY NOW!!!!!" |
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| 16:13.00 | jameswf | Caller ID gets to Asterisk but does not pass to the SIP phone so it is automatically the cards fault... makes perfect sense |
| 16:14.08 | MCCob | hello everyone, with Asterisk 1.6 I Get "Unable to create channel of type 'DAHDI'" when I try to transfer a call to my PBX, someone know why ? |
| 16:14.26 | jameswf | MCCob: core chow channeltypes |
| 16:14.32 | MCCob | channels are In Serv |
| 16:14.44 | jameswf | mmmm chow.. |
| 16:14.59 | MCCob | DAHDI DAHDI Telephony Driver w/PRI no yes no |
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| 16:15.24 | MCCob | Type Description Devicestate Indications Transfer |
| 16:15.54 | MCCob | it's core show channeltype output |
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| 16:18.04 | SuPrSluG | well dev state in no so it's not enabled |
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| 16:57.40 | kamh | hello |
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| 17:05.58 | nullable_type | When i bridge call using a voip provider, i sense a network delay than usual, What's the best way to debug network issues in Asterisk |
| 17:12.56 | pif | tzafrir_laptop: hi, do you have a zaptel-source that compiles with 2.6.29 ? I tried 1:1.4.12.9.svn.r4635~dfsg-0.7224 from your repository but it breaks at "/usr/src/modules/zaptel/kernel/vzaphfc/vzaphfc_main.c:569: error: 'struct net_device' has no member named 'priv'" |
| 17:13.24 | tzafrir_laptop | pif, hmm... yes, it's something I need to fix |
| 17:13.42 | tzafrir_laptop | there's an open bug for it, and I should hopefully merge it soon |
| 17:13.48 | pif | do you have a patch somewhere maybe? |
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| 17:24.33 | tzafrir_laptop | pif, look at that open bug (in bugs.debian.org/zaptel-source , IIRC) . GTG |
| 17:25.01 | pif | oki doki |
| 17:31.56 | vi390 | can I change the Chmod of "monitor" voice files written by Asterisk ? |
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| 17:52.40 | jameswf | here they come to save the Day !! http://trixbox.org/forums/trixbox-forums/help/new-trixbox-ce-support |
| 17:57.30 | ruben23 | hi when i set the recording on my asterisk server where i could find the final records... |
| 17:58.29 | Aiatek | .var/spool/asterisk/monitor |
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| 18:08.23 | nullable_type | How can i debug a delay in sip conversations using Asterisk? i have sip debug on but not all the requests have timestamps |
| 18:11.33 | nullable_type | Or it might be RTP session initiating delay |
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| 18:58.24 | EGBlue | does anyone know where I can download voiceglue from? the link at the website isn't working |
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| 18:59.36 | jasonwoot | I dunno what voiceglue does, but it sounds cool |
| 19:00.39 | EGBlue | it is voicexml support for asterisk |
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| 19:32.23 | therealcircut | grrr |
| 19:32.30 | therealcircut | sipsak is pissin me off |
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| 20:07.43 | saxa | if I want to connect 2 * boxes, and one has a static IP but the other one has a dyn Ip, how is the best way to do that ? |
| 20:08.04 | carrar | dynamic IP box registers to the stativ |
| 20:08.06 | carrar | c |
| 20:08.14 | saxa | ok |
| 20:08.33 | saxa | but then will i be able to pass calls from one location to another ? |
| 20:08.39 | carrar | yes |
| 20:08.42 | saxa | and viceversa ? |
| 20:08.46 | carrar | & yes |
| 20:08.50 | saxa | good |
| 20:09.05 | EGBlue | how can I direct a SIP call that is being received from a non-register source to a specific context? |
| 20:09.34 | carrar | use the default context |
| 20:09.59 | carrar | then part it out in the dialplan |
| 20:10.01 | carrar | parse |
| 20:10.09 | saxa | another question i have is, is there a way to configure a dialplan to each incoming call on Zap/1 be immediately redirected to some sip on the other static IP box ? |
| 20:10.31 | EGBlue | thanks carrar |
| 20:11.08 | carrar | yes |
| 20:15.37 | saxa | carrar: can you direct me ? |
| 20:16.26 | [TK]D-Fender | ~book |
| 20:16.27 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:16.29 | [TK]D-Fender | saxa: ^^^^ |
| 20:16.49 | [TK]D-Fender | saxa: This is all jsut dialplan. Answer the call and DIAL the other box. 2-3 lines |
| 20:19.16 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
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| 20:19.29 | *** join/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
| 20:19.49 | dshap | hey drmessano and or carrar - are you guys here? |
| 20:20.16 | dshap | ? |
| 20:20.33 | dshap | is anyone here who knows stuff about callerID? |
| 20:21.26 | [TK]D-Fender | ~ask |
| 20:21.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:21.39 | dshap | oh or D-Fender |
| 20:21.40 | dshap | alright then |
| 20:22.01 | dshap | my question is specifically related to the callerID headers that are sent from outgoing calls from an asterisk box |
| 20:22.18 | dshap | can they be manipulated such that the receiving phone/system can ONLY see what you want them to see? |
| 20:23.57 | dshap | specifically, I want to design an outgoing call context that dials into an AT&T voicemail backdoor number |
| 20:24.09 | dshap | and then leaves a voicemail in someone's voicemail box |
| 20:24.30 | dshap | but i want the voicemail system to think the voicemail came from whoever left the message by dialing into my asterisk box |
| 20:24.32 | dshap | if that makes sense |
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| 20:24.49 | saxa | [TK]D-Fender: thx |
| 20:24.51 | [TK]D-Fender | dshap: the ability to set your callerid depends who you are calling THROUGH |
| 20:24.52 | dshap | is this possible? |
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| 20:25.03 | ZikiBe | hi |
| 20:25.11 | dshap | what if i am calling through a SIP trunk provider |
| 20:25.17 | dshap | i would have to ask them, huh? |
| 20:25.56 | [TK]D-Fender | dshap: Yes |
| 20:25.59 | ZikiBe | does anyone know if there's a CLI command to list users from users.conf? |
| 20:26.42 | dshap | so if i am going to ask a potential provider, what should i ask? |
| 20:26.53 | dshap | "do you guys pass the CID headers that I send out?" |
| 20:26.54 | [TK]D-Fender | dshap: "can I set Callerid |
| 20:26.58 | dshap | ok |
| 20:27.02 | dshap | and if they say yes |
| 20:27.09 | dshap | that means i can make my calls seem like they are from whatever number i want |
| 20:27.14 | [TK]D-Fender | dshap: just SET IT |
| 20:27.18 | dshap | gotcha |
| 20:27.21 | dshap | thank you very much |
| 20:28.52 | dshap | also an unrelated question...but if i wanted to make an outgoing call context that dials the AT&T voicemail backdoor number and has to use touch-tone input to interact with it to select the mailbox and start the recording, would that be feasible to do with asterisk? |
| 20:29.23 | [TK]D-Fender | dshap: Yes |
| 20:29.29 | dshap | okay cool |
| 20:29.39 | dshap | thanks [TK]D-Fender, as always, for your help |
| 20:29.52 | [TK]D-Fender | dshap: "core show application dial" |
| 20:29.53 | dshap | oh actually one more thing |
| 20:30.27 | Kobaz | do de do |
| 20:30.32 | dshap | i'm currently searching for a SIP provider that sends me the diversion SIP headers, in order to get the CID of a phone/device that is FORWARDING another call |
| 20:30.37 | dshap | my current provider does not support this |
| 20:30.42 | dshap | is it uncommon? |
| 20:30.57 | dshap | i called up bandwidth.com and they simply told me "we pass everything we receive" |
| 20:31.04 | [TK]D-Fender | dshap: No, not uncommon. |
| 20:31.15 | dshap | okay so then it sounds like my current provider is the odd one out |
| 20:31.39 | dshap | when i receive a forwarded call with my current provider and check ${CALLERID(rdnis)} |
| 20:31.41 | dshap | it is empty |
| 20:31.51 | dshap | but it should be the CID of the forwarding phone |
| 20:31.54 | dshap | correct? |
| 20:32.25 | [TK]D-Fender | dshap: Maybe. depends how fields get passed. "core show function SIP_HEADER" |
| 20:32.33 | [TK]D-Fender | ok, G2G, BBL |
| 20:38.12 | jaytee | G2G2, BBLAD |
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| 20:41.33 | spck | trying to get * to connect to postgresql, i get this error when registering a sip client: update_pgsql: PostgreSQL RealTime: Failed to query database. Check debug for more info. |
| 20:41.47 | spck | what does it mean by check debug? |
| 20:42.04 | spck | i've already checked /var/log/debug|messages|syslog etc... |
| 20:43.43 | spck | does it mean sip set debug on? |
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| 21:00.46 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 21:01.47 | nny_1 | quick straneg question, I know I can use Dial to do two numbers at once (number1&number2,etc). I want to do two calls at the same time with two seperate dial statements. The purpose is to set the callerid differently on the second number, any advice? |
| 21:03.07 | nny_1 | exten => s,1,Set(CALLERID(num)=${COMPANYNUMBER}) exten => s,2,Dial(${ARG2}&SIP/${CELL${ARG1}}@${SIPOUTBOUND},50,wm) |
| 21:03.53 | nny_1 | is what I have right now. I was hoping to set the second dial argument with a different caller id. Reason is we use the system to also ring my cell, and the circle of jerks friends thing that they have makes them free :) |
| 21:04.11 | nny_1 | however it is nice to have the proper caller id on my desk phone |
| 21:05.06 | outtolunc | if you are going as far as setting a SIPOUTBOUND var, why not set SIPOUTCLID and set(callerid(num)=${SIPOUTCLID}) ? |
| 21:05.48 | nny_1 | outtolunc: i use COMPANYNUMBER in other places in the dialplan |
| 21:06.14 | nny_1 | outtolunc: although your variable seems to make more sense |
| 21:06.19 | outtolunc | you can name it whatever you like <G> |
| 21:07.22 | nny_1 | i think i may be SOL on it, i don't see a way to run Dial and set just one identifier to have a specific CID |
| 21:07.53 | nny_1 | is there a way to hard code any outbound calls to a specific zap group? |
| 21:07.57 | nny_1 | er nm |
| 21:08.20 | outtolunc | if you are only hitting the dial() once, you have to set the channel var before it, but nothing says your dialplan can't call a macro/sub that contains the dial |
| 21:08.51 | outtolunc | zapata/dahdi.conf lets you set group=x |
| 21:08.52 | nny_1 | hmmm |
| 21:09.18 | outtolunc | (you could set your callerid there also) |
| 21:09.29 | nny_1 | yeah but would need to set outbound CID for the outbound sip provider |
| 21:09.42 | outtolunc | sip <> zap |
| 21:09.51 | outtolunc | hehe |
| 21:10.06 | nny_1 | so your saying my SIPOUTBOUND could be a specific CID. |
| 21:10.07 | nny_1 | i see |
| 21:10.11 | nny_1 | (in sip.conf) |
| 21:10.24 | nny_1 | so no matter what dials out that channel, it will always show the CID as what I set it to |
| 21:10.45 | outtolunc | you can set a per TECH type callerid |
| 21:11.02 | outtolunc | but it will only fill (usually) if not already |
| 21:11.12 | outtolunc | meaning overridden down the line |
| 21:11.39 | outtolunc | in this case tech/dev being sip/provider |
| 21:12.12 | nny_1 | yeah i see callerid=XXXXX is only used if no CID is available in sip.conf |
| 21:12.14 | nny_1 | hmmph |
| 21:12.42 | nny_1 | heh |
| 21:13.38 | nny_1 | well, i am looking into a way to SMS the CID to my phone before, may just have to live with my desk phone not having inbound caller list. God I am cheap ;) |
| 21:14.05 | outtolunc | system(echo... ) |
| 21:14.17 | outtolunc | asterisk wiki callerid sms |
| 21:14.22 | nny_1 | kk looking |
| 21:14.44 | nny_1 | if anyone thinks of a claver way to preserve the callerid to ${ARG1} only in that statement, lemme know. |
| 21:14.47 | nny_1 | clever* |
| 21:14.56 | outtolunc | afk, need coffee only half-slept last night |
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| 21:18.13 | kkemp | Anyone notice that msg.bypassInstantMessage="1" is ignored on the Polycom 330 phones? |
| 21:19.00 | melaleuca | Anybody can recommend eitheir a HP or IBM for a small Asterisk Office? Or is there a website with some current specs etc? |
| 21:22.25 | timeshell_atwork | Is there any magic needed to make asterisk work with video? In other words, do you need to add in any channels for H263/4 or is it included already in 1.6.x? |
| 21:23.53 | drmessano | videosupport=yes and the proper allow lines for h263 and h264 |
| 21:24.06 | timeshell_atwork | That's all? |
| 21:24.26 | drmessano | Yes |
| 21:24.27 | timeshell_atwork | Don't have to install any other channels? |
| 21:24.38 | drmessano | I gave you a complete answer |
| 21:24.42 | timeshell_atwork | wow |
| 21:24.55 | timeshell_atwork | I remember having to do other stuff last time I tried to install video |
| 21:25.12 | drmessano | Its worked out of the box for me since 1.2 |
| 21:25.22 | timeshell_atwork | hrmmm |
| 21:25.55 | timeshell_atwork | Allow lines for h263/4 are in sip.conf? |
| 21:26.11 | drmessano | and the videosupport line |
| 21:27.03 | timeshell_atwork | Great. Thansk |
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| 21:35.24 | melaleuca | Anybody can recommend eitheir a HP or IBM for a small Asterisk Office? Or is there a website with some current specs etc? |
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| 21:48.56 | init6 | Where can I find out what are allowed for hint extension names. For example 4000 and 1400 work but 140, 40, and 4 do not. By not work I mean "core show hints" shows "Watchers 0" and the SPA932 lights blink orange. |
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| 21:59.08 | nny_1 | i love sendmail |
| 21:59.31 | nny_1 | i am trying to discern how to do mail -s EMAIL BODY and have it just auto send in a script |
| 21:59.37 | nny_1 | yet still i am here :\ |
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| 22:07.50 | beek | nny_1: Don't use the 'mail' command, use the 'sendmail -oi -t' command. |
| 22:09.43 | timeshell_atwork | l |
| 22:11.26 | nny_1 | beek: hmm ok |
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| 22:11.50 | kkemp | On polycom phones is it possible to have it display the name from the phone's directory instead of just the number when dialing the extension directly? |
| 22:12.02 | beek | nny_1: let me know if you need an example. |
| 22:12.29 | nny_1 | beek: did System(echo "${CALLERID(num)}" | mail "${CELL${ARG1}}@tmomail.net") |
| 22:12.35 | nny_1 | but that's not right? it works fwiw |
| 22:12.57 | nny_1 | sends an sms to my cell phone telling me the CID |
| 22:13.27 | beek | If that's all you want it should work fine. |
| 22:13.44 | nny_1 | yeah i think so, the sms messages don't seem to stack right now, looking into that, but other than that it works |
| 22:14.32 | nny_1 | hah now they do |
| 22:14.35 | nny_1 | i <3 asterisk |
| 22:15.18 | Pan3D | :) |
| 22:15.18 | nny_1 | 3 lines, 1 to sms me the CID, 1 to change the CID to my company number, and one to ring my deskphone and cellphone. Add company number to Myfaves/circle and boom, free calls |
| 22:15.34 | nny_1 | i lose CID at deskphone, i'll work on that tomorrow |
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| 22:20.09 | nny_1 | any reason exten => s,1,System(echo "${CALLERID(num)} called at ${TIMESTAMP}" | mail "${CELL${ARG1}}@tmomail.net") |
| 22:20.09 | nny_1 | doesn't have the timestamp? |
| 22:20.51 | nny_1 | er oops |
| 22:20.58 | nny_1 | TIMSTAMP is deprectated heh |
| 22:23.12 | nny_1 | fixed |
| 22:23.20 | nny_1 | alright i am done spamming channel, later all thanks for the help |
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| 22:26.09 | nullable_type | Can i query flat file CDR record from the CLI? for example to get the record with accountcode 1112 |
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| 22:33.34 | Corydon76-dig | nullable_type: nope |
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| 22:34.28 | Corydon76-dig | nullable_type: you could do a grep... CLI> !grep 1112 /var/log/asterisk/cdr-csv/Master.csv |
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| 22:49.51 | TokyoJimu | Can I have zaptel not load unneeded modules like wct4xxp? We've been experiencing some panic()s and I'd like to make it as simple as possible. |
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| 22:53.04 | nullable_type | Cory >> Thank you so much for your idea |
| 22:53.22 | nullable_type | as obvious, I am new to linux, i was thinking of writing a mini app to parse the CSV |
| 22:54.19 | TokyoJimu | Ahh, seems I need to edit /etc/sysconfig/zaptel |
| 23:02.20 | nullable_type | Cory >> Can i create custom commands? like set MyCommand=!grep |
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| 23:09.02 | saxa | I need to understand on thing, if I want to connect my dynamic IP aserisk box to my fixed IP asterisk box, I should do it within iax.conf on one side, but on the fixed IP box where do I need to set up the password for the other box ? |
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| 23:12.32 | nullable_type | Core are you there |
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| 23:25.39 | mattbUK | Hello folks this one is for people using G729? - how do you find it impacts processor performance is there a noticable increase? |
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