| 00:00.52 | [TK]D-Fender | NO, it isn't |
| 00:00.52 | mattwj2002 | I am not connecting to asterisk |
| 00:00.52 | [TK]D-Fender | .... |
| 00:01.39 | mattwj2002 | I want to get it to connect to a DID directly |
| 00:03.50 | [TK]D-Fender | A DID is a *PHONE NUMBER*. there is no such thing as "connect directly". |
| 00:04.01 | [TK]D-Fender | That term is pure rubbish |
| 00:04.18 | mattwj2002 | I meant without going through asterisk |
| 00:04.27 | mattwj2002 | of my own |
| 00:04.37 | [TK]D-Fender | Also vague |
| 00:04.42 | [TK]D-Fender | SIP is SIP |
| 00:04.55 | mattwj2002 | are you going to help are be a pain in the ass? |
| 00:04.56 | [TK]D-Fender | that CALLER can 404, the called NUMBER can 404. |
| 00:05.16 | mattwj2002 | *or |
| 00:05.26 | mattwj2002 | screw it |
| 00:05.28 | [TK]D-Fender | And YOU aren't giving a useful difinitive answer about WHAT is failing <- |
| 00:05.38 | mattwj2002 | screw it |
| 00:05.43 | Kevin` | mattwj2002: what will be called here? your provider will connect to your phone directly? does your phone register with the provider? |
| 00:05.44 | mattwj2002 | I am out of here |
| 00:05.48 | [TK]D-Fender | 404 can be a DIFFERENT kind of response for MULTIPLE things |
| 00:06.00 | florz | *lol* |
| 00:06.17 | [TK]D-Fender | mattwj2002: Your info is incomplete and you're asking about help with things that are outside of * |
| 00:06.28 | mattwj2002 | I understand |
| 00:06.30 | mattwj2002 | sorry |
| 00:06.41 | FutureWeb | Hey everyone, My ISP modem uses VoIP, and I need to figure out what settings (SIP/etc) its using :P anyone know a method how ? I can provide Modem info etc etc |
| 00:06.53 | mattwj2002 | freenode needs a general voip channel |
| 00:07.14 | [TK]D-Fender | mattwj2002: Your problems is specifics. Or more directly your lack of having them. |
| 00:07.16 | Kevin` | if you asked your question better you'd probably get an answer here |
| 00:07.22 | mattwj2002 | [TK]D-Fender I am sorry I got mad at you |
| 00:07.29 | [TK]D-Fender | mattwj2002: What is that 404 as response TO? |
| 00:07.41 | [TK]D-Fender | mattwj2002: Can you even tell? |
| 00:07.48 | [TK]D-Fender | mattwj2002: otherwise asking anything mroe is pointless |
| 00:08.01 | mattwj2002 | I would assume it can't connect to the address |
| 00:08.05 | mattwj2002 | but I can ping it |
| 00:08.13 | [TK]D-Fender | mattwj2002: "Assume" doesn't help |
| 00:08.16 | mattwj2002 | from my pc anyways |
| 00:08.27 | [TK]D-Fender | mattwj2002: Yes it "connects. and is being REFUSED |
| 00:08.37 | [TK]D-Fender | mattwj2002: For one of the 2 things I already told you |
| 00:08.37 | FutureWeb | so anyone has an Idea about my quesion ? please :) |
| 00:08.44 | [TK]D-Fender | mattwj2002: You don't seem to be paying attention |
| 00:09.00 | [TK]D-Fender | mattwj2002: They are telling you to GTFO <- And I told you the 1 REASONS why that might be. |
| 00:09.23 | [TK]D-Fender | FutureWeb: ISP's tend to lock you out. You are likely in a dead-end with this |
| 00:09.35 | [TK]D-Fender | 2* |
| 00:09.35 | mattwj2002 | oh |
| 00:09.49 | [TK]D-Fender | mattwj2002: 404 is an ANSWER |
| 00:09.50 | mattwj2002 | interest |
| 00:09.56 | mattwj2002 | *interesting |
| 00:10.19 | [TK]D-Fender | [20:07]<[TK]D-Fender>mattwj2002: What is that 404 as response TO? <----RESPONSE |
| 00:10.38 | FutureWeb | no idea :/ the modem has VoIP in it, and Im sure it authenicates using SIP (they said so lol), so if I got get the settings (Address and Secret) I could bypass the modem and make asterisk use the DID :) |
| 00:11.02 | [TK]D-Fender | FutureWeb: Maybe. of course there are MULTIPLE ways they can lock you out. |
| 00:11.09 | [TK]D-Fender | FutureWeb: Could eb a whole lot more than that |
| 00:11.20 | Kevin` | FutureWeb: have you tried asking them? |
| 00:11.24 | mattwj2002 | so [TK]D-Fender question |
| 00:11.37 | mattwj2002 | is it possible too many failed attempts? |
| 00:11.49 | FutureWeb | they dont know what VoIP etc is... only 1 tech knew :/ and he seems to h ave vanished away now :| |
| 00:11.58 | FutureWeb | that one tech told me they use SIP |
| 00:11.59 | FutureWeb | ;D |
| 00:13.35 | [TK]D-Fender | mattwj2002: NO |
| 00:20.35 | *** part/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 00:21.23 | [TK]D-Fender | douchebaf |
| 00:21.29 | [TK]D-Fender | g even :0 |
| 00:32.43 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 00:34.16 | *** join/#asterisk ivanvujisic (5b9450be@gateway/web/freenode/ip.91.148.80.190) |
| 00:35.00 | *** join/#asterisk misc-- (~misc@202-154-80-42.people.net.au) |
| 00:35.56 | ivanvujisic | I'm trying to move dialplan to agi shell script. echo "EXEC GOTO $1|4" works, but I can not find related agi syntax for GOTOIF |
| 00:37.05 | ivanvujisic | in dialplan I put AGI(/path/to/script|${EXTEN}) |
| 00:37.33 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 00:38.04 | [TK]D-Fender | iviGOTOIF has parms jsut like every other app |
| 00:38.30 | [TK]D-Fender | ivanvujisic: Go rad them. EXEC can call it |
| 00:39.00 | [TK]D-Fender | ivanvujisic: although normally you would EXIT your AGI at a different context/exten/prio instead |
| 00:40.19 | *** join/#asterisk ldiamond (60167564@gateway/web/freenode/ip.96.22.117.100) |
| 00:40.19 | ivanvujisic | so how can we translate this from dialplan GotoIf($["${CALLERID(num):1}" = "0628030729" ]?CID_OK) to agi |
| 00:40.52 | [TK]D-Fender | ivanvujisic: First why are you even THINKING about jumping out of your AGI? Second you wouldn't need to use a dialplan app to do it <- |
| 00:42.06 | ivanvujisic | actually I'm trying to move part of dialplan to shell script to hide it (encrypt it by shc) |
| 00:42.15 | ldiamond | Does Asterisk support receiving text messages through SIP? |
| 00:42.47 | Alton35 | you mean sms? |
| 00:43.37 | ivanvujisic | maybe echo "EXEC GOTOIF $["${CALLERID(num):1}" = "0628030729" | CID_OK" is right agi syntax? |
| 00:44.00 | ldiamond | Well, in the SIP protocol it is not called SMS. That would be up to the provider to support SMS right? |
| 00:44.46 | *** join/#asterisk xheliox (~jeff@i216-58-41-253.cybersurf.com) |
| 00:46.02 | ivanvujisic | [TK]D-Fender: I want to jump somewhere else from agi script |
| 00:46.33 | florz | *lol2* |
| 00:46.40 | florz | _encrypted dialplan_? |
| 00:46.49 | ivanvujisic | why not?! |
| 00:46.56 | Kevin` | why so? |
| 00:46.58 | ivanvujisic | do you know what is shc? |
| 00:47.14 | florz | must be something idiotic, if you ask me |
| 00:47.16 | ivanvujisic | to protect my knowledge |
| 00:47.43 | florz | a helmet protects your knowledge |
| 00:47.47 | Kevin` | i'm all for job security, but I would try to find someone else if I saw my employee doing that |
| 00:48.03 | florz | not some idiotic code obfuscation device |
| 00:48.53 | ivanvujisic | did you tried shc to encrypt shell script? |
| 00:48.55 | florz | (and, damn, don't EVER confuse "encryption" with "obfuscation" - that's really basic knowledge, actually) |
| 00:49.16 | Kevin` | ivanvujisic: why would that matter? |
| 00:49.25 | Kevin` | shc looks like it makes a binary out the shell script |
| 00:49.32 | Kevin` | would have to read the code to see how |
| 00:49.46 | ivanvujisic | I'm not the only owner of root account |
| 00:50.14 | Kevin` | that said, why don't you just write the script in c from the start |
| 00:50.15 | [TK]D-Fender | ivanvujisic: There is another AGI command to set the exit point <- |
| 00:50.21 | [TK]D-Fender | ivanvujisic: Go read your basics again |
| 00:50.37 | florz | whatever your problem is, code obfuscation is not the solution |
| 00:51.02 | florz | and encryption protects secrets, not knowledge |
| 00:51.26 | florz | real encryption that is, the stuff with keys, you know? |
| 00:51.26 | [TK]D-Fender | [20:42]<ldiamond>Does Asterisk support receiving text messages through SIP? <- Asterisk is NOT a messaging platform. If this is what you're looking for, look elsewhere |
| 00:51.59 | ivanvujisic | I use to convert shell scripts to C by shc, did anyone use it? |
| 00:52.19 | Kevin` | ldiamond: asterisk has support for sms |
| 00:52.25 | florz | ivanvujisic: I doubt anyone wants to admit they are an idiot ... |
| 00:52.42 | Kevin` | oh, I get it |
| 00:53.16 | Kevin` | ivanvujisic doesn't want to answer the questions, he wants to talk in pm to someone who already "understands" why you would obfuscate everything randomly |
| 00:54.04 | ldiamond | Kevin`: Thanks. I suppose that in order to receive SMS, it is required that the VOIP provider has the capability to forward it to your Asterisk server, which is apparently not the case with mine. |
| 00:54.11 | ivanvujisic | anyway, I really need agi GOTOIF syntax |
| 00:54.24 | florz | ivanvujisic: NO, YOU DON'T |
| 00:54.28 | pepselap | it's on voip-info |
| 00:54.34 | [TK]D-Fender | ivanvujisic: No, you DON'T |
| 00:54.42 | [TK]D-Fender | ivanvujisic: There is another AGI command to set the exit point <- |
| 00:54.54 | [TK]D-Fender | ivanvujisic: Read the BOOK |
| 00:55.16 | ivanvujisic | thanks, you were most helpfull |
| 00:55.22 | Kevin` | ivanvujisic: exactly what secret are you trying to hide? it's obviously not elite asterisk configuration, but some specific passwords or such? |
| 00:57.00 | ivanvujisic | it's not password, it's my dialplan for listening recorded calls |
| 00:57.10 | florz | gee, the man page of this shc stuff even more-or-less explicitly says that anyone with a remote clue can "decrypt" any scripts "encrypted" with it |
| 00:57.14 | pepselap | exten => username,1,GotoIf($["${CALLERID(num)}" = "+18005551212"]?reject:allow) |
| 00:57.24 | Kevin` | ivanvujisic: and exacrtly what secret does it contain? |
| 00:58.04 | ivanvujisic | t's my dialplan for listening recorded calls |
| 00:58.13 | ivanvujisic | it's my dialplan for listening recorded calls |
| 00:58.31 | Kevin` | and what part of it is secret information? |
| 00:58.34 | Kevin` | "so what?" |
| 00:58.36 | [TK]D-Fender | ivanvujisic: We heard youthe first 3 times |
| 00:58.54 | ivanvujisic | anyway, nobody here don't know how to convert GOTOIF to agi |
| 00:59.06 | Kevin` | they are saying it isn't necessary |
| 00:59.16 | florz | "BUT I REALLY WANT TO BE AN IDIOT!!!1" |
| 00:59.31 | pepselap | ivanvujisic: ... |
| 00:59.50 | [TK]D-Fender | [20:58]<ivanvujisic>anyway, nobody here don't know how to convert GOTOIF to agi <- YOU on the otherhand do NOT appear to be listening |
| 01:00.35 | ivanvujisic | I can tell you same, I just asked if somebody know the syntax |
| 01:00.58 | pepselap | ivanvujisic: < pepselap> it's on voip-info |
| 01:01.22 | pepselap | ivanvujisic: < pepselap> exten => username,1,GotoIf($["${CALLERID(num)}" = "+18005551212"]?reject:allow) |
| 01:01.34 | ChannelZ | see SET EXTENSION and SET PRIORITY et al |
| 01:01.47 | troy42 | ivanvujisic: in agi, you can use whatever language you want, do get variable callerid(num), and check it and act (go to a different extension, etc) |
| 01:02.19 | *** join/#asterisk thissucks (~crazyhick@gw.its-my.net) |
| 01:02.25 | troy42 | also, you'll avoid blindness from having to deal with syntax like that :o |
| 01:02.58 | Kevin` | 'but I want to use sh converted to c that only uses dialplan-like commands' |
| 01:03.04 | ivanvujisic | you dont have to search voip-info, show application GotoIf is ok |
| 01:03.39 | troy42 | Kevin`: well, we all know that scripting languages were created exclusively for system, exec, and popen :-) |
| 01:03.49 | troy42 | who needs other calls anyway |
| 01:04.41 | ivanvujisic | I know it'll be easier to write it in C, but I'm familiar with shell scripting |
| 01:04.49 | florz | *lol* |
| 01:04.55 | Kevin` | both sh and c have conditionals |
| 01:05.11 | ivanvujisic | really? |
| 01:05.12 | troy42 | use asterisk::fastagi or adhearsion or phpagi or whatever |
| 01:05.31 | Kevin` | ivanvujisic: of course, they are proper lanugages |
| 01:05.36 | troy42 | if you know bash or csh, a perl example is going to be easy to follow |
| 01:05.41 | *** join/#asterisk coppice (~chatzilla@m121-203-231-175.smartone-vodafone.com) |
| 01:06.42 | *** join/#asterisk Belgarath (~Belgarath@banda.pl) |
| 01:07.23 | florz | ivanvujisic: by the way, in case you haven't noticed yet, your great shc "protection" puts your extremely secret script into the process list for every user(!) on the system to see |
| 01:08.02 | ivanvujisic | ok, but I have to push asterisk command from C to stdout and I dont know the exact gotoif syntax |
| 01:08.03 | [TK]D-Fender | Some people feel the need to hide the evidence that they can't code.... |
| 01:08.25 | Kevin` | I think this may be hopeless |
| 01:08.26 | [TK]D-Fender | ivanvujisic: YOU DON'T FUCKING USE GOTOIF. WAKE. THE. FUCK. UP, |
| 01:08.43 | xheliox | jumps out of his chair |
| 01:08.47 | [TK]D-Fender | [21:01]<ChannelZ>see SET EXTENSION and SET PRIORITY et al |
| 01:08.51 | xheliox | you have angered the Fender.. |
| 01:09.13 | ivanvujisic | I already use SET VARIABLE |
| 01:09.16 | [TK]D-Fender | ivanvujisic: Stop trying to use a wrench like a fucking hammer. ChannelZ even HANDED you the God-damn answer |
| 01:09.33 | ChannelZ | I give and I give |
| 01:09.43 | [TK]D-Fender | ivanvujisic: set the CONTEX, and EXTEN, and PRIORIT with the damn AGI command you already have and just END your fucking script. |
| 01:10.08 | [TK]D-Fender | ivanvujisic: GotoIF is NOT for AGI. Round peg, square hole |
| 01:10.35 | [TK]D-Fender | ivanvujisic: Appendix C : READ IT |
| 01:11.04 | ivanvujisic | ok, finaly you helped me |
| 01:11.09 | [TK]D-Fender | ivanvujisic: Give how adept yous eem at this, you don't have code WORTH stealing. |
| 01:11.31 | ChannelZ | The Goto and If in GotoIf are two different things in AGI - If is whatever construct conditionals have in whatever language you're using, and Goto is setting the context/extension/priority as I mentioned |
| 01:11.49 | *** join/#asterisk _Eagle_ (~nick@wasteland.net) |
| 01:12.00 | ChannelZ | goes back to fapping |
| 01:12.29 | _Eagle_ | is there a definitive list of everything in asterisk that requires a dahdi timing source? |
| 01:15.50 | WIMPy | _Eagle_: Meetme and page. That should be it. |
| 01:16.08 | [TK]D-Fender | IAX2 Trunk Mode |
| 01:16.50 | WIMPy | Hmm. Does it automagically fall back to normal mode then? |
| 01:16.53 | _Eagle_ | meetme, page, and iax2 trunking? is that all? |
| 01:17.35 | [TK]D-Fender | _Eagle_: Pretty mch. Got a SPECIFC thing you're worried about? |
| 01:17.36 | WIMPy | I have an iax peer in trunk mode bit no dahdi any more. But I haven't seen any failure. |
| 01:17.38 | xheliox | I thought transcoding in general was improved with a timing source? |
| 01:17.53 | [TK]D-Fender | WIMPy: it will simply NOT trunk. |
| 01:17.59 | [TK]D-Fender | WIMPy: They will go independent |
| 01:18.11 | [TK]D-Fender | xheliox: No relation |
| 01:18.15 | xheliox | j |
| 01:18.16 | xheliox | k |
| 01:18.23 | xheliox | no j/k, just k :) |
| 01:18.29 | xheliox | goes back to his cave |
| 01:18.48 | _Eagle_ | i wasn't sure it if was needed to play audio files accurately (playback/background), or for music on hold, or for mp3 playback for music on hold, or anything else like that? |
| 01:19.00 | WIMPy | I guess I can live with a few bits of overhead. |
| 01:19.20 | _Eagle_ | i'm using SIP, not IAX... and only one box will have MeetMe running... so is it 100% safe to not install dahdi_dummy on my boxes that don't use iax or meetme? |
| 01:19.48 | [TK]D-Fender | _Eagle_: or Page |
| 01:19.58 | _Eagle_ | what exactly is page? |
| 01:20.02 | _Eagle_ | is that for pagers? |
| 01:20.21 | florz | and "accurately" is completely irrelevant anyhow in this context |
| 01:20.53 | _Eagle_ | what i meant, florz, is... will the audio playback quality be distorted at all? or less than top quality? without timing? |
| 01:21.39 | WIMPy | And Meetme can be replaced by Confbridge. It just won't do announcements ATM. |
| 01:22.09 | ChannelZ | Page is like a one-way conference so you can yell at people |
| 01:22.18 | _Eagle_ | i've been using asterisk since around 2002 or 2003... but every single box i've set up has had a digium card or ztdummy/dahdi_dummy... so i'm nervous about the idea of setting up production boxes without timing... just want to be sure i'm not making a mistake |
| 01:22.48 | [TK]D-Fender | _Eagle_: and the reason for NOT doing it? |
| 01:23.37 | _Eagle_ | tkd-fender: well first, no digium card... and second, dahdi_test is showing low numbers 99.6%, etc |
| 01:25.23 | [TK]D-Fender | _Eagle_: What do you think is worse? Having a minor variation in a timer.. or NONE AT ALL? |
| 01:26.04 | _Eagle_ | tkd-fender: well that's why i'm asking if anything else uses the timing... if nothing USES it, then why do i need it at all? |
| 01:26.36 | _Eagle_ | adding the dahdi drivers is an extra step that is unnecessary, at least i hope |
| 01:27.50 | _Eagle_ | if i'm sure these machines won't be using iax trunking, meetme, or page... then i just want to be clear that not having dahdi_dummy loaded won't be a bad thing? no negative impact on everything else in asterisk? |
| 01:29.26 | WIMPy | has lived very well without it for several weeks now. |
| 01:30.03 | [TK]D-Fender | _Eagle_: No. |
| 01:30.25 | _Eagle_ | are you sure? |
| 01:31.33 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 01:31.34 | _Eagle_ | do you use asterisk in production without any dahdi drivers installed on the machines? |
| 01:31.58 | [TK]D-Fender | _Eagle_: Now you can't even take "yes" as an answer. |
| 01:32.06 | [TK]D-Fender | Good fucking grief |
| 01:32.33 | [TK]D-Fender | _Eagle_: Another 20 or 30 times to tell you the 3 things that require it? |
| 01:32.55 | _Eagle_ | tkd-fender: i'm sorry if that offended you, i just don't like short answers of "no" after i've typed 3 or 4 lines of questions |
| 01:32.59 | [TK]D-Fender | _Eagle_: If you don't need them then you should be fine without it |
| 01:33.11 | [TK]D-Fender | _Eagle_: Its the same question 4 times.... |
| 01:33.16 | _Eagle_ | ok |
| 01:33.18 | _Eagle_ | thank you |
| 01:34.09 | WIMPy | Could it be that res_timing_timerfd is good enough for iax trunking? |
| 01:34.23 | *** part/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 01:35.31 | florz | given that the intarwebs are gonna add jitter anyhow ... |
| 01:36.15 | _Eagle_ | i just wanted to be thorough... i need to buy about 10 more servers... and this is going to affect that decision... thousands of dollars... and if there was any misunderstanding, and i come back here in 2 weeks and complain, i doubt you or anyone else is going to volunteer to pay for the unusable servers :-) you're gonna say "well, you should have been absolutely sure and not just take the first answer from someone you've never talked to before" |
| 01:36.55 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 01:38.55 | [TK]D-Fender | _Eagle_: if this could cost you thousands of dollars perhaps you shouldn't be looking to cut corners |
| 01:38.55 | WIMPy | didn't know there where special servers that wouldn't let you install dahdi it you wanted to. |
| 01:40.41 | _Eagle_ | tkd-fender: how is it cutting corners if i don't need iax trunking, meetme, or page? why spend an extra $xxx for a digium card or whatever just for timing if it isn't needed? |
| 01:40.55 | _Eagle_ | especially when we're talking 10 servers = 10 cards |
| 01:41.42 | Kevin` | there's a dummy dahdi timer source that works fine, especially if you aren't actually using it normally |
| 01:41.58 | _Eagle_ | i'm also looking at using machines that don't have a pci or pci-e slot in the first place... |
| 01:42.14 | WIMPy | Are you sure hardware helps if it's not coneected to a timing source? I'd doubt it.s better than dahdi-dummy. |
| 01:42.15 | _Eagle_ | kevin: yes, i know about dahdi_dummy... thank you |
| 01:42.23 | [TK]D-Fender | _Eagle_: Timing helps keep MPG123 in-line. * can whine without it. What were planning on using? |
| 01:43.17 | florz | well, the crystals on the digium cards are probably higher quality than your average CPU clock ;-) |
| 01:43.35 | _Eagle_ | i don't think i need mpg123... ill either use wav files for music on hold, or the mp3 add on from asterisk-addons |
| 01:44.00 | _Eagle_ | but last time i tried using the mp3 add on, it crashed my machine... so i switched back to wav files |
| 01:44.51 | _Eagle_ | tkd-fender: but yes, that's the kind of info i'm looking for... a definitive list of why timing is needed in asterisk... so now we have meetme, iax trunking, page, and mpg123 |
| 01:44.59 | Kevin` | florz: is the timer interrupt used based on the adjusted rate that ntp corrects for? |
| 01:45.50 | WIMPy | florz: In that case a GPS receiver might be better than a telephony interface. |
| 01:45.54 | _Eagle_ | anything else i should add to that list? |
| 01:47.44 | Kevin` | why hasn't (or have they?) someone made an adapter with a crystal and divider that connects to serial, parallel, pci interrupt, or something else and just ticks away, for asterisk |
| 01:48.21 | *** join/#asterisk shtoom (~shtoom@115.117.248.164) |
| 01:48.35 | florz | Kevin`: I don't really have a clue - and I guess it doesn't really matter, as ntp probably still has quite a bit of low-frequency jitter anyhow |
| 01:48.48 | drmessano | I believe Sangoma has such a beast |
| 01:49.21 | Kevin` | ntp is just used to correct the drift rate of the high speed oscillator in the computer |
| 01:49.40 | drmessano | http://www.sangoma.com/products/hardware_products/specialty_tools.html |
| 01:50.35 | florz | WIMPy: well, only if the telephony interface is unconnected and you are actually talking to the PSTN somewhere through IP |
| 01:51.01 | [TK]D-Fender | drmessano: VERY nifty |
| 01:51.16 | drmessano | $73 from the first google hit |
| 01:51.43 | WIMPy | florz: Unconnected was my precondition. |
| 01:51.44 | [TK]D-Fender | drmessano: Though they should have added the MB header>USB-B as an adapter and not a separate unit altogether |
| 01:51.50 | drmessano | $65 for the UT51 from the first hit |
| 01:52.23 | Kevin` | that's a bit expensive, but I suppose it's a relatively specialty product |
| 01:52.35 | Kevin` | i'm tempted to make a clone, but what's the point |
| 01:52.36 | drmessano | Ok, that $73 was high... It's $65 for either from a few other sites |
| 01:54.00 | WIMPy | Even a serial port should have an adequate timing source. |
| 01:54.07 | florz | Kevin`: well, yeah, but that correction is not instant and not perfect, obviously - and also the purpose is long-term stability, not short-term stability, the latter being the important thing for telephony |
| 01:54.57 | WIMPy | It's still best if everyone uses the same clock source, off course. |
| 01:56.35 | Kevin` | florz: of course. this is assuming the computer's oscillator has fairly good stability but is off by a constant amount |
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| 01:59.32 | _Eagle_ | thanks for the help, guys |
| 01:59.36 | _Eagle_ | seeya |
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| 05:42.43 | Junior | yello ;) |
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| 06:31.12 | timahvo1 | Hi guys |
| 06:31.23 | timahvo1 | can anybody help me with this http://paste.pocoo.org/show/230809/ |
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| 07:01.26 | athom | Hi guys, is there anyone who is using Vitelity services? |
| 07:01.29 | athom | am I going to pay if I dial my friend's mobile phone (Bulgaria) and he don't answer the call? Are they charge for un-answered calls? |
| 07:03.22 | ChannelZ | I guess it depends on how much they know about call progress |
| 07:04.02 | athom | because Rapidvox are charging |
| 07:04.08 | athom | my friend didn't answer the call |
| 07:04.12 | athom | and I'm still charging.. |
| 07:04.15 | athom | this is terrible :( |
| 07:04.47 | ChannelZ | depending on the path the call takes one can't always know when a call has been answered |
| 07:05.24 | athom | so they maybe will be charging |
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| 07:05.48 | timahvo1 | can anyone please point me in the right direction towards resolving this http://paste.pocoo.org/show/230809/ |
| 07:06.11 | timahvo1 | can make calls between sip phones on the same LAN |
| 07:06.30 | timahvo1 | but can't dial out on the PSTN line |
| 07:06.59 | ChannelZ | Need to see actual dial string |
| 07:08.20 | timahvo1 | ok |
| 07:10.30 | timahvo1 | http://paste.pocoo.org/show/230822/ |
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| 07:10.40 | timahvo1 | ChannelZ: ^ |
| 07:13.56 | ChannelZ | Is DAHDI running and configured correctly? What does 'dahdi show channel 1' say? |
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| 07:18.19 | timahvo1 | ChannelZ: no dahdi channel 1 , channel 25 |
| 07:18.37 | ChannelZ | eh? |
| 07:19.08 | ChannelZ | you're trying to dial out DAHDI channel 1 |
| 07:20.38 | timahvo1 | ChannelZ: am sorry am really new at this can you give me an example of how my extensions.conf should look like to make a call on 25 ? |
| 07:21.46 | ChannelZ | Dial(DAHDI/25/whatevernumber) |
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| 07:24.36 | tzafrir_laptop | that's odd. Is the "," interpreted as part of the dial string? |
| 07:25.52 | ChannelZ | hmm no that's the way it always looks |
| 07:26.02 | tzafrir_laptop | timahvo1, though it's a bit odd if you don't have channel 1 and do have channel 25 |
| 07:26.44 | timahvo1 | tzafrir_laptop: how so ? |
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| 07:27.07 | tzafrir_laptop | timahvo1, what device(s) do you have? |
| 07:27.40 | ChannelZ | will venture a guess of a 2-port PRI card and things are configged on the second port... |
| 07:30.34 | timahvo1 | tzafrir_laptop: Ethernet controller: Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) (rev 11) |
| 07:30.46 | timahvo1 | thats waht I get from lspci |
| 07:30.50 | timahvo1 | what* |
| 07:31.17 | tzafrir_laptop | what's the output from dahdi_hardware ? |
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| 07:31.49 | timahvo1 | pci:0000:14:08.0 wctdm24xxp+ d161:8006 Wildcard AEX410P |
| 07:31.49 | timahvo1 | pci:0000:18:08.0 wcte12xp+ d161:8000 Wildcard TE121 |
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| 07:38.38 | timahvo1 | mmade the change in extensions.conf and get this now --> http://paste.pocoo.org/show/230827/ |
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| 07:43.51 | timahvo1 | tzafrir_laptop: is there away I can force dahdi to configure my channels sequentially, i.e start with 1 instead of 25 ? |
| 07:44.24 | *** join/#asterisk matagou_ (566afda1@gateway/web/freenode/ip.86.106.253.161) |
| 07:44.32 | tzafrir_laptop | timahvo1, what do you have in /etc/dahdi/modules ? |
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| 07:44.52 | matagou_ | hello |
| 07:45.46 | matagou_ | i have an asterisk 1.4.32 + chan_mobile installed from trunk |
| 07:46.45 | ChannelZ | timahvo1: you could unplug your TE121 |
| 07:47.00 | ChannelZ | timahvo1: it's being seen first, consuming channels 1-24 |
| 07:47.29 | matagou_ | an issue appears when two simultaneous calls through chan_mobile are done. It is described here https://issues.asterisk.org/view.php?id=17554 |
| 07:48.26 | matagou_ | so, i want to upgrade the asterisk up to latest 1.6 version, to use chan_mobile from asterisk-addons |
| 07:48.54 | timahvo1 | tzafrir_laptop: ChannelZ http://paste.pocoo.org/show/230828/ |
| 07:49.21 | timahvo1 | thats my /etc/dahdi/modules |
| 07:49.47 | ChannelZ | you might be able to comment out wcte12xp so it won't be seen |
| 07:50.10 | matagou_ | the dialplan under Asterisk 1.4 is compatible with latest asterisk 1.6 ? |
| 07:50.20 | ChannelZ | But then if you decide you need to use that card again in the future all your channel numbers will change again, so why bother? |
| 07:50.37 | timahvo1 | ChannelZ: |
| 07:50.41 | timahvo1 | ok |
| 07:50.58 | ChannelZ | matagou_: mostly but there are some syntax changes, read the UPGRADE-* files |
| 07:51.16 | timahvo1 | what about the error am still getting even after I changed the channel to 25 in extensions.conf ? |
| 07:51.21 | timahvo1 | still the same as before |
| 07:51.52 | ChannelZ | what does "dahdi show channels" tell you? |
| 07:52.29 | matagou_ | ChannelZ: thanks for tip |
| 07:52.34 | tzafrir_laptop | timahvo1, list only those two modules there, in the order you want |
| 07:52.45 | tzafrir_laptop | that is: |
| 07:53.07 | tzafrir_laptop | wctdm24xxp and then, in the next line wcte12xp |
| 07:54.02 | timahvo1 | ChannelZ: Chan Extension Context Language MOH Interpret Blocked State |
| 07:54.05 | timahvo1 | <PROTECTED> |
| 07:54.09 | timahvo1 | <PROTECTED> |
| 07:54.12 | timahvo1 | sorry |
| 07:54.14 | timahvo1 | I hope am not flooding |
| 07:54.40 | timahvo1 | tzafrir_laptop: ok |
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| 07:55.25 | ChannelZ | hmmm |
| 07:55.54 | tzafrir_laptop | timahvo1, after you've edited that file, use: |
| 07:56.32 | tzafrir_laptop | /etc/init.d/asterisk stop; /etc/init.d/dahdi restart; /etc/init.d/asterisk start |
| 07:56.51 | tzafrir_laptop | though I suspect you'll need to reconfigure both, as channel numbering has changed |
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| 07:58.31 | sawgood | By default, when one installed Asterisk 1.6.x, do you then have the ability to send/receive calls in any codec other than G.711 ulaw (or alaw)? |
| 08:00.16 | ChannelZ | yeah.. gsm, g726, g722... |
| 08:00.30 | ChannelZ | some others |
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| 08:01.38 | sawgood | Is G.711 the samething as PCM? |
| 08:02.26 | ChannelZ | g711 is ulaw/alaw - it is a PCM codec yes |
| 08:03.13 | sawgood | ChannelZ: cool ... so what does one have to 'do' in order to have their Asterisk box be able to send/receive calls using G.729? |
| 08:03.30 | ChannelZ | buy licenses for it |
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| 08:03.54 | tzafrir_laptop | have a working g729 codec :-( |
| 08:04.03 | ChannelZ | actually if the two end points can do g729 and Asterisk doesn't need to be in the media stream, I think it can pass thru |
| 08:04.05 | tzafrir_laptop | Buying licenses alone won't help |
| 08:04.07 | ChannelZ | but I could be wrong |
| 08:04.14 | sawgood | Oh ... well, I know when using other (non Asterisk IP PBX boxes) (like for example TalkSwitch and /or Allworx) you can simply choose a different codec in the setup of either their phones or the IP PBX |
| 08:04.55 | sawgood | Maybe the 'license' for them is built into their software??? |
| 08:05.08 | ChannelZ | You can restrict peers to whatever codecs you feel like that they support.. but g729 is commercial so you need to download and install the codec and buy license(s) for Asterisk |
| 08:05.33 | sawgood | oh ok ... I get it better now ... thanks ... I'll read about it at Digium then |
| 08:05.50 | ChannelZ | Yes a lot of devices are licensed for g729 out of the box; other PBXs might bundle them into the cost |
| 08:06.47 | *** join/#asterisk Polysics (~luca@ip-192-59.sn3.eutelia.it) |
| 08:06.49 | Polysics | hello |
| 08:07.09 | Polysics | anyone knows if x-lite supports the URL parameter of dial() |
| 08:07.24 | Polysics | i was trying to make the browser pop up a page with some info when a call comes in |
| 08:07.40 | Polysics | i could also do that with an HTTP client and some AMI magic, but was jsut curious |
| 08:08.23 | ChannelZ | Ask Counterpath. My guess is either no, or only in a paid version of one of their products |
| 08:09.14 | Polysics | is there any client that does the above? |
| 08:09.22 | Polysics | if not i'll just go the HTTP route |
| 08:12.12 | ChannelZ | Zoiper Biz might |
| 08:12.45 | ChannelZ | http://www.zoiper.com/feature_list_zoiper_communicator.php |
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| 08:13.41 | ChannelZ | actually Communicator Free might |
| 08:13.49 | ChannelZ | by the sounds of it |
| 08:13.50 | Polysics | ChannelZ, from your experience, is this kind of stuff best handled through HTTP services? |
| 08:14.05 | ChannelZ | ?? I have no experience with this |
| 08:14.29 | Polysics | :-) |
| 08:14.38 | Polysics | you looked like someone that has seen a lot of stuff :-) |
| 08:14.58 | ChannelZ | you are wanting to pull up info about someone on an incoming call or something? |
| 08:15.25 | Polysics | yes |
| 08:15.39 | Polysics | classic call-center functionality, i suppose |
| 08:16.14 | Polysics | unless they do it by hand, when you call a phone company they already have your info up from your number |
| 08:19.26 | ChannelZ | Half the time that shit makes me punch in my phone number and account number and then they STILL ask me both |
| 08:21.37 | ChannelZ | hmm. Well in spite of their feature grid, all this stuff seems to be ghosted out in the preferences |
| 08:22.44 | ChannelZ | it didn't do anything special providing a URL on Dial() and the options for it to open a URL its self are ghosted out. Hmm. |
| 08:26.43 | sawgood | The G.729 licensing from Digium seems rather straight forward and simple ... did I get the right impression? |
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| 08:30.39 | ChannelZ | yeah.. $10/channel IIRC |
| 08:31.44 | sawgood | nice .... so basically if I give them $10 bucks I could make one call using G.729 ... |
| 08:32.05 | sawgood | What do I have to 'do' on the Asterisk box to make a G.729 call (outside of buying the license) |
| 08:33.25 | Polysics | pricing is pretty honest then |
| 08:33.45 | ChannelZ | you download the g729 module and install it, and then install the license |
| 08:34.37 | sawgood | right ... that is why they allow you to d/l the software from their website for 'free' ... you'll need a license key before it starts to work |
| 08:34.41 | sawgood | got it now ... thanks! |
| 08:34.54 | ChannelZ | there is a separate license tool |
| 08:35.28 | ChannelZ | When you buy licenses you get a code; You run the license tool, type in the code, and then it generates a unique license for your machine and installs it |
| 08:36.28 | sawgood | So, if I had to do a re-install of the box (because it is a LAB unit) that would still be ok (for future testing) |
| 08:39.23 | ChannelZ | Not sure what all infomation they use to generate a fingerprint for your machine |
| 08:39.41 | ChannelZ | MAC address no doubt but not sure what else it may or may not use |
| 08:40.46 | sawgood | thanks |
| 08:41.00 | ChannelZ | "These license files are tied to the |
| 08:41.00 | ChannelZ | <PROTECTED> |
| 08:41.48 | ChannelZ | You can re-register once with a different MAC, any more than that and you have to call Digium and explain |
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| 08:48.55 | Godfather_ | hi |
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| 08:51.57 | ChannelZ | hi.. bye |
| 08:51.58 | ChannelZ | (bed time) |
| 08:53.54 | geemee | Hey Folks... Anyone got experience of Virtualising Asterisk.. Would be for smallish office 40 users and 10-15 remote users. would also probably host small conferences |
| 08:55.12 | sawgood | Any ideas or suggestions on how to make this need happen ... A front desk work has a SIP phone (either a Grandstream or an Aastra), and she needs to know when a voicemails has been left in the general voicemail box (in addtion to her own VM). Does anyone know of a way to make a button on the phone light up if a message is left in the general voicemail box? |
| 08:55.38 | sawgood | Or, to light up her MWI button if a message is left in EITHER the general voicemail and/or her own voicemail box |
| 08:56.12 | sawgood | As a 'work around' I have her SIP phone registering as two different accounts on the same phone |
| 08:56.46 | sawgood | If a VM is left for 'line 1' the MWI light comes up. If a voicemail has been left for line 2 (the VM MWI light does not come on) |
| 08:57.20 | sawgood | Luckly, the Grandstream phone will put a 'mail icon' in the LCD if a VM is left on 'line 2' |
| 08:57.35 | sawgood | it is as close as I've been able to make things work |
| 08:57.35 | geemee | sawgood: not the same but setup email alert? can also have wav of voicemail attached. |
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| 08:58.02 | sawgood | geemee: that is a decent approach too ... |
| 08:58.22 | sawgood | but the company might not always allow her access to a PC and/or a PC might not be available near her phone at all times |
| 08:58.53 | sawgood | You would think this should be as easy as a 'hint' for BLF ... but its not |
| 08:58.53 | geemee | We have it setup here.. works well and remote users can check voicemail via email.. I realise its not what you are after but just a suggestion. |
| 08:59.27 | sawgood | It would be 'cool' if somehow I could make a voicemail box be an 'extension' and then map the 'extension' to a BLF key on the Grandstream phone |
| 09:00.12 | sawgood | geemee: by default (for your Asterisk box sending SMTP messages) are you using sendmail on the box? |
| 09:00.46 | geemee | sawgood: errr.. cant recall exact setup.. I think sendmail with using our exchange server as smarthost. Was simple to setup. |
| 09:01.07 | sawgood | smarthost was pretty easy for me to configure ... |
| 09:01.22 | geemee | simple as in had it up and running in 5 minutes |
| 09:01.26 | sawgood | But now, I disable sendmail on the boxes, and I use a light weight SMTP front end on the box |
| 09:01.56 | sawgood | by-pass sendmail and basically dump emails out as a smart host client to a hosted mail server running sendmail |
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| 09:53.40 | Martinblr | have anybody tried integrating Nokia E63 with Asterisk..? |
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| 10:43.07 | Blackvel | hi all. any IVR gurus here for brainstorming? i am in the need to extend/optimize my current IVR system and status routing playbacks (out off office, vacation, available times). i am looking for the best way to dynamically change times (A-B, C-D or A-D) and to combine the playback prompt with time information (one of my prompts uses fixed times as well as fixed IVR GotoIfTime and would need to be re-recorded for every change - |
| 10:45.10 | Blackvel | is it best to use only fixed times with fixed prompts (like i have right now)? then it is a nightmare to change for e.g 1 week. or how to do it dynamically (including specific time playbacks) |
| 10:45.32 | Blackvel | ? |
| 10:47.31 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 10:47.36 | tzafrir_laptop | Blackvel, "dynamically change" as in edit extensions.conf and reload? |
| 10:47.42 | tzafrir_laptop | GotoIfTime? |
| 10:49.21 | tzafrir_laptop | Also, "playback prompt with time": see the application SayUnixTime |
| 10:50.11 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 10:50.41 | *** join/#asterisk mrchrisadams (~Adium@CPE-203-51-2-76.lns5.cht.bigpond.net.au) |
| 10:51.39 | Blackvel | dynamically change time routings as of variables (gotoiftime, e.g asterisk database) and dynamically circuit switching as of calling 401, 402, 403 which writes variable IVR_CIRCUIT_AVAILIBILITY (this is done already) but would need manual extensions.conf modification for new routings |
| 10:53.04 | Blackvel | e.g for this week i need to have the system open up 10-15 and let the caller know about that time. this is not possible yet with any extensions.conf routing or my current fixed time programming :) |
| 10:53.54 | Blackvel | does it suck when the "normal ivr playback" gets interrupted by a different / standard prompt? |
| 10:54.03 | Blackvel | e.g for saying the time? |
| 10:56.04 | Blackvel | would you record all time prompts for sayunixtime manually? :) |
| 10:57.01 | Chainsaw | Blackvel: Change voice mid-sentence is highly noticeable and tends to sound tacky. |
| 10:57.12 | *** join/#asterisk s14ck (~s14ck@190.72.27.63) |
| 10:57.51 | Chainsaw | Blackvel: Like on some UK train stations: *female voice* The next *male voice* First Capital Connect *female voice* service departs from platform 12. |
| 10:58.22 | Blackvel | hehe yeah ;) |
| 10:58.44 | Chainsaw | Blackvel: It makes you subconsciously question what happened to the female announcer. Was she off sick? Did she get fired? |
| 10:59.03 | Chainsaw | Blackvel: It's likely that your customers will do the same. I'd get both recorded by the same person. |
| 10:59.10 | Blackvel | i feel thisk about this programming too :) |
| 10:59.21 | Blackvel | sick |
| 11:00.41 | Blackvel | does that make sense playback(a) + playback(b) + playback(c) + playback(d) where a is normal introduction, b time from, c "word to" and d time to |
| 11:01.17 | Chainsaw | Blackvel: Should work, yes. You can string things together like that. |
| 11:01.42 | Blackvel | bad thing to record all prompts manually one after one (as you can hear that). |
| 11:02.22 | Blackvel | there is no easy way in german to express english language like : office closed but you can reach us in the following time + a to b |
| 11:02.46 | Blackvel | german always combines the words ...so time a to b is always in the middle of the sentence |
| 11:02.47 | Chainsaw | Your word ordering is completely different, yes. |
| 11:03.02 | Chainsaw | Be glad you don't have to write it. Capitals everywhere ;) |
| 11:04.29 | Blackvel | "You can string things together like that." in one playback? i think i used multiple playbacks |
| 11:04.31 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 11:04.47 | Blackvel | but i fail to do the time thing...didnt use sayunixtime yet |
| 11:04.58 | Chainsaw | Blackvel: In multiple playbacks, yes. That's basically what SayUnixTime does internally anyway. |
| 11:05.28 | Blackvel | would you recommend going back to one prompt? and if time really has to change (in asterisk database) then its a 1 minute thing to re-record 1 prompt which includes time information? |
| 11:05.34 | Chainsaw | Blackvel: I wonder how that is going to work with drei-und-zwanzig vs twenty-three though. |
| 11:06.11 | Blackvel | well e.g its 13:00pm to 14:00pm and 18:00pm to 19:00pm |
| 11:06.46 | Blackvel | and for this week i need 10:00am to 15:00pm |
| 11:07.26 | Blackvel | from or to time can be one recording (not 1 + 3) |
| 11:07.50 | Blackvel | the fastest way probably would be to use only one prompt |
| 11:08.14 | Blackvel | otherwise i would have to record at least 10am, "to", 15pm |
| 11:10.01 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 11:14.06 | Blackvel | honestly i have to say that i do not change time too often |
| 11:14.37 | *** join/#asterisk edgars (edgars@ns.dtg.lv) |
| 11:14.40 | edgars | yo |
| 11:15.11 | edgars | which cpu better performs with voice transcoding amd or intel? |
| 11:19.18 | coppice | well, a 12 core AMD definitely beats the intel atom |
| 11:20.10 | *** join/#asterisk fauxalliance (~gerald@207.231.237.59) |
| 11:21.25 | edgars | :> |
| 11:21.40 | edgars | it also can beat 166mmx too |
| 11:22.21 | coppice | well, if you ask a silly question, you should expect a silly answer |
| 11:22.52 | edgars | very fundamental question |
| 11:24.30 | FutureWeb | hmm anyone knows how to make zaptel display my cards information ? such as version etc etc pls |
| 11:25.12 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 11:25.21 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 11:26.07 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 11:26.26 | tzafrir_laptop | edgars, with an Intel CPU you can do AMD() . With an AMD one you can't do INTEL() |
| 11:26.45 | tzafrir_laptop | So I guess that answers the fundenmental question |
| 11:26.51 | edgars | mhh |
| 11:27.13 | sawgood | If I have Asterisk 1.6.2.9 installed on a box with an unknown type of TDM PCI card, is there an easy way from the CLI to determine which card is installed in the PC? |
| 11:27.23 | tzafrir_laptop | FutureWeb, there's zaptel_hardware, though it's not really as reliable as dahdi_hardware |
| 11:27.39 | tzafrir_laptop | sawgood, dahdi_hardware |
| 11:28.48 | sawgood | well, from the CLI there is a dahdi command |
| 11:28.58 | sawgood | not sure what dahdi_hardware is |
| 11:30.12 | sawgood | found it |
| 11:30.15 | tzafrir_laptop | sawgood, !dahdi_hardware #? |
| 11:30.25 | sawgood | you meant from the Linux CLI ... |
| 11:30.29 | tzafrir_laptop | !/usr/sbin//dahdi_hardware |
| 11:31.03 | edgars | tzafrir_laptop: so intels iX is a good choice :> |
| 11:31.18 | sawgood | tzafrir_laptop: thank you ... I did not know you could run a standard command from the CLI (using a !) |
| 11:31.20 | sawgood | very nice! |
| 11:31.21 | sawgood | thank you! |
| 11:31.40 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 11:31.58 | coppice | edgars: define a goal and you might get a more meaningful answer |
| 11:32.55 | edgars | coppice: main idea is to support ~20-30 simultaneus calls |
| 11:33.20 | edgars | :> |
| 11:35.22 | coppice | G.729A? |
| 11:35.46 | edgars | 711 |
| 11:35.49 | edgars | t38 |
| 11:36.14 | edgars | g.729 is a bad choice :) |
| 11:36.51 | coppice | so, its mostly fax you want? |
| 11:39.09 | edgars | it's not a local office pbx, thats why our sip guy uses 711. Fax and calling server |
| 11:39.38 | edgars | mostly voice calls |
| 11:41.54 | coppice | spandsp will do 100 channels of fax on a single core of any respectable processor |
| 11:44.46 | edgars | and how about voice calls? :) |
| 11:45.01 | FutureWeb | [pdx.ftwb-networks.net ~]# zaptel_hardware |
| 11:45.01 | FutureWeb | pci:0000:03:05.0 wcfxo- 1057:5600 Wildcard X100P |
| 11:45.18 | FutureWeb | I have that anyone know how the zaptel.conf in /etc/ should be especialy the span part? :/ |
| 11:46.21 | coppice | edgars: you said they are all 711 |
| 11:49.59 | edgars | coppice: as i understand 711 is for voice calls too |
| 11:50.23 | coppice | just if everything is 711, there is no transcoding |
| 11:50.37 | edgars | ahh |
| 11:50.54 | edgars | why did he didnt tell me about that :/ |
| 11:53.34 | tzafrir_laptop | FutureWeb, no need for any "span" part for that card |
| 11:54.18 | tzafrir_laptop | FutureWeb, 'zapconf zaptel' should generate one for you... |
| 11:54.35 | tzafrir_laptop | FutureWeb, but then again, do use DAHDI. It's the latest version of Zaptel |
| 11:55.29 | FutureWeb | well yum wont install dahdu lol |
| 11:55.32 | FutureWeb | *dahdi |
| 11:55.39 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 11:57.19 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 11:58.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:00.04 | FutureWeb | my dahdi seems to be messed up or somethin :S |
| 12:01.01 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 12:01.23 | tzafrir_laptop | FutureWeb, please be more specific |
| 12:01.40 | tzafrir_laptop | What distro do you use? |
| 12:02.11 | FutureWeb | CentOS |
| 12:02.37 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 12:02.49 | tzafrir_laptop | I suspect package called 'dahdi' only includes userspace parts |
| 12:03.04 | tzafrir_laptop | And not the kernel modules |
| 12:03.18 | tzafrir_laptop | Where did you install dahdi from? From what yum repo? |
| 12:03.39 | SiNGLer | didn't see the question, but dahdi-tools is userspace utils |
| 12:06.29 | SiNGLer | is there a way to play announcement to both callee and caller after call is answered? |
| 12:09.32 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:10.00 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 12:15.35 | [TK]D-Fender | SiNGLer: Not directly. You could use M() to playback for the CALLEE and Originate() a new local channel to ChanSpy Whisper to the CALLER. |
| 12:16.02 | FutureWeb | grea dahdi didnt find the card :p |
| 12:19.35 | *** join/#asterisk stope (~nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 12:20.38 | stope | I'm trying to compile the addons package with mysql support but the module is not being created, I have the latest source, am I missing extra parameters on ./configure ? |
| 12:22.00 | SiNGLer | you need mysqlclient dev package |
| 12:22.28 | SiNGLer | check if "make menuselect" allows you to select mysql addon |
| 12:25.08 | stope | it's all XXX'd out |
| 12:25.29 | stope | I'll look for that package and install it |
| 12:26.18 | SiNGLer | on debian it's libmysqlclient-dev, not sure about other distros |
| 12:26.25 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:26.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:26.38 | stope | ok, I know the one you're talking about... installing it now |
| 12:27.08 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 12:27.12 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 12:27.16 | nextime | mumble: latest asterisk in sid: if i try to use chan_alsa ( or chan_oss with oss emulation in the alsa support ) i get a zombie process of asterisk if i try to restart it, making it unusable and needing a reboot of the whole server |
| 12:27.30 | stope | thanks SiNGLer, that did the trick :) |
| 12:27.38 | nextime | ( sid => packaged for debian unstable ) |
| 12:28.05 | SiNGLer | np |
| 12:28.22 | [TK]D-Fender | nextime: And what version of * is that precisely? |
| 12:28.43 | nextime | [TK]D-Fender : 1.6.2 |
| 12:28.51 | [TK]D-Fender | nextime: And what version of * is that precisely? <-------------- |
| 12:29.25 | nextime | let me reboot the server and i will check at core show version |
| 12:33.24 | nextime | [TK]D-Fender: *CLI> core show version |
| 12:33.24 | nextime | Asterisk 1.6.2.7-1 built by buildd @ biber on a i686 running Linux on 2010-05-07 11:31:33 UTC |
| 12:33.33 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:34.02 | SiNGLer | [TK]D-Fender: thnx for suggestion, today I'll try to do it |
| 12:38.45 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
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| 12:45.42 | *** join/#asterisk guven (~guven@47.168.55.73) |
| 12:48.38 | *** join/#asterisk anonymouz666 (~anonymouz@201.23.222.159) |
| 12:51.30 | *** join/#asterisk matagou_ (566afda1@gateway/web/freenode/ip.86.106.253.161) |
| 12:51.59 | matagou_ | hello |
| 12:52.38 | matagou_ | can i ask here a question about Asterisk GSM trunks ? |
| 12:53.01 | [TK]D-Fender | matagou_: No such thing... but go ahead |
| 12:54.16 | matagou_ | [TK]D-Fender: i meant things like chan_mobile trunks via Bluetooth GSM phones |
| 12:54.30 | *** join/#asterisk n3hxs (~HAMming@75-145-10-45-Delmarva.hfc.comcastbusiness.net) |
| 12:57.24 | guven | in a system like Core-GWC-Proxy-Asterisk Pbx-Users , is it possible to refer(transfer) a call to someone who is out of the network.I mean A and B are talking and they are in the same network,B is going to refer C and A&C will start talking,but C is in another network.So can asterisk refer a call to outer users ? |
| 12:57.40 | guven | i mean is there a way to send refer through a proxy |
| 12:58.30 | pabelanger | [TK]D-Fender: Actually there is, beroNet has one (http://www.beronet.com). But like most channel drivers, its not integrated into asterisk.org |
| 12:58.59 | matagou_ | i have an asterisk 1.4.32 and chan_mobile revision 421 from asterisk-addon-trunk. The chan_mobile module iq quite unstable - it can crash Asterisk when using 2 simultaneous call through chan_mobile FXOs, it can give errors like mbl_read() read error 104, or so on |
| 12:59.40 | pabelanger | guven: Yes, that is how SIP refers work. |
| 13:00.04 | Chainsaw | pabelanger: They are proud to be using Realtek chips. I'll give that a miss if you don't mind. |
| 13:00.44 | matagou_ | my question is: latest Asterisk 1.6 + chan_mobile from asterisk-addons is more stable and is ready to use in SOHO environment? what alternatives are? |
| 13:01.17 | pabelanger | Chainsaw: There GSM modules use a Siemens chip |
| 13:01.23 | *** join/#asterisk clintc (~clintc@n128-227-87-199.xlate.ufl.edu) |
| 13:01.30 | guven | pabelanger: Thanks for the answer.Interesting thing is that,i'm doing my summer training in a telecommunication company which uses asterisk 1.4x and they say they are not able to refer to outer users because askterisk can't sen refer messages to proxy and asking me to solve the situation |
| 13:01.45 | guven | send* |
| 13:01.48 | Chainsaw | pabelanger: They could use a chip personally hand-crafted by the archangel Gabriel for the GSM side... |
| 13:02.03 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 13:02.13 | Chainsaw | pabelanger: Using a Realtek chip on the PC interface means it'll still fall down under any level of real traffic. |
| 13:03.01 | pabelanger | Chainsaw: Regard less, each card only support 2 channels |
| 13:04.14 | pabelanger | guven: Now you know more then your boss :) |
| 13:04.39 | pabelanger | guven: It is possible, but without any knowledge of your network we are in the dark |
| 13:05.03 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:08.59 | guven | pabelanger: :) Well,simply network is like Core-Gwc-SSL(as proxy server) and Asterisk Pbx |
| 13:09.19 | guven | and they say when they use pi..(forgot the name,an open source pbx again) |
| 13:09.43 | guven | they can use refer without any problem that's why they think asterisk is the problem. |
| 13:09.53 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:10.07 | pabelanger | guven: Can you dial the SIP extension directly, via the Proxy? |
| 13:11.05 | guven | pabelanger: Must be positive,because system works as it should except the refer outer side |
| 13:11.51 | pabelanger | guven: Then, a REFER should work too. What is the actual problem when you transfer to the proxy? |
| 13:13.29 | guven | pabelanger: They were keep saying that asterisk can't send refer to the proxy but seems like noone actually know the exact error..Which makes me wonder around searching for an answer without knowing the exact problem |
| 13:13.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:13.44 | guven | sorry for asking questions without giving too much descriptions by the way.. I don't like this situation either.. Just trying hard now to get the logs |
| 13:13.56 | guven | will ask you guys after having them |
| 13:14.00 | [TK]D-Fender | \o/ |
| 13:14.40 | pabelanger | guven: Sounds like somebody on your staff is lazy! ;) Either way, you got the right idea. Find out the actual problem / error and we can help debug then. |
| 13:14.45 | pabelanger | ~collectdebug |
| 13:14.46 | infobot | methinks collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 13:14.50 | pabelanger | guven: ^^ |
| 13:15.53 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 13:16.08 | guven | pabelanger: Thank you for the informations,i'll get those logs for sure this time,and yes they are pretty lazy.. (: |
| 13:16.14 | matagou_ | please review my posts |
| 13:17.12 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 13:19.36 | pabelanger | matagou_: chan_moblie is not very popular, not sure how stable it is. |
| 13:19.50 | *** join/#asterisk asphere (~ardavis@rrcs-69-193-16-71.nys.biz.rr.com) |
| 13:21.05 | matagou_ | pabelanger: what solution is stable for make/receive calls to GSM networks from asterisk? |
| 13:21.17 | *** part/#asterisk grummund (~grummund@unaffiliated/grummund) |
| 13:21.44 | [TK]D-Fender | matagou_: Some other GSM device that talks SIP to *. |
| 13:22.33 | [TK]D-Fender | matagou_: because * doesn't speak "GSM" |
| 13:23.21 | matagou_ | [TK]D-Fender: please give examples. What about pci GSM adapters or GSM gateway that connects to Asterisk FXOs ? |
| 13:23.23 | pabelanger | matagou_: ^^ A GSM gateway with SIP support |
| 13:24.01 | matagou_ | * to asterisk via FXOs |
| 13:25.03 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 13:25.08 | [TK]D-Fender | matagou_: FXO= CRAP. PCI should work, but that requires a slot in your server and puts a lot of the more interference susceptible bits too close to well... things that might interfere |
| 13:25.28 | [TK]D-Fender | matagou_: SIP would be preferable |
| 13:25.49 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:29.15 | matagou_ | exit |
| 13:29.21 | matagou_ | sorry |
| 13:30.10 | Chainsaw | Right over there sir, next to the toilets. |
| 13:30.28 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:31.56 | E-bola | Do anybody know a way to make hint subscriptions react to DND on/off on snom phones. My blf's work great on my snom phones except they doesnt take DND status into consideration |
| 13:32.50 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:32.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:36.05 | [TK]D-Fender | E-bola: Not possible. |
| 13:36.11 | *** join/#asterisk iamy_china (~yang@221.223.48.204) |
| 13:36.46 | *** join/#asterisk iamy_china (~yang@221.223.48.204) |
| 13:37.22 | *** join/#asterisk iamy_china (~yang@221.223.48.204) |
| 13:37.47 | E-bola | [TK]D-Fender: Mmmm well im pretty positive that u can do it with custom headers, enabling and disabling the hint |
| 13:37.55 | E-bola | but i was hoping for a better/simpler way to do it |
| 13:38.03 | [TK]D-Fender | E-bola: There is no message sent to * for its state <- |
| 13:38.21 | [TK]D-Fender | E-bola: Nothing to track/trigger on |
| 13:38.25 | E-bola | There is when u define the DND on and off codes |
| 13:38.34 | E-bola | which i already to do sync DND status on phone and asterisk |
| 13:38.35 | pabelanger | E-bola: Or have Asterisk control your DND states. |
| 13:38.37 | [TK]D-Fender | E-bola: DND is a "reaction" from the phone, |
| 13:38.48 | E-bola | so asterisk already knows if the phone is on DND or not |
| 13:38.57 | Chainsaw | E-bola: Not until it sends a call there. |
| 13:39.05 | E-bola | wrong all of you im afraid :) |
| 13:39.07 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 13:39.08 | Chainsaw | E-bola: And the phone responds with "No! I'm on DND" |
| 13:39.08 | E-bola | lemme paste my dialplan |
| 13:39.12 | [TK]D-Fender | E-bola: So what happens on * side wthn you hit DND? |
| 13:39.19 | E-bola | hoes to pastebin it |
| 13:39.21 | [TK]D-Fender | E-bola: You saying it dials an extension as specified? |
| 13:39.59 | E-bola | http://pastebin.com/LSjhkUWa |
| 13:40.12 | E-bola | as u can see asterisk simply checks if the phone is on DND before it dials it |
| 13:40.29 | E-bola | the snom phone sends *79 and *78 to asterisk when turning dnd on/off |
| 13:40.39 | DND | good day. i need some help with g729. we have 2 asterisk connected via iax. server A can call Server B with G729 but the problem is B can call A but cant switch to g729 only other codecs |
| 13:41.04 | E-bola | so as I said, asterisk already has the DND status in its DB, im just trying to find the best way to "notify" a hint listener when a phone goes on DND |
| 13:41.09 | DND | i tried restricting it to g729&gsm but server A is switching to gsm. |
| 13:41.09 | [TK]D-Fender | E-bola: that is not "DND". That is "I'm using AsDB with some dialplan to implement something I'd like to call an integraetd feature but isn't fixed" |
| 13:41.27 | [TK]D-Fender | E-bola: If that's how you're doing it.... |
| 13:41.29 | DND | you talking about me? :D |
| 13:41.30 | [TK]D-Fender | ~devicestate |
| 13:41.31 | E-bola | [TK]D-Fender: thats just definitiona nonesence, in effect its DND |
| 13:41.38 | [TK]D-Fender | ~devstate |
| 13:41.39 | infobot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/community/russell/asterisk-1.4/func_devstate-1.4 , or http://www.asterisk.org/node/48325 |
| 13:41.41 | [TK]D-Fender | ^^^^^^ |
| 13:41.51 | [TK]D-Fender | E-bola: native in 1.6.0+ |
| 13:42.02 | E-bola | Yep, already using it to monitor queue's |
| 13:42.17 | [TK]D-Fender | [09:41]<E-bola>[TK]D-Fender: thats just definitiona nonesence, in effect its DND <- definitions are details, and in this world details can get you KILLED |
| 13:42.18 | E-bola | but im not sure how to use it in combination with "normal" hints |
| 13:42.25 | freezey | for the follow me feature. Say i forward a phonecall to a cellphone how do i allow the cellphone voicemail to take the phonecall not the asterisk system? |
| 13:42.50 | [TK]D-Fender | E-bola: add a CUSTOM one in your dialplan code to match the toggle |
| 13:42.58 | [TK]D-Fender | E-bola: Its documented. Go read. |
| 13:43.07 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:43.28 | E-bola | [TK]D-Fender: I know, but sometimes somebody else has already done the exact same thing, and its easier to share |
| 13:43.30 | [TK]D-Fender | freezey: Cell will ANSWER when it takes VM. Why would * take over? |
| 13:43.33 | E-bola | Sharing is good, mmmkkkk :) |
| 13:43.44 | [TK]D-Fender | E-bola: Go read you lazy fuck :p |
| 13:43.47 | E-bola | hehe |
| 13:43.52 | [TK]D-Fender | E-bola: for 1 bloody line. |
| 13:44.03 | [TK]D-Fender | reaches for his trusty ClueBat (tm) |
| 13:46.36 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 13:47.13 | freezey | [TK]D-Fender: because what i am getting is the call dies right before the vmail on their side picks up |
| 13:47.44 | freezey | fot eh destination if no answer |
| 13:48.38 | E-bola | [TK]D-Fender: Im not so sure devstate can be used for this. Because as it is now im monitoring the devicestate of the phone. If i change it to a custom devicestate and try to combine that with the phone's state i dunno how.... |
| 13:48.57 | E-bola | A custom device state doesnt "turn off" the hint automatically |
| 13:49.47 | [TK]D-Fender | freezey: Perhaps you should actually LOOK at the call <--- |
| 13:50.12 | E-bola | Hmm actualy you could get around BUSY and DND with dialplan tricks, but the ringing part i dont see how would be possible |
| 13:50.23 | [TK]D-Fender | E-bola: There is no "automatic" You're doing this in your DND enable/disable code |
| 13:50.40 | [TK]D-Fender | E-bola: And you don't need "ringing" |
| 13:51.16 | freezey | [TK]D-Fender: for the destination if no answer settings pushes to the local system VM but i want it to notdo that lol |
| 13:51.32 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:51.52 | [TK]D-Fender | freezey: Perhaps you should actually LOOK at the call <--- |
| 13:52.25 | freezey | [TK]D-Fender: yeah i am looking at the call and i see where it spawns the end destination in the new stack |
| 13:53.31 | [TK]D-Fender | freezey: The most worthless of emssages |
| 13:53.34 | [TK]D-Fender | messages* |
| 13:53.38 | cusco | hi |
| 13:53.52 | cusco | I am having troubles with echo cancellers in dahdi |
| 13:54.25 | cusco | if i set up other than mg2, ti is either not recognised when I run dahdi_cfg or kernel freezes... |
| 13:55.38 | freezey | [TK]D-Fender: got it... it was all just a timing issue. increased the ring time |
| 13:55.39 | freezey | good now |
| 13:55.43 | cusco | that pri card always worked well, now we moved it phisically to another machine with a new PRI line, and operators are hearing echo |
| 13:55.50 | cusco | so im guessing I need to change the echo canceller |
| 13:58.24 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 13:59.30 | cusco | what can I do? |
| 14:00.25 | E-bola | [TK]D-Fender: Mmmm well i dont NEED to show if a phone is ringing, but its nice after all for the receptionist |
| 14:00.53 | [TK]D-Fender | E-bola: Where exactly is ringing coming into this? |
| 14:00.58 | E-bola | but if i change the hint to a custom one, i have to turn it on and off myself in the dialplan, unlike if u use the normal hints for a SIP account |
| 14:01.16 | E-bola | [TK]D-Fender: notifyringing = yes |
| 14:01.32 | [TK]D-Fender | E-bola: single hint, checking TWO things <---- |
| 14:01.36 | E-bola | I dont JUST need DND monitoring i still need to monitor the extensions normal events (busy,ringing) |
| 14:02.47 | E-bola | So i guess i can make the dialplan do Set(DEVICE_STATE(Custom:MyCustomHint)=INUSE) whenever i do soemthing with the phone |
| 14:03.08 | E-bola | it "ought" to work, but i dont feel confident it wont get out of sync or some other weird stuff hehe |
| 14:03.29 | E-bola | it wouldnt work for queues either |
| 14:03.38 | E-bola | since they dont go through dialplans :( |
| 14:03.47 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:03.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:03.56 | E-bola | Thats a gamestopper :( |
| 14:03.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.00 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
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| 14:04.58 | [TK]D-Fender | E-bola: You only need to set your CUSTOM one for DND purposes. your HINT will look at TWO things. |
| 14:05.24 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 14:05.34 | E-bola | [TK]D-Fender: I didnt know a hint can check 2 different states? |
| 14:09.56 | [TK]D-Fender | reaches for his trusty ClueBat (tm) |
| 14:10.04 | [TK]D-Fender | E-bola: Read. The. DOCS |
| 14:11.07 | E-bola | .... reading |
| 14:11.10 | E-bola | first page didnt mention it |
| 14:11.23 | *** join/#asterisk sulex (~sulex@firewall.blindata.ch) |
| 14:12.40 | [TK]D-Fender | E-bola: & <- |
| 14:12.43 | E-bola | [TK]D-Fender: http://svncommunity.digium.com/community/russell/asterisk-1.4/func_devstate-1.4 is a 404 btw |
| 14:12.49 | [TK]D-Fender | http://www.asterisk.org/search/node/devstate |
| 14:12.51 | [TK]D-Fender | ^^^^^ |
| 14:13.39 | cusco | ok so im reading on how to install oslec |
| 14:15.01 | E-bola | [TK]D-Fender: sorry but unless im blind i cant find anything about how to specify a 1 hint for multiple devices |
| 14:15.11 | [TK]D-Fender | [10:12]<[TK]D-Fender>E-bola: & <- |
| 14:16.00 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:16.04 | E-bola | is not listed ANYWHERE!! |
| 14:16.05 | E-bola | :) |
| 14:16.47 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 14:17.37 | [TK]D-Fender | E-bola: It is. The quality and depth of your research is evident. |
| 14:19.30 | E-bola | not on anywhere you pointed me too |
| 14:20.08 | cusco | it says that I should copy `drivers/staging/echo` from a recent kernel tree (at least 2.6.28-rc1) |
| 14:20.15 | cusco | but Im running kernel 2.6.26 |
| 14:20.29 | [TK]D-Fender | E-bola: Like I said... NO DEPTH. If I don't hand it to you you won't get off your ass to look. |
| 14:21.36 | E-bola | [TK]D-Fender: I obviously did look, and already read what i coudl find before, since i already use the function |
| 14:21.37 | [TK]D-Fender | E-bola: This is even in the basic WIKI page for Asterisk Standard Extensions. |
| 14:21.40 | [TK]D-Fender | ^^^^^^^^ |
| 14:21.46 | E-bola | you even prooved it by showing how a search for the function doesnt provide detailed docs |
| 14:22.29 | [TK]D-Fender | E-bola: And put the app name into asterisk.org immediately produces the good link I gave. Stop being a lazy ass. |
| 14:22.32 | [TK]D-Fender | E-bola: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 14:22.39 | [TK]D-Fender | ^^^^^ clear instructions on multiple devices. |
| 14:22.49 | [TK]D-Fender | E-bola: Separate by "&" |
| 14:23.00 | [TK]D-Fender | E-bola: Its listed elswhere are well |
| 14:23.33 | E-bola | well contrary to what u might think i dont find it obvious to search for asterisk standard exstentions when im looking for info on devicestate :) |
| 14:23.42 | [TK]D-Fender | E-bola: LAZY ASS. |
| 14:23.48 | E-bola | not to mention i was adviced against using voip-infio.org in this very channel |
| 14:23.58 | E-bola | i even think u recomended against it :) |
| 14:24.07 | E-bola | but NEVERMIND hehe |
| 14:24.09 | E-bola | and thanks |
| 14:24.15 | [TK]D-Fender | E-bola: It should simply be the LOWER on the list, not off it entirely. |
| 14:24.43 | Naikrovek | yes |
| 14:24.54 | Naikrovek | it's based on reality but requires references from other sources |
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| 14:53.00 | hurdman | if i have to send some log from asterisk via an udp socket, is there an existing class or i need to make a patch for log ? |
| 14:53.35 | fenrus | do you want it to syslog to a remote host ? |
| 14:53.51 | hurdman | fenrus: to a QT gui |
| 14:53.58 | hurdman | ( c++ ) |
| 14:54.28 | Beave | hurdman: just need a popup when errors occur or something? |
| 14:54.29 | fenrus | hm, perhaps you can pass it to syslog att localhost, and from syslog route the stuff you want to another host/port |
| 14:56.38 | Beave | hurdman: set syslog in the logger.conf, load Sagan (http://sagan.softiwnk.com).. enabled the asterisk.rules (not many there).. use "external program:" in Sagan with a program like "xmessage" (or whatever) |
| 14:56.54 | hurdman | Beave: slot into an ui to "flash some button" |
| 14:56.59 | Beave | if your just looking for popups... |
| 14:57.07 | Beave | ah.. nevermind then. |
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| 14:59.19 | hurdman | i'll extend the ast_log function or something like this |
| 14:59.53 | Beave | errr http://sagan.softwink.com (incorrect link).. anyways... |
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| 15:05.07 | pabelanger | hurdman: why duplicate what syslog can do? |
| 15:05.32 | Beave | pabelanger: I think he just wants a hook into a custom gui app, not sure. |
| 15:05.38 | Beave | but yeah. |
| 15:05.47 | Dksaarth | Hi guys - I have a strange problem - I am trying to originate a call from a sip extension over a pstn line - the destination cellphone rings, but the call is hung up immediatly, resulting in a missed call before anybody can answer the phone. Anybody seen something like this before ? |
| 15:06.32 | Dksaarth | the command i am using in the asterisk cli is - originate SIP/2001 extension 082xxxx317@from-internal |
| 15:06.44 | hurdman | Beave: pabelanger i juste need critical and warning into a QT gui app for a stupid user |
| 15:07.34 | Beave | hurdman: does the QT gui app do anything else? |
| 15:08.04 | hurdman | Beave: yes of course :) |
| 15:08.15 | Beave | heh. okay.. |
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| 15:40.06 | FutureWeb | Just to let you all know, I am offering shells capable of running Asterisk, with a Dedicated Public IP aviable, incase anyone wants one, let me know in a Pm :) |
| 15:40.20 | FutureWeb | thats just to help people who cant host it them self, not to spam btw ;D |
| 15:41.09 | pabelanger | free? |
| 15:41.33 | Qwell | unless it's free, it's spam. so yeah |
| 15:41.40 | FutureWeb | I cant give it out for free, but since I wanna help people instead of making profit |
| 15:41.45 | FutureWeb | if your not makin a profit |
| 15:41.48 | FutureWeb | its not spam :P |
| 15:41.54 | FutureWeb | well thats how I think of it anyhow |
| 15:42.00 | FutureWeb | if you consider it as spam sorry then :/ |
| 15:42.01 | Qwell | yes it is. please refrain from any paid advertisements here |
| 15:42.07 | FutureWeb | kk sorry |
| 15:46.00 | Naikrovek | this Sriracha hot sauce is going to kill me |
| 15:46.43 | Qwell | Naikrovek: worth it. |
| 15:47.10 | Qwell | Naikrovek: ever actually used it as a hamburger, as the bottle suggests? |
| 15:47.13 | Qwell | err, on |
| 15:47.27 | Naikrovek | not yet |
| 15:47.30 | Qwell | it's actually pretty awesome |
| 15:47.33 | Naikrovek | it is awesome |
| 15:47.35 | Qwell | use it instead of ketchup |
| 15:47.38 | Naikrovek | but oh my stomach |
| 15:47.46 | Naikrovek | i must have an ulcer or something |
| 15:47.53 | Naikrovek | it's some of the best hot sauce i've ever eaten |
| 15:48.14 | Naikrovek | and i got it at walmart of all places |
| 15:48.20 | Naikrovek | like $2.50 or something |
| 15:48.22 | Qwell | for like $2.50, heh |
| 15:48.22 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 15:48.23 | Qwell | eyah |
| 15:48.25 | QubeZ | hello all |
| 15:48.40 | Naikrovek | it'll take at least 6 months to get through this bottle |
| 15:48.44 | Qwell | I use the hell out of that stuff... |
| 15:49.07 | Qwell | I get those frozen chimichangas, and smother them with it. |
| 15:49.12 | Naikrovek | smother?! |
| 15:49.14 | Naikrovek | whoa |
| 15:49.16 | Qwell | add some horseradish... |
| 15:49.31 | Naikrovek | you like to have clear sinuses |
| 15:49.38 | Qwell | not really smother. :p I put like 3-4 lines across it |
| 15:49.38 | Naikrovek | sounds like you require them |
| 15:49.52 | Naikrovek | 3-4 lines for me on turkey sandwiches |
| 15:49.54 | Naikrovek | mmmm |
| 15:49.55 | QubeZ | we have a server running Centos 5.3 kernel 2.6.18-92.1.22.el5 and after turning on recordings, the server locks up with random memory errors but never does this when recordings aren't enabled. We have 2 servers exhibiting this behavior and both are set to record calls fulltime. We are currently recording to a ramdisk. Any ideas? |
| 15:50.29 | Qwell | QubeZ: run memtest on that server? sounds like you've got some bad RAM there, that's only getting used when you record |
| 15:50.48 | QubeZ | Qwell: we tested ram and it came back fine |
| 15:51.38 | Qwell | Naikrovek: makes me want to go get thai food today.. |
| 15:51.53 | Qwell | QubeZ: are the other boxes running the same kernel? |
| 15:51.58 | QubeZ | not sure what else it could be, as soon as we enable recordings... boom issues |
| 15:52.30 | QubeZ | Qwell: we've thought about that too, reluctant to upgrade just yet but wanted to shoot out my issue here before proceeding |
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| 15:55.19 | p3nguin | qubez: Did you run memtest86+ on the systems unchanged from the way you use them in production? |
| 15:57.24 | QubeZ | p3nguin: yup, both prod servers have had memtest run on them. Even tested another server that doesn't do recordings and as soon as we enabled recordings, it began to freeze too |
| 15:57.32 | QubeZ | same kernel so what Qwell eluded too many be the real issue |
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| 16:04.46 | wcselby | o/ |
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| 16:34.17 | zxvff | hi, i have an asterisk setup using asterisk and FreePBX. I have a user who claims he called a customer and heard a message that said "the extension xxxx has been routed to phone number xxxxxxxxxx" |
| 16:34.27 | zxvff | any clue what this is or how I can correct it? |
| 16:34.35 | zxvff | tyia |
| 16:34.35 | wcselby | ~freepbx |
| 16:34.36 | 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 |
| 16:34.38 | wcselby | kicks infobot |
| 16:34.40 | wcselby | there you are |
| 16:34.57 | zxvff | okay, wasn't sure if it was freepbx or asterisk related. thanks |
| 16:42.46 | anonymouz666 | ok guys, gotta go. time to win another world cup. |
| 16:42.57 | anonymouz666 | brazil again. |
| 16:59.21 | ChannelZ | I have a cup right here |
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| 17:03.19 | [TK]D-Fender | goes to find 2 girls... |
| 17:03.54 | ChannelZ | Make sure they're from different countries |
| 17:05.40 | p3nguin | I don't know whether to be concerned or turned on. |
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| 17:14.39 | drmessano | 2g1wc? |
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| 17:30.07 | bran | guys |I can't get my Polycom to show up in FreePBX, any ideas? |
| 17:30.22 | pabelanger | ~freepbx |
| 17:30.23 | 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 |
| 17:30.27 | pabelanger | bran: ^^^ |
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| 17:42.15 | Krolik13 | there is any distance limit for an telephone cable between FXS and phone? |
| 17:44.55 | Kyosh | using cat1? |
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| 17:46.38 | [TK]D-Fender | Krolik13: Of course. |
| 17:46.45 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 17:47.25 | [TK]D-Fender | Krolik13: Dependent on the grade of wire, the interface used, the phone used, solar flare activity, and flux in the aurora borealis |
| 17:47.35 | TheSov | dont forget neutrinos |
| 17:48.24 | SiNGLer | and possition of Mars and Venus |
| 17:48.44 | Kyosh | and how much an ant can shit in a day |
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| 18:22.40 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.33.1 (2010/06/22), *-Addons 1.6.1.4, 1.6.0.6 (2010/06/08), 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.2 (2010/06/08) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-b |
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| 18:26.56 | [TK]D-Fender | Krolik13: See all of the OTHER factors as well |
| 18:27.09 | [TK]D-Fender | Krolik13: I'd recommend calling the manufacturer |
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| 18:30.25 | Krolik13 | [TK]D-Fender> thanks |
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| 18:32.35 | Krolik13 | [TK]D-Fender> what E1 (ISDN PRI) equipment for Voice Gateway do you recommend? |
| 18:33.17 | [TK]D-Fender | Krolik13: In 99% of cases, Sangoma A-Series cards |
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| 18:34.42 | Krolik13 | [TK]D-Fender> thanks again for your advice! |
| 18:35.10 | Krolik13 | [TK]D-Fender> what about Digium E1 pci? who many % of 100%? :) |
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| 18:35.35 | Qwell | Krolik13: some people prefer different things. you'll get different answers if you ask different people. |
| 18:36.13 | Qwell | of course, buying Digium hardware helps support Asterisk. |
| 18:36.45 | [TK]D-Fender | Indeed |
| 18:36.46 | Krolik13 | what is better, to user E1 Pci cards, or an external gateway voip hardware? |
| 18:36.53 | [TK]D-Fender | Krolik13: Depends on your needs |
| 18:36.54 | Krolik13 | i'm just new in it. sorry for stupid questions |
| 18:37.08 | Krolik13 | need is: ISDN PRi - 30 channels, E1 |
| 18:37.21 | Krolik13 | i don't mind use PCI card, ori hardware gateways |
| 18:37.28 | Krolik13 | just tell me what is more stable and better |
| 18:37.31 | [TK]D-Fender | Krolik13: For a single PRI and a rather simlpe server, PCI is considerably more cost effective. I recommend SIP gateways for large HA type setups though |
| 18:37.31 | russellb | Digium E1 card! :-) |
| 18:37.37 | Krolik13 | because you have great experience |
| 18:38.35 | Krolik13 | [TK]D-Fender> from the gateways, what brands/models do you recommend? |
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| 18:38.56 | russellb | he recommends a server with a Digium E1 card in it as a gateway :-D |
| 18:39.25 | [TK]D-Fender | needs a new "dummy" ... this one is going all Chuchy-like |
| 18:39.29 | [TK]D-Fender | Chucky* |
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| 18:39.45 | [TK]D-Fender | russellb: Put down that knife! |
| 18:40.00 | Krolik13 | sorry |
| 18:40.22 | Tim_Toady | eggs digium bacon digium digium digium sausage digium and digium |
| 18:42.17 | Krolik13 | [TK]D-Fender> is it this one: Sangoma A101 Single Voice and Data Card (one E1) port? |
| 18:42.45 | [TK]D-Fender | Krolik13: A101d <- w/ HWEC |
| 18:43.33 | russellb | pouts |
| 18:43.53 | wcselby | lol |
| 18:44.07 | Krolik13 | [TK]D-Fender>what about A101DE ? |
| 18:44.08 | [TK]D-Fender | russellb: "I'm gonna pout at you until I get my way..." - Hootie & The Blowfish |
| 18:44.20 | [TK]D-Fender | Krolik13: Sure if you need it in PCI-E |
| 18:44.42 | russellb | [TK]D-Fender: that band is from my hometown. |
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| 18:45.18 | [TK]D-Fender | russellb: That song is on my perform list. |
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| 18:47.22 | Krolik13 | [TK]D-Fender> sorry for borring you, but just to confirm if i understand right: you wanna say that A101DE is better then digium's TE121B ? |
| 18:47.31 | wcselby | just got invited to SW:TOR beta, kinda |
| 18:47.36 | Krolik13 | thanks again for all your recommandations |
| 18:47.42 | [TK]D-Fender | Krolik13: IMO |
| 18:48.20 | Krolik13 | thanks mate |
| 18:48.22 | Krolik13 | ! |
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| 18:49.52 | evilbit | hi, reading up on asterisk and receiving SMS txt messages. I'm a little confused. Can it be done with asterisk using just VOIP and a DID number? |
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| 18:50.41 | Krolik13 | [TK]D-Fender> just forget to ask you about client side: FXO ports.. what gateways should be find for about 8-24 FXO ports? |
| 18:51.50 | SiNGLer | Krolik13: you can try Audiocodes SIP GW |
| 18:52.27 | pabelanger | evilbit: define SMS |
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| 18:52.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:53.02 | evilbit | pabelanger: cell phone txt messages |
| 18:53.09 | [TK]D-Fender | Krolik13: anything above 8 I'd look at PRI options very hard. 8-12 I'd aim for PCI probably. 12+ would be SIP gateway. Either AudioCodes or Mediatrix |
| 18:53.22 | evilbit | so, I'd like to txt back and forth from my cell to asterisk |
| 18:54.05 | Krolik13 | [TK]D-Fender> asterisk and PCI E1 will be quite far from the clients. So i need to place several gateways, far away on the LAN, closer to the clients... |
| 18:54.27 | pabelanger | I don't recommend Audiocodec. Only because configuring them the first time stinks. |
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| 18:55.07 | Krolik13 | Telecom -> E1->PCI->Asterisk->LAn->(Switches)->Lan->FXO_Gateways->Phones |
| 18:55.20 | pabelanger | evilbit: You will need a SMS service provider for your Asterisk box. |
| 18:56.15 | evilbit | ah, ok... so if I have a DID from a iax provider there's no way to use that same number for SMS? |
| 18:56.18 | [TK]D-Fender | pabelanger: Startup curve isn't a great detminating factor. Think long term quality |
| 18:56.55 | pabelanger | evilbit: No, because your provider does not supply SMS services. |
| 18:57.01 | Krolik13 | [TK]D-Fender> if to choose between AudioCodes or Mediatrix, what do you recommend? |
| 18:57.10 | Krolik13 | for FXO_Gateways |
| 18:57.27 | [TK]D-Fender | Krolik13: Kinda a toss-up. Might say either... |
| 18:57.39 | pabelanger | [TK]D-Fender: They are good boxes; just terrible to use in a lab environment. |
| 18:58.24 | Krolik13 | [TK]D-Fender> i should relay on your experiense, as i don't have option to test any of those gateways. Thanks! |
| 19:00.41 | [TK]D-Fender | Krolik13: I don't have a lot of experience with these external gateways, but a decent viewpoint as to what makes to use and for what kind of usage. |
| 19:00.59 | [TK]D-Fender | Krolik13: Your specific needs may vary. |
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| 19:05.04 | Deeewayne | What's the highest number of digium/sangoma quadspans that people have running in a single production box without problems? |
| 19:05.14 | jdoe | Why would the feature codes for ChanSpy not work? Voicemail works, so I assume button presses are being sent correctly. |
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| 19:05.48 | WIMPy | Deeewayne: There are also 8xE1 and E3 interfaces. |
| 19:06.39 | Deeewayne | WIMPy, sangoma? |
| 19:08.18 | WIMPy | Indeed |
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| 19:11.48 | Krolik13 | [TK]D-Fender> does E1 PRI ISDN type of connection has specific signals (like SIP has) for disconnect event, instead of detecting disconnect tone? |
| 19:12.10 | russellb | yes, but only on digium cards |
| 19:12.10 | russellb | ducks |
| 19:12.21 | SiNGLer | it does, by protocol |
| 19:12.33 | [TK]D-Fender | Krolik13: Yes, this is the point of PRI. OOB signalling |
| 19:12.42 | Krolik13 | [TK]D-Fender> wandefull!!! |
| 19:13.02 | [TK]D-Fender | Deeewayne: Digium has always maintained you shouldn't have more than 2 cards in a given server. |
| 19:13.17 | [TK]D-Fender | Deeewayne: That in mind one could say 16 Ports |
| 19:14.03 | Krolik13 | russellb> are you fan of digium? ;) |
| 19:14.17 | russellb | Krolik13: that's my employer |
| 19:14.26 | Krolik13 | oh, i see :) |
| 19:14.27 | Deeewayne | [TK]D-Fender, I've heard that before not sure that digium agrees w/ that limitation |
| 19:14.30 | leifmadsen | and he's a fan |
| 19:14.56 | leifmadsen | Deeewayne: isn't there some sort of satisfaction guarantee if it didn't work? :) |
| 19:15.41 | Deeewayne | leifmadsen, wouldn't the guarantee be more like if it doesn't work, we'll give you your money back? |
| 19:15.55 | [TK]D-Fender | Deeewayne: these recommedations were directly from asterisk.org |
| 19:15.59 | leifmadsen | Deeewayne: if multiple cards didn't work, that sounds like "not working" |
| 19:16.05 | leifmadsen | Deeewayne: I'd check with sales first |
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| 19:31.10 | jdoe | Anyone? 1.6.2.9, ChanSpy with an explicit channel works, #/* don't work for scanning though. |
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| 19:37.41 | [TK]D-Fender | jdoe: Show us |
| 19:38.00 | jdoe | [TK]D-Fender: sure, what would you like to see? |
| 19:38.14 | [TK]D-Fender | jdoe: I dunno.. that actual PROBLEM maybe? |
| 19:38.22 | [TK]D-Fender | jdoe: Show us it "not working" |
| 19:38.33 | [TK]D-Fender | jdoe: And enough backup to clearly indicate that it should have |
| 19:38.54 | jdoe | [TK]D-Fender: I'm not sure how to demonstrate that when I push a button nothing happens. |
| 19:39.12 | *** join/#asterisk n3hxs (~HAMming@75-145-10-45-Delmarva.hfc.comcastbusiness.net) |
| 19:39.36 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:39.50 | jdoe | I can pastebin configs, but core/sip debugging didn't show anything obviously relevant. |
| 19:42.59 | evilbit | when a sip user dials a outside (of asterisk) number where does the CID get set? |
| 19:45.02 | jdoe | [TK]D-Fender: http://pastebin.com/qYbLEjGy is extensions.conf. What I want is for an extension to be able to dial 999 and cycle through the channels in the CSR group. In practice what happens is that I get dropped into ChanSpy, get dead air, and no button presses do anything. |
| 19:45.13 | [TK]D-Fender | jdoe: I want to SEE the channels that are active, and I want to SEE you calling chanspy |
| 19:45.22 | jdoe | sec. |
| 19:45.57 | *** join/#asterisk chuckf_ (~chuckf@ubuntu/member/chuckf) |
| 19:46.52 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 19:46.58 | *** part/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 19:47.51 | jdoe | [TK]D-Fender: http://pastebin.com/x1fwP990 |
| 19:48.29 | [TK]D-Fender | jdoe: "core show channels concise" |
| 19:48.36 | [TK]D-Fender | jdoe: sip show channels = crap |
| 19:48.43 | jdoe | sure, one sec. |
| 19:48.50 | pabelanger | Anybody using Broadvoice? Do they support / supply DNS SRV records for SIP phones? |
| 19:49.07 | jdoe | ... well shit, that IS more useful. |
| 19:49.08 | jdoe | heh. |
| 19:49.58 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
| 19:50.00 | jdoe | [TK]D-Fender: http://pastebin.com/X226Muvs |
| 19:50.40 | [TK]D-Fender | jdoe: options b: Only spy on channels involved in a bridged call. |
| 19:50.54 | [TK]D-Fender | jdoe: Nobody is BRIDGED |
| 19:51.10 | [TK]D-Fender | jdoe: Perhaps yous hould actually read the options you are using with your applications |
| 19:51.20 | jdoe | [TK]D-Fender: the queue doesn't bridge calls? |
| 19:51.40 | *** join/#asterisk residentgrey (~residentg@174-152-236-250.pools.spcsdns.net) |
| 19:54.11 | jdoe | [TK]D-Fender: 3343 and 3288 are caller and callee respectively, my understanding was that when a queue call gets picked up the two are bridged. |
| 19:54.26 | [TK]D-Fender | jdoe: 2 PEOPLE aren't talking together. |
| 19:54.44 | [TK]D-Fender | jdoe: When the schmuck gets ANSWERED then there is something to spy on |
| 19:55.02 | [TK]D-Fender | jdoe: He's sitting IN LINE to be answered |
| 19:55.12 | jdoe | [TK]D-Fender: no he's not, they're in a call. |
| 19:56.56 | [TK]D-Fender | jdoe: Not sure if you can do that. Try a more direct call. |
| 19:57.11 | jdoe | more direct? how do you mean? |
| 19:57.14 | Krolik13 | [TK]D-Fender> about E1 PCI A101d, there is any processing load over the server's CPU during voice calls? if so, how can i calculate the resources that i need for an specific amount of concurent calls? |
| 19:57.18 | [TK]D-Fender | jdoe: DIAL |
| 19:57.45 | jdoe | [TK]D-Fender: dial what? 3343 dialed to enter the queue. |
| 19:57.58 | [TK]D-Fender | Krolik13: the card doesn't place any load on your server, and you can fully load a quad-port card on anything worth even thinking about for * in general |
| 19:58.11 | [TK]D-Fender | jdoe: ... Direct Dial. No Queue |
| 19:58.32 | [TK]D-Fender | jdoe: Could be that Chanspy will only look at Dial-based calls |
| 19:58.49 | jdoe | I don't think that's the case, if I give ChanSpy SIP/3343 it spies on the channel |
| 19:58.53 | jdoe | it's only the cycling that's not working. |
| 19:58.58 | [TK]D-Fender | jdoe: When using that "bridged option" anyway |
| 19:58.58 | Krolik13 | [TK]D-Fender> great! what abou *? what resouces does it need for a specific amount of concurent calls? |
| 19:59.06 | jdoe | [TK]D-Fender: oh, hmm |
| 19:59.17 | [TK]D-Fender | Krolik13: What do you intend to do exactly>? |
| 19:59.26 | jdoe | [TK]D-Fender: even that would be a little strange. MixMonitor is smart enough to use "b" to only record when two people are actually SPEAKING on the call. |
| 20:00.02 | Krolik13 | [TK]D-Fender> kind of call-center. People to call to some numbers, to be set on queue, and wait for answers from support team. |
| 20:00.08 | jdoe | [TK]D-Fender: was kinda hoping I could do the same here... realistically I could just just do something silly like 99XX in extensions.conf and use that to give exact channels to ChanSpy I guess... |
| 20:00.50 | [TK]D-Fender | Krolik13: call recording? Where are you agents erlative to your server? |
| 20:01.51 | Krolik13 | [TK]D-Fender> yep, call recoding, agents are in the LAN (soft or hardphones) |
| 20:02.46 | [TK]D-Fender | Krolik13: Core2 Duo type server, pair of decently fast HDs in RAID 1 or a 5+ array |
| 20:03.10 | [TK]D-Fender | Krolik13: 4 gig ram. Which when you get down to it is practically common spec for analog watches these days |
| 20:03.19 | Krolik13 | [TK]D-Fender> what CPU frequesncy? |
| 20:03.32 | [TK]D-Fender | Krolik13: Won't really matter. |
| 20:03.45 | [TK]D-Fender | Krolik13: Just go for what looks like a fairly decent PC today |
| 20:03.48 | Krolik13 | why exactly 4 gb? how did you calculate it? |
| 20:04.07 | [TK]D-Fender | Krolik13: I calculate by "max out 32bit OS" cost effective |
| 20:04.26 | [TK]D-Fender | Krolik13: You won't haev to think about it for a long time |
| 20:04.40 | Krolik13 | [TK]D-Fender> thanks much for you support and time! |
| 20:04.50 | [TK]D-Fender | Krolik13: You're welcome. |
| 20:05.18 | [TK]D-Fender | Krolik13: My advise is free, and my support very accessible :) |
| 20:05.49 | Krolik13 | [TK]D-Fender> you are like gift for us! God bless you |
| 20:05.56 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:05.56 | [sr] | hi |
| 20:06.14 | [TK]D-Fender | is a regular 'ole Jack-In-The-Box |
| 20:07.22 | Qwell | orders a Sourdough Jack and seasoned curly fries |
| 20:09.11 | wcselby | lol |
| 20:10.56 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:12.10 | [sr] | haha |
| 20:12.16 | [sr] | just discovered something interesting |
| 20:12.20 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 20:12.30 | [sr] | the state phone company uses & support asterisk |
| 20:12.40 | [sr] | it's on the asterisk official video |
| 20:12.50 | Qwell | there's an Asterisk official video? |
| 20:13.14 | [sr] | well |
| 20:13.21 | [sr] | the video that is on asterisk.org front page |
| 20:15.11 | wcselby | asterisk.org front page recently changed again? |
| 20:15.43 | [sr] | "get started video" |
| 20:15.44 | [sr] | :p |
| 20:31.47 | *** join/#asterisk guilhermebr (~Guilherme@189.63.64.235) |
| 20:32.29 | *** join/#asterisk twanny796 (~twanny@78.133.65.141) |
| 20:37.36 | *** part/#asterisk zxvff (shahid@vps.wtfux.org) |
| 20:39.35 | pabelanger | ~book |
| 20:39.35 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:42.31 | wcselby | wakka wakka |
| 20:43.34 | twanny796 | Skype for asterisk could not be loaded? Any skype users? |
| 20:45.41 | *** join/#asterisk rvleij (robin@rsus.fx-services.com) |
| 20:46.52 | *** join/#asterisk mza- (~adam@hypnos.fscker.com) |
| 20:47.04 | mza- | is there a good howto on setting up TLS? |
| 20:47.10 | mza- | im sure it's been asked 100's of times |
| 20:47.12 | *** join/#asterisk cesar_CR (~cesar@201.196.220.82) |
| 20:47.31 | mza- | SSL3_READ_BYTES:tlsv1 alert unknown ca |
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| 20:57.45 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 20:57.45 | *** mode/#asterisk [+o file] by ChanServ |
| 20:59.27 | boodu | hello |
| 20:59.40 | mza- | hello |
| 21:03.41 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
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| 21:19.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 21:49.46 | WIMPy | San someone spot why this incomming sip call gets disconnected immediately? http://wimpy.yeti.dk/pastebin |
| 21:49.50 | WIMPy | Can |
| 21:52.07 | Dksaarth | wimpy, have you got sdpignoreversion in the peer definition? see https://issues.asterisk.org/view.php?id=16238 |
| 21:52.13 | Dksaarth | (total random guess) |
| 21:54.43 | WIMPy | Outgoing is ok for me, but I'll give it a tr. |
| 21:54.48 | WIMPy | y |
| 21:54.54 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 21:55.55 | WIMPy | Nope, no difference. |
| 21:55.58 | Joe_CoT | is there any way to change the formatting of cli command output? I'm trying to parse the output from cli commands over the manager interface, and it's rather hard to separate fields in the output |
| 21:56.12 | Joe_CoT | something like csv, or even tab delimited, would make things much easier. |
| 21:57.30 | tzafrir_laptop | Joe_CoT, not really. Consider using the manager interface? |
| 21:58.39 | Joe_CoT | tzafrir_laptop, I am using the manager interface. But there not all the commands i need are manager commands, so I have to run them as cli commands through the manager. |
| 21:59.03 | Joe_CoT | And that works fine, but it outputs it like the cli would, and that output isn't very clean to parse. |
| 21:59.23 | daemon | hey all im just setting up queing for asterisk |
| 21:59.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:59.37 | daemon | I use ilbc, how much bandwidth per user shall I allow fr |
| 21:59.38 | daemon | for |
| 21:59.49 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 22:01.59 | *** join/#asterisk JAMMAN2110 (~James@unaffiliated/jamman2110) |
| 22:06.20 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 22:12.34 | *** join/#asterisk cweagans (~cweagans@67.42.166.69) |
| 22:13.03 | cweagans | is it possible to have two pbx boxes behind a single firewall on a single public IP address and still have them both register and make calls and such? |
| 22:13.26 | cweagans | register to an external sip provider, that is |
| 22:15.34 | [TK]D-Fender | cweagans: run them on separate SIP ports & RTP ranges |
| 22:15.58 | cweagans | [TK]D-Fender: I'm not really sure how to do that.. |
| 22:16.20 | [TK]D-Fender | cweagans: si.conf, rtp.conf. blatantly obvoius parms |
| 22:16.20 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 22:16.29 | cweagans | [TK]D-Fender: heh, thanks |
| 22:19.38 | Micc | is sip info the same as sip notify? |
| 22:19.39 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 22:20.15 | Blackvel | what would be the fastest way to manually record my own voice for various times which i want to use prompts for? create extensions to record every hour manually? i have no programs to record all hours by one and then split the wav/gsm afterwords :( |
| 22:20.31 | cweagans | Blackvel: Audacity.sourceforge.net |
| 22:21.36 | jdoe | [TK]D-Fender: well I figured out at least part of my problem. The SPYGROUP channel variable is somehow, somewhere, getting unset. I'm getting dead air in chanspy etc. because it has no 'valid' channels to pick from. Any insight on how/when variables get cleared? |
| 22:21.48 | Blackvel | would you do it like this ? would you use sayunixtime or would you try to just to pass the variables from database e.g _myxyztimehhfrom to playback? |
| 22:22.24 | Blackvel | e.g announce_abc_timefrom1300 |
| 22:23.05 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 22:23.30 | Micc | it looks like sip info and sip notify dtmf methods are different, does asterisk support sip notify dtmf method? |
| 22:25.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:26.34 | *** join/#asterisk bran (~chatzilla@unaffiliated/bran) |
| 22:26.43 | bran | why doesn't my polycom 330 auth with asterisk? |
| 22:26.46 | bran | <--- Transmitting (no NAT) to 192.168.1.201:5060 ---> |
| 22:26.48 | bran | SIP/2.0 403 Forbidden (Bad auth) |
| 22:26.50 | bran | Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK91c3010e28424E09;received=192.168.1.201 |
| 22:27.06 | fenrus | looks like incorrect username/pw |
| 22:27.39 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:27.46 | bran | i use the extension # as username and the secret as password right? |
| 22:28.09 | [TK]D-Fender | bran: YES |
| 22:30.06 | jdoe | [TK]D-Fender: nevermind. Root cause is that I'm an idiot and it was a dial plan issue. SPYGROUP was only being set some of the time. Facepalm. Thanks for your help earlier. |
| 22:30.39 | *** join/#asterisk Scorcerer (root@czlug.icis.pcz.pl) |
| 22:31.22 | bran | [TK]D-Fender: but i am using the right username and password.... |
| 22:31.51 | bran | <phone1> |
| 22:31.53 | bran | <PROTECTED> |
| 22:31.55 | bran | reg.1.label="201" reg.1.type="private" reg.1.lcs="" reg.1.csta="" reg.1.thirdPartyName="" reg.1.auth.userId="201" |
| 22:31.57 | bran | reg.1.auth.password="lol123" |
| 22:32.13 | bran | so that's 201/lol123 |
| 22:32.21 | bran | i don't know why it's not working :( |
| 22:33.37 | jdoe | sec. |
| 22:34.44 | jdoe | bran: try reg.1.address="201" |
| 22:35.15 | bran | oh? |
| 22:35.52 | jdoe | bran: I don't have the docs handy right now to confirm that's correct, it's how my configs are though and they work. |
| 22:36.10 | [TK]D-Fender | because addess is NOT the IP of your server |
| 22:36.14 | [TK]D-Fender | it is there USER |
| 22:36.19 | [TK]D-Fender | the* |
| 22:36.22 | bran | damn |
| 22:36.28 | bran | if that's it i'll kill myself |
| 22:36.50 | [TK]D-Fender | bran: Please stand on that plastic sheet over there |
| 22:36.50 | jdoe | bran: that was frequently my response while dealing with polycom configs. |
| 22:36.56 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 22:37.12 | [TK]D-Fender | bran: And press the barrel firmly to the temple |
| 22:37.21 | [TK]D-Fender | loads up a few more sub-sonic rounds |
| 22:39.22 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 22:41.02 | bran | holy fuck it works |
| 22:41.09 | bran | wow.... |
| 22:41.50 | bran | alright |
| 22:41.55 | bran | i love you guys |
| 22:42.27 | bran | how do I dial to a landline? |
| 22:42.37 | bran | there's a default in here that's 9|. |
| 22:42.39 | jdoe | pick up receiver, press buttons? |
| 22:42.43 | *** join/#asterisk aidinb (~Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 22:42.57 | bran | but i can only seem to dial up to 10 digits into the phone |
| 22:43.05 | bran | if I dial 9 first, don't I need 11? |
| 22:44.21 | cweagans | bran: 1) Install AsteriskNow |
| 22:44.23 | cweagans | 2) ??? |
| 22:44.25 | cweagans | 3) Profit! |
| 22:46.41 | bran | i am using asterisk now! |
| 22:48.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:50.53 | cweagans | can anybody help me get my 2 pbx's working? |
| 22:51.33 | pabelanger | ~ask |
| 22:51.34 | 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. |
| 22:51.42 | pabelanger | cweagans: ^^^ |
| 22:51.54 | cweagans | I have two PBX systems behind a single firewall with a single public IP. One of them completes calls through Speakeasy sip trunks intermittently (when it doesn't, it's asterisk error code 21 - call rejected). The other PBX will not complete a call at all. Same error code. |
| 22:51.55 | cweagans | :) |
| 22:51.57 | [TK]D-Fender | bran: Go fix the dialplan on the phone |
| 22:52.32 | cweagans | I've tried changing the SIP ports and the RTP ports. One of them is on 5060 and the other is on 5061 for sip |
| 22:52.41 | cweagans | for rtp, pbx1 is 10000-20000 |
| 22:52.48 | cweagans | pbx2 is 20001 to 30000 |
| 22:53.27 | cweagans | I don't know enough about VoIP systems I guess....I may be able to convince my boss to pay somebody for fixing it, if anybody is interested |
| 22:55.24 | bran | [TK]D-Fender: got it |
| 22:55.46 | bran | right now if somebody calls in, Asterisk picks up and says "This number is not in service..." |
| 22:55.53 | bran | where is the setting for this? |
| 22:56.12 | cweagans | create an inbound route (or whatever it's called in FreePBX) |
| 22:57.22 | [TK]D-Fender | bran: #freepbx <---------------- |
| 22:58.16 | bran | hmm k |
| 22:59.22 | Trixboxer | Hi, does anyone have got success for cpanel and asterisk installation on the same system .. i'm going to do it.. just want to have some reviews :) |
| 23:01.29 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:03.10 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 23:06.37 | *** join/#asterisk radic (~radic@dslb-094-216-241-201.pools.arcor-ip.net) |
| 23:07.03 | radic | will that work? >> http://bravo.hopto.org/~radic/asd.txt |
| 23:13.45 | [TK]D-Fender | radic: exten => asd,n,Set(NUM=Math(${NUM}+1,i)) <- massacred syntax on a priority that will never execute anyway |
| 23:14.10 | [TK]D-Fender | radic: exten => asd,n,GoTo(asd,2) <- Also never going to execute |
| 23:15.42 | radic | [TK]D-Fender: I have many loops like this and I naver had a problem |
| 23:16.01 | WIMPy | Can someone spot why this incomming sip call gets disconnected immediately? http://wimpy.yeti.dk/pastebin |
| 23:18.28 | radic | WIMPy: dialplan waere noch hilfreich |
| 23:19.45 | [TK]D-Fender | WIMPy: 1st allowing all codecs = ICK. Second, I'm betting you didn't prevent reinvites. |
| 23:19.54 | WIMPy | You can see it being executed there. I does an Answer(), send an OK and receives a CANCEL, but I don't see why the call gets cancelled at the moment of answering. |
| 23:19.54 | [TK]D-Fender | [2010-06-28 23:39:37] Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90d (g723|ulaw|alaw|g726|g729) <- BAD |
| 23:20.54 | p3nguin | That's a LOT of codecs that will never get used. |
| 23:21.08 | WIMPy | I tried alaw only and I also tried without reinvites. But I changed to just Answer and Musiconhold for testing. |
| 23:21.27 | [TK]D-Fender | WIMPy: What you've shown begs to differ and looks bad |
| 23:21.42 | [TK]D-Fender | WIMPy: And doesn't include configs |
| 23:21.56 | WIMPy | It made no difference so I allowed all again. |
| 23:22.56 | WIMPy | Unfortunaletly I don't knpw since when that's broken, as its a number I don't uasually use. |
| 23:23.24 | [TK]D-Fender | WIMPy: Also why do we only see a PART of the OpenSER part of this? |
| 23:23.54 | WIMPy | That's the full call. |
| 23:24.02 | [TK]D-Fender | WIMPy: the ONLY part of that comm we see is the CANCEL. |
| 23:24.25 | WIMPy | And yes, I also noticed huawei and openser being mixed.. |
| 23:24.34 | [TK]D-Fender | WIMPy: text search your pastebin. ONE occurence |
| 23:24.50 | [TK]D-Fender | WIMPy: My trust has shunk to microscopic proportions |
| 23:26.01 | WIMPy | The only interesting thing I see is more than one IP, which looks quite interesting. |
| 23:27.27 | WIMPy | The call seems to come in twice from different IPs. |
| 23:29.12 | bran | so i did a yum install asterisk16-skypeforasterisk |
| 23:29.15 | bran | where did that stuff go? |
| 23:29.20 | bran | it's now showing up under modules |
| 23:31.23 | *** join/#asterisk kotp (~vgoff@96.2.187.67) |
| 23:31.53 | [TK]D-Fender | bran: Did you buy and install your licenses? |
| 23:32.01 | bran | not yet |
| 23:33.05 | [TK]D-Fender | bran: Then don't expect to see the module load |
| 23:35.37 | WIMPy | Hmm. Is it possible that the provider actually isn't cancelling the call answered call, but the other call, that was also generated? |
| 23:36.01 | bran | [TK]D-Fender: ok i just used register to enter my key |
| 23:36.08 | WIMPy | And * getting confused by that double call? |
| 23:36.49 | WIMPy | At least it confuses me, but that doesn't mean much :-) |
| 23:37.10 | *** join/#asterisk mrchrisadams (~Adium@CPE-58-168-41-94.lns7.cht.bigpond.net.au) |
| 23:40.05 | bran | i still don't see the module listed in asterisk tho |
| 23:42.51 | cweagans | bran: asterisk != freepbx |
| 23:43.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:43.14 | bran | how do I check what modules are loaded from asterisk's prompt? |
| 23:43.25 | WIMPy | Ok, so I guess that CANCEL is ok. So the question is why the BYE comes immediately. |
| 23:53.54 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 23:54.10 | ManxPower | Anyone have any ideas on where in the actual ael file this error occurs? "LOG: lev:4 file:ael.flex line:647 func: ael_yylex Unhandled char(s):" |
| 23:55.20 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 23:56.15 | bran | can i configure skype lines in freepbx at all? |