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| 00:15.58 | [TK]D-Fender | [19:46]<bjhaid>yes, wondering if it could be a FXS or FXO port too <- no |
| 00:16.10 | p3nguin | (1855.47) -!- bjhaid ~IceChat7@41.206.15.1 has quit |
| 00:16.18 | [TK]D-Fender | \o/ |
| 00:18.25 | coppice | bjhaid: thousands want to use a winmodem as an FXO port. nobody wants to do the work to make it so. :-( |
| 00:18.42 | lost_soul | lol |
| 00:19.01 | lost_soul | coppice: gone |
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| 00:21.36 | [TK]D-Fender | high-5's coppice |
| 00:40.27 | p3nguin | Is there a way in dialplan to decrease or increase the playback/background volume of sound files? |
| 00:43.00 | lost_soul | not that I've ever seen |
| 00:43.33 | [TK]D-Fender | p3nguin: "core show function VOLUME" |
| 00:44.14 | p3nguin | asterisk*CLI> core show function VOLUME |
| 00:44.14 | p3nguin | No function by that name registered. |
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| 00:45.43 | [TK]D-Fender | p3nguin: ver? |
| 00:45.49 | p3nguin | 1.4.33 |
| 00:46.12 | [TK]D-Fender | p3nguin: You ppor, poor sap :0 |
| 00:46.32 | [TK]D-Fender | p3nguin: module unload res_decrepit.so |
| 00:46.56 | p3nguin | It's brand-spankin'-new! |
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| 00:48.07 | p3nguin | I am kind of surprised that some of the more popular functions of the 1.6.x branches have not been added in the newest 1.4 versions. |
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| 00:49.40 | [TK]D-Fender | p3nguin: I think you have failed to grasp how *'s branches work. |
| 00:49.51 | [TK]D-Fender | p3nguin: Once a branch is released.. NO new features |
| 00:50.14 | [TK]D-Fender | p3nguin: 1.6.2 has something 1.6.1 doesn't? Well its not going back. |
| 00:52.48 | sip7 | Does anybody know how to resume execution of a dial plan after the Page() app concludes? |
| 00:54.43 | [TK]D-Fender | sip7: And how does it "conclude"? |
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| 00:58.35 | sip7 | TK: I've included the code on pastebin here: http://pastebin.com/H4ugzhFT |
| 00:58.45 | sip7 | It is for overhead and extension paging |
| 00:58.52 | sip7 | Included are comments as well |
| 00:59.53 | sip7 | Here is a better version: http://pastebin.com/XGWXb7NK |
| 01:02.17 | path | is it possible to change the verbosity level on CLI without restart *¡ |
| 01:02.18 | path | ? |
| 01:03.08 | [TK]D-Fender | sip7: wrong answer. I asked HOW it "ends". |
| 01:03.10 | pabelanger-lap | *CLI> core set verbose 15 |
| 01:04.42 | path | thats exactly what I wanted. |
| 01:04.46 | path | ty pabelanger-lap |
| 01:05.03 | path | I was looking on help core already :] |
| 01:06.01 | sip7 | [TK]D-Fender: Oh... well, the logs look like this.. http://pastebin.com/MkdX34JD |
| 01:07.14 | [TK]D-Fender | SipWhat is that mountain of crap? I asked HOW the damn PAGE gets stopped? What friggen ACTION? |
| 01:07.54 | pabelanger-lap | sip7: exten => 170*,n,Hangup() your hanging up the channel |
| 01:07.57 | sip7 | My apologies... A hangup by the user |
| 01:09.09 | path | whats the module/app for t.38 faxing? |
| 01:09.32 | [TK]D-Fender | sip7: Hangup = dead call = "H' Asterisk Standard Extension |
| 01:10.05 | path | nevermind |
| 01:16.06 | sip7 | [TK]D-Fender: Right.. so I tried something like this instead http://pastebin.com/4QqKYWQp |
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| 01:17.28 | sip7 | The respective debug log for that setup is: http://pastebin.com/mZ6U9TrN |
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| 01:17.58 | sip7 | As you can see on line 60, the unmute command is executed before the user even begins to speak |
| 01:18.22 | sip7 | I'm not sure why the h extension is executed so prematurely as there is no hangup |
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| 01:21.05 | [TK]D-Fender | exten => 170,1,Answer() |
| 01:21.11 | [TK]D-Fender | <PROTECTED> |
| 01:21.24 | [TK]D-Fender | Calling 170 at the same time as anything else = RETARDED |
| 01:21.33 | [TK]D-Fender | 170 is going to instantly win. |
| 01:22.56 | sip7 | So, how can I set it up to page all extensions (via 167) and the overhead system (170)? |
| 01:23.18 | sip7 | A user does not want to have to page the same thing twice.. |
| 01:24.02 | [TK]D-Fender | sip7: You're also knee-deep in #freepbx territory. You should be taking it up there |
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| 01:26.19 | sip7 | Alright, thank you, TK |
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| 02:12.00 | jaytee | [TK]D-Fender, hey quick one for you. @ my new job they have Vonage. I have a TDM04B card with 3 FXO mods on it. Can I take the fxs lines from the Vonage ATA's and run them into the FXO ports to use Asterisk internally until I switch over to another SIP provider? |
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| 02:38.25 | [TK]D-Fender | jaytee: Yes. Or have them use the softphone accont for it and take it into * STRAIGHT |
| 02:38.47 | [TK]D-Fender | jaytee: And... whats the new job like? Long time no hear... |
| 02:40.04 | jaytee | it's pretty good, lots less stress cuz my boss is an old friend and he's not a dick like my last boss. |
| 02:40.25 | [TK]D-Fender | jaytee: Pay acceptable? |
| 02:40.30 | jaytee | [TK]D-Fender, I was thinking of using a Vonage softphone account |
| 02:40.54 | jaytee | pay is about the same but not the same benefits package at the old job |
| 02:41.09 | jaytee | I'm ok with that. better to have work than no work |
| 02:41.29 | [TK]D-Fender | jaytee: Yeah, but its not paired with the RISK package of the old job :) I guess I'm happy for you... |
| 02:42.52 | jaytee | I feel a lot less stressed out normally. Plus I'm doing IT work and a mix of development so I get to sharpen my coding skills and a project I'm working on in spare time has the potential to earn extra dollars. |
| 02:43.46 | [TK]D-Fender | jaytee: Cool... |
| 02:44.42 | [TK]D-Fender | jaytee: My work is same as always (15 years running), but I finally got up and tried out for an actual band. Was accepted the following day, had 33 song setlist to be ready for 8 days following and the the gig went great. We've already been re-booked |
| 02:44.59 | [TK]D-Fender | So I'm now technically a "professional musician" :p |
| 02:45.17 | [TK]D-Fender | NOW WHERE ARE THE CHEAP FLOOZIES! |
| 02:45.24 | jaytee | that's awesome!! you'll have to post some vid clips |
| 02:45.32 | [TK]D-Fender | jaytee: 1 SEC |
| 02:45.55 | [TK]D-Fender | jaytee: http://www.facebook.com/?ref=home#!/video/video.php?v=405472060956 |
| 02:47.05 | [TK]D-Fender | jaytee: Me in black in the back :) |
| 02:47.53 | jaytee | in the black shirt |
| 02:48.02 | jaytee | good sounding covers |
| 02:48.28 | [TK]D-Fender | its a 10 min mix of stuff we did that night |
| 02:49.02 | jaytee | the quality is very good. I'd pay the cover charge |
| 02:49.21 | [TK]D-Fender | jaytee: LOL... no cover :0 |
| 02:49.46 | [TK]D-Fender | This was unfortunately on the same night as the F1 races... so it was a smaller crowd.. and the buggers were all out on the terasse! |
| 02:50.00 | [TK]D-Fender | until like 2Am.... thats when the hotties hit the floor :) |
| 02:50.05 | jaytee | I paid a 3 dollar cover in 1973 to see Aerosmith at a local bar in my town just before they released Get Your Wings |
| 02:51.20 | jaytee | the vocalist is good but doesn't really have the high range for a Guns and Roses tune :-) |
| 02:53.41 | jaytee | my buddy Vic has been a drummer for about 20 years. just got a gig with a band named "5 Minutes to Wapner" |
| 02:57.01 | [TK]D-Fender | jaytee: Drummer & bassis swap roles on occasion. I'm on Guitar & keys (mostly rhythm) |
| 02:57.16 | [TK]D-Fender | jaytee: And I may pick up vocals. Our lead is a big of a power-rocker indeed. |
| 02:57.47 | jaytee | yeah, I could tell by watching that you were playing rhythm while the bald guy was playing lead. |
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| 02:58.20 | jaytee | you both sound solid and the drummer is good too |
| 02:59.09 | jaytee | the power rocker likes to improvise a bit. some people hate that but I tend to like it as it makes oldies less monotonous |
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| 03:00.39 | [TK]D-Fender | jaytee: Yeah, they took a lot of older tunes and revamped them, also inserted interludes of other solgs... |
| 03:01.17 | ChannelZ | PLAY FREE BIRD! |
| 03:01.53 | [TK]D-Fender | jaytee: Somebody To Love (Jefferson Airplane) they intro with the guitar from "You've Got Another Thing Coming" (Judas Priest) and has a part-way interlude of 8 bars from Crazy Train (Ozzy) |
| 03:02.13 | [TK]D-Fender | ChannelZ: God you sound like one of those Guitar Hero retards :p |
| 03:02.33 | [TK]D-Fender | "I beat it on hard, I R KOOL!!!!!" |
| 03:02.40 | jaytee | hehehe |
| 03:03.16 | jaytee | I was at a city park last year and they had a polka band playing and someone shouted "Freebird!!!" |
| 03:03.34 | ChannelZ | I can barely play on Medium |
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| 03:05.26 | [TK]D-Fender | I beat something on medium. I only tried like on 3 differnt occasions. The first time my hand cramped so hard I was looking to self-amputate |
| 03:05.40 | [TK]D-Fender | And I've played guitar for 20 years.... |
| 03:06.08 | [TK]D-Fender | PLASTIC SHIT TOY IS *MOCKING ME*. DESTROY! |
| 03:06.46 | jaytee | lol |
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| 03:10.19 | [TK]D-Fender | jaytee: Go look up the "Guitar Queer-O" South Park episode on Comedy Central. Its teh SHIZNIT y0 |
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| 03:13.35 | ChannelZ | Yay, I am free of Qwest! |
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| 03:27.22 | jaytee | ahahaaa, "real guitars are for old people" |
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| 03:28.25 | jaytee | "game over! you suck!" |
| 03:28.50 | p3nguin | lol |
| 03:30.30 | jaytee | topless girls in the pool at the "giant mansion" |
| 03:32.50 | [TK]D-Fender | jaytee: Watching now? |
| 03:32.58 | jaytee | yeah |
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| 03:47.38 | jaytee | ahahaaaaaa, "you are fags!" |
| 03:47.43 | jaytee | priceless!!!! |
| 03:47.47 | [TK]D-Fender | :D |
| 03:48.13 | jaytee | thanks for the link, man. that made my night |
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| 04:23.43 | drclue | Sorta curious as to what folks might like to see in a generic HTML5/AMI interface. I already have the server side written and XML based transactions going between the browser and Asterisk via HTML5 WebSockets and it supports many concurrent browser connections. This next week I'm working on refining the browser UI framework. |
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| 05:00.24 | joobie | hey guys.. is there a way in asterisk queue to play a different periodic-greeting after a caller has been on hold for more than 5 mins? I want to have a stndard periodic-greeting play, but if they are on hold beyond a threshold, play a different periodic greeting |
| 05:02.18 | p3nguin | You could set up a second queue. |
| 05:02.19 | [TK]D-Fender | joobie: No. What you CAN do is use an MoH class that points to a SINGLE recording with all of your prompts properly placed within it |
| 05:02.38 | [TK]D-Fender | p3nguin: Second queue? EW. Nasty Stats killer... and a race condition :p |
| 05:04.04 | p3nguin | I was thinking of it more like automatic escalation after being on hold for too long. |
| 05:04.22 | p3nguin | It would involve more work, though. |
| 05:07.43 | carrar | work amazes me |
| 05:07.50 | carrar | I can sit and watch it for hours |
| 05:07.58 | p3nguin | Is there any typical timeout length that people use on a queue before dumping calls to some other place? |
| 05:08.18 | carrar | 24 hours |
| 05:09.50 | p3nguin | That's a long-ass time to wait on hold. |
| 05:10.06 | p3nguin | I think that's way too long. |
| 05:10.19 | p3nguin | I better go with more like 20 hours just to be safe. |
| 05:10.50 | carrar | 20..0625 |
| 05:10.52 | carrar | err |
| 05:10.54 | carrar | 20.0625 |
| 05:12.49 | p3nguin | See... |
| 05:12.53 | p3nguin | That's MUCH more reasonable than 24 hours. |
| 05:13.41 | carrar | Or you really might go nuts with 19.9375 |
| 05:13.55 | carrar | but thats probably too short |
| 05:13.56 | drmessano | What's the best way to transfer queue calls to the competition after they've been on hold with me for 5+ minutes? |
| 05:15.03 | p3nguin | haha |
| 05:15.07 | p3nguin | nice |
| 05:15.10 | carrar | It's easier to just have the competition port your number over to them |
| 05:18.00 | drmessano | So can if my queues have average wait times of 10+ minutes, can I port my number to the competition using an http request with curl, then turn in my business license with app_fax? |
| 05:22.14 | [TK]D-Fender | drmessano: Fear not.. they'll transfer themselves :) |
| 05:24.36 | tzanger | yay for asterisk manager interface |
| 05:24.44 | tzanger | libami is very handy too |
| 05:25.31 | joobie | TK, reason i dont have a single recording is the "ive been on hold for ages" recording advertises that we can call them back |
| 05:25.43 | joobie | but dont wnat that to be an option until they have been waiting a while in queue.. to reduce our own callbacks |
| 05:26.28 | joobie | .. how come the suggestion p3nguin put out there would potentially trigger a race condition? |
| 05:26.35 | p3nguin | If your single recording doesn't play that callback option for at least five minutes, that's how you handle it. |
| 05:27.11 | joobie | p3nguin, how would you time the queue handover to the next queue after X mins? |
| 05:27.16 | [TK]D-Fender | joobie: What part of "MoH starts from the beginning, make it ONE recording of the music AND announcements PRE-TIMED" are you not getting? |
| 05:27.51 | joobie | TK, i get it.. it just doesnt support what I want to do - which is play a different message for callers on hold for an extended period of time |
| 05:27.58 | p3nguin | You were concerned that a subsequent call would start in the middle of the recording? |
| 05:28.35 | p3nguin | If the single MoH file is, say, 10 minutes long... |
| 05:28.43 | joobie | p3nguin, naa.. once the user is in the queue, not sure how to achieve "after X seconds, pull them out of QueueA and put them into QueueB" |
| 05:28.56 | p3nguin | And at about 8 minutes in, it plays "we can call you back"... |
| 05:29.14 | joobie | ahhhhhh |
| 05:29.16 | joobie | sorry |
| 05:29.19 | joobie | i think i get you |
| 05:29.23 | joobie | sorry TK :P |
| 05:29.45 | [TK]D-Fender | swats joobie with his trusty ClueBat (tm) anyways |
| 05:29.46 | joobie | so you're suggesting to record the first announcement into the moh itself.. and then use periodic announcement after that? |
| 05:29.53 | [TK]D-Fender | joobie: NO |
| 05:30.13 | p3nguin | There's no reason you can't make a single .wav file that is 20 minutes in length. |
| 05:30.16 | joobie | turns the other cheek for the next cluebat swat |
| 05:30.26 | joobie | ahh |
| 05:30.28 | [TK]D-Fender | joobie: NO "announcement" file. 1 FUCKING MoH that STARTS with music, says message 1, CONTINUES music, then plays MESSAGE 2 |
| 05:30.42 | joobie | ok |
| 05:31.01 | joobie | i have multiple mp3's all as seperate files for this moh context.. mainly so i can start the moh and a random mp3 each time |
| 05:31.01 | p3nguin | Make it a full hour if you want. |
| 05:31.03 | [TK]D-Fender | resumes swatting |
| 05:31.05 | joobie | (to keep the queue music fresh) |
| 05:31.34 | [TK]D-Fender | joobie: MoH is NOT sealed for freshness... please observe "Best Before" dates |
| 05:31.47 | joobie | so unless i time it in each mp3 seperately (which sounds painful) or switch to one standard long mp3, i think im screwed? the one standard long mp3 i couldn't find a way to start at a random point |
| 05:32.11 | [TK]D-Fender | joobie: If you care, make your recordings a few seconds longer than your queue timeout and have MULTIPLE files all similarly timed. |
| 05:32.25 | p3nguin | You wouldn'te want it to start at a random point if you have spaced your voice messages at predetermined time intervals. |
| 05:32.47 | joobie | true |
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| 05:33.42 | [TK]D-Fender | joobie: When you dump from queue to queu you run into prioritization issues, cluttered queue logs, and worse |
| 05:33.43 | joobie | yea i can see how the queue log will start to get messy |
| 05:34.06 | joobie | also if i do a 20 min recroding and it getrs to the end.. then im screwed |
| 05:34.13 | joobie | will need some massive length mp3s |
| 05:34.31 | joobie | just thinking, i can probably create a few massive length mp3s.. all with the recordings embeded |
| 05:34.47 | joobie | then still cycle thru all the various massive mp3s to achieve the same.. sounds like a heap of work |
| 05:34.51 | p3nguin | The bat finally made an impression! |
| 05:35.00 | joobie | yes i can see the light now |
| 05:35.15 | joobie | thanks |
| 05:35.32 | joobie | .. the light is off-white tho |
| 05:35.34 | joobie | which sucls |
| 05:35.37 | joobie | -l+k |
| 05:35.52 | p3nguin | weird. |
| 05:37.25 | [TK]D-Fender | lets the hate flow through him... |
| 05:37.30 | [TK]D-Fender | Goood.. yes.... |
| 05:37.51 | joobie | deep breaths TK |
| 05:37.54 | joobie | in.. and out.. |
| 05:41.53 | [TK]D-Fender | fries joobie with FORCE LIGHTNING |
| 05:42.17 | joobie | change your nick to zeus |
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| 05:53.41 | [TK]D-Fender | checkout time, later all |
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| 07:33.38 | poller | Hello. Is there any simple way to log all outgoing calls? Like cdr-csv for incomming calls? |
| 07:34.01 | poller | (someone is making expensive long distance calls |
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| 07:40.15 | ChannelZ | cdr should already log outgoing calls to |
| 07:40.17 | ChannelZ | too |
| 07:40.39 | ChannelZ | you just have to determine what that means based on the calling and called channels for instance |
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| 07:58.58 | poller | ChannelZ: Thanks, yes it did. But the calls we're being charged for isn't in the logfile (i noticed now) |
| 07:59.41 | poller | Is there any setup that could result in an outgoing call not being logged? |
| 08:00.12 | poller | Or should I ask the SIP-provider to check their logs (maybe someone else is using our credentials) |
| 08:00.41 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
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| 08:08.52 | ChannelZ | The calls should be logged unless someone is going in and editing the files or something... if those calls are passing through Asterisk that is. |
| 08:09.44 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 08:09.48 | ChannelZ | Has someone perhaps registered their device directly with your ITSP? (maybe your credentials have been compromised and they are calling out directly, not through your system at all) |
| 08:10.06 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-dmjlebgwuylcpsar) |
| 08:10.19 | *** join/#asterisk Velfor (~master.or@velfor1.velfor.com) |
| 08:10.21 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:10.40 | Velfor | Hello to all Asterisk's users |
| 08:11.00 | ChannelZ | ahoyhoy |
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| 08:13.34 | Velfor | I just started a training session in a medium firm last week |
| 08:14.01 | ChannelZ | A medium firm? Like, psychics? |
| 08:14.07 | Velfor | I'm setting up a project about Videoconference |
| 08:14.39 | Velfor | (Sorry my english is kinda crappy, I've no vocabulary when I talk 'bout work |
| 08:14.41 | Velfor | ) |
| 08:15.26 | Velfor | so basically I'm in an office of a medium plant |
| 08:16.34 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 08:16.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:17.41 | *** join/#asterisk plutopathz (~plutopath@38.126.213.64) |
| 08:17.48 | Velfor | brb, I have to reboot |
| 08:19.10 | tzafrir | ChannelZ, now you'd have to wonder what a "medium plant" is |
| 08:19.39 | plutopathz | hi all, has anyone here ever successfully setup a linksys wip330? |
| 08:19.49 | tzafrir | A plant you have to smoke to be able to communicate? |
| 08:19.59 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:20.49 | ChannelZ | tzafrir: sounds like a harem... |
| 08:21.24 | ChannelZ | wonders if he was going to ask a question, or was calling out for help? "I'm being held in a medium plant..." |
| 08:21.33 | *** join/#asterisk Velfor (~master.or@velfor1.velfor.com) |
| 08:21.46 | Velfor | I'm back. |
| 08:22.27 | Velfor | So, has any of you ever used Asterisk for a videoconferencing solution ? |
| 08:23.04 | Velfor | I'm rather new in that world |
| 08:24.09 | Velfor | If anyone has answers and tips for me, do not hesitate, I'd be glad to have a quick chat. |
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| 08:34.46 | angryuser | i need to change a billing, can someone advice me one ? I need to review some, (ser + mera mvts ..) Names please |
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| 09:07.35 | Martinblr | what could be the problem of noise in PSTN line. This is happening only for some places... |
| 09:11.27 | Faustov | strong electromagnetic fields |
| 09:11.38 | Faustov | low quality cabling |
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| 09:19.47 | Martinblr | Faustov: its a low noise.. |
| 09:20.01 | Martinblr | Faustov: for low noise also the same cause.. |
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| 09:21.18 | mcr_mv | Martinblr, if you connect a phone in your pstn line, the noise disappears? |
| 09:21.35 | Martinblr | mcr_mv: yes |
| 09:22.41 | mcr_mv | try to set lower db gain in fxo/fxs channels |
| 09:23.18 | Martinblr | mcr_mv: ok |
| 09:24.26 | Martinblr | mcr_mv: you mean tx rx gain? |
| 09:25.05 | mcr_mv | yeas |
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| 09:32.26 | mcr_mv | (note: db gain is logaritmic, not lineal ) |
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| 10:15.36 | Godfather_ | hi |
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| 11:17.05 | cuco | tzafrir: ping |
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| 11:18.09 | tzafrir | cuco, pong |
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| 11:26.39 | SonOfGods | Hey, guys, what live cd is the best for using Asterisk? |
| 11:27.04 | SonOfGods | I'm using Asterisk only for dial-center |
| 11:27.23 | SonOfGods | is it need to be persistent? |
| 11:29.39 | MAbbas | how can an agent already talking to customer, invite another agent in conference? I only want to use asterisk feature for this? |
| 11:32.33 | MAbbas | ChannelZ: how can an agent already talking to customer, invite another agent in conference? I only want to use asterisk's feature for this? |
| 11:33.47 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:33.51 | tzafrir | mloe, do you ask for a live CD? For something persistent? What do you need the live CD for? |
| 11:38.28 | *** join/#asterisk ms110 (~shehzad@122.170.58.187) |
| 11:38.30 | mloe | well, I use Asterisk for connecting to SIP server and making a calls |
| 11:38.49 | ms110 | hellow |
| 11:38.54 | mloe | so I wanted to try to install Asterisk to Live Linux distribution (on USB Drive) |
| 11:39.03 | mloe | and set-up Asterisk |
| 11:39.21 | mloe | so when I load the Linux from the USB, just to start Asterisk and start doing my job |
| 11:39.23 | mloe | :) |
| 11:40.13 | ms110 | i can enable dtmf in meetme application using 'f' option, and that is working pretty cool. now i just need to silent the beep voice which is being heard while pressing dtmf... is that anyway we can do? |
| 11:40.40 | mloe | I'm going to install by this guide: type "do it yourself free caller id spoofing" in Google |
| 11:42.18 | mloe | I'm not sure is it a good idea, because if I load the distribution and start asterisk from some laptop for example.. it will be very hard to configure the wireless settings |
| 11:42.51 | mloe | I know that is better to run Asterisk to a permanent server but I don't have so.. I'll try with live linux distribution :) |
| 11:42.58 | mloe | I'm downloading Ubuntu right now |
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| 11:45.32 | mloe | Ubuntu NetBook remix looks nice |
| 11:45.37 | tzafrir | mloe, that's fine and dandy. I even built my own distro. But I'm not sure that's a good way to deploy Asterisk |
| 11:45.45 | tzafrir | What happens when you reboot? |
| 11:46.44 | mloe | I don't know :)) |
| 11:47.00 | mloe | that's why I need persisten feature |
| 11:47.10 | mloe | maybe this will be the way to save all of the information |
| 11:47.17 | mloe | I was thinking to download the smallest distribution |
| 11:47.23 | mloe | and just install asterisk |
| 11:47.31 | tzafrir | FWIW, my distro i sbased on Debian Live. I believe you should also be able to build latest Debian-Live on Ubuntu |
| 11:47.51 | mloe | how can I build a distro? :)) |
| 11:48.09 | mloe | that's not look easy |
| 11:48.12 | tzafrir | My config files are at http://updates.xorcom.com/iso/live-2.1.0_config/ |
| 11:48.13 | mloe | I'm new in Linux |
| 11:48.39 | tzafrir | hmm.. if you're new to Linux - don't waste your time on building a live CD. |
| 11:48.55 | mloe | yeah |
| 11:48.56 | mloe | :) |
| 11:49.05 | tzafrir | Interesting exercise. Can teach you a lot. But there are more useful things to do :-) |
| 11:49.36 | mloe | I just want to install Asterisk and run auto-dial center :) |
| 11:49.47 | mloe | but I downloaded a small linux distro |
| 11:50.03 | mloe | and there was a problem with my internet driver |
| 11:50.09 | mloe | and I didn't have internet.. haha |
| 11:50.16 | mloe | I'll try with another |
| 11:51.46 | MAbbas | how can an agent already talking to customer, invite another agent in conference? I only want to use asterisk's feature for this? |
| 11:52.06 | mloe | what do you think? |
| 11:52.08 | mloe | this: http://www.pendrivelinux.com/put-xubuntu-10-04-on-a-flash-drive-with-windows/ |
| 11:52.14 | mloe | or this: http://www.pendrivelinux.com/put-ubuntu-netbook-remix-10-04-on-flash-drive-in-windows/ |
| 11:53.36 | mloe | I'll try the second one |
| 11:53.42 | mloe | I'm coming back after 30 minutes |
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| 12:24.15 | UserL00ser | hello |
| 12:24.46 | UserL00ser | have some problem with making calls from sip via h323 trunk, there is one way audio only |
| 12:25.51 | UserL00ser | any idea what i should check? |
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| 12:29.13 | tzafrir | UserL00ser, on the same LAN? |
| 12:35.51 | UserL00ser | no H323 is on internet, but rtp port 10k-20k and 1720 are forwarded |
| 12:37.56 | UserL00ser | tzafrir, if you want i can show you how config looks |
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| 12:47.40 | elred_ | join #freeswitch |
| 12:47.42 | elred_ | lol |
| 12:47.43 | elred_ | sorry |
| 12:47.48 | Chainsaw | I refuse! |
| 12:48.00 | mcr_mv | ¬_¬U |
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| 13:06.50 | puzzled | hi |
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| 13:11.42 | mcr_mv | i've troubles with audio on asterisk. when a play a locution with cmd playback, background or read. Asterisk stops playing at random time and don't continue the dialplan. Only when I hangup the call, asterisk do the 'h' extension. |
| 13:12.08 | mcr_mv | any ideas what's wrong ? |
| 13:13.12 | [TK]D-Fender | mcr_mv: Show us the failed call with debug enabled for the channel placing the ccall |
| 13:13.14 | [TK]D-Fender | PASETBIN <- |
| 13:13.16 | [TK]D-Fender | ~pb |
| 13:13.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:13.29 | [TK]D-Fender | ^^^^^^^ |
| 13:13.30 | ms110 | hi |
| 13:15.24 | mcr_mv | ok, making debug info, in few minutes will be available... |
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| 13:24.51 | mcr_mv | http://pastebin.com/CznB4Y0g |
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| 13:30.06 | mcr_mv | [TK]D-Fender, as you can see asterisk don't drop anything to CLI, any idea? |
| 13:31.16 | [TK]D-Fender | mcr_mv: What kind of channel? pastebin your dialplan. Also confirm the size of that file you're playing |
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| 13:40.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.07 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 13:43.14 | [sr] | i'm in love with asterisk & ISDN |
| 13:43.14 | [sr] | :D |
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| 13:46.03 | mcr_mv | [TK]D-Fender, i was simplifing the dialplan to show u the problem and i discovered that the problem is the dial local |
| 13:46.22 | mcr_mv | now i will check if is a missconfiguration or a bug, thanks |
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| 13:49.31 | WIMPy | [sr]: You're easiliy impressed, eh? :-) |
| 13:49.46 | [sr] | WIMPy: yes, more with LCR than properly with asterisk :p |
| 13:51.05 | WIMPy | Yes, I thought about doing something similar for quite a while, but with LCR available, there's no motivation left. |
| 13:51.29 | WIMPy | But there are some configuration options on lcr I don't like. |
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| 13:57.19 | [sr] | WIMPy: hum like what for example? |
| 13:58.24 | WIMPy | Compatibility things like siemens date per MSN. I'd like such things per port. |
| 13:59.49 | [sr] | its would be better by port for sure |
| 14:00.22 | [sr] | the physical phones are connected by port, in my opinion makes more sense by port also |
| 14:00.46 | WIMPy | Well, I can see the idea. You might have a normal and a buggy phone on the same port. |
| 14:01.31 | WIMPy | But as I do all the dialplan in Asterisk, I don't have (and don't want) any peer configurations in lcr. |
| 14:02.10 | WIMPy | And in fact I'm very likely to have the same MSN configured on both phones in th above case. |
| 14:02.51 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:03.35 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 14:04.19 | WIMPy | I don't know which part is responsible, but I can't switch from en-block to overlap dialling any more. Which used to work. |
| 14:05.03 | [sr] | WIMPy: maybe if we suggest this to Andreas maybe he'll listen to us, what do you say? |
| 14:06.02 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
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| 14:06.22 | WIMPy | He took some weeks to reply to a bugfix I sent to him. |
| 14:06.48 | WIMPy | Maybe I should meet him for a longer discussion. |
| 14:06.54 | *** join/#asterisk Khratos (~jespinal@66.128.60.148) |
| 14:07.22 | Khratos | Good morning |
| 14:07.43 | Khratos | I would like to know if it is imperative to recompile asterisk-addons after upgrading asterisk |
| 14:08.15 | [TK]D-Fender | Khratos: Within the same brach... very rarely |
| 14:08.19 | [TK]D-Fender | branch |
| 14:08.53 | Khratos | Thank you [TK]D-Fender, I'm building slackware packages, and had that doubt |
| 14:09.20 | [sr] | WIMPy: i think he may need some help, not in coding but other stuff, he seems the only one in coding/bug report etc etc |
| 14:09.53 | WIMPy | Yes, I think he's done it all by himself. |
| 14:10.03 | WIMPy | And he does a LOT of other stuff as well. |
| 14:10.57 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 14:14.51 | [sr] | WIMPy: i think he doesn't have a bug tracker even, like mantis |
| 14:15.03 | [sr] | going to confirm and offer myself to setup one :) |
| 14:15.12 | WIMPy | None, that I'd know about. |
| 14:15.29 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 14:16.27 | WIMPy | I used to see him quite regularly a loooong time ago. That really needs re-activation, I think. |
| 14:19.23 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 14:20.29 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:24.56 | [sr] | WIMPy: sent him an email sending my help to setup a mantis for LCR+misdn bugtracker! |
| 14:24.58 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:26.29 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 14:26.49 | *** join/#asterisk mifadir (29891eb3@gateway/web/freenode/ip.41.137.30.179) |
| 14:26.52 | *** join/#asterisk Khratos (~jespinal@66.128.60.148) |
| 14:27.22 | [sr] | WIMPy: unless asterisk may have native support for ISDN/PSTN, LCR is the best for ISDN i'm sure |
| 14:30.11 | *** join/#asterisk asphere (~ardavis@rrcs-69-193-16-71.nys.biz.rr.com) |
| 14:30.58 | WIMPy | Err, what do you call dahdi then? |
| 14:31.32 | MAbbas | [TK]D-Fender: is there any simpler way for n-way transfer than http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 14:32.10 | [TK]D-Fender | MAbbas: First that isn't "transfer", next you should provide som actual important details about the devices involved in your scenario |
| 14:32.13 | grandpapadot | Woot! Check out my new app - http://pocketoperator.com |
| 14:33.41 | MAbbas | <[TK]D-Fender> I have 5 sip softphones ..and want to do n-way transfer(or whatever they call it) |
| 14:34.09 | [TK]D-Fender | MAbbas: Who is"they"? And that says N-way CALL, not TRANSFER |
| 14:34.40 | MAbbas | <[TK]D-Fender> ok, n-way call? |
| 14:35.03 | [TK]D-Fender | MAbbas: That sample just uses a Meetme. |
| 14:35.05 | [sr] | WIMPy: more nativelly, ok i know the modules will have always to be kernel mode do it won't be possible to skip the kernel part, but something easier.. |
| 14:35.17 | [TK]D-Fender | MAbbas: So FFS just place calls and transfer them yourself |
| 14:35.32 | MAbbas | <[TK]D-Fender> FFS? |
| 14:35.36 | [TK]D-Fender | MAbbas: Or get a device that can do more than a 3-way call |
| 14:35.39 | [TK]D-Fender | ~ffs |
| 14:35.39 | infobot | i heard ffs is for f**k's sake, or for fine's sake. UCB's Fast File System |
| 14:36.35 | tzafrir | WIMPy, there's also chan_misdn and the external chan_capi |
| 14:36.37 | MAbbas | <[TK]D-Fender> well, I only want to use n-way feature of asterisk |
| 14:37.20 | [TK]D-Fender | MAbbas: There is no such feature |
| 14:38.41 | MAbbas | <[TK]D-Fender> then is this "http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO" some kind of hack to get n-way call functionality? |
| 14:39.30 | [TK]D-Fender | MAbbas: Do you have EYES? Are you even LOOKING at what its doing? Do you see some fascinating channel-driver options saying "magic button for N-way calling" in there? |
| 14:39.30 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:39.56 | *** join/#asterisk s519 (~steve@87-194-151-213.bethere.co.uk) |
| 14:40.21 | [TK]D-Fender | MAbbas: Since when is the MeetMe dialplan app a "call feature"? Place call. TRANSFER caller there. Place another call, trasnfer them in. Join YOURSELF. |
| 14:43.21 | MAbbas | <[TK]D-Fender> Can we have such thing, we have customer and agent talking and agent invite supervisor in .. and now all 3 are in conference now. Agent can leave this conference to let supervisor and customer talking? |
| 14:43.22 | *** join/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 14:43.50 | [TK]D-Fender | MAbbas: Its a meetme, ANYONE can leave with the rest staying. |
| 14:43.52 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:44.36 | WIMPy | tzafrir: chan_misdn was, not is, but chan_capi is an option, but doesn't support NT mode, IIRC. |
| 14:48.57 | MAbbas | <[TK]D-Fender> lets say, agent transferred both(supervisor n customer) to meetme. Now how can agent join the bridge himself? As When a logged-in agent dials something nothing happens? |
| 14:50.02 | [TK]D-Fender | MAbbas: Now how can agent join the bridge himself <- Dial an extension that joinds the meetme |
| 14:50.28 | mifadir | how to make call forking in asterisk? |
| 14:51.23 | [TK]D-Fender | mifadir: Meaning? |
| 14:52.14 | mifadir | if 1 personne's away or anvailble it pass the call to 2nd, or 3rd |
| 14:52.29 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 14:52.29 | [TK]D-Fender | mifadir: Just dial them back to back. |
| 14:53.12 | mifadir | no with gotoif status |
| 14:53.22 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:54.08 | MAbbas | <[TK]D-Fender> When agent is logged in, if he dails something on the same line. It does not executes in dialplan. However when agent is taking calls he can do that. Are u trying to say the logged-in agent should dial bridge extension from other line? |
| 14:55.50 | ms110 | hi |
| 14:55.52 | ms110 | <PROTECTED> |
| 14:56.51 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 14:58.03 | [TK]D-Fender | MAbbas: How observant. |
| 14:58.20 | [TK]D-Fender | ms110: nope |
| 14:58.26 | MAbbas | <[TK]D-Fender> How can a logged-in agent join the bridge. I dial bridge extension and nothing happens? |
| 14:58.43 | *** part/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 14:58.48 | [TK]D-Fender | mifadir: You may or may not need to check status at all. If you feel like it ... GotoIf <- |
| 14:59.17 | [TK]D-Fender | MAbbas: AgentLogin = SHIT. You ahve to LEAVE the damn queue and place a NEW CALL to the MeetMe. |
| 14:59.53 | ms110 | thanks for replying, [TK]D-Fender> |
| 15:00.52 | [TK]D-Fender | ms110: Wish it was better enws. |
| 15:00.52 | MAbbas | <[TK]D-Fender> so a logged-in agent can not join a bridge.. |
| 15:01.10 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:01.18 | [TK]D-Fender | MAbbas: You have to call MeetMe. This isn't Raw-Cat Sigh-Hence |
| 15:01.29 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 15:01.46 | MAbbas | <[TK]D-Fender> ok, thanks ! |
| 15:02.20 | ms110 | [TK]D-Fender: it would be much better if i could silent beep, :) |
| 15:03.03 | wcselby | o/ |
| 15:03.23 | [TK]D-Fender | \o |
| 15:03.52 | drmessano | Real guitars are for old people |
| 15:04.02 | wcselby | that raw-cat sigh-hence is purty hard sumtimes |
| 15:04.16 | [TK]D-Fender | drmessano: Followed that chat from last night did you? :) |
| 15:04.22 | wcselby | queue show csvc_initial |
| 15:04.24 | wcselby | bleh |
| 15:04.40 | drmessano | I watched the whole thing |
| 15:04.52 | drmessano | Funniest one since the WoW episode |
| 15:05.08 | [TK]D-Fender | drmessano: I should look that one up then... |
| 15:05.19 | drmessano | Now we just need an Asterisk equiv of Heroin Hero |
| 15:05.34 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:05.46 | drmessano | "I'm chasing the call, but it never completes.. Whoa, where did it go.." |
| 15:08.54 | [TK]D-Fender | drmessano: AT&T <- |
| 15:09.18 | [TK]D-Fender | CAN YOU HEAR ME N...*CLICK* |
| 15:09.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:13.34 | [sr] | this chan_capi may be interesting |
| 15:14.19 | *** join/#asterisk Warp4 (~robert@firewall-a.buf.ny.i-evolve.net) |
| 15:14.22 | drmessano | Yeah, tell me about it |
| 15:14.48 | [sr] | but LCR is what im going to use :P |
| 15:14.59 | *** join/#asterisk grEvenX (~even@188.148.131.74) |
| 15:16.24 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 15:17.15 | *** join/#asterisk imox1234 (~imox1234@p4FC5C514.dip0.t-ipconnect.de) |
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| 15:21.27 | [sr] | LCR seems much more flexible than this native asterisk chan_capi |
| 15:22.24 | imox1234 | hello, i get this error, then a peer call another peer on my asterisk http://pastebin.com/TEgsCzQb |
| 15:22.33 | *** join/#asterisk telnettech (~telnettec@cpe-24-33-229-14.insight.res.rr.com) |
| 15:23.54 | *** join/#asterisk bluebug (~ernix@61.206.115.177.static.zoot.jp) |
| 15:24.47 | imox1234 | what can i do :-) ? |
| 15:25.19 | WIMPy | [sr]: And it supports two features, which seems to be twice as much as the others. |
| 15:25.29 | WIMPy | But still very little :-( |
| 15:28.15 | bluebug | Hi, I'm a newbe about asterisk. I installed asterisk and asterisknow via macports on mac OSX 10.6. It seems no problems while installing but I don't know how to use them now. Does anyone know a good reference? |
| 15:28.54 | wcselby | ~book |
| 15:28.55 | 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 |
| 15:29.00 | wcselby | bluebug ^^^^^ |
| 15:29.10 | [sr] | WIMPy: the two features, you're saying chan_capi or lcr? |
| 15:29.48 | WIMPy | lcr |
| 15:29.49 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:30.03 | [sr] | WIMPy: ah, you'r saying TE and NT mode's right? |
| 15:30.29 | WIMPy | No, HOLD and TP :-) |
| 15:30.33 | bluebug | wcselby, I couldn't find Japanese books about asterisk though. hehe. |
| 15:30.49 | [sr] | WIMPy: hum.. don't know that yeat :P |
| 15:30.53 | *** join/#asterisk goodjoke (~thayes@66.195.70.253) |
| 15:30.55 | WIMPy | Where TP interestingly works globally, not per port. |
| 15:31.45 | [sr] | hey an offtopic, did you saw portugal-korea, 7-0 :D |
| 15:32.43 | WIMPy | does not watch any footie, ever. |
| 15:33.22 | [sr] | heeh, well, i don't like football at all... only in world cup, after that... dont see |
| 15:34.26 | WIMPy | can't take sports events seriousely, until they understand ISO-3166 ;-) |
| 15:34.47 | [sr] | lol |
| 15:34.52 | wcselby | bluebug - did you request a reference in Japanese? |
| 15:37.41 | bluebug | wcselby, I don't care. It seems work well asterisk in a process list. But I don't know how to handle asterisk now. I can find only the html file "index.html" under my /opt directory. |
| 15:40.03 | wcselby | bluebug - asterisknow comes with a web gui. you would point your web browser to the IP address of your asterisk box and then go from there. |
| 15:40.17 | *** part/#asterisk goodjoke (~thayes@66.195.70.253) |
| 15:41.58 | WIMPy | Ok, time to bring some hot coal and slices of animals together. o/~ |
| 15:43.31 | tzafrir | WIMPy, together? Is that a good idea? |
| 15:43.51 | tzafrir | recommends putting the animals in a cage |
| 15:43.58 | bluebug | wcselby, thank you for your reply, that means asterisknow has its own http daemon inside, right? |
| 15:44.33 | wcselby | bluebug - not sure, I haven't used it in a long time. |
| 15:44.38 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 15:44.56 | wcselby | tzafrir - I think he simply meant he was going to bring the hot coals near the slices of animals |
| 15:45.22 | tzafrir | wcselby, animals in a cage, coal in a cage, whatever |
| 15:45.38 | wcselby | although i have seen a few shows that suggest putting the animal slices directly on the hot coals, but that just seemed crazy to me |
| 15:46.58 | tzafrir | WIMPy, I believe chan_dahdi/trunk supports those as well |
| 15:47.27 | tzafrir | (and bristuff does as well. but I try to keep away from it) |
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| 15:48.54 | [TK]D-Fender | bluebug: AsteriskNOW is of course only if you have a sparesystem laying around |
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| 16:02.09 | Khratos | [TK]D-Fender: in your oppinion, would it be reliable for a script to use the links named as 'asterisk-1.4-current' to get latest version of each package? |
| 16:02.55 | [TK]D-Fender | Khratos: I don't do "packaging, sorry |
| 16:03.49 | Khratos | No prob, I asked cause in the download server, there are links like asterisk-1.4-current, and I don't know if they can be removed in the future |
| 16:05.18 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:07.19 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 16:08.10 | tzafrir | Khratos, generally, yes. However, don't assume that the name of the directory inside it is asterisk-1.4.33 |
| 16:08.17 | tzafrir | As it may change |
| 16:10.05 | Khratos | Thanks!, for the directory inside I normally use PKGDIR=`tar -tf some-pagage.tar.gz | head -n1` |
| 16:10.47 | p3nguin | I explicitly list the version number when I am going to compile and package a new version. |
| 16:11.20 | p3nguin | What distro are you making a package for? |
| 16:12.05 | Khratos | Slackware |
| 16:12.35 | tzafrir | Khratos, I prefer to download the version from releases/ , to be sure of the exact version |
| 16:15.19 | Khratos | I'll better try that, and to get the latest stable version for the last changelog of each package |
| 16:18.12 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 16:23.04 | grinder13 | hello guys! i am trying to compile Asterisk from the SVN trunk. I am interested specifically in the security features of IAX and SIP/RTP. do i need to install any special development libraries eg for TLS? I am working on Debian/Ubuntu servers |
| 16:24.57 | [TK]D-Fender | [12:03]<Khratos>No prob, I asked cause in the download server, there are links like asterisk-1.4-current, and I don't know if they can be removed in the future <- module load res_psychic.so |
| 16:34.11 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
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| 16:40.45 | imox1234 | i want use asterisk realtime. i have set the settings in the res_config_mysql and in the Extconfig.conf. i think i have to set the general information like in the sip.conf or ? |
| 16:41.01 | imox1234 | when i load the module i get this error MySQL realtime: no requirements setting found, using 'warn' as default. |
| 16:41.49 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-cpaoeurowxzcwupx) |
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| 16:42.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:44.31 | imox1234 | ??Ã |
| 16:48.29 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:48.30 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:49.44 | thehar | welcome back to power, digium. |
| 16:52.03 | *** join/#asterisk ariel_ (~chatzilla@216.7.159.250) |
| 16:53.12 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:53.27 | ariel_ | tzafrir: you have a few minutes to help out with a xorcom astribank? |
| 16:54.07 | tzafrir | ariel_, yes |
| 16:54.17 | ariel_ | can I PM you? |
| 16:54.40 | tzafrir | yes |
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| 16:58.24 | *** join/#asterisk structz (~structz@gandalf.ai.unesp.br) |
| 16:58.27 | structz | Hello all |
| 16:58.46 | structz | does anyone knows how to get the ip from the asterisk server which is originating the call!? |
| 17:01.34 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
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| 17:03.58 | structz | ? |
| 17:04.23 | wcselby | logon to the box, run ifconfig? |
| 17:04.39 | wcselby | tcpdump? |
| 17:04.42 | wcselby | wireshark? |
| 17:04.52 | structz | wcselby, no, when a call is receibed |
| 17:04.54 | structz | received* |
| 17:05.04 | wcselby | by who? |
| 17:05.13 | structz | server 1 > server2 |
| 17:05.22 | structz | onto server2 get ip from the iax call |
| 17:05.45 | structz | from server1 |
| 17:06.04 | imox1234 | can somebody help me by asterisk realtime ? |
| 17:06.09 | wcselby | netstat ? |
| 17:06.17 | wcselby | ~ask |
| 17:06.18 | 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. |
| 17:06.24 | *** join/#asterisk Alagar (~Administr@122.164.33.116) |
| 17:06.28 | wcselby | imox1234, structz - ^^^^^^^^ |
| 17:06.44 | imox1234 | member:structz - ^^^^^^^^ ?? |
| 17:06.58 | structz | wcselby, that will show a lot of stuff, i need the the ongoing call |
| 17:07.06 | structz | ? |
| 17:07.36 | imox1234 | what ? |
| 17:07.38 | imox1234 | sorry |
| 17:07.40 | imox1234 | :D |
| 17:07.42 | structz | :( |
| 17:07.57 | structz | confused |
| 17:08.09 | grinder13 | anyone for that guys, please? i am trying to compile Asterisk from the SVN trunk. I am interested specifically in the security features of IAX and SIP/RTP. do i need to install any special development libraries eg for TLS? I am working on Debian/Ubuntu servers |
| 17:08.12 | imox1234 | wcselby: i have configurate the .conf |
| 17:08.15 | wcselby | structz - think about your question and then ask it again, giving more details and specifics. |
| 17:08.19 | imox1234 | but what should i do know ? |
| 17:08.29 | structz | wcselby, i'll exemplify |
| 17:08.45 | structz | server1 ip = 1.1.1.1 , server2 ip = 2.2.2.2 |
| 17:08.47 | wcselby | grinder13 - are you getting any errors when trying to compile? |
| 17:09.35 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:10.33 | structz | when a user registered on server 1 place a call and goes to server2, via IAX2(no trunk) before it dials to the destination exten i need to get the server2 ip and put into a var(this is a example with 2 servers only, in my scenario a have 23) |
| 17:10.33 | structz | wcselby, see if you can understand now |
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| 17:11.16 | wcselby | structz - phone is registered to server1, you need IP of server2, to put into a variable on server1 or server2 or somewhere else entirely? |
| 17:11.35 | *** join/#asterisk skymeyer (~skymeyer@91.183.54.9) |
| 17:11.51 | grinder13 | wcselby: no errors during configure and make, but I am trying to be sure that everything is fine. i 've checked the config.log and it says that openssl is there, but i am just wondering if it needs something more, like gnutls |
| 17:12.00 | structz | wcselby, lets try again |
| 17:13.44 | wcselby | grinder13 - best way to find out is to try it, and then if it works you're good, if not you'll have an idea what's missing |
| 17:14.09 | structz | wcselby, try this http://pastebin.com/NaLc3HQb |
| 17:14.14 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:14.40 | grinder13 | well ok, thanx wcselby |
| 17:14.48 | thehar | iOS4 released! |
| 17:16.07 | structz | thehar, IPHONE OS4 update released? |
| 17:16.10 | thehar | yes |
| 17:16.14 | thehar | http://forums.macrumors.com/showthread.php?t=943997 |
| 17:16.16 | thehar | links there |
| 17:16.26 | imox1234 | structz: can you give me a good link to configurate asterisk realtime ? |
| 17:17.04 | structz | i mean insetad using sip.conf to add users, use some database like mysql? |
| 17:17.04 | wcselby | ~book |
| 17:17.05 | 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 |
| 17:17.08 | thehar | and pandora was updated this morning for multi-tasking |
| 17:17.11 | wcselby | imox1234 ^^^^ |
| 17:17.16 | wcselby | read the book, the chapter on realtime |
| 17:18.11 | *** join/#asterisk ariel_ (~chatzilla@216.7.159.250) |
| 17:18.33 | structz | wcselby, did you understand my need? |
| 17:18.44 | structz | from the pastbin!? |
| 17:18.48 | structz | pastebin* |
| 17:19.09 | structz | thehar, where is the news? |
| 17:19.28 | thehar | structz: for pandora? |
| 17:19.35 | structz | iOS4 |
| 17:19.49 | thehar | http://forums.macrumors.com/showthread.php?t=943997 |
| 17:19.56 | thehar | links there to direct ipsw |
| 17:20.32 | structz | tkz |
| 17:21.21 | thehar | np |
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| 17:23.06 | *** join/#asterisk |Rain| (rain@ev.il.net) |
| 17:23.42 | |Rain| | I don't suppose anyone here does T.38 with a MAXAscendLucent TNT? |
| 17:25.15 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 17:25.53 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-171-203.clienti.tiscali.it) |
| 17:26.07 | structz | wcselby, there? |
| 17:26.12 | grinder13 | does anyone know which is the standard solution for compiling g729 in Asterisk? Google revealed some cases. |
| 17:26.54 | *** join/#asterisk clintc (~clintc@n128-227-87-199.xlate.ufl.edu) |
| 17:27.06 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-178-117.mia.bellsouth.net) |
| 17:28.25 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 17:29.32 | *** join/#asterisk Hotpuppy (~brian@75.148.177.89) |
| 17:32.24 | structz | wcselby, there ? |
| 17:36.30 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 17:38.06 | [TK]D-Fender | grinder13: There is none. You download and install the binary provided direct from Digiuum |
| 17:41.44 | *** join/#asterisk VultureZ (~chuck@c-76-106-198-76.hsd1.fl.comcast.net) |
| 17:44.25 | grinder13 | [TK]D-Fender, what about that then: http://asterisk.hosting.lv/ is it trustworthy? i just care for educational purposes, specifficaly for my project in Uni. alternatively, I 'll just use speex and g711 |
| 17:45.09 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 17:45.14 | [TK]D-Fender | "educational purposes" .... *COUGH*BULLSHIT*COUGH* |
| 17:45.22 | [TK]D-Fender | grinder13: No "education" there |
| 17:45.35 | [TK]D-Fender | has this nasty allergy |
| 17:45.51 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:47.04 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 17:47.22 | grinder13 | I don't care if you believe me. you can easily check my ip if you want to see from where I am connecting |
| 17:47.55 | grinder13 | my project is about voip performance with and without encryption and MPLS. happy now? |
| 17:48.14 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 17:48.19 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 17:49.17 | *** join/#asterisk gavimobile (~gavriel@77.124.156.153) |
| 17:49.41 | luke-jr | just curious, but any idea why ${CDR(clid)} is always "" in the CDR? |
| 17:49.53 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 17:51.03 | [TK]D-Fender | grinder13: Then go buy a license |
| 17:51.21 | [TK]D-Fender | luke-jr: Dump the call you're logging to compare |
| 17:53.35 | luke-jr | [TK]D-Fender: it shows a correct value in 'show channel' |
| 17:53.58 | [TK]D-Fender | luke-jr: Are they correctly wrotten to CDR? |
| 17:54.01 | [TK]D-Fender | written* |
| 17:54.17 | [TK]D-Fender | (the actual file) |
| 17:54.29 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 17:55.27 | luke-jr | [TK]D-Fender: that's what I'm saying is wrong |
| 17:55.34 | luke-jr | what is there besides the file? |
| 17:55.42 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 17:55.54 | [TK]D-Fender | luke-jr: Hrm... rather odd... what backend aer you using? |
| 17:56.31 | luke-jr | the default in 1.2 |
| 17:56.34 | luke-jr | just a raw text file |
| 17:56.50 | [TK]D-Fender | luke-jr: Hrm.... You at least on the altest? |
| 17:57.03 | [TK]D-Fender | luke-jr: Given it is pretty decrepit.... |
| 17:57.05 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 17:57.09 | [TK]D-Fender | latest* |
| 17:57.21 | luke-jr | 1.4.18.1 actually |
| 17:57.28 | luke-jr | 1.4.19 broke with my setup |
| 17:57.50 | [TK]D-Fender | luke-jr: You just jumped branches there. Also that is massively behind |
| 17:57.56 | [TK]D-Fender | @ 33 now |
| 17:58.00 | luke-jr | yeah |
| 17:58.08 | luke-jr | nobody ever fixed the bugs introduced in 1.4.19 AFAIK |
| 17:58.31 | gavimobile | which type of asterisk compatible hardware is the easiest to setup/install? |
| 17:58.33 | luke-jr | but it's been so long, I don't recall what they were anymore |
| 17:58.45 | luke-jr | one of these days I'll get around to redoing my whole system with 1.6 |
| 17:59.16 | gavimobile | for example I have an spa2102, and its rather easy to set up. I need to upgrade my hardware and im not sure what type of hardware to get. im looking for cheap and easy to install |
| 18:00.09 | imox1234 | structz: i dont see any peer :-) sorry can you help me again please |
| 18:00.17 | drmessano | That narrows it down to 126,963 products |
| 18:00.28 | drmessano | What type of hardware do you NEED? |
| 18:00.34 | luke-jr | curiously, my configured format in cdr_custom doesn't seem to match the CDR entries at all |
| 18:01.43 | luke-jr | accountcode is skipped, and there's an extra blank field before lastapp |
| 18:02.14 | *** join/#asterisk ariel_ (d8079ffa@gateway/web/freenode/ip.216.7.159.250) |
| 18:02.53 | gavimobile | drmessano: something similar to the spa2102 and I can connect more than 4 extentions to it and 2 lines |
| 18:03.27 | ariel_ | tzafrir: I got disconnected and could not log back in via the chatzilla |
| 18:03.28 | drmessano | Maybe an 8 port digium card, or a channel bank |
| 18:03.35 | luke-jr | wait no, the blank field is dstchannel; the clid seems to be after dcontext |
| 18:03.35 | gavimobile | does that narrow down to any less than 126,963 products? |
| 18:03.54 | [TK]D-Fender | luke-jr: Upgrade and collect yourself... |
| 18:04.03 | luke-jr | ... |
| 18:04.12 | drmessano | GET AHOLD OF YOURSELF |
| 18:04.16 | luke-jr | [TK]D-Fender: too risky when I don't have a bunch of free time to troubleshoot :( |
| 18:04.32 | drmessano | Over macho grande? |
| 18:04.40 | drmessano | No, you never get over macho grande |
| 18:04.48 | gavimobile | drmessano: http://cgi.ebay.com/Digium-TDM814BF-1-FXS-4-FXO-Card-1TDM814BF-/130326137511?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item1e580b0aa7 like this |
| 18:04.54 | [TK]D-Fender | luke-jr: Well its going to be challenging to find people to help diagnose a wheel that's likely already been fixed. |
| 18:05.01 | luke-jr | true |
| 18:05.04 | gavimobile | hi tzafrir |
| 18:05.11 | luke-jr | any way to ask a running Asterisk what *it* thinks the field order is? |
| 18:05.16 | [TK]D-Fender | gavimobile: Forget PCI for FXS. |
| 18:05.30 | [TK]D-Fender | luke-jr: check yoru CDR confs |
| 18:05.39 | luke-jr | I did, this doesn't match it |
| 18:05.47 | [TK]D-Fender | gavimobile: For lines... depends what you need. |
| 18:06.05 | [TK]D-Fender | luke-jr: go prep something comprehensive for us then. |
| 18:07.02 | tzafrir | gavimobile, hi |
| 18:07.23 | gavimobile | tzafrir: still havnt found hardware yet |
| 18:08.28 | gavimobile | [TK]D-Fender: I am not sure |
| 18:08.51 | structz | wcselby, dead? |
| 18:09.34 | [TK]D-Fender | gavimobile: Don't ask for advise on equipment to accomplish a goal you haven't defined |
| 18:10.08 | wcselby | structz - no, I'm very much alive, thanks for asking though. |
| 18:10.15 | gavimobile | [TK]D-Fender: ok.. |
| 18:10.22 | wcselby | i don't know how to help you, maybe ask some of the other people in the channel |
| 18:10.24 | wcselby | ~ask |
| 18:10.25 | 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. |
| 18:11.06 | structz | wcselby, thanks for the reply, sorry to bother |
| 18:12.23 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:13.02 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-178-117.mia.bellsouth.net) |
| 18:17.54 | jdoe | any disadvantage to using dnat to map a couple of ports to asterisk? (5060, 5061 etc.) |
| 18:18.08 | gavimobile | [TK]D-Fender: I would like to acomplish the tasks of managing a basic midsized automated ivr. |
| 18:18.10 | Mark22 | Hello, I do get messages like "file.c:650 ast_openstream_full: File avond_em.wav does not exist in any format". I get the message when I try to use Background() and when I do "moh show files" I see avond_em listed (as expected) :S |
| 18:18.28 | [TK]D-Fender | gavimobile: Loose terms, no meat to it |
| 18:18.46 | [TK]D-Fender | Mark22: show us your complete attempt |
| 18:18.46 | gavimobile | [TK]D-Fender: lol ill find some meat |
| 18:19.28 | [TK]D-Fender | gavimobile: I have an IVR as well. Clearly you need a channelized DS3. Go call your telco. |
| 18:20.26 | grinder13 | does anyone know which library is needed in order to compile Asterisk with OGG support? i 've installed libogg and libogg-dev, but even so it seems that something is missing |
| 18:21.03 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 18:21.28 | Mark22 | [TK]D-Fender: http://yourpaste.net/5690/ or what do you need? |
| 18:22.04 | [TK]D-Fender | Mark22: Executing [s@ivr:5] BackGround("SIP/trunk-sinnerg-00000004", "avond_em.wav") in new stack <--- do NOT specify extensions with apps like this |
| 18:23.37 | Mark22 | same error after changing that http://yourpaste.net/5691/ (.wav/.ulaw/.alaw/.gsm for avond_em is availble) |
| 18:24.00 | gavimobile | clearly I need the channelized ds3 |
| 18:24.04 | gavimobile | clearly |
| 18:24.05 | [TK]D-Fender | Mark22: Show us the file in detail |
| 18:24.21 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 18:24.41 | Mark22 | extensions.conf or another file? |
| 18:24.43 | *** join/#asterisk picard1421 (~chatzilla@ip70-181-112-218.oc.oc.cox.net) |
| 18:24.47 | [TK]D-Fender | Mark22: the SOUND file |
| 18:24.58 | picard1421 | hey guys i have a quick question that is really bugging me with the callback module |
| 18:24.59 | Mark22 | ok |
| 18:25.01 | [TK]D-Fender | Mark22: full dump of the folder it's in, spec, etc |
| 18:25.13 | [TK]D-Fender | picard1421: What "callback module"? |
| 18:25.25 | picard1421 | basically i have international calls comming in and im using the callback module and i have a2billign setup so that the 011 is added to everycall.. but i want the 011 to be added to the callback module |
| 18:25.42 | picard1421 | so if someone from usa like 1xxxxxxxxxx calls in the call back will go to 011 1 xxx xxx xxxx |
| 18:25.58 | [TK]D-Fender | picard1421: a2billing isn't supported here. |
| 18:26.01 | picard1421 | or if someone form uk calls in 44xxxxxxxx then the callback will go to 011 44 xxx xxxxxxx |
| 18:26.12 | picard1421 | its not a2billing this is within asterisk itself.. im using the asterisk callback module not a2billing's |
| 18:26.24 | picard1421 | just want to prefix all callback call's with 011 |
| 18:26.26 | [TK]D-Fender | picard1421: what "callback module"? |
| 18:26.33 | [TK]D-Fender | picard1421: I see no such thing |
| 18:26.37 | picard1421 | hmm? |
| 18:26.42 | picard1421 | its a script |
| 18:26.45 | picard1421 | the caomes with asterisk |
| 18:26.47 | picard1421 | one sec |
| 18:26.52 | [TK]D-Fender | picard1421: then it isn't Asterisk. Its YOUR script |
| 18:27.28 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:27.52 | picard1421 | i gues my question is... where would i add the 011 within the script itself or in dialing rules etc.. |
| 18:28.08 | Micc | Is there a volume setting in IAX2 or SIP on asterisk 1.6.2? I'm getting a lot more echo after upgrading to 1.6.2 from 1.6.1. |
| 18:28.10 | Mark22 | http://yourpaste.net/5692/ << information about the sound files |
| 18:28.19 | [TK]D-Fender | picard1421: What " |
| 18:28.31 | [TK]D-Fender | picard1421: What "script"? Again... there is no such thing in Asterisk |
| 18:28.52 | [TK]D-Fender | Micc: SIP/IAX2 are not sources of echo |
| 18:29.09 | Micc | TKD-Fender, too much audio are sources of echo. |
| 18:29.36 | picard1421 | i gues it might be more a trixbox thing then? |
| 18:29.37 | [TK]D-Fender | Micc: Endpoints are, nt * itself |
| 18:29.48 | [TK]D-Fender | picard1421: Indeed. Which is also not supported here |
| 18:30.02 | picard1421 | kk thanks |
| 18:30.14 | Micc | TKD-Fender, right, but it can be changed by lowering the audio output to the PSTN. |
| 18:30.17 | [TK]D-Fender | Mark22: -- Executing [s@ivr:5] BackGround("SIP/trunk-sinnerg-00000005", "avond_em") in new stack <-- you didn't specify the FOLDER therefor * uses the defaul sounds folder which is clearly not the MoH folder |
| 18:30.27 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 18:30.44 | [TK]D-Fender | Micc: taht should be done in the termination point's interface config. Not at the SIP level |
| 18:30.48 | Mark22 | how should/could I specify the folder or what folder does asterisk use? |
| 18:31.00 | [TK]D-Fender | Micc: VoIP protocols are expected to be pre-normalized |
| 18:31.10 | [TK]D-Fender | Mark22: Specify an absolute path |
| 18:31.13 | Micc | TKD-Fender, could it be a change in dahdi? |
| 18:31.42 | [TK]D-Fender | Micc: For levels, shouldn't be. For EC effectiveness, very possible |
| 18:31.50 | Micc | I did upgrade dahdi too. |
| 18:32.05 | [TK]D-Fender | Micc: try out different EC options |
| 18:32.22 | Mark22 | [TK]D-Fender: thank you, that works :) |
| 18:32.25 | Micc | Its using the same mg2 option it was before. I could try some others I guess. |
| 18:33.12 | [TK]D-Fender | Micc: OSLEC seems to be pretty good. I'd try that next. If your card is under warranty then you're likely eligible for HPEC as well |
| 18:34.55 | Micc | Its an older card, not under warranty, not even supported anymore I guess. |
| 18:35.05 | Micc | Thats what they told me last time I called digium. |
| 18:35.20 | Micc | They said it was end of lifed a few years ago. |
| 18:35.37 | [TK]D-Fender | Micc: that surpasses even warranty by far |
| 18:35.41 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 18:35.52 | [TK]D-Fender | Micc: OSLEC time first. HPEC is also $10/channel.. could be viable |
| 18:36.05 | [TK]D-Fender | Micc: Fire it off in order |
| 18:36.43 | structz | Hey all can somebody help me with this issue >> http://pastebin.com/NaLc3HQb |
| 18:37.42 | Micc | system.conf says Valid echo cancellers are: mg2, kb1, sec2, and sec. |
| 18:38.42 | Micc | DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) |
| 18:38.55 | [TK]D-Fender | structz: then set a channel variable with that IP before placing the IAX2 call. The receiving end should be able to access it via standard variable inheritance rules |
| 18:39.13 | [TK]D-Fender | Micc: OSLEC is a 3rd-party addon <- |
| 18:39.33 | [TK]D-Fender | Micc: Google-up some instructions |
| 18:39.35 | structz | [TK]D-Fender, my problem is that the route is found via DUNDi so i can't do anything before the Dial... |
| 18:39.38 | Micc | ok |
| 18:39.47 | [TK]D-Fender | structz: Then change your dialplan. |
| 18:40.12 | [TK]D-Fender | structz: Match it internally first then bring DUNDi into the picture |
| 18:40.13 | structz | in order to change i have to quit DUNDi |
| 18:40.33 | structz | becuse is too much numberss |
| 18:40.39 | structz | i mean several locations |
| 18:40.45 | structz | lots of servers to do it |
| 18:40.55 | structz | ain't functional |
| 18:42.21 | [TK]D-Fender | structz: Sure it is. Separate all your dundi searches from internal patterns. Set the var in a catchall then JUMP to a DUNDi lookup |
| 18:43.17 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 18:43.18 | structz | [TK]D-Fender, looks like a solution but kinda don't know how to can you help me out?! |
| 18:43.18 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 18:43.44 | [TK]D-Fender | structz: Go make your catch-all for patterns you'd look elsewhere for. |
| 18:44.22 | structz | hmm might have a problem with it.. because if use jump DUNDI LOOKUP must be the last think to look right?! |
| 18:44.40 | structz | or else everything else after dundi include won't execute |
| 18:45.32 | [TK]D-Fender | structz: You clearly don't DO a DUNDi include at that level. CATCH-ALL EXTENSION first. |
| 18:46.34 | structz | kidna confused |
| 18:46.39 | structz | let me learn about the cacth-all |
| 18:47.16 | [TK]D-Fender | structz: Sounds like you don't comprehend the dialplan yet. |
| 18:47.17 | [TK]D-Fender | ~book |
| 18:47.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:47.20 | [TK]D-Fender | ^^^^^^^^ |
| 18:48.31 | *** part/#asterisk gloin (me@unaffiliated/gloin) |
| 18:49.03 | structz | i hv :D |
| 18:49.35 | [TK]D-Fender | structz: Keep reading. If this is even a question for you then its a bad sign... |
| 18:50.52 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 18:50.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:51.20 | structz | [TK]D-Fender, still new at it!! |
| 18:54.34 | *** join/#asterisk Adylas (~jfbouchar@accedian2-136-114.cust.b2b2c.ca) |
| 18:56.30 | *** join/#asterisk jasonwert (~jasonwert@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net) |
| 18:56.54 | Adylas | Hello ! I just did an upgrade from zaptel to dahdi. The dahdi test seems to work fine, I did restart my asterisk. I still get "Unable to open pseudo device". In /dev/dahdi I got : channel ctl pseudo timer transcode ... I am making an error ? Any config to do ? |
| 19:01.22 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 19:01.35 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
| 19:01.55 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 19:01.59 | nix8n82 | Can anyone recommend a simple dialer for asterisk? |
| 19:02.08 | [TK]D-Fender | nix8n82: Your PHONE |
| 19:02.24 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:02.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:02.38 | nix8n82 | funny..I was thinking something along the lines that would maintain 100 connections |
| 19:02.40 | [TK]D-Fender | Adylas: initiale DAHDI FIRST. Also DAHDI must have been compiled and installed BEFORE * for it to function |
| 19:03.04 | [TK]D-Fender | nix8n82: What is a "dialer" conencted to on the other end for these 100 "connections"? |
| 19:03.16 | nix8n82 | sip provider |
| 19:03.52 | [TK]D-Fender | nix8n82: there are 2 ends of a call. Please start over and be more thorought about what you're looking to do. |
| 19:04.21 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-sysluaitfbgmqrkd) |
| 19:05.16 | Adylas | [TK]D-Fender, initiale ? |
| 19:05.32 | [TK]D-Fender | initialize <- |
| 19:05.36 | [TK]D-Fender | dahdi_cfg -vvvv |
| 19:05.38 | [TK]D-Fender | ^^^66 |
| 19:05.42 | nix8n82 | I'm looking a dialing mechanism to read a database of numbers and present the other end with an ivr menu |
| 19:05.46 | [TK]D-Fender | Adylas: THEN start * and see if it's OK |
| 19:06.06 | [TK]D-Fender | nix8n82: this is complete 3rd-party scripting then. |
| 19:06.22 | Adylas | [TK]D-Fender, Yeah, I did and it does generate a quite emtpy system.conf. Also I only need this for the meetup conf. |
| 19:06.40 | nix8n82 | I know it's like vicidial but I was hoping for something less database dependant |
| 19:07.11 | Adylas | [TK]D-Fender, Also, I got some bad things when I do dahdi show status, there is nothing. |
| 19:07.18 | structz | [TK]D-Fender, when you have some time and patience i'll explain my dialplan for you, and you can help me out kinda drowned here |
| 19:07.25 | structz | tks for the attention for now |
| 19:07.45 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 19:08.00 | *** join/#asterisk clintc (~clintc@n128-227-87-199.xlate.ufl.edu) |
| 19:08.04 | [TK]D-Fender | Adylas: the command I gave you does not generate ANY conf |
| 19:08.15 | [TK]D-Fender | Adylas: If your confs are screwed up, go fix them |
| 19:08.32 | Adylas | [TK]D-Fender, Nope, but as I said I only use it for the dummy for meetup |
| 19:08.39 | [TK]D-Fender | nix8n82: Feel free to write on up yourself. |
| 19:08.57 | [TK]D-Fender | Adylas: Yes. I got that. Now fix your configs |
| 19:09.12 | [TK]D-Fender | Adylas: And confirm that DAHDI config exits without error |
| 19:10.16 | Adylas | [TK]D-Fender, It does exit without any error. I just write not much (Some zone = US) |
| 19:11.09 | [TK]D-Fender | Adylas: PASTEBIN is your friend... |
| 19:11.26 | [TK]D-Fender | Adylas: Also confirm that you have completely recompiled * from scratch AFTER having installed DAHDI |
| 19:12.32 | nix8n82 | [TK]D-Fender, I am trying to but it's failing miserably at this point |
| 19:12.53 | nix8n82 | most of it in php |
| 19:12.53 | Adylas | [TK]D-Fender, k, thanks for your help. So, there is no glue between asterisk and dahdi ? (LIke a config that tell what to use) |
| 19:16.05 | [TK]D-Fender | Adylas: Did you initialize dahdi_dummy as well? |
| 19:16.17 | *** join/#asterisk imox1234 (~imox1234@p4FC5C45A.dip0.t-ipconnect.de) |
| 19:16.33 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-wtnsskxvosucblsk) |
| 19:17.14 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 19:17.42 | imox1234 | hello i try to use asterisk realtime, it seems that all its ok but i dont see any peer in my asterisk can somebody help me ? |
| 19:18.25 | [TK]D-Fender | imox1234: Don't se "where"? |
| 19:18.28 | [TK]D-Fender | see |
| 19:18.40 | imox1234 | on the asterisk CLI |
| 19:18.52 | imox1234 | sip show peers |
| 19:19.04 | [TK]D-Fender | imox1234: That won't show them. |
| 19:19.10 | [TK]D-Fender | imox1234: thats jsut how realtime works |
| 19:19.31 | Adylas | [TK]D-Fender, Humm, I dont think. I did some dahdi_cfg and dahdi_genconf but thats it |
| 19:19.35 | imox1234 | hmm i think it works :D |
| 19:19.52 | imox1234 | when i start i get this == Binding sippeers to mysql/general/sip_buddies |
| 19:19.58 | [TK]D-Fender | imox1234: keep "thinking". Let see how far you get. |
| 19:20.06 | imox1234 | and when i unload and load the module i dont get any errors |
| 19:20.49 | imox1234 | where can i see if it work ? |
| 19:21.10 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:21.20 | Gugge | imox1234: make some calls |
| 19:21.41 | imox1234 | ;) |
| 19:22.14 | imox1234 | but i must see the peers. without peers i cant call ;) |
| 19:22.28 | Adylas | [TK]D-Fender, I dont find much thing about initialize dahdi_dummy .. ;-( |
| 19:22.38 | [TK]D-Fender | imox1234: I don't see you showing us a FAILURE. |
| 19:22.43 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 19:22.49 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:22.49 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:22.56 | imox1234 | i know |
| 19:22.58 | [TK]D-Fender | Adylas: modprobe dahdi_dummy |
| 19:23.00 | [TK]D-Fender | Adylas: modprobe dahdi |
| 19:23.01 | imox1234 | but i dont know how ? |
| 19:23.08 | [TK]D-Fender | Adylas: dahdi_cfg -vvvv |
| 19:23.09 | imox1234 | because of that i ask here ;) |
| 19:23.11 | [TK]D-Fender | Adylas: Start * |
| 19:23.31 | [TK]D-Fender | [15:23]<imox1234>but i dont know how ? <- Ok, You're not seeming too bright here. |
| 19:23.39 | [TK]D-Fender | imox1234: AShow us the FAILURE <- |
| 19:23.54 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 19:23.57 | Micc | alright I got oslec built, just waiting now for calls to end so I can restart asterisk. |
| 19:23.57 | QubeZ | hello all |
| 19:24.17 | QubeZ | when you have same extensions in sip.conf and iax.conf, which takes precedence? |
| 19:24.48 | Adylas | [TK]D-Fender, This is the module I got .... http://pastebin.com/BCknhKjA |
| 19:24.56 | QubeZ | also when I add the extensions into sip.conf, and do a sip reload... it gives me inheritance error stating the category cannot be found and to make sure sip.conf exists even if its empty. It's not empty and it does exist. |
| 19:25.11 | Adylas | [TK]D-Fender, When I modprobe dahdi_dummy, it seems to work but I dont see it in lsmod |
| 19:25.39 | [TK]D-Fender | Adylas: Do you have a card? |
| 19:25.51 | imox1234 | [TK]D-Fender: why you don't say me how i can check if it work |
| 19:26.08 | [TK]D-Fender | QubeZ: Extensions aren't in sip.conf. They are in EXTENSIONS.CONF |
| 19:26.24 | [TK]D-Fender | imox1234: How to check if it works? PLACE A CALL. |
| 19:26.33 | Adylas | [TK]D-Fender, Nope, its only for the dummy. for Meetup. |
| 19:26.46 | [TK]D-Fender | QubeZ: There is no such thing as "precedence" |
| 19:27.02 | [TK]D-Fender | Adylas: then initialize it like I showed you. Restart * and TRY IT |
| 19:27.15 | QubeZ | [TK]D-Fender: i also added them into extensions.conf. I have exten => 12313,hint,IAX2/12313 and exten => 12313,hint,SIP/12313 |
| 19:27.30 | imox1234 | member:identifier:[tk]d-fender: you want kidding me ? how should i make a call without peer and without anything ? |
| 19:28.11 | [TK]D-Fender | [15:27]<imox1234>member:identifier:[tk]d-fender: you want kidding me ? how should i make a call without peer and without anything ? <- "sip show peers" won't fucknig LIST THEM. It does not mean it will not WORK. Now get off you ass and place a CALL. |
| 19:28.36 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 19:29.00 | [TK]D-Fender | QubeZ: The 2nd one takes over. taht is a dialplan screwup. Same priority twice. Total no-no |
| 19:29.45 | QubeZ | [TK]D-Fender: so I added the extensions into sip.conf but when I do a sip reload is when I get that error about inheritance |
| 19:30.06 | [TK]D-Fender | QubeZ: Show us |
| 19:30.10 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 19:30.44 | *** join/#asterisk Chris-NB (~chris@home.fuerstaller.com) |
| 19:30.45 | QubeZ | [TK]D-Fender: going to paste to pastebin shortly |
| 19:30.54 | Micc | TKD-Fender, do you get paid to help people with asterisk? |
| 19:31.19 | [TK]D-Fender | Micc: for consulting yes, and the occasional donation from here... |
| 19:31.26 | imox1234 | [TK]D-Fender: ok thanks. it is possible to show the peers on the cli ? |
| 19:31.26 | [TK]D-Fender | Micc: But not as a job or anything |
| 19:31.30 | [TK]D-Fender | (just in general) |
| 19:31.36 | jdoe | no I think it's clear he just derives joy from it. Can't you tell from his glee? ;) |
| 19:31.46 | [TK]D-Fender | imox1234: gor ead up on the realtime chapters in THE BOOK. |
| 19:31.48 | [TK]D-Fender | ~book |
| 19:31.48 | 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 |
| 19:31.49 | t_dot_zilla | Asterisk doesn't seem to provide an option to wait for more keys after something matches in features.conf |
| 19:32.10 | [TK]D-Fender | jdoe: My kind of happyness KILLS |
| 19:32.19 | t_dot_zilla | we have attended xfre set to # |
| 19:32.27 | t_dot_zilla | and blind xfer set to ## |
| 19:32.29 | QubeZ | [TK]D-Fender: http://pastebin.com/zErNi6K8 |
| 19:32.52 | t_dot_zilla | if you hit ##, it does an attended xfer looking for extension beginning wtih # |
| 19:33.01 | *** join/#asterisk gavimobile (~gavriel@77.124.156.153) |
| 19:33.02 | QubeZ | [TK]D-Fender: and /includes/ca-wc01/sip.incl does exist |
| 19:33.15 | [TK]D-Fender | quaBad pastebin I have no proof of what is reading that, what's IN there, and whats in the file calling it. |
| 19:33.21 | [TK]D-Fender | QubeZ: ^^^ |
| 19:33.37 | QubeZ | [TK]D-Fender: one moment |
| 19:34.10 | gavimobile | [TK]D-Fender: could I just attach a router to my spa2102 and than connect several ip phones to the router and configure each phone as a separate extention? |
| 19:34.43 | Adylas | [TK]D-Fender, It seems that dahdi_dummy never load. It wont stick in. |
| 19:36.07 | Adylas | [TK]D-Fender, I got some more strange things in dmesg : http://pastebin.com/ufuZDUpT |
| 19:37.48 | [TK]D-Fender | yhave you restarted since install DAHDI? |
| 19:37.57 | [TK]D-Fender | Adylas: ^^ |
| 19:38.00 | *** join/#asterisk d4rkstar (~bruno@93-41-29-219.ip79.fastwebnet.it) |
| 19:38.29 | Adylas | [TK]D-Fender, Yup, I did 1- Reboot the machine 2- Restart asterisk |
| 19:38.56 | Adylas | [TK]D-Fender, Anyway there is no way the module can play nice with zaptel. you need to reboot |
| 19:44.06 | tzafrir | Adylas, what do you mean? |
| 19:44.33 | tzafrir | As of dahdi-linux 2.3.0 dahdi_dummy is gone (moved into the main dahdi module) |
| 19:44.47 | *** join/#asterisk n3hxs (~HAMming@phx-69-171-161-170.evdo.leapwireless.net) |
| 19:45.16 | Adylas | tzafrir, Ha okay. Sitll I got "No Zaptel interface found" after all reset and the config commands. |
| 19:45.32 | [TK]D-Fender | Adylas: PASTEBIN <- |
| 19:45.35 | t_dot_zilla | how long does asterisk wait before executing a feature in features.conf ? |
| 19:46.17 | t_dot_zilla | let's say attended transfer is set to '#', how long does it wait do to an attended transfer? |
| 19:46.29 | Qwell | t_dot_zilla: whatever the timeout is set to |
| 19:46.43 | t_dot_zilla | where would i find that? |
| 19:46.49 | Qwell | features.conf.. |
| 19:47.09 | drmessano | Where do I find features.conf? |
| 19:47.21 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 19:47.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:47.39 | [TK]D-Fender | drmessano: Somewhere around Uranus :p |
| 19:48.04 | jdoe | hrm. |
| 19:48.27 | Trixboxer | Hi, Im compiling asterisk 1.4.33 but res_odbc is disabled in "make menuselect". Is there any way I can enable it ? |
| 19:48.39 | drmessano | E: Couldn't find package uranus |
| 19:48.56 | drmessano | Trixboxer: Install the needed dependencies (bottom) |
| 19:48.59 | t_dot_zilla | what if timeout is not set in features.conf? what is the default timeout? |
| 19:49.08 | Trixboxer | they are installed |
| 19:49.10 | wcselby | Trixboxer - have you installed the odbc requirements? |
| 19:49.13 | [TK]D-Fender | Trixboxer: Install the pre-requisites |
| 19:49.28 | wcselby | rerun ./configure |
| 19:49.36 | jdoe | I'm trying to do least cost routing with a provider that does auth based on source ip/port... two trunks to them means two ips/ports or a second port on the same ip I guess... I can't bind or use multiple sources from asterisk can I? Is there a less hackish way to do this than running a second asterisk instance? |
| 19:51.23 | Trixboxer | unixODBC unixODBC-devel libtool-ltdl everything is installed |
| 19:51.33 | *** join/#asterisk Krolik13 (~Krolik13@5ac3136f.bb.sky.com) |
| 19:51.35 | Trixboxer | but it still says "Depends on: unixodbc(E), ltdl(E)" |
| 19:51.37 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:51.43 | Trixboxer | done a ./configure |
| 19:51.55 | Krolik13 | hi. how can i match the recorder calls via Monitor app, with the CDR records from the database? |
| 19:52.05 | wcselby | libtool-ltdl-devel? |
| 19:52.12 | Trixboxer | hmm |
| 19:52.12 | [TK]D-Fender | Krolik13: What database? |
| 19:52.16 | t_dot_zilla | Qwell: are you refering to transferdigittimeout in features.conf ? |
| 19:52.37 | Krolik13 | [TK]D-Fender> mysql, database that is used by asterisk to insert the CDRs |
| 19:53.02 | [TK]D-Fender | Krolik13: Well go look at the names you give your recordings with Monitor. |
| 19:53.51 | Krolik13 | [TK]D-Fender> the names are related to the time.. There is any distinct parameter that i can use in the recorded filename and the CDR records. Just to be able to match the right call details with the recorded files |
| 19:54.00 | wcselby | Krolik13 - if you add the ${UNIQUEID} variable to the monitor'ed filenames, you can match on that field in the CDR database. You might need "logunique=yes" in your cdr.conf file |
| 19:54.28 | Trixboxer | wcselby, Thanks that helped .. now doing install ... |
| 19:54.30 | Krolik13 | wcselby> good point. let me try :) thanks |
| 19:54.40 | Krolik13 | [TK]D-Fender, thanks |
| 19:54.46 | wcselby | Trixboxer - np |
| 19:54.49 | wcselby | Krolik13 - np |
| 19:54.54 | wcselby | wcselby - np |
| 19:54.59 | wcselby | wait....that's a problem |
| 19:54.59 | Trixboxer | :) |
| 19:55.04 | wcselby | :) |
| 19:55.43 | Krolik13 | wcselby, sorry for such stupid questin, but how can i see the all variables that are set in the env during the call? (Like running "SET" command in windows, to see all variables and values) |
| 19:56.07 | wcselby | core show application DumpChan |
| 19:56.49 | Krolik13 | wcselby> thanks again |
| 19:56.51 | wcselby | Krolik13 - sorry, DumpChan(). Do the above command from the asterisk cli to read about it. |
| 19:57.05 | Krolik13 | i understand your mean :) |
| 19:57.15 | Krolik13 | i understand your thought :) |
| 19:57.18 | Krolik13 | sorry, my english |
| 20:02.36 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 20:03.56 | Deeewayne | cd |
| 20:04.19 | drmessano | ll |
| 20:08.31 | *** join/#asterisk imox1234 (~imox1234@p4FC5C45A.dip0.t-ipconnect.de) |
| 20:14.49 | imox1234 | hm realtime extensions dont work |
| 20:14.52 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 20:15.10 | imox1234 | i can register with a peer but i cant call |
| 20:15.37 | [sr] | u mean, "my realtime doesnt work" |
| 20:15.38 | [sr] | :p |
| 20:16.09 | imox1234 | yes yes ;( |
| 20:16.13 | [TK]D-Fender | imox1234: Where is the call? |
| 20:16.26 | imox1234 | how where ? |
| 20:17.03 | [TK]D-Fender | imox1234: Where is the call? |
| 20:17.03 | imox1234 | i make in the extension_table a exten _. s Wait 100 for test |
| 20:18.54 | imox1234 | when I write it to my extensions.conf thats works |
| 20:19.04 | imox1234 | but then i write it to my database dont work ;( |
| 20:19.20 | [TK]D-Fender | imox1234: Where is the call? <----------- |
| 20:19.47 | wcselby | imox1234 - show us a failed call attempt from the CLI |
| 20:19.55 | gr0mit | hey guys - can anyone recommend a provider who can provide a 212 area DID please? |
| 20:20.15 | Krolik13 | what is the best (more functionalities like statistics, filters, details) open source web-app for processing and displaying content of CDRs? (not interested in billing) |
| 20:20.18 | imox1234 | [Jun 22 00:20:14] NOTICE[1774]: chan_sip.c:20059 handle_request_invite: Call from '111' to extension '222' rejected because extension not found. |
| 20:20.34 | Krolik13 | which one do you recommend |
| 20:20.42 | wcselby | imox1234 - the entire call, from start to finish, pastebin it |
| 20:20.44 | wcselby | ~pb |
| 20:20.45 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:21.03 | imox1234 | NO MORE its all |
| 20:21.08 | imox1234 | i use pastebin |
| 20:21.14 | imox1234 | but not by one line |
| 20:22.54 | [TK]D-Fender | imox1234: Show us your extensions.conf and a failed call where you actually LOOKED at the SIP DEBUG to see what context it's [pointing to... and then also decide to show us your damn DB entries |
| 20:23.15 | wcselby | thanks [TK]D-Fender |
| 20:23.27 | wcselby | was having the damndest time typing that out |
| 20:24.06 | Krolik13 | sorry, for disturbing... any opinions to my question? |
| 20:24.21 | [TK]D-Fender | Krolik13: Areski |
| 20:25.11 | Krolik13 | [TK]D-Fender> thank you! |
| 20:25.44 | imox1234 | http://pastebin.com/gmCNNSRb |
| 20:26.01 | Krolik13 | guys, you are doing a great job here!!! |
| 20:26.54 | [TK]D-Fender | imox1234: Where is your EXTENSIONS.CONF like I asked for? |
| 20:27.02 | *** join/#asterisk Tim_Toady (~moi@77.49.107.115.dsl.dyn.forthnet.gr) |
| 20:27.25 | imox1234 | in the extensions.conf is nothing. i have the table ? |
| 20:27.29 | imox1234 | or its wrong ? |
| 20:28.18 | [TK]D-Fender | imox1234: WRONG. |
| 20:28.31 | imox1234 | hmm ok |
| 20:28.33 | wcselby | you've at least got to have a switch statement in there, right? |
| 20:28.38 | [TK]D-Fender | Yes |
| 20:29.21 | [TK]D-Fender | Nobody ever reads the fucking instructions. |
| 20:29.25 | [TK]D-Fender | imox1234: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 20:29.30 | [TK]D-Fender | imox1234: SWITCH <---------------------- |
| 20:33.32 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:35.05 | [TK]D-Fender | checkout time, back later. |
| 20:35.58 | imox1234 | ok sorry, now it works :D |
| 20:36.17 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:36.52 | imox1234 | it is right that i need for any context the mysql table a context with a switch in the extensions.conf. or can i set only one switch for all context ? |
| 20:38.26 | *** join/#asterisk MohsenSaeedi (~mohsen@unaffiliated/mohsensaeedi) |
| 20:41.20 | imox1234 | ?? |
| 20:41.57 | Gugge | imox1234: one pr context |
| 20:42.11 | imox1234 | its dont possible ? |
| 20:42.50 | Gugge | try again in english. |
| 20:43.38 | Gugge | you need one pr context as far as i can tell from the docs |
| 20:43.39 | imox1234 | it is not possible ? :D |
| 20:43.49 | imox1234 | hmm |
| 20:43.52 | imox1234 | its stupid |
| 20:44.00 | Gugge | yes, one pr context is perfectly possible |
| 20:44.30 | imox1234 | but i want ONLY one switch for all |
| 20:45.07 | imox1234 | now i have to write every time that i create a new context a switch in the extensions.conf |
| 20:45.07 | Gugge | and i want to win the lottery |
| 20:45.41 | MohsenSaeedi | hello all, i have bad problem with elastix 1.6 and asterisk 1.4 . i configured it with openvox A400 . it works without any error . but after 1 week it stopped to work with trunk zap/g0 . 9 defined for outside call and i define one IVR. both of them didn't work now!!! |
| 20:46.22 | Gugge | MohsenSaeedi: what error do you see in the CLI? |
| 20:46.52 | Gugge | imox1234: im sure a patch is welcome if you want to change how it works :) |
| 20:47.45 | oc80z | someoen said dont use PCI FXS |
| 20:48.08 | MohsenSaeedi | Gugge: for incoming call or outgoing call? for incoming nothing! i think openvox couldn't detect ring from pstn! and for outside call i should trace it again. |
| 20:48.34 | imox1234 | Gugge: ;) |
| 20:52.34 | [sr] | oc80z: why not? |
| 20:53.38 | [sr] | example, if i need a FXS card, why shouldn't i not use it? |
| 20:54.05 | Gugge | why you shouldn't not use it? |
| 20:54.09 | WIMPy | [sr]: BTW: How's your fax doing? |
| 20:55.58 | [sr] | Gugge: 'cause i don't need it yet :P |
| 20:56.20 | Gugge | thats why you shouldn't use it |
| 20:56.25 | Gugge | or why you should not use it |
| 20:56.32 | Gugge | not why you shouldn't not use it |
| 20:56.35 | Gugge | :P |
| 20:57.00 | [sr] | WIMPy: hi there, didn't conf it yet...1st i'm puting my isdn stuff stable |
| 20:57.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:57.44 | [sr] | Gugge: for example, as asterisk instalation, with as 8port FXS card, it's cheaper to not change the existing phones, and it's gonna be cheaper |
| 20:58.10 | [sr] | depends on the cases |
| 20:59.32 | [TK]D-Fender | PCI FXS = ASS |
| 20:59.54 | t_dot_zilla | has anybody gotten attended transfer (set to #) and blind transfer (set to ##) to work? could someone else try it? i can't figure out why it won't work, i've set transferdigittimeout and featuredigittimeout |
| 21:02.21 | [TK]D-Fender | t_dot_zilla: Show us the complete failed call |
| 21:08.06 | t_dot_zilla | [TK]D-Fender: http://pastebin.com/RdNtUErx |
| 21:09.09 | [TK]D-Fender | t_dot_zilla: Nowhere do I see that feature enabled via DYNAMIC_FEATURES |
| 21:10.47 | t_dot_zilla | here is my features.conf http://pastebin.com/VEVNLYt9 |
| 21:12.09 | [TK]D-Fender | t_dot_zilla: No. the CHANNEL VARIABLE that you need to set to specify WHICH features are permitted <- Go read the docs on this again |
| 21:16.08 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
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| 21:31.14 | MohsenSaeedi | Gugge: that is my log : http://pastebin.org/349432 |
| 21:31.37 | *** join/#asterisk Holos (~cosmond@static-pppoe-209-91-139-211.vianet.ca) |
| 21:32.02 | Holos | Anyone have a suggestion for an FXO ATA that is standalone and can do ring-detection ie.. don't answer short-short rings? |
| 21:34.14 | Holos | Or even a good quality FXO adapter that's dependable? |
| 21:35.07 | WIMPy | POTS should really have been left in the past century for good, that's where it belongs. |
| 21:36.06 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 21:36.17 | Holos | WIMPy: yah.. except it's pretty dependable compared to the Internet for low-budget connections |
| 21:37.38 | WIMPy | There has been a successor since the 80s. |
| 21:38.34 | [TK]D-Fender | Holos: SPA-3102 isn't too bad |
| 21:38.42 | [TK]D-Fender | Holos: If you're looking for jsut 1 port. |
| 21:39.56 | Holos | [TK]D-Fender: thanks.. I think I can do just 1 port, wanted to get someone who'd used them as some of the FXO's can be pretty crappy at times. |
| 21:43.04 | picard1421 | hey.. does asterisk have any compatability with the USSD protocal? |
| 21:44.00 | WIMPy | picard1421: If you're talking about service codes: You can make it do so. |
| 21:44.01 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 21:45.10 | picard1421 | like sending data through USSD |
| 21:45.20 | [TK]D-Fender | picard1421: No. |
| 21:45.22 | picard1421 | and recieving USSD data |
| 21:45.29 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 21:45.52 | picard1421 | ohh you cant? is there any type of hardware you need to buy or this protocol cannot be used with asterisk period |
| 21:46.24 | WIMPy | I don't think there is any possibility. |
| 21:52.36 | [TK]D-Fender | picard1421: * does not speak "data" or "sms". |
| 21:52.53 | picard1421 | gotchya ok well thanks .. |
| 21:53.11 | MohsenSaeedi | hello all, i have bad problem with elastix 1.6 and asterisk 1.4 . i configured it with openvox A400 . it works without any error . but after 1 week it stopped to work with trunk zap/g0 . 9 defined for outside call and i define one IVR. both of them didn't work now!!! |
| 21:53.33 | [TK]D-Fender | MohsenSaeedi: .... |
| 21:53.35 | [TK]D-Fender | ~freepbx |
| 21:53.36 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 21:53.37 | [TK]D-Fender | ^^^^^^^ |
| 21:54.27 | MohsenSaeedi | [TK]D-Fender: my question are not related too freepbx |
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| 21:56.03 | [TK]D-Fender | MohsenSaeedi: You've shown nothing to the contrary |
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| 21:56.20 | Micc | I installed oslec for dahdi like it said on the web page, but adding oslec to system.conf still give me an error. |
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| 21:59.02 | Micc | They don't ever say anything about editing /etc/dahdi/system.conf so maybe its always on? |
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| 22:01.56 | [TK]D-Fender | Micc: that is where you specify that your channels use it |
| 22:03.01 | Micc | Thats what I thought, but I still get this when running dahdi_cfg, Setting echocan for channel 1 to oslec DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) |
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| 22:05.41 | [TK]D-Fender | Micc: And it works immediately if you switch back to MG2? |
| 22:05.53 | Micc | yes. |
| 22:05.55 | Micc | or kb1 |
| 22:06.24 | [TK]D-Fender | Micc: Ok, I can't help with the fine points on this one. Tighten up your Google-ing and You should probably find your answer |
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| 22:15.50 | [sr] | going to sleep |
| 22:15.52 | [sr] | see ya all |
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| 22:19.25 | [sr] | oh |
| 22:19.31 | [sr] | no one whises me night night |
| 22:19.33 | [sr] | :p |
| 22:19.53 | [sr] | wishes |
| 22:20.31 | Chainsaw | [sr]: [TK]D-Fender will tuck you in momentarily. |
| 22:20.54 | WIMPy | Already asleep :-) |
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| 22:22.08 | [sr] | hehe |
| 22:22.21 | [sr] | was just trying to put some fun here |
| 22:22.30 | [sr] | ok now im really going |
| 22:22.31 | [sr] | see ya |
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| 22:39.43 | markfeatherston_ | Using the manager interface, how would I request the active channels from an extension? |
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| 23:17.38 | d4rkstar | hi all |
| 23:18.11 | d4rkstar | someone can tell me if t.38 pass-thru on asterisk 1.6.2 is broken? |
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| 23:30.15 | [TK]D-Fender | d4rkstar: Why would it be? Don't asking leading question like that. |
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| 23:30.58 | Krolik13 | [TK]D-Fender> my Monitor app is recoding mp3 but it doesn't have any voice, but the length is right. What it could be? |
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| 23:31.25 | d4rkstar | [TK]D-Fender: i'm reading better my sip.conf example file and it seems that t38 configuration is a bit changed |
| 23:31.34 | d4rkstar | i didn't see that |
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| 23:32.07 | d4rkstar | i'm having troubles with t.38 passthru because of that T38FaxMaxDatagram |
| 23:32.30 | d4rkstar | it seems that now that value can be set from sip.conf t38pt_udptl setting |
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| 23:32.40 | d4rkstar | and no more from udptl.conf |
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| 23:35.15 | d4rkstar | after that it seems to work fine :) |
| 23:35.27 | d4rkstar | sorry for that leading question |
| 23:36.01 | [TK]D-Fender | Krolik13: 1st, Monitor() can't record in MP3, and it has no audio when you try playing it back through ASTERISK? |
| 23:36.31 | Krolik13 | i'm trying to play it locally on my PC |
| 23:36.53 | Krolik13 | an it has no voice... |
| 23:37.01 | Krolik13 | but the lenght of the call is right |
| 23:37.22 | [TK]D-Fender | Krolik13: Perhaps your PC player has issues <- |
| 23:37.37 | [TK]D-Fender | Krolik13: so remove that as a factor and play it back with ASTERISK |
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