| 00:09.39 | *** join/#asterisk XVampireX (n=serge@unaffiliated/xvampirex) |
| 00:09.46 | XVampireX | Are there any interesting voip channels here? |
| 00:09.56 | XVampireX | other than this one |
| 00:10.35 | crimethinker | this one is interesting? |
| 00:11.13 | XVampireX | I guess |
| 00:11.30 | XVampireX | I can't even find a better client than twinkle :P |
| 00:11.41 | crimethinker | uh, epic? |
| 00:12.36 | XVampireX | Say, asterisk aside, is it possible to setup a "ringtone" which plays music while someone calls you, i mean, so that they hear the ringtone while they wait for me... like it is possible to do that with regular cellphones... |
| 00:12.41 | XVampireX | I don't know how this feature is called |
| 00:12.50 | XVampireX | I mean to do it in SIP |
| 00:14.29 | JT | i'm not sure if it's possible to give callers the shits in that way yet |
| 00:14.30 | crimethinker | Your callers will get upset when they get billed starting the moment they finish dialing, whether you answer or not. |
| 00:14.50 | crimethinker | as you will have to answer a call the moment it is received to do that. |
| 00:14.51 | JT | crimethinker: that depends on the setup |
| 00:14.54 | XVampireX | no, not pc to phone |
| 00:14.56 | JT | crimethinker: not true |
| 00:14.57 | XVampireX | or phone to pc |
| 00:15.14 | XVampireX | I just want a platform for telephony :) |
| 00:15.20 | crimethinker | ... |
| 00:15.34 | XVampireX | I'm also really interested in something like skype which supports large conference rooms |
| 00:15.40 | XVampireX | with presence |
| 00:16.31 | *** join/#asterisk dalfry (n=dalfry@70.89.177.109) |
| 00:16.39 | dalfry | hello |
| 00:16.52 | dalfry | need help with some agi stream_file issues |
| 00:16.57 | JT | crimethinker: telcos don't bill it from the start of ringing indication |
| 00:17.03 | JT | crimethinker: it's called early media |
| 00:17.34 | crimethinker | sweet! so I can program early media with my message of the day, and distribute for free? |
| 00:18.18 | XVampireX | cr4z3d, what? |
| 00:18.34 | cr4z3d | what? |
| 00:18.42 | cr4z3d | how did i come aprt of this conversation haha |
| 00:18.45 | dalfry | running asterisk 1.4.4 release with phpagi playing mp3 files using the stream_file call |
| 00:19.01 | XVampireX | agi or api? :P |
| 00:19.22 | JT | crimethinker: however it's normally provided by telco systems, using intelligent networking features |
| 00:19.41 | dalfry | and when two people are listening to the same mp3 file, at times, the playback gets robotic |
| 00:20.00 | dalfry | are there known issues with format_mp3 which might be causeing this? |
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| 00:20.22 | JT | i wouldn't mind if "caller tones" died |
| 00:20.30 | JT | playing music for ringing indication is uber annoying |
| 00:20.43 | XVampireX | ? |
| 00:20.45 | XVampireX | no |
| 00:20.50 | XVampireX | it's not |
| 00:21.06 | JT | yes, if i call people and they have it on, it gives me the shits hardcore |
| 00:21.15 | JT | i don't want to listen to their shit taste in music |
| 00:21.23 | JT | i want to listen to a ringing indication |
| 00:21.59 | XVampireX | lol |
| 00:22.29 | JT | it's more annoying than people with stupid ringtones |
| 00:22.38 | JT | on their mobile phones |
| 00:23.22 | *** join/#asterisk psi0n (n=123@109.80-202-238.nextgentel.com) |
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| 00:23.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:23.59 | XVampireX | JT, Maybe, i actually didn't think of it much... |
| 00:24.02 | Nugget | heh, from a review of a seagate hard drive: "Began making a loud, annoying noise after about 2 years. Died without warning after 5 years." |
| 00:24.09 | Nugget | "without warning" ftw :) |
| 00:24.10 | XVampireX | But i learned about a pretty good song from those things :P |
| 00:24.31 | XVampireX | I'm also wondering how to setup a good voicemail message |
| 00:27.43 | psi0n | does anyone have any experience with the asterisk-1.2.4-silence_suppression-4.patch ( http://bugs.digium.com/view.php?id=5374 ), which supposedly fixes stuttering MoH caused by silence suppression? |
| 00:29.26 | rikstah | how is a loud noise without warning? or is that your point |
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| 00:35.00 | Nugget | that's the point. |
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| 00:38.24 | lee_is_me | LOL, my wife has a ringtone for everyone in our family from country songs to rock. I'm guilty of having having Bob Marely and Rare Earth only. |
| 00:39.39 | lee_is_me | Rare Earths version of "feeling alright" is worth the space on my phone... |
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| 00:43.05 | leleobhz | have someone linked to asteriskbrasil.org staff here? |
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| 00:43.57 | JT | i see |
| 00:44.40 | leleobhz | JT: ? |
| 00:45.00 | JT | leleobhz: i wasn't replying to you |
| 00:45.08 | leleobhz | ah, sorry |
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| 00:45.56 | XVampireX | Has anyone here tried Jingle? |
| 00:46.15 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 00:46.17 | Strom_M | has anyone here ever dialed a phone ??!?!?!?!?!?!?!?!?!?!?!?!!??!?!!?!?!?!!!!! need help here people |
| 00:46.21 | Damin | Wee... |
| 00:46.33 | Damin | Strom_M: What is this "phone" you speak of? |
| 00:47.00 | Strom_M | its this thing plugged into my cat that has the numbers 1 through jizz on it |
| 00:47.13 | *** part/#asterisk leleobhz (n=leleobhz@unaffiliated/leleobhz) |
| 00:48.47 | XVampireX | steven|c, I work in phone surveys place, so... yeah... |
| 00:50.18 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 00:50.58 | JT | XVampireX: does your autocomplete ever get the right person? |
| 00:51.42 | XVampireX | I am used to konversation, so no. |
| 00:52.04 | cnile | where can I set the wait time before my zaptel answers the phone? |
| 00:52.07 | XVampireX | Strom_M, I work in phone surveys place, so... yeah... |
| 00:53.57 | Strom_M | XVampireX: it was a bad joke |
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| 00:57.30 | psi0n | hmm can i upgrade asterisk from 1.2 to 1.4 using yum? |
| 00:58.16 | crimethinker | of course. Will it keep your current configuration? Who knows. |
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| 00:59.39 | Strom_M | is your current configuration going to break horribly in 1.4? who knows. |
| 00:59.50 | psi0n | lol |
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| 01:32.54 | _charly_ | hi, i have a problem with asterisk v1.2 connected to a siemens hipath pbx using oh323 v0.7.3. everything is working except for one thing: if i make a call from asterisk out to a busy extension of the hipath i don't get busy signalled on my phone, it just times out after a minute. the hipath is sending user busy in q931, i've checked this with ethereal and also in the tracefile. is there something i have to change in the config? |
| 01:33.42 | Strom_M | _charly_: you've got to handle the busy condition in your dialplan |
| 01:33.56 | Strom_M | use gotoif() along with the ${DIALSTATUS} variable |
| 01:37.12 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 01:38.40 | _charly_ | Strom_M: that doesn't work, i would have already tried that if Dial() would recognize the busy state. ${DIALSTATUS} contains NOANSWER |
| 01:40.50 | _charly_ | the hipath signals user busy in a q.931 progress message, don't know if that could the problem |
| 01:44.52 | _charly_ | any other idea? |
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| 01:54.37 | _charly_ | the hipath sends a callProceeding directly after asterisks setup, 0.1sec later there's the progress with user busy (cause 17), and 35 secs later it sends a release complete with normal call clearing (cause 16). the dialplan is continued after cause 16, and ${DIALSTATUS} is NOANSWER although the called user is busy |
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| 02:03.46 | Waverly360 | Hey guys, I'm running version 1.2.14 of asterisk but for some reason agi_calleridname isn't being set when I call in. That should have been added in 1.2.11 from what I've read. Any ideas? |
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| 02:11.16 | Waverly360 | Anyone? |
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| 02:33.12 | _charly_ | i just tried it with yate, and it's the same, no busy although the called phone is busy :/ |
| 02:35.39 | Hmmhesays | my god die hard 2 has some horrible acting |
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| 02:44.44 | killfill_ | hey, what means NOTICE[93113]: chan_sip.c:1796 ast_sip_ouraddrfor: Warning: Re-lookup of '' failed!? |
| 02:46.50 | killfill_ | im trying to call from my ekiga |
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| 03:03.37 | killfill_ | damn.. ekiga wont dial out to the pstn.. :S |
| 03:05.46 | PioneerVM2 | anyone know what happens if you have two matching patterns in a context? Like: exten => _1X. and exten => _X. |
| 03:05.52 | PioneerVM2 | will it do both in order? |
| 03:05.57 | PioneerVM2 | or just go to the first matching one? |
| 03:06.14 | killfill_ | it will go in the first match, ordered by priority |
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| 03:20.58 | PioneerVM2 | kill: sorry had to check on baby |
| 03:21.11 | PioneerVM2 | so the first thing it matches, it will only stick with that match? |
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| 03:21.40 | PioneerVM2 | so once it matches _1X., it will stick in that match by priority and not match anything else? |
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| 03:26.02 | *** join/#asterisk minesh (i=minesh@203.88.149.166) |
| 03:26.12 | minesh | hello thr |
| 03:31.12 | minesh | n e body can help me out to know about asterisk..? |
| 03:32.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:33.15 | Strom_M | only if you agree not to ever butcher the English language again by abbreviating simple words like "anybody" |
| 03:33.15 | Strom_M | because, seriously, you ADDED keystrokes there |
| 03:33.48 | minesh | yes sure |
| 03:34.10 | minesh | i will try not to butcher English... |
| 03:35.00 | Strom_M | so what do you want to know? |
| 03:35.27 | minesh | i would like to implement the asterisk... |
| 03:36.05 | minesh | i am little bit confuse about the cli configuration...is there graphical user interface that will help me out to understand the thing |
| 03:36.21 | Strom_M | well, there are GUIs, but they're all horrible |
| 03:37.04 | Strom_M | and they all take away the immense flexibility you get from learning to use the dialplan configuration |
| 03:38.07 | demlak | minesh there are config files... you donīt have to configure asterisk on the fly |
| 03:38.30 | minesh | yes dear i know that... |
| 03:38.49 | demlak | hmkay |
| 03:38.56 | minesh | but i have little bit confusion about same.. i have gone through the files which are made as samples while installation... |
| 03:39.12 | Strom_M | "yes dear"? are you two married? |
| 03:39.42 | minesh | yes...i am too married... |
| 03:39.46 | minesh | what about you.? |
| 03:40.00 | Strom_M | you completely misinterpreted what I was asking |
| 03:40.29 | Strom_M | check one: English is my [ ] third [ ] fifth [ ] next language |
| 03:41.13 | minesh | [x] next |
| 03:41.18 | Strom_M | cocks. |
| 03:41.22 | *** join/#asterisk putnopvut (n=putnopvu@69-94-204-193.biltmorecomm.com) |
| 03:41.46 | demlak | i searched weeks for a GUI-config solution... all was crap.. finaly i learned most by reviewing configs of other people.. and by using this links: http://www.asteriskguru.com/tutorials/ and for people understanding german language http://www.das-asterisk-buch.de/ |
| 03:42.12 | Strom_M | minesh: just read the book |
| 03:42.16 | Strom_M | ~thebook |
| 03:42.26 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:42.26 | demlak | these two links are all i need |
| 03:43.30 | minesh | thaks for the book reference... |
| 03:43.52 | minesh | can you give me some references about the .conf files that can give me some ideas.. |
| 03:44.00 | demlak | i did |
| 03:44.03 | Strom_M | minesh: I did |
| 03:44.38 | minesh | yes but i need actual implemented .conf files if possible.. |
| 03:45.26 | Strom_M | minesh: um |
| 03:45.33 | Strom_M | READ WHAT WE LINKED YOU TO |
| 03:45.51 | Strom_M | or hire a consultant if you're too lazy |
| 03:46.11 | demlak | $50/10min im your asterisk save |
| 03:46.17 | demlak | slave |
| 03:46.21 | nestAr | lol |
| 03:46.28 | Strom_M | i'm a relative bargain at $150 an hour |
| 03:46.49 | minesh | Thaks dear |
| 03:46.59 | JT | ? |
| 03:47.06 | JT | in English? |
| 03:47.07 | Strom_M | just a hint: don't address strangers as "dear" |
| 03:47.16 | JT | indeed |
| 03:47.22 | Strom_M | it's very unnerving |
| 03:47.39 | demlak | free hugs for all! |
| 03:47.41 | demlak | =) |
| 03:48.00 | minesh | hey he has given me valuable suggestion so i called dear.... |
| 03:48.10 | Strom_M | no |
| 03:48.24 | minesh | ok if you dont like.. |
| 03:48.27 | Strom_M | you don't do that. |
| 03:48.41 | demlak | let me say it in rough words... you donīt fuck me.. you donīt call me "dear" |
| 03:48.53 | sbingner | demlak, yes dear |
| 03:49.09 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.10 | demlak | pfff =) |
| 03:49.20 | sbingner | somebody had to do it... |
| 03:49.25 | demlak | sure.. |
| 03:49.49 | JT | minesh: calling a stranger "dear" is highly patronising |
| 03:50.01 | JT | and makes people think you have female fatitude |
| 03:50.11 | sbingner | it might be a girl... |
| 03:50.13 | *** join/#asterisk stridernzl (n=neville@125-237-100-97.jetstream.xtra.co.nz) |
| 03:50.14 | Strom_M | it's insulting, to put it mildly |
| 03:50.21 | minesh | fine JT..and thanks |
| 03:50.23 | Strom_M | even if minesh /is/ a girl |
| 03:50.38 | sbingner | if she's got and wants to come by, she can call me dear! |
| 03:50.44 | sbingner | s/got/hot/ |
| 03:51.05 | sbingner | but yea :p |
| 03:51.18 | JT | sbingner: only females with fatitude say dear in such a context :) |
| 03:51.57 | sbingner | what IS fatitude supposed to mean? lol |
| 03:52.09 | stridernzl | when setting up a remote extension using routers/NAT @ both ends would i have to open up a port for the * server? |
| 03:52.24 | Strom_M | stridernzl: yes |
| 03:52.28 | sbingner | stridernzl, yes |
| 03:52.35 | Strom_M | and that's a recipe for disaster if you're using SIP |
| 03:52.35 | stridernzl | what the port number ? |
| 03:52.47 | sbingner | stridernzl, that would depend on what port number |
| 03:52.54 | stridernzl | me using sip .. I think so why ? |
| 03:53.05 | stridernzl | if that was meant for me ? |
| 03:53.05 | JT | sbingner: overweight females who overcompensate with bad attitude... fat attitude |
| 03:53.16 | demlak | <- using openvpn.. never have problems with ports =) |
| 03:53.25 | sbingner | SIP is primarily 4569, but you;ll need more ports... |
| 03:53.37 | JT | that's IAX2 |
| 03:53.40 | sbingner | er |
| 03:53.51 | Strom_M | cocks |
| 03:53.56 | demlak | .oO(poor wall!) |
| 03:54.05 | sbingner | I meant 5060 |
| 03:54.28 | Strom_M | 5060 UDP for the signaling + 10000-20000 UDP for the media |
| 03:54.31 | stridernzl | is there a doc file i can read then :), I don't really want to be opening stuff i don't have too :) |
| 03:54.55 | stridernzl | so 5060 will handle everything to the remote ? |
| 03:55.35 | JT | sbingner: open udp 5060 and 10000-20000 and you're all done |
| 03:55.55 | sbingner | JT, you mean stridernzl |
| 03:56.01 | JT | stridernzl: |
| 03:56.03 | JT | yes |
| 03:56.19 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-225.ph.ph.cox.net) |
| 03:56.35 | stridernzl | JT: Cheers just writing that down ... :) , It may be why i cannot register ... |
| 03:58.06 | PioneerVM2 | Anyone know how to share a match -- for example I want to do: exten => _X.,1,SetCallerID, then exten => _1X.,1,Call Exten A, exten => _2X.,1,Call Exten B and have "SetCallerID" run before both |
| 03:58.16 | PioneerVM2 | so i dont have to duplicate the SetCallerID twice |
| 03:58.22 | JT | stridernzl: should only need 5060 to register |
| 03:58.36 | stridernzl | JT: Would i just open that up @ the * end / or both sets of routers ? |
| 03:58.46 | JT | both ends |
| 03:58.56 | JT | i don't know what your setup is |
| 03:59.06 | stridernzl | JT: thats nice I will take it 1 step at a time then .. theres another couple of days :) |
| 03:59.10 | Strom_M | PioneerVM2: no, you can't match multiple things like that |
| 03:59.19 | PioneerVM2 | yea i know that it cant be done that way |
| 03:59.25 | PioneerVM2 | but what is the easiest way to accomplish that |
| 03:59.37 | Strom_M | you'd have to match one extension and then use conditional branching to jump elsewhere |
| 03:59.53 | PioneerVM2 | i was thinking that but can u match on a pattern? |
| 03:59.59 | stridernzl | JT: in N.Z ... 2 x DI-704P's (1 Each end) .. couple of DI 304P's as modems |
| 04:00.13 | PioneerVM2 | so exten => _X.,Set Caller ID then do "if match _1X. go here" |
| 04:00.23 | stridernzl | JT: thats dlink gear |
| 04:00.26 | Strom_M | PioneerVM2: sigh |
| 04:00.30 | Strom_M | it's ONE EXTRA LINE |
| 04:00.55 | Strom_M | don't overcomplicate your dialplan by having clever tricks |
| 04:00.57 | PioneerVM2 | not sure what you mean by that comment |
| 04:01.06 | PioneerVM2 | Strom -- its not a "clever trick" im using a simple example |
| 04:01.20 | PioneerVM2 | if I want to have multiple initial things and not have to duplicate and keep them all the same |
| 04:01.25 | PioneerVM2 | it's smart coding |
| 04:01.51 | Strom_M | well how about having one extension which then does different things based on what you dial? |
| 04:01.55 | Strom_M | that would be smarter coding |
| 04:02.07 | PioneerVM2 | yes that was what i was asking, for alternatives |
| 04:02.27 | PioneerVM2 | What command do i do to test the pattern and jump? |
| 04:02.34 | Strom_M | GotoIf() |
| 04:02.51 | Strom_M | or perhaps the IF() function |
| 04:03.06 | PioneerVM2 | im ,looking but dont see it uses patterns only static #'s |
| 04:03.16 | PioneerVM2 | need something that uses patterns |
| 04:03.31 | Strom_M | your expressions can use regular expressions |
| 04:03.38 | PioneerVM2 | ahh |
| 04:03.46 | PioneerVM2 | oh wait, i see you can jump to "subroutines" |
| 04:03.46 | stridernzl | JT: but the routhers might just need a fiddle , somstuff works o.k so stuff needs a bit of tweaking, so when 10 mins ago tried and did not register, thought what next , - I had not tried the Router so howto do .. so port 5060 it is - thats a big help I'd say :) |
| 04:03.47 | Strom_M | so $[expr1:expr2] where expr2 is a regexp |
| 04:04.06 | PioneerVM2 | ahh ok, dont see it in this example ill look that up |
| 04:04.59 | *** join/#asterisk bbryant (i=Brett@65-182-39-142.cre.bil.biltmorecommunications.net) |
| 04:05.33 | PioneerVM2 | hey strom, maybe u can help with this -- check out this problem im having with caller ID |
| 04:05.35 | PioneerVM2 | http://www.pastebin.ca/501219 |
| 04:06.13 | PioneerVM2 | Set(CALLERID(num)= is not working with SIP, but works with IAX... |
| 04:07.44 | PioneerVM2 | I'm thinking it's either a Asterisk SIP issue, or a VoicePulse Connect issue -- strange part is that if i dont set it at all, the phone # of caller passes thru |
| 04:08.09 | PioneerVM2 | unless VoicePulse for SIP is incorrectly taking the number from the (name) rather than (num) |
| 04:08.37 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:08.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:08.42 | Strom_M | i'd blame voicepulse |
| 04:11.24 | *** join/#asterisk basilisk (n=jerry@192.18.43.225) |
| 04:14.11 | PioneerVM2 | yea im starting to think its on there end |
| 04:14.42 | PioneerVM2 | it worked when I changed to "SetCallerID(123)" however i think that was for the incorrect reason, as that really set the "name" field, i think they are pulling the # from the name field in SIP for some reason |
| 04:14.48 | PioneerVM2 | i just wrote them about it |
| 04:15.51 | Strom_M | SetCallerID is very deprecated |
| 04:16.02 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 04:18.05 | adorah | Hi does anyone know how to use an extension to open door? |
| 04:18.32 | rudholm | Strom_M: have you ever thought about the use of "deprecated" to mean "no longer supported" or "discontinued"? |
| 04:19.58 | Strom_M | rudholm: yes, but in this case i'm being silly in the same way as calling something "very dead", "extremely dead", or "deader than the soviet union" |
| 04:20.08 | rudholm | I see |
| 04:21.07 | Strom_M | you know, i'm obviously completely bonkers. I've gone out twice to a sushi place in west LA, and it's been closed both times, and i'm now considering making the drive a third time in two days because I know they're open now |
| 04:21.32 | rudholm | you could call them first. you *do* have a telephone, don't you :-) |
| 04:22.28 | rudholm | I recall there's a WE 2D2 towering over your desk, threatening to squash you during the next earthquate |
| 04:22.32 | rudholm | quake |
| 04:22.44 | Strom_M | oh, like you don't have one also? |
| 04:22.45 | Strom_M | ;) |
| 04:23.15 | rudholm | Mine's a 2C2 |
| 04:23.30 | rudholm | totally different. |
| 04:23.39 | rudholm | I mean, does your support ground start??? |
| 04:23.40 | Strom_M | boners |
| 04:23.48 | rudholm | (and mine's not hanging over me) |
| 04:24.41 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-64.plaicmtc01.tx.dh.suddenlink.net) |
| 04:25.00 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 04:26.04 | Strom_M | heh |
| 04:26.09 | Strom_M | http://www.calrollfactory.com/ |
| 04:26.15 | Strom_M | it's a yellow page |
| 04:27.02 | rudholm | you better go to the Beverly Hills location. I mean, do you really want to eat in GTE-land? |
| 04:27.09 | pabs3 | with sip, how does asterisk handle people connecting from different locations at the same time? would it dial both peers at once? I'm wondering if an IP phone at an office will prevent people dialling in with a softphone from home? |
| 04:27.31 | aptura | the 5 min web page made in yellow |
| 04:27.38 | rudholm | you mean using the same authentication credentials? |
| 04:27.42 | JT | pabs3: i don't understand that question at all |
| 04:28.32 | rudholm | yeah, there's more information in the meta tags than in the visible html :) |
| 04:29.04 | aptura | Cannot wait for the day of electrically aligned ink based news papers. |
| 04:29.19 | rudholm | I like clean and simple HTML. It reminds me of 1995 |
| 04:29.24 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
| 04:30.05 | rudholm | speaking of Japanese food in Beverly Hills, I wonder if Gonpachi has finally opened... |
| 04:30.29 | *** join/#asterisk darkpixel (n=kvirc@c-71-59-168-108.hsd1.wa.comcast.net) |
| 04:31.07 | aptura | yea just opened |
| 04:31.13 | rudholm | sweet! |
| 04:31.18 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 04:31.19 | pabs3 | JT: a sip phone registers itself as user foo to the bar.com asterisk server, a softphone does the same. now sip:john@bar.com calls sip:foo@bar.com. Will the softphone or the sip phone receive the call? |
| 04:31.35 | rudholm | only, what, two years behind schedule? |
| 04:31.35 | aptura | very nice looking resteraunt. |
| 04:31.41 | rudholm | you been in yet? |
| 04:32.04 | rudholm | the company that owns it tends to have great restaurants |
| 04:32.09 | rudholm | (Global Dining) |
| 04:32.12 | aptura | no I was just reading about it on some critics page :) |
| 04:32.17 | rudholm | ah |
| 04:32.22 | rudholm | what'd the critic have to say? |
| 04:32.25 | aptura | http://www.global-dining.com/en/news/la_gonpachi/index.cfm |
| 04:32.55 | rudholm | cool |
| 04:32.56 | aptura | I used to work in a japanese resteraunt as a teen. Interesting experaince. |
| 04:33.05 | rudholm | in LA |
| 04:33.05 | rudholm | ? |
| 04:33.08 | aptura | no |
| 04:33.09 | aptura | Tacoma |
| 04:33.10 | aptura | :) |
| 04:33.13 | rudholm | ah |
| 04:33.46 | aptura | Then the two men in black came in looking for the manager. I did not know but thay were INS agents :) |
| 04:33.57 | rudholm | heh |
| 04:34.08 | rudholm | this review doesn't make sense |
| 04:34.37 | rudholm | oh, I guess they mean the first Gonpachi |
| 04:34.42 | JT | pabs3: you don't register 2 sip clients under the same account at the one time. |
| 04:34.45 | rudholm | not the first GD restaurant |
| 04:34.51 | JT | pabs3: use different sip accounts |
| 04:39.15 | pabs3 | JT: hmm, ok. thats a bit annoying. was hoping to be able to connect from multiple locations |
| 04:39.24 | JT | pabs3: what for? |
| 04:39.48 | rudholm | pabs3: you can connect from multiple locations |
| 04:40.05 | rudholm | just create a sip user for each location/device/whatever |
| 04:40.10 | pabs3 | JT: mainly so people can answer work calls when they are at home sick or something |
| 04:40.29 | JT | pabs3: so call the relevant locations |
| 04:40.31 | rudholm | there are simple ways to address that |
| 04:40.40 | JT | pabs3: you can call multiple phones at once |
| 04:40.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 04:40.46 | JT | or call certain phones first |
| 04:40.47 | pigpen | Hi all...I was looking into the custom device state using presence. I am trying to figure out the syntax if I wanted to "monitor" if a db value was 1 (light on) or 0 (light off). Ideas? |
| 04:40.55 | darkpixel | If anyone's up to answering a T1 question, it would be greatly appreciated... The company I work for decided they wanted to drop their 10Mb DSL connection and replace it with a T1 line. When I asked why, I was told I would be setting up a phone system along with internet access over the T1. To be honest, I'm not much of a telco guy, but I've played around with Asterisk at home.... |
| 04:41.03 | darkpixel | Is it possible to take a (multiplexed?) T1 that has both Data and Voice channels and connect it into one of Digium's cards and be able to access both the voice and data channels? In other words, can the Asterisk box use the voice channels in Asterisk and the data channels for internet access? |
| 04:41.22 | JT | darkpixel: i know the sangoma cards can |
| 04:42.27 | darkpixel | JT: I dug through google a bit, but I'm coming up short on any details or hints on how to set this up...do you have any ideas? |
| 04:42.51 | JT | darkpixel: for sangoma, it's something you setup at the wanpipe level |
| 04:44.02 | pabs3 | rudholm, JT: any examples of such a setup on the voip-info wiki? |
| 04:44.57 | shido6 | ZzZZ |
| 04:45.21 | darkpixel | JT: Thanks for the pointer. I have a stack of reading sitting on the printer about the telco side of things and about interfacing a T1 to a Linux box. Nerd fun. ;) |
| 04:45.23 | shido6 | um |
| 04:45.32 | shido6 | darkpixel, yes |
| 04:46.20 | shido6 | do u have the info on what channels are what? |
| 04:46.27 | shido6 | for the T1 |
| 04:47.39 | darkpixel | shido6: The LEC (qwest) is trying to sell me on their "Integrated T1" which they say will dynamically shrink/grow the voice channels as needed when calls come in or are placed. |
| 04:48.12 | shido6 | LOL!!!!! |
| 04:48.15 | shido6 | i know that one... |
| 04:48.16 | rudholm | pabs3: not sure. but it's easy, just do Dial(SIP/user1&SIP/user2) |
| 04:48.33 | shido6 | u can make the Zaptel card an "Integrated T1" |
| 04:48.42 | shido6 | and tell Qwest to shove it |
| 04:49.31 | *** join/#asterisk Mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 04:49.35 | Mattwj2005 | hey guys |
| 04:49.37 | Mattwj2005 | :) |
| 04:50.14 | pigpen | darkpixel got me thinking...I am needing to bring a PRI into my * box...it will provide some fax services then pass a PRI out to another PBX.... |
| 04:50.30 | pigpen | this is possible with Asterisk & Digium right? (I forget) |
| 04:51.14 | Mattwj2005 | just playing with my Asterisk box |
| 04:51.19 | Mattwj2005 | setting it up again |
| 04:52.26 | darkpixel | shido6: I love telling telco's to shove it. I should get to reading through my stack of printouts before I start asking a ton of lame T1 questions... Thanks for your help. |
| 04:52.50 | shido6 | pigpen, yes |
| 04:52.54 | *** join/#asterisk Octoban (i=paranoid@202.155.92.26) |
| 04:53.09 | pigpen | remind me what that is called... |
| 04:53.45 | pigpen | ...and I guess since I can do it with a digium card, I should be able to do it with the redphone product. |
| 04:54.05 | pigpen | I thought it was like T1 bridging or something... |
| 04:54.22 | shido6 | how CLOSE is the redphone ? |
| 04:54.30 | shido6 | same switch? |
| 04:54.38 | pigpen | 2 feet. |
| 04:54.40 | pigpen | :P |
| 04:54.58 | pigpen | ethernet would be probably a crossover |
| 04:55.07 | pigpen | or direct connect anyway. |
| 04:55.25 | [TK]D-Fender | Redfone = waste, just buy a T1 card and be done with it |
| 04:55.47 | pigpen | really...I thought it sounded cool to allow for redundant asterisk boxes. |
| 04:55.52 | pigpen | but I have never used one. |
| 04:57.33 | [TK]D-Fender | pigpen, Yeah, and TDMoE = an * only entity. |
| 04:57.50 | aptura | darkpixel ever work with att? |
| 04:58.02 | pigpen | but I agree, the digiums just work....never used sangoma |
| 04:58.20 | aptura | pigpen lucky you ;) |
| 04:58.31 | [TK]D-Fender | Sangoma = great. 0 echo, 0 PCI issues. |
| 04:58.33 | pigpen | I would feel that using sangoma would be kinda biting the hand that feeds me. |
| 04:58.37 | pigpen | really? |
| 04:58.47 | [TK]D-Fender | pigpen, Yup. |
| 04:58.47 | pigpen | so better than digium? |
| 04:58.50 | aptura | yea I year nothing but 100% good things with sangoma. |
| 04:58.56 | JT | pigpen: no shit :P |
| 04:58.58 | rudholm | [TK]D-Fender: 0 echo even without their hardware echo canceller? |
| 04:59.03 | JT | but anyway |
| 04:59.13 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 04:59.16 | darkpixel | aptura: Not willingly. At my last job, my ISP had several services through ATT, but I was the one and only developer and didn't get much chance to touch the network. |
| 04:59.16 | JT | pigpen: you want to do add drop multiplexing basically |
| 04:59.33 | *** join/#asterisk bbryant (i=Brett@65-182-39-142.cre.bil.biltmorecommunications.net) |
| 04:59.37 | [TK]D-Fender | rudholm, I always buy with the HWEC. Wouldn't know otherwise. Why screw around? Buy it right the first time and you won't spend a lot more time & money fixing it later. |
| 04:59.51 | rudholm | how much $? |
| 04:59.54 | pigpen | k...this config is still a few months off...just thinking about it. |
| 05:00.00 | pigpen | sangoma better priced? |
| 05:00.17 | rudholm | [TK]D-Fender: I understand the TDM800 is going to have a hardware EC soon. |
| 05:00.25 | [TK]D-Fender | pigpen, comparably priced |
| 05:00.33 | pigpen | works for me. |
| 05:00.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:00.55 | rudholm | [TK]D-Fender: that sounds good, although I thought the Sangona HWEC was expensive |
| 05:00.56 | pigpen | who is a good distributor for resellers? |
| 05:01.19 | JT | pigpen: yes, generally slightly cheaper |
| 05:01.58 | JT | rudholm: sangoma is also the only company that makes a zap card that is single PRI with HW EC |
| 05:02.06 | pigpen | k. Thanks for the info.. |
| 05:02.49 | JT | rudholm: the hw ec is no more expensive than digium |
| 05:02.50 | [TK]D-Fender | rudholm, on par with Digium |
| 05:02.50 | JT | i've heard they use identical chipsets for the HW EC side of things anyway |
| 05:02.50 | pigpen | does the hw ec actually work? the digium was a waste and caused dtmf issues on mine. |
| 05:02.50 | rudholm | I just need about 6-8 analog ports but I have echo problems with my current TDM400 |
| 05:03.07 | aptura | Is there a reason why a rj45 jack would deviate from the tia 568b standard because it starts with blue and messed up my t1 termination. |
| 05:03.16 | rudholm | what I really want to do is go to BRI, but I can't. |
| 05:03.37 | JT | rudholm: no hardware? |
| 05:03.40 | [TK]D-Fender | rudholm, all but 1 project of mine that used TDM400/2400 were forcibly replaced with A200d's because of static, noise, and echo problems. All were instantly overjoyed at the difference |
| 05:04.01 | rudholm | JT: there's no support for American BRI, only EuroISDN BRI. |
| 05:04.15 | JT | rudholm: that's what i thought |
| 05:04.30 | rudholm | [TK]D-Fender: yeah, my TDM400 basically sucks. |
| 05:04.40 | JT | aptura: no idea, i don't think it matters if the jack complies with the A or B version |
| 05:04.49 | rudholm | [TK]D-Fender: I ordered a TDM800 yesterday because they have better echo performance. |
| 05:05.14 | aptura | JT the way the jack is wired is strang. example 1 - 5 2-4 ect. not strait though. |
| 05:05.24 | rudholm | [TK]D-Fender: but now you have me thinking maybe I should have ordered a Sangoma |
| 05:05.40 | [TK]D-Fender | rudholm, time will tell... |
| 05:05.43 | rudholm | [TK]D-Fender: Strom_M seems to prefer the Digium hardware though. |
| 05:05.43 | JT | aptura: sorry, be more clear |
| 05:06.02 | rudholm | [TK]D-Fender: I tried a TDM800 in my system and the echo was significantly better |
| 05:06.03 | pigpen | rudholm, I have managed many different digium pri cards and analog cards...if it is done right, it should be fine. |
| 05:06.07 | JT | aptura: you cannot use ethernet crossover stuff with T-carrier and E-carrier circuits |
| 05:06.15 | pigpen | but yes, I do want to try the sangoma. |
| 05:06.21 | [TK]D-Fender | rudholm, there are those for whom they work just fine, and if they are promoted as such, hey, thats their experience. |
| 05:06.33 | rudholm | pigpen: I think my problem is partly the fact that my CO is 26,000 feet away |
| 05:06.38 | JT | pigpen: 5 vs 2 yr warrant is another thing |
| 05:06.47 | pigpen | no shit... |
| 05:06.47 | JT | warranty |
| 05:06.48 | pigpen | dam. |
| 05:06.59 | [TK]D-Fender | rudholm, I can tell you that in all of my Sangoma installs the echo was significantlty GONE. Not better... GONE. |
| 05:07.12 | rudholm | [TK]D-Fender: I don't really care *what* I use, I just don't want my guests to ask me why my phone is all echo-y |
| 05:07.21 | pigpen | ok..I am sold. if they work as good as digium with better warranty...hey..no brainer... |
| 05:07.21 | rudholm | [TK]D-Fender: nice |
| 05:07.28 | pigpen | sweet. |
| 05:07.46 | [TK]D-Fender | rudholm, Right on. When I hear enough reports of trouble free cards from someone else I'll give thema shot. |
| 05:07.54 | JT | on the digium plus side, their analogue boards are avilable in higher density per pci slot space consumed |
| 05:08.03 | pigpen | we modifyed the echo cancel "algorithm"..whatever it is called....it took care of it... |
| 05:08.20 | rudholm | [TK]D-Fender: what's a good low-density Sangoma card model? (I'm trying to parse the taxonomy here on their website) |
| 05:08.29 | [TK]D-Fender | JT : I prefer to think of it as incentive to ditch analog all-together... and Sangoma has higher density boards now :) |
| 05:08.50 | JT | A200, rudholm, 4 ports a board, up to 6 boards piggyback on the one slot |
| 05:08.57 | [TK]D-Fender | rudholm, A200 does 4 ports in 1 PCI space. A400 doubles that |
| 05:08.58 | rudholm | [TK]D-Fender: I'd *love* to ditch analog and go to digital entrance facilities entirely, but that brings me back to my BRI problem :( |
| 05:09.14 | [TK]D-Fender | rudholm, PRI, not BRI :) |
| 05:09.16 | JT | rudholm: 8 ports...., pri? |
| 05:09.20 | aptura | JT ended a t1 extention to some off brand rj45 jack/bisket box and then plug in the cisco router. Did not link up. Checked the jack last nigh on a dmm meter. When doing a continuity check came up with pins 1-5 2-4 4-2 5-3. I asume AMP jacks are strait though. Would have made my install simpler. |
| 05:09.37 | JT | heh |
| 05:09.41 | rudholm | [TK]D-Fender: most of my ports are FXS, I don't need 8 CO lines |
| 05:09.50 | rudholm | JT: two CO lines is plenty for me |
| 05:09.59 | [TK]D-Fender | rudholm, using PCI for FXS ?!?! EWWWW!!! |
| 05:10.17 | [TK]D-Fender | rudholm, expensive, inflexible, and more a PITA |
| 05:10.33 | aptura | JT what jack to you use for t1 extentions |
| 05:10.33 | rudholm | [TK]D-Fender: you prefer ATAs or something else? |
| 05:10.39 | minesh | JT:while running command zttool, error comes "unable to open /dev/zap/ctl: No such file or directory" pl suggest |
| 05:10.42 | JT | aptura: 1-4 2-5 is correct pri crossover cable pinning, so it must be ethernet crossover |
| 05:10.52 | JT | aptura: umm, normal ones |
| 05:10.55 | [TK]D-Fender | rudholm, Definately |
| 05:11.07 | JT | aptura: use straight through cables for a telco connection |
| 05:11.16 | [TK]D-Fender | rudholm, non-dtmf transfer without mucking with yout dialplan. |
| 05:11.21 | rudholm | [TK]D-Fender: the performance seems to be the same |
| 05:11.36 | [TK]D-Fender | rudholm, And all that.... better redundency, remote deployment, and a fraction of the cost. |
| 05:12.00 | rudholm | [TK]D-Fender: this is my house at issue here :) |
| 05:12.02 | [TK]D-Fender | rudholm, PCI FXS = 100% loss for user phones. |
| 05:12.32 | rudholm | [TK]D-Fender: what do you mean? |
| 05:12.32 | aptura | JT I did. I made the cables and just put on the plugs on both ends. worked fine. Im saying the jack is strange. I have not really had a reason to question jacks because thay are color coded but are most of them basicly wired strait though? |
| 05:12.51 | [TK]D-Fender | rudholm, SPA-2102 = $35/port. Do do 3-way calliong, blind/attended transfer, etc you don't need to mess around with your dialplan & features.conf and all that. |
| 05:13.29 | [TK]D-Fender | rudholm, PCI FXS offers nothing to the user and forces everything to terminate directly into the server. ATA's also allow for redundency. |
| 05:13.33 | rudholm | [TK]D-Fender: I don't recall having to do anything special to enable trasnfers |
| 05:13.50 | [TK]D-Fender | rudholm, and for a blind transfer? |
| 05:14.05 | rudholm | [TK]D-Fender: flash, dial, hangup... ? |
| 05:14.28 | pigpen | anyone running * on a soekris 5501 yet? |
| 05:14.42 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 05:14.51 | sumasuma | pigpen: yes 5501 working with * |
| 05:15.10 | aptura | sumasuma what case are you using? |
| 05:15.19 | pigpen | yeah... |
| 05:15.26 | sumasuma | not yet got a case yet, just bare board |
| 05:15.41 | pigpen | yeah..that is what I was thinking. |
| 05:15.52 | pigpen | what card do you have on it? |
| 05:15.54 | sumasuma | works fine with TDM400P & TDM800P |
| 05:15.55 | aptura | find one big enough to accomidate a ups |
| 05:16.51 | sumasuma | aptura, big one ? just only 15W is enough to run the board, no need for very big one |
| 05:16.51 | rudholm | [TK]D-Fender: if I was setting up an office, I would use IP phones. right now I use a mix of PCI FXS and ATAs, but they're all centrally located anyway. I don't install the ATAs at the end locations. |
| 05:16.58 | aptura | sumasuma enough to run the phones and the board. |
| 05:16.59 | pigpen | sumasuma, cool..how many exten's? |
| 05:17.13 | sumasuma | pigpen: i used with 4 works fine |
| 05:17.19 | pigpen | not bad. |
| 05:17.35 | sumasuma | yes |
| 05:17.40 | pigpen | I plan to be sticking T1's on it, so the power load would be less... |
| 05:17.48 | pigpen | for fxs's anyway. |
| 05:17.50 | sumasuma | yes |
| 05:17.59 | JT | aptura: i don't know, it is impossible to tell what the pin config of the jack is without both ends of the jack |
| 05:18.01 | sumasuma | FXS with T1 ? |
| 05:18.15 | pigpen | and the 5501 should be able to handle a pri of calls. |
| 05:18.24 | pigpen | sumasuma, no...with the tdm400 |
| 05:18.36 | sumasuma | oh ok |
| 05:18.45 | pigpen | sorry, I may be rambling... |
| 05:18.45 | sumasuma | 5501 works fine with sangoma too ! |
| 05:18.55 | pigpen | custom image? |
| 05:18.56 | sumasuma | I mean sangoma A102 |
| 05:19.10 | sumasuma | yes custom one |
| 05:19.15 | pigpen | flash or hdd? |
| 05:19.30 | sumasuma | digium PRI also works fine |
| 05:19.33 | sumasuma | I use HDD |
| 05:19.41 | sumasuma | 2.5" |
| 05:19.55 | pigpen | I am going to try to run it from flash...we will see.... |
| 05:19.58 | pigpen | gentoo I am sure... |
| 05:21.02 | sumasuma | cool |
| 05:21.14 | sumasuma | flash, you mean CF ? |
| 05:21.15 | pigpen | I have a gentoo dev in house...gotta love it. |
| 05:21.18 | pigpen | yeah. |
| 05:21.22 | sumasuma | oh ok |
| 05:21.32 | sumasuma | should work fine though |
| 05:21.54 | pigpen | we are already running 500-600 firewalls with vpn and squid/dansguardian using flash... |
| 05:22.01 | pigpen | on the 4801's... |
| 05:22.20 | pigpen | so we just need to add asterisk and the hardware support in it. |
| 05:23.05 | *** part/#asterisk Mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 05:23.59 | sumasuma | cool |
| 05:24.07 | sumasuma | 4801 is not having enough power to drive asterisk |
| 05:24.19 | sumasuma | especially when coming to codecs |
| 05:24.22 | pigpen | yeah..that is what I heard...I didn't even try it. |
| 05:24.35 | aptura | what do you mean to "drive" asterisk? |
| 05:24.42 | pigpen | I figure if I need fxo/fxs's I would just use a channel bank... |
| 05:24.49 | pigpen | proc |
| 05:25.00 | pigpen | 4801 is kinda slow. |
| 05:25.01 | aptura | you mean not enough cycles |
| 05:25.02 | aptura | ? |
| 05:25.03 | sumasuma | aptura, processing power |
| 05:25.05 | aptura | k |
| 05:25.29 | aptura | how many channels have you tested with no transcoding and gsm or ulaw? |
| 05:26.19 | *** join/#asterisk phocus (n=phocus@67.32.20.66) |
| 05:26.26 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 05:26.37 | pigpen | aptura, just fyi, it is marketed for embedded pbx usage....so I hope it can at least handle a pri... |
| 05:26.52 | phocus | hey guys, has anyone used jspphone , i cant seem to get it to work, I have my server behind nat, and i am going nuts |
| 05:27.19 | pigpen | phocus, have a beer...things will get better. |
| 05:27.30 | pigpen | if it doesn't ... have another one. |
| 05:27.45 | pigpen | but sorry. never used it. |
| 05:27.47 | JT | what is jspphone? |
| 05:28.09 | pigpen | JT, thanks....at least I am not the only one. |
| 05:28.12 | phocus | http://www.sjphone.org/ |
| 05:28.16 | aptura | pigpen dont know. I like the idea of embeded but have yet to test it. Read its very reliable and can be rebooted with no fear of corruption. |
| 05:28.21 | phocus | its the first windows based sip phone i found, is there a better one? |
| 05:28.35 | JT | phocus: helps to type it correctly the first time :) |
| 05:28.41 | phocus | yep |
| 05:28.42 | aptura | rebooted from a power off case is what I was refering to. |
| 05:29.02 | pigpen | aptura, yeah...also no moving parts doesn't hurt. |
| 05:29.56 | pigpen | phocus, now I know who the "world leader in softphone production" is... :-P |
| 05:29.58 | aptura | Im just a power concios person. Can see one of these used in a remote area with small solar panel and a wifi mesh setup. |
| 05:30.30 | phocus | pigpen do you know how to configure it by chance |
| 05:30.39 | phocus | all the docs are of a differnt version, i cant get anything to work |
| 05:30.45 | pigpen | this is the first I have heard of it....so I have no clue. |
| 05:30.53 | pigpen | I use idefisk or eyebeam. |
| 05:31.12 | pigpen | but they do have a mac version..that is cool. |
| 05:31.28 | phocus | pigpen i have to give this to a novis windows user, wich eas easeier for her? |
| 05:31.53 | pigpen | well, personally, I don't like the look of it... |
| 05:31.58 | pigpen | idefisk is dam easy. |
| 05:32.03 | pigpen | works real good too. |
| 05:32.12 | pigpen | windows/linux/mac |
| 05:32.20 | pigpen | sip/iax |
| 05:35.21 | pigpen | So have you ever been so consumed working, you realize that you are listening to Christmas music in May? |
| 05:35.43 | aptura | is there a combo ata/wireless 900 mhz phone on the market |
| 05:36.12 | pigpen | I know that panasonic was to come out with one... |
| 05:36.28 | aptura | pigpen I am so often consumed my wife shouts 3 times its dinner time and I go and ask her why she has not made dinner yet ;) |
| 05:37.20 | aptura | Some long documentry on tv that state men suffer from tone deafness when there wives talk to them ;) |
| 05:38.01 | *** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net) |
| 05:38.19 | obnauticus | any free IAX providers anyone knows 9of? |
| 05:40.05 | rudholm | sure, I run a free IAX service for a few friends. |
| 05:40.09 | phocus | pigpen do you have time to render some asstance |
| 05:40.10 | phocus | ? |
| 05:41.00 | obnauticus | rudholm ugh |
| 05:41.05 | obnauticus | Why don't people do that |
| 05:41.07 | obnauticus | like caompanies |
| 05:41.37 | rudholm | well, who do you want to call or get calls from? |
| 05:42.06 | obnauticus | just the US |
| 05:42.12 | obnauticus | I' |
| 05:42.17 | obnauticus | I'm 15 |
| 05:42.18 | obnauticus | just wanna setup a pbx |
| 05:42.34 | rudholm | there are free SIP services |
| 05:42.37 | obnauticus | nothing bandwidth consuming just mainly myself i use free world dialup right now but it has no outbound |
| 05:42.40 | rudholm | like Freeworld Dialup |
| 05:42.44 | obnauticus | I already use that |
| 05:42.49 | rudholm | it has outbound |
| 05:42.52 | obnauticus | I want to configure asterisk and stuff |
| 05:43.06 | Octoban | hi, im new to asterisk, and want to buy cheap fxo card, and i came across this ATCOM AX100P, has anyone heard of it? |
| 05:43.33 | obnauticus | rudholm umm |
| 05:43.36 | obnauticus | FWD supports oubound calling |
| 05:43.39 | obnauticus | ? |
| 05:43.47 | rudholm | sure, you can make calls |
| 05:44.04 | obnauticus | it's not working lol. |
| 05:48.17 | phocus | i keep getting an all circutes are busy now, please try your call gain later,?? what odes that mean? |
| 05:48.22 | phocus | do i not have a port open? |
| 05:50.22 | pigpen | what does everyone think of the "softecho" software sangoma is providing... |
| 05:51.51 | pigpen | cool, sangoma has a ds3 card... |
| 05:51.56 | pigpen | not much to look at...but cool. |
| 05:52.06 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 05:52.26 | JT | pigpen: i think it only does data, afaik |
| 05:52.34 | pigpen | right. |
| 05:53.17 | pigpen | I have a few ds3 customers... |
| 05:53.36 | obnauticus | rudholm why do companies make you pay for a PTSN hookup? |
| 05:53.43 | pigpen | nice to have a linux option over pricey cisco solutions. |
| 05:53.54 | rudholm | obnauticus: because it costs them money to provide it? |
| 05:54.05 | obnauticus | why do you do it for freen then |
| 05:57.29 | obnauticus | is what i mean |
| 05:57.31 | obnauticus | lol. |
| 05:57.32 | rudholm | I do it for free for my friends. |
| 05:57.33 | obnauticus | oh, ok |
| 05:57.34 | obnauticus | it's basically just bandwidth right? |
| 05:57.37 | mosty | no |
| 05:57.43 | obnauticus | then what is it |
| 05:57.43 | rudholm | no, it costs me money to complete PSTN calls for them |
| 05:57.44 | mosty | terminating calls costs more than bandwidth |
| 05:57.44 | rudholm | PSTN connections and calls cost money |
| 05:57.45 | rudholm | per minute |
| 05:57.47 | obnauticus | I can't wait until everyone switches to voip |
| 05:57.50 | rudholm | yeah, that would be nice. |
| 05:57.54 | phocus | I cant wait till i get it working |
| 05:57.55 | obnauticus | this PTSN stuff is retarded. |
| 05:57.57 | JT | perhaps you are |
| 05:57.57 | JT | the pstn is awesome |
| 05:57.58 | obnauticus | JT, no. |
| 05:57.59 | obnauticus | Why use the phone networ if there's an even better one... |
| 05:58.00 | obnauticus | the internet |
| 05:58.00 | obnauticus | network* |
| 05:58.01 | rudholm | obnauticus: the way PSTN pricing is done is complicated |
| 05:58.02 | JT | it's NOT a better network |
| 05:58.02 | JT | yes it can be cheaper |
| 05:58.03 | JT | not more reliable |
| 05:58.04 | pigpen | ...I can call my pstn provider and say fix it. |
| 05:58.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:58.16 | pigpen | I call my ISP and say fix it...they laugh. |
| 05:58.17 | pigpen | bastards. |
| 05:58.17 | JT | heh |
| 05:58.18 | phocus | why does it keep telling me "all circuts are busy" |
| 05:58.18 | rudholm | yeah, really the PSTN is very good |
| 05:58.18 | obnauticus | PSTN* |
| 05:58.18 | obnauticus | lol. |
| 05:58.19 | obnauticus | I keep thinking Public Telephone Switch Network |
| 05:58.19 | pigpen | FCC also ensures that the pstn is working for the customer...mostly... |
| 05:59.34 | rudholm | I'm trying to get my employer to open up a public SIP gateway so that people can call them via SIP, rather than having to use the PSTN. |
| 05:59.35 | JT | it shows a gross lack of technical understanding when people saying "the Internet is a better network than the PSTN" |
| 05:59.37 | obnauticus | well JT it is more reliable |
| 05:59.40 | mosty | rudholm, it costs your employer money to do that though, they don't save any money |
| 05:59.40 | obnauticus | I know that. |
| 05:59.41 | pigpen | rudholm, setup an irc server..it is easier...and you don't have to use the phone. |
| 05:59.41 | JT | obnauticus: the pstn is, yes |
| 05:59.42 | obnauticus | That's about it. |
| 05:59.42 | obnauticus | From what I know. |
| 05:59.42 | aptura | JT I agree on that front. |
| 05:59.42 | rudholm | yeah, or "the internet is going to put the phone companies out of business" --as if some entity other than "the phone companies" are providing all that bandwidth... |
| 05:59.43 | JT | obnauticus: quality too |
| 05:59.43 | aptura | will never happen |
| 05:59.44 | obnauticus | Depends, doesn't it? |
| 05:59.44 | JT | obnauticus: the telcos put in the infrastructure for it |
| 05:59.44 | rudholm | pigpen: we have an IRC server |
| 05:59.44 | obnauticus | dunno |
| 05:59.45 | JT | what you do know about? |
| 05:59.45 | aptura | Most people will never go voip because it is not 99.9999% reliable. |
| 05:59.45 | pigpen | there you go..it is paid for! :) |
| 06:00.06 | rudholm | mosty: there might be some savings in eventual reduction of PRI usage |
| 06:00.06 | obnauticus | I'll go VoIP because it's 100% cheaper |
| 06:00.07 | rudholm | mosty: some organizations are already doing it |
| 06:00.10 | obnauticus | Well if EVERYONe were to switch to VoIP it would force better reliability |
| 06:00.11 | obnauticus | lol. |
| 06:00.11 | JT | i prefer my phones to work, so i won't go 100% voip |
| 06:00.16 | JT | lol, good argument |
| 06:00.26 | obnauticus | just like almost everyone isn't using VoIP |
| 06:00.28 | JT | umm, there's very little accountability with VoIP over Internet |
| 06:00.36 | obnauticus | People obviously need 911 and shit |
| 06:00.38 | aptura | JT my wife switched to shaw digital. Thay own the cable system. Lagely its been good with little issues. |
| 06:00.40 | JT | the Internet is run by many different interests |
| 06:00.44 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 06:00.48 | rudholm | JT: there's some validity to that argument. my employer uses VoIP PBXes and they had to totally harden the IP network to do it. |
| 06:00.50 | JT | aptura: i said VoIPoI |
| 06:01.24 | JT | also, the Internet is mostly run by noobs, so it's not the most reliable of networks |
| 06:01.46 | obnauticus | Well |
| 06:01.46 | obnauticus | It's because most people use the PSTN that it is so reliable. |
| 06:01.58 | obnauticus | So if everyone were to switch to VoIP it is likely the quality and reliablity would go up also. |
| 06:02.05 | pigpen | rudholm, the network must have been pretty screwed if they had to go to that extent. |
| 06:02.08 | obnauticus | because of pissed off people. |
| 06:02.17 | JT | no, it's because telcos spend a lot of money on building and designing it properly |
| 06:02.22 | JT | telco stuff is engineered |
| 06:02.29 | JT | IT stuff is not usually carrier grade |
| 06:02.29 | obnauticus | k |
| 06:02.42 | obnauticus | ya, that's probably why |
| 06:02.42 | obnauticus | lol. |
| 06:02.57 | JT | you can't guarantee the Internet unless you run all the relevant sections |
| 06:04.27 | pigpen | yeah..I guarantee my internet right up to my firewall. |
| 06:04.33 | rudholm | pigpen: no, it wasn't. but they had to put UPSes on all network gear, support 802.1q VLANs, 802.3 Power Over Ethernet, and automatic port trunking/VLAN selecting. |
| 06:04.42 | obnauticus | pigpen i wish i could say the same |
| 06:04.52 | obnauticus | I rely on comcast to provide excellent uptime |
| 06:04.56 | obnauticus | which is a pain in the ass |
| 06:05.55 | pigpen | rudholm, ah...yeah...ups's are a good thing. We always try to deploy completely seperate wiring.....try anyway. |
| 06:06.45 | rudholm | pigpen: the phone on my desk is an Avaya H.323 (I think it does SIP also) phone |
| 06:07.21 | obnauticus | rudholm setting up a working PBX and shit with outbound calling and stuff |
| 06:07.23 | obnauticus | is that hard |
| 06:07.25 | obnauticus | for a noob |
| 06:07.30 | obnauticus | I'm a phone noob... |
| 06:07.46 | rudholm | obnauticus: how smart are you? :) |
| 06:07.53 | obnauticus | I'm pretty good at networking |
| 06:07.55 | obnauticus | I run web server |
| 06:07.57 | obnauticus | s |
| 06:07.58 | obnauticus | lol. |
| 06:07.58 | pigpen | rudholm, the phone on my desk is a dixie cup with a string. It also holds water. Sometimes orange juice. |
| 06:08.00 | obnauticus | and mail servers |
| 06:08.04 | obnauticus | on FreeBSD |
| 06:08.05 | obnauticus | and Gentoo |
| 06:08.08 | JT | loling doesn't help that much |
| 06:08.13 | pigpen | AH...not Gentoo! |
| 06:08.18 | obnauticus | I lol all the time it's a habbit I must break. |
| 06:08.26 | obnauticus | pigpen i have it partitioned on here |
| 06:08.29 | obnauticus | i don't run my serers on Gentoo |
| 06:08.32 | obnauticus | i run them on FreeBSD |
| 06:08.36 | obnauticus | and one on Debian |
| 06:08.39 | obnauticus | my firewall is on m0n0wall |
| 06:08.42 | rudholm | yeah, I wouldn't recommend Gentoo for anything other than the desktop |
| 06:08.50 | obnauticus | I rarely use it |
| 06:08.58 | obnauticus | Most of the time I boot into server 2k3 for stuff |
| 06:09.05 | obnauticus | I totally underestimaged 2K3 |
| 06:09.12 | rudholm | the "developers" are too interested in New and Shiny, and not interested enough in Stability and Predictability. |
| 06:09.32 | obnauticus | well Gentoo-Hardened is pretty secure |
| 06:09.34 | obnauticus | so |
| 06:09.34 | pigpen | Well, people who don't understand Gentoo typically don't use it much. |
| 06:09.37 | obnauticus | Debian is pretty good too |
| 06:09.43 | obnauticus | naw I understand it |
| 06:09.47 | obnauticus | I used to use it ALL THE TIME |
| 06:09.53 | pigpen | not for servers. |
| 06:09.58 | obnauticus | hell no |
| 06:10.01 | rudholm | pigpen: yeah, and people who don't understand it criticize it a lot |
| 06:10.18 | JT | debian is fine for servers |
| 06:10.47 | pigpen | Well, I will put it this way: I have a dev of the gentoo hardened sources as a business partner....do the math. |
| 06:10.50 | rudholm | Debian, FreeBSD, RHEL, Solaris, HP-UX, AIX... all make good servers. |
| 06:10.57 | obnauticus | I love freebsd |
| 06:10.58 | obnauticus | I started on it |
| 06:11.02 | obnauticus | don't plan to quit on it |
| 06:11.33 | rudholm | pigpen: my team manages about 40,000 servers and 6000 CIDR blocs --do the math. |
| 06:11.36 | rudholm | :) |
| 06:11.56 | obnauticus | 6000 CIDR chunks |
| 06:11.58 | obnauticus | 32 bit? |
| 06:11.59 | obnauticus | rofl |
| 06:12.13 | florz | .o( with all the servers behind PAT ) |
| 06:12.25 | pigpen | my team is contracted to come into placed like yours and fix the stuff that on staff people can't fix. |
| 06:12.30 | pigpen | :) |
| 06:12.35 | rudholm | uh huh |
| 06:12.56 | pigpen | Oh..and I am not kidding. |
| 06:13.12 | obnauticus | pigpen lucky |
| 06:13.18 | obnauticus | i wish I could work with hardware like mine everywhere |
| 06:13.25 | rudholm | I work for a really big internet company with lots and lots of people who really know their shit, I highly doubt it. |
| 06:13.27 | obnauticus | lol. |
| 06:13.48 | obnauticus | I'm 15 and I got a summer inturnship at HP and IBM |
| 06:13.51 | obnauticus | so I think I'm good. |
| 06:13.54 | rudholm | cool |
| 06:13.59 | rudholm | then you should have no problem with Asterisk |
| 06:14.17 | Strom_M | just learn to spell internship and you're good to go ;) |
| 06:14.21 | obnauticus | Ya... |
| 06:14.24 | obnauticus | You don't need to spell to network. |
| 06:14.25 | rudholm | haha |
| 06:14.25 | pigpen | Yeah..sounds like you have "big business attitude" |
| 06:14.34 | rudholm | ?? |
| 06:14.39 | obnauticus | most of the time |
| 06:14.40 | obnauticus | rofl. |
| 06:14.46 | rudholm | pigpen: as evidenced by what? |
| 06:14.47 | obnauticus | in which cases google comes in handy |
| 06:14.59 | Strom_M | google is not a dictionary |
| 06:15.02 | rudholm | pigpen: my attitude toward Gentoo? (I'm on a Gentoo box right now, fwiw) |
| 06:15.08 | Strom_M | there's dictionary.yahoo.com for that ;) |
| 06:15.14 | obnauticus | yup yup |
| 06:15.39 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:15.46 | obnauticus | At least I don't abbreviate words and speak in complete sentences (allthough they are spread throughout a few lines).... unlike most people my age. |
| 06:15.48 | rudholm | obnauticus: you don't need to spell to network, but you need to spell to be taken seriously as a professional. |
| 06:16.02 | obnauticus | Ya, I'm still in school. |
| 06:16.12 | obnauticus | I wanna goto MIT, and it's looking good I got a 4.0GPA |
| 06:16.25 | obnauticus | actually it's a 4.01 atm |
| 06:16.33 | rudholm | not 5.0? bah! |
| 06:16.38 | rudholm | slacker! |
| 06:16.43 | obnauticus | we have a 4 point grading scale where I live. |
| 06:16.55 | rudholm | then how is it 4.01? |
| 06:17.00 | rudholm | rounding error? :) |
| 06:17.00 | obnauticus | Extra credit |
| 06:17.03 | rudholm | ah |
| 06:17.08 | obnauticus | 102% in math |
| 06:17.09 | obnauticus | cal 1 |
| 06:17.21 | obnauticus | brb |
| 06:18.58 | obnauticus | back |
| 06:19.46 | rudholm | have you had DEs yet? |
| 06:19.57 | obnauticus | We're covering Derivatives |
| 06:20.21 | rudholm | cool |
| 06:20.40 | rudholm | I was in a meeting the other day, talking about monitoring network performance |
| 06:20.48 | rudholm | looking at interface error counts |
| 06:21.10 | obnauticus | and? |
| 06:21.15 | rudholm | I pointed out that we should be looking at the first or second order derivatives rather than just have error count thresholds :) |
| 06:21.26 | obnauticus | Excel for the win |
| 06:21.47 | rudholm | Matlab, please... |
| 06:21.57 | obnauticus | I would much wrather have numbers than graphs while looking at network performance |
| 06:22.25 | obnauticus | But I guess with a mass amount of hosts |
| 06:22.27 | obnauticus | it would help |
| 06:22.30 | rudholm | yeah, there'd be no graphing involved |
| 06:22.48 | obnauticus | it was kinda a random remark |
| 06:22.51 | obnauticus | anyway ugh |
| 06:23.01 | mosty | is there a way to make a one-way call? i want to be able to call all the phones in the office and have them act as intercoms, eg for emergency warnings or unclaimed pizzas at reception. anyone know a way to do that? |
| 06:23.17 | rudholm | Strom_M: where's that new Asterisk book you were gonna write? |
| 06:23.22 | rudholm | obnauticus needs it. |
| 06:23.27 | obnauticus | lol |
| 06:23.31 | obnauticus | unclaimed pizzas? |
| 06:23.35 | mosty | i can call a single phone like this already, but i don't want an office full of people speaking on the line too |
| 06:23.36 | pigpen | mosty, check out allpage |
| 06:23.36 | obnauticus | mosty can you send me that call? |
| 06:23.40 | Strom_M | oh, it's all in my head |
| 06:23.40 | rudholm | no really, he's gonna write one |
| 06:23.51 | obnauticus | Strom_M when it's done |
| 06:23.58 | obnauticus | make sure you include a step by ste |
| 06:23.59 | obnauticus | step* |
| 06:24.00 | mosty | pigpen, thanks |
| 06:24.01 | obnauticus | lol. |
| 06:24.12 | rudholm | and I'm going to edit the parts on Shannon and Nyquist, so it's nice and pedantic :) |
| 06:24.14 | obnauticus | I learn best with step by steps explaining what to do there and stuff |
| 06:24.23 | obnauticus | Strom_M admin@aaopwner.com when you are done with it lol. |
| 06:24.26 | *** join/#asterisk CyberMad (n=jack@222.124.69.180) |
| 06:24.36 | JT | ~thebook |
| 06:25.01 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:25.01 | CyberMad | does asterisk support password protection? so when user want to make call outside.. must hit 9 then password, after that can make a call.. i want to log the usage too based on that password, because each employee have 1 password |
| 06:25.13 | Strom_M | CyberMad: you can write something like that into your dialplan |
| 06:25.20 | rudholm | thebook is now quite out of date, though. |
| 06:25.27 | JT | CyberMad: yes, using DISA or the auth app |
| 06:25.31 | Strom_M | rudholm: a second edition of thebook is on its way shortly |
| 06:25.42 | rudholm | same authors? |
| 06:25.45 | CyberMad | Strom_M is there tutorial of it on internet? do you have any link of it? |
| 06:25.45 | JT | the book is fine for learning about most of the fundamentals |
| 06:25.48 | obnauticus | ugh |
| 06:25.59 | obnauticus | Comcast needs some fucking real ddos protection |
| 06:26.08 | obnauticus | I thought cisco put that shit into their CMTS's now a days |
| 06:26.13 | rudholm | JT: yeah, it is, but it'd be really nice to have something current (and now with 1.4, a lot of the syntax won't work) |
| 06:26.23 | CyberMad | well.. uncle google i think is the answer.. but if someone here have great web link of it, that would be nice :) |
| 06:26.25 | JT | heh |
| 06:26.38 | JT | CyberMad: just look it up on the wiki |
| 06:26.39 | Strom_M | CyberMad: we already told you how |
| 06:26.47 | CyberMad | JT thanks :) |
| 06:26.58 | CyberMad | Strom_M thanks.. for the keyword.. it's ok |
| 06:27.38 | obnauticus | anyone here have a good up-to-date guide of Asterisk on FreeBSD |
| 06:27.51 | mosty | pigpen, the link to allpage.agi 404's :( would you happen to have a copy? |
| 06:27.52 | JT | what keyword? |
| 06:28.19 | pigpen | what ver of * are you running? |
| 06:28.40 | obnauticus | oh.. |
| 06:28.42 | obnauticus | i dunno. |
| 06:28.44 | obnauticus | I wanna install it |
| 06:28.47 | obnauticus | i'm fairly new to this shiz |
| 06:29.35 | pigpen | mosty, what ver of * are you running? |
| 06:30.09 | obnauticus | I haven't even installed it yet |
| 06:30.19 | obnauticus | the last time i tried the freebsd support for asterisk was shiz |
| 06:30.20 | mosty | pigpen, 1.2.something |
| 06:30.21 | CyberMad | JT using DISA or the auth app << that's the keyword.. ;) |
| 06:30.29 | JT | CyberMad: right |
| 06:33.24 | pigpen | mosty, on 1.2.x I was using allpage.agi....on 1.4.x I am using the Page app with SIPAddHeader |
| 06:33.36 | pigpen | in 1.4.x it makes much more sense. |
| 06:34.17 | *** join/#asterisk wunderkin- (n=wunderki@dslstat-ppp-95.fastq.com) |
| 06:34.38 | mosty | pigpen, hmm ok, we cannot upgrade to 1.4 yet, i will try to find another source for that allpage.agi |
| 06:35.10 | pigpen | http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom |
| 06:35.25 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:35.34 | pigpen | ah..the aussievoip.com is dead. |
| 06:35.47 | mosty | yes |
| 06:36.02 | pigpen | hmm..here it says that 1.2 supports the Page command... |
| 06:38.01 | JT | aussievoip works fine |
| 06:38.55 | pigpen | the one site on it that references the allpage.agi. |
| 06:39.50 | *** join/#asterisk KaiHanari (n=kai@CPE0013a3bd89d2-CM0011e6c7e1cf.cpe.net.cable.rogers.com) |
| 06:40.32 | _charly_ | does chan_oh323 interpret the inband informations like user busy? |
| 06:42.27 | pigpen | rudholm, how many datacenters? |
| 06:43.01 | rudholm | lots. I don't know off the top of my head. |
| 06:43.14 | pigpen | Rackspace maybe? |
| 06:43.19 | rudholm | huh?? |
| 06:43.27 | pigpen | ie: do you work for Rackspace? |
| 06:43.27 | rudholm | you're kidding, right? |
| 06:43.30 | rudholm | no |
| 06:43.35 | pigpen | just guessing. |
| 06:43.38 | rudholm | ah |
| 06:43.52 | JT | he said he works for an isp, not a dedihost provider |
| 06:43.59 | obnauticus | lucky :( |
| 06:43.59 | pigpen | ha... |
| 06:44.01 | rudholm | I said "internet company" |
| 06:45.36 | obnauticus | rudholm how fast? |
| 06:45.37 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:45.38 | JT | oh |
| 06:45.38 | obnauticus | Gimme like |
| 06:45.38 | rudholm | obnauticus: ho fast what? |
| 06:45.38 | rudholm | s/ho/how/ |
| 06:45.38 | *** join/#asterisk Rahail (i=Oh-Ya@12.191.5.194) |
| 06:45.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:45.38 | obnauticus | Your average like.. |
| 06:45.39 | obnauticus | coustomer |
| 06:45.39 | obnauticus | buys this speed xxx |
| 06:45.39 | obnauticus | xxx = ? |
| 06:45.39 | rudholm | not an ISP |
| 06:45.39 | obnauticus | oh |
| 06:45.39 | rudholm | sometimes I wish we were, I can't get decent connectivity at my house |
| 06:45.39 | rudholm | argh |
| 06:45.39 | obnauticus | I do. |
| 06:45.39 | obnauticus | I got 30/2 |
| 06:45.49 | rudholm | that's nice, how well does it perform during peak hours? |
| 06:45.51 | pigpen | Ok..not an ISP, not a Hosting provider...maybe telco? |
| 06:45.55 | rudholm | my cable modem starts to suck at night |
| 06:45.56 | obnauticus | How fast is the bandwidth does he provide |
| 06:46.03 | JT | jbot: go back to sleep |
| 06:46.24 | rudholm | 10/1 |
| 06:46.24 | obnauticus | rudholm during peak hours it's like 23/1.5 |
| 06:46.24 | obnauticus | and im on cable rudholm |
| 06:46.25 | obnauticus | what isp rudholm? |
| 06:46.25 | JT | he said not an isp |
| 06:46.25 | rudholm | my cable provider is Charter Communications |
| 06:46.26 | obnauticus | oh |
| 06:46.30 | obnauticus | you must do the fun stuff |
| 06:46.31 | obnauticus | like me |
| 06:46.34 | rudholm | I used to live in a FIOS area :( |
| 06:46.39 | obnauticus | Well |
| 06:46.41 | obnauticus | fios isn't haxable |
| 06:46.45 | obnauticus | like cable |
| 06:46.49 | rudholm | haxable? |
| 06:46.52 | obnauticus | how's the latency on FIOS |
| 06:46.56 | obnauticus | herd of modem uncapping? |
| 06:47.00 | rudholm | are you talking about uncapping? |
| 06:47.32 | obnauticus | Definetally not. |
| 06:47.40 | rudholm | yeah, well, uncapping won't help you if your head end is suffering from aggregation problems. |
| 06:47.57 | rudholm | FIOS latency is about what you'd expect |
| 06:48.00 | obnauticus | i reside in a domestic zone.. |
| 06:48.02 | obnauticus | good? |
| 06:48.04 | obnauticus | T1 good? |
| 06:48.09 | rudholm | single-digit milliseconds to the default gateway typically |
| 06:48.20 | obnauticus | I'd hope so |
| 06:48.24 | obnauticus | It's fiber |
| 06:48.33 | obnauticus | I can |
| 06:48.33 | rudholm | what's that got to do with it? |
| 06:48.35 | rudholm | C is C man |
| 06:48.41 | obnauticus | dunno |
| 06:48.43 | obnauticus | i'm tired |
| 06:48.48 | obnauticus | umm what's that |
| 06:48.56 | obnauticus | I can't wait for FiOS to get near me |
| 06:48.57 | rudholm | the speed of light in a vacuum. |
| 06:49.04 | obnauticus | is the fastest... |
| 06:49.05 | obnauticus | apperentally |
| 06:49.07 | KaiHanari | whats a good tutorial on how to create an ivr? |
| 06:49.18 | obnauticus | well rudholm umm what's that we got FiOS in portland, OR |
| 06:49.22 | obnauticus | which is right across from me |
| 06:49.30 | rudholm | cool |
| 06:49.41 | rudholm | we have it in parts of L.A. |
| 06:49.42 | pigpen | shit, I want a tutorial to pass data over light in a vacuum. |
| 06:49.47 | JT | obnauticus: so what do you mean about hackable? |
| 06:49.50 | pigpen | I got a dyson vac... |
| 06:49.53 | obnauticus | JT cable is hackable |
| 06:50.00 | *** join/#asterisk olinux (n=olinux@ip68-107-12-15.sd.sd.cox.net) |
| 06:50.00 | rudholm | haha |
| 06:50.02 | obnauticus | you can get free service and etc, but i won't talk about it here... |
| 06:50.02 | JT | obnauticus: please be more specific |
| 06:50.10 | obnauticus | i won't elaborate.. |
| 06:50.17 | KaiHanari | nvm. found one |
| 06:50.21 | rudholm | it's hard to get free service when they disconnect your drop line |
| 06:50.21 | JT | obnauticus: capping or surveillence? |
| 06:50.25 | JT | or else? |
| 06:50.26 | obnauticus | capping |
| 06:50.32 | obnauticus | rudholm well.. |
| 06:50.38 | olinux | trying to use x-lite on fedora 6 and i do not have sound, any ideas? |
| 06:50.46 | obnauticus | on DOCSIS 2 it's actually hard to get found |
| 06:50.51 | obnauticus | from what I understand |
| 06:50.58 | obnauticus | I haven't herd of one case where someone got caught |
| 06:51.14 | rudholm | I've heard of some. not sure which version of DOCSIS was involved, though. |
| 06:51.15 | olinux | logs, Warning: /dev/dsp appears to be a valid audio device, but I cannot open it. Please ensurusing the audio device (perhaps by trying ``lsof /dev/dsp'').e that no other applications are |
| 06:51.21 | obnauticus | probably 3 |
| 06:51.41 | obnauticus | from what i've seen you have to like TRY to get caught |
| 06:51.49 | obnauticus | and im on a public forum hosting config files over tftp |
| 06:51.51 | obnauticus | with my ip on there |
| 06:51.55 | obnauticus | and i haven't gotten a call or a high bill |
| 06:51.57 | *** join/#asterisk johngalt (n=chatzill@h460773f1.area7.spcsdns.net) |
| 06:51.57 | rudholm | and with the later versions of the spec, it's harder to uncap by spoofing the tftp config server |
| 06:51.58 | obnauticus | so im waiting |
| 06:52.00 | JT | smart |
| 06:52.06 | obnauticus | you can't do that anymore rudholm |
| 06:52.28 | obnauticus | you have to change the address of your tftp server by modifying the firmware of your modem. |
| 06:52.48 | obnauticus | they disallow umm internal tftp spoofing... nobody is sure of what causes it |
| 06:52.49 | rudholm | so you hardcode the address of the config server into the modem? |
| 06:52.55 | obnauticus | no |
| 06:52.58 | obnauticus | you jtag a new firmware on |
| 06:53.16 | obnauticus | that's enough from me |
| 06:53.20 | obnauticus | if you want more you'll have to pm |
| 06:53.31 | rudholm | well, if you can load a new firmware, why do you need a config server at all? couldn't you hardcode the config settings into the firmware? |
| 06:53.34 | obnauticus | I don't want to PUBICALLY make myself look like a retard |
| 06:53.37 | rudholm | hahah |
| 06:53.44 | johngalt | anyone know where to get china did numbers that forward to sip or where to ask? |
| 06:53.53 | obnauticus | rudholm i don't know about that. |
| 06:53.57 | rudholm | "hello, Comcast..." |
| 06:54.13 | obnauticus | lol. |
| 06:54.23 | rudholm | "yeah, there's this guy on 71.59.162.60..." |
| 06:54.33 | obnauticus | rudholm that won't work lol. |
| 06:54.39 | rudholm | hehe |
| 06:54.42 | obnauticus | umm |
| 06:54.47 | obnauticus | if you wanna know pm me |
| 06:54.53 | rudholm | that's ok |
| 06:54.55 | obnauticus | k |
| 06:54.59 | rudholm | I'm not interested in uncapping my modem |
| 06:55.04 | rudholm | that's not the problem |
| 06:55.12 | obnauticus | Well |
| 06:55.17 | obnauticus | I'll be getting FiOS once it gets here |
| 06:55.18 | pigpen | yeah..same here..my isp's upstream sucks. |
| 06:55.24 | obnauticus | ya |
| 06:55.28 | rudholm | I have 10Mb service that doesn't deliver 10Mb because the network is saturated |
| 06:55.31 | rudholm | uncapping won't help me |
| 06:55.36 | obnauticus | rudholm umm |
| 06:55.39 | obnauticus | you're in LA you said.. |
| 06:55.41 | obnauticus | or you WERE in LA |
| 06:55.41 | pigpen | yeah..dam sad. |
| 06:55.48 | rudholm | I am in L.A. |
| 06:55.54 | obnauticus | rudholm complain to your ISP |
| 06:55.56 | obnauticus | like YELL at them |
| 06:56.02 | obnauticus | tell them to provide what they advertise |
| 06:56.02 | obnauticus | lol. |
| 06:56.03 | rudholm | I've spoken to the CTO already. |
| 06:56.14 | obnauticus | they would need to like install new hardware |
| 06:56.16 | obnauticus | actually DO something |
| 06:56.30 | rudholm | they upgraded already. and it made a huge difference, but it's still not what it should be. |
| 06:56.40 | obnauticus | Probably not.. |
| 06:57.03 | obnauticus | what's your cidr chunk at |
| 06:57.03 | obnauticus | on your node |
| 06:57.03 | obnauticus | in LA probably like /16 lol. |
| 06:58.43 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 06:58.58 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 06:59.15 | obnauticus | anyway what i find is pretty stupid is that you can actually download your config for your cable isp at an uncapped rate.. |
| 06:59.32 | obnauticus | so technically everyone uncaps every time they download the config from their isp |
| 07:00.00 | rudholm | well, that's before the cap goes into place |
| 07:00.05 | obnauticus | ya |
| 07:00.25 | obnauticus | wow umm |
| 07:00.31 | obnauticus | the alarm on my watch just rang at 00:00 |
| 07:00.47 | pigpen | 2:00am here. |
| 07:01.53 | olinux | no xlite+linux users? |
| 07:01.54 | *** part/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 07:02.03 | obnauticus | ugh rudholm... |
| 07:02.11 | obnauticus | Opinion on Extreme networks Alpine switches |
| 07:03.13 | obnauticus | I got one and a Catalyst, and you probably have more experience |
| 07:03.51 | *** part/#asterisk minesh (i=minesh@203.88.149.166) |
| 07:05.39 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 07:07.07 | olinux | had to kill artsd |
| 07:14.11 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 07:14.15 | drrt | hello |
| 07:17.28 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 07:18.03 | angryuser | hi |
| 07:19.20 | angryuser | have anything changed in mysql structure from version 1.4.0 > 1.4.4 because iu have a lot of "s" values in destioation field in mysql |
| 07:19.26 | angryuser | ? |
| 07:19.54 | angryuser | do i need to make a new table? |
| 07:20.47 | drrt | i ve such rows in my pgsql cdr base too |
| 07:21.45 | angryuser | it was clean in "dst" field before, no other values but destination number |
| 07:23.20 | angryuser | and sometimes i have nothing in "From" field, same thing in 1.4.0 it was stable and allways filled |
| 07:23.37 | drrt | yep. same thing for me. |
| 07:24.16 | *** join/#asterisk friedrich| (n=friedric@e177253100.adsl.alicedsl.de) |
| 07:24.48 | drrt | do u use macro for dialing ? |
| 07:25.24 | angryuser | yes |
| 07:25.58 | drrt | can u share it ? |
| 07:25.58 | angryuser | going to read changelog, who knows |
| 07:26.51 | angryuser | <drrt> of course give me a sec |
| 07:29.22 | Daejeo1 | anylink for codec g726. i want to install on my box |
| 07:29.45 | mosty | Daejeo1, www.digium.com |
| 07:30.44 | drrt | is it free to use with asterisk ? |
| 07:30.56 | Daejeo1 | mosty: is it free |
| 07:30.58 | Daejeo1 | ? |
| 07:31.33 | mosty | sorry, i misread- did you mean g729? |
| 07:32.56 | JT | g.726 is free |
| 07:32.57 | *** join/#asterisk saftsack (n=oliver@p54A7ED09.dip.t-dialin.net) |
| 07:33.04 | JT | it's adpcm if my memory serves me right |
| 07:33.28 | JT | be good if more equipment like ip phones and ITSPs supported it |
| 07:33.49 | JT | almost identical quality to g.711, half the bandwidth |
| 07:36.36 | Daejeo1 | ? jt |
| 07:36.55 | JT | Daejeo1: sorry, was that a question? |
| 07:36.55 | Daejeo1 | ? JT |
| 07:36.59 | JT | ... |
| 07:37.07 | JT | stop doing that |
| 07:38.24 | Daejeo1 | JT: link for 726 |
| 07:39.30 | JT | i don't have a link |
| 07:42.14 | *** join/#asterisk fujin (i=aj@unaffiliated/fujin) |
| 07:42.50 | angryuser | drrt you still here? |
| 07:43.19 | angryuser | http://www.pastebin.ca/501857 my little macro to call out if it helps |
| 07:43.29 | angryuser | <drrt> |
| 07:43.54 | drrt | <PROTECTED> |
| 07:44.40 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:45.53 | drrt | angryuser, do u ve any channel state like s-NOANSWER in your context ? |
| 07:46.31 | angryuser | <drrt> no dont need then i have a really simple dialplan |
| 07:47.30 | drrt | i tried to use NoCDR to clean such garbage cdrs |
| 07:47.48 | drrt | but ve no success |
| 07:51.12 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:54.33 | angryuser | drrt yea , upset here, orked just fine before |
| 07:58.11 | drrt | angryuser, actually i ve no idea how to debug this |
| 07:58.40 | drrt | angryuser, but i will try to find out something using high level of debug and verbose |
| 08:00.57 | angryuser | <drrt> if you find anything.... |
| 08:02.44 | _charly_ | hmm... asterisk 1.2 with chan_oh323 seems to work best, with 1.2 and chan_h323 i only have one way audio, with asterisk 1.4 i don't have any audio... but chan_oh323 seems not supporting inband busy signalling recognition :( does anyone has some hints for me to get this working, please? |
| 08:02.47 | *** join/#asterisk octoban (i=paranoid@219.83.106.152) |
| 08:04.20 | drrt | angryuser, i ll post it to voip-info.org then |
| 08:04.58 | angryuser | drrt ok |
| 08:05.53 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:06.56 | _charly_ | btw, is chan_oh323 still in development? i guess it's not, is it? |
| 08:08.00 | Strom_M | _charly_: inband? BUSY messages should come out-of-band |
| 08:10.53 | _charly_ | Strom_M: i'm not really sure if it's inband or not, i'm a little confused by h323 :/ the user busy comes in the first progress message from the peer |
| 08:11.00 | *** join/#asterisk blmn (n=blmn@S0106006097940f68.vw.shawcable.net) |
| 08:12.28 | Strom_M | if they other end is sending PROCEEDING and a busy tone, but never sending BUSY or releasing with cause BUSY, then something is quite fucked up on the terminating end of the call |
| 08:13.18 | blmn | is there apt or rpm or ebuild for asterisk 1.4.X anywhere, could not find this anywhere, also whats the official linux distro for asterisk anyone ? |
| 08:13.38 | mosty | blmn: debian unstable has 1.4 |
| 08:13.38 | jm|work | blmn: Debian has 1.4 in apt |
| 08:13.38 | Strom_M | blmn: build it yourself |
| 08:13.44 | Strom_M | also, there is no official distro |
| 08:13.45 | jm|work | mosty: etch, too |
| 08:13.48 | Strom_M | use whatever you like best |
| 08:14.03 | mosty | jm|work, 1.2.13 is in etch, last time i looked |
| 08:14.15 | Strom_M | build it yourself |
| 08:14.25 | jm|work | Connected to Asterisk 1.4.2 currently running on macintel (pid = 13322) |
| 08:14.28 | *** join/#asterisk saftsack (n=oliver@p54A7FAD9.dip.t-dialin.net) |
| 08:14.38 | Strom_M | thats uber old |
| 08:14.38 | jm|work | Linux macintel 2.6.18-4-686 #1 SMP Mon Mar 26 17:17:36 UTC 2007 i686 GNU/Linux |
| 08:14.44 | _charly_ | i have an outgoing setup, then an incoming callProceeding followed by an incoming progress (with cause = user busy), 30 seconds later i'll get an incoming releaseComplete (with cause = normal call clearing). i could send you a tcpdump file if that would help |
| 08:14.49 | jm|work | etch: still 1.4, though :) |
| 08:14.59 | Strom_M | cocks |
| 08:15.10 | jm|work | hmm? |
| 08:15.18 | jm|work | like roosters? |
| 08:15.23 | jm|work | boy-hens? |
| 08:15.23 | Strom_M | like penises |
| 08:15.30 | jm|work | oh, right. |
| 08:15.44 | blmn | jm|work, is it stable enough ( the debian 1.4 unstable ) ? |
| 08:15.45 | jm|work | and how is that relevant? |
| 08:15.54 | jm|work | blmn: I have absolutely no problems |
| 08:16.03 | Strom_M | blmn: please listen to me. build it yourself on debian stable |
| 08:16.09 | jm|work | blmn: the only problems I had were with my X100P which was rubbish |
| 08:16.21 | blmn | jm|work, which distro are you on ? |
| 08:16.30 | jm|work | but I've replaced that with a Linksys 3102 now |
| 08:16.37 | Strom_M | ok fine, dont listen to me |
| 08:16.48 | jm|work | blmn: etch/4.0/stable |
| 08:16.50 | blmn | Strom_M, I dont like to build stuff myself, thats what the package mangement system is for |
| 08:16.56 | Strom_M | boo hoo |
| 08:17.04 | mosty | blmn: apt-get install asterisk |
| 08:17.09 | Strom_M | ./configure; make clean; make install |
| 08:17.14 | Strom_M | SOOOOO MUCH WORK |
| 08:17.40 | blmn | Strom_M, thats not the point, you lose all the version tracking, etc.... |
| 08:17.41 | mosty | managing upgrages of source packages, is not so easy Strom_M |
| 08:17.52 | Strom_M | you break into a sweat somewhere between "clean" and "install" |
| 08:18.18 | Strom_M | and also, whoever packages asterisk for debian is full of crap and puts the files in all the wrong places |
| 08:18.22 | blmn | Strom_M, there is a reason why apt, rpm and other systems exist |
| 08:18.42 | Strom_M | blmn: I use debian exclusively; I know the joy of apt |
| 08:19.47 | mosty | Strom_M, the only thing that bugs me about debian's asterisk packages are the config files, they should be in /usr/share/doc/asterisk/examples/ instead |
| 08:19.47 | Strom_M | but my asterisk box is exclusively my PBX, and it's not exactly a hassle to upgrade to the latest subversion release |
| 08:19.47 | blmn | mosty, which distro are you on ? |
| 08:19.48 | Strom_M | I run one perl script and, catsex, i'm done |
| 08:19.53 | Strom_M | s/perl/bash/ |
| 08:20.05 | mosty | blmn, debian |
| 08:20.20 | Strom_M | _charly_: well, hmm, ive never used h323 to be honest |
| 08:20.25 | Strom_M | but I have a copy of the spec |
| 08:20.31 | jm|work | (asterisk extras don't compile in etch atm) |
| 08:20.46 | jm|work | so if you need mysql connection, you might need to build "by hand" |
| 08:20.52 | Strom_M | NO! |
| 08:20.56 | Strom_M | HE CAN'T DO THAT!!!!! |
| 08:21.07 | Strom_M | he'll lose his VERSION MANAGEMENT DOGBALLS! |
| 08:21.19 | mosty | use postgresql instead, better db, better licenced libs |
| 08:21.23 | jm|work | well, they will build against the apt installed version but you'll have to point it to the right headers |
| 08:21.55 | Strom_M | woot. |
| 08:22.28 | Strom_M | i think coming up with the phrase "version management dogballs" was worth it |
| 08:22.52 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 08:23.16 | _charly_ | Strom_M: ah, ok. i have a trunk to a siemens hipath, i tried with sip before switching to h323, but sip didn't work because the hipath uses sip over tcp, and that's not yet supported by asterisk :/ |
| 08:24.41 | Strom_M | _charly_: http://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.323-200606-I!!PDF-E&type=items |
| 08:25.29 | _charly_ | thanks :) |
| 08:26.37 | Strom_M | good light reading at only 304 pages |
| 08:28.05 | jm|work | something feels a little uncomfortable about running 'the latest SVN' on a OS that gladly calls itself 'unstable' in an enterprise application ... |
| 08:28.08 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:28.12 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:28.30 | Strom_M | jm|work: hence why I tend to run svn checkouts of the release branch |
| 08:28.39 | Strom_M | not the development branch |
| 08:28.46 | Strom_M | and i do it on stable debian |
| 08:30.04 | Strom_M | so, ok, i lose my version management dogballs, but it keeps me and my clients happy |
| 08:33.29 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 08:33.33 | zdrulio | hello all |
| 08:33.36 | Strom_M | hi |
| 08:35.10 | zdrulio | in asterisk1.2v have "sounds" pack, but in asterisk1.4 haven`t ? why ? this sounds are include in aster source or ? |
| 08:35.30 | Strom_M | zdrulio: in 1.4 you select the extra sounds when you run "make menuselect" |
| 08:36.20 | zdrulio | ahm |
| 08:36.23 | zdrulio | ok |
| 08:36.23 | zdrulio | thx |
| 08:37.00 | zdrulio | asterisk-addons ? what is this ? |
| 08:37.49 | Strom_M | extra gibberish you probably dont need |
| 08:38.28 | zdrulio | ok |
| 08:38.39 | punani | zdrulio, where you get your sound pack from btw |
| 08:38.42 | punani | free or ĢĢĢ? |
| 08:39.48 | zdrulio | http://asterisk.org/downloads |
| 08:43.04 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 08:43.19 | ptiggerdine | anyone know if misdn is in the kernel for FC6? |
| 08:46.59 | ghenry | If you're using say Realtime Pg or MySQL, can you still add things to sip.conf and in the database? |
| 08:47.07 | ghenry | so things in sip.conf will still get honoured? |
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| 08:58.51 | *** join/#asterisk matsk (n=mk@194.68.102.171) |
| 09:01.20 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 09:01.49 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-44-193.red.bezeqint.net) |
| 09:01.56 | JT | blmn: new here? |
| 09:02.37 | fourcheeze | ghenry: what sort of things? |
| 09:02.42 | Dovid | if I am having NAT issues with certain phones **COUGH COUGH** Polycom - will a SBC server help ? |
| 09:02.47 | Strom_M | JT: I don't think he liked my version management dogballs |
| 09:02.52 | JT | Strom_M: hehe |
| 09:02.59 | ghenry | fourcheeze: like a sip outbound trunk |
| 09:03.13 | JT | Strom_M: i find public /ignore announced lame and attention seeking |
| 09:03.53 | Strom_M | duh |
| 09:04.32 | fourcheeze | ghenry: any reason you can't have that in your database? |
| 09:04.33 | drrt | does anybody use pickup groups ? |
| 09:04.38 | Strom_M | now i can finally complete my life's work of dialing a phone |
| 09:04.41 | fourcheeze | drrt: yep |
| 09:04.53 | *** join/#asterisk saftsack (n=oliver@p54A7EA2C.dip.t-dialin.net) |
| 09:05.01 | ghenry | fourcheeze: nope, just wondering |
| 09:05.20 | fourcheeze | ghenry: it would be fairly easy to test - add it to your sip.conf then reload sip and see if you have a peer |
| 09:05.26 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:05.33 | ghenry | fourcheeze: thanks |
| 09:05.40 | drrt | fourcheeze, do u use soft or hard phones ? |
| 09:06.11 | fourcheeze | hard |
| 09:07.14 | *** join/#asterisk ^shunz^ (n=SkHate@217.72.32.19) |
| 09:08.20 | drrt | can u describe how to pickup a call using same callgroup? do i need to enter magic sequence *8 at client side? |
| 09:09.35 | fourcheeze | drrt: yep, the phone should just dial *8 normally |
| 09:10.10 | fourcheeze | unless you've overridden *8 or * with something else |
| 09:11.51 | drrt | fourcheeze, and how should i call if i ve no call button? should i just pickup and enter the sequence ? |
| 09:12.07 | drrt | fourcheeze, so stupid questions but i can get it by google ) |
| 09:12.25 | fourcheeze | drrt: which phone are you using? |
| 09:13.15 | angryuser | <drrt> i use picjing groups |
| 09:13.28 | angryuser | *picking |
| 09:14.12 | angryuser | <drrt> i have *8 to pickupgroup assigned, so just enter sequence |
| 09:15.16 | drrt | angryuser, call and pickup groups arent different |
| 09:15.22 | drrt | angryuser, i c |
| 09:15.56 | drrt | fourcheeze, i m using f |
| 09:16.05 | *** join/#asterisk nyaya (n=gangelop@213.5.47.201) |
| 09:16.12 | drrt | fourcheeze, i m using dlink voip gateways at the moment |
| 09:17.27 | fourcheeze | I don't know those. But if you are setup ok it should be just like dialling a normal number |
| 09:17.39 | fourcheeze | make sure that the dlink will call a short number like that |
| 09:17.57 | fourcheeze | in otherwords check its dialplan isn't stopping it |
| 09:18.34 | drrt | fourcheeze, and i cant call them hardphones but they ve no call button as softphones. it confuses me |
| 09:19.18 | drrt | fourcheeze, yeah. i decide to check it using softphone for first |
| 09:20.05 | fourcheeze | but do the phones have a dialplan themselves? |
| 09:20.08 | fourcheeze | or the gateway? |
| 09:21.53 | drrt | fourcheeze, no they are stupid equipment |
| 09:24.01 | fourcheeze | ok |
| 09:24.08 | fourcheeze | never tried them I'm afraid |
| 09:26.53 | drrt | fourcheeze, thx for advice |
| 09:29.06 | *** join/#asterisk sebastian|foo (n=sebastia@61.151.249.123) |
| 09:29.45 | sebastian|foo | hi |
| 09:30.36 | *** join/#asterisk saftsack (n=oliver@p54A7E62C.dip.t-dialin.net) |
| 09:31.56 | sebastian|foo | i've got a single hfcpci card - what is the best version to take? astersik(now) 1.4.x or asterisk 1.2.x? |
| 09:35.02 | sebastian|foo | and which kernel versions work best with misdn? |
| 09:41.09 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
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| 10:22.18 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 10:22.54 | kippi | how can I recored all calls on asterisk, I just need the files dated with times |
| 10:23.48 | phpboy | kippi: read up about monitor() |
| 10:23.53 | phpboy | that should sort you out |
| 10:25.09 | phpboy | example: _0.,1,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) \n _0.,2,Monitor(wav,${CALLFILENAME},m) |
| 10:25.37 | phpboy | the first ofcourse sets ur pref for the filename and the second is the actual recording |
| 10:26.39 | *** join/#asterisk oej (n=olle@65-182-39-213.cre.bil.biltmorecommunications.net) |
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| 10:30.56 | *** join/#asterisk Marquel (n=marquel@e177158179.adsl.alicedsl.de) |
| 10:31.00 | Strom_M | and then you grow up and stop labeling priorities like we did in the dark ages of asterisk 1.0 |
| 10:31.02 | Marquel | morning |
| 10:32.09 | Marquel | is it possible w/ asterisk to receive and record a call and afterwards automatically call some people until one answers? |
| 10:32.24 | Strom_M | very possible |
| 10:32.29 | Strom_M | quite doable, in fact |
| 10:33.30 | Marquel | including sending emails, faxes, using SMS and somehow alert extraterrestrial life support and such things? ;) |
| 10:34.25 | Strom_M | that's a bit trickier |
| 10:34.33 | Strom_M | but i hear they're working on chan_aricebo |
| 10:35.39 | Marquel | well... emails, faxes, sms and extraterrestrial life support are not as important as the automated calls after the recorded call is termintated ;) |
| 10:35.56 | Strom_M | yeah, it's doable |
| 10:36.05 | Marquel | that's great. thanks. |
| 10:36.57 | *** join/#asterisk kvidell (i=id5814@adsl-63-204-157-35.dsl.snfc21.pacbell.net) |
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| 10:40.47 | *** join/#asterisk berktr (n=canberk@teknopet.com) |
| 10:41.04 | berktr | which ports should I forward to my asterisk server for a complete sip communication |
| 10:41.10 | berktr | port 5060 and rtp ports right? |
| 10:41.49 | Strom_M | 5060 UDP |
| 10:41.52 | Strom_M | 10000-20000 RTP |
| 10:42.05 | Strom_M | and then make sure you set externip in sip.conf |
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| 10:53.27 | ghenry | and nat=yes |
| 10:53.36 | Strom_M | yes, that too |
| 10:53.59 | ghenry | ;-) |
| 10:54.08 | drrt | fourcheeze, thx for advice |
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| 11:07.40 | *** join/#asterisk samarora (i=minesh@203.88.149.166) |
| 11:07.54 | samarora | ?? |
| 11:12.37 | *** part/#asterisk samarora (i=minesh@203.88.149.166) |
| 11:16.43 | *** join/#asterisk Morlac (n=Miranda@217.23.37.40) |
| 11:18.03 | Morlac | any one has good experience with ATAs? I am looking for a good quality ATA with 2 FXS and possible 1 FXO... I looked at GrandStream HD-488 but do not know how stable it is. |
| 11:23.25 | zdrulio | i install asterisk but if i reboot, i must start asterisk . how can i start it automaticly ? |
| 11:23.31 | mosty | Morlac, the general experience is that grandstream is total crap |
| 11:23.46 | mosty | zdrulio: depends on your linux dist, man init |
| 11:26.24 | *** join/#asterisk Hypn0tek (n=Hypn0tek@196.203.247.132) |
| 11:27.16 | *** join/#asterisk saftsack (n=oliver@p54a7f648.dip.t-dialin.net) |
| 11:28.18 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 11:30.58 | Morlac | mosty: thats what worried me...I need those for a customer and I dont want to disappoint him... any recommendations on good ATAs? |
| 11:31.20 | mosty | i guess linksys maybe |
| 11:32.08 | Morlac | linksys? they bought sipura out right? |
| 11:32.12 | Hypn0tek|Away | Morlac: I agree linksys |
| 11:33.00 | cpm | I like the digium, but it's very limited, and too spendy |
| 11:33.16 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 11:33.24 | Morlac | ok, you see, am going to buy about 50 of them... anything better, if there is any? |
| 11:34.40 | cpm | you should buy at least one of anything you are intending to roll out, and test. |
| 11:34.54 | cpm | steer clear of the x100p things you see on ebay. |
| 11:35.31 | Morlac | yah, know about them....my other options was either astribanks or sangoma a400 |
| 11:35.32 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 11:35.44 | Morlac | but cost is important, thats why I was thinking ATA's |
| 11:35.47 | mosty | Morlac, this for a business? i'd buy real voip phones |
| 11:36.41 | jeremy_g | hi |
| 11:36.52 | Morlac | Mosty: yes, 3 to 4 sites with 50 users avarage and 8 external lines at each location...replacing their Samsung based PBX.. I wanted to use the most of the current infrastructure |
| 11:36.53 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 11:37.18 | jeremy_g | Morlac:is this the channel for the italian movie asterisk fans |
| 11:37.58 | jeremy_g | why lots of telephones here |
| 11:38.18 | Morlac | jerem_g: it is Asterisk support and discussion |
| 11:38.44 | jeremy_g | Morlac:is a PBX some kinda above block buster |
| 11:38.48 | mosty | Morlac, buy one or two first and test them |
| 11:39.30 | jeremy_g | asterisk 2 is much better, it has justin stevens |
| 11:39.33 | cpm | or channel banks |
| 11:40.00 | Morlac | I agree.... I researched all those....but worried about the price difference |
| 11:40.20 | Morlac | VoIP phones are not cheap around here....snom 300's are like 200$ here |
| 11:41.09 | jeremy_g | why do you buy a voip phone, it doesnt work outside your home or office |
| 11:41.29 | cpm | an ata + power supply, plus decent desk phone, plus whatever, might be a wash |
| 11:42.00 | Morlac | jeremy: I connect VoIP phones to Asterisk and let asterisk handle outgoing trunks |
| 11:42.16 | Hypn0tek|Away | Morlac: I'm facing the same problem for a business, and I'm asking you if u'll do it with one asterisk server or u'll link many ? |
| 11:42.47 | jeremy_g | Morlac:but asterisk is the name of a movie, how do you connect a phone to a movie |
| 11:42.53 | Morlac | Ill link many, spanning multiple countries |
| 11:43.21 | Morlac | jeremy: Asterisk is not a movie, it is a PBX |
| 11:43.28 | Hypn0tek|Away | Morlac: each site with one server ? |
| 11:43.33 | Morlac | yes |
| 11:44.40 | Hypn0tek|Away | Morlac : in my business case, the have 9 analog telephone lines and they want to setup them in a PBX |
| 11:45.18 | jeremy_g | Morlac:aha so pbx is that box with all the phone wires going into at our office |
| 11:45.21 | Morlac | oh, in my case, each site is already running a PBX... with 8 lines and about 50 extensions... They looking for ways to reduce cost |
| 11:45.27 | djmarin | anyone ever heard of a company named signate who does setup of Asterisk system? |
| 11:45.37 | Hypn0tek|Away | Morlac: I'm thinking about 9 ATAs adapter cause I dont want deguim cards they consume a lot of cpu resources, and then offer Ip phones or ATAs adapters n analog phones |
| 11:46.21 | jeremy_g | Morlac:how many sip calls at maximum this asterisk box can take |
| 11:46.51 | Morlac | thats what I was thinking and trying to find the best solution for.... as far as ATA are concerned, I have been adviced to use Linksys...but I have to get one or two and test them |
| 11:47.11 | Morlac | Jeremy: it depends on your hardware.Theoritically, there is no limit |
| 11:47.32 | jeremy_g | considering a 3Ghz box with 1GB ram |
| 11:47.48 | Morlac | although, asteribanks sounds like a good choice and reason |
| 11:48.16 | *** join/#asterisk nkrasko (n=nkrasko@213.33.238.142) |
| 11:48.51 | Morlac | Jeremy: you have to consider the codecs and if you doing codec translation....also, the feature these users are using...but In my case, I can run 100 calls on that hardware comfortably |
| 11:49.26 | *** join/#asterisk saftsack (n=oliver@p54A7F12A.dip.t-dialin.net) |
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| 11:51.11 | djmarin | . |
| 11:53.01 | Hypn0tek|Away | Morlac: U don't think that we will need an analog Gateway for the 9 or 8 analog lines ? |
| 11:53.28 | Morlac | ofcourse you will need them |
| 11:53.56 | Hypn0tek|Away | Morlac: cause in my case there is no E1/T1 connection so I cant use an E1/T1 card |
| 11:53.57 | Morlac | currently, am evaluating the TDM2400 or Sangoma A200 |
| 11:55.37 | *** join/#asterisk fujin (i=aj@unaffiliated/fujin) |
| 11:58.22 | Hypn0tek|Away | Morlac: in your case u'll need 2 Sangoma A200 cards then each with 4 ports, but in mine ... I have 9 lines .... |
| 11:59.03 | Morlac | Hypn0tek: Actual, only one A200 but with extra remora addins....A200 can go up to 24 ports |
| 11:59.26 | mosty | Hypn0tek|Away, get a sangoma card with hardware echo cancellation |
| 11:59.52 | mosty | sangoma a400 can do 12 ports on a single card |
| 12:00.29 | Hypn0tek|Away | mosty : mmmm, n wich box config will I have to use ?! |
| 12:00.45 | Morlac | yah I know...am also looking at that....trouble with a400 is they are bit lengthy....and might not fit our 2U rackmount casing |
| 12:01.32 | Hypn0tek|Away | mosty : with echo cancellation, it will consume a lot of cpu charge ! |
| 12:01.53 | mosty | Hypn0tek|Away, hardware echo cancellation doesn't use the cpu |
| 12:02.45 | Morlac | Hardware echocan is a must.... |
| 12:03.20 | Hypn0tek|Away | Sangoma Remora A20005D PCI Card 0 FXS / 10 FXO Ports + Echo Cancellation |
| 12:04.02 | Hypn0tek|Away | Morlac: why a must ??! |
| 12:04.45 | Morlac | in my envronment, I have lots of users....If I run echocan on software, I loose some calls sometimes |
| 12:04.46 | Hypn0tek|Away | mosty : what about this A2005D ? |
| 12:05.23 | defswork | is echo cancellation only needed on analog lines? |
| 12:05.46 | Morlac | Hypn0tek: Check the A400 card, the 10 ports will be on the same card...but its a bit long, so, depending on the chasis you will have, A400 can be a better solution |
| 12:05.57 | Morlac | defswork: no, even digital |
| 12:06.13 | Morlac | defswork: there are cases where you dont need any |
| 12:06.18 | redax | hi, |
| 12:06.21 | defswork | Morlac: I have Sangoma E1 card with no echo can and not had problems |
| 12:06.35 | redax | anybody using here Snom 320/360 phone? |
| 12:06.46 | Morlac | redax: I have 20xsnom 320 |
| 12:07.06 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:07.40 | redax | Morlac: cool, do you use the hint (devstates) feature? |
| 12:07.50 | Morlac | redax: no |
| 12:08.03 | mosty | redax, i use snoms |
| 12:08.20 | *** join/#asterisk berktr (n=canberk@teknopet.com) |
| 12:08.28 | berktr | hello friends |
| 12:08.52 | *** join/#asterisk saftsack (n=oliver@p54A7E956.dip.t-dialin.net) |
| 12:08.54 | Hypn0tek|Away | Morlac, in this A400 I'm not seeing the ports ?! there a large parallel likes port |
| 12:08.59 | berktr | so, I forwarded UDP & TCP 5060 and 10000-20000 UDP & TCP, however external users still are not able to log in to my server |
| 12:09.03 | redax | is there a way to display "some event text" immediatly on the snom's? |
| 12:09.05 | berktr | why do you think is this |
| 12:09.08 | Morlac | defswork: which is the same as I have....I only have echo when I call from my snom320 to an external number that is landline and analog |
| 12:09.24 | *** join/#asterisk Divious1 (i=chatzill@65.112.134.160) |
| 12:09.30 | Divious1 | hello everyone |
| 12:09.44 | redax | mosty: do you use the hint feature? |
| 12:09.44 | Morlac | Hypn0tek: yah, it comes with a special cable |
| 12:10.00 | mosty | redax, yes |
| 12:10.04 | Divious1 | I was actually looking for an advice, which one would be considered the best distro for asterisk 1.4? |
| 12:10.14 | berktr | [07-05-22]15:08:59.353 | Info | STUN | "STUN: Requested FW Type discovery using STUN server: 85.105.49.124:3478" | |
| 12:10.14 | berktr | [07-05-22]15:09:03.125 | Info | STUN | "STUN: OK FW Type discovery: Block : 85.105.49.124:3478" | |
| 12:10.14 | berktr | [07-05-22]15:09:03.125 | Info | AbstractPhone | "Receiving notification about firewall IP address: 0.0.0.0, voip always possible: 0" | |
| 12:10.18 | berktr | what does this mean? |
| 12:10.55 | mosty | redax, have you looked at http://www.voip-info.org/wiki-Asterisk+phone+snom ? |
| 12:12.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:12.30 | Divious1 | any recommendation?A distro for asterisk? |
| 12:12.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:12.46 | defswork | Divious1: I use trixbox |
| 12:13.12 | mosty | Divious1, use whichever you know how to admin best |
| 12:13.12 | Morlac | divious: I use rpath |
| 12:13.18 | redax | mosty: basicly it works. just today morning it happened the whole stuff stopped working. after restarting asterisk it worked again |
| 12:13.26 | redax | mosty: did you ever noticed such a thing? |
| 12:13.45 | mosty | redax, asterisk is buggy *shrug* |
| 12:13.45 | Divious1 | k thank you guys i take your word for it |
| 12:14.21 | redax | mosty: shhh... |
| 12:14.38 | berktr | any help for me? |
| 12:15.27 | mosty | berktr, perhaps ask a stun channel/mailing list? |
| 12:16.15 | berktr | i don't want to use stun |
| 12:16.17 | redax | mosty: did you tried the Custom devstates stuff on your snom ? (there's an article on asterisk.org about that) I'd like to use it to display the Nightmode on/off state |
| 12:16.45 | mosty | i've only used blf for line monitoring |
| 12:17.53 | Divious1 | isn't tribox already prebuilt? |
| 12:18.13 | *** join/#asterisk ssokol (n=ssokol@65-182-39-203.cre.bil.biltmorecommunications.net) |
| 12:18.37 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
| 12:23.07 | angryuser | tzafrir here? |
| 12:23.52 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 12:25.32 | *** join/#asterisk friedrich| (n=friedric@e177253110.adsl.alicedsl.de) |
| 12:28.18 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:28.37 | DrukenLPY | ok, it's official... wireless SUCKS ASS! |
| 12:29.10 | [TK]D-Fender | DrukenLPY: What kind/use? |
| 12:29.21 | DrukenLPY | 802.11, personal |
| 12:29.49 | LeddyHM | holy crap |
| 12:29.58 | LeddyHM | isn't it a bit early for you to be on irc tk? |
| 12:30.23 | DrukenLPY | i have two ap's in my little 1800 sq ft house.. and both of them disapear all the time, and 8/10 the don't work worth shit |
| 12:30.31 | [TK]D-Fender | LeddyHM: I'm at work... its too early to be awake, but strangely I've been doing this the past 12 years here :) |
| 12:30.47 | Daejeo1 | any link for g726 codec. I want to install |
| 12:31.00 | [TK]D-Fender | Daejeo1: No link, * has it included |
| 12:31.25 | DrukenLPY | [TK]D-Fender: and voip isn't all great over it either :) |
| 12:31.26 | Daejeo1 | how can I check? |
| 12:31.45 | Daejeo1 | show translation? |
| 12:31.46 | [TK]D-Fender | Daejeo1: "shor translation" "show codecs" |
| 12:31.50 | [TK]D-Fender | show* |
| 12:32.09 | [TK]D-Fender | Daejeo1: "show modules" |
| 12:32.12 | [TK]D-Fender | take your pick |
| 12:34.05 | Daejeo1 | g.726-32 |
| 12:34.15 | Daejeo1 | yes it is installed |
| 12:34.47 | Daejeo1 | how can I inable in sip.conf? |
| 12:35.00 | Daejeo1 | *enable |
| 12:35.13 | [TK]D-Fender | allow=g726 |
| 12:35.33 | *** join/#asterisk bbryant (i=Brett@69-94-197-78.biltmorecomm.com) |
| 12:35.45 | *** join/#asterisk NirS_ (i=Nir@87.68.0.156.cable.012.net.il) |
| 12:36.42 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 12:37.10 | Daejeo1 | what should be the sequence? g729, g726, gsm, ilbc, g723, ulaw, alaw |
| 12:37.14 | fourcheeze | can I change codec in a dialplan? |
| 12:37.23 | fourcheeze | e.g. |
| 12:37.34 | Daejeo1 | lower to higher? |
| 12:37.34 | fourcheeze | I have incoming from a DDI provider |
| 12:37.53 | fourcheeze | I have users who prefer ulaw and others who prefer g729 |
| 12:38.01 | fourcheeze | I don't want to transcode if I don't have to |
| 12:38.27 | fourcheeze | incoming provider can send the call in either g729 or ulaw but I can't choose per number |
| 12:38.41 | fourcheeze | is there a way to renegotiate once the call has arrived? |
| 12:39.01 | fourcheeze | Daejeo1: lower to higher what? |
| 12:39.15 | Daejeo1 | sip.conf |
| 12:39.15 | mosty | fourcheeze, that is done in sip.conf (for sip channels) |
| 12:39.21 | [TK]D-Fender | Daejeo1: depends on your priorities |
| 12:39.41 | fourcheeze | mosty: what do you mean? |
| 12:40.25 | fourcheeze | mosty: I realise how I can choose priorities for codecs |
| 12:40.31 | [TK]D-Fender | fourcheeze: You can choose per peer/user what they are allowed to do. So go do it. (disallow=all , allow=ulaw , etc...) |
| 12:40.41 | fourcheeze | that's not what I'm asking |
| 12:40.57 | mosty | fourcheeze, once the call has entered the dialplan, the codec is already selected |
| 12:41.13 | fourcheeze | sure, I'm wondering if it can be renegotiated |
| 12:41.19 | [TK]D-Fender | fourcheeze: No. |
| 12:41.20 | mosty | as far as i know, it can't |
| 12:41.48 | fourcheeze | is there some other way around this? |
| 12:41.51 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 12:42.16 | fourcheeze | bascially I want the provider to default to ulaw so that people who want high quality can have it |
| 12:42.19 | *** join/#asterisk DirtyD (n=DigiD@ool-18bddad8.dyn.optonline.net) |
| 12:42.23 | DirtyD | Hi.. |
| 12:42.34 | fourcheeze | but if they are calling an end user who wants g729 I would rather they provided that |
| 12:42.39 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 12:42.39 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 12:42.48 | fourcheeze | I don't see how that can be done in sip.conf (or realtime equivalent) |
| 12:43.05 | fourcheeze | but I'd love to know how if its possible |
| 12:43.12 | DirtyD | Anyone know of an IP phone that has a programmable display? I'd like to do a database lookup and display customer information on the display of an ip phone... |
| 12:43.52 | fourcheeze | DirtyD: don't the snoms have a url for idle text? |
| 12:44.21 | *** join/#asterisk saftsack (n=oliver@p54A7DD7E.dip.t-dialin.net) |
| 12:44.27 | [TK]D-Fender | fourcheeze: You can choose per user & per peer. thats it. |
| 12:45.07 | [TK]D-Fender | DirtyD: Only cisco's have an XML push to my awareness, but I'd hate for that to be a reason to buy them. |
| 12:45.20 | [TK]D-Fender | DirtyD: this is something far better done on a PC. |
| 12:45.41 | Qwell | well, it has the xml services button |
| 12:46.21 | fourcheeze | http://snom.com/wiki/index.php/Xmlobjects |
| 12:46.52 | fourcheeze | gotta be something there you can use |
| 12:48.23 | DirtyD | Thanks! I'll check it out. Thanks for the start. |
| 12:49.28 | DirtyD | Oh cool. The phone user can even "login" to the phone? using the snom.. I like that. |
| 12:50.17 | *** join/#asterisk ssokol (n=ssokol@69-94-197-22.biltmorecomm.com) |
| 12:50.37 | *** join/#asterisk jmacz (n=jmacz@201.244.170.3) |
| 12:51.21 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:51.29 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 12:52.07 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
| 12:55.14 | fourcheeze | does anyone find that their users are completely happy with g729? |
| 12:55.43 | deegan | mr roboto? who doesnt like him. |
| 12:55.46 | Qwell | fourcheeze: users are never completely happy |
| 12:55.52 | Qwell | Deeewayne: g729 isn't bad... |
| 12:55.57 | Qwell | erm, deegan |
| 12:56.05 | DrukenLPY | fourcheeze: why don't you want to do translation? |
| 12:56.13 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-97.biltmorecomm.com) |
| 12:56.23 | fourcheeze | just want to save my cpu cycles :-) |
| 12:56.46 | mosty | cpu cycles are cheaper than your time |
| 12:56.48 | fourcheeze | I will do it if I have to |
| 12:56.58 | fourcheeze | that may be true |
| 12:57.13 | fourcheeze | anyone g726 a lot? |
| 12:57.18 | drrt | i dont be dramatic. but you ve too |
| 12:57.19 | drrt | to |
| 12:57.35 | *** join/#asterisk Fieldy (i=1u3GGsMe@gentoo/contributor/Fieldy) |
| 12:58.33 | [TK]D-Fender | fourcheeze: few do. |
| 12:58.59 | *** join/#asterisk Scrumps (n=scrumpy@smurfnet.xs4all.nl) |
| 13:02.10 | *** join/#asterisk oej (n=olle@69-94-197-20.biltmorecomm.com) |
| 13:03.40 | fourcheeze | personally I like GSM but I seem to be alone in that |
| 13:04.46 | fourcheeze | so what do people normally use for incoming? ulaw/alaw or g729? |
| 13:04.57 | Qwell | lpc10 |
| 13:05.24 | fourcheeze | never used that one |
| 13:05.33 | fourcheeze | which clients support it? |
| 13:05.54 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 13:06.07 | mosty | fourcheeze, g711, g729, gsm is pretty much all we use |
| 13:06.12 | [TK]D-Fender | Qwell Styx to the good stuff ;) |
| 13:06.12 | *** join/#asterisk saftsack (n=oliver@p54A7F7D2.dip.t-dialin.net) |
| 13:06.30 | *** join/#asterisk Mavvie (n=edwin@ppp1-208.lns1.syd7.internode.on.net) |
| 13:06.42 | fourcheeze | I wish there was a codec that was just a bit better quality than g729 |
| 13:06.51 | [TK]D-Fender | fourcheeze: G.726 |
| 13:07.12 | fourcheeze | yeah, I'm trying to get this sipura to use it but it doesn't want to |
| 13:08.41 | *** join/#asterisk seele_ (n=seele@dns.tennis.com.co) |
| 13:09.31 | fourcheeze | hmm |
| 13:09.32 | fourcheeze | Capabilities: us - 0x10c (ulaw|alaw|g729), |
| 13:09.51 | fourcheeze | any idea why asterisk would think itself incapable of g726 |
| 13:09.54 | [TK]D-Fender | fourcheeze: welcome to TFB Land, population YOU. |
| 13:10.12 | [TK]D-Fender | fourcheeze: check your allow statements |
| 13:10.21 | fourcheeze | i've got allow=g726 |
| 13:10.22 | seele_ | please help with freepbx reports ... in the report panel show No data found !!! and my Master.csv is full of calls |
| 13:10.36 | Qwell | ~freepbx |
| 13:10.39 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:11.17 | seele_ | jbot, ok thanks |
| 13:11.19 | jbot | Just send money... |
| 13:11.29 | seele_ | jbot, LOL |
| 13:11.31 | jbot | lol is probably stands for Laughs Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
| 13:11.31 | Katty | jbot: i love you |
| 13:11.33 | jbot | You love you? |
| 13:12.00 | fourcheeze | [TK]D-Fender: also "sip show settings" tells me |
| 13:12.01 | fourcheeze | <PROTECTED> |
| 13:12.10 | Katty | Qwell: so we complained to our isp over crappy bandwidth, and took screenshots with a SINGLE laptop connected to the t1. |
| 13:12.11 | [TK]D-Fender | jbot: not in public, this is a family channel! |
| 13:12.13 | jbot | I think you lost me on that one, [TK]D-Fender |
| 13:12.16 | Katty | Qwell: you know what their Resolution was? |
| 13:12.40 | Katty | Qwell: you /obivously/ have viruses running rampant on your network. please install a copy of mcafee or norton and scan every machine. |
| 13:12.43 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 13:12.49 | [TK]D-Fender | seele_: .... |
| 13:12.53 | [TK]D-Fender | ~freepbx |
| 13:12.55 | jbot | well, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:13.04 | Katty | [TK]D-Fender: do you see what i have to deal with?! |
| 13:13.08 | seele_ | [TK]D-Fender, yes |
| 13:13.10 | Katty | [TK]D-Fender: i need to hire ninjas. |
| 13:13.17 | Qwell | assassins |
| 13:13.22 | Katty | those too. |
| 13:13.23 | Qwell | or pirates |
| 13:13.26 | Katty | and some pirates, just for good measure |
| 13:13.29 | Qwell | exactly |
| 13:13.41 | Katty | there's a pirate/ninja resturant in las vegas |
| 13:13.43 | cpm | yaar |
| 13:13.47 | Qwell | eh? |
| 13:13.57 | Katty | one day the waiters all dress up like pirates |
| 13:13.59 | Katty | the next ninjas |
| 13:14.02 | Qwell | umm |
| 13:14.04 | Qwell | okay |
| 13:14.07 | Katty | except for the su chefs |
| 13:14.09 | cpm | howz the food? |
| 13:14.11 | Katty | sue |
| 13:14.14 | Katty | whatever |
| 13:14.17 | Katty | i don't know |
| 13:14.38 | fourcheeze | sous chef |
| 13:14.57 | fourcheeze | sous being the french for "under" |
| 13:15.04 | cpm | what makes sauce special? |
| 13:15.04 | Katty | thanks. |
| 13:15.09 | Katty | weed? |
| 13:15.17 | cpm | indeed! |
| 13:15.19 | Katty | sorry, that's the southern missouri talking |
| 13:15.31 | Katty | in other mews, my boss wants to be a fonality reseller. |
| 13:15.42 | cpm | fonality? |
| 13:15.50 | Katty | they use an old asterisk version |
| 13:15.51 | DirtyD | gar |
| 13:15.53 | Katty | and their own software |
| 13:15.54 | Qwell | Katty: ugh, quit |
| 13:16.01 | Katty | Qwell: it might not be such a bad idea... |
| 13:16.02 | DirtyD | Butt Pirate |
| 13:16.12 | fourcheeze | Katty: come and work for me |
| 13:16.13 | Katty | Qwell: especially considering i can't seem to locate pretty software for asterisk |
| 13:16.18 | cpm | sounds like good way to lose money |
| 13:16.36 | Katty | Qwell: and if linux splodes, the only thing i know to do is wipe it, put all the /src back in and restore config files |
| 13:16.49 | cpm | fonality is probably another outfit that can't spell gpl |
| 13:16.49 | Katty | fourcheeze: you wouldn't want me. |
| 13:16.56 | [TK]D-Fender | Katty: Congrats... you know all you need to :) |
| 13:16.56 | fourcheeze | are you cheap ;-) |
| 13:17.02 | Katty | no |
| 13:17.08 | Katty | i'm priceless |
| 13:17.22 | Katty | [TK]D-Fender: but...but |
| 13:17.30 | Katty | [TK]D-Fender: that means their phone server is down for at least 3 hours )= |
| 13:17.37 | Katty | [TK]D-Fender: unacceptable! |
| 13:17.55 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 13:18.00 | Katty | and asterisk, too |
| 13:18.07 | Katty | some things happen and, well, i just don't know what to do |
| 13:18.17 | Katty | so maybe this fonality thing will help. |
| 13:18.26 | Qwell | eww |
| 13:18.28 | Katty | gives me someone to pass the blame onto, if nothing else. |
| 13:18.45 | fourcheeze | are there organisations who would setup/manage/support an asterisk cluster on behalf of someone, and if so does anyone recommend one? |
| 13:18.46 | Katty | and it'd probably be quicker than flying mister fender out :P |
| 13:18.46 | cpm | Katty fonality has put all kinds of press into doing everything cheap. |
| 13:18.56 | Katty | cpm: i know. |
| 13:19.00 | Qwell | not inexpensive either |
| 13:19.00 | Qwell | cheap |
| 13:19.01 | Katty | cpm: but southern missouri is cheap :/ |
| 13:19.02 | cpm | so, you the reseller has to break the news, that 'No, you can't have a pony' |
| 13:19.03 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 13:19.05 | Qwell | cheap != inexpensive |
| 13:19.19 | Katty | Qwell: but the /point/ is i don't know enough. |
| 13:19.22 | Katty | Qwell: AND software |
| 13:19.23 | cpm | customers dont' like hearing they can't have a pony |
| 13:19.25 | Katty | Qwell: pretty software!! |
| 13:19.37 | Katty | cpm: i don't like hearing i can't have a pony either :< |
| 13:19.59 | Katty | Qwell: tech support and software is their selling point with the boss man. |
| 13:20.03 | cpm | http://www.brainfuel.tv/wp-content/uploads/2006/03/nopony.jpg |
| 13:20.06 | Katty | Qwell: two things i can't really do myself yet. |
| 13:20.29 | cpm | Anyway, if you look at their site, the claim a fully functioning pbx for less than $1k, |
| 13:20.42 | cpm | so, you, the reseller has to explain, "err, no, , " |
| 13:21.09 | plasmid | I am trying to record incoming calls from a company tech support but I am not entirely sure what to press when I begin conversing with them and WHERE do these recorded calls end up (pathwise) |
| 13:21.25 | mosty | plasmid, show features |
| 13:21.58 | mosty | one-touch-monitor'ed calls are put in /var/spool/asterisk/monitor/ on my dist |
| 13:22.37 | plasmid | and what key do u press on your phone to start recording (default)? |
| 13:22.47 | Katty | plasmid: it's specified in features.conf |
| 13:22.59 | Katty | plasmid: or you can do Show Features at the cli |
| 13:23.02 | *** join/#asterisk dwmw2_gone (i=ctrlprox@81.187.2.161) |
| 13:23.34 | mosty | plasmid, show features, that will tell you |
| 13:24.00 | plasmid | checking... |
| 13:24.52 | fourcheeze | so no-one really wants to recommend their own (or someone else's) asterisk consultancy? |
| 13:25.00 | Qwell | fourcheeze: Digium |
| 13:25.11 | *** join/#asterisk saftsack (n=oliver@84.167.196.42) |
| 13:25.15 | fourcheeze | do they do stuff like remote management? |
| 13:25.18 | Qwell | dunno |
| 13:25.19 | mosty | fourcheeze, depends what you need, and what currency you pay in |
| 13:25.34 | fourcheeze | I can probably pay in $ or ÂĢ |
| 13:25.44 | fourcheeze | but I want a cluster with full management |
| 13:26.10 | mosty | sounds like a big job, you probably need someone local |
| 13:26.17 | Daejeo1 | ping JT |
| 13:26.23 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 13:26.24 | fourcheeze | yeah, that's what I'm thinking |
| 13:26.25 | nestAr | plasmid: for my tech support call center, i didn't have a push button, it just recorded all the time for those calls. |
| 13:26.29 | fourcheeze | anyone know anyone in the UK? |
| 13:26.35 | plasmid | also, for some reason my DID# is private and there are plenty of times where i have to press *82 just to make a phone call. I think I would like my # to be visible. Where do I turn off that "privacy" feature? FOUnd the ONE-touch Monitor = *1 to record? |
| 13:26.38 | nestAr | never trust someone in tech support to push a button. |
| 13:26.46 | *** part/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 13:26.50 | plasmid | nestAr, that's nice |
| 13:26.55 | fourcheeze | nestAr: never trust someone in tech support. |
| 13:29.01 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 13:29.46 | fourcheeze | anyone using asterisknow ? |
| 13:29.54 | DirtyD | I'm really torn.. Should I get the Lenovo x60t with XGA or SXGA The XGA has is MultiTouch the SXGA is not not,. |
| 13:30.18 | fourcheeze | sxga |
| 13:30.45 | fourcheeze | more pixels is always better |
| 13:31.20 | *** join/#asterisk x-blur (n=walterkl@bb219-75-58-28.singnet.com.sg) |
| 13:31.37 | x-blur | hello, can anybody help me with my TC400B ? |
| 13:31.50 | mosty | x-blur, what's the problem? |
| 13:32.20 | x-blur | lspci shows me the card, but modprobe wctc4xxp doesn't do anything |
| 13:32.31 | x-blur | insmod the module gives me following error in dmesg |
| 13:32.49 | Qwell | x-blur: I'd suggest calling Digium support |
| 13:32.55 | x-blur | TC400B: firmware tc400m-firmware.bin not available from userspace |
| 13:32.55 | x-blur | wctc4xxp: probe of 0000:02:0b.0 failed with error -1 |
| 13:33.08 | x-blur | well, they point me here... as one of the options... |
| 13:33.14 | Qwell | what? |
| 13:33.22 | Qwell | You called, and they said to come here? |
| 13:33.42 | x-blur | no, I go to their web-site... they're not awake I presume |
| 13:33.47 | Qwell | oh, right... |
| 13:33.49 | x-blur | I am in Singapore, GMT +8 |
| 13:33.51 | Qwell | yeah, it's 7:30 there |
| 13:34.04 | mosty | x-blur, tc400m-firmware.bin exists? |
| 13:34.12 | Qwell | x-blur: give it likt an hour and a half |
| 13:34.14 | x-blur | where do I find it? |
| 13:34.33 | mosty | x-blur, find / -type f name 'tc400m-firmware.bin' |
| 13:34.36 | x-blur | it's a tc400b, now I need to find tc400m ? :p |
| 13:34.57 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:36.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:36.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:37.03 | x-blur | it's in /usr/src/zaptel-1.4.2.1/wctc4xxp/tc400m-firmware.bin |
| 13:37.41 | *** join/#asterisk bbryant (n=brett@69-94-196-169.biltmorecomm.com) |
| 13:37.53 | mosty | x-blur, i suspect that the file hasn't been installed properly, it should have been copied somewher |
| 13:38.19 | x-blur | ehm, where is it supposed to be? |
| 13:38.29 | *** join/#asterisk bagheaduk (n=billybob@host217-34-48-25.in-addr.btopenworld.com) |
| 13:38.37 | x-blur | in the same directory as the .ko driver? |
| 13:38.42 | mosty | not sure. did you do make install? |
| 13:38.46 | x-blur | of course :p |
| 13:39.44 | DirtyD | fourcheese: You think on a tablet, sxga (more pixels) is better over and XGA with MultiTouch? I can't make up my freaking mind! I've been dwelling on this for 3 months now. |
| 13:40.14 | bagheaduk | hey - ive got a var set called OPER in a dialplan which is a number, and i need to access a field called try1 / try2 etc -- im trying to get it like: ${TRY${OPER}} however its not working - any ideas? using ${TRY1} works.. |
| 13:40.24 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 13:40.34 | DirtyD | Why can't they make a SXGA with multitouch.. bastards |
| 13:40.36 | mosty | x-blur, strace the modprobe command and see where it's looking |
| 13:40.52 | shadebob | hi, I have a problem with asterisk, FAI with 2vlan (1 for voice, 1 for data) and my linux box |
| 13:41.09 | shadebob | someone can help me? |
| 13:41.17 | [TK]D-Fender | DirtyD: Get a shrink ... and a wireless mouse. Geez. Touchpads are for when you portable mouse dies :) |
| 13:42.00 | [TK]D-Fender | Katty: I'm taking the plunge! |
| 13:42.04 | [TK]D-Fender | Katty: http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=2887690&CatId=1751 |
| 13:42.12 | [TK]D-Fender | Katty: http://www.insight.ca/apps/productpresentation/index.php?alert=categoryresults&product_id=IF4052284 |
| 13:42.26 | [TK]D-Fender | Katty: Size isn't everything, unless its 120" ;) |
| 13:42.56 | Qwell | 120"? wtf? |
| 13:43.52 | [TK]D-Fender | Qwell[]: Ditching my 52" rear-projection HDTV for a projection setup |
| 13:43.57 | Qwell | oh |
| 13:43.58 | Katty | 120"? |
| 13:44.04 | *** join/#asterisk Vec2 (n=Vec@dsl-243-90-187.telkomadsl.co.za) |
| 13:44.22 | Katty | i see |
| 13:44.29 | bagheaduk | sorry to be a pain - any ideas re above q? |
| 13:44.53 | Qwell | bagheaduk: field called "try1"? |
| 13:44.56 | Qwell | It's case sensitive |
| 13:45.02 | Qwell | So, ${try${OPER}} |
| 13:45.18 | *** join/#asterisk okkar (n=andy@213.165.233.139) |
| 13:45.29 | *** join/#asterisk wyoming (n=steve_mu@69-94-197-138.biltmorecomm.com) |
| 13:45.39 | x-blur | mosty: that gives me a shitload of info... what am I looking for? |
| 13:45.41 | bagheaduk | Qwell - it is actually the right case in the config - i just changed it when typin into irc (oops) |
| 13:46.23 | Qwell | bagheaduk: NoOp(${try1} - ${OPER}) |
| 13:46.31 | Qwell | Do the values of both of those show what you expect? |
| 13:47.22 | x-blur | mosty: by the way, a make install shows me this: |
| 13:47.23 | x-blur | make -C firmware hotplug-install |
| 13:47.24 | x-blur | make[1]: Entering directory `/usr/src/zaptel-1.4.2.1/firmware' |
| 13:47.24 | x-blur | Firmware zaptel-fw-oct6114-064.bin is already installed with required version 1.05.01 |
| 13:47.24 | x-blur | Firmware zaptel-fw-oct6114-128.bin is already installed with required version 1.05.01 |
| 13:47.24 | x-blur | Firmware zaptel-fw-tc400m.bin is already installed with required version MR5.6 |
| 13:47.43 | *** join/#asterisk sashion (n=djbdsf@196.33.37.1) |
| 13:47.57 | sashion | tzafrir: Are you here? |
| 13:48.04 | bagheaduk | qwell - yes, try1 = IAX2/.... oper = 1 |
| 13:48.26 | bagheaduk | qwell - however ${try${OPER}} gives nothing |
| 13:48.40 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-27-9.bflony.east.verizon.net) |
| 13:48.43 | Qwell | oper, or OPER? |
| 13:48.47 | Qwell | again, BIG difference :) |
| 13:48.51 | Qwell | ...literally? |
| 13:49.39 | bagheaduk | qwell - exten => s,1,set(TRIES=0) ; |
| 13:50.04 | bagheaduk | qwell - its TRIES, not OPER in this case - however same thing |
| 13:50.08 | sashion | I'm getting a ticking noise on my Analog line with asterisk... any ideas? |
| 13:50.51 | redax | uh.. don't you have the standalone app_devstate patch for asterisk 1.2 ? |
| 13:51.09 | redax | the link on the voip-info is broken. |
| 13:51.23 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 13:51.50 | [TK]D-Fender | bagheaduk: Pastebin the whole mess. |
| 13:51.53 | [TK]D-Fender | ~pb |
| 13:52.11 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:53.42 | redax | I try to take out the app_devstate stuff from bristuff.. but it changed the ast_device_state_changed_literal() to 2+ args. Do I need that patch also ? |
| 13:53.58 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 13:54.09 | bagheaduk | qwell / Fender = http://pastebin.ca/502324 |
| 13:55.04 | bagheaduk | bagheaduk - noting that TRIES is set to 0 to begin with (not shown on page) |
| 13:55.06 | *** join/#asterisk mkl1525 (n=qwertz@i59F7099D.versanet.de) |
| 13:56.05 | mkl1525 | Hi, I can use asterisk db to put somehting in it and then use an if to decide something, but how can I set the db value when starting asterisk? don't want to do this by hand. |
| 13:57.36 | bagheaduk | qwell - are you allowed to do that kind of string setting in asterisk? |
| 13:57.58 | mosty | is there mkl1525 call file? |
| 13:58.07 | mosty | woops |
| 13:58.14 | mosty | mkl1525, you could use a call file |
| 13:59.19 | sashion | mkl1525: when starting asterisk, you could write a little module that can do that... |
| 13:59.20 | x-blur | nobodhy can help me with my TC400B ? |
| 13:59.24 | sashion | check out skel.c |
| 13:59.28 | sashion | app_skel.c |
| 13:59.35 | redax | why does junghanns changed the ast_device_state_changed_literal() to accept cid_num and cid_name additionally at all? |
| 13:59.40 | mkl1525 | sashion, thanks will have a look at it |
| 13:59.42 | mosty | x-blur, do you have hotplug and/or udev installed? |
| 13:59.45 | redax | in bristuff ... |
| 13:59.47 | *** join/#asterisk saftsack (n=oliver@p54a7c6d8.dip.t-dialin.net) |
| 13:59.57 | x-blur | ehm, I believe so... |
| 14:00.07 | x-blur | by default installed on FC6 |
| 14:00.22 | x-blur | zaptel works though... |
| 14:00.36 | x-blur | and my Sangoma A104 as well... |
| 14:01.07 | mosty | x-blur, is the tc400b the only digium card in the box? |
| 14:01.12 | x-blur | yep |
| 14:03.28 | mosty | x-blur, i think i heard that there may be problems using sangoma and digium cards in the same machine |
| 14:04.05 | x-blur | well, i asked the digium guys at the asternic conference here in Singapore 2 weeks ago and they say there is no problem with that... |
| 14:04.11 | mosty | i just sent out a box with a tc400b, but it had a digium e1 card in it instead of the sangoma we normally use, for this very reason |
| 14:04.12 | x-blur | the transcoding card only transcodes |
| 14:04.25 | x-blur | doesn't interact in any other way directly with the other cards apparently |
| 14:04.44 | mosty | wanpipe patches zaptel though, i am not sure if that might cause problems |
| 14:05.41 | x-blur | I have looked at the patch (apparently the latest wanpipe can't patch the latest zaptel), and there doesn't look to be anything that interferes with the zttranscode |
| 14:06.26 | x-blur | it's only a few lines added in the zaptel-base.c ... |
| 14:06.37 | mosty | x-blur, have you tried stracing modprobe/insmod? it might show you where it's failing |
| 14:07.09 | x-blur | yes, it doesn't seem to be failing... you want me to paste the output somewhere? |
| 14:07.18 | x-blur | I can't seem to find anything wrong... and the driver is loaded... |
| 14:07.34 | x-blur | just nothing in dmesg and show transcoder in asterisk doesn't work |
| 14:07.43 | [TK]D-Fender | bagheaduk: "show function EVAL" |
| 14:09.19 | *** join/#asterisk saftsack (n=oliver@p54A7F649.dip.t-dialin.net) |
| 14:10.14 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:10.51 | irule | is there a way to add the current time and date to a noop message? I tried timestamp but it is blank thanks |
| 14:11.31 | [TK]D-Fender | irule: Check upgrade.txt . Things changed, read the notices they've been putting out for ages. |
| 14:11.35 | x-blur | mosty: http://pastebin.ca/502348 |
| 14:11.36 | mosty | irule: you can get the date with System i guess |
| 14:11.44 | [TK]D-Fender | irule: and the docs. Ther is a new functio for this sort of stuff. |
| 14:11.57 | irule | btw im on 1.4 |
| 14:15.12 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:15.39 | brodiem | I have a question on BLF -- what defines standards of how BLF statuses are passed? Is this part of the SIP standards? |
| 14:16.00 | [TK]D-Fender | brodiem: yes |
| 14:16.48 | *** join/#asterisk Acidcrawl (n=Miranda@12.168.96.254) |
| 14:16.55 | x-blur | mosty: any idea? |
| 14:17.44 | mosty | x-blur, no sorry. i shipped my box with the tc400b this morning, can't have a look at it now |
| 14:17.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:18.12 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-197-125.biltmorecomm.com) |
| 14:18.24 | Acidcrawl | If I wanted to have 2 queues, and have an agent log into both queues, is there a way for the agent to know from which queue a call came from? |
| 14:19.47 | mosty | Acidcrawl, set callerid |
| 14:19.53 | x-blur | pk, thanks so far. i guess i'll call digium when they wake up |
| 14:20.04 | *** join/#asterisk Cresl1n (n=matt@69-94-196-9.biltmorecomm.com) |
| 14:20.04 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:21.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:24.13 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:26.17 | nkrasko | Acidcrawl: you can add prefix to CallerID for calls, so it will be displayed on phones |
| 14:26.47 | Acidcrawl | cool, I got it, thanks for the hel |
| 14:28.26 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:29.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:29.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:29.23 | [TK]D-Fender | Katty: I wasted over an hour on hold with HP for printer software support only to get hung up on about 2 weeks ago |
| 14:30.13 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:30.33 | coolbeans | Hey guys, do you run with qualify on or off for phones? |
| 14:30.33 | [TK]D-Fender | Katty: And my little piece of wisdom for you today : "Life is like a penis ... if it gets too hard, %#@ it" :) |
| 14:30.34 | coolbeans | And what MS value is typical? |
| 14:30.36 | [TK]D-Fender | coolbeans: On personally. |
| 14:30.49 | [TK]D-Fender | coolbeans: typicaly = 2000 (what "yes' does) |
| 14:30.57 | coolbeans | [TK]D-Fender: Thanks. |
| 14:34.53 | Katty | [TK]D-Fender: :< |
| 14:35.59 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:36.06 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 14:37.02 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-61bd5a081e167e75) |
| 14:39.48 | redax | is there a way to query a devstate in asterisk cli ? |
| 14:41.40 | *** join/#asterisk saftsack (n=oliver@p54A7D9C7.dip.t-dialin.net) |
| 14:44.33 | drrt | redax, which device are u interested for ? |
| 14:44.51 | redax | actually SIP... |
| 14:45.04 | redax | like what's the state of SIP/110 ... |
| 14:45.24 | drrt | sip show peer 110 |
| 14:45.30 | drrt | sip show user 110 |
| 14:45.36 | mosty | redax, i would be interested to hear if you figure out how to do that from the dialplan |
| 14:46.21 | drrt | system(asterisk -rx "sip show peer 110) :) |
| 14:46.29 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:46.36 | redax | + grep ;-) |
| 14:47.36 | redax | why, there's no such a CLI command which returns the output of ast_device_state() ? |
| 14:47.56 | redax | gr. not CLI but dialplan |
| 14:47.57 | redax | app |
| 14:49.01 | mosty | redax, only if you use the bristuff patch |
| 14:50.34 | redax | mosty: seems like no. the bristuff app_devstate is suitable only for the Fake DS/xxx |
| 14:50.43 | redax | and only for settings |
| 14:51.05 | redax | although it's stores the fake DS/xxx state in the astdb, as DEVSTATES/xxx |
| 14:51.06 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:51.29 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:51.51 | *** join/#asterisk edguy (n=edguy@69-94-196-221.biltmorecomm.com) |
| 14:51.54 | redax | sorry. missinformation :) |
| 14:52.10 | redax | it has some kind of devstate query as well. but only for the fake device |
| 14:54.27 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 14:57.27 | *** join/#asterisk penguinFunk (n=penguin@89-145-196-28.xdsl.murphx.net) |
| 14:57.41 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 14:59.02 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:02.31 | phillipk | I have my asterisk box connected to an NEC PBX which is providing a PRI connection to it. Roughly every hour, asterisk shows the dchan on the PRI dropping, which interrupts service. The guy who supports the NEC for me thinks it's a timing issue. I've set the timing source in zaptel.conf for that span to 1 and 0, and I have the same problem either way. Is there any other timing adjustment I can/should make? |
| 15:02.31 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 15:04.47 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
| 15:06.11 | sashion | phillipk: have you enable extensive debugging on the span ? |
| 15:06.22 | sashion | it will give you an ISDN message on why it dropped |
| 15:06.47 | sashion | and who is the master of the link? Is it the NEC or the * ? |
| 15:07.11 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:07.20 | phillipk | I have not enabled extensive debugging. |
| 15:07.27 | phillipk | * is the master |
| 15:09.00 | sashion | phillipk: enable debuggin on the PRI, might be getting a message, or the NEC might be dropping the link when the PRI restarts |
| 15:09.16 | bkw_ | is it every hour on the hour? |
| 15:10.05 | phillipk | No. The drops have been as close together as 15 minutes. |
| 15:10.29 | sashion | phillipk: also try changing resetinterval = 3600 to something like resetinterval = 86400 |
| 15:10.35 | irule | is it possible to do something like call waiting within sip phones? the idea is to make sure users dont miss calls that come from sip peers |
| 15:10.39 | sashion | under zapata.conf |
| 15:10.40 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 15:10.49 | [TK]D-Fender | phillipk: make sure your card isn't sharing an IRQ and check your zttest score |
| 15:11.08 | [TK]D-Fender | irule: thats the phones job, not *'s |
| 15:12.34 | phillipk | It's not sharing an IRQ. What kind of score is too low on zttest? |
| 15:13.19 | sashion | 99.999% is the bare acceptable :P |
| 15:13.44 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:14.51 | puzzled | hi |
| 15:16.02 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 15:16.12 | *** join/#asterisk JoeMoes (n=JoeMoes@wolverine.vcc.de) |
| 15:20.33 | irule | [TK]D-Fender You have confused me, because currently if I dial from exten 102 to a busy exten 123 in a call with 145, 102 will get a busy signal and sends the call to voicemail. What I waould like to try is make a distinctive sound in 123 to be heard by the callee, and see 102 in the caller ID display or something, with an option to answer 102 leaving 145 in a MOH and then allow to put 102 at MOH to re-take 145 etc |
| 15:21.35 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:21.40 | [TK]D-Fender | irule: If you try to call the phone and the phone refuses you, that it. End of story. |
| 15:21.43 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 15:22.39 | [TK]D-Fender | irule: its the phones job to support multiple simultaneous calls and offer a beep like CW. If it does well... fix its config or pick a more flexible phone |
| 15:22.55 | phillipk | zttest looks good to me. I've got extensive debugging turned on on that span, so I guess I'll just wait for it to drop. Thanks everybody. |
| 15:23.26 | shadebob | hi. I have a problem with SIP and RTP flow. I have 120 sip phones, 12 telco lines. After a random time, RTP flow don't work and "unable to create socket : too many open files" appear in the CLI. I had fix the ulimit to 65535, the /etc/security/limits.conf to 65535 but no way... Probleme stay |
| 15:23.27 | irule | [TK]D-Fender oh I see thanks |
| 15:23.31 | shadebob | any idea? |
| 15:23.57 | NOT_guru | will zttest effect people in calls? |
| 15:25.31 | *** join/#asterisk berktr (n=canberk@teknopet.com) |
| 15:26.49 | drrt | shadebob, u are a new victim ) |
| 15:27.05 | drrt | shadebob, welcome ) do u use 1.4 branch ? |
| 15:27.11 | shadebob | yes |
| 15:27.14 | shadebob | 1.4.0 |
| 15:27.33 | [TK]D-Fender | shadebob: perhaps you should upgrade to a more stable release |
| 15:28.24 | shadebob | drrt TK : So it's a well known problem... I don't fond it in the bug.digium.com |
| 15:28.37 | drrt | shadebob, let me give u the number |
| 15:28.56 | shadebob | ok thanks drrt |
| 15:29.56 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 15:30.32 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 15:31.00 | drrt | shadebob, http://bugs.digium.com/view.php?id=9235 |
| 15:32.17 | shadebob | <PROTECTED> |
| 15:32.23 | shadebob | I will see |
| 15:32.48 | berktr | hello friends, i have extensions starting from 1001 to 1011 and all the other numbers are forwarded to the carrier for pstn |
| 15:33.04 | berktr | i want to do this however when i call the 1004 number for example |
| 15:33.09 | berktr | asterisk forwards it to carrier too |
| 15:33.12 | berktr | how can we solve this? |
| 15:33.39 | mosty | <PROTECTED> |
| 15:33.43 | berktr | exten => 1004,1,Dial(SIP/1004) |
| 15:33.58 | berktr | exten => _0.,1,Dial,SIP/carrier/9${EXTEN} |
| 15:34.08 | berktr | when i call 1004, it automatically calls 91004 |
| 15:34.09 | drrt | shadebob, yw |
| 15:34.41 | mosty | berktr, see what i just said? |
| 15:35.04 | berktr | i just saw it, but didn't understand what you meant by |
| 15:35.20 | *** join/#asterisk saftsack (n=oliver@p54A7F6FB.dip.t-dialin.net) |
| 15:35.53 | mosty | berktr, a single dial command can ring multiple devices at once |
| 15:36.04 | berktr | yes but i need it to call 1004 only |
| 15:36.11 | berktr | i don't want it to carry it to carrier |
| 15:36.26 | mosty | oh sorry, i misread your question |
| 15:37.06 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 15:37.20 | mosty | what you need to do is create one context for your local extensions, and another context for outgoing calls, then include => local-extensions followed by include => dial-out |
| 15:37.45 | drrt | what is the sequence in the context ? |
| 15:39.13 | drrt | berktr, ? |
| 15:39.31 | kombi | would any of you people have or know where to find the sip firmware for cisco's 7941? |
| 15:39.37 | mosty | berktr, see this page: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
| 15:39.47 | berktr | first is the carrier |
| 15:39.50 | berktr | then the extensions |
| 15:39.56 | berktr | is it important |
| 15:40.20 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-ee5927bb7af4817f) |
| 15:40.33 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 15:40.33 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 15:41.53 | [TK]D-Fender | kombi: www.cisco.com |
| 15:41.56 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 15:42.28 | kombi | TK-D-Fender: already done, only takes 2-3 weeks to be delivered.. |
| 15:42.44 | coppice | [TK]D-Fender: I saw something rather disgusting at the weekend. |
| 15:42.52 | [TK]D-Fender | kombi: Should be able to download direct with your smartnet contract |
| 15:42.54 | shadebob | drrt : it's my problem |
| 15:42.59 | coppice | Fender are making the Strat in pink with Hello kitty on it |
| 15:43.20 | kombi | TK-D-Fender: The contract is what takes that whole time.. |
| 15:43.24 | shadebob | drrt : have you test the 1.4.4 patch for th eUDP lingering |
| 15:44.10 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 15:44.12 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 15:44.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:44.33 | _VoiceMeUp_COM | can one point me to where i can see an example of setting a variable in manager ? |
| 15:44.41 | _VoiceMeUp_COM | i need more then one |
| 15:44.52 | _VoiceMeUp_COM | SetVariable: var1=val1;var2=val2 ? |
| 15:45.04 | drrt | shadebob, not yet. i m still using patch made by reporter. going to switch tonight |
| 15:45.23 | kombi | TK-D-Fender: ordered smartnet from our distri (ingram), they say it takes 3 weeks |
| 15:46.32 | kombi | whatever takes cisco that long.. |
| 15:48.07 | kombi | I gladly pay the money when smartnet eventually arrives, just need that firmware now for things to work.. |
| 15:48.12 | lee_is_me | <_VoiceMeUp_COM>: AMI info doesn't seem to indicate the ability to set multiple vars: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+SetVar |
| 15:48.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:48.26 | lee_is_me | someone can correct me if i'm wrong |
| 15:48.37 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:49.32 | mosty | lee_is_me, just do it multiple times? |
| 15:51.06 | berktr | what can i do if my carrier is trying to use comfort noise support and asterisk does not support it |
| 15:53.06 | [TK]D-Fender | berktr: tell them to stop, live with it, or change providers |
| 15:53.20 | berktr | does it affect the quality? |
| 15:53.47 | lee_is_me | that is what I would do...I haven't played with the AMI much yet |
| 15:54.14 | drrt | berktr, are u talking about VAD ? |
| 15:54.17 | lee_is_me | if you have a windows box, you can download a Manager API test utility that I wrote to try it out without having to write code |
| 15:54.17 | berktr | eys |
| 15:54.19 | berktr | yes i think |
| 15:54.49 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 15:55.12 | drrt | you can switch in on directly at the client`s side |
| 15:55.32 | *** join/#asterisk GuruJee (n=tilde@12.186.161.117) |
| 15:55.39 | *** join/#asterisk saftsack (n=oliver@p54a7f973.dip.t-dialin.net) |
| 15:55.44 | *** join/#asterisk Carlis4 (n=Nisse@c-b525e455.02-36-6c6b7012.cust.bredbandsbolaget.se) |
| 15:55.45 | berktr | no drrt, this is something different |
| 15:55.52 | lee_is_me | mosty: http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx | link is bottom of page |
| 15:56.24 | mosty | lee_is_me, i don't have a windows machine |
| 15:56.46 | lee_is_me | OK, sorry. Haven't had time to port to linux yet. |
| 15:58.54 | berktr | so, having comfort noise support at carrier side turned on affect the quality? |
| 15:59.28 | drrt | berktr, i think it does |
| 15:59.48 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:00.09 | mosty | lee_is_me, i don't run random code from people on irc anyway |
| 16:00.13 | *** join/#asterisk NirS (i=Nir@87.68.0.156.cable.012.net.il) |
| 16:00.35 | lee_is_me | mosty: does IRC have a random function? ;) |
| 16:01.02 | lee_is_me | i don't blame you. |
| 16:04.19 | _VoiceMeUp_COM | no way to set a header via manager right ? |
| 16:04.35 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 16:05.39 | mosty | _VoiceMeUp_COM, you could set a variable, and have your dialplan use that variable? |
| 16:05.53 | _VoiceMeUp_COM | well proplem .. |
| 16:05.58 | _VoiceMeUp_COM | its a pstn to pstn call module |
| 16:06.15 | mosty | so? |
| 16:06.18 | _VoiceMeUp_COM | so...it needs to either to sip/box2/number OR local/number@mynewconext |
| 16:06.23 | _VoiceMeUp_COM | i DONT want to use local |
| 16:06.26 | GuruJee | people |
| 16:06.29 | GuruJee | yoyoyoyoyoyo |
| 16:06.30 | _VoiceMeUp_COM | it casues deadlocks 56|% more |
| 16:06.32 | [TK]D-Fender | _VoiceMeUp_COM: Nope. The call is already in progress, too late to think about adding headers now. |
| 16:06.38 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 16:06.45 | GuruJee | can anyone please point me to a good resource about configuring dialplans? |
| 16:06.48 | _VoiceMeUp_COM | not in progreess |
| 16:06.52 | _VoiceMeUp_COM | its before the originate command |
| 16:06.55 | [TK]D-Fender | GuruJee: ... |
| 16:06.56 | [TK]D-Fender | ~book |
| 16:07.08 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:07.09 | [TK]D-Fender | ~wikis |
| 16:07.13 | jbot | [wikis] http://www.voip-info.org |
| 16:07.13 | GuruJee | ok |
| 16:07.17 | vAd0r | can someone help me with an authentication problem. when i am at my house using the private ip it works fine. when im offsite and use the public ip w/ ports 5060 10000-20000 open i get wrong password |
| 16:07.18 | [TK]D-Fender | _VoiceMeUp_COM: so NOT happening... Chan_local or bust. |
| 16:07.20 | GuruJee | hey TK, do you know anything about DUNDi |
| 16:07.21 | GuruJee | ? |
| 16:07.28 | [TK]D-Fender | GuruJee: Nope. |
| 16:07.59 | [TK]D-Fender | vAd0r: Then your user/pass or other off is wrong, plain & simple. |
| 16:08.16 | _VoiceMeUp_COM | hmmm becasue asterisk has created a call upon calling the manager itself ? |
| 16:08.16 | GuruJee | basically, i have a DUNDi system working over IAX2 its working great for itnernal calls. All I want to do is to be able to route outgoing calls on a pbx with a PRI line |
| 16:08.22 | vAd0r | yes i would think that but all i have to do is change domain to my local ip on site and it works |
| 16:08.30 | vAd0r | i never change the password |
| 16:08.37 | vAd0r | just the domain ip in xlite |
| 16:08.54 | vAd0r | do i use some other sort of username when i am offsite |
| 16:09.06 | [TK]D-Fender | vAd0r: Nope |
| 16:09.31 | vAd0r | so i dont understand then why it would say wrong password |
| 16:10.12 | tutt9876 | hi , sorry I can't get a dialtone when connecting to asterisk 1.4.2 |
| 16:10.59 | tutt9876 | any idea? |
| 16:11.37 | PioneerVM2 | Manx: you here? |
| 16:12.16 | tutt9876 | Do you use 1.4.2? |
| 16:13.13 | mosty | tutt9876, is your phone registered? |
| 16:13.32 | cpm | phone registration leads to confiscation! |
| 16:13.39 | *** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net) |
| 16:13.59 | tutt9876 | yes i can make calls but can't get a dialtone until connected |
| 16:15.30 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:15.32 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 16:15.51 | mosty | tutt9876, you did not answer my question |
| 16:16.18 | tutt9876 | I use Xlite and I get registred on asterisk$* |
| 16:16.26 | ManxPower | PioneerVM2: sort of |
| 16:16.44 | mosty | tutt9876, ok so if the sip client is registered and you can make calls, what is the problem? |
| 16:16.46 | ManxPower | tutt9876: and "sip show peers" shows the X-lite with the correct IP address |
| 16:17.22 | ManxPower | tutt9876: you can MAKE calls without being registered to asterisk. You just can't receive calls if your phone is on a dynamic IP and it does not register. |
| 16:17.26 | tutt9876 | No in the headset I don't have ringing tone until the connectin is established |
| 16:17.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:17.53 | mosty | tutt9876, that is normal, isn't it? |
| 16:18.38 | errr | Im having some trouble with echo on certian incoming calls. Whats the best way to troubleshoot that? |
| 16:18.47 | tutt9876 | no when using 1.2 version I think I had ringing tone |
| 16:18.55 | PioneerVM2 | If you get an incoming call and immediately "Dial" it (forward) out to a new number, is CallerID automatically passed from the incoming caller (if you dont set it to anything specifically) |
| 16:19.05 | Qwell | tutt9876: dialtone is not the same thing as ringing tone... |
| 16:19.07 | ManxPower | tutt9876: make sure yo have a /etc/asterisk/indications.conf |
| 16:19.19 | tutt9876 | Sorry I made a mix |
| 16:19.20 | ManxPower | PioneerVM2: yes, that is correct. |
| 16:21.11 | *** join/#asterisk saftsack (n=oliver@p54a7c64d.dip.t-dialin.net) |
| 16:21.35 | *** join/#asterisk suhler (n=suhler@69-94-196-219.biltmorecomm.com) |
| 16:24.27 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 16:24.46 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:24.59 | Qwell | Mercestes: *cough* |
| 16:25.08 | [TK]D-Fender | ManxPower: You can receive calls on a dynamic IP without registering... with a few minutes of thought ;) |
| 16:25.12 | Mercestes | >.> |
| 16:25.17 | Mercestes | I know, I know..... |
| 16:25.26 | vAd0r | Here is my problem. It's telling the phone to send auth to sip:5001@172.17.2.51 .... but 172.17.2.51 = not locateable. so... xlite says hi to asterisk... asterisk says auth to 172.blah ... it tries to connect to 172... and 401 |
| 16:25.28 | Mercestes | IT is having an audit. |
| 16:25.45 | *** join/#asterisk nahirean (n=ninja@unaffiliated/nahirean) |
| 16:25.53 | vAd0r | that happens when i try to connect to my asterisk from outside w/ my pub ip |
| 16:26.21 | [TK]D-Fender | vAd0r: Your remote phone is clearly behind another NAT and you have not told * about that. |
| 16:26.35 | vAd0r | where do i tell it |
| 16:26.43 | vAd0r | that is correct |
| 16:26.54 | [TK]D-Fender | vAd0r: "nat=yes" in the phone's user/peer setup |
| 16:27.01 | PioneerVM2 | Manx: sorry was on with voicepulse... I had written you and sent you that data yesterday but you were not here and then you tried to find me but i had left by then. |
| 16:27.20 | *** join/#asterisk ixx (i=foobar@cpe-70-112-123-132.austin.res.rr.com) |
| 16:27.22 | vAd0r | in the extensions file? |
| 16:27.26 | PioneerVM2 | manx: it looks as if there maybe a glitch in Asterisk, but not sure regarding the caller ID issue |
| 16:27.48 | PioneerVM2 | voicepulse sniffed packets and we ran all the tests -- the "invite" packets to them were showing the caller ID info was coming from my end |
| 16:27.56 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:28.32 | *** join/#asterisk dr_decimal (n=stefan@c-68-45-144-101.hsd1.pa.comcast.net) |
| 16:29.31 | vAd0r | where do i set that [TK]D-Fender |
| 16:30.11 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-9-117.w81-251.abo.wanadoo.fr) |
| 16:30.13 | [TK]D-Fender | vAd0r: I just told you. |
| 16:30.22 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 16:30.31 | vAd0r | i dont know where the phone's user/peer setup is |
| 16:30.37 | vAd0r | sip.conf |
| 16:32.54 | tutt9876 | sorry I have check indications.conf and restarted asterisk but still have no riging tone: any idea? |
| 16:33.13 | tutt9876 | ringing |
| 16:34.49 | rene- | hey guys, i was playing with vlans yesterday, my switch asigns VLAN2 to my phone, and VLAN 3 to the desktop behind it, however the desktop should be on a network that originally didnt had any VLAN, so now even if i set the desktop to the correct subnet, it cant see any gear, |
| 16:35.10 | rene- | it seems that the moment you start using vlans in an equipment, you have to set up a vlan in every port |
| 16:35.24 | rene- | it doesnt seem like you can use VLAN 0 (no vlan) on those |
| 16:36.14 | tutt9876 | is my question a dummy one? |
| 16:36.30 | rene- | i still dont get how i can connect the NO VLAN network to the desktops that are getting VLAN3 because they are connected in the switch of the ip phone |
| 16:36.30 | GuruJee | can anyone help me with a DUNDi dialplan, please? |
| 16:37.01 | blitzrage | GuruJee: pastebin what you have, and ask your question |
| 16:37.03 | ManxPower | rene-: this is really a switch issue, not an asterisk issue. |
| 16:37.15 | ManxPower | try assigning the default VLAN to be VLAN 0 or VLAN 1 |
| 16:37.19 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-97.biltmorecomm.com) |
| 16:37.25 | GuruJee | blitzrage: where can I paste and hwo can I paste my configs? |
| 16:37.43 | GuruJee | I have my extensions_custom.conf right here - do you want me to paste it in ur pvt? |
| 16:37.56 | ManxPower | ~pb |
| 16:38.00 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:38.14 | GuruJee | k thanks |
| 16:38.16 | GuruJee | let me do that |
| 16:38.19 | [TK]D-Fender | yay FreePBX... |
| 16:38.27 | ManxPower | GuruJee: extensions_custom.conf? That sounds like FreePBX AMP, etc |
| 16:38.47 | rene- | ManxPower: i know, sorry |
| 16:39.16 | ManxPower | rene-: on Ciscos VLAN 1 is the "default vlan" |
| 16:39.53 | rene- | ManxPower: default vlan, as in no VLAN? |
| 16:40.03 | GuruJee | bltirage: http://paste.debian.net/28589 |
| 16:40.16 | GuruJee | thats file on 1 system. This pbx has the terminating PRIs |
| 16:40.28 | ManxPower | rene-: more or less. |
| 16:40.34 | tutt9876 | sorry I have check indications.conf and restarted asterisk but still have no ringing tone: any idea? |
| 16:40.38 | ManxPower | rene-: what brand of switch are you using? |
| 16:40.40 | rene- | cisco |
| 16:41.16 | rene- | it has a profile for desktop+phone, that gives the phone the vlan 2, and you can select what you want to use for the desktop connected to the phone |
| 16:41.26 | rene- | i have been able to assign vlan3 to the desktop |
| 16:41.31 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-97.biltmorecomm.com) |
| 16:41.32 | ManxPower | rene-: what SPECIFIC model? |
| 16:41.39 | *** join/#asterisk saftsack (n=saftsack@pd9e07185.dip.t-dialin.net) |
| 16:41.41 | tutt9876 | do you have ringing tone with your asterisk? |
| 16:41.55 | GuruJee | bltirage: http://paste.debian.net/28590 thats the second file |
| 16:41.55 | rene- | cisco catalyst express 500 24 ports poe |
| 16:42.13 | rene- | not as powerful as real ciscos |
| 16:42.17 | rene- | just a web interface |
| 16:42.33 | GuruJee | bltirage: http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 16:42.42 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 16:44.08 | ManxPower | rene-: Oh. Don't know those. We have Catalyst 550x switches |
| 16:45.04 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 16:45.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 16:45.29 | rene- | but, like, if the desktops are getting vlan3, what could i do to get the packets untagged, (maybe using linux, maybe using the switch) |
| 16:45.53 | rene- | i was thinking about nat |
| 16:45.59 | rene- | but seems overly complicated |
| 16:47.04 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 16:47.27 | rene- | ManxPower: sorry to be a but offtopic |
| 16:47.50 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:48.53 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 16:49.46 | justdave | my local asterisk box at home is being strange. Can't get it to register with my sip provider for some reason. |
| 16:49.59 | rene- | i was wondering, i can setup a linux box connected to VLAN2 in interface0 connected to the switch, that solves asterisk connectivity, then i could connect VLAN3 to that same switch interface so i can talk from the desktops to the linux box, and then finally i could plug the interface2 of the linux server to the No Vlan network, configure a suitable IP and then enable routing. |
| 16:50.00 | justdave | it's been working fine for several months and I haven't touched the config. :) |
| 16:50.15 | tutt9876 | I ma using sip connextion to asterisk but no ringing tone |
| 16:50.21 | tutt9876 | connexion |
| 16:50.36 | rene- | so if eth0.3 and eth1 are on the same subnet, maybe they can magically talk? |
| 16:50.42 | justdave | sip debugging shows packets going back and forth between me and the sip provider (so the network isn't blocked) but none of those packets have a registration attempt. |
| 16:50.46 | rene- | or do i need to do bridging inside the linux box? |
| 16:50.51 | justdave | outgoing calls work, incoming doesn't, because it's not registered |
| 16:51.06 | *** join/#asterisk edguy3 (n=edguy3@69-94-196-190.biltmorecomm.com) |
| 16:51.57 | tutt9876 | have you ringing tone in your asterisk? |
| 16:52.31 | justdave | not sure what you mean by that |
| 16:52.59 | nahirean | what is the result of a sip show registry? |
| 16:53.05 | nahirean | request sent? anything? |
| 16:53.10 | justdave | my sip show registry shows nothing |
| 16:53.16 | justdave | just the table headers |
| 16:53.54 | tutt9876 | dring dring: have you some? |
| 16:54.01 | nahirean | is this straight asterisk? not a gui such as trixbox? |
| 16:54.05 | ManxPower | "sip show registry" shows you devices that Asterisk is registered TO |
| 16:54.14 | *** join/#asterisk umay (n=chris@71-208-167-161.hlrn.qwest.net) |
| 16:54.21 | nahirean | ManxPower: wouldn't that be valid if he were having inbound call issues? |
| 16:54.22 | ManxPower | "sip show peers" shows you what devices have registered TO asterisk. |
| 16:54.25 | justdave | ManxPower: right, that'd be what we're trying to figure out |
| 16:54.25 | tutt9876 | I am using Xlite with asterisk |
| 16:54.42 | justdave | outbound registration from my asterisk box to an external sip provider |
| 16:54.53 | justdave | not a phone registring to asterisk (those all work fine :) |
| 16:54.53 | nahirean | justdave: is this straight asterisk? |
| 16:55.04 | ManxPower | nahirean: "sip show registry" would be what you want if he is having trouble with calls from a service provider getting to Asterisk. |
| 16:55.17 | nahirean | ManxPower: right, perhaps I misunderstood, but I thought that was the issue? |
| 16:55.23 | ManxPower | justdave: Is Asterisk behind NAT? |
| 16:55.47 | phillipk | OK, my PRI just dropped again. Can anyone look at a log excerpt for me? http://pastebin.ca/502797 |
| 16:55.51 | justdave | yeah, it's behind NAT. network works, I can make outgoing calls. |
| 16:55.58 | ManxPower | justdave: if "sip show registry" is empty, then you have a problem with sip.conf |
| 16:56.34 | ManxPower | justdave: If Asterisk is behind NAT then you need localnet=, externip= and forward UDP ports 5060 and 10,000 - 20,000 |
| 16:56.37 | ManxPower | justdave: have you done that? |
| 16:56.44 | justdave | yes, it's worked fine for months |
| 16:56.50 | justdave | just suddenly stopped the last day or two |
| 16:56.52 | ManxPower | justdave: what changed? |
| 16:56.54 | nahirean | if the registration lines are in sip.conf and it's showing nothing i doubt it's a networking issue |
| 16:56.57 | justdave | not a thing that I know of |
| 16:57.07 | ManxPower | justdave: then your provider is prolly down |
| 16:57.19 | justdave | how come I can make outgoing calls then? |
| 16:57.28 | nahirean | if the provider were down it would say "request sent" |
| 16:57.36 | nahirean | or at least have something |
| 16:57.36 | ManxPower | frequently providers have different servers for inbound .vs. outbound |
| 16:57.36 | justdave | yeah, it's not getting that far |
| 16:57.53 | ManxPower | justdave: what is the actual host portion of the register => line? |
| 16:57.54 | justdave | sip show registry shows nothing |
| 16:58.11 | justdave | actually, the machine did just get rebooted because of a power failure the other day |
| 16:58.32 | ManxPower | justdave: if your internet was down, at that time then you need to stop and start asterisk |
| 16:58.36 | nahirean | try a sip reload to see if it'll parse the registration lines? |
| 16:58.39 | justdave | only thing I can think of that a reboot would affect is zap drivers |
| 16:58.45 | justdave | but zap shouldn't affect sip |
| 16:58.49 | ManxPower | specifically if Asterisk fails to look up a hostname it will never try again |
| 16:59.01 | justdave | yeah, I restarted asterisk a couple times already |
| 16:59.02 | *** join/#asterisk renier (n=renier@69.79.111.24) |
| 16:59.11 | justdave | although I told it to restart, I didn't actually stop it |
| 16:59.23 | nahirean | pkill -9 asterisk :) |
| 16:59.38 | ManxPower | I'm still waiting for that hostname |
| 17:00.18 | kombi | anyone maybe know where a cisco 7941 sip firmware might be flying about? |
| 17:00.32 | Qwell | kombi: cisco.com |
| 17:01.02 | kombi | Qwell: only with a smartnet contract that takes my reseller 2 weeks to get.. |
| 17:01.10 | Qwell | Welcome to Cisco. |
| 17:01.21 | kombi | Qwell: lol,, |
| 17:01.26 | kombi | had I known.. |
| 17:01.34 | Qwell | indeed |
| 17:01.45 | kombi | torrent? p2p somewhere? |
| 17:02.32 | kombi | how can they even sell a sip phone and charge extra for it to work as a sip phone? |
| 17:02.39 | justdave | dns and dhcp for the lan are on the same box with asterisk, and those both start before asterisk does in the boot order |
| 17:02.59 | ManxPower | kombi: they don't sell SIP phones. They sell SCCP phones |
| 17:03.20 | kombi | ManxPower: they advertise them as sip phones though.. |
| 17:03.37 | justdave | this was an Asterisk@Home originally, it's been manually upgraded in pieces (Asterisk 1.2.18, CentOS 4.5) and I usually ignore the GUI and edit the files manually anyway, because I'm used to doing that on the big * servers at work. :) |
| 17:04.47 | *** join/#asterisk dasuberdavid (i=david@nat/digium/x-f12659ef84c53826) |
| 17:04.52 | deeperror | i have recently upgraded from 1.2.8 - 1.2.18 and have lost the ringback tones on inbound calls. Where is ringback generated at the ATA or within asterisk? I'm also using a RT31P2 that worked fine prior to the update. Any clues? |
| 17:05.05 | [TK]D-Fender | kombi: Cisco is more trouble than they're worth. Polycom > All. |
| 17:05.15 | ManxPower | deeperror: it depends on if the call has been answered or not. |
| 17:05.28 | deeperror | the call rings to my handsets and i can answer them |
| 17:06.24 | deeperror | but the caller hears silence until the call is answered or sent to voicemail |
| 17:06.25 | cpm | em onto 24 |
| 17:06.25 | nahirean | Put a ",Ringing" in the syntax? |
| 17:06.25 | kombi | [TK]D-Fender: I am beginning to see that..;) (you have a tough name to type) |
| 17:06.25 | [TK]D-Fender | kombi: Thats what auto-complete is for.... |
| 17:06.25 | [TK]D-Fender | kombi: 3 keystrokes. |
| 17:07.12 | [TK]D-Fender | cpm: ? |
| 17:07.12 | kombi | [TK]D-Fender: that is the first time I realized there was auto complete on a chat client... |
| 17:07.16 | kombi | even on epic (which is what I use normally)? |
| 17:07.26 | deeperror | nahirean i have done this and even added playback no joy |
| 17:07.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:07.51 | cpm | [TK]D-Fender, the show, 24, where all cisco telcom stuff works like complete magic, raising the expectation that this crap works |
| 17:08.13 | [TK]D-Fender | cpm: The phones work... its users are lazy and avoid work wherever possible :) |
| 17:08.17 | kombi | that is exactly why i got it, jack bauer.. |
| 17:08.31 | PioneerVM2 | that cisco conference system is pretty cool (the video one) |
| 17:08.38 | kombi | I'll call division! |
| 17:08.53 | cpm | yeah, but the network that you need to support it, its pretty specific |
| 17:08.53 | ManxPower | If you don't hear ringback, adding the Ringing app to your dialplan almost NEVER fixes the problem |
| 17:08.56 | anonymouz666 | who tested the HPEC? |
| 17:08.57 | mrdigital | cpm: whats the easist way to do faxes in asterisk? |
| 17:08.58 | [TK]D-Fender | kombi: My Polycom's all use the Cisco "24" ringer (the only good part anyways), so :p |
| 17:09.20 | cpm | remember that system only works with <250ms latency, *including* the 85ms latency of the codecs, per end |
| 17:09.21 | ManxPower | anonymouz666: define "tested" |
| 17:09.28 | kombi | [TK]D-Fender: no polycom distribution in this country, unfortunately |
| 17:09.31 | [TK]D-Fender | mrdigital: with an analog fax on an analog line that has NOTHING to do with *. |
| 17:09.33 | ManxPower | mrdigital: there is no easy way to do faxes in Asterisk |
| 17:09.34 | justdave | I should wipe this machine and just put asterisk 1.4 on it without the GUI crap one of these days. |
| 17:09.35 | cpm | mrdigital, I don't do faxes with asterisk |
| 17:09.42 | mrdigital | ok |
| 17:09.46 | [TK]D-Fender | kombi: Whereabouts? |
| 17:09.58 | kombi | [TK]D-Fender: germany |
| 17:09.58 | ManxPower | The way I do faxes is EASY and RELIABLE, but it is not cheap. |
| 17:10.17 | [TK]D-Fender | kombi: There are resellers there, but a fair bit more pricy. |
| 17:10.28 | deeperror | ManxPower: that seems to be a line right from the docs haha yea i know playback doesn't fix this but i've tried quit a few things |
| 17:10.31 | kombi | [TK]D-Fender: which? |
| 17:10.40 | justdave | pots ftw for faxes :( |
| 17:10.48 | deeperror | could it be the provider? or is it on my end? |
| 17:10.51 | [TK]D-Fender | kombi: Can't recall off-hand, just know I've run across them in googling. |
| 17:11.07 | ManxPower | deeperror: make sure you have a /etc/asterisk/indications.conf This file controls the tones used AFTER a call has been answered (Background and several other apps automagically answer the line) |
| 17:11.24 | ManxPower | This file does NOT control ringback if the call has not been answered yet. |
| 17:11.29 | deeperror | yep that is in there and setup fine |
| 17:11.44 | deeperror | everything worked in 1.2.8 |
| 17:11.55 | deeperror | i setup a new box moved configurations to it |
| 17:12.03 | ManxPower | deeperror: just because it worked in a previsous version doesn't mean anything..... |
| 17:12.06 | deeperror | everything works 100% as before except ringback tones to inbound callers |
| 17:12.26 | anonymouz666 | [TK]D-Fender: tested the HPEC and got good results |
| 17:12.26 | ManxPower | deeperror: make sure your indications.conf was not overwritten. |
| 17:12.30 | deeperror | it means that it has something to do with the new version of the software or something that has changed i need to find |
| 17:12.33 | ManxPower | I don't use 1.2.18 because it kept crashing on us. |
| 17:12.45 | tutt9876 | I ma using sip connexion with xlite to asterisk but no ringing tone |
| 17:12.48 | deeperror | it is there |
| 17:12.50 | deeperror | looking at it now |
| 17:12.56 | kombi | [TK]D-Fender: I checked, none of the big distributers does them here, not even ingram |
| 17:14.07 | ManxPower | deeperror: there's a good chance someone quoted me for the docs. |
| 17:14.12 | kombi | anyways, got to find the damn cisco firmware somewhere.. |
| 17:14.14 | tutt9876 | have you any ringing tone until connected? |
| 17:14.18 | flujan | hi guys... |
| 17:14.23 | ManxPower | I know at least one of the dialplan examples I posted is on the Wiki |
| 17:14.33 | flujan | I need to block collect calls on my asterisk box. |
| 17:14.34 | tutt9876 | Where can i see the bugs of 1.4.2 ? |
| 17:14.38 | flujan | i found this information: http://www.nanpa.com/number_resource_info/ani_ii_assignments.html |
| 17:14.39 | Qwell | tutt9876: upgrade |
| 17:14.40 | deeperror | the caller doesn't hear anything until the call goes to voicemail or connected |
| 17:14.45 | ManxPower | kombi: You bought Cisco, you will have to deal with things the Cisco Way and that means waiting for your firmware |
| 17:15.08 | kombi | ManxPower: I'm so psyched..;) |
| 17:15.09 | tutt9876 | Qwell: to upgrade I need to install again? |
| 17:15.16 | tutt9876 | Where can i see the bugs of 1.4.2 ? |
| 17:15.34 | ManxPower | tutt9876: in the 1.4.4 and 1.4.;3 changelog |
| 17:15.41 | ManxPower | or do you mean open bugs? |
| 17:15.46 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 17:16.38 | ManxPower | deeperror: try 1.2.15, that is what I use on my servers. |
| 17:17.02 | ManxPower | I've not tried 1.2.16, but 1.2.17 and 1.2.18 both crash at least once per day on my servers |
| 17:17.57 | deeperror | ManxPower: do i also need to download the same versions of zap and libpri? |
| 17:18.15 | deeperror | recompile them all ? |
| 17:18.15 | ManxPower | deeperror: I doun't think so, but it would not hurt to do so. |
| 17:18.27 | ManxPower | deeperror: it's not THAT hard to do. |
| 17:18.48 | deeperror | yea i know |
| 17:18.51 | deeperror | not an issue here |
| 17:19.01 | deeperror | will do it and see if that fixes the problem |
| 17:19.10 | ManxPower | deeperror: I can only say what works for ME. |
| 17:19.19 | deeperror | its just for my home system |
| 17:19.23 | ManxPower | your enviroment might trigger bugs in 1.2.15 that I do not have. |
| 17:19.42 | *** join/#asterisk tonycarstens (n=oper@206.135.21.162) |
| 17:21.06 | *** join/#asterisk ssokol (n=ssokol@69-94-196-106.biltmorecomm.com) |
| 17:21.12 | flujan | but dunno how to collect this code from the calls... |
| 17:23.11 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 17:24.20 | *** join/#asterisk saftsack (n=oliver@p54a7e6f2.dip.t-dialin.net) |
| 17:24.55 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
| 17:24.57 | tonycarstens | can anyone help me with some zap problems |
| 17:26.33 | ManxPower | tonycarstens: ask your question |
| 17:27.29 | tonycarstens | well i tried to configure * so that if line 1 is busy it will auto jump to 2 and 3 then 4 |
| 17:27.40 | tonycarstens | and now i am getting a chan_zap.c error |
| 17:27.50 | tonycarstens | stating that the device or resource is busy |
| 17:28.13 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 17:28.34 | deeperror | pastebin extensions.conf |
| 17:28.40 | tonycarstens | ok |
| 17:28.42 | tutt9876 | sorry to bother you but I can't make it out to have a ringing when dialing out |
| 17:29.07 | tutt9876 | indications.conf r option bug ? |
| 17:29.56 | tutt9876 | indications.conf bad config, r option , bug in 1.4.2? |
| 17:30.06 | [TK]D-Fender | tutt9876: You shouldn't have to sue that option. it means you aren't being passed back proper progress indications |
| 17:30.11 | [TK]D-Fender | use* |
| 17:30.22 | tonycarstens | http://www.pastebin.ca/502902 |
| 17:30.30 | tonycarstens | i also included zapata.conf in there |
| 17:30.44 | *** join/#asterisk |Tiger| (n=Tiger@213.201.58.8) |
| 17:31.23 | tutt9876 | [TK]D-Fender>: i have tried without r option but same result |
| 17:31.53 | |Tiger| | can some one help me i get "call failed Service unavailble" but its see to be like im connectet whit my server or? |
| 17:32.53 | flujan | guys, It is possible to block incoming collect calls on asterisk? |
| 17:33.18 | tutt9876 | <PROTECTED> |
| 17:33.19 | flujan | I found a page that says to collect the ANI II DIGITS |
| 17:33.29 | flujan | but how can I get this digit? |
| 17:33.44 | tutt9876 | [TK]D-Fender>: use * ? |
| 17:33.54 | |Tiger| | i did startet whit asterisk -gvvvc |
| 17:34.20 | [TK]D-Fender | tutt9876: Corrected typo |
| 17:34.27 | deeperror | tonycarstens: is the issue on outbound calls? |
| 17:34.37 | tonycarstens | both now |
| 17:34.43 | tonycarstens | can't call in/out |
| 17:34.51 | tutt9876 | [TK]D-Fender>: so which option in the dial command ? |
| 17:35.08 | deeperror | you can't call between sip - sip? |
| 17:35.25 | tonycarstens | yes |
| 17:35.27 | docelmo | say whats the command to pull out headers from a sip packet? |
| 17:35.33 | [TK]D-Fender | tutt9876: Shouldn't need anything. |
| 17:35.33 | docelmo | or application? |
| 17:35.42 | tonycarstens | i can dial through sip |
| 17:35.52 | tutt9876 | [TK]D-Fender>: except I have no ringing tone |
| 17:36.12 | docelmo | nevermind.. its a function.. Sorry |
| 17:36.23 | vAd0r | does anyone here have a pix 501 that they are able to connect through to your asterisk server |
| 17:36.39 | vAd0r | maybe my pix is messing up my authentication |
| 17:36.40 | [TK]D-Fender | tutt9876: Externally or between 2 registered local phones? |
| 17:36.51 | tutt9876 | externally |
| 17:37.21 | [TK]D-Fender | vAd0r: PIX is one of the nastiest firewalls to try to get SIP working through. Go check it out ont he WIKI |
| 17:37.29 | [TK]D-Fender | tutt9876: internally is ok? |
| 17:37.47 | deeperror | tony: are you connected up to your outbound provider? |
| 17:37.50 | vAd0r | im about to go insane because i think it is setup right |
| 17:38.03 | vAd0r | about to download pfsense |
| 17:38.05 | tonycarstens | i have a regular pstn line connected to channel one |
| 17:38.22 | tonycarstens | will have 4 lines pluged in once deployed |
| 17:39.16 | deeperror | tony: so in your sip.conf what is the context setup for your users? is this [sip]? |
| 17:39.51 | Katty | so |
| 17:39.53 | Katty | i wanna hire someone |
| 17:39.56 | Katty | to write me some software |
| 17:39.59 | tonycarstens | deeperror: yes |
| 17:40.01 | Katty | so dumb people can use asterisk too! |
| 17:40.06 | Mercestes | What kind of software?? |
| 17:40.06 | Katty | mainly, our 50 year old recpetionist |
| 17:40.21 | Mercestes | kill the receptionist. get a new one. |
| 17:40.28 | Katty | yeah, but our clients are dumb too |
| 17:40.32 | Mercestes | what do you want your receptionist to do?? |
| 17:40.34 | Katty | and i can't fire them |
| 17:40.57 | [TK]D-Fender | Katty: You can however drug their coffee.... |
| 17:41.06 | nahirean | wish someone would drug mine |
| 17:41.09 | tutt9876 | [TK]D-Fender> yes when dialing a sip address I have a ringing tone |
| 17:41.41 | [TK]D-Fender | tutt9876: Then your provider is not sending back the proper progress. Ask them about it |
| 17:42.02 | Katty | [TK]D-Fender: oooh, i do like that approach. |
| 17:42.24 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 17:42.26 | tutt9876 | [TK]D-Fender> is it a codec question? |
| 17:42.54 | [TK]D-Fender | tutt9876: No, its a SIP progress code issue |
| 17:43.31 | tutt9876 | [TK]D-Fender>thanks very much will ask my provider |
| 17:43.33 | *** part/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 17:43.45 | *** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net) |
| 17:44.18 | tonycarstens | deeperror: something went wrong with the zap configuration |
| 17:44.33 | tonycarstens | deeperror: i reloaded them and it is working |
| 17:44.48 | *** join/#asterisk candyman50 (n=mdecandi@pool69-59-255-25.kewr1.s.vonagenetworks.net) |
| 17:45.08 | ManxPower | My questions about TeX formatted Docs was mostly answered. |
| 17:46.24 | deeperror | tony: working now? |
| 17:46.24 | candyman50 | anyone know how to get the whole channel name out of "show channels verbose"? |
| 17:46.24 | [TK]D-Fender | tonycarstens: You are dialing out Zap/g1 but did not define any of your zap channels as belonging to that group (or any other for that matter). That is your problem. |
| 17:46.27 | candyman50 | i.e. The cli cuts the name... how do I programatically get the whole thing? |
| 17:46.39 | [TK]D-Fender | tonycarstens: And jsut a side note : your dialplan is hugely redundent and could be reduced to at least 25% of its current size. |
| 17:47.05 | tonycarstens | yeah my programming skills are weak |
| 17:47.07 | Qwell | candyman50: core show channels concise |
| 17:47.08 | [TK]D-Fender | ManxPower: So have they indeed moved to TeX only? |
| 17:49.17 | tonycarstens | [TK]D-Fender: anywhere you could point me to reduce it? |
| 17:49.24 | |Tiger| | i get 2 erro line shen i tryede to start asterisk -gvvvc "May 22 19:07:02 WARNING[17796]: config.c:502 process_text_line: parse error: No" and secound one " WARNING[17796]: app_voicemail.c:6356 load_config: Failed to load" |
| 17:49.43 | bill4242 | In zapata.conf can I define different analog ports to different groups? and would it all be done in the same "context" under [trunkgroups] or do i create a new one like [group2] ? |
| 17:49.48 | candyman50 | Qwell: thanks |
| 17:49.57 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:51.04 | bill4242 | or would it look something like, signalling = fxo_ks / group=2 / channel => 1-4 |
| 17:51.07 | candyman50 | Qwell: is there a time field in the output? |
| 17:51.53 | bill4242 | and under that signalling=fxs_ks / group=1 / channel => 1-4 ... |
| 17:51.53 | bill4242 | [macro-stdexten]; |
| 17:52.00 | [TK]D-Fender | tonycarstens: First read up on macro's for all those extens that dial your phones. then learn to use "include => [context]" to give your IVR's access to your internal extensions without those Goto's (ICK!). Then realize that all of your zap lines lead to seperate yet functionally identical IVR's. |
| 17:54.31 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 17:55.23 | poppo | I have and asterisk box and recently got a 1800 number with voipstreet, I have outgoing working with voipstreet and that works fine i would like to know how i can setup my 1888 number to foward to a number |
| 17:55.37 | kombi | I never dared to ask, do you killall -1 asterisk after every conf change? |
| 17:55.50 | poppo | when i dial the number i get a request 'number@default' does not exist |
| 17:55.56 | poppo | on the console |
| 17:58.09 | deeperror | poppo: the inbound is point to your default context...you need to define an inbound context for the 888 number or put some type of logic in the default context to handle this |
| 17:58.32 | poppo | deeperror: ok were do i do that |
| 17:59.08 | poppo | i have and context in the extension.conf |
| 18:01.50 | poppo | Ok i think i figure it out now i am getting chan_iax2.c:6874 socket_process: No best format in 0xe000??? |
| 18:02.32 | *** join/#asterisk Mdsp (n=tradeshi@mail.tradeshield.co.za) |
| 18:02.36 | Mdsp | Hello |
| 18:02.48 | Mdsp | Has anybody ever configured a TE410P card before ? |
| 18:03.09 | *** join/#asterisk edguy3 (n=edguy3@69-94-196-221.biltmorecomm.com) |
| 18:03.12 | b11d|bbl | no.. no one has EVER done that. |
| 18:03.13 | b11d|bbl | EVER. |
| 18:03.30 | Mdsp | i take it u havent ;) |
| 18:03.31 | b11d|bbl | Not even the guys who made it.. |
| 18:03.34 | b11d|bbl | :) |
| 18:04.44 | [TK]D-Fender | Mdsp: Now's the point where you should realize that a more useful specific quesiton might get you what you really want to know. |
| 18:05.03 | b11d|bbl | thanks for bringing him up to speed TK |
| 18:05.04 | b11d|bbl | :) |
| 18:05.25 | Mdsp | well i've got the te410p connected to a pri line in south africa (normally uk / de standards) but when im running genzaptelconf my spans are all commented out |
| 18:05.58 | [TK]D-Fender | Mdsp: And if you try configuring it by hand? |
| 18:06.01 | Mdsp | also have a 2400 wildcard (FXOS) but thats fine, in the same box |
| 18:06.15 | Mdsp | well were do i start.. (n00bie) |
| 18:06.28 | [TK]D-Fender | Mdsp: www.voip-info.org |
| 18:06.56 | [TK]D-Fender | Mdsp: go lookup "config zaptel.conf" and "config zapata.conf" |
| 18:06.57 | bill4242 | I can't call out on my TDM800P using Asterisk 1.4.4.. All I get is a busy when i do "zap show channel 5" which is the channel for my FXO line it shows the hookstate: Offhook when the line is connected. When it is not connected the hookstate changes to Onhook.. |
| 18:07.07 | *** join/#asterisk sysreq (n=sysreq@H144.C72.B0.tor.eicat.ca) |
| 18:07.30 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 18:08.00 | bill4242 | but obviously with no line connected no outgoing calls go through.. |
| 18:08.06 | poppo | OK i am getting socket_process: Rejected connect attempt from *.*74.24, requested/capability 0x4/0xf804 incompatible with our capability 0xe703. |
| 18:08.13 | poppo | when i call my did |
| 18:08.28 | kombi | sorry, got knocked off by x-lite.. how are configuration changes activated? restart each time? |
| 18:08.39 | [TK]D-Fender | kombi: To answer your earlier question, most changes except Zatel related can be put into effect with a "reload". The rest "restart gracefully" or "restart now" (if I'm feeling hostile/impatient) |
| 18:09.29 | kombi | [TK]D-Fender: thanks! that is from whithin the cli I assume |
| 18:09.31 | [TK]D-Fender | poppo: Looks like incompatible codecs |
| 18:09.42 | [TK]D-Fender | kombi: Yes |
| 18:10.00 | poppo | is that something i dont have installed? |
| 18:10.50 | [TK]D-Fender | poppo: Thats you misconfiguring your sip.conf or them using codecs that your setup doesn't support |
| 18:11.02 | [TK]D-Fender | poppo: Likely the former |
| 18:11.11 | poppo | i am not using sip only aix |
| 18:11.14 | |Tiger| | i realy dont andrestand x-lite say its connected but when i trying to call from 6001 to 6002 i get "call failed Service unavailable" |
| 18:11.14 | kvidell | oh I hate highlighting on "kel" |
| 18:11.21 | kvidell | everytime somethign is "likely" I flash |
| 18:11.32 | [TK]D-Fender | poppo: Whatever, transpose the two, but your issue is the same |
| 18:11.41 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-164-170-51.hvc.res.rr.com) |
| 18:12.02 | poppo | is the a command to see what codec i have installed |
| 18:12.03 | flujan | hi guys, How can I detect and block collect calls using asterisk? |
| 18:12.21 | flujan | I found some information about using ANI digits |
| 18:12.24 | [TK]D-Fender | Kelloggs Frosted Flakes .... they're gggrrrrrrrreat! |
| 18:12.42 | [TK]D-Fender | :D |
| 18:12.42 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 18:12.42 | flujan | I am using a PRI/ISDN TE406P card |
| 18:12.57 | kvidell | I should tell you everything I highlight on to see how creative you can get |
| 18:12.57 | poppo | found the isssue show g729 |
| 18:12.57 | poppo | thats what i am missing |
| 18:12.57 | [TK]D-Fender | poppo: its not what you have INSTALLED, its what you have ENABLED. Go get a clue, check your configs, then come back |
| 18:13.09 | kvidell | (kel kvidell thumper plant) |
| 18:13.12 | kvidell | go! |
| 18:13.23 | b11d|bbl | Go Gadget Go! |
| 18:13.37 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 18:14.18 | [TK]D-Fender | kvidell: |
| 18:14.24 | b11d|bbl | I always wanted that Computer Book that Penny owned. |
| 18:14.31 | flujan | any ideas guys? to block collect calls in asterisk? |
| 18:14.33 | flujan | :( |
| 18:14.35 | b11d|bbl | It could somehow sieze control of helicopters in the air and stuff :) |
| 18:14.45 | tuan_modulis | im trying to find the wiki page that shows how make the system perform outgoing calls automatically... forgot the term... what is it? |
| 18:14.45 | [TK]D-Fender | kvidell: "plant" is too easy, and the only use of "thumper" would be Bambi referrences :) |
| 18:14.53 | b11d|bbl | dont you do that at the telco flujan? |
| 18:14.54 | kvidell | lol |
| 18:14.56 | tuan_modulis | like call placing or something |
| 18:14.58 | b11d|bbl | dont you just say to them "no collect calls on these lines" / |
| 18:14.59 | b11d|bbl | ? |
| 18:15.09 | [TK]D-Fender | flujan: CallerID is all you've got. If that isn't enough, TFB |
| 18:15.36 | b11d|bbl | tuan_modulis.. call files? |
| 18:15.40 | tuan_modulis | that's it! |
| 18:15.47 | b11d|bbl | they rock :) |
| 18:15.52 | tuan_modulis | thx |
| 18:16.08 | [TK]D-Fender | AMI Originate > Call Files. |
| 18:16.16 | b11d|bbl | pleasure++ |
| 18:16.17 | *** join/#asterisk Massimiliano (n=administ@81-208-83-242.fastres.net) |
| 18:16.34 | tuan_modulis | Fender, u might be onto something there... .checking it out |
| 18:16.43 | Massimiliano | hi all |
| 18:16.46 | b11d|bbl | HI |
| 18:17.16 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 18:17.20 | flujan | b11d|bbl, yeap... but I need to block the incoming collect calls depending of the location... for this, I need to know IF the call is a collect call AND IF it is a collect CALL accept it depending on the caller id... |
| 18:17.24 | flujan | this is the problema... |
| 18:17.34 | b11d|bbl | ohh |
| 18:17.37 | b11d|bbl | good luck :() |
| 18:17.38 | Massimiliano | anyone know iaxclient? |
| 18:19.09 | flujan | [TK]D-Fender, there is no way to detect a incoming collect call without the callerid? |
| 18:19.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:20.21 | anonymouz666 | flash() |
| 18:20.22 | anonymouz666 | ? |
| 18:21.23 | bill4242 | I can't call out on my TDM800P using Asterisk 1.4.4.. All I get is a busy signal when calling out. When i do "zap show channel 5", which is the channel for my FXO line, from the CLI it shows the hookstate: Offhook when the line is plugged in. When it is not plugged in the hookstate changes to Onhook and calls attempt to go through but obviously go no where... |
| 18:21.54 | [TK]D-Fender | flujan: I'll give you a hint when you can tell me how you expect to know that its a collect call within the dialplan. |
| 18:22.15 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:22.55 | [TK]D-Fender | bill4242: pastebin your configs & dialplan and the CLI output of a failed call. If you want us to help you we shouldn't even have to ASK for this. |
| 18:23.55 | b11d|bbl | hey flujan.. have you made a collect call into your system just to see how the system handles it? and myabe gather some info? |
| 18:25.01 | bill4242 | D-Fender: well i'm fairly new to this channel so i guess i would have to be asked.... doing that now.. |
| 18:27.49 | flujan | [TK]D-Fender, the telco said they cannot send me calls with any other digits with the caller id |
| 18:28.04 | flujan | b11d|bbl, yeap... the call enters normally no additional info |
| 18:28.13 | flujan | the telco answers the call and redirects it to me |
| 18:28.25 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 18:28.38 | flujan | so my business rules depends to know the callerid and IF the call is a collect call... |
| 18:28.55 | [TK]D-Fender | flujan: That does not sound helpful. Perhaps you should add "load => chan_psychic.so" to modules.conf... |
| 18:29.02 | b11d|bbl | how was it handled prior to this? |
| 18:29.29 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:29.30 | flujan | b11d|bbl, the legacy pbx handled it... |
| 18:29.39 | b11d|bbl | yeah but in what way? |
| 18:29.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 18:29.41 | b11d|bbl | how exactly? |
| 18:29.56 | b11d|bbl | cant you "see" what it's doing? |
| 18:30.00 | Strom_M | flujan: you want something called "ANI II" |
| 18:30.30 | Strom_M | not "eigh en eye two" but "eigh en eye eye eye" |
| 18:31.11 | Strom_M | http://nanpa.com/number_resource_info/ani_ii_assignments.html |
| 18:31.12 | flujan | [TK]D-Fender, lol... It will be cool... This module will also discover why I am getting this errors: http://pastie.caboo.se/63613 |
| 18:31.27 | flujan | Strom_M, yeap... I read about it... but how can I get the value using asterisk? |
| 18:31.38 | Strom_M | your telco has to deliver it |
| 18:31.52 | flujan | Strom_M, http://www.nanpa.com/number_resource_info/ani_ii_assignments.html |
| 18:31.55 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:31.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:31.57 | phillipk | Can anyone look at a log excerpt for me and see if you can tell what causes my PRI to drop? http://pastebin.ca/502797 |
| 18:32.12 | Strom_M | flujan: thats the same URL I just pasted |
| 18:32.16 | flujan | Strom_M, according the them... They already gives me everything that I need to identify a collect call |
| 18:32.18 | flujan | ops |
| 18:32.38 | Strom_M | flujan: is it a PRI? |
| 18:32.40 | b11d|bbl | umm |
| 18:32.40 | b11d|bbl | chan_zap.c: No D-channels available! |
| 18:32.43 | b11d|bbl | would be my guess |
| 18:32.46 | flujan | Strom_M, for sure... I didn't see your pastie... I also found this information... Yes it is a PRI. |
| 18:33.09 | Strom_M | flujan: well then run PRI debug on inbound calls and see what happens |
| 18:33.10 | [TK]D-Fender | b11d|bbl: His PRI seems to reset spontaneously every >HR |
| 18:33.15 | [TK]D-Fender | <HR * |
| 18:33.57 | b11d|bbl | crazy.. |
| 18:34.03 | flujan | Strom_M, I dunno how Strom_M OK... I can get this value from the Dialplan? |
| 18:34.06 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:34.07 | b11d|bbl | we had a non-PRI t1 doing that, but it was the NIU card.. |
| 18:34.17 | b11d|bbl | which isnt CPE. |
| 18:34.26 | b11d|bbl | at least, not here. |
| 18:34.29 | Strom_M | flujan: at the CLI: "pri intense debug span 1" |
| 18:34.33 | Strom_M | or whatever your span is |
| 18:36.11 | rikstah | tzafrir_laptop, ping |
| 18:36.45 | flujan | Strom_M, thanks I will check it... Just cannot check it now since the server is online and I receive a lot of calls |
| 18:36.49 | flujan | Strom_M, :( |
| 18:36.56 | Strom_M | oops |
| 18:37.04 | *** join/#asterisk fd__ (n=fd@finnishplatoon.org) |
| 18:37.36 | flujan | Strom_M, thanks anyway. :) |
| 18:37.47 | phillipk | b11d|bbl: yeah, if you check down toward the bottom of the pastebin, the D-channel comes back up. That's what I'm trying to fix. |
| 18:37.56 | flujan | hey [TK]D-Fender ... do you have a clue about this error? http://pastie.caboo.se/63613 |
| 18:39.04 | tzafrir_laptop | rikstah, pong |
| 18:39.08 | b11d|bbl | yeah.. still.. |
| 18:39.11 | b11d|bbl | its suspect. |
| 18:39.12 | [TK]D-Fender | flujan: yes... you've been a VERY bad boy! |
| 18:39.12 | b11d|bbl | :) |
| 18:39.22 | b11d|bbl | BAD! BAD! BAD! boy! |
| 18:39.37 | flujan | b11d|bbl, [TK]D-Fender ;( |
| 18:39.42 | b11d|bbl | haha.. |
| 18:39.52 | Strom_M | flujan: what version of asterisk are you running? |
| 18:39.53 | b11d|bbl | well im off to northern ontario (ahh!! sweet relief) for a few days.. ttyl all |
| 18:39.59 | flujan | Strom_M, 1.2.18 |
| 18:40.01 | [TK]D-Fender | <- Zen master of the blatantly obvious |
| 18:40.42 | rikstah | tzafrir_laptop, ,re: your post to *-users about the firefox greasemonkey script. You make a good point. If this is being used in an untrusted source, add a password check to the .php script and then pass the password as a param from the .user.js GM script. |
| 18:40.44 | flujan | this starts to happen after i switch my extensions to SIP. |
| 18:40.48 | rikstah | tzafrir_laptop, unless you have some better suggestion? :) |
| 18:42.12 | flujan | b11d|bbl, did you already see those erros about SIP and ACK? |
| 18:42.26 | b11d|bbl | i left.. see above. |
| 18:42.45 | flujan | b11d|bbl, ok |
| 18:42.51 | b11d|bbl | :) |
| 18:43.24 | *** join/#asterisk ploieel (n=manni@Fb20f.f.ppp-pool.de) |
| 18:43.48 | `Sean | Anyone in here got TollFree termination that allows you to set youre own callerID? |
| 18:49.48 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 18:50.08 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:50.25 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 18:50.31 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 18:50.41 | *** part/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 18:50.54 | *** part/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 18:53.00 | flujan | ping oej |
| 18:54.03 | *** join/#asterisk Remenic (n=Richard@cc1222307-a.frane1.fr.home.nl) |
| 18:54.03 | Remenic | hi |
| 18:54.17 | oej | pong |
| 18:54.23 | Remenic | is it possible to play a sound, after a Dial()? Like a voice saying "this call was powered by asterisk" |
| 18:54.38 | Remenic | I know it sounds stupid, but I need it to test reinvite on a client :) |
| 18:54.50 | [TK]D-Fender | Remenic: "show application dial" |
| 18:55.14 | flujan | hi oej |
| 18:56.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:56.36 | Remenic | ah, g might help me |
| 18:57.18 | flujan | oej, I am having some sip errors could you please have a look at it? |
| 18:59.13 | bill4242 | <PROTECTED> |
| 19:01.42 | bill4242 | oops...i forgot to post the CLI output from the analog extension. its essentially the same.. |
| 19:01.42 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 19:01.43 | Strom_M | is it a busy signal or a reorder tone? |
| 19:01.43 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:02.09 | bill4242 | used to be busy singal.. now its silence.. |
| 19:02.27 | bill4242 | the output from the analog line is a little different.. i'll postbin it.. |
| 19:02.34 | bkw_ | postbin? |
| 19:02.35 | bkw_ | haha |
| 19:03.02 | Strom_M | well, a busy signal would indicate that the called line is actually busy; reorder ("fast busy") indicates a problem with call setup |
| 19:03.03 | bkw_ | bill4242, just giving you a hard time |
| 19:05.56 | bill4242 | strom - yeah, and the strange thing is ... when the line is plugged in asterisk console reports it as off hook.. here's the output from the CLI for an analog phone - http://pastebin.ca/503094#comments |
| 19:05.56 | bill4242 | see the comments.. |
| 19:06.14 | bill4242 | saying called, but i hear no ringing, nothing just silence now.. |
| 19:06.40 | bill4242 | SIP gives me the fast busy. |
| 19:06.56 | xpot | anyone know of a good solution for load-balancing mulitple asterisk servers? |
| 19:06.56 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-242.ph.ph.cox.net) |
| 19:09.20 | *** join/#asterisk dwmw2_gone (i=ctrlprox@81.187.2.161) |
| 19:13.20 | fd__ | hulloh, if anyone can shed some light on a trunk thingy i'm battling with, please drop an email or msg me up |
| 19:13.20 | fd__ | i assure you this is not advertisement or spam or viruses: http://gle.fi/setup.html |
| 19:14.28 | bill4242 | bkw_: pastebin .. blah.. my heads spinning.. it took me awhile to realize what you were pointing out.. lol |
| 19:14.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:14.50 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 19:16.56 | [TK]D-Fender | bill4242: Looks like a codec error on your SIP device's side. Change codecs and retest |
| 19:18.54 | [TK]D-Fender | bkw_: postbin ... where you realize you should have gone this first time ;) |
| 19:19.14 | Katty | [TK]D-Fender: you any good at looking at a tcpdump? |
| 19:19.28 | [TK]D-Fender | Katty: I have eyes... but whats your actual problem? |
| 19:19.29 | Katty | [TK]D-Fender: my telco tells me we're putting out a ton of SYN requests. |
| 19:19.36 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 19:19.42 | Katty | [TK]D-Fender: so i did a packet capture for 5 minutes, and put a tcp syn filter on it |
| 19:19.58 | [TK]D-Fender | Katty: I'm betting either no return path or someone inside your LAN is DoS'ing someone |
| 19:20.00 | Katty | [TK]D-Fender: from the machine who i think is the culprit, after comparing their log to my outgoing firewall. |
| 19:20.24 | [TK]D-Fender | Katty: if you suspect it, pullt he plug outright and see what happens |
| 19:20.34 | Katty | [TK]D-Fender: hmm. |
| 19:20.36 | Katty | [TK]D-Fender: good point. |
| 19:20.38 | Katty | [TK]D-Fender: however! |
| 19:20.44 | Katty | [TK]D-Fender: i'm noting a steady progression... |
| 19:20.50 | Katty | from 5419 on up |
| 19:21.01 | Katty | [TK]D-Fender: i don't know enough about what i'm looking at ti tell me if it's a port number |
| 19:21.06 | Katty | [TK]D-Fender: also, this an exchange server. |
| 19:21.17 | Katty | [TK]D-Fender: so seeing the dest port of 25 isnt' really useful here |
| 19:21.50 | funxion | I'm trying to send a call through a PRI intoa a cisco 3660 and out to my asterisk box via sip I have cisco in sip.conf but asterisk keeps rejecting the call from the cisco I think because of the callerid can someone help? |
| 19:25.19 | funxion | please look at my dial peer and sip.conf entries http://pastebin.ca/503159 |
| 19:26.22 | [TK]D-Fender | funxion: don't name your peer entry like an IP, thats just TROUBLE |
| 19:26.34 | funxion | ok |
| 19:26.54 | [TK]D-Fender | funxion: and that should be "insecure=very" |
| 19:27.21 | *** join/#asterisk Cresl1n (n=matt@69-94-196-9.biltmorecomm.com) |
| 19:27.21 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:28.14 | funxion | lol\ |
| 19:28.16 | Cresl1n | phillipk: what version of asterisk are you running? |
| 19:28.16 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-97.biltmorecomm.com) |
| 19:28.16 | funxion | oops |
| 19:28.33 | Cresl1n | and what does asterisk say when the span drops? |
| 19:29.15 | ManxPower | Cresl1n: "Help! The line has fallen and I can't get up!"? |
| 19:29.27 | phillipk | Cresl1n: 1.2.14 |
| 19:29.52 | Cresl1n | what is the message when the span drops? |
| 19:30.29 | phillipk | Primary D-Channel on span 1 down |
| 19:30.59 | Cresl1n | can you get anything more verbose than that? |
| 19:31.14 | Cresl1n | like enabling verbose and debug output to the console |
| 19:31.21 | Strom_M | the telco calls up and says "boners" |
| 19:31.45 | ManxPower | phillipk: no yellow alarms, no red alarms? |
| 19:32.01 | [TK]D-Fender | ManxPower: ... Who's on first? :) |
| 19:32.05 | phillipk | no alarms |
| 19:32.46 | ManxPower | phillipk: I have seen this when I had a significant mismatch between the zaptel verison and the asterisk verison |
| 19:33.37 | phillipk | I've got verbosity set to 20 and intense debug on. I pastebinned the output earlier. |
| 19:33.38 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 19:34.04 | funxion | thnx TK |
| 19:34.07 | funxion | that did it |
| 19:34.08 | ManxPower | phillipk: look at /etc/asterisk/logger.conf to see what the "console" is set for. |
| 19:34.42 | Cresl1n | phillipk: not only intense debug, but also add the debug option in logger.conf to the console section |
| 19:34.44 | Cresl1n | thanks ManxPower |
| 19:35.00 | ManxPower | <-- smarter than he looks. |
| 19:35.10 | phillipk | ok, I have that set now. |
| 19:35.26 | ManxPower | Cresl1n: so what advantages does HPEC 9.x have over 8.x? |
| 19:36.13 | Cresl1n | Various algorithm improvements |
| 19:36.13 | coppice | 1.x? |
| 19:36.13 | Cresl1n | fixing a few bugs |
| 19:36.13 | Cresl1n | it's actually a pretty good sized list |
| 19:36.13 | *** join/#asterisk ta^3 (n=tacvbo@189.146.195.139) |
| 19:36.13 | phillipk | how do I check my zaptel version? |
| 19:36.16 | Cresl1n | I wasn't even sure the best way to summarize it all |
| 19:36.20 | Cresl1n | I got a pretty verbose changelog |
| 19:36.41 | ManxPower | phillipk: as far as I know, you can't. |
| 19:36.54 | ManxPower | unless you have the source from which you built it. |
| 19:37.06 | ManxPower | Cresl1n: would know for sure. |
| 19:37.07 | brodiem | Anyone use Aastra (notably 480i) behind NAT? |
| 19:37.22 | *** join/#asterisk kram (n=markster@pdpc/sponsor/digium/kram) |
| 19:37.22 | *** mode/#asterisk [+o kram] by ChanServ |
| 19:37.26 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 19:37.30 | Cresl1n | kram: /1111\\ |
| 19:37.52 | ManxPower | GuruJee: I doubt anyone can help you. |
| 19:38.03 | ManxPower | since your system was originally AMP/FReePBX |
| 19:38.07 | kram | maybe |
| 19:38.19 | tonycarstens | does anyone know the variable that is made from Waitexten? |
| 19:38.26 | russellb | it's .... kra! |
| 19:38.29 | russellb | er. kram, even. |
| 19:38.32 | ManxPower | nobody wants to spend the day or so trying to figure out what that bastardized config is doing before being able to help you. |
| 19:38.45 | ManxPower | Are you sure it's not kram's cat logging into IRC as him? |
| 19:39.02 | ManxPower | tonycarstens: that would be EXTEN |
| 19:39.02 | *** join/#asterisk mrdigital (n=mrdigita@207-172-228-21.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 19:39.12 | tonycarstens | thank you |
| 19:39.31 | mrdigital | anyone use tun in linux? |
| 19:39.35 | ManxPower | waitexten will jump to whatever extension you dial. EXTEN always contains the currently executing extension |
| 19:39.55 | [TK]D-Fender | mrdigital: Clearly as its the basis of OpenVPN |
| 19:40.07 | ManxPower | This is usually the digits you dialed, but it does not have to be when using gotos, etc |
| 19:40.45 | tonycarstens | so can i use waitexten |
| 19:40.53 | tonycarstens | then use {EXTEN} in a goto |
| 19:41.14 | [TK]D-Fender | tonycarstens: Why? |
| 19:41.23 | mrdigital | tun was working fine for me then all of a sudden i got 22 15:40:21.272 [ 0] [ 5053] tap: connect() failed 2 (No such file or directory) |
| 19:41.25 | mrdigital | <PROTECTED> |
| 19:41.37 | mrdigital | i use vpn to remotely connect to asterisk behind a nat |
| 19:41.43 | [TK]D-Fender | mrdigital: Perhaps you should ask in ##linux |
| 19:41.53 | Cresl1n | or #2,000 |
| 19:42.19 | tonycarstens | TK: i'm trying to get rid of all those goto's in my dialplan so i was going to have just one "exten => s,n,Goto(sip,${EXTEN},1) |
| 19:42.41 | [TK]D-Fender | tonycarstens: ICK |
| 19:42.51 | tonycarstens | bad ida |
| 19:42.52 | tonycarstens | idea |
| 19:42.58 | [TK]D-Fender | tonycarstens: Dear God, pastebin the whole thing and I'll give you a head start... |
| 19:43.18 | [TK]D-Fender | tonycarstens: VERY. you dont. do. gotos. like. that! |
| 19:44.27 | tonycarstens | http://www.pastebin.ca/503202 |
| 19:44.50 | ManxPower | tonycarstens: EXTEN contains the currently executing extension. In your example that would be extension "s". |
| 19:44.54 | *** part/#asterisk dr_decimal (n=stefan@c-68-45-144-101.hsd1.pa.comcast.net) |
| 19:45.01 | ManxPower | So you would be going to context "sip", extension "s", priority "1". |
| 19:45.33 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
| 19:45.40 | ManxPower | If the dialplan is on extension "s" then IT DOES NOT KNOW WHAT NUMBER WAS DIALED |
| 19:45.47 | tonycarstens | ok, i thought that it would go to whatever extension they dialed during the waitexten |
| 19:46.19 | ManxPower | tonycarstens: No, WaitExten will send the call to whatever extension was dialed. |
| 19:46.49 | ManxPower | Extension s is only ever matched when there are no dialed digits, like on an FXO channel that does not have DID |
| 19:46.59 | tonycarstens | so i could just have a simple waitexten() and whatever the user dials it would call that extension? |
| 19:47.38 | [TK]D-Fender | tonycarstens: http://www.pastebin.ca/503208 |
| 19:47.47 | [TK]D-Fender | tonycarstens: Complete replacement. |
| 19:47.54 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:48.09 | ManxPower | tonycarstens: that is the way waitexten works, so you can see by the output of "show application waitexten" in the asterisk CLI |
| 19:48.10 | [TK]D-Fender | ManxPower: that (n) priority will never get called anyways :0 |
| 19:48.43 | ManxPower | [TK]D-Fender: I didn't even look at his paste. It is obvious he has some serious misunderstandings about stuff. |
| 19:48.55 | [TK]D-Fender | ManxPower: yup |
| 19:50.36 | *** join/#asterisk WindBack (n=jorge@host44.200-117-61.telecom.net.ar) |
| 19:51.44 | *** join/#asterisk bbryant (n=brett@69-94-196-94.biltmorecomm.com) |
| 19:53.30 | WindBack | Somebody can recomendme a good softphone for Linux Gnome who can handle sip protocol |
| 19:53.33 | WindBack | ?? |
| 19:53.40 | [TK]D-Fender | WindBack: Ekiga |
| 19:54.51 | WindBack | [TK]D-Fender, yes, I have it installed, but it don't recognize the press of buttom on interactives menus |
| 19:54.53 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM] |
| 19:55.02 | [TK]D-Fender | WindBack: Fix your DTMF mode then. |
| 19:55.21 | [TK]D-Fender | WindBack: Should set to RFC2833 on both sides |
| 19:57.25 | WindBack | [TK]D-Fender, thank you |
| 19:57.34 | justdave | hmm, so I got my sip registration working. |
| 19:58.03 | justdave | best I can tell, at some point in the recent security updates to the 1.2 branch, asterisk started getting pickier about where the register=> line could be in the sip.conf file |
| 19:58.12 | justdave | apparently it used to work anywhere and not it has to be in the [general] section |
| 19:58.22 | WindBack | [TK]D-Fender, in the ekiga were I can change that?? There are any configuration file?? |
| 19:58.24 | justdave | s/not/now/ |
| 19:58.38 | *** join/#asterisk pawel (n=pawel@87.243.195.236) |
| 19:58.40 | justdave | heh jbot rocks |
| 19:58.44 | [TK]D-Fender | WindBack: go LOOK. |
| 19:58.56 | [TK]D-Fender | WindBack: and check your * SIP setup for that peer/user |
| 20:00.32 | *** join/#asterisk DeadYak (i=rene@newbabe.pobox.com) |
| 20:00.50 | DeadYak | what would cause cdr_addon_mysql to try to insert CDRs with a blank uniqueid? |
| 20:00.51 | WindBack | [TK]D-Fender, ok, I'm asking because I was searching before I asked you, but I didm?t find it. But.. ok, I'll continue searching |
| 20:01.05 | WindBack | didn't |
| 20:02.36 | WindBack | [TK]D-Fender, sorry, it was silly |
| 20:02.48 | WindBack | [TK]D-Fender, I found it |
| 20:02.57 | WindBack | [TK]D-Fender, thank you |
| 20:03.11 | [TK]D-Fender | tonycarstens: Here, final revision for you since I'm doing some favours.. http://www.pastebin.ca/503240 |
| 20:03.39 | [TK]D-Fender | tonycarstens: Shrunk to tiny proportions |
| 20:03.51 | *** join/#asterisk Assid (n=assid@59.165.14.35) |
| 20:04.50 | DeadYak | never mind, found the problem |
| 20:04.52 | *** part/#asterisk DeadYak (i=rene@newbabe.pobox.com) |
| 20:06.22 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 20:10.37 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 20:11.00 | Jingles | having a serious Asterisk issue, and need a hand. |
| 20:11.20 | Jingles | Getting 200 OK on REGISTER that is a register msgs every 30 seconds from one of my SIP providers. |
| 20:11.28 | Jingles | and it's causing all my other SIP connections to thrash. |
| 20:11.56 | Jingles | at this point, I'll take any advice. |
| 20:12.05 | Jingles | the SIP provider hasn't been helpful in the least. |
| 20:12.58 | johann8384 | may I ask what provider? |
| 20:13.02 | Jingles | Broadvoice |
| 20:13.16 | Jingles | they're claiming 'we dont' really support Asterisk' |
| 20:13.20 | johann8384 | I see |
| 20:13.41 | johann8384 | You aren't registering every 30 seconds are you? |
| 20:14.25 | Jingles | there's nothing in sip.conf (in either the register line or the later [broadvoice] entry to cause that' |
| 20:15.27 | johann8384 | <shameless plug>I'm not sure how to help you fix that right off but NetLogic would support you.</shamelessplug> :) |
| 20:16.00 | Jingles | really? |
| 20:17.46 | Assid | Jingles: whats your reigster line ? |
| 20:18.04 | Assid | or rather your refresh |
| 20:18.28 | johann8384 | 120 seconds iirc |
| 20:18.58 | Assid | and your defaultexpiry ? |
| 20:19.03 | johann8384 | defaultexpiry=120 (is the default value in Asterisk) |
| 20:19.23 | Assid | do you have a sip reload being run in cron ? |
| 20:19.33 | *** join/#asterisk saftsack (n=saftsack@pd9e04530.dip.t-dialin.net) |
| 20:19.46 | *** join/#asterisk dalfry (n=dalfry@c-67-189-95-238.hsd1.or.comcast.net) |
| 20:19.47 | johann8384 | heh, i read that as if Jingles: was asking me :) |
| 20:19.57 | rikstah | i dont really know why once every 30 seconds would be that much of a problem |
| 20:20.03 | rikstah | (even though it's not desired) |
| 20:20.11 | Jingles | I'll answer your questions in order |
| 20:20.12 | Assid | also check for registertimeout |
| 20:20.13 | rikstah | one of my peers sipgate does it |
| 20:20.41 | Jingles | I don't know where to find my 'refresh' value. |
| 20:20.55 | Assid | sip.conf |
| 20:20.59 | Assid | read the basic lines |
| 20:21.02 | Jingles | but both maxexpirey and defaultexpirey are 36000 |
| 20:21.15 | Assid | one of them says registertimeout |
| 20:21.15 | Assid | set that to 120 |
| 20:21.17 | Assid | that will give you 2 mins |
| 20:21.30 | Assid | thedefault is 20 seconds.. not 30 |
| 20:21.37 | Jingles | mine was set to 50. |
| 20:21.40 | tonycarstens | In order to test outbound calling to a zap group do i have to have all lines attatched for the searching to work? |
| 20:22.06 | Assid | okay then you should be fine.. unless your per peer config is overriding them |
| 20:22.11 | Assid | check the peer/context |
| 20:22.44 | Jingles | ok. there's no 'registertimeout' value in the peer section. |
| 20:22.59 | Assid | any kind of timeout/refresh |
| 20:23.04 | Jingles | no |
| 20:23.28 | Assid | dunno.. then |
| 20:23.37 | *** join/#asterisk cazze (n=pc@unaffiliated/kammicazze) |
| 20:23.50 | Jingles | *nods* I've got their engineers on the problem - but they don't seem inclined to help much, since I'm on an Asterisk system. |
| 20:24.34 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:24.51 | Assid | sorry mate.. today is just one of those days i dont have the drive but to sit and stare out of the window.. or into the screen |
| 20:25.01 | Jingles | hehehe |
| 20:32.19 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 20:33.01 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:35.33 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 20:37.28 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 20:37.30 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 20:37.40 | nahirean | omg |
| 20:37.47 | GuruJee | omg what? |
| 20:37.49 | GuruJee | nahirean |
| 20:37.57 | nahirean | stop spamming |
| 20:38.04 | GuruJee | i am not spamming |
| 20:38.14 | GuruJee | thatst the monty python in you tthats spamming |
| 20:38.21 | GuruJee | i am just in a desperate situation |
| 20:38.23 | nahirean | ... Right. |
| 20:38.25 | GuruJee | trying to find help |
| 20:38.35 | GuruJee | i have helped like 2 dozen people since yestarday on this chan |
| 20:38.42 | GuruJee | but no one has heard my cries |
| 20:39.03 | GuruJee | hey guys |
| 20:39.12 | GuruJee | any one know how to create dial plans? |
| 20:39.16 | GuruJee | its not even about dudni |
| 20:39.19 | GuruJee | dundi |
| 20:39.36 | GuruJee | its just creating a dialplan that will forward a call on iax2 trunk |
| 20:39.39 | GuruJee | now who can help me? |
| 20:40.26 | dalfry | GuruJee: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction |
| 20:40.35 | GuruJee | dude |
| 20:40.40 | *** join/#asterisk linmax (n=maxi@p549156d3.dip.t-dialin.net) |
| 20:40.42 | GuruJee | u shouldnt eat dal at all |
| 20:40.50 | GuruJee | thanks!! |
| 20:41.11 | dalfry | Example outgoing channel names: |
| 20:41.12 | dalfry | IAX/mark:asdf@myserver/6275@default ÃĒâŽâ Call to "myserver" using "mark" as username and "asdf" as password, and requesting extension 6275 in default context |
| 20:41.19 | kiscokid | can you describe the problem in 25 words or less? |
| 20:41.32 | GuruJee | dalfry: have u had a chance to look at my extensions_custom.conf files? |
| 20:41.49 | GuruJee | basically, i have all the internal extensions working |
| 20:41.56 | GuruJee | what isnt working is the outside calls to pstn |
| 20:42.00 | dalfry | nope. I am trying to fix a problem of my own. freakin format_mp3 is making asterisk segfault |
| 20:42.41 | GuruJee | so i guess it would be something like exten => _NXXNXXXXXX,1,Macro (dial, ${EXTEN}) |
| 20:42.42 | GuruJee | right? |
| 20:42.47 | dalfry | GuruJee: you should be able to trace the call coming in from the console |
| 20:42.53 | tuan_modulis | okay.... I did an unbelievaly stupid mistake |
| 20:42.59 | GuruJee | how do i trace it? |
| 20:43.04 | dalfry | asterisk -r |
| 20:43.06 | GuruJee | debug doesnt work |
| 20:43.10 | GuruJee | no, i dotn see anything at all |
| 20:43.11 | dalfry | and watch where the call goes |
| 20:43.13 | GuruJee | on asterisk -r |
| 20:43.18 | GuruJee | thats the problem |
| 20:43.20 | dalfry | increase debug level |
| 20:43.23 | tuan_modulis | anyone have a recommendation for a program to undelete files? |
| 20:43.30 | GuruJee | my phone goes to busy after 10 seconds but i dont see nothing |
| 20:44.53 | tuan_modulis | gah... for ext3, I have very little hope |
| 20:46.05 | GuruJee | dalfry: change the ext to *.gsm instead |
| 20:46.12 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:46.19 | GuruJee | whats the command and whats the debug level? |
| 20:47.00 | dalfry | change to gsm? |
| 20:47.00 | dalfry | set verbose X |
| 20:47.02 | GuruJee | ok |
| 20:47.02 | nahirean | change ext3 to gsm? hahaha |
| 20:47.02 | GuruJee | yeah |
| 20:47.09 | GuruJee | not ext3 dumbo |
| 20:47.14 | GuruJee | the mp3 extension to gsm |
| 20:47.19 | dalfry | right |
| 20:47.27 | GuruJee | i dotn know what version u have |
| 20:47.35 | GuruJee | but i remember running into this problem and thats what helped |
| 20:47.36 | dalfry | what version of what? |
| 20:47.39 | GuruJee | asterisk |
| 20:47.49 | GuruJee | are u configuring ivr? |
| 20:47.56 | dalfry | 1.4.4 |
| 20:48.04 | GuruJee | hmm |
| 20:48.12 | dalfry | nope. have a phpagi script playing mp3 files that I want to play / pause using key presses |
| 20:48.17 | GuruJee | i have 1.2.13 and it works with mp3 now |
| 20:48.50 | GuruJee | have u backgrounded the service? |
| 20:48.51 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 20:48.51 | *** join/#asterisk ssokol (n=ssokol@65-182-39-203.cre.bil.biltmorecommunications.net) |
| 20:48.51 | GuruJee | not the service, i mean the mp3 file? |
| 20:48.54 | dalfry | backgrounded? |
| 20:48.57 | GuruJee | yeah |
| 20:49.02 | dalfry | what does that mean? |
| 20:49.07 | GuruJee | if u dont background it, then it wont take ur key inputs |
| 20:49.11 | GuruJee | it will always play |
| 20:49.17 | *** part/#asterisk linmax (n=maxi@p549156d3.dip.t-dialin.net) |
| 20:49.21 | dalfry | heh :) |
| 20:49.23 | GuruJee | is that what ur problem is? |
| 20:49.27 | nahirean | ... |
| 20:49.27 | dalfry | nope |
| 20:49.41 | GuruJee | well, sorry i troubleshooted the wrong problem :D |
| 20:49.43 | GuruJee | whats ur problem? |
| 20:49.44 | kiscokid | I think its usually exten => _NXXNXXXXXX,1,Dial(ZAP/4, ${EXTEN}) to dial out |
| 20:49.55 | *** join/#asterisk madeinny (n=Steve@70.88.255.133) |
| 20:50.22 | dalfry | I told you. asterisk segfaults when playing mp3 files for parallel calls using format_mp3 |
| 20:50.23 | GuruJee | kisco: what if my aix2 trunk is called priv. Will it be exten => _NXXXNXXXXX,1,Dial (priv,&{EXTEN}) |
| 20:50.24 | GuruJee | ? |
| 20:50.38 | madeinny | /who am I |
| 20:50.45 | GuruJee | $ |
| 20:50.48 | dalfry | madeinny: you are madeinny |
| 20:50.53 | madeinny | Thanks. |
| 20:50.56 | dalfry | np |
| 20:51.10 | GuruJee | wow man, u helped maideinny with self-realization |
| 20:51.13 | GuruJee | are u buddha? |
| 20:51.19 | GuruJee | :) |
| 20:51.38 | CVirus | http://rafb.net/p/HRu6O554.html .... Will channel 3 use context=incoming and hidecallerid=yes (please note the default options) ? |
| 20:51.43 | kiscokid | sorry, don't know what an aix2 trunk is |
| 20:52.03 | GuruJee | iax2 dude |
| 20:52.09 | GuruJee | its something like SIP |
| 20:52.10 | *** join/#asterisk saftsack (n=saftsack@pd9e07dc8.dip.t-dialin.net) |
| 20:53.07 | kiscokid | no it would be exten => _NXXNXXXXXX,1,Dial(IAX2/priv,${EXTEN}) |
| 20:53.08 | madeinny | I'd like to ask a real question now that I botched up my into. Is it possible to do something like Goto(XXX) in a dialplan? Goto(123), and Goto(XXX) didn't work for me. |
| 20:53.39 | GuruJee | cool kisco |
| 20:53.39 | GuruJee | let me try that |
| 20:53.39 | GuruJee | kisco: do u have any experience with DUNDi |
| 20:53.39 | GuruJee | ? |
| 20:53.51 | kiscokid | no, haven't read that chapter yet |
| 20:53.59 | GuruJee | :) aah |
| 20:54.21 | GuruJee | but dude, can you look up my extensions_custom.conf file. all the internal extensions work. I just cant dial outside |
| 20:54.49 | GuruJee | not ext3 dumbo http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 20:54.49 | GuruJee | oops sorry |
| 20:55.02 | GuruJee | http://paste.debian.net/28590 and http://paste.debian.net/28589 are the extensions_custom.conf I need some dial plan help with DUNDi please |
| 20:55.30 | justdave | anyone know if there's any good single-span T1 cards compatible with asterisk that have hardware echo cancellation? |
| 20:55.40 | GuruJee | dude |
| 20:55.44 | GuruJee | thats a tough question |
| 20:55.53 | justdave | The product list on Digium's site dosn't list echo cancellation on anything less than 2 spans |
| 20:55.54 | GuruJee | I know that digium and sangoma dont make it |
| 20:56.04 | GuruJee | yeah and two span is quite expensive |
| 20:56.26 | GuruJee | u know , if u are having echo problems and u have tried everything but nothing helps then its your provider |
| 20:56.46 | justdave | haven't even set up the T1 yet, just wanting to make sure we have hardware on hand when we do. |
| 20:56.54 | justdave | no idea if it'll be echoey or not |
| 20:56.58 | GuruJee | u wont need it |
| 20:57.12 | GuruJee | i have setup a dozen systems and ran into echo issue only once |
| 20:57.18 | justdave | ok. |
| 20:57.19 | GuruJee | and it turned out to be the providers fault |
| 20:57.25 | justdave | so if the provider's decent it won't be an issue :) |
| 20:57.29 | GuruJee | use Digium though |
| 20:57.33 | GuruJee | dotn go with anything else |
| 20:57.39 | GuruJee | use t110p |
| 20:57.46 | justdave | sounds like TE120P then |
| 20:57.50 | justdave | only single-span they still make |
| 20:57.59 | GuruJee | amm |
| 20:58.06 | GuruJee | i think its TE110 |
| 20:58.07 | GuruJee | :D |
| 20:58.15 | GuruJee | may be te120 is neww |
| 20:58.20 | GuruJee | but anyways, its a good card |
| 20:59.49 | justdave | TE120P is the only single-span listed on digium's site. |
| 20:59.49 | justdave | hmm, it does data/voice combo |
| 20:59.49 | GuruJee | yeah it does |
| 20:59.49 | justdave | that could handle our existing T1 |
| 20:59.49 | GuruJee | ok |
| 20:59.49 | GuruJee | T1 is different than PRI |
| 20:59.52 | justdave | we have a T1 now with 5 analog phone lines terminated on it, and the remaining channels are data right now |
| 21:00.00 | GuruJee | if u are gonna T1 it means u are going with sip trunking or something |
| 21:00.06 | justdave | we're wanting to switch the whole thing over to PRI because we have fiber for net now |
| 21:00.07 | GuruJee | then u dont need a PRI card |
| 21:00.18 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:00.25 | GuruJee | but if u wanna keep a T1 then any network card will work |
| 21:00.35 | GuruJee | even a dlink! |
| 21:00.42 | GuruJee | anything that flashes and passes on data |
| 21:00.43 | GuruJee | for T1 |
| 21:01.18 | Greek-Boy | any telco guys around/ |
| 21:01.25 | justdave | pointless to reconfigure for that though, can wait to get rid of the existing router until we have the PRI :) |
| 21:02.02 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 21:02.04 | johann8384 | GuruJee: I think your smoking crack...what T1 are you going to plugin to an ethernet jack? I assume you mean with a T1 router going to Ethernet then to the PBX as data... |
| 21:02.18 | justdave | otherwise we'd just be reconfiguring the card again when it gets converted. |
| 21:03.00 | *** join/#asterisk mazpe (n=lesterm@c-71-206-91-61.hsd1.fl.comcast.net) |
| 21:03.24 | GuruJee | johaan8384: I think u are a nooob. We are talking about Datalink layer, so router is irrelevent. Ofcourse, u cant plug in T1 card into a ethernet jack. |
| 21:03.49 | syzygyBSD | hmmm, so is having an extra asterisk server in between zap and my other asterisk server going to help me debug why calls are being dropped? |
| 21:03.57 | GuruJee | u are one of those students in class who raise their hands and waste time of entire class just to point out that the professor had forgot to put the dot on an i |
| 21:04.19 | syzygyBSD | GuruJee: rather learn right then take shortcuts |
| 21:04.27 | GuruJee | syz |
| 21:04.35 | syzygyBSD | like most of the other students are really in class to learn |
| 21:04.52 | GuruJee | we were talking about layer 2 , layer 3 was irrelevent. And I dont mind it. It was just his choice of words |
| 21:04.57 | GuruJee | which gave away that he is a noob |
| 21:05.07 | GuruJee | actually they are |
| 21:05.11 | GuruJee | later dawgs |
| 21:05.21 | GuruJee | hey johann good luck with ur asterisk server |
| 21:06.27 | *** join/#asterisk WindBack (n=jorge@host44.200-117-61.telecom.net.ar) |
| 21:07.14 | WindBack | Somebody can helpme?? |
| 21:08.02 | WindBack | I configure Asterisk with ekiga, but I couldn't use the interactive menu of asterisk |
| 21:08.23 | WindBack | I put the dtmfmode to rcf2833 in both, but it didn't work |
| 21:10.04 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 21:10.54 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 21:12.34 | Greek-Boy | whats a good class 5 ss7 to voip switch? |
| 21:16.59 | tuan_modulis | I opened a can of whupass on mysel... deleted /var/lib/mysql on a ext3 partition |
| 21:17.04 | tuan_modulis | myself* |
| 21:17.17 | tuan_modulis | *bashes head* |
| 21:17.21 | bkw_ | Greek-Boy, their are many to pick from |
| 21:17.30 | bkw_ | Greek-Boy, what kind of ss7 are you wanting to speak? |
| 21:22.27 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:26.20 | *** part/#asterisk madeinny (n=Steve@70.88.255.133) |
| 21:26.32 | *** join/#asterisk tonycarstens (n=oper@206.135.21.162) |
| 21:26.57 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 21:28.29 | tonycarstens | i'm unable to make outbound calls except for on line one. i have a zap group set up and the outbound call command to dial the group but it still will only work on line 1 |
| 21:29.15 | tonycarstens | any ideas? |
| 21:29.32 | [TK]D-Fender | tonycarstens, Pastebin the new dialplan & zapata |
| 21:31.23 | tonycarstens | http://www.pastebin.ca/503431 |
| 21:31.44 | [TK]D-Fender | tonycarstens, You clearly did not take the replacement I made for you... |
| 21:32.51 | tonycarstens | only thing i changed was the include => incoming |
| 21:33.05 | [TK]D-Fender | tonycarstens, http://www.pastebin.ca/503435 <- better full replacement |
| 21:33.07 | tonycarstens | i added that under the sip context, other than that it is |
| 21:34.10 | [TK]D-Fender | tonycarstens, mod this up and then reload. if it fails, pastebin the CLI output of the attempt |
| 21:34.36 | tonycarstens | ok |
| 21:36.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:37.03 | *** join/#asterisk mazpe (n=lesterm@c-71-206-91-61.hsd1.fl.comcast.net) |
| 21:37.17 | *** join/#asterisk Onyfex (n=ben@S0106000d5637e38a.cg.shawcable.net) |
| 21:38.06 | tonycarstens | http://www.pastebin.ca/503449 |
| 21:38.21 | Onyfex | Hello, Does anyone know how to make asterisk send a tone back across the line? I am trying to make it unlock my apartment door when I enter a code. |
| 21:38.53 | Greek-Boy | when will we get to see wimax phones? Nokia says 2008. |
| 21:39.19 | [TK]D-Fender | tonycarstens, Notive that the context in the error doesn't match your config? |
| 21:39.39 | [TK]D-Fender | tonycarstens, Zapata changes will NOT take effect on a "reload". |
| 21:39.56 | [TK]D-Fender | tonycarstens, You need to either completely restart * or reload chan_zap.so |
| 21:40.10 | [TK]D-Fender | tonycarstens, if all's clear go do "restart now" |
| 21:41.22 | [TK]D-Fender | Onyfex, "show application senddtmf" |
| 21:41.26 | *** join/#asterisk basty (n=basty@dome-city-rockers.sunblast.de) |
| 21:41.30 | basty | Hi |
| 21:42.26 | basty | anyone using Asterisk with Kirk DECT ? I am running into several Problems using Kirk with SCCP. |
| 21:43.12 | tonycarstens | same problem http://www.pastebin.ca/503459 |
| 21:44.13 | [TK]D-Fender | tonycarstens, not like that. Restart asterisk COMPLETELY |
| 21:44.22 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 21:44.26 | [TK]D-Fender | tonycarstens, "restart now" - |
| 21:44.40 | *** join/#asterisk drrt (n=junior@ppp-static2-140.tis-dialog.ru) |
| 21:44.42 | Strom_C | "restart yesterday" |
| 21:45.07 | [TK]D-Fender | Strom_C, : He hasn't compiled res_fluxcapacitor.so yet.... |
| 21:45.09 | tonycarstens | its still lookin for the default context |
| 21:45.16 | Strom_C | oh, ok |
| 21:45.41 | [TK]D-Fender | tonycarstens, kill * completely and restart it. your half-way attempts are bad. |
| 21:45.54 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:46.48 | [hC] | Anyone have any idea why dialing out FXO onto Qwest, i'll dial a 10 digit number (NXX-NXX-XXXX) and it will play a recording that i dont need to dial a 1 (i didnt) i'll hit redial and it will work? |
| 21:46.49 | [hC] | totally random. |
| 21:47.02 | [hC] | other times i'll dial with a 1 and it will work, or say 'you dont need a 1' |
| 21:47.18 | Defraz | sounds like a dtmf problem. |
| 21:47.21 | tonycarstens | how do i kill it |
| 21:47.23 | Strom_C | [hC]: well, what's actually being dialed out onto the circuit? |
| 21:47.27 | [TK]D-Fender | [hC], My gess it its fudging the first digit. Add a "w" or two in front |
| 21:47.29 | Defraz | FX0 is on what type of equipment |
| 21:48.26 | [hC] | Strom_C the correct number is being dialed out. |
| 21:48.26 | [hC] | [TK]D-Fender: I'll try that. Thanks. |
| 21:48.33 | [hC] | [TK]D-Fender: so its missing it between offhook and dial? |
| 21:48.35 | [hC] | you think? |
| 21:48.36 | Strom_C | [hC]: you've confirmed this by clipping a buttset onto the line and listening? |
| 21:48.53 | [hC] | Strom_C: no. I've confirmed it by watching asterisk tell me what number its dialing, so far. |
| 21:49.07 | [hC] | I'm not physically near the pbx, unfortunately. |
| 21:49.25 | [TK]D-Fender | [hC], Yup |
| 21:49.56 | [TK]D-Fender | [hC], I've seen telco switches that are slow to give tone and this fixed nice & easy |
| 21:50.25 | Katty | dooby do! |
| 21:50.55 | *** join/#asterisk ManxPower (n=manxpowe@69.sub-70-220-176.myvzw.com) |
| 21:51.04 | mrdigital | kool |
| 21:51.06 | Katty | ManxPower: ! |
| 21:51.26 | [hC] | [TK]D-Fender: so instead of dialing Zap/g0/6021234567, I dial Zap/g0/ww6021234567 ? |
| 21:51.57 | Katty | good ole ww on analog. |
| 21:52.02 | Katty | [hC]: that's precisely what we did. |
| 21:52.05 | Strom_C | in area code six oh two |
| 21:52.20 | [hC] | I decided to install trixbox at this site to give it a shot and make it easy. Man am I already regretting it. everything is all over hte place, i have no idea where to set that. haha. |
| 21:52.33 | Katty | [hC]: trixbox eh? |
| 21:52.40 | Katty | [hC]: isnt' that the non support version of fonality? |
| 21:52.52 | [hC] | basically, now... although its a different platform entirely |
| 21:52.54 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 21:52.58 | Katty | yepyep |
| 21:53.00 | [hC] | trixbox just includes freepbx basically |
| 21:53.02 | Katty | lemme read up |
| 21:53.06 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 21:53.18 | [hC] | it came with the sangoma a200d that i threw in so i thought id actually give it a go |
| 21:53.18 | Katty | [hC]: ah yes, we have those problems too |
| 21:53.22 | Katty | [hC]: sporatic issues. |
| 21:53.27 | [hC] | its not TERRIBLE, but .... its not fantastic either. |
| 21:53.29 | Katty | [hC]: is that sangoma card for analog lines? |
| 21:53.34 | [hC] | yeah, FXO |
| 21:53.39 | [hC] | Katty: what issues do you have? |
| 21:53.41 | [TK]D-Fender | [hC], Yes |
| 21:53.41 | Katty | [hC]: well, here's your problem. |
| 21:53.47 | Katty | [hC]: it's not sangoma, it's analog |
| 21:53.53 | Katty | [hC]: analog causes all sorts of stupid little issues |
| 21:53.58 | Katty | [hC]: it doesn't have pretty signalling |
| 21:54.04 | Katty | [hC]: think of your problem like a modem |
| 21:54.06 | [hC] | Katty: oh i know. |
| 21:54.11 | Katty | [hC]: you wouldn't tell the modem to dial out without a dial tone |
| 21:54.16 | [hC] | Katty: Im by no means new to being angry at analog installs. |
| 21:54.24 | Katty | [hC]: the modem waits for dialtone, then dials. |
| 21:54.32 | Katty | [hC]: otherwise, it might get ahead of itself and go too quickly. |
| 21:54.34 | [hC] | Katty: The frustrating thing is, all the crappy old nortel pbx's do JUST GREAT with these crappy analog lines. |
| 21:54.41 | Katty | [hC]: same thing with making a call on an analog line.... |
| 21:54.41 | [hC] | Katty: makes perfect sense. |
| 21:54.48 | Katty | [hC]: ya gotta ww a second for a dialtone |
| 21:54.59 | Katty | [hC]: yeah, but they're designed for analog |
| 21:55.04 | Katty | [hC]: they've got the bugs worked out |
| 21:55.14 | Katty | [hC]: and they've had years to do it. |
| 21:55.17 | Katty | [hC]: voip is designed for voip |
| 21:55.19 | [TK]D-Fender | [hC], thats because nortel either waits before sending (intercom default), or gives you RAW dialtone and hangs up if it doesn't like what you dial. |
| 21:55.25 | Katty | [hC]: and shiny digital stuffs. |
| 21:55.30 | *** join/#asterisk mutilator (i=WebChat@the.drinkproject.com) |
| 21:55.39 | Katty | [hC]: analog is just trying to squeeze in a backwards compatibility feature for you. |
| 21:55.56 | Katty | [hC]: we're technically on a t1 now, tho it gets turned into a channel bank and analog lines. |
| 21:56.14 | [hC] | Katty: ew. |
| 21:56.19 | Katty | [hC]: odds are, if you've got more than 8 lines...you're better off (and often spend less money) by getting your numbers ported through a t1 or a pri |
| 21:56.34 | [hC] | Katty: I never understood why anyone would want to do that.. unless you need to pull an analog line off for like.. fax or a credit card terminal or something |
| 21:56.45 | Katty | yeah, we have an analog line here. |
| 21:56.49 | irule | I was TOLD to make asterisk (I think it should be Meetme application) act like this: I am ext 222, 1) ext 222 calls 1800123456. 2) 222 sends 1800123456 to MOH 3) ext 222 call dials another number, this time 1800987654, and once 1800987654 answers the call, just press something to make 1800123456 join 222 and 1800987654. does this make sense? any thoughts? thanks |
| 21:56.55 | Katty | it's for fax, credit card, and the occasion fear of 911 problems. |
| 21:56.57 | [TK]D-Fender | Katty, Your setup is like refried beans.... just can't get the job done right the first time :) |
| 21:57.09 | Katty | ooh mexican |
| 21:57.11 | Katty | hm, dinner |
| 21:57.16 | Katty | hmmm hmmh mm |
| 21:57.23 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 21:57.25 | Katty | i do like some mexican foods. |
| 21:57.27 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 21:57.39 | [TK]D-Fender | irule, You basicaly wannt to do a 3-way call? |
| 21:57.43 | Katty | [hC]: we had all sorts of issues with the ole regular analog setup. |
| 21:57.44 | irule | btw, I am in an instant bean processing factory, really! |
| 21:58.00 | Katty | [hC]: even a few times on occasion, our telco thought we were trying to do call tracing... |
| 21:58.14 | irule | [TK]D-Fender yes a 3 way call just like the ones made by old PBX's |
| 21:58.42 | Katty | [hC]: sometimes the lines would stay open... |
| 21:58.45 | [TK]D-Fender | irule, Your phone should offer you that functionailty all by itself without anything special to do in *. |
| 21:58.49 | [hC] | Katty: nice. out of the ~80 or so installs ive done, maybe.. 5 of them involved analog. every time ive wanted to kill myself. dropped calls, static, calls not being answered, echo (moved to sangoma's d series) |
| 21:58.51 | Katty | [hC]: not to mention that 1 or 2 seconds of echo |
| 21:58.58 | [hC] | I prefer terminating via IAX2 back to my pri |
| 21:59.12 | *** join/#asterisk drega (n=drega@80-47-195-117.lond-th.dynamic.dial.as9105.com) |
| 21:59.13 | Katty | [hC]: and my all time favorite...when going to make an outgoing call, you pick up an incoming call at random! |
| 21:59.22 | [hC] | Katty: haha. we've had that too. |
| 21:59.31 | Katty | that one's great ^_^ |
| 21:59.32 | drega | anyone here been at the dev conference in atlanta today? |
| 21:59.33 | [TK]D-Fender | Katty, he has an A200d ... he doesn't GET echo :) |
| 21:59.35 | [hC] | Katty: I solved that by dialing out using "G" instead of "g" |
| 21:59.38 | drega | hows it going? |
| 21:59.44 | [hC] | Katty: in my zap line, so it starts at the end of the line pool. |
| 21:59.46 | Katty | [hC]: try explaining why the boss answered an incoming call ;) |
| 21:59.52 | Katty | [hC]: that's /real/ fun |
| 21:59.54 | [hC] | Katty: been there. :) |
| 22:00.04 | [hC] | Katty: how did you solve all of those issues? Ive either worked around them or ditched the analog setup |
| 22:00.05 | Katty | [TK]D-Fender: sigh. |
| 22:00.09 | Katty | [TK]D-Fender: come brainwash my boss for me. |
| 22:00.09 | irule | [TK]D-Fender the problem is that I have old analog phones connected to an ata so I need * to simulate this bs |
| 22:00.22 | Katty | [hC]: well, you can't really work around them much |
| 22:00.27 | Katty | [hC]: it's just the doom of analog. |
| 22:00.31 | [hC] | no no, i have had echo on an a200d. not often, but it happens. <3 gaintuning lines over and over. |
| 22:00.34 | [TK]D-Fender | Katty, Actually using "G" instead of "g" should statistcally shrink your analog line selection collision events to near 0.... |
| 22:00.43 | *** join/#asterisk boch (n=fran@190.48.217.195) |
| 22:00.43 | Katty | [hC]: tweaking things help reduce it. getting a t1 to analog fixed our echoy stuff |
| 22:01.01 | Katty | [TK]D-Fender: yeah well the new server w/ t1 card is going to fix everything. |
| 22:01.06 | Katty | [TK]D-Fender: well, it won't fix my stupidity... |
| 22:01.17 | Katty | [TK]D-Fender: but i can dream, regardless :P |
| 22:01.56 | Katty | [hC]: but if i were you, run screaming from analog (= |
| 22:02.15 | Katty | [hC]: with your arms straight out in front of you, like in pirates of the carribean |
| 22:02.31 | [TK]D-Fender | Ok, off to martial arts, back in a few.... |
| 22:02.34 | Katty | later |
| 22:02.44 | Katty | and i'm goin to dinner! |
| 22:02.45 | Katty | g'later |
| 22:02.52 | Katty | [hC]: hope you got whatever answer you were looking for (= |
| 22:05.36 | irule | how can I set a speciffic caller ID? |
| 22:08.26 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 22:09.04 | wwalker | anyone able to confirm or deny telasip still being in business? they don't answer any of their phone numbers. |
| 22:11.54 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-28.plaicmtc01.tx.dh.suddenlink.net) |
| 22:12.40 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 22:12.44 | linagee | LOL. anyone read slashdot? |
| 22:12.53 | *** join/#asterisk mutilator (i=WebChat@the.drinkproject.com) |
| 22:12.56 | linagee | " Nortel Strong-Arms Open Source Vendor Fonality" |
| 22:13.05 | linagee | bastards. hah |
| 22:13.25 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 22:13.29 | mazpe | anyone using the cisco 7960g with asterisk |
| 22:13.30 | mazpe | ? |
| 22:15.27 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 22:16.23 | clyrrad | Hey all, can anyone link me to a doc or let me know what the differnce is between greet.WAV greet.wav and greet.gsm in /var/spool/asterisk/voicemail/peer |
| 22:20.22 | clyrrad | anyone? |
| 22:27.00 | JT | clyrrad: the difference between .wav and .gsm should be obvious, but why you have different .wavs with different cases, i don't know |
| 22:28.31 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:30.02 | clyrrad | JT: its there by default when a mailbox is created |
| 22:30.19 | clyrrad | and yes .wav and .gsm is sort obvoius |
| 22:30.33 | clyrrad | becase I found a page that sais .wav is not compressed |
| 22:30.58 | clyrrad | I take it .WAV is the compressed version |
| 22:31.25 | syzygyBSD | no wavs are compressed |
| 22:32.12 | syzygyBSD | my guess is they are formatted differently, maybe one has stereo, or 8000 sample rate or ... one of a bunch of other formats |
| 22:34.11 | clyrrad | so yea this is what I am tryign to find out |
| 22:34.24 | clyrrad | I had a recording done for voicemail box in gsm format |
| 22:34.35 | Greek-Boy | JT it seems that most carrier grade switches only support SIP |
| 22:34.36 | clyrrad | so I need t convert it to whatever format .WAV and .wav are..... |
| 22:34.49 | clyrrad | the problem is I dont know what format .WAV and .wav are in........ |
| 22:34.52 | JT | Greek-Boy: you must be joking |
| 22:35.06 | Greek-Boy | i mean the voip ones |
| 22:35.35 | JT | clyrrad: you shouldn't get recordings from prompts done in .gsm |
| 22:35.43 | JT | Greek-Boy: switches or phones? |
| 22:35.51 | Greek-Boy | switches |
| 22:36.03 | JT | they will most definitely support H.323 |
| 22:36.11 | JT | H.323 is much more carrier grade than SIP |
| 22:37.39 | clyrrad | jT: what is the reason for that? |
| 22:37.56 | clyrrad | JT: also I still need to find what format .wav and.WAV are..... |
| 22:37.59 | JT | clyrrad: poor audio quality in gsm |
| 22:38.14 | clyrrad | jT: strange these sound great?? |
| 22:38.26 | JT | with .wav, it's good quality and easily converted by asterisk automatically |
| 22:38.32 | Defraz | It is all a matter of preference really h323 and SIP. |
| 22:38.38 | JT | the .wav or the .gsm? :P |
| 22:38.57 | clyrrad | JT: what is the difference between .wav and .WAV? |
| 22:39.01 | JT | Defraz: they each have their advantages, H.323 clearly leads for carrier use, SIP for PBX |
| 22:39.13 | JT | clyrrad: if i knew, i would've told you |
| 22:39.14 | Defraz | JT true true |
| 22:39.25 | clyrrad | jT: k |
| 22:39.50 | JT | sip has pretty poor signalling capabilities compared to q.931 |
| 22:42.15 | clyrrad | JT: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 22:42.45 | rene- | h323 is older |
| 22:42.52 | Greek-Boy | JT: so which protocol will wimax startups use? |
| 22:43.51 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 22:43.59 | JT | rene-: correct |
| 22:44.20 | rene- | i would say they will keep a variety of protocols, but customer premises equipment will be sip likely and they might have upstream connections in PRI, SIP, h323 and other forms |
| 22:44.45 | JT | Greek-Boy: no idea, i think wimax is more a data protcol, i don't really see what the big deal about wimax is |
| 22:44.54 | JT | another wireless communication standard, hooray |
| 22:45.28 | JT | rene-: well that sounds right |
| 22:45.35 | Greek-Boy | wimax is the future of mobile communication as it provides ultimate convergence |
| 22:45.51 | JT | straight out of a glowing press release |
| 22:45.59 | rene- | heh |
| 22:46.00 | JT | it's just another wireless standard |
| 22:46.01 | JT | big whoop |
| 22:46.09 | rene- | what about the folks at gnu radio |
| 22:46.18 | rene- | are they onto something big? |
| 22:46.23 | rene- | a software radio? |
| 22:46.44 | rene- | they say they are decoding hdtv of the air |
| 22:46.45 | rene- | well they are |
| 22:46.45 | rene- | ithey posted pictures |
| 22:46.54 | rene- | but like the hardware they need is pretty powerful |
| 22:46.54 | JT | software radio is already happening quite a bit commercially |
| 22:46.58 | Strom_C | wimax is the future of my boner |
| 22:47.10 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 22:48.07 | rene- | it could be cool to have a box that would do like wimax, wifi dect gsm and what not |
| 22:48.15 | rene- | and was based on linux and cheap to make |
| 22:48.21 | Greek-Boy | wimax long range and high speed... |
| 22:48.41 | JT | Greek-Boy: just copying and pasting out of press releases are we? |
| 22:49.50 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 22:50.47 | Greek-Boy | lol |
| 22:50.48 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:51.22 | JT | Greek-Boy: you haven't provided any reasoning and facts, just fancy sounding grablines |
| 22:51.28 | JT | wimax will come and go |
| 22:51.34 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 22:51.39 | JT | like all other standards for wireless data transmission |
| 22:52.17 | Greek-Boy | JT: whats more promising than wimax? |
| 22:52.24 | Strom_C | no! the future of wireless communications is surely AM RADIO |
| 22:52.43 | kvidell | 2 meter! |
| 22:54.25 | JT | Greek-Boy: i didn't say anything was more promising, it doesn't mean that there isn't something more promising though |
| 22:55.18 | Greek-Boy | WiMAX could replace GSM and CDMA |
| 22:56.22 | JT | but probably won't |
| 22:56.26 | JT | so who cares :) |
| 22:56.53 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 22:56.57 | JT | WCDMA is more likely to replace gsm |
| 22:57.00 | Greek-Boy | lol |
| 22:59.28 | JT | aren't there more interesting things to chat about than wireless standards with stupid name? :) |
| 23:00.34 | Greek-Boy | how else can we get voip mobile? |
| 23:00.44 | Greek-Boy | at a operator level standard |
| 23:00.45 | Strom_C | UMTS |
| 23:01.04 | JT | Greek-Boy: i don't want voip mobile |
| 23:01.22 | JT | i want proper mobile phone service if i'm paying for it |
| 23:01.22 | Mercestes | Strom_C Does IAX2 run over AM?? |
| 23:01.43 | Strom_C | if you modulate it properly? |
| 23:01.46 | Mercestes | yay |
| 23:02.12 | *** join/#asterisk ixx (i=foobar@cpe-70-112-123-132.austin.res.rr.com) |
| 23:02.16 | Greek-Boy | JT: voip is the future, will replace everything and give operators the ability to offer tariffs @ fraction of current ones |
| 23:02.39 | JT | Greek-Boy: voip is just another method of moving voice traffic |
| 23:02.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:02.59 | Greek-Boy | a more sophisticated and feature-rich method, yes :) |
| 23:03.11 | JT | Greek-Boy: why the hell would i want to pay for phone service to move voip packets to me instead of circuit switched voice frames? |
| 23:03.26 | JT | Greek-Boy: bullcrap... what is more sophisticated about it? |
| 23:03.38 | *** join/#asterisk drega (n=drega@80-47-238-236.lond-th.dynamic.dial.as9105.com) |
| 23:03.52 | ixx | looking for recommendations on reliable SIP provider for residential service. by reliable i mean both on the network side and the customer support side |
| 23:03.55 | ixx | any suggestions? |
| 23:04.11 | ixx | LNP if possible also |
| 23:04.28 | Strom_C | and let me guess - no more than half a cent per minute |
| 23:04.39 | Greek-Boy | JT: what about your voip pbx? why do u have asterisk instead of the conventional stuff? |
| 23:04.50 | JT | Greek-Boy: asterisk is NOT a voip pbx |
| 23:04.59 | JT | it is a PBX that CAN do voip |
| 23:05.02 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 23:06.26 | JT | i have it due to flexibility at the price level, however that is NOT a result of VoIP really |
| 23:06.26 | ixx | was considering Teliax... who I have used in the past on their Pay as you go plan. But I have not placed a large number of calls through them to see reliability |
| 23:06.26 | JT | it can do analogue and digital circuits and phones just fine also |
| 23:06.53 | JT | its internals are not pure voip in any close |
| 23:07.03 | JT | closer to a circuit switched softswitch anyway |
| 23:07.07 | ixx | Strom_C: hah... monthly flat rate plans would be a nice option... but 0.005 is not required :) |
| 23:07.32 | ixx | 1-2c/min for minute plans are OK |
| 23:07.53 | ixx | I need both options though (because both are being requested) |
| 23:07.54 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 23:08.12 | Strom_C | teliax has been good to me |
| 23:08.46 | ixx | I just don't feel confident recommending some providers I have been happy with in the past |
| 23:08.51 | CoffeeIV | For each extension I have (say 103), in extensions_additional.conf there is a line "E103 = SIP" or "E103 = IAX2" -- but these variables are not being set for some reason -- does this sound like a familar problem to anyone ? |
| 23:08.53 | b1shop | ok. i have a test server set up. i can call it from my cell and forward the calls to a sip phone. but i cannot seem to make an outbound call from the sip phone to cell. (using ekiga as software sip client). |
| 23:09.00 | ixx | So I am hoping to find some new good ones |
| 23:09.10 | JT | Greek-Boy: so i think you are mistaking the advantages of asterisk over a conventional pbx as being somehow a product of voip... which it isn't |
| 23:09.18 | ixx | Strom_C: have you done LNP with teliax? |
| 23:09.18 | JT | voip is just a method of moving voice |
| 23:09.42 | Greek-Boy | JT: a cheap method, yes? |
| 23:09.48 | Strom_C | CoffeeIV: let me guess - freepbx? trixbox? |
| 23:09.58 | Greek-Boy | yet voice quality is comparable to conventional methods? |
| 23:10.21 | b1shop | do you *need* outbound calling rules? |
| 23:10.21 | CoffeeIV | Strom_C: the dialplan started that way, in the Asterisk@home days, but it has been hacked up beyound all recognition |
| 23:10.50 | JT | Greek-Boy: yes, it can be a cheap method, albeit a less reliable one |
| 23:11.07 | *** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net) |
| 23:11.19 | JT | Greek-Boy: if i'm paying a phone provider for mobile phone service, why would i want the data transmitted to me as voip instead of circuit switched data? |
| 23:13.12 | Greek-Boy | you're paying a quarter of local tarrifs and probably a tenth of international tarrifs |
| 23:13.24 | Greek-Boy | and voip is becoming more reliable and comparable |
| 23:13.55 | ixx | hmmm 1500 softcap is not good for many families I know |
| 23:14.03 | ixx | mainly when there are lots of kids |
| 23:14.09 | ixx | or kids in college etc |
| 23:14.20 | ixx | thats the one thing that is bugging me with teliax |
| 23:14.34 | JT | Greek-Boy: if i'm paying for mobile phone service, it's irrelevant if the voice is delivered to me via voip or not to my handset, and voip certainly won't make it cheaper |
| 23:14.37 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:15.11 | mutilator | yea |
| 23:15.17 | mutilator | just get My Circle from alltel |
| 23:15.36 | mutilator | :P~ |
| 23:16.14 | Greek-Boy | JT: what do u use voip for? |
| 23:17.05 | JT | what it's good for |
| 23:17.06 | ixx | hard to convince all extended family and friends to go with one provider like that |
| 23:17.11 | JT | making some cheap calls from asterisk |
| 23:17.17 | JT | connecting sip handsets |
| 23:17.21 | ixx | or alltel, cingular, or whatever would be fine |
| 23:17.27 | ixx | or even 1 voip provider |
| 23:17.35 | JT | you should try and be more logical with your voip argument :) |
| 23:17.40 | mutilator | i dunno, the 10 people free calling |
| 23:17.48 | mutilator | i use almost 0 of my anytime air minutes |
| 23:18.15 | mutilator | i even put a dialup POP in one of them so i can dialup for however long i need if 'free' wifi isnt available |
| 23:18.50 | lee_is_me | i'm trying to install a sangom A200 and on port 1 (out of 2 ports) I get a busy signal...anyone have an idea what might cause this? |
| 23:19.48 | lee_is_me | that is when dialing into the system...to be more specific |
| 23:20.15 | Strom_C | lee_is_me: a busy signal, or a reorder? |
| 23:20.31 | lee_is_me | Strom_C: not sure what the difference is |
| 23:20.37 | lee_is_me | should be ringing either way |
| 23:20.43 | Strom_C | reorder is sometimes misnamed "fast busy" |
| 23:20.55 | lee_is_me | no, sounds like a regular busy signal |
| 23:21.02 | lee_is_me | port 2 works just fine |
| 23:21.20 | lee_is_me | originally tried 3-4 and 4 would have a terrirble humming noise in the background |
| 23:21.29 | lee_is_me | maybe the module?.... |
| 23:21.42 | Strom_C | possibly |
| 23:21.59 | lee_is_me | the last 5 days of dealing with analog cards....sheesh, loll |
| 23:22.14 | lee_is_me | not sure what I can try next |
| 23:22.31 | lee_is_me | maybe stomping on the card... |
| 23:22.53 | lee_is_me | I RMA'd a TDM400 before I tried the sangoma |
| 23:23.15 | lee_is_me | what would the fast busy indicate when dialing into the system, just curious |
| 23:23.36 | Strom_C | lee_is_me: a call that failed to set up properly |
| 23:23.45 | lee_is_me | ah |
| 23:24.07 | Strom_C | pastebin your console output |
| 23:24.18 | lee_is_me | sure, hold 1 |
| 23:24.33 | lee_is_me | well, there is no output when i use port 1 |
| 23:24.42 | lee_is_me | port 2 works as advertized |
| 23:24.53 | lesouvage | I'm using T,1,Playback(meetmecall/no_credits_left) exten => T,n,Goto(h,1) but when the absolute time out is reached Asterisk jumps directly to exten h,1 . and not first to T,1 . |
| 23:25.27 | lesouvage | Is tis an error or does the h extensions prevails above the T extension. |
| 23:25.34 | lee_is_me | I'll try putting the module back on 3-4 where it almost worked, but with a load humming in the background. There WERE a couple of errors that popped up |
| 23:26.24 | Strom_C | lesouvage: is there a priority 1 for T? |
| 23:26.42 | ixx | thx for info |
| 23:26.45 | ixx | time to eat |
| 23:27.12 | lesouvage | Strom_C: yes the message that there are no credits left. |
| 23:27.23 | Strom_C | er, yeah |
| 23:29.02 | lesouvage | Strom_C: I think I find it. T is in wrong context. |
| 23:29.28 | Strom_C | oh, well that was easy |
| 23:33.02 | lee_is_me | one thing I did notice is that when asterisk loads (during boot) there is an error in zaptel.c:759 SIG 0000000 |
| 23:33.16 | lee_is_me | not sure what it means though |
| 23:35.21 | CoffeeIV | A question about dialplan syntax -- should the line "E103 = SIP" in extensions.conf set the global variable E103 to "SIP" everywhere, or should I have the call actually pass through a line that does Set(E103=SIP) ? |
| 23:38.04 | lee_is_me | Good news: Card starting working after I put it back on 3-4 where it acted up in the first place. Bad news is that I'm gonna go stomp on it anyway for not giving me a reason for it working now |
| 23:38.30 | JT | CoffeeIV: no.... did you just guess |
| 23:38.31 | JT | ? |
| 23:38.42 | lee_is_me | I find it very disturbing when things just start working without a reason where they were not working before, lol |
| 23:39.18 | b1shop | can anyone help me with outbound dialing from a software sip client? |
| 23:39.46 | *** join/#asterisk saftsack (n=saftsack@pd9e07dc8.dip.t-dialin.net) |
| 23:39.52 | b1shop | dialing in from my cell, i can make the sip phone ring.. but not the otehr way aorund |
| 23:40.04 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 23:41.35 | lee_is_me | Strom_C: Do you do many analog installs? |
| 23:41.52 | Strom_C | a few here and there |
| 23:42.05 | lee_is_me | ever had problems with the cards? |
| 23:42.12 | Strom_C | no |
| 23:42.19 | Strom_C | ive had problems with the circuits, but not the cards |
| 23:42.35 | lee_is_me | ah, you mean circuits on the card or the modules? |
| 23:42.41 | Strom_C | uh no |
| 23:42.43 | JT | no |
| 23:42.44 | Strom_C | the circuit from the telco |
| 23:42.45 | JT | the circuit |
| 23:43.22 | lee_is_me | lol, i thought you meant that something wasn't soldered corectly |
| 23:43.30 | Strom_C | no |
| 23:43.49 | Strom_C | in telcoese, "circuit" /always/ means the circuit you lease from the telco |
| 23:44.02 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 23:44.13 | lee_is_me | nice. sorry the only thing that kept me out of college was...high school |
| 23:44.46 | lee_is_me | b1shop: do you mean that you cannot dialout? |
| 23:45.12 | b1shop | lee_is_me, correct... it's analog fxo card |
| 23:45.43 | lee_is_me | did you try inserting a pause "w" before dialing the number? I had that problem a couple of months ago... |
| 23:46.11 | Strom_C | lee_is_me: FWIW, I've only ever used the digium tdm400/tdm800 |
| 23:46.23 | Strom_C | I like the TDM800 a bit more than the 400 |
| 23:46.35 | b1shop | lee_is_me, trying with ekiga (linux sip phone). it just says user not found |
| 23:46.59 | b1shop | lee_is_me, i can make ekiga ring from my cell... but i cannot return the call |
| 23:47.23 | lee_is_me | Strom_C: I've only tried the TDM400 so far. This is my first time with Sangoma |
| 23:47.29 | CoffeeIV | JT: I am guessing based on examining some example dialplans; does a definitive reference of the dialplan language exist on the web or voip-info somewhere ? |
| 23:47.40 | lee_is_me | b1shop: you should pastebin your configs for everyone to see |
| 23:48.00 | lee_is_me | b1shop: www.pastebin.ca |
| 23:48.48 | b1shop | not much there really.. |
| 23:49.07 | b1shop | its all in extensions.conf right? |
| 23:49.17 | lee_is_me | b1shop: and sip.conf |
| 23:49.33 | b1shop | i bet the prob is in sip.conf. have not changed that from default |
| 23:49.55 | lee_is_me | b1shop: look there. you need to setup the extension in there. |
| 23:50.26 | lee_is_me | Strom_C: after tomorrow's install, I'll have a whopping 5 installs under my belt for PBX, lol |
| 23:51.10 | lee_is_me | I have over 1000 P.O.S. install under my belt but they don't count here... |
| 23:52.06 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:53.39 | JT | CoffeeIV: check out the book |
| 23:53.41 | JT | ~thebook |
| 23:53.43 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:56.19 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 23:56.30 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:58.21 | *** join/#asterisk trixjames (n=Jamestri@fw.newsnet.com) |
| 23:58.38 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 23:59.02 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 23:59.05 | trixjames | has any one used NVLineDetect ? |
