00:00.03 | *** part/#asterisk [dc] (n=dc@24-205-223-175.dhcp.slto.ca.charter.com) |
00:00.58 | mzo_ | iwhat's the going rate for a paypal thank you to imlement a call block list and some fancy stuff? :P |
00:01.13 | Qwell[] | mzo_: "fancy stuff"? |
00:01.14 | Qwell[] | :p |
00:01.47 | mzo_ | well, like, a custom set of voice mail prompts when people on the block list so it'll say 'the caller you hae called does not accept calls from facist companies like citibank. please try your call again later when you aren't an ass' and have it hang up |
00:02.26 | Qwell[] | that could be done in dialplan |
00:02.37 | mzo_ | i'm sure it can be done a bunch of awyas but i'm wary of fucking up my system again ;) |
00:02.38 | Skumling | Qwell: are you one of the *-devs? |
00:02.52 | Qwell[] | Skumling: I'm a bug marshall |
00:03.14 | Qwell[] | I do write code though, sure |
00:03.29 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
00:03.39 | Skumling | Qwell: okay... must make friends with you ;) |
00:03.43 | *** join/#asterisk dominix (n=dominix@CA03FD17.adsl.mana.pf) |
00:04.16 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
00:04.40 | lucasjb | When a SIP call connects to my Asterisk system, the process events/0 jumps to 99% CPU usage. This has only just started happening, I can't figure what configuration change I've made that's caused it... |
00:05.59 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
00:06.18 | AJMn | <PROTECTED> |
00:06.47 | Qwell[] | AJMn: What sip phone? |
00:07.00 | Qwell[] | There was an echo issue with a certain firmware version of...umm...polycom? |
00:07.01 | knight_ | ; switch => IAX2/user:password@bigserver/local <--- will this let me pass traffic between asterisk servers? |
00:07.05 | *** join/#asterisk family (n=family@c-67-163-169-5.hsd1.ct.comcast.net) |
00:07.14 | Qwell[] | knight_: Should, but it's better to use a peer |
00:07.17 | family | anyone in here have astcc working |
00:07.28 | Qwell[] | family: I've gotten it working...I don't currently though |
00:08.13 | family | i have it running it all, but when i try to use it, , using the sampels, dialign 1234, its says 20 then hangs up |
00:08.34 | Qwell[] | ugh, I saw that before...what was it? |
00:08.43 | Qwell[] | it just tells you the length of the card |
00:08.53 | Qwell[] | or, rather, the length it expects a card to be |
00:09.27 | Qwell[] | family: You'd have to pastebin your CLI output. I think you missed a step |
00:09.35 | family | so isnt there suppsoed to be a prompt askign for the # |
00:09.43 | family | sure brb |
00:09.44 | Qwell[] | I think that's what that is, is the prompt |
00:09.45 | knight_ | Qwell, I have an asterisk box here at home, but I want my asterisk on a colo box to handle all the calls and pass them to my home... I have 8+ trunks, and various dialplans, so I'd hate to have to duplicate my dialplans |
00:09.59 | Qwell[] | knight_: read up on switch => |
00:10.06 | knight_ | Qwell, yeah that's what I thought |
00:10.35 | *** join/#asterisk ardor (n=vircuser@las-cust-66.18.135.148.mpowercom.net) |
00:10.43 | AJMn | Qwell[] Zyxel P2000W, but both have updated firmware, it only happens when i call each other, but if i call an outside line from a sip phone or outside to sip theres no echo, just calls that stay inside the asterisk system |
00:11.02 | Qwell[] | AJMn: Are they reinviting? |
00:11.14 | ardor | how do shutdown asterisk via the command line... asterisk -xstopnow |
00:11.17 | ardor | doesnt work.. |
00:11.21 | family | Qwell[] http://pastebin.ca/39182 |
00:11.29 | Qwell[] | ardor: asterisk -rx "stop now" |
00:11.31 | Qwell[] | should do the trick |
00:11.34 | Math[laptop] | ardor: asterisk -rx "stop now" |
00:11.37 | ardor | thanks |
00:11.41 | Qwell[] | Math[laptop] slow :p |
00:11.45 | Math[laptop] | lol |
00:11.55 | ardor | sweet!! |
00:11.59 | Qwell[] | family: Yeah, I'd say you missed a step |
00:12.04 | AJMn | <Qwell[]> set to no |
00:12.11 | family | what in particular |
00:12.12 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
00:12.14 | Math[laptop] | ardor: actually its asterisk -rx "cli command here" |
00:12.25 | Qwell[] | family: Dunno...I'd have to install it to try to figure it out |
00:12.31 | Qwell[] | or see yours |
00:12.36 | family | you still ahve your old config files from it |
00:12.41 | family | have |
00:12.42 | Qwell[] | no, it was for somebody else |
00:12.52 | family | damn |
00:13.12 | AJMn | <Qwell[]> reinviting set to NO on both |
00:13.35 | *** join/#asterisk troy (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
00:14.12 | Zodiacal | anyone know if my line assignments can be both speed dial -AND- show the status of ext.'s? i.e. on/off hook using hint. if i use sccp with my cisco 7960 phones? |
00:14.45 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
00:15.02 | Qwell[] | Zodiacal: chan_sccp? |
00:15.08 | Zodiacal | ya |
00:15.10 | Qwell[] | yep |
00:15.15 | Zodiacal | coolness |
00:15.19 | Zodiacal | qwell thank you! |
00:15.22 | Qwell[] | 1234,name,hintexten |
00:15.51 | *** part/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
00:15.51 | Zodiacal | i havn't tried sccp yet ,but i have heard good things about it.. |
00:15.57 | Zodiacal | like that feature for instance. |
00:16.01 | Qwell[] | I love chan_sccp |
00:16.25 | Qwell[] | especially with my realtime patch...mmm |
00:16.46 | Zodiacal | can i have the latest firmware :P j/k im still waiting for my retailer to give me my cisco account so i can d/l it.. i can only play with sip right now :P and its ok.. |
00:17.33 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
00:17.33 | Zodiacal | qwelll realtime patch? |
00:17.33 | Qwell[] | mmhmm |
00:17.33 | Qwell[] | let's me put the sccp configs in a database |
00:17.33 | Zodiacal | oic |
00:17.33 | Qwell[] | Sergio is going to add it "any time now"...heh |
00:17.33 | Qwell[] | I need to get on him about that |
00:17.48 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
00:18.12 | Zodiacal | hints and programable soft keys is what sounds nice |
00:18.23 | Qwell[] | They aren't exactly programmable |
00:18.27 | Zodiacal | qwell have you tried changing the softkeys? how easy is it? |
00:18.33 | Qwell[] | I mean, form the services menu...sure |
00:18.41 | Qwell[] | I looked a few times...I wasn't feeling too ambitious |
00:18.42 | Zodiacal | like i wanta add a page button |
00:18.56 | Zodiacal | to just dial an ext. to get loud speaker paging.. |
00:18.56 | Qwell[] | it's not impossible |
00:19.14 | Qwell[] | certainly isn't easy though |
00:20.45 | dmz | hey y'all, i enabled caller announcing on meetme and whenever i enter the conference, it says to say name & press #, however right after it does that, it goes directly into conference. i don't hit # and it doesn't capture my name...also once in a conference how can i get into administrative mode? or do i have to setup another extension and join as an admin? |
00:21.21 | Qwell[] | dmz: You have to join as admin |
00:21.51 | *** part/#asterisk bertd (n=admin@adsl-220-179-181.mob.bellsouth.net) |
00:22.35 | Qwell[] | brb |
00:22.58 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
00:23.47 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:24.36 | *** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu) |
00:25.34 | *** part/#asterisk jaike (n=a@203.131.137.76) |
00:28.43 | *** join/#asterisk Math[laptop]_ (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
00:30.26 | *** join/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
00:31.41 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
00:32.45 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
00:38.38 | dmz | Qwell, thanks. so i need to setup a specific extension for each conference for admin's to join as? hmm i might look into the code, should be able to use admin code instead of pin and have it recognize admin vs regular pin-auth'd user |
00:39.41 | [av]bani | wow, the asterisk phpbb forums got thoroughly trashed |
00:40.56 | drumkilla | um ... which ones? |
00:41.07 | drumkilla | forums.digium.com looks fine. |
00:41.19 | [av]bani | i cant login anymore, says wrong password |
00:41.42 | drumkilla | well, I wouldn't classify that as thoroughly trashed |
00:41.51 | [av]bani | of course they got really screwed on friday... someone redirected them to the sales forum for allison |
00:42.09 | drumkilla | i can log in just fine |
00:42.21 | Qwell[] | people actually use forums? |
00:42.46 | drumkilla | almost 13,000 posts ... |
00:42.54 | Qwell[] | how many are answered? ;/ |
00:42.55 | [av]bani | well, somehow my account got corrupted since friday |
00:43.12 | [av]bani | after the big mangle |
00:43.15 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
00:43.47 | drumkilla | what is your username |
00:44.00 | [av]bani | bani |
00:44.33 | [av]bani | well, guess i get to wait another 30 minutes because it says i've exceded login attempts |
00:45.52 | drumkilla | well, I can reset your password if you need it ..... |
00:46.01 | [av]bani | i already tried. |
00:46.05 | [av]bani | then it locked me out. |
00:46.39 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
00:48.39 | Frawg | hey |
00:48.47 | Frawg | g729 uses 8kbit/sec correct? |
00:48.52 | Frawg | is there anyway to change it's timeslicing? |
00:48.56 | Math[laptop] | plus overhead |
00:49.19 | Frawg | Math[laptop]: what's the overhead? |
00:49.31 | Math[laptop] | it uses around 20kbps total |
00:49.44 | Math[laptop] | http://www.packetizer.com/voip/diagnostics/bandcalc.html |
00:53.56 | neon_kl | i want to setup sms server using gsm modem can suggest a gsm modem 4 sim card support |
00:54.41 | litage | is it possible to limit a particular extension to a certain number of concurrent/simultaneous calls? |
00:54.57 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
00:55.10 | Frawg | Math[laptop]: can you change the frames/packet easily? |
00:56.30 | dlynes | litage: sip.conf/iax.conf/...: incominglimit=n |
00:57.04 | Frawg | Math[laptop]: of g729 |
00:57.14 | Math[laptop] | uhm let me check |
00:57.16 | *** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
00:57.42 | Math[laptop] | Frawg: no idea |
00:59.44 | litage | dlynes: thanks for that. i looke dup incominglimit on voip-info.org and it turns out that it's been deprecated in favour of setgroup and checkgroup |
00:59.56 | litage | s/looke dup/looked up/ |
01:01.05 | Qwell[] | Frawg: There is/was a patch on the tracker to allow changing packet sizes |
01:01.33 | Math[laptop] | litage/dlynes: which have then been deprecate for the GROUP() and CHECKGROUP() dialplan functions |
01:01.42 | Math[laptop] | er, not CHECKGROUP() but GROUP_COUNT() |
01:06.05 | litage | thanks Math[laptop] |
01:06.58 | Frawg | Math[laptop]: cool, you don't happen to know the overhead that an ssh tunnel adds(perpacket) ? |
01:07.55 | Math[laptop] | er, you're tunnelling a voice conversation over ssh? |
01:08.09 | trixter | I dont know about the IP layer, presumably it has some (I fail to see how it cant) but there is also the cipher chosen, most are block ciphers which will padd to make an even block, that will be at most a couple bytes per packet.. |
01:08.26 | trixter | it shouldnt be that hard to figure out what you are sending and what ssh packet sizes are and do the math |
01:09.13 | trixter | Math[laptop]: if its a local network the tcp difference wont matter that much, by local I mean direct connection could be a wan could be physically local, either way the tcp part of it shouldnt intefere if the network is managed |
01:09.29 | trixter | however if, and I suspect this is what you were getting at, you tunnel via the inet then there may be problems |
01:09.53 | trixter | I didnt know his configuration so I didnt comment on that |
01:10.25 | *** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
01:12.16 | Err | you don't want to tunnel VoIP over ssh |
01:12.31 | *** join/#asterisk p0g0__ (n=pogo@mrtc-dsl-610045.mis.net) |
01:12.32 | trixter | that depends |
01:13.01 | Err | not really - it's pretty much a universal constant |
01:13.02 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
01:13.14 | Err | if you're on a network with virtually NO loss, then you might be able to get away with it |
01:13.20 | trixter | if its a local network as described above on a managed network it wont be a problem |
01:13.31 | trixter | ok thanks for confirming what I said |
01:13.47 | Netgeeks | I'll provide a third confirmation if you need it |
01:13.51 | Err | otherwise, if TCP's congestion control ever drops below the required bitrate, you'll experience strangeness |
01:14.10 | trixter | that doesnt sound like a managed network to me |
01:14.16 | trixter | why I specifically added that qualifier |
01:14.19 | Err | that sounds like packet switching to me |
01:14.46 | trixter | some people maintain networks that are more than just internet links |
01:14.53 | trixter | why I specifically commented on those |
01:14.58 | trixter | I wasnt talking about home dsl |
01:15.28 | Err | TCP's congestion window won't ever grow much above the committed bit rate, which means that ANY loss (including corruption, collision, ARP timeout, whatever) will cause your congestion window to possibly drop below the CBR |
01:15.33 | Err | I'm not either |
01:15.39 | Err | I'm just talking about TCP dynamics |
01:16.47 | trixter | on a non home network there are ways to deal with that |
01:16.53 | *** join/#asterisk ctooley (n=ctooley@24-155-179-239.dyn.grandenetworks.net) |
01:16.56 | Err | to deal with TCP's congestion window? |
01:16.58 | trixter | however if you want to talk about home dsl then you are 100% right you shouldnt use ssh |
01:17.12 | Err | your implication that I've never seen a non-home network is misguided |
01:17.12 | trixter | I however see little point in arguing the difference between a managed network and home dsl with you |
01:17.21 | ctooley | There a good client for the Nokia 770 tablet available yet? |
01:17.27 | trixter | I never said you hadnt seen one |
01:17.45 | trixter | I am saying that you, like many in the asterisk community, think in terms of a home install and try to make assumptions and comments that may not apply |
01:17.50 | Err | no, I don't |
01:17.56 | trixter | its a common asterisk thing, dont worry about it |
01:18.12 | Err | I happen to do TCP research, and I'm well aware of what can fail with TCP and CBR streams |
01:18.16 | trixter | most people do what you do, think solely in terms of what runs fine on their box at home on their home network applies to everyone else |
01:18.29 | trixter | I dont think less of you for doing that |
01:18.44 | mzo_ | my work asterisk is flakey :P |
01:18.47 | Err | that's big of you |
01:19.11 | trixter | well I didnt want you to start boasting about some mythical work you do to justify what you said, after all that isnt proof that you are even right |
01:19.11 | Err | I don't think less of you for assuming that I'm an idiot who's never worked on a "real" network |
01:19.14 | trixter | its just ego boosting |
01:19.20 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net) |
01:19.27 | Netgeeks | whoah |
01:19.29 | trixter | so I felt that if I boosted your ego by saying I didnt look down of you for thinking opnly in terms of home networks you wouldnt do that |
01:19.47 | trixter | after all only idiots will say some mythical work they do as proof they are right |
01:19.49 | Err | yes, you are indeed God's gift to networking - it's clear to me now |
01:19.59 | mzo_ | no wait, i'm gods gift to networking! |
01:20.02 | Netgeeks | there is some massive testosterone flying about |
01:20.17 | trixter | I never claimed that I did mythical research work in tcp and thus was gods gift to networking |
01:20.20 | [av]bani | penis wars |
01:20.20 | trixter | I left that for the ego impaired |
01:20.21 | Qwell[] | Netgeeks: NO, you're thinking the other one... |
01:20.25 | Qwell[] | what was it called? |
01:20.32 | Qwell[] | ahh yes...bullshit |
01:20.43 | Netgeeks | hehe, it's hard to tell the difference, Qwell |
01:20.48 | Qwell[] | Netgeeks: indeed |
01:20.50 | trixter | after all what validity does a comment about some mythical made up research job have in proving anything? |
01:21.08 | [av]bani | this reminds me of the rednecks who were redlining their engines in an attempt to prove who had the larger penis |
01:21.29 | [av]bani | whoever had the louder engine had a bigger penis |
01:21.31 | trixter | there *is* a difference between a managed network and home dsl or just inet links, that is all I said, and I was quite clear about it |
01:21.39 | Err | trixter: http://www.ietf.org/internet-drafts/draft-allman-rto-backoff-02.txt <-- I'm the first author |
01:21.54 | Err | it's not "mythical" |
01:22.00 | Qwell[] | best comeback EVER |
01:22.03 | Qwell[] | :P |
01:22.08 | trixter | I dont know that you are true, I could pull up a link to the constitution and claim I am the first signer |
01:22.19 | trixter | lemme see if 'err' is listed |
01:22.34 | trixter | nope its not |
01:22.37 | [av]bani | yay penis |
01:22.41 | Err | whatever |
01:22.42 | trixter | you sure proved me wrong |
01:22.54 | Qwell[] | hostname matches |
01:22.57 | trixter | that still doesnt address the core issue which you are now trying to avoid |
01:22.59 | Err | of course my hostname matches |
01:23.10 | Err | what core issue might that be? |
01:23.11 | trixter | I really dont think less of you for thinking only in terms of home dsl and not properly managed networks |
01:23.15 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-4.nas28.salt-lake-city1.ut.us.da.qwest.net) |
01:23.54 | fiber0pti | What is the best way to include include or goto a context depending on the hour of the day, day of the week or date? |
01:24.01 | trixter | oh that you were trying to base decisions and comments on home networks despite my continued comments about properly managed networks |
01:24.11 | trixter | fiber0pti: gotoiftime ? |
01:24.15 | Err | oh, right, because packet loss never happens in "properly managed" networks |
01:24.26 | Netgeeks | fiber: are you asking if there is an option in addition to GotoIfTime? |
01:24.34 | fiber0pti | ah..thank you.. didn't know gotoiftime existed |
01:24.36 | Qwell[] | can't include do stuff based on time? |
01:24.44 | Qwell[] | or something like that |
01:25.04 | trixter | Err: depends on the network manager I guess |
01:25.07 | Netgeeks | I lost a packet the other day at the store, I had to go back and get it, but by the time I got there they had discarded it. |
01:25.14 | Netgeeks | They gave me a new one tho |
01:25.26 | trixter | you can build a network that doesnt have such issues, you just have to think about the problem a little differently |
01:26.00 | Qwell[] | mythical no packetloss network? |
01:26.11 | dudes | I've never heard of one |
01:26.13 | Err | yeah - Shannon talks about it a lot in his information theory books :-) |
01:26.16 | Netgeeks | don't use packets, and you can't lose any |
01:26.34 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
01:27.28 | Math[laptop] | Qwell[]: LANs usually don't suffer from packet loss |
01:27.35 | Qwell[] | usually != never |
01:27.37 | trixter | if you send packets in the same old way and deal with them in the same old way you have the same old problems |
01:27.59 | trixter | there are solutions but they dont work well over unmanaged networks like the internet (unmanaged from the endpoints perspective) |
01:28.11 | Math[laptop] | well as long as you don't do rate-limiting and that your links aren't maxed-out, there's none |
01:28.16 | Math[laptop] | except in a case of hardware failure |
01:29.04 | wilymage | ...or crap hardware |
01:29.36 | Err | or bit corruption - which MUST happen |
01:30.12 | Math[laptop] | bit corruptions? |
01:30.35 | Math[laptop] | transmission errors on an ethernet network is handled by the card itself, the faulty packet is discarded and retransmitted automaticly |
01:30.41 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
01:30.43 | mzo_ | hehe, i just made a 75 minute call on asterisk, using my fxo. yay. ;) it's so nice when it just works. |
01:30.51 | Math[laptop] | hehe :) |
01:30.52 | Netgeeks | cosmic ray beans a bit in the wire, changing it |
01:30.58 | rayvd | Anyone here using adelphia.net? |
01:30.59 | Qwell[] | Math[laptop]: I think that's the issue |
01:31.04 | Qwell[] | rayvd: god no |
01:31.09 | Qwell[] | rayvd: Need help? |
01:31.11 | Err | Math[laptop]: not necessarily - COLLISIONS are detected by CSMA/CD networks, but bit corruption can be unrelated to collisions |
01:31.21 | Netgeeks | I used to work for adelphia, and I stayed at a holiday inn express last night |
01:31.24 | mzo_ | what's the other kind of network besides CSMA/CD? there's some other kind |
01:31.25 | rayvd | just want to prove to my boss that email from adelphia.net is not being rejected by our servers :) |
01:31.26 | Qwell[] | I'm on a first name basis with their CEO and a bunch of execs :p |
01:31.39 | mzo_ | haha, and i'm calling another asterisk server to leave voice mail now. ;) |
01:31.41 | rayvd | so wanted to find someone taht can use their mx server |
01:31.46 | *** join/#asterisk dan_ (n=dan@202.147.139.26) |
01:31.50 | Err | well, wireless is CSMA/CA |
01:31.57 | Qwell[] | rayvd: call support, tell them to try |
01:32.08 | Err | there are TDM networks that don't do any CSMA whatsoever |
01:32.15 | rayvd | ooh, that's a swell plan |
01:32.27 | Err | (there are a million types of networks) |
01:32.36 | dan__ | howdi peoples... just wondering if anyone knew if a cisco AS5200 loaded with MICA modems can be used in conjunction with an asterisk to do SIP ? |
01:32.40 | mzo_ | yeah that's it CA. I forgot. It's something i read years and years and years ago. ;) |
01:33.27 | mzo_ | is there a reccomendation for an OSX Softphone application? |
01:33.35 | Qwell[] | mzo_: idefisk works |
01:33.35 | litage | what does it mean when someone says "this channel has only one leg"? |
01:33.48 | mzo_ | ill try that. ;) |
01:34.04 | mzo_ | is that spelled right? |
01:34.07 | Qwell[] | litage: means it only goes between a phone and the pbx |
01:34.12 | trixter | I am still waiting for the great research god of all things networking to explain this comment: Err or bit corruption - which MUST happen |
01:34.23 | trixter | specifically his emphasis on the use of the word 'must' |
01:34.37 | mzo_ | oh it's with a c, not a k |
01:34.46 | Qwell[] | no, it's idefisk |
01:34.56 | Err | trixter: see, this dude named Shannon wrote a bunch of papers on information theory, which proves that bit corruption must exist in any network - you can use ECC to lower the rate, but you can never totally eliminate it |
01:34.57 | mzo_ | heh google had it the other way :P |
01:35.03 | Qwell[] | well, google is wrong :p |
01:35.21 | trixter | so when using ECC a retransmission is required? |
01:35.26 | litage | Qwell[]: where/what else could the channel go between? |
01:35.33 | rt | well, that's sort of what it says, yes. |
01:35.34 | Qwell[] | litage: another phone |
01:35.40 | Qwell[] | two phones, no pbx between |
01:35.42 | trixter | your whole point was about retransmissions, so taken in context according to what you have said parity checking wouldnt be able to compensate |
01:35.46 | Err | no - link-layer retransmission is completely different - and also doesn't play particularly nicely with TCP, because it jacks with the RTT calculations |
01:36.00 | rt | what it really means is that for any given communication channel, there is an associated capacity that depends on the bandwidth and the signal to noise ratio of the channel. |
01:36.01 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
01:36.18 | litage | Qwell[]: how can 2 phones communicate without some sort of pbx/softswitch in the middle? |
01:36.19 | Err | no, *TCP* retransmissions are unrelated to link-level retransmissions, and care caused by packet loss |
01:36.22 | rt | any attempt to transmit beyond this capacity cannot succeed with greater chance than just guessing the bits. |
01:36.25 | Qwell[] | litage: easily |
01:36.38 | trixter | ahh so this stuff about bit corruption doesnt really apply |
01:36.39 | trixter | I see |
01:36.43 | litage | Qwell[]: oh, using something like enum? |
01:36.49 | Qwell[] | litage: no |
01:37.03 | trixter | I think I finally understand you now, when you couldnt back up your comments you went on a tangent to try to divert attention away |
01:37.04 | Qwell[] | just dial the user@hostname of another phone |
01:37.04 | litage | Qwell[]: woud you care to shed some more light on this for me please? |
01:37.08 | trixter | you sure showed me up |
01:37.09 | Err | trixter: yes, you're right - I don't know anything about computer networks; I'm sorry for having tried to help, when I'm clearly a fool |
01:37.18 | litage | ah =P |
01:37.30 | trixter | well trying to compare apples to oranges after you couldnt back up your claim doesnt help your image any |
01:37.33 | Qwell[] | litage: works fine with sip and iax, most phones |
01:37.37 | litage | Err: don't insult yourself dude. it doesn't do anyone any good |
01:37.41 | trixter | it gave me a laugh though |
01:37.45 | trixter | oh and btw now I do think less of you |
01:37.59 | Netgeeks | I'm laughing too, but not for the same reason |
01:38.25 | trixter | not becuase you only think in terms of home dsl links going over the internet but instead becuase you tried to cover up when you were wrong and hide the fact later finally admitting that bit corruption (your holy grail to prove your point) was not quite accurate for that point |
01:39.51 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
01:40.12 | family | yea so hmmm looks liek my problem with astcc is it cant connect to the database |
01:40.22 | Qwell[] | family: That'd do it |
01:42.02 | family | but on the astcc wiki it doesnt say whats needed for the db connected... odbc and what not |
01:42.10 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
01:42.12 | inv_Arp | damn i need some good caribbean rates |
01:42.23 | *** join/#asterisk prufrock (n=orange80@252.84.cm.sunflower.com) |
01:42.25 | mzo_ | ooh, ty for the softphone thing it works |
01:42.36 | *** join/#asterisk Hali_303 (n=surfk@dsl51B6ACC5.pool.t-online.hu) |
01:42.42 | Hali_303 | hi! |
01:43.33 | Netgeeks | what a quotable quote! "any attempt to transmit beyond this capacity cannot succeed with greater chance than just guessing the bits." not sure what it meant in context, but out of context, it's pretty funny |
01:43.49 | Err | heh, it's actually true, too |
01:43.53 | Err | Shannon ruled |
01:44.30 | [av]bani | guh, can the cross conversation leakage stop? i put err and trixster on /ignore so i wouldnt get this stupid babble |
01:44.56 | Netgeeks | Back in 'the day' I worked with some sun hardware running SunOS 4.1.3 or so, I don't remember the exact version, and they had a great quote for the option to shutdown that would stop the computer without syncing disks |
01:45.11 | mzo_ | suicide? :P |
01:45.19 | trixter | Netgeeks: yeah it also goes to prove my point about 'properly managed networks' which by definition dont exceed their allowable bandwidth |
01:45.28 | Hali_303 | does anyone here know SIP on a protocol level? my question: how to setup a SIP request like "send the reply back to the same IP and same port from where you have received this" and also for the RTP: "initiate an RTP connection to the ip and port where you receive and rtp connection".. this way I could get around if only one of the hosts is behind a symmetric NAT. any ideas? |
01:45.32 | *** join/#asterisk dsasda (n=OIGEORGE@ool-43540102.dyn.optonline.net) |
01:45.36 | dsasda | http://www.valvehacks.zaccum.com/ - great cs dod hl2 (valve) hacks |
01:45.39 | Netgeeks | it said something like this: "this option is used to shut down the computer without syncing the disks. You should only use this when the CPU is on fire" |
01:45.43 | Qwell[] | Netgeeks: like shutdown -n? |
01:45.49 | mzo_ | hahahah, that's a real man(tm) option |
01:45.56 | trixter | Netgeeks: are you refering to going into prom mode by pressing stop-a (or l1-a depending on keyboard) |
01:46.16 | trixter | that sounds like something that was in the prom |
01:46.20 | Netgeeks | nah, it was in the man page for the shutdown command |
01:46.33 | trixter | ahh |
01:46.39 | trixter | there were a few things that sun did back in those days |
01:47.09 | trixter | the screensaver had a backdoor of 'hasta la vista' to bypass anyones password, the prom actually was interesting in itself becuase you could do forth (it still should support that at least) |
01:47.10 | Netgeeks | now I also worked with some Apollo computers, and they had a error message in the error.txt file that gave you text error descriptions for error codes |
01:47.13 | trixter | and a few other things |
01:47.24 | trixter | vax had the best WAY back in the day |
01:47.26 | Netgeeks | I don't remember the code number, but the description was "Won't fit through an 18 inch hatch" |
01:47.38 | trixter | if 'love' was not a valid make target and you typed 'make love' it would reply 'not war?' |
01:47.48 | trixter | likewise if you did make war it would reply not love? |
01:48.06 | Netgeeks | apparently they had lost a navy contract after putting a ton of cash into the development because they failed to meet the spec requirement that the computer fit through a standard 18 inch hatch |
01:48.18 | *** part/#asterisk dsasda (n=OIGEORGE@ool-43540102.dyn.optonline.net) |
01:49.13 | trixter | all I have to say is I am glad the 70s are over and I no longer have to 'attach' to a drive before using anything off it |
01:49.42 | Qwell[] | What, you mean like mounting it? |
01:49.53 | trixter | no in a prime you have to attach its like doing c: |
01:50.40 | Qwell[] | oh, so a command you have to type before you can access files on it |
01:50.47 | Netgeeks | At my first real post college job, we had a computer that you had to boot by entering binary based commands using 10 flip-switched and 10 lamps on the front panel |
01:50.51 | Qwell[] | not like mount at all |
01:50.56 | trixter | the seperate drives like windows does is actually rooted WAY back when, its not anything remotely current or even that handy, but I guess its less confusing to some |
01:50.58 | justinu | prime? |
01:51.11 | trixter | prime computer made mainframes back in the day they went under in the 80s |
01:51.18 | trixter | right after a buyout the new owner drove it into the ground |
01:51.24 | justinu | yeah, i've read about them |
01:51.43 | trixter | their pixel was used in 1 star trek TNG episode though... they were doing a trade show in LA and a producer saw some stupid ass grfx and wanted one |
01:51.58 | trixter | yeah I was using em in the 70s... your 'read about em' comment is um ... :P |
01:52.14 | trixter | prime also had their own private world wide network which was also fun |
01:52.28 | trixter | not much to do then the intarweb wasnt around.. but it was fun to talk to people |
01:52.36 | Qwell[] | bbl, hom |
01:52.37 | Qwell[] | e |
01:52.52 | justinu | i was born in 76 |
01:52.57 | trixter | I was born in 72 |
01:53.16 | trixter | it just so happens in the late 70s my father got a job doing education for prime so I got to play |
01:53.37 | trixter | that is also when I started programming, I must say dartmouth basic on a prime isnt as much fun as languages today |
01:53.49 | justinu | so you were playing with a corporate mainframe before age 10? |
01:53.54 | trixter | yup |
01:53.58 | justinu | heh, nice |
01:54.25 | Math[laptop] | heh I was coding under the age of 10 |
01:54.31 | trixter | my first computer was built by my dad (mom soldered the keyboard together) |
01:54.34 | Math[laptop] | well I started some qbasic at 8 |
01:54.43 | justinu | i was keying in basic programs on my ti/994a from "99er magazine" |
01:54.51 | trixter | MS made a lot of money rewriting the io routines in dartmouth basic |
01:55.00 | justinu | i dunno if you would call that programming |
01:55.04 | trixter | qbasic wasnt that bad compared to what I used.. infact it was a lot better in many ways |
01:55.16 | Math[laptop] | heh |
01:55.36 | Math[laptop] | the first thing I did is make some sort of siren sound using a for loop and the SOUND instruction |
01:55.54 | Netgeeks | heh |
01:55.54 | trixter | I took a programming class in um 7th grade and we had to type in code from a magazine that was soooo boring |
01:56.03 | justinu | i also had one of those radio shack 3000-n-1 electronics kits |
01:56.42 | trixter | the teacher and I didnt get along because she was trying to teach the difference between print 4+2 and print "4+2" and I was making the result blink and fly across the screen, that lasted 1 week before I switched to theaqter (it was that or AG and I didnt want to raise livestock - yes it was a rural farming area school) |
01:56.46 | SibRphrek | how does asterisk handle 911 calls? |
01:56.54 | trixter | justinu: lucky I only had a 50:1 |
01:57.06 | justinu | something like that |
01:57.07 | trixter | SibRphrek: it technically doesnt, it hands it off to someone who does |
01:57.09 | justinu | it was really cool tho |
01:57.10 | Err | SibRphrek: it doesn't do anything out-of-the-box - you have to put in rules for it |
01:57.12 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
01:57.19 | SibRphrek | Err: which conf is that? |
01:57.34 | trixter | SibRphrek: what exactly do you want to do? |
01:57.35 | Netgeeks | sib: Asterisk doesn't know the difference between 911 and 123 or xxx until you tell it to |
01:57.41 | dlynes | SibRphrek: Your upstream has to be able to handle 911, or you have to have a contract with a call center that does it |
01:57.57 | trixter | map it so when someone dials 911 it calls? or set up the E911 information data? it doesnt do the E911 information stuff out of the box, you have to get a service provider for that |
01:57.58 | SibRphrek | i just wanna make sure if i dial 911, it gets somewhere |
01:58.08 | dlynes | SibRphrek: and of course you have to submit your 411 data to them so that they have something to send to the 911 call center |
01:58.12 | trixter | extensions.conf typicaly |
01:58.31 | Netgeeks | Then you will need to create a dialplan that captures your 911 calls and routes them where you want |
01:58.39 | dlynes | SibR: exten => _911,1,... |
01:59.05 | trixter | I prefer to route them to the local pizza place, figured since its a take and bake if they call about a fire it would all work out |
01:59.12 | Math[laptop] | dlynes: no _ |
01:59.14 | Netgeeks | the very basic for a home office install with an analog card would be something as simple as 'exten => 911,1,Dial(Zap/1/911) |
01:59.31 | trixter | Math[laptop]: that shouldnt hurt anything since there isnt any wildcard stuff ... but yeah |
02:00.01 | Netgeeks | but then you need to make sure all of the inbound context for any handset you might dial 911 from has the above extension available to it |
02:00.20 | trixter | Netgeeks: what if someone is on line 1? like the burglar outside your business trying to break in just so you cant call out :P |
02:00.20 | Netgeeks | and in order, so if you are allowing _9X. you better check 911 before that |
02:00.48 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
02:00.51 | Netgeeks | trixter: thus my qualifying 'very basic' prefix to the rest of the comment |
02:01.04 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
02:01.22 | Netgeeks | i didn't want to lead into a possible discussion of what G1 meant |
02:01.28 | dlynes | We generally won't sell a client a system unless they have at least one hard line |
02:01.39 | trixter | extensions handling doesnt do real good matching where multiple targets exist that both wildcard match, it may match correctly one time it may not the other, version seems to be the biggest determiner of that |
02:01.45 | dlynes | That way they can still dial 911, when there's no network or power or whatever else |
02:02.06 | trixter | so its better to extend your wildcard _9X. to _9XX. (. means 1 or more) so you dont have a dup match |
02:02.17 | *** join/#asterisk Flyboy (i=rsears@gateway.adnc.com) |
02:02.40 | Flyboy | Hey Everyone |
02:02.52 | Netgeeks | did any of that help at all SibRphrek? |
02:02.53 | justinu | hi flyboy |
02:03.00 | Flyboy | got a Digium TDM40B quiestion is anyone has some time |
02:03.07 | Math[laptop] | just ask it |
02:03.12 | trixter | I have one of those in a box waiting to be given away |
02:03.17 | clyrrad | I am having a really strange problem, I have 6 DID's all configured the exact same way in iax.conf, and extension.conf. Only 3 of the DID's are able to accept incomming, and all can make outgoing. I get the error Rejected while trying to reach 'DID@' instead of it saying 'DID@incoming' Can anyone suggest what can be the problem? |
02:03.25 | Netgeeks | I'd use it as a paper weight |
02:03.29 | trixter | it looks quite secksi but .. |
02:04.03 | dlynes | clyrr: make sure in your channel config that you have a context |
02:04.21 | clyrrad | dlynes, i have the context in iax.conf |
02:04.25 | clyrrad | is that what you mean? |
02:04.46 | dlynes | yeah...make sure all of your iax peers/friends/users have a context specified |
02:04.46 | Flyboy | Installed the 40B card, got all modules loaded, ztcfg -vv checks good, running 1.2.3, all the latest drivers from cvs installed no problem and I get no dialtone on my phone attached to the ports (which are greeg) and when I try to dial the extension attached I get a "Unable to create a channel type of 'Zap" |
02:04.52 | dlynes | and that's it's not overridden |
02:04.59 | clyrrad | dlynes, I have checked they all have context's |
02:05.04 | Netgeeks | do you see any CLI message that looks something like 'unable to find match for <insert something here>, using default context.? |
02:05.19 | clyrrad | Netgeeks, nope nothing like that |
02:05.26 | dlynes | you might have it coming into a non-existent extension, too |
02:05.31 | trixter | Flyboy: zap show channels does that show anything? I am almost thinking that chan_zap.so isnt loaded in modules.conf |
02:05.44 | dlynes | which is probably what netgeeks was insinuating |
02:05.52 | clyrrad | dlynes, sorry didnt quite follow you on the last part |
02:06.03 | SibRphrek | WTF |
02:06.05 | clyrrad | they all have context=incoming in iax.conf |
02:06.06 | Flyboy | trixter, when I run zap show channels from asterisk, it tells me No such caooman 'zap' |
02:06.17 | mzo_ | caooman? |
02:06.19 | trixter | you dont have chan_zap.so loaded |
02:06.23 | Netgeeks | sounds like the peer/user matching algorithm for determining which iax entry to use when getting a call from an remote iax speeking device may not be able to determine then entry you want it to use |
02:06.24 | trixter | edit modules.conf to make that happen |
02:06.40 | dlynes | clyrrad what is the extension they're trying to reach though? Is it a valid extension in your incoming context? |
02:06.42 | clyrrad | Netgeeks, what do you suggest? |
02:06.44 | SibRphrek | http://pastebin.com/531497 someone please help me understand this? |
02:06.51 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
02:06.57 | dlynes | clyrrad: make sure you don't have a #include statement that includes another context inside of your [incoming] context, too |
02:07.01 | clyrrad | dlynes, yes its the DID in the form of [DID] |
02:07.28 | Netgeeks | clyrrad capture cli output, with iax debug on, verbosity set 3 or higher, and include the iax.conf section and the context in extensions.conf in a pastebin |
02:07.45 | trixter | ~pb |
02:07.47 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
02:07.52 | justinu | http://www.sky.com/skynews/picture_gallery/picture_gallery/0,,70141-1210748-1,00.html |
02:08.09 | Netgeeks | that should give enough info to get a handle on the issue |
02:08.15 | clyrrad | Netgeeks, I have done that, the error i get is CAUSE: No authority found |
02:09.06 | clyrrad | and all the DID's can make outgoing calls with out issue, its just when they come in. So I think the problem is in iax.conf somehow? But all DID contexts are identical just username and secret are different |
02:09.52 | Flyboy | trixter, I added load => chan_zap.so to /etc/asterisk/modules.conf and restarted Asterisk - it crashed |
02:09.55 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
02:10.40 | litage | Math[laptop]: you said earlier that SETGROUP() and CHECKGROUP() have been deprecated by GROUP() and CHECKGROUP(). however, voip-info.org says that CHECKGROUP() is only in v1.0.x |
02:11.00 | Netgeeks | clyrrad: that leads me close to the belief that the incoming iax request is not finding a match (peer, user, or friend) from all the entries in your iax.conf |
02:11.14 | austinnichols10 | do start/end media ports on the server need to match the phone (cisco 7960)? Server is 10000/20000, phone is 16384/32766. |
02:11.15 | Flyboy | trixter - also there is no such file on my system (chzn_zap.so) |
02:11.33 | Netgeeks | I'd have to see the above requested output to be any more help, clyrrad |
02:11.43 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
02:12.04 | clyrrad | Netgeeks doing it now |
02:12.09 | Seedy | I am a total newb so please excuse my ignorance... |
02:12.19 | Netgeeks | the wiki at www.voip-info.org has some valid text on how the selection is made when looking for a user/peer/friend match, that may help... |
02:12.27 | Err | SibRphrek: it looks to me like your SIP control packets aren't getting through to the other end, or the other end's responses aren't getting back, and so the channel is being torn down |
02:13.02 | Seedy | Can an sip.conf file be used for incoming calls too? Or does iax.conf handle that? |
02:13.04 | Netgeeks | Seedy, you are in good company, we are all newbs at something here |
02:13.13 | trixter | Flyboy: I hpe that you see the typo :P |
02:13.17 | trixter | er hope |
02:13.21 | Netgeeks | sip and iax are two different signalling protocols |
02:13.26 | clyrrad | Netgeeks, here it is http://pastebin.ca/39196 |
02:13.53 | trixter | if you dont have a chan_zap.so you need to build asterisk with that |
02:14.12 | Netgeeks | and I'm half wrong there, iax actually does signalling as well as media |
02:14.20 | Netgeeks | looking at it now cly |
02:14.25 | Flyboy | trixter, :-) WHat type.. |
02:14.33 | Flyboy | trixter, :-) WHat typo |
02:14.47 | trixter | was it a typo or is it really not there? |
02:14.49 | Netgeeks | so yes, sip handles both 'incomming' and 'outgoing' calls as well as iax |
02:15.17 | Flyboy | trixter, I did an updatedb and locate chan_zap and just the the chan_zap.c file |
02:15.25 | Flyboy | trixter, nothing else |
02:15.40 | trixter | when you built asterisk did you already have the zaptel stuff installed? |
02:15.46 | trixter | and most likely libpri |
02:15.49 | Flyboy | nope, but I can build again if necessary |
02:15.58 | trixter | you may have to |
02:16.04 | Flyboy | trixter, OK - I will try that now |
02:16.09 | trixter | becuase it may not build that if the dependancies arent t here, I know I wouldnt if they arent there :P |
02:16.12 | Flyboy | trixter, thanks and I will elt you know how it goes |
02:16.17 | *** join/#asterisk dijit0 (n=eric@adsl-69-106-42-147.dsl.pltn13.pacbell.net) |
02:16.38 | Netgeeks | clyrrad, and you have a iax.conf entry that starts with [4168484163] and has host=64.26.157.230 ? |
02:17.06 | dijit0 | is nufone or iax.cc any good? or can anyone recommend something better/cheaper? |
02:17.11 | clyrrad | yes, I have it matched exactly as you just said |
02:17.26 | Netgeeks | clyrrad and what is type= set to? |
02:17.29 | Flyboy | dijit0, I am using NuFOne |
02:17.37 | clyrrad | peer |
02:17.39 | Flyboy | dijit0, seems ok |
02:17.48 | Flyboy | I have about a dozen DIDs |
02:17.58 | clyrrad | just like all the other DID's 3 of which can get incomming calls |
02:18.04 | Netgeeks | clyrrad do you have a matching iax.conf entry with type=user? |
02:18.11 | Flyboy | dijit0, would be nice if they were closer to my servers in CA |
02:18.30 | clyrrad | Netgeeks, no they are all peer, the working ones and non working ones, thats how its been working all along |
02:18.41 | Netgeeks | okay, here is the issue |
02:18.42 | dijit0 | cali or canada? |
02:18.55 | Flyboy | dijit0, Calif |
02:18.59 | *** join/#asterisk svenl_ (n=sven@AStrasbourg-251-1-35-113.w82-126.abo.wanadoo.fr) |
02:19.03 | dijit0 | ahh alright, lol, thats where im at |
02:19.14 | Netgeeks | when an iax request comes in, it's going to look for a iax.conf type=user or type=friend entry with a matching username, if it can't find that, it looks for a matching host= entry |
02:19.20 | dijit0 | i notice there are cheaper rates than nufone around... but i dont know how good any of them really are |
02:19.40 | Flyboy | dijit0, same here, I was going to try another carrier or two to see how well they worked |
02:19.54 | Netgeeks | if it finds neither, it is *supposed* to check the peer entries based on secret.... don't ask, I don't know who was on what drugs when they did that |
02:20.08 | clyrrad | lol |
02:20.14 | Flyboy | dijit0, where are you at in Calif..? |
02:20.24 | Netgeeks | anyway, the fact that you have no peer or interpeted peer via friend statement, is bad |
02:20.26 | clyrrad | Any idea why 3 are working setup like this but not the other 3? |
02:20.30 | Netgeeks | sorry, I mean user |
02:20.42 | Netgeeks | I would try first changing the non-working ones to type=friend |
02:20.44 | austinnichols10 | saw a good review of providers on mundy.org |
02:20.52 | dijit0 | bay area |
02:21.00 | Netgeeks | restart (I don't like reloads) your asterisk if possible and test |
02:21.03 | Flyboy | I am in San Diego |
02:21.06 | austinnichols10 | currently using voxee.com |
02:21.18 | Flyboy | thanks austinnichols10 |
02:21.20 | Netgeeks | clyrrad: bit corruption is as likely as anything |
02:21.21 | clyrrad | Netgeeks, ok going to give that a try, becase the contexts are used for incomming and ougtoing, what i mean is its all IAX for calls incomming or outgoing |
02:21.39 | Flyboy | how do you like voxee.com and do they hit you for multiple calls at once..? |
02:21.51 | Flyboy | I run a Friends and Family system with about 50 users |
02:21.54 | Netgeeks | the context statement is only valid for 'incomming' calls, it's ignored when you make an outbound call referencing a iax.conf entry that has one |
02:21.55 | Flyboy | non-profit |
02:22.13 | Flyboy | I like NuFOne becuase they could care less how many inbound and outbound calls I have at once |
02:22.31 | dijit0 | flyboy, and those are toll free numbers? |
02:22.32 | clyrrad | well i'll be damned! :p |
02:22.51 | clyrrad | set as type friend it works |
02:22.59 | austinnichols10 | flyboy: no - voxee is outgoing only |
02:23.04 | Netgeeks | Good to hear, Cly |
02:23.12 | austinnichols10 | flyboy: but they don't care how many concurrent |
02:23.20 | Flyboy | do they care how many outgoing connections you ahve at once..? |
02:23.26 | clyrrad | any security or related issues with using friend instead of peer? |
02:23.31 | Flyboy | dijit0, My inbound calls are 800 |
02:23.39 | Netgeeks | I read 'I like NuFOne becuase they could care less.....' and I stopped reading and agreed with the later haf of that phrase |
02:23.52 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
02:23.57 | dijit0 | id REALLY like a service that has inbound caller id, WITH NAME! lol |
02:24.31 | Flyboy | dijit0, sometimes we get caller ID from NuFone and sometimes we don't :-( |
02:25.06 | Netgeeks | clyrrad: type=friend is less perfect that having a type=peer and type=user, but the issue isn't that big, if you run into a problem with authentication then I would look into breaking the friend entries into peer and user, but for now... I wouldn't worry |
02:25.22 | Flyboy | Netgeeks, HA |
02:25.26 | dijit0 | i c... |
02:25.41 | clyrrad | Netgeeks, so type user is used for incoming and peer for outgoing? |
02:26.02 | *** join/#asterisk unixgeek_ (n=unixgeek@216-220-234-197.exploremaine.com) |
02:26.11 | Netgeeks | type=user is scanned when a call request is recieved from a remote party (read iax box calling you in this case) |
02:26.53 | Netgeeks | type=peer is used when you reference it via a dial command, such as Dial(IAX2/acme/${EXTEN}) |
02:27.05 | Netgeeks | type=friend means I'm both a peer and user entry |
02:27.17 | wilymage | hmm, any good papers on stopping denial of service attacks using asterisk? (i.e. someone spawning multiple IAX calls from multiple hosts simultaneously) |
02:27.20 | clyrrad | gotcha, thanks so much for your help much appreciated ) |
02:27.48 | Netgeeks | in some cases, you need a username and secret for calling through a provider, but they expect you not to challenge them when they send a call back to you |
02:27.55 | Netgeeks | in that case a friend entry wouldn't work |
02:28.03 | Netgeeks | you'd need to split into user and peer |
02:28.16 | family | argh |
02:28.29 | clyrrad | Got it, thanks for clarifying :) |
02:28.35 | family | i upgraded mysql connectors perl dbi etc etc and astcc stillc ant connect to the db |
02:28.38 | clyrrad | Makes sense now |
02:28.45 | Netgeeks | no worries, good luck! |
02:28.54 | clyrrad | thanks again :) |
02:29.36 | Netgeeks | glad I could help, the problem you had can be pesky if you don't know what you are looking for |
02:29.39 | *** part/#asterisk unixgeek_ (n=unixgeek@216-220-234-197.exploremaine.com) |
02:30.15 | austinnichols10 | Anyone know if start/end media ports on the server need to match the phone (cisco 7960)? Server is currently 10000/20000, phone is 16384/32766 |
02:30.26 | clyrrad | yea tell me about it, i was looking at this for a couple hours before i decided to ask here |
02:30.56 | Flyboy | austinnichols10, I have a 7960 - How owould I tell |
02:31.00 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
02:31.27 | Netgeeks | the phone and the server should be able to negotiate for a valid media port |
02:31.35 | austinnichols10 | settings / sip / scroll down to 16 or 17 where the ports are shown |
02:31.42 | Netgeeks | can't say I've ever tried testing the fact |
02:31.49 | austinnichols10 | netgeeks: tks - that's kind of what I thought |
02:32.13 | Netgeeks | I wouldn't be suprised if either asterisk or the cisco or both even ignored the setting altogether |
02:32.15 | Err | it would be broken if it required the ports to match (since you're not guaranteed that the server has any ports free in whatever range is specified) |
02:32.15 | Seedy | I'm having trouble getting asterisk to recognize DTMF. I have a simple exten => _x,1,Playback(beep) in my extensions.conf. But it never gets triggered. Is this a common problem? |
02:32.40 | austinnichols10 | I'm just trying to debug a problem with SER on my Linksys (running dd-wrt). The start/end media ports are about the last thing on my list to check |
02:33.10 | Netgeeks | Seedy, there have been many many pubs who have been saved by bankruptcy by people trying to get asterisk to work with DTMF in a simple manner |
02:33.23 | Netgeeks | save from bankruptcy that is |
02:33.42 | *** join/#asterisk CoiL (n=bah@68.62.165.236) |
02:34.32 | Seedy | Netgeeks: So it should be easy? |
02:34.41 | wilymage | or perhaps given their woe saved *by* bankruptcy would be more appropriate ;) |
02:34.54 | Err | hm, can you do that without using some command to prompt for an extension to be input first? |
02:35.08 | Netgeeks | one could only wish it was easy, but IMHO echo and DTMF are the two most difficult asterisk issues to resolve |
02:35.17 | Err | (i.e. shouldn't use use Background() or some other mechanism to listen for tones before trying to key off of them...?) |
02:35.32 | Netgeeks | WaitExten would work as well |
02:36.02 | Netgeeks | I just assumed you had one.... |
02:36.02 | wilymage | musiconhold.c is the most buggy component, in my experience; it's the only thing that crashes our boxes on a regular basis |
02:36.23 | justinu | wow |
02:36.26 | Netgeeks | yeah, but that issue is easily fixed |
02:36.34 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
02:36.42 | Netgeeks | noload => res_musiconhold.so |
02:37.01 | Netgeeks | I think it's a resource |
02:37.01 | wilymage | and if one wishes to utilise musiconhold? |
02:37.11 | Flyboy | trixter, rebuild asterisk - still fails - messages now say - undefinded symboy: ast_pickup_call |
02:37.49 | wilymage | we wrote another module relying moreso on sox, but it merely stays up for longer before leaking then dying. |
02:38.02 | Netgeeks | make sure you are using the perfect mpg123 version (make mpg123 in asterisk source directory) and make sure you convert all the mp3's to the right format, mono 8k I believe |
02:38.11 | Math[laptop] | wilymage: musiconhold is crashing? or mpg123 is? |
02:38.17 | Netgeeks | the native stuff in 1.2 seems to be nice |
02:38.36 | *** join/#asterisk Triffid_Hunter (n=Splat@funkmunch.net) |
02:39.06 | wilymage | but yeah, with the native moh worked very poorly |
02:39.24 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
02:39.38 | wilymage | all of the standard distortion, etc., that timers and silence suppression are purported to fix. |
02:39.40 | Seedy | Netgeeks: So how does one debug dtmf. I don't even think asterisk is getting my phones signals. |
02:40.19 | Err | Seedy: does your dialplan have a default entry that waits for DTMF entries? |
02:40.25 | Flyboy | trixter, HA |
02:40.26 | wilymage | Triffid_Hunter ended up writing a piece of software that replaced using mpg123, which works amazingly for a few days, then eats the CPU and kills asterisk. |
02:40.37 | Flyboy | trixter, interesting - it fails to load the module, but now it works |
02:40.45 | Triffid_Hunter | lol asterisk eats the cpu, not sox |
02:41.09 | wilymage | Netgeeks: version of mpg123 was the correct clean version, as per the gentoo ebuild. |
02:41.20 | Triffid_Hunter | the one that baffles me is why asterisk insists on starting seven or more moh processes, even when sending only one stream to one place |
02:42.12 | Seedy | Err: I have this exten => _x,n,SayDigits(${EXTEN}), which I thought would echo my button being pressed. But it does nothing |
02:42.41 | [TK]D-Fender | Seedy : Pastebin the entire context for us please.... |
02:42.42 | [TK]D-Fender | ~pb |
02:42.44 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
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02:42.52 | Err | Seedy: is this coming from an analog line? you'll need something like "exten => s,1,Background(file-to-play)" or use WaitExten() (I don't know anything about it...) |
02:43.22 | Seedy | Err: Thanks... I'll try that |
02:43.30 | Err | [TK]D-Fender |
02:43.32 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
02:43.35 | Err | 's suggestion is probably better :-) |
02:43.39 | Netgeeks | waitExten(x) waits for a keypress for the period of <x> seconds |
02:43.52 | Err | if you show us your dialplan, someone here can almost certainly help out |
02:44.02 | [TK]D-Fender | Err : yeah... I don't trust single lines pasted like that.. I have no idea what other BS is in the breaking it :) |
02:44.04 | Seedy | Ok... One second |
02:44.26 | Err | [TK]D-Fender: I completely understand :-) |
02:44.49 | Netgeeks | Wily: how busy is your asterisk system that has this MOH / CPU eating problem? |
02:45.02 | family | exit |
02:45.35 | Seedy | Here is the conf file http://pastebin.com/531533 |
02:46.16 | Flyboy | lithi, I can't send private messages |
02:46.32 | Flyboy | zap show channels |
02:46.39 | lithi | Flyboy, ah 1 sec |
02:46.40 | [TK]D-Fender | Seedy : Got Autofalltrouhg=off I hope...... |
02:46.56 | litage | after a call has finished, how would you get asterisk to send the call's CDR to another box, or write to another file? |
02:47.10 | *** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
02:47.26 | Netgeeks | litage, cdr_mysql pgsql odbc, etc |
02:47.49 | Seedy | [TK]d-fener: Autofalltrouhg In my sip.conf file? |
02:48.54 | Seedy | the _NXXNXXXX numbers are replaced with my reail number in that file |
02:49.06 | Seedy | like 15551212 |
02:49.18 | Seedy | or 12125551212 i mean |
02:49.39 | Netgeeks | you are dialing in using a sip phone? |
02:49.58 | [TK]D-Fender | Seedy : in extensions.conf |
02:50.30 | Seedy | I am using a regular analog phone |
02:51.10 | Seedy | [TK]d-Fender: what does Autofalltrough do? |
02:51.53 | litage | Netgeeks: to be more specific, my CDRs are currently being stored in a mysql db using cdr_mysql, but i also need each a program to parse each cdr['s data] |
02:52.50 | Err | Seedy: I suspect that he typo'd Autofallthrough |
02:52.55 | *** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net) |
02:53.19 | Seedy | So, will my conf not work with an analog phone? |
02:53.41 | Netgeeks | litage, you could use mysqldump to remotely dump the contents of the cdr table, or you could even use a perl/php/python/ruby/c program to go and grab the specific entries you want... once in the db they should be easy to get at |
02:54.05 | Netgeeks | you could use scp to copy the text file cdr's from the log section |
02:54.05 | Err | well, you won't get an extension when dialed from an analog phone adapter |
02:54.38 | Err | (at least, that's my understanding - I don't have one) |
02:55.27 | Netgeeks | see my addition onto the end of your pastebin, seedy, swap out what you have for what I added, and you should get farther |
02:55.54 | Netgeeks | if you aren't sure, in incoming you could add this: |
02:56.01 | Netgeeks | exten => _X.,1,Goto(s,1) |
02:56.38 | Flyboy | trixter, THANKS MUCH - system works wonderful |
02:56.52 | Netgeeks | you could actually add _.,1,.... but you would suffer the ire of the asterisk warning gods in doing so |
02:56.52 | Flyboy | trixter, now I can hook it into my NEX IPS2000 :-) |
02:57.01 | SibRphrek | where do voicemail passwords get kept? |
02:57.07 | Netgeeks | voicemail.conf |
02:57.14 | Flyboy | or a MySQL db |
02:57.19 | Flyboy | :-) |
02:57.21 | Netgeeks | or a database if you are using such a..... |
02:57.30 | ptiggerdine | <PROTECTED> |
02:57.30 | trixter | Flyboy: np |
02:57.43 | wilymage | ptiggerdine: err? |
02:57.50 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
02:57.53 | Err | your /etc/passwd has extensions in it? :-) |
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02:58.05 | wilymage | whoa freaky |
02:58.17 | ptiggerdine | don't worry I just realised it was a bad idea as I pressed enter :) |
02:58.45 | Netgeeks | sounds like my first marriage |
02:58.52 | trixter | I wrote a pam module to tie the unix logins into the database system we had at a unified messaging company I worked at in 1999 |
02:58.55 | ptiggerdine | ROFL! |
02:58.59 | wilymage | Netgeeks: on-line marriage? |
02:59.10 | SibRphrek | i found my error |
02:59.11 | SibRphrek | thanks |
02:59.11 | ptiggerdine | he marrige a PBX |
02:59.14 | SibRphrek | still having problems calling out |
02:59.15 | Netgeeks | nope, not online.... |
02:59.21 | trixter | people werent fond of that, they wanted them to be seperate even if they were the same ... mostly I think its becuase it was 'strange code they didnt undertand' |
02:59.30 | ptiggerdine | ROFL! |
02:59.30 | Netgeeks | still, the similarities are amazing.... |
02:59.56 | wilymage | Netgeeks: just curious as to how you sealed your fate with the press of a key |
03:00.11 | ptiggerdine | pr0n |
03:00.20 | trixter | they freaked when I wrote javascript client to directly interact with the database and basically bypassed some of the stupid things the tool the DB guys gave tech support (which meant he couldnt do his job) iut was sanctioned code they just didnt know that you could do such a thing |
03:00.24 | trixter | live connecti s wonderful |
03:00.57 | Netgeeks | I'm just extending the thought... I would have made the same comment to 'I knew it was a bad idea as the front wheels passed over the edge of the cliff' |
03:01.11 | trixter | I think that scared them the most because it also showed them a huge gaping hole in their design |
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03:01.20 | wilymage | trixter: I'd freak out if I saw javascript anywhere near my important systesm . . . |
03:01.25 | trixter | and by gaping hole I mean like you might find on a 50 cent whore on the docks on payday |
03:01.55 | trixter | well this was for a webbrowser to parse, it just allowed network connectivity to the oracle webserver (which is art of the DB) |
03:01.59 | trixter | er part |
03:02.50 | wilymage | so you didn't use perl because ... ? |
03:03.05 | Err | probably because web browsers don't do client-side perl |
03:03.11 | Err | just a guess, here :-) |
03:03.17 | trixter | perl doesnt execute within a browser nearly as well |
03:03.18 | litage | Netgeeks: that requires checking the db every second though. i was thinking more along the lines of asterisk sending the CDR data to a program rather than a program periodically grabbing the data |
03:04.02 | wilymage | perl doesn't execute within a browser at all, that's the wonder of it. |
03:04.14 | Netgeeks | litage: I think you can dump cdr to the manager api (don't quote me on this). you could then have your remote app connected and listening on the manager api |
03:04.15 | trixter | it might, wouldnt suprise me if someone by now has written a plugin of some sort |
03:04.25 | trixter | but it certainly didnt in 1999, at least none that I was aware of |
03:04.52 | trixter | Netgeeks: do you mean concurrent? if it werent I would be afraid of losing CDR |
03:04.56 | Netgeeks | litage: my knowledge in cdr to manager api comes from either reading it somewhere or having dreamt that i read it somewhere |
03:05.20 | Netgeeks | trixter: I would hope it could be done in parallel with permanent storage methods |
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03:06.03 | Seedy | So... An analog phone won't register extensions with asterisk from what I have heard. So how can I get it to intercept what numbers I am pressing!?! |
03:07.11 | Netgeeks | the only time I worked with analog phones, I used DISA and immediate=yes to capture the dtmf entries |
03:07.28 | Netgeeks | see immediate= directive information in your sample zapata.conf file |
03:07.40 | Netgeeks | and DISA is an app you can read about at www.voip-info.org |
03:07.46 | trixter | at the very least you can do a 'userevent' |
03:07.59 | trixter | in your dial plan, but it might be better to watch call creation and termination |
03:08.07 | Netgeeks | again, I warn you that I am no where near an expert in analog phone and asterisk... more like a newbie |
03:08.13 | Math[laptop] | I just make up a digitmap for the ATA |
03:08.18 | Netgeeks | so take anything I say with a grain of salt and do your homework |
03:08.58 | trixter | well what you proposed can be done |
03:08.58 | *** join/#asterisk whoknows (n=nav_swt@cpe-70-117-5-47.satx.res.rr.com) |
03:09.08 | trixter | however it may require a userevent if the functionality isnt already there |
03:09.42 | whoknows | need some help in configuring pri for two separate phone lines using separate context |
03:09.45 | trixter | although for the most part you can watch the call setup/teardown with the manager api and do it that way, you may not have *all* the cdr info but you would have a general overview |
03:09.52 | trixter | I personally like tossing CDR into a DB and accessing it that way |
03:10.03 | Seedy | Oh, just to make things clear too. My asterisk only accepts incoming calls. And I am making these calls from an analog phone |
03:10.06 | *** join/#asterisk {ss}Another (n=dob@kec1130-01.engr.oregonstate.edu) |
03:10.07 | {ss}Another | <PROTECTED> |
03:10.09 | *** part/#asterisk {ss}Another (n=dob@kec1130-01.engr.oregonstate.edu) |
03:10.17 | Netgeeks | he wants push methodology versus a pull, though, Trixter |
03:10.37 | trixter | ahh there is an app for that |
03:10.42 | trixter | forget its name (broadcast?) |
03:11.08 | trixter | basically its a message thing where the server sends arbitrary messages as defined in your dial plan, its not even in add-ons but it does exist, I recall the author saying something um ... a year ago on asterisk-users |
03:11.11 | trixter | maybe only 6 months ago |
03:11.16 | whoknows | can anyone help this newbie with pri card |
03:12.23 | whoknows | configuring pri for two different companies phone number |
03:12.28 | whoknows | pri is 16 channel |
03:12.41 | whoknows | need help in zapata conf file |
03:13.08 | trixter | by send I mean it has a bunch of programs that listen and you select which get the message ... thought my explanation wasnt exactly clear earlier |
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03:34.32 | JunK-Y | hey hey angler__ . |
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03:39.40 | mattwj2006 | so can you make a call on an actually asterisk box (not a sip phone or iax softphone) using a soundcard |
03:39.49 | mattwj2006 | if so how do you do it? |
03:40.41 | Math[laptop] | dial extension@context |
03:40.42 | Math[laptop] | on the cli |
03:41.24 | mattwj2006 | do you have to load any special kernel mods? |
03:41.50 | mattwj2006 | ex zaptel |
03:42.20 | Err | you need soundcard drivers that provide the oss or alsa API, and load chan_oss.so or chan_alsa.so (whichever is appropriate) |
03:43.33 | mattwj2006 | awesome :) |
03:44.31 | litage | Netgeeks: hahah thanks for that. i'll look into the manager api |
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03:57.16 | whoknows | asterisk cannot detect if the phone is busy and go directly into voicemail |
03:57.28 | whoknows | is there any way to fix this guys |
03:57.33 | Qwell | whoknows: define busy |
03:57.55 | whoknows | thks qwell, but how do i do that |
03:58.00 | whoknows | in extensions.conf file |
03:58.11 | Qwell | by typing words, that explain to me what you think "busy" means |
03:58.37 | Math[laptop] | whoknows: what kind of phone |
03:59.44 | whoknows | if the extensions is busy then the call should directly go into voicemail rather than ringing and then going to voicemail |
03:59.53 | whoknows | hope i make myself clear |
04:00.19 | Math[laptop] | what do you mean by busy |
04:00.25 | Math[laptop] | that's what he's asking |
04:01.20 | whoknows | well the person at 103 is on phone and talking and if you are calling 103 then it should go to voicemail |
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04:21.31 | *** topic/#asterisk by drumkilla -> Asterisk 1.2.4 and Zaptel 1.2.3 Have been released! => Includes a significant memory leak fix for Asterisk |
04:22.08 | *** join/#asterisk coppice (n=chatzill@46.155.17.210.dyn.pacific.net.hk) |
04:22.15 | opsys | what had the mem leak asterisk or Zaptel? |
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04:22.27 | drumkilla | opsys: Asterisk |
04:22.31 | drumkilla | it was the expression parser |
04:22.39 | Qwell | like $[]? |
04:22.42 | drumkilla | yes |
04:22.44 | Qwell | nice |
04:23.03 | drumkilla | thank murf for fixing it and Corydon for getting it merged ... |
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04:33.25 | FuriousGeorge | wadup kids |
04:34.25 | FuriousGeorge | mogorman: que pasa |
04:36.47 | clyrrad | I have call forwarding app enabled on my * box and a sipura ata, the ata does not let me do the *21*forward_number# like other voip phones do, anyone know how to use call forard with this ATA? |
04:38.11 | Math[laptop] | *21*? isnt that on cells |
04:38.35 | clyrrad | no i have that enabled on my asterisk box |
04:38.40 | clyrrad | as teh call-forward app |
04:38.48 | Corydon76-home | and blame Corydon for not getting it into 1.2.3 |
04:38.58 | Flyboy | good night everyone !! |
04:41.41 | wasim | woo hoo ... 1.2.4 |
04:42.57 | SwK | 1.2.4? |
04:42.59 | SwK | already? |
04:43.05 | fugitivo | well, i knew this was going to happen |
04:43.17 | fugitivo | that's why i stick with 1.2.1 |
04:43.17 | fugitivo | i'll wait for 1.2.5 |
04:43.28 | Corydon76-home | 1.2.1 also has the memory leak |
04:43.33 | fugitivo | i know |
04:43.39 | fugitivo | but who knows what has 1.2.4 |
04:43.40 | SwK | asterisk 1.X.X has a memory leak |
04:43.57 | Corydon76-home | No, it's just 1.2.x prior to 1.2.4 |
04:44.16 | Corydon76-home | 1.0.x had a significantly different expression parser |
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04:47.15 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
04:47.34 | __Krush__ | Hi all, anyone know a source for Grandstream ringtones? |
04:48.12 | *** part/#asterisk __Krush__ (n=chatzill@195.158.84.170) |
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04:48.47 | __Krush__ | Hi all, anyone know of a source for good Grandstream rintones? |
04:48.56 | __Krush__ | ringtones that is |
04:49.48 | robbyt | krush: a moog and an ftp server |
04:50.07 | __Krush__ | moog? |
04:50.08 | *** part/#asterisk mattwj2006 (n=Matt@dialup-4.254.83.80.Dial1.Chicago1.Level3.net) |
04:50.18 | robbyt | google it ;) |
04:52.10 | __Krush__ | google "grandstream ringtones" yields only 2 entries in a forum with a similar request... |
04:52.35 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
04:52.48 | knight_ | hey SwK! |
04:53.00 | robbyt | krush: my point is, i haven't been able to find much for ring tones outside of making them my self |
04:53.19 | __Krush__ | ah...I'm not the only one then... |
04:53.29 | robbyt | krush: http://freesound.iua.upf.edu/ |
04:53.32 | Corydon76-home | robbyt: Wendy Carlos would be proud |
04:53.43 | robbyt | there might be something there, but be ready to dig! |
04:55.22 | robbyt | krush: grab some samples off of free sound and convert them in audacity |
04:56.58 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.53.122) |
04:56.59 | __Krush__ | thanks robby...which specific format as seems quite nitpicky? |
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04:57.29 | robbyt | krush: i'm not sure what the GS uses, i'm sure the format is listed on voip-info though |
04:57.52 | __Krush__ | Ok 10x will have a browse |
04:57.56 | robbyt | haha here you go: http://freesound.iua.upf.edu/samplesViewSingle.php?id=14252 |
04:59.28 | __Krush__ | :) |
05:00.30 | __Krush__ | Is there some utility to convert mobile phone ringtones to GS format? |
05:00.42 | robbyt | midi files? |
05:00.58 | __Krush__ | MIDI or polyphonic.. |
05:01.11 | robbyt | audacity is a free audio editor |
05:01.17 | robbyt | but you can't import midi files into it |
05:01.29 | robbyt | midi is basicly sheet music |
05:01.52 | robbyt | i think winamp might let you dump midi files to disk as wavs |
05:02.08 | robbyt | then, for example on the cisco phones, i beleive they need to be compressed in GSM |
05:02.15 | robbyt | as 22khz 8bit files |
05:02.21 | robbyt | so that's where audacity would come in |
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05:02.47 | __Krush__ | good...will consider this route than...have tons of mobile phone ringtones.... |
05:03.19 | robbyt | haha here's another great one: |
05:03.20 | robbyt | http://freesound.iua.upf.edu/samplesViewSingle.php?id=11729 |
05:03.30 | robbyt | wow, know how i'm going to spend my morning at work |
05:03.35 | {zombie} | you can use timidity to convert midi to wav |
05:03.46 | robbyt | making up new amazing annoying rings for my cisco deskphone |
05:03.50 | {zombie} | and gs provide a hacked version of sox to convert to their ringtone format |
05:04.51 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
05:06.51 | __Krush__ | thanks zombie will have a look |
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05:09.04 | *** mode/#asterisk [+o denon] by ChanServ |
05:10.15 | __Krush__ | What are you opinions about the different SIP phones money can buy? |
05:10.55 | robbyt | i like the polycom 501s |
05:11.30 | CANO-1982 | __Krush__, you could try timidity |
05:11.33 | robbyt | i've worked with the newer sipura, cisco 7940s, and polycoms |
05:11.39 | CANO-1982 | is the best choice |
05:11.47 | __Krush__ | Bought GXP2000s as a low cost intro to Asterisk |
05:12.06 | __Krush__ | robby new sipura is 941? |
05:12.21 | robbyt | umm, the one with the cisco up/down button |
05:12.25 | robbyt | hold on, i'll check |
05:12.28 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
05:12.46 | __Krush__ | cisco/linksys....? |
05:12.53 | austinnichols10 | wooHoo! double-nat finally working! |
05:13.08 | robbyt | 841 |
05:13.15 | robbyt | it's a toy |
05:13.18 | __Krush__ | price-wise same as polycom 301 |
05:13.45 | robbyt | no wait |
05:13.54 | robbyt | it is the 941 |
05:13.55 | robbyt | sorry |
05:14.01 | robbyt | it's a toy- |
05:14.08 | robbyt | the poly301 is nice for the money |
05:14.16 | robbyt | but the kicker is that there's no speaker phone Mic |
05:14.34 | robbyt | the 501 is the best i've used though |
05:14.40 | robbyt | amazing speaker phone |
05:14.59 | __Krush__ | ok... |
05:15.08 | __Krush__ | 501 not yet ce certified in Europe here |
05:15.14 | __Krush__ | 500 ok but not 501 |
05:15.22 | robbyt | ahh |
05:15.34 | robbyt | not sure what the difference is- |
05:15.49 | __Krush__ | seems more mem for bigger better firmware |
05:16.24 | robbyt | <shrug> the newest firmware has https support for the services buttons |
05:16.30 | robbyt | guess that takes up some room |
05:16.44 | __Krush__ | yeps |
05:17.27 | __Krush__ | Would you know of some AAH resource to upgrade to latest asterisk without breaking config? |
05:17.52 | *** part/#asterisk CANO-1982 (i=alejandr@201.255.53.122) |
05:18.17 | robbyt | AAH? |
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05:18.29 | __Krush__ | Asterisk@Home |
05:18.35 | wasim | * @ ~ |
05:19.21 | robbyt | ahh |
05:19.28 | robbyt | *@~ hehe |
05:19.47 | robbyt | not sure krush- never used AAH |
05:19.54 | austinnichols10 | krush: try tom vile at baldwintech.com |
05:20.15 | austinnichols10 | wait - wrong url |
05:20.30 | austinnichols10 | baldwintechsolutions.com |
05:20.39 | austinnichols10 | He helped me get started with my AAH |
05:21.01 | austinnichols10 | tell him michael dyer reffered you |
05:21.39 | austinnichols10 | there are a few more names in the AAH forums - do a search on baldwintechsolution.com and you'll find the others |
05:22.02 | __Krush__ | Ok...will give it a browse |
05:22.11 | austinnichols10 | which version you running? |
05:22.38 | __Krush__ | 2.2 but read that 2.4 has addressed major mem leaks |
05:22.54 | austinnichols10 | I just upgraded my 2.2 to 2.4 |
05:23.04 | __Krush__ | how...? |
05:23.10 | austinnichols10 | let me get the link |
05:23.15 | __Krush__ | any reference docs? |
05:23.19 | austinnichols10 | yes |
05:23.45 | austinnichols10 | do you know how to do the yum updates? |
05:23.50 | __Krush__ | yes |
05:24.50 | __Krush__ | anything really tangible from the change 2.2 > 2.4? |
05:25.13 | austinnichols10 | nothing that I've needed |
05:25.57 | austinnichols10 | you need to be careful because AAH uses AMP. AMP has a lot of deprecated commands and if those commands go away between 2.2 and 2.4 your SOL |
05:28.11 | __Krush__ | What is the better alternative to AMP...besides hand changing the configs? |
05:28.20 | wasim | vi |
05:28.36 | Qwell | hire a monkey to do it |
05:28.57 | rob0 | tt-monkeys.gsm |
05:29.24 | __Krush__ | I see there are some monkeys dangling on the tree... |
05:29.47 | austinnichols10 | Here's the link to what I used: http://sourceforge.net/forum/message.php?msg_id=3542026 |
05:29.49 | rob0 | I resemble that remark! |
05:30.02 | austinnichols10 | you should readh the whole thread so you can see what's really happening |
05:30.31 | austinnichols10 | afterwards I did a yum update which included some kernel stuff so I had to rebuild zaptel afterwards |
05:31.03 | FuriousGeorge | where are those modules i gotta delete before upgrading. i thought they were in /var/lib/modules/asterisk |
05:31.06 | austinnichols10 | I read that doing this manual update may break the fax over IP receiving stuff |
05:31.56 | austinnichols10 | in /usr/src |
05:32.24 | FuriousGeorge | ? |
05:32.28 | FuriousGeorge | no its not |
05:32.29 | __Krush__ | OK...thanks |
05:33.31 | austinnichols10 | sorry george - was on mine |
05:34.20 | FuriousGeorge | you were close though, theyre in usr lib |
05:36.13 | __Krush__ | Bye alll... |
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05:55.10 | mattwj2006 | I just made my first call from cli |
05:55.20 | tzafrir_laptop | using what? |
05:55.25 | mattwj2006 | pretty sweet....I have always been just doing sip or iax |
05:55.35 | tzafrir_laptop | dial? |
05:55.37 | mattwj2006 | alsa |
05:55.40 | mattwj2006 | yup |
05:56.22 | mattwj2006 | I am still quite the noob with asterisk |
05:56.27 | inv_Arp | mattwj2006: how? |
05:56.28 | mattwj2006 | :) |
05:57.18 | mattwj2006 | well you have to load => chan_alsa.so |
05:57.24 | mattwj2006 | reboot it |
05:57.32 | mattwj2006 | dial extension@context |
05:58.21 | mattwj2006 | does that help inv_arp? |
05:58.21 | inv_Arp | ahh |
05:58.26 | inv_Arp | thx |
05:58.39 | mattwj2006 | yup |
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05:59.55 | libila | I'm setting up my dgium card for the first time. I have four FX0 ports on it. When I do ztcfg -vv I get 'Channel map: Channel 01: FXO Kewlstart (default) (Slaves: 01) etc etc etc 4 channels configure. Notice: Configuartion file is /etc/zaptel.conf line 26: Unable to open master device `/dev/zap/ctl`' line 26 is 'defaultzone=us' I did ls /dev/zap and there isn't a directory by that name. Could someone explain why I'm getting that error? |
06:00.29 | *** join/#asterisk coppice (n=chatzill@30.196.17.210.dyn.pacific.net.hk) |
06:01.08 | libila | or why it's looking for /dev/zap/ctl when it doesn't exist? lspci detects my card: Communication controller: Tiger Jet Network Inc. Tiger3xx Modem/ISDN interface |
06:01.12 | Qwell | libila: README.udev |
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06:03.55 | mattwj2006 | anyone know how to give my sound card an extension? |
06:03.59 | mattwj2006 | I am using ALSA |
06:04.06 | Qwell | Dial the alsa channel |
06:08.25 | *** join/#asterisk dissolutions_ (n=rgff@h24-207-70-68.dlt.dccnet.com) |
06:15.14 | mattwj2006 | Qwell you do know what the dial syntax is? |
06:15.47 | mattwj2006 | a brief look at voip-info.org didn't turn up anything |
06:15.48 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
06:15.55 | Qwell | Dial(Console/dsp) ? |
06:16.07 | *** join/#asterisk yoyoma (n=S@mbl-99-58-31.dsl.net.pk) |
06:16.50 | yoyoma | hello |
06:18.21 | yoyoma | when i make an external call, there is no audible ring for the called party at the other end. can anyone help here? |
06:19.39 | *** part/#asterisk yoyoma (n=S@mbl-99-58-31.dsl.net.pk) |
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06:19.58 | tzafrir_laptop | mattwj2006, how application dial |
06:20.03 | yoyoma | hello |
06:20.38 | tzafrir_laptop | yoyoma, what type of call? from what phone to what phone? |
06:20.59 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
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06:22.23 | yoyoma | tzafrir: I have a TDM04B with asterisk 1.2. Internal calls, ie. extension to extension calls, work fine. When calls come in from the outside, all internal phones ring. When I place an external call using the zap channel to a regual pstn number, the phone at the other end does not ring |
06:22.59 | yoyoma | if they happen to pick up the phone at the moment i'm trying to call them, the call is connected... |
06:23.07 | yoyoma | at my end i hear the ring back tone as well... |
06:23.17 | yoyoma | it's just that the phone does not ring for them |
06:23.18 | Qwell | yoyoma: Did you plug in the power connector? |
06:23.29 | *** join/#asterisk OloBola (n=not@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
06:23.33 | Qwell | it won't be able to generate ring voltage otherwise |
06:24.07 | austinnichols10 | or could be fxs/fxo swapped |
06:24.30 | yoyoma | Qwell: excellent point.... recently the digium board was reinstalled... it may have been left unplugged. let me check |
06:25.06 | mattwj2006 | it worked.....but I got a notice |
06:25.24 | mattwj2006 | NOTICE[5509]: rtp.c:510 ast_rtp_read: Unknown RTP codec 72 received |
06:25.56 | yoyoma | Qwell: the power connector is plugged and the lights at the ports are on |
06:25.59 | austinnichols10 | anyone know if PRI circuits need to be rx/tx gain tuned (for echo)? |
06:27.12 | OloBola | I would like a caller to be able to "press 2" to speak with whoever, which is then forwarded to their cell phone. How can I do this from my php script? |
06:28.01 | opsys | austinnichols10: PRIs shoudl NOT need to be turned as they are a balanced cirquit |
06:28.14 | austinnichols10 | that's what I thought |
06:28.18 | yoyoma | i had an older version of asterisk using the same hardware and this problem was not present then |
06:28.30 | opsys | austinnichols10: PRIs should NOT need to be tuned as they are a balanced circuit (sorry late) |
06:29.27 | austinnichols10 | opsys: I'm having echo issues with cisco 7960s going to PSTN endpoints via the PRI. Any idea where else to look since the PRI should be clean? |
06:30.03 | opsys | Are you in FLorida, using a Bell PRI?? |
06:30.17 | austinnichols10 | florida using an FDN PRI |
06:31.31 | robbyt | pri isn't always echo free! :) |
06:31.55 | opsys | FDNs are trick, they don;t put good echo cans on them. YOu can try calling from a landline a DID that runs echo. DO NOT use a Cell. If you have echo there its on your Telco side, Yuo can try using a dirrect echo can in Asteisk. What kind of hardware do you have? |
06:32.08 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
06:32.41 | austinnichols10 | TE110P (was thinking to switch to an EC card) |
06:33.04 | opsys | what are you getting as far as interupts in /proc, have you run zttest? |
06:33.08 | austinnichols10 | FDN = Florida Digital Network |
06:34.27 | austinnichols10 | zttest runs at either 99.987793 or 100 |
06:34.33 | austinnichols10 | jumps back and forth |
06:34.48 | austinnichols10 | avg 99.991708 |
06:35.31 | opsys | anything above 99.975 is good. Congrats your PRI to PCI interface is solid, now on to the the next problem. |
06:36.10 | austinnichols10 | what am I looking for on interrupts? |
06:37.49 | litage | are there any other options/settings you can specify in cdr_manager.conf besides "enabled=" ? |
06:38.16 | opsys | austinnichols10: did you get my reply? |
06:41.05 | austinnichols10 | not to the x600 question |
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06:42.31 | austinnichols10 | opsys: you still there? |
06:42.56 | opsys | austinnichols10: I'm still here |
06:43.16 | austinnichols10 | k - interrupts seem to be > 1000/sec (dual proc box) |
06:43.43 | opsys | Your OK on interupts too. |
06:44.09 | austinnichols10 | is there a better way to test other than just cat | interrups? |
06:44.59 | opsys | you can do.. while: do; cat /proc/interrupts >> /tmp/out; sleep1 ; done |
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06:45.42 | opsys | after about 10sec ctrl-c and the grep for the module (wcte1xxp) and add up the differences. |
06:46.15 | yoyoma | what reasons could there be for the no audible ring at an external number? i have a pstn line coming in to one port of my tdm04b. when i place an outside call using that pstn line, the phone at the other end does not ring but i get a ring back tone. |
06:46.35 | austinnichols10 | I'm running off of an aah build so there's no 600, but I do have dids. What does 600 normally do? |
06:47.17 | opsys | add this into your menu context. exten => 600,1,Echo() |
06:48.09 | opsys | echo does just that it echos everything it gets back to you. If you are on a landline there is a chance you will get an echo if it on the Teclo side. |
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06:50.09 | austinnichols10 | got it |
06:51.29 | opsys | You could also try this on the VoIP side and see if you get an echo also. |
06:51.37 | opsys | How BAD is the echo? |
06:53.39 | austinnichols10 | testing now |
06:53.57 | opsys | yoyoma: once you place the call on the PSTN the ringing comes from a ring gen at YOUR CO. Unless of course you have Asterisk set to generate rings. |
06:54.17 | austinnichols10 | opsys: check your PM. I don't think it's bad at all |
06:54.48 | opsys | What PM |
06:55.06 | austinnichols10 | I opened up a separate window - hmmm |
06:55.26 | trixter | I would open a window but its cold outside |
06:55.53 | opsys | I know it about 67F hear on the coast of Florida, BRRRRRRR!! |
06:56.24 | opsys | trixter: Are you in SFO? |
06:56.32 | austinnichols10 | going down to 56 tonight |
06:56.37 | trixter | no about 150 miles or so away |
06:57.21 | opsys | trixter: Thanks for some of the info on your site, really helpfull, got me out of a crunch the other day |
06:57.30 | austinnichols10 | trixter: have you tried hooking up a 'real' phone and making a test call? |
06:57.43 | trixter | um which site and what info? |
06:57.44 | austinnichols10 | sorry - not trixter, yoyo |
06:57.56 | trixter | austinnichols10: define 'real' |
06:57.58 | yoyoma | yes.. |
06:58.11 | austinnichols10 | yoyoma: princess phone |
06:58.30 | opsys | the only way its going down to 56 here is if my A/C mal-functions. I am on the Beach. GulfStream BABY |
06:58.47 | yoyoma | austinnichols10: i have a second line here that is not connected to my asterisk server... |
06:58.59 | Qwell | meh, you people and your "weather" |
06:59.11 | Qwell | "snow" and "rain" and the like |
06:59.21 | opsys | yoyo: are both lines off of the same CO? |
06:59.39 | opsys | Qwell: (err North) |
06:59.52 | Qwell | opsys: hrm? |
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07:01.27 | opsys | Qwell: I figured cause your name is North that your up there Lat wise. |
07:01.41 | Qwell | not so much, no.. |
07:01.51 | Qwell | pretty far south actually |
07:01.59 | opsys | Sorry, How far?? |
07:02.03 | Qwell | southern CA |
07:02.48 | opsys | Ahh SoCal the area where the weather is great but you may find your house at the bottom of the Vally!! |
07:03.23 | austinnichols10 | opsys: what do you do for work? |
07:03.49 | opsys | austinnichols10: Asterisk Consulting and integration help. |
07:04.04 | austinnichols10 | cool - send me info |
07:04.13 | austinnichols10 | ALWAYS helps to have someone local |
07:04.30 | opsys | go to www.opsys.com. Site outdated but contacts are still good. And yourself. |
07:04.36 | yoyoma | opsys: yes both lines are from the same co |
07:04.49 | austinnichols10 | www.tieronehosting.net |
07:05.03 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
07:05.25 | [TK]D-Fender | Wow... another quick release.... |
07:05.27 | austinnichols10 | ha - I tried to call you guys when I was first thinking about setting up a server |
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07:05.52 | opsys | austinnichols10: How long ago?? |
07:06.04 | Supaplex | austinnichols10: but the line was busy eh? |
07:06.06 | Supaplex | ;) |
07:06.38 | austinnichols10 | a couple of months ago |
07:06.43 | austinnichols10 | right before Christmas |
07:06.49 | Math[laptop] | thats 1 month ago |
07:06.58 | austinnichols10 | it's 2:05 |
07:06.59 | opsys | yoyo: If asterisk is ringing and you pick up the other phone are the calls bridged. |
07:07.56 | opsys | opsys: sorry we missed ya. |
07:07.57 | yoyoma | opsys: yes the calls are bridged. i get ring back tone when i call the other number. the other phone does not ring, but if i pick it up i'm connected |
07:08.01 | austinnichols10 | np |
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07:08.48 | opsys | yoyo: try swapping lines and see if problem continues, alt. can you call second line from other phone (ie cell)? |
07:09.17 | opsys | Austinnichols10: Call me at 305-503-3000 ext 122. Lets see if we can kill the echo prob. |
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07:11.00 | yoyoma | opsys: i'll give that a shot. thanks a lot for your help! |
07:11.13 | opsys | yoyoma: no probblem |
07:11.48 | austinnichols10 | opsys: when is a good time to call? |
07:12.06 | opsys | I'm up now. |
07:12.21 | austinnichols10 | cool |
07:13.10 | clive- | found it |
07:14.13 | *** part/#asterisk dataworm (n=dataworm@modemcable192.46-130-66.mc.videotron.ca) |
07:15.04 | lucasjb | Hiyas, just a quick one, how do I tell WaitExten() I want more than one digit? I've set my Digit timeout to four seconds, but the system always continues when my user presses their first key. |
07:17.10 | *** join/#asterisk Psykick (n=anon@203-167-215-33.dsl.clear.net.nz) |
07:17.12 | Psykick | hi guys |
07:17.23 | Psykick | I got a weird yet kind of interesting question for ya'll |
07:17.44 | Psykick | just wondering if its possible to change the firmware in an avaya phone to talk either SIP or IAX2 |
07:18.03 | Psykick | I know avaya makes SIP phones but just wondering if it's possible using other avaya models |
07:18.30 | Psykick | I've got about 20+ avaya 5420 phones from our previous phone system |
07:18.45 | [TK]D-Fender | Google it... |
07:19.04 | Psykick | been googling it |
07:19.18 | Psykick | thought I'd come to the one place I might get an answer |
07:24.16 | coppice | coming here sounds really desperate :-) |
07:24.49 | Psykick | kinda am :) |
07:25.21 | dpryo | Psykick: avaya.com probably knows ;) |
07:26.46 | Psykick | not like they'd tell anyone |
07:26.52 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
07:27.12 | Psykick | they'd have people changing firmware left right and center then be expected to fix the problems of those people |
07:27.18 | dpryo | I've found sip-images for 4620 on their site. |
07:27.26 | Psykick | so I don't think they'd mention it |
07:27.35 | Psykick | yeah I found those as well |
07:27.42 | dpryo | All my 40 phones are running sip now :) |
07:27.56 | Psykick | do you have original 4620's? |
07:28.03 | Psykick | or another model? |
07:28.17 | dpryo | I have 4620 phones. |
07:28.28 | Psykick | well there ya go ... I have 5420 phones |
07:28.36 | Psykick | and around 20+ of them |
07:28.39 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
07:28.47 | dpryo | I'm sure you find what you look for if you put your mind to it :) |
07:29.40 | Psykick | well considering I have 5420 phones that are not advertised as SIP phones I don't think I'd be able to do what I want with them which is change the firmware so that it supports SIP |
07:30.13 | dpryo | My 4620 were not sip-phones, until i downloaded another firmware from support.avaya.com. |
07:31.18 | lucasjb | Can anyone explain to me why WaitExten() is returning after only one digit is entered? |
07:31.28 | Psykick | I suppose I could at least screw up 1 phone ... an expensive test though |
07:31.36 | [TK]D-Fender | lucasjb : Pastebin you extensions.conn |
07:31.37 | [TK]D-Fender | ~pb |
07:31.39 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
07:31.41 | Psykick | damned phones cost $550 |
07:31.46 | *** join/#asterisk pobre (n=seymore@203.215.73.192) |
07:31.59 | Psykick | that's each |
07:32.40 | dpryo | Heh |
07:33.18 | dpryo | So why buy them in the first place? |
07:33.27 | dpryo | Sounds like a really bad deal :) |
07:33.31 | Psykick | we originally had an entire avaya phone system |
07:33.56 | Psykick | but ... for basic call reports the suppliers wanted us to pay $30K |
07:34.05 | Psykick | on top of what we had already paid |
07:34.13 | lucasjb | [TK]D-Fender, http://pastebin.com/531775 |
07:34.15 | dpryo | Heard that one before :) |
07:34.25 | Psykick | even though we were supposed to get CCC |
07:35.48 | Psykick | considering the 5400 isn't exactly advertised as an IP phone I wonder if it would be possible to change the firmware to another firmware that supports SIP |
07:36.17 | [TK]D-Fender | lucasjb : Don't just show me a little clip of what YOU think is relevent. Pastebin it ALL |
07:39.04 | lucasjb | [TK]D-Fender, hmm... I'll try... |
07:39.21 | *** join/#asterisk EriSan (n=erisan@151.8.109.88) |
07:40.35 | coppice | most PBX phones only work as RIP phones, especially when they are OFF :-) |
07:41.48 | bigjb | psykick, you can get call reports for next to nothing |
07:41.55 | bigjb | on a an avaya system |
07:41.57 | Psykick | not on avaya you can't |
07:42.06 | bigjb | yes you can |
07:42.07 | Psykick | cbc ... yeah fine |
07:42.08 | lucasjb | [TK]D-Fender, http://pastebin.com/531780 |
07:42.20 | Psykick | what we were supposed to get was CCC |
07:42.26 | Psykick | cbc is just a waste of time |
07:42.31 | lucasjb | [TK]D-Fender, the interesting context is [atp-incoming] |
07:42.46 | bigjb | nope, when i get to work i will find the name of the company that makes the software |
07:43.09 | bigjb | in fact i might even be able to find you the site now |
07:43.12 | Psykick | bigjb: doesn't matter now .... using asterisk and we have all the reports we need |
07:43.30 | Psykick | only thing is just trying to re-use the equipment |
07:43.38 | bigjb | http://www.ctidata.co.uk/office_product.htm |
07:44.15 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
07:45.19 | Psykick | bigjb: errr ... don't see any kind of reference to integrates with avaya IP office ..... |
07:45.30 | bigjb | it does |
07:45.33 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
07:45.38 | [TK]D-Fender | lucasjb : You are trying to use "n" as the first priority on several extens in there which is something you can not do. You therefor have NO valid extens in taht menu and everything leads to "i"nvalid |
07:45.41 | bigjb | ive installed it to use several systems |
07:46.46 | Psykick | bigjb: doesn't matter anymore |
07:47.17 | bigjb | heh definitely not if your using asterisk |
07:47.32 | lucasjb | [TK]D-Fender, Ah, ok let me fix that... |
07:47.59 | Psykick | be good if I could change the firmware in these 5400 phones to use Avaya's SIP firmware |
07:48.21 | Psykick | assuming that the phones still work after changing the firmware |
07:48.22 | dpryo | What is the difference on 4600 and 5400? |
07:49.04 | lucasjb | [TK]D-Fender, Ah, that's fixed it - thank you! |
07:49.06 | Psykick | other than the 4600 series being IP phones .... can't really differentiate between the two |
07:49.17 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
07:49.37 | [av]bani | bleh |
07:49.54 | bigjb | that is the difference |
07:49.57 | knight_ | I am having problems getting ast to ast via iax nat to auth |
07:50.03 | Psykick | wouldn't it just really rip your shorts if there is no difference between the different series other than firmware |
07:50.09 | dpryo | Psykick: But they do have an ethernet interface? |
07:50.19 | Psykick | dpyro: yip |
07:50.22 | [TK]D-Fender | lucasjb : ywc |
07:50.23 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
07:50.33 | Psykick | power is supplied over ethernet as well as firmware updates |
07:50.43 | dpryo | Just as with 4600. |
07:51.00 | dpryo | I'd try to put in a 4600-sip-image :) |
07:51.03 | bigjb | thats the same with all ipoffice digital phones |
07:51.13 | Psykick | it's tempting ... but $550 is a fair bit o cash to waste if it screws the phone |
07:51.27 | dpryo | It probably won't. You can always put back the old image. |
07:51.52 | dpryo | The netbootloader is probably never changed. |
07:52.18 | bigjb | give me 3 or 4 houts and once im at work i can get a request in with my supplier |
07:52.21 | dpryo | And I bet they have some kind of a system to check if the image is compatible |
07:52.39 | Psykick | hmm ... |
07:52.48 | Psykick | can you email me? |
07:52.54 | Psykick | or leave me a memo |
07:52.54 | bigjb | yup |
07:52.55 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
07:53.09 | bigjb | not a problem |
07:53.12 | Psykick | kewl |
07:53.19 | bigjb | i would be interested to see myself |
07:53.35 | Psykick | THANKS! bigjb |
07:54.22 | bigjb | we have a ipoffice 406 v3 with 5402 and 5420 handsets |
07:54.24 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13) |
07:54.40 | bigjb | i also have a asterisk box running on the same network |
07:54.45 | *** join/#asterisk tomas_ (n=tomas@78.121.broadband3.iol.cz) |
07:55.04 | bigjb | been meaning to speak to supplier about best way to get them to talk to each other anyway |
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08:01.10 | Psykick | bigjb: I found a doc somewhere on the net for getting an asterisk box to act as a gateway for avaya ip office |
08:02.47 | *** join/#asterisk S3RG3US|W (n=SERGEUS@ippe-245.ippe.ru) |
08:04.23 | knight_ | k |
08:05.25 | dpryo | Psykick: care to share? |
08:06.19 | *** join/#asterisk gevious (n=chatzill@dsl-146-112-82.telkomadsl.co.za) |
08:06.41 | gevious | Hi All |
08:07.49 | *** join/#asterisk acehunky (n=chat_jok@221-128-138-148.exatt.net) |
08:08.10 | Psykick | dpyro: http://www.inventigo.co.uk/home/forum/index.php/topic,20.new.html |
08:08.22 | Psykick | they just talk about their setup ... |
08:08.45 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
08:08.57 | Psykick | I believe it works quite well though ... have heard of someone else doing the same thing |
08:09.16 | Psykick | ok well ... goin home now |
08:09.21 | Psykick | talk to ya'll tomorrow |
08:11.14 | *** join/#asterisk webmind (n=webmind@feather.perl6.nl) |
08:12.16 | knight_ | an asterisk over iax over a nat does not seem to authenticate to a remote ast server.... ideas? |
08:13.43 | OloBola | I need to use call files to forward calls from an AGI script? |
08:14.26 | *** join/#asterisk diLLec (n=dillec@a15182648.alturo-server.de) |
08:14.32 | OloBola | so person calls, recording: "press 1 to call cell phone", user presses 2, AGI script drops a call file? |
08:15.48 | OloBola | press 2, |
08:17.07 | lucasjb | [TK]D-Fender, thanks again for your help. |
08:17.08 | *** join/#asterisk Bambr (n=Bambr@213-35-235-26-dsl.end.estpak.ee) |
08:18.27 | *** join/#asterisk coppice_ (n=chatzill@103.194.17.210.dyn.pacific.net.hk) |
08:19.36 | mmmToop | hi...anyone got an idea what this WARNING is: "Received mini frame before first full voice frame" |
08:20.16 | clive- | mmmtttooop, ignore that,,,just bad connectivity normally |
08:22.05 | mmmToop | what do you mean by bad connectivity...? LAN issues? |
08:23.16 | clive- | I see that on international linmks, but I don't realy have expereince over a LAN |
08:23.28 | clive- | you can usually ignore those warning messages |
08:24.47 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
08:25.38 | *** join/#asterisk OnuR (n=HaLuK_Le@62.220.216.163) |
08:25.46 | mmmToop | maybe the soft phones that we are using are doing it...? |
08:27.52 | OloBola | can someone suggest a way to forward a call from an AGI script? |
08:28.22 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
08:29.01 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
08:29.08 | *** join/#asterisk Password (n=oFF@62.220.216.163) |
08:30.25 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
08:32.27 | pobre | im using TDM400P im know on asterisk and dont know how to make a call via this zap card? |
08:32.39 | pobre | im using TDM400P im new on asterisk and dont know how to make a call via this zap card? |
08:33.01 | pobre | it is configure no alarms |
08:35.30 | *** join/#asterisk robbie2 (n=rob@CPE-144-137-188-224.qld.bigpond.net.au) |
08:35.35 | robbie2 | helllo |
08:35.45 | *** join/#asterisk O-Zone (n=O-Zone@moloch.asb.unisi.it) |
08:35.47 | O-Zone | hi all |
08:36.39 | OloBola | can someone suggest a way to forward a call from an AGI script? |
08:37.40 | O-Zone | there's a way to do ringing on two extn ? |
08:38.13 | robbie2 | anyone here configured an isdn 10 ? |
08:38.52 | *** join/#asterisk leopardus (n=leopardu@217.22.179.69) |
08:40.01 | OloBola | I think you need to use & symbol or something to ring two extensions |
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08:44.02 | *** join/#asterisk KeX_WorX (n=chris@ng1.kurtkrenn.com) |
08:44.05 | KeX_WorX | hi |
08:44.28 | KeX_WorX | i'v a problem getting a variable from an agi script back into the dialplan |
08:44.35 | KeX_WorX | here are my scripts: http://phpfi.com/99391 |
08:44.54 | KeX_WorX | can someone pls look at them and give me a hint where the problem is/could be ? |
08:45.49 | KeX_WorX | when i call the sh script from the cmd line, i get the expected result, but if i call it from within the dialplan i just get nothin : / |
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08:50.56 | *** part/#asterisk O-Zone (n=O-Zone@moloch.asb.unisi.it) |
08:51.56 | robbie2 | are rtp ports udp or tcp ? |
08:52.00 | robbie2 | for forwarding ? |
08:54.14 | Krill | udp i'd imagine |
08:54.53 | robbie2 | loos so |
08:55.47 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
08:57.12 | KeX_WorX | can someone pls look at that and probably find an error? http://phpfi.com/99391 |
09:04.19 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
09:07.13 | *** join/#asterisk enemy^x (n=null@85.196.70.98) |
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09:12.30 | jhiver | hi all |
09:12.33 | dogtanian | <PROTECTED> |
09:13.07 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13) |
09:13.22 | enemy^x | I`ve been trying to find out why my MOH is causing so much misbehaviours with asterisk. When running mpg123-0.59r, it seems more stable than any other mpg123 release. But still, it seems to crash once in a while... Anyone have any good suggestions to keep this stable? I`m running 1.2.3 |
09:13.30 | jhiver | is there a way to have the current time (in seconds or so) in a variable so that I can get asterisk to record each of the calls I place through it? |
09:18.53 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
09:23.19 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
09:23.31 | jhiver | seriously I can't I do something like Servar(TIME,Now()) ? |
09:24.06 | *** part/#asterisk Hali_303 (n=surfk@dsl51B6ACC5.pool.t-online.hu) |
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09:30.39 | *** join/#asterisk justnulling (i=justnull@ool-18bab443.dyn.optonline.net) |
09:30.48 | justnulling | what is this error Auto-congesting call due to slow response? |
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09:34.49 | reza | filberts, eh? |
09:35.08 | *** join/#asterisk gvag11 (n=gvag11@ipa201.5.tellas.gr) |
09:35.11 | gvag11 | Hi all |
09:35.52 | gvag11 | Any suggestions of how can i monitor an Asterisk box ? Monitor the process and the ZAP channels.... |
09:35.55 | *** join/#asterisk P0L0 (n=n0n3@140.Red-83-58-255.dynamicIP.rima-tde.net) |
09:38.18 | diLLec | connect via manager account. there are zap status commands available |
09:39.02 | gvag11 | dillec : yes i know but i would like to find an application which can do this 24h and report any alarms, any idea ? |
09:40.04 | diLLec | i think that alarms are sent by events to the manager |
09:41.28 | gvag11 | dillec: yes they are comming like events, but instead of make something from scratch (catch the event and then report) i am looking to find if there is something ready to do that. |
09:43.06 | diLLec | ah ok - i thought you need a way to implement something :-) |
09:43.49 | gvag11 | no ... ;-) |
09:43.58 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
09:45.16 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
09:46.02 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
09:50.34 | areski | good morning |
09:51.42 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
09:52.14 | *** join/#asterisk phpboy (n=shane@196.26.21.106) |
09:52.57 | phpboy | Hey all, I'm trying to make a call with my Swissvoice hardphone... I can call local extensions but I can't call out to my PSTN... these are the errors I get on asterisk |
09:52.58 | phpboy | Jan 31 11:51:36 WARNING[7101]: chan_sip.c:703 retrans_pkt: Maximum retries exceeded on call 106b3030-aa10a8c0-13c4-17d-5b5b6-72cc@192.168.16.170 for seqno 1 (Critical Response) |
09:55.15 | phpboy | :T |
10:00.06 | phpboy | I really love you guys |
10:00.10 | phpboy | but I need your help :< |
10:01.36 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
10:01.36 | *** mode/#asterisk [+o drumkilla] by ChanServ |
10:01.41 | tronix | I imagine * doesn't support 7960G's extension mobility |
10:01.51 | tronix | it's probably a CCM thing... and relies on SCCP specific features? |
10:02.02 | areski | phpboy, peoples have never been so quite |
10:02.41 | phpboy | I know |
10:02.46 | phpboy | that's why I'm so heart sore :< |
10:03.19 | tronix | slackers in north america sleeping. how dare they! :-) |
10:03.23 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13) |
10:03.30 | *** join/#asterisk secure75 (n=mic@host-62-245-230-34.customer.m-online.net) |
10:03.47 | tronix | for some reason, I keep writing down 'twinkie' instead of 'twinkle' in my extensions.conf. freudian slip? :P |
10:03.51 | phpboy | If I call to my pstn(via ISDN) I cometimes get an echo... how do I avoid this? |
10:04.24 | tronix | hmm the usual means is to adjust rx/tx gain, I understand. beyond that, not too familiar with echo issues. |
10:06.20 | tronix | (unrelated), looks like I found something on extension mobility at voip-info.org's wiki. cool. |
10:07.16 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
10:07.37 | *** join/#asterisk viperdude (n=jon@borat.enta.net) |
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10:20.56 | tronix | hmm. where does one find NVFaxDetect? |
10:21.37 | tronix | n/m |
10:21.39 | chapeaurouge | hmm.. why would i have no sound, if i have * setup in my internal network, with no firewall whatsoever? |
10:21.43 | tronix | I see the info on how to find it via wiki. |
10:23.49 | *** join/#asterisk grey (n=grey@193.220.84.198) |
10:23.55 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
10:23.55 | grey | hi all |
10:24.11 | grey | can anyone give me some help with amp ? |
10:25.20 | *** join/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com) |
10:25.23 | *** part/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com) |
10:25.31 | tronix | ~amp |
10:25.37 | jbot | hmm... amp is NOT supported here! people using it should join #amportal |
10:25.45 | tronix | :) |
10:25.53 | tronix | best place for it. |
10:26.35 | tronix | chapeauro: not sure. call establish ok? no messages in logs that looks unusual? debug messages enabled? |
10:27.07 | tronix | sic tcpdump/ethereal/etc and watch traffic to see if RTP established ok |
10:27.37 | tronix | grey: no offense intended. I'd help if I knew anything about amp. sorry. |
10:27.42 | chapeaurouge | tronix, calls establish ok. sip debug gives no real good info |
10:28.01 | tronix | hmm. |
10:28.10 | chapeaurouge | messages are being played, but i hear nothing. |
10:28.44 | tronix | sip softphone or hardphone used? |
10:29.02 | chapeaurouge | spftphone |
10:29.08 | *** join/#asterisk BSDaemon (i=hbf@CPE00032f0d286f-CM014380004179.cpe.net.cable.rogers.com) |
10:29.10 | tronix | hmm. |
10:29.24 | chapeaurouge | i had specified an RTC port.. |
10:29.27 | *** join/#asterisk fulgas (n=fulgas@209.8.233.254) |
10:29.27 | chapeaurouge | im gonna try without it |
10:29.47 | chapeaurouge | bleh. i can't :) |
10:31.50 | chapeaurouge | it works at home (differnt * install, but very similar)... odd. |
10:32.52 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
10:33.00 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13) |
10:33.38 | chapeaurouge | btw, for a 1.2.3 to 1.2.4 upgrade, nothing in particular to worry about? |
10:33.51 | fourcheeze | if I have N asterisks all looking at the same database (realtime) what else is required for me to get dynamic routing between them? |
10:35.26 | fourcheeze | is there some way to make * save the name of the host that each client is registered with to the database? |
10:36.05 | Dabba | if $fred contains 12345 how do i make it contain null ? |
10:36.15 | fourcheeze | it seems a waste of a RDBMS otherwise |
10:37.32 | X-Rob_ | Dabba, - 'Set(fred=)' |
10:38.09 | X-Rob_ | ANyone here got a GXP-2000? |
10:39.12 | susin | some with astgui experience? |
10:39.27 | fourcheeze | any realtime experts around? |
10:39.32 | fourcheeze | any realtime hackers around? |
10:39.38 | fourcheeze | I'd like to know where it's going |
10:39.54 | fourcheeze | there doesn't seem to be much more work to have automatic clustering |
10:40.10 | fourcheeze | or is that available already with some module I don't know about? |
10:40.54 | oej | fourcheeze: It's not automatic, but if you have the proper fields in the database, Asterisk will save information for you |
10:41.14 | fourcheeze | oej: which fields do I need? |
10:41.35 | oej | chapeaurouge: No, there should not be anything to worry about more than upgrading your binaries |
10:41.47 | oej | fourcheeze: Can't remember of the top of my head. Wait. |
10:42.09 | fourcheeze | oej: because then I'm 90% of the way towards my goal |
10:42.31 | fourcheeze | oej: BTW my snoms still are not *subscribing* for less than 3600 seconds |
10:42.38 | oej | Read doc/README.extconfig to find the fields |
10:42.44 | fourcheeze | oej: ok thanks |
10:43.13 | oej | In Addition, a field named "fullcontact" is used |
10:43.27 | oej | For saving the Contact: header of the registered peer |
10:43.41 | oej | Most people use dundi in combination with regcontext/regexten |
10:43.53 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13) |
10:46.11 | fourcheeze | oej: but fullcontact doesn't tell me where a client registered |
10:46.13 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
10:46.23 | fourcheeze | or don't I need to know that? |
10:46.43 | *** join/#asterisk Vinsik (i=vinsik@84-240-73-181.dsl.maxinetti.fi) |
10:46.50 | chapeaurouge | hmmm.... i can now hear, the 'bye' part of good-bye, but nothing else before... :\ |
10:47.25 | Vinsik | Need a bit of help. How to dial out with Realtime peer added to MySQL? |
10:48.53 | chapeaurouge | voicemail isn't playing mp3 file, is it? |
10:52.00 | chapeaurouge | no. hmm. |
10:52.26 | Dabba | X-Rob thanks |
10:52.51 | *** join/#asterisk FastJack (n=fastjack@reverse-82-141-60-53.dialin.kamp-dsl.de) |
10:53.32 | *** join/#asterisk ful|work (n=fulgas@209.8.233.254) |
10:55.25 | *** join/#asterisk sdgusler (n=animenod@65.111.201.79) |
10:55.48 | mutilator | :O |
10:55.49 | mutilator | no +r? |
10:57.13 | X-Rob_ | Oooh. |
10:57.19 | X-Rob_ | This new GXP-2000 firmware is interesting |
10:57.27 | X-Rob_ | OOh! Pretty! |
10:57.29 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
10:58.45 | X-Rob_ | Fuck me. I think they've fixed it. |
10:59.10 | *** join/#asterisk cfh (n=luca@82.193.23.6) |
10:59.10 | gaupe | X-Rob_: what has been fixed? |
10:59.13 | Vinsik | Need a bit of help. How to dial out with Realtime peer added to MySQL? |
10:59.36 | X-Rob_ | gaupe, well, the AGC that was causing handset feedback seems to be fixed. |
10:59.39 | X-Rob_ | BLF's work |
10:59.43 | X-Rob_ | haven't tested call pickup yet. |
10:59.43 | RoyK | Vinsik: it's all in the docs :P |
10:59.55 | Vinsik | Royk: cant find it.. |
11:00.04 | gaupe | RoyK: takk |
11:00.07 | RoyK | ~realtime |
11:00.09 | jbot | rumour has it, realtime is http://www.voip-info.org/wiki-Asterisk+RealTime |
11:00.41 | gaupe | X-Rob_: I will try to load in on the only one I have, but it's still not a good phone :) |
11:00.55 | X-Rob_ | For the money, they seem pretty reasonable. |
11:01.08 | cfh | what can i do to configure a sip trunk with a server asterisk and a cisco router? |
11:01.33 | RoyK | methinks grandstream is quite reasonable, only perhaps a little lightweight, for dorstoppers |
11:02.00 | gaupe | X-Rob_: I like the thomson 2030 for that money |
11:02.48 | Vinsik | RoyK: the problem is .. when i put Dial(SIP/${EXTEN}@user-out,20,r) <= this says no route to host. isnt it suppose to look for the user in database? |
11:03.33 | RoyK | hm... |
11:03.45 | RoyK | the syntax is Dial(SIP/user) |
11:03.53 | RoyK | or Dial(SIP/user/${EXTEN}) |
11:04.02 | Vinsik | bah |
11:04.07 | RoyK | :) |
11:04.18 | Vinsik | RoyK: im calling through a SIP provider.. |
11:05.24 | *** join/#asterisk P0L0 (n=n0n3@140.Red-83-58-255.dynamicIP.rima-tde.net) |
11:05.34 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
11:05.45 | backblue | hi. |
11:06.52 | RoyK | Vinsik: ithen use the latter |
11:06.58 | RoyK | < |
11:06.58 | RoyK | > |
11:07.03 | *** join/#asterisk drumkill1 (n=russell@host-12-179-65-65.nctv.com) |
11:07.15 | Vinsik | RoyK: nope |
11:07.23 | Vinsik | not workin |
11:07.25 | RoyK | then pastebin the output |
11:07.34 | Vinsik | w8 |
11:07.36 | phpboy | I need to get rid of the echo on my line |
11:07.37 | phpboy | pomple :< |
11:08.04 | X-Rob_ | phpboy, you using -trunk? |
11:08.43 | X-Rob_ | -trunk has _shithot_ echo cancellation now. Well. Shithot in comparison to what it used to be. Still woefully inadequate in comparison to a hardware EC. |
11:08.45 | RoyK | Vinsik: why are you using realtime if you're just dialling through a provider? that's just one peer.... |
11:10.27 | Vinsik | RoyK: got.. thanx!!! |
11:10.30 | Vinsik | RoyK: got it |
11:10.46 | RoyK | X-Rob_: what's so cool about it? |
11:10.52 | RoyK | X-Rob_: how's cpu load? |
11:10.58 | X-Rob_ | RoyK, it works, is what's cool about it 8) |
11:11.08 | RoyK | doesn't the one in 1.2 work? |
11:11.28 | X-Rob_ | cpu load's not that bad. I don't have any machines that are loaded up enough for me to notice difference in cpu load. |
11:11.51 | X-Rob_ | 10 line PRI into a Via Eden 1Ghz machine with software ec == bugger all cpu utilisation |
11:14.18 | *** join/#asterisk simondotsi (n=simon@mindtrip.entered.net) |
11:14.58 | robin_sz | OK girls |
11:15.21 | X-Rob_ | heh |
11:15.25 | X-Rob_ | damn that tab expansion |
11:15.28 | robin_sz | arse biscuits |
11:15.43 | simondotsi | Hello, I'm having troubles with echo while doing MixMonitor on a call (between two Sirrix ports), anyone aware of this ? |
11:15.44 | robin_sz | so where was I .. oh yes * |
11:15.58 | robin_sz | damn thing keeps quitting on me :( |
11:16.14 | robin_sz | im running monit now on a 1 minute cycle to keep it alive |
11:16.28 | robin_sz | but tis quit twice in two weeks now ... |
11:16.32 | robin_sz | that aitn good :( |
11:16.35 | X-Rob_ | robin_sz, there's a thing called 'safe_asterisk' that does that for you. |
11:16.42 | robin_sz | ahh .... |
11:16.57 | robin_sz | no other changes required? |
11:17.06 | robin_sz | just run safe_asterisk?? |
11:21.25 | X-Rob_ | that restarts asterisk when it crashes, and saves the core in /tmp |
11:21.27 | X-Rob_ | then read README.backtrace |
11:22.02 | *** join/#asterisk Johan (n=kvirc@194.151.113.2) |
11:22.04 | Johan | Hi all |
11:22.14 | grey | what is the channel for AMP help ? |
11:24.09 | X-Rob_ | grey, #amportal |
11:24.14 | X-Rob_ | but you can ask me |
11:25.13 | grey | thanks Rob |
11:25.16 | susin | some installed astgui? |
11:25.42 | grey | what is the best GUI for asterisk ? |
11:26.10 | DarkFlibble | grey, depends what you are trying to do... |
11:26.44 | Johan | Is here someone with experience with mISDN? I am trying to dial-out (or recieve a call) but it doesn't work. It seems asterisk failes to get an index or so, but the output is pretty verbose. |
11:27.01 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
11:27.01 | grey | setup an asterisk gateway to provide cheap international calls |
11:27.26 | DarkFlibble | grey, to the public? |
11:27.54 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
11:28.06 | X-Rob_ | Holy shit. |
11:28.12 | X-Rob_ | The EC actually works! |
11:28.16 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
11:28.23 | DarkFlibble | X-Rob_?!? |
11:28.26 | X-Rob_ | I have two phones, next to eachother, on hands free, and they're not feeding back! |
11:28.34 | DarkFlibble | ahhh |
11:28.35 | X-Rob_ | (well, only a little bit, for 1/2 a second or so before it chops it) |
11:28.46 | DarkFlibble | EC as in Echo Cancellation.. |
11:28.50 | grey | yes |
11:29.25 | mutilator | wish it worked that well for me |
11:29.32 | mutilator | i had to put my cisco box back into play |
11:29.34 | DarkFlibble | grey, then I would write a custom one or hack someone elses code... since every business I've seen works differently... |
11:29.43 | mutilator | because the echo on my new te405p i put in was so bad |
11:36.35 | Money5ack | hey ho |
11:36.42 | X-Rob_ | mutilator, were you using -trunk? |
11:37.26 | mutilator | 1,2,1 |
11:37.28 | X-Rob_ | (and, hint, echo problems -> http://bugs.digium.com/view.php?id=5520 ) |
11:37.30 | mutilator | and trunk |
11:37.57 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
11:39.27 | mutilator | i hope nothing in trunk is broken right now |
11:39.42 | X-Rob_ | trunk seems pretty good. |
11:40.49 | mutilator | nah |
11:41.03 | mutilator | that was already posted to trunk |
11:41.09 | mutilator | before i had my echo problems |
11:41.26 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.220) |
11:41.27 | mutilator | this was like last... |
11:41.37 | mutilator | wed morning |
11:41.44 | mutilator | 25th in the early AM |
11:43.56 | *** join/#asterisk moreece (n=m@196.46.142.23) |
11:44.02 | *** join/#asterisk leopardus (n=leopardu@217.22.179.69) |
11:44.33 | moreece | help!?@! - does anyone have an idea how to disable my Call rating engine and routing -> rate_engine.conf? |
11:44.43 | moreece | my asterisk is crashing cause their is no database |
11:45.18 | clive- | moreece howzit |
11:45.33 | moreece | howzit clive |
11:45.33 | moreece | any ideas??? |
11:46.22 | clive- | I am not familiar with that software working with * but why dont you try changing your dialplan so it doesnt use this cal rating stuff |
11:47.08 | moreece | hmmm, not sure I've configured my sip.conf and extensions.conf and I dont see it within there however this was preconfigured ---> let me check quick |
11:47.28 | DarkFlibble | moreece, what rating engine are you using? |
11:47.35 | DarkFlibble | I know of two... |
11:47.42 | moreece | *checking* |
11:47.46 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
11:49.04 | moreece | ; Call Rating Engine Configuration File |
11:49.04 | moreece | ; |
11:49.04 | moreece | ; Copyright (C) 2003 by Troll Phone Networks AS |
11:49.04 | moreece | ; |
11:49.04 | moreece | ; This program is distributed under the terms of the GNU General Public License |
11:49.05 | moreece | ; as published by the Free Software Foundation; either version 2, or (at your |
11:49.08 | moreece | ; option) any later version. |
11:49.18 | moreece | ; |
11:49.18 | moreece | ; $Id: rate_engine.conf.sample,v 1.5 2003/11/28 16:20:46 tholo Exp $ |
11:49.21 | moreece | sorry for the paste, should have used pastebin |
11:49.24 | DarkFlibble | ahh... the troll phone one.. |
11:49.30 | DarkFlibble | nope, never used it.. |
11:49.58 | DarkFlibble | http://www.voip-info.org/wiki/view/Asterisk+addon+rate-engine <-- might help tho |
11:51.13 | moreece | ta |
11:51.18 | moreece | *checking* |
11:51.46 | DarkFlibble | http://www.trollphone.org/files/ <-- might be useful.. |
11:55.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
11:56.28 | *** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk) |
11:56.43 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.220) |
11:57.40 | buzzyd | Does anyone know how I can play a message when I get Got SIP response 404 "Not Found" message ? |
11:59.40 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
12:00.55 | *** join/#asterisk Modcuts (n=sam@ppwood.gotadsl.co.uk) |
12:01.29 | *** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.220) |
12:01.47 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
12:02.51 | moreece | all righty, simply remote the entire rate_engine package from the systems starts up without it |
12:02.58 | moreece | *removed* |
12:03.30 | *** join/#asterisk Krill (n=majestic@210-84-11-13.dyn.iinet.net.au) |
12:14.04 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
12:15.20 | *** part/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
12:15.35 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
12:16.07 | *** join/#asterisk Tribastian (n=tribasti@62-2-138-202.business.cablecom.ch) |
12:16.15 | Johan | what is the difference between ptp and ptm? When I have an ISDN-card and want to dial to a phone number, then I am talking about ptp, is that right? |
12:19.49 | *** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com) |
12:20.22 | Tribastian | hello all |
12:20.37 | *** join/#asterisk mko-025 (n=korpim@p5498BE5D.dip0.t-ipconnect.de) |
12:23.54 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
12:25.03 | *** join/#asterisk snoops (n=blah@202-0-37-253.cable.paradise.net.nz) |
12:27.41 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
12:28.52 | snoops | hey - I've just purchased a linksys SPA-3000 Sipura, and was wondering if you guys could point me in the right direction (I grabbed the O'Reilly Asterisk - The Future of Telephony book, since it was published under creative commons license, which I am currently reading through) of some resource links/information I should know, etc |
12:29.29 | *** join/#asterisk beside (n=beside@tigbis.lt) |
12:29.38 | DarkFlibble | a linksys sipura? |
12:29.53 | beside | i would like to ask if someone here worked on passive monitoring using hfc-e1 board? |
12:29.56 | DarkFlibble | voip-info.org |
12:30.21 | bigjb | does anyone know of a guide to intergrating asterisk with a h.232 gatekeeper? |
12:30.36 | DarkFlibble | voip-info.org |
12:30.44 | bigjb | =oP |
12:30.50 | DarkFlibble | its h.323 |
12:30.58 | Skumling | what are people doing this "monitoring" things for? for logging purposes or for debugging? |
12:31.11 | beside | Skumling: logging |
12:31.12 | DarkFlibble | Skumling, recording all calls normally... |
12:31.17 | gaupe | DarkFlibble: well it probably says cisco linksys on the box and sipura on the software |
12:31.22 | DarkFlibble | so they can laugh at them later... |
12:31.35 | beside | in the case I do, I have E1 stream copy and give only RX to hfc-e1 board |
12:31.36 | snoops | That's one of the things I'll be doing - recording calls when I call power company etc(after telling them of course) |
12:31.44 | Skumling | beside, DarkFlibble: hum okay... not that nice :-/ |
12:31.52 | beside | I need a way to read D channel HDLC packets and open and read B channel |
12:32.18 | Tribastian | hello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!! |
12:33.08 | beside | as I load multihfc module with debugging |
12:33.15 | beside | I can see hdlc packets in kern.log |
12:33.34 | beside | but how can I take them using for example C |
12:38.24 | *** join/#asterisk Wiiz (n=nick@host217-34-132-179.in-addr.btopenworld.com) |
12:38.27 | Wiiz | hi |
12:38.57 | *** part/#asterisk snoops (n=blah@202-0-37-253.cable.paradise.net.nz) |
12:38.57 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
12:44.33 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
12:46.26 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
12:47.03 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
12:47.53 | Dr-Linux | hi |
12:48.00 | Dr-Linux | exten => _81NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
12:48.33 | Dr-Linux | i don't want my users to dial _81 << this "1" |
12:48.43 | Dr-Linux | what do i need to change in this line? |
12:48.45 | Dr-Linux | exten => _81NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
12:51.15 | Tribastian | hello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!! |
12:52.15 | *** join/#asterisk cpm (n=Chip@border1.avitecture.net) |
12:52.27 | Wiiz | Dr-Linux |
12:52.30 | Wiiz | what do u awnt them to dial |
12:52.38 | Dr-Linux | Wiiz: yes sir |
12:52.40 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
12:53.09 | Dr-Linux | i want them to dial _8NXXXXXX the other 1 is needed by telco |
12:53.35 | Wiiz | so delete the 1 |
12:53.42 | Wiiz | add an X |
12:54.33 | Dr-Linux | Wiiz: no, but user will still need to dial 816504500 |
12:54.44 | Dr-Linux | i don't want to dial 1 |
12:56.08 | Err | so *asterisk* has to dial the 1, right, and not the user? |
12:56.22 | Dr-Linux | Err: yessssssssss you are right |
12:56.25 | Dr-Linux | thats what i want |
12:56.35 | Dr-Linux | but sorry i can't explain with my bad english |
12:56.40 | Err | so you have asterisk dial 81${EXTEN:1} |
12:56.59 | Err | (which dials 8, then 1, then everything that the user dialed except the first number (the 8) |
12:56.59 | Dr-Linux | Err: and what would be the petren ? |
12:57.26 | Err | Dr-Linux: hm? what's a petren? |
12:57.39 | Dr-Linux | exten => _81NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
12:58.00 | tbs | Skumling: *You* think recording calls is mean? ;-) |
12:58.01 | Err | exten => _8Nxxxxxx,1,Dial(Zap/g1/81${EXTEN:1}) |
12:58.51 | DarkFlibble | Skumling, in the UK a private individual can record a call as long as one party is aware of it... |
12:59.04 | DarkFlibble | .. at least according to my last emplyer... |
12:59.06 | Dr-Linux | Err: so now user will not dial 8 and 1 ? |
12:59.09 | tbs | DarkFlibble: It's the same in .dk |
12:59.29 | Err | Dr-Linux: yes, with that line, users would dial 8<number>, and asterisk would dial 81<number> |
12:59.32 | tbs | It's a weird law... |
12:59.56 | Dr-Linux | Err: okey great . let me try this |
13:00.14 | Err | heh, in the US the *federal* law requires that one of the parties know - but many states have laws that require both parties to know |
13:00.15 | dpryo | hehe, I hate that my cellphone sends *BEEP* every minute when recording calls |
13:00.22 | viperdude | Darklibble: thats how I understand it but have you noticed how many calls centers now say calls may be recorded for training purposes.... i am guessing they are fearing litigation |
13:00.26 | DarkFlibble | it sometimes comes in useful, since most big companies record all calls... and you can prove that one of their representitives said x but did y |
13:00.49 | Err | common courtesy suggests that you should tell people that you're recording their calls |
13:00.52 | DarkFlibble | call centres aren't individuals... |
13:01.05 | Err | (not that common courtesy has anything to do with telemarketing ;-) |
13:01.21 | fugitivo | viperdude: call centers use recordings for training, that's true |
13:01.32 | cron | "your message might be recorded for training purposes" |
13:01.52 | cron | ooh .4 |
13:01.54 | dpryo | «all you say will be held against you» |
13:02.04 | DarkFlibble | "your call maybe recorded because we are paranoid" |
13:02.09 | cron | well |
13:02.28 | cron | different laws require user information to be recorded and held for ammounts of time |
13:02.34 | viperdude | i think they say training purposes as it sounds less big brother |
13:02.45 | dpryo | of course :) |
13:02.47 | Tribastian | or 1982 |
13:02.53 | Err | 84 even ;-) |
13:02.59 | viperdude | lol |
13:03.00 | cron | example, I wonder if PIPEDA would go into play for social engineerign |
13:03.00 | beside | DarkFlibble: in my case I'm not talking about individuals or companies, but about Law Enforcement Agency |
13:03.30 | DarkFlibble | Law enforcement hold all calls in the uk |
13:03.47 | DarkFlibble | since they may need them to "safeguard the life of the caller" |
13:03.54 | cron | man thats sad :( |
13:04.03 | viperdude | the BBC was talking about the NSA and the wire tap scandel in the US, last night. Does anyone reckon VoIP taps are in common use by the law enforcement yet? |
13:04.37 | DarkFlibble | I recon voip taps are used...but nowhere near to the extent normal landline taps are |
13:04.40 | cron | i wouldn't doubt it |
13:04.41 | Err | I'm certain that VoIP calls are monitored in and out of the US; internally, maybe so and maybe not |
13:04.44 | warmcat | Use a VPN if you're worried |
13:04.46 | Dr-Linux | Err: 8 is also going out, i wanna strip it |
13:04.54 | Dr-Linux | Executing Dial("SIP/4092-26e9", "Zap/g1/815935400") in new stack |
13:04.57 | beside | in my country no, but in some yes, VoIP are monitored |
13:04.57 | DarkFlibble | since a lot of voip is untappable without major problems |
13:04.57 | viperdude | does a VPN worry the NSA? |
13:04.57 | tbs | DarkFlibble: A few months ago a Danish consumer tv-programme described a case with a person, who had been called by a tele marketing company, who wanted to subscribe him to a new ADSL-line. He agreed to have some more info sent to his home. A few days later, he received an order confirmation. When he called them, they claimed to hold a recordring of him agreeing to subscribe. |
13:05.05 | Err | Dr-Linux: oh, so you want outbound to dial only 1<number>? that's not what I thought you said |
13:05.14 | warmcat | The question is how much budget you attract |
13:05.16 | Err | exten => _8Nxxxxxx,1,Dial(Zap/g1/1${EXTEN:1}) |
13:05.33 | tbs | DarkFlibble: however, they wouldn't let him hear the recorded conversation, because it "contained sensitive information" |
13:05.44 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
13:05.53 | Dr-Linux | Err: 8 is my patren |
13:05.56 | warmcat | There's only so much cracking capability around, it will be focussed on the Really Bad Guys |
13:05.59 | fugitivo | tbs: lol |
13:06.06 | DarkFlibble | tbs, surely that would be covered in the data protection act in the uk, and across the EU |
13:06.11 | tbs | DarkFlibble: They couldn't quite explain what the point of the recording was then :D |
13:06.21 | warmcat | if you are not a bad guy at all but interested in privacy there's no reason to burn that limited resource on your VPN |
13:06.48 | *** join/#asterisk gevious (n=chatzill@dsl-146-112-82.telkomadsl.co.za) |
13:07.01 | Err | Dr-Linux: so, give me an example of what you want to dial internally, and what needs to be dialed externally; the last line I wrote will take an internally-dialed 8<number> and dial, outbound, 1<number> - if that's not what you want, what *do* you want? |
13:07.08 | Tribastian | hello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!! |
13:07.14 | Dr-Linux | Err: this is the way 1(number) i wanna strip 8 |
13:07.43 | Dr-Linux | i don't want 8 to send out, user will dial 8 but asterisk will not send it out |
13:07.57 | Err | that last line I wrote should work, then |
13:08.08 | Err | (a concrete example with numbers would help out a bunch, here) |
13:08.31 | Err | you're using too many pronouns for what you're trying to describe :-) |
13:08.42 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:08.51 | Dr-Linux | Err: sorry for my bad english |
13:09.16 | Err | Dr-Linux: it's not a problem - just give me an example of a number dialed internally, and what you want asterisk to actually dial on the outbound line |
13:10.23 | Dr-Linux | well, i want like this 85935400 << 8 is a patren and 5935400 is a number |
13:10.52 | Dr-Linux | Err: but the damn telco says dial 1 before the number |
13:11.26 | Err | ok, so "exten => _8Nxxxxxx,1,Dial(Zap/g1/1${EXTEN:1})" should work, I think |
13:11.28 | Dr-Linux | so it will be like this 8+1+5935400 |
13:12.21 | *** part/#asterisk frenzy (n=frenzy@196.45.144.41) |
13:13.21 | Dr-Linux | Err: yeah, thanks it works, i tried same before, but i was missing an X in patren |
13:13.51 | Dr-Linux | Err: i wanna ask you a another question |
13:14.32 | Dr-Linux | Err: one of my x-lite client has low bandwidth, so when he calls me on my x-lite extension, he hears me good, but i can't hear him fine |
13:14.50 | Dr-Linux | but when i call him on his xlite extension, we hear fine each other |
13:14.59 | Dr-Linux | what could be happen? :S |
13:15.23 | DarkFlibble | different negotiation depending on the call setup? |
13:15.46 | Err | yeah, that'd be my guess - you probably need to force a codec for his extension |
13:16.40 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
13:16.51 | DarkFlibble | just dropped my lappy... |
13:17.10 | Dr-Linux | hhm.. |
13:17.17 | Dr-Linux | Err: which one? |
13:17.43 | DarkFlibble | landed open on the screen and base like an A... seems alright tho.. |
13:18.00 | DarkFlibble | Dr-Linux, what codecs does x-lite support? |
13:19.04 | Dr-Linux | ulaw, alaw, iLBC and GSM |
13:19.16 | DarkFlibble | hmmm... supposedly speex and ilbc need a reg hack to work with asterisk in xlite |
13:19.35 | DarkFlibble | gsm would be the codec I would force for low bandwidth |
13:19.47 | Dr-Linux | DarkFlibble: i don't think ilbc needs registration |
13:19.55 | Dr-Linux | yeah, i know |
13:19.58 | DarkFlibble | works out about 20kbits/sec per channel |
13:20.12 | DarkFlibble | reg hack == registry hack |
13:20.18 | DarkFlibble | in windows... |
13:20.21 | Dr-Linux | but my codecs sequence is something like this >> iLBC >> GSM >> ulah >> alaw |
13:20.27 | DarkFlibble | http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite |
13:21.02 | Dr-Linux | i also have g729, but that doesn't work with x-lite |
13:21.25 | *** part/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
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13:32.10 | *** part/#asterisk cfh (n=luca@82.193.23.6) |
13:34.18 | jaike | wow..new release? |
13:34.20 | bigjb | whats the easiest way to add a few hundredths of a second pause into the dialplan before it goes into playing an automated messasge? |
13:35.52 | mutilator | make a perl script and exec it via agi to pause for a few hundredths |
13:36.02 | mutilator | or just pause for 1 second with Wait() |
13:36.45 | tronix | the mere act of exec'ing might very well provide the delay. :-) |
13:37.10 | bigjb | heh |
13:37.24 | mutilator | yea |
13:37.41 | mutilator | dunno about a few hundredths tho |
13:37.43 | tronix | can probably count on at least 10ms (1/100th sec) delay alone |
13:37.43 | mutilator | it's not that slow |
13:37.49 | tronix | due to disk i/o |
13:38.10 | cpm | Umm, can you stack Wait()Wait() ? |
13:38.30 | tronix | it's just that I don't think you can get that granular with Wait() |
13:38.50 | mutilator | only deals in seconds as far as i know |
13:39.10 | mutilator | bigjb: why ya need the pause? |
13:39.20 | tronix | could be for callerid or other stuff |
13:39.59 | Money5ack | anybody here who gets t.38 running with latest svn ? |
13:40.09 | mutilator | but for what.. usually theres always something that executes before you'de need to use callerid |
13:41.27 | bigjb | when it goes into auto attendant and asks "please dial the extension of the person" it just seems to imidieate |
13:41.30 | bigjb | imediate |
13:41.35 | bigjb | immediate |
13:41.40 | bigjb | bollocks |
13:41.41 | *** join/#asterisk fugitivo (n=ajf@201.255.176.83) |
13:41.44 | bigjb | you know what i mean |
13:41.50 | Tribastian | hello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!! |
13:41.57 | bigjb | its probably a lot to do with the fact that im using the pc without a headset |
13:44.35 | bigjb | Wait doesnt seem to affect it anywho =oS |
13:44.44 | bigjb | woman is just there straight away |
13:44.45 | Err | Tribastian: your firewall is blocking inbound calls |
13:45.15 | Err | Tribastian: your "firehol" rules are far too simple - you'll need to allow the UDP block that asterisk uses for incoming SIP data streams |
13:45.34 | fugitivo | what is "firehol" ? |
13:45.41 | bigjb | ill have a gander at sounds directory when back from customers, pretty sure that there is a silence in there somewhere |
13:45.48 | Err | it's some firewalling frontend, according to google |
13:46.01 | fugitivo | so, it's just iptables? |
13:46.06 | *** join/#asterisk saftsack (n=saftsack@p54A7E4D2.dip.t-dialin.net) |
13:46.13 | saftsack | _Sam--, hi |
13:46.21 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
13:46.58 | Err | it's a front-end to iptables, I guess |
13:47.08 | Err | Yet Antoher Program to make firewalls "easy" |
13:47.58 | beside | it's sounds like Firewall Hole :) |
13:48.24 | *** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br) |
13:48.38 | Drew___ | not only do you need to allow the udp block - but if its a NAT you need portforwarding |
13:48.52 | Tribastian | Err how would the line in the firehol would look like? btw thanks for answering |
13:49.02 | Err | Tribastian: I have no idea - I don't know anything about firehol |
13:49.02 | iCEBrkr | yay 1.2.4 |
13:49.12 | Err | Drew___: it doesn't look like a NAT |
13:49.18 | Err | of course, I don't know anything about firehol |
13:49.32 | Tribastian | Err: thanks anyway, at least somebody did talk to me... |
13:49.44 | iCEBrkr | lol |
13:49.52 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
13:49.54 | Drew___ | 1.2.4???? holy shit - more updates... ;) |
13:49.56 | iCEBrkr | WTF is Firehol? |
13:50.04 | Stephnie | hi |
13:50.05 | iCEBrkr | Drew___: Don't blink man! |
13:50.11 | Err | Tribastian: I've seen your question three or four times, but I don't really know what the answer is - which I'm sure is why nobody else answered either |
13:50.17 | Err | "I don't know" isn't a very useful answer |
13:50.24 | Drew___ | hi Stephnie |
13:50.33 | Stephnie | If outbound call is connected/Answered ...Can I play a FILE (.wav) ????? |
13:50.33 | _Paulo_ | <PROTECTED> |
13:50.43 | coppice | Firehol == a firewall that only protects you for 15 minutes |
13:50.44 | iCEBrkr | Stephnie: Sure |
13:50.54 | iCEBrkr | coppice: 15mins?! Woah! Crazy stuff! |
13:50.55 | Stephnie | wait wait |
13:50.56 | Tribastian | thanks again, so i see why nobody did answer... |
13:51.20 | iCEBrkr | Tribastian: What's wrong with iptables? Why not use it instead? |
13:51.46 | Flyboy-SR22 | Tribastian, have you tried dropping your firehol to make sure the system is operating properly without the firewall in place..? |
13:52.02 | Stephnie | iCEBrkr: actually I need to first check that I need to play a file or not....I mean not every time I want to play a file |
13:52.07 | Flyboy-SR22 | I use iptables on all of my * servers, works great |
13:52.09 | Err | well, it looks like it's also operating as the NAT box itself, so I'm not sure that he can |
13:52.15 | Flyboy-SR22 | ah |
13:52.17 | Tribastian | ICEbrKr: well i first suggested that but my team said that it is stupid what i am doing and i shall use the firehol because it should be much easier |
13:52.21 | iCEBrkr | Stephnie: Ok? You can check for conditions |
13:52.31 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
13:52.31 | Stephnie | what should I check ? |
13:52.36 | Flyboy-SR22 | no way to move the * box to a public IP protectec by iptables..? |
13:52.36 | Stephnie | Dial application? |
13:52.42 | iCEBrkr | Tribastian: You'd get more help/support using iptables.. It's pretty standard. |
13:52.42 | Err | tell firehol not to block any inbound connections (forward them all to the machine), and see if it works |
13:52.51 | iCEBrkr | Stephnie: I dunno, what are you checking before you play the file? |
13:53.49 | iCEBrkr | Tribastian: First of all, if your team is telling you what you're doing is stupid-- Don't take advice from them :) |
13:53.58 | Tribastian | i will try, but even as i said in my conf that everything should not be blocked it still did not work, sadly we do need a masquarating wich is in the firehol... |
13:54.01 | Stephnie | if there is a voice mail...then I want to play a file...otherwise NOT... |
13:54.18 | iCEBrkr | Stephnie: Simple |
13:54.29 | iCEBrkr | Stephnie: Start here.. http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
13:54.52 | iCEBrkr | Stephnie: HasNewVoicemail() |
13:55.39 | Flyboy-SR22 | Tribastian: Have you verified that the traffic is actually making it through firehol..? Do you have logging enabled on the firewall..? |
13:56.21 | Stephnie | iCEBrkr : it's not an incoming call.....It's an outbound call |
13:56.32 | Stephnie | iCEBrkr : for example |
13:56.41 | iCEBrkr | Stephnie: OK, I think you need to describe what your end result is going to be.. |
13:56.59 | Tribastian | it is enabled, i will have to check again, lot of changes have been taking place over the last days, so did not have time to check... |
13:57.01 | iCEBrkr | Stephnie: A lot of people have obfuscated ideas how things work and go about doing it the wrong way. |
13:57.01 | Stephnie | iCEBrkr : I dial your number.....if you pick it up then I'll talk to you...otherwise I will leave a message through a .WAV file... |
13:57.22 | Stephnie | :) |
13:57.29 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
13:57.54 | iCEBrkr | Stephnie: So you're trying to make it call someone when they have a voicemail? |
13:58.16 | Flyboy-SR22 | Tribastian: Here is what I would do..check your logs on the firewall and fire off asterisk -rvvvvvvv and watch to see if you see anything coming in at all while you are attempting the connection. If the logs on the firewall are no help, maybe you will see the traffic hit the * box.. |
13:58.30 | Stephnie | I want to play .WAV manually when there is a voice mail..... |
13:58.33 | fugitivo | is xlite free for a company? |
13:58.42 | fugitivo | for internal use |
13:58.55 | Stephnie | I want to play .WAV manually when there is a voice mail at dialed number |
13:59.06 | Flyboy-SR22 | Tribastian: You could also throw iptables on the * box and allow all but log it to get a better picture of what you were seeing. TCPDUMP or Ethereal may also help you pinpoint where the traffic is blocked, but my first suggestion would be the firewall logging |
13:59.11 | iCEBrkr | Stephnie: What's the wav file gonna say? |
13:59.18 | Tribastian | but we could one time as we turned of the firehol in a very early status (before everything was connected with our callcenter) call in, so it must be the firewall, but i will do as you said an come back here again, thanks... |
13:59.41 | Stephnie | iCEBrkr : "LALALALAL ....Laaaaa...Lets make things better" |
13:59.42 | Flyboy-SR22 | Tribastian: NP |
14:00.16 | Stephnie | iCEBrkr : :-) |
14:01.06 | iCEBrkr | Stephnie: You're still not being descriptive enough for me to help you do exactly what you want to do |
14:01.31 | Stephnie | iCEBrkr : sorry...ok now let me explain... |
14:01.34 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
14:02.53 | Stephnie | iCEBrkr : for example: I dial your number ...if you are available then ofcourse We'll talk....but if you are unavailable then I want to PLAY a WAV file at your voice mail... |
14:03.29 | Stephnie | iCEBrkr : that's the as simple as I can explain..... |
14:03.41 | iCEBrkr | Stephnie: So why not use VoiceMail()? |
14:05.15 | Stephnie | is that what I need? |
14:05.26 | iCEBrkr | Stephnie: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail |
14:06.28 | Stephnie | yep reading .. |
14:08.14 | Stephnie | that is for incoming voice mails |
14:08.28 | iCEBrkr | That's for when someone is on the phone or doesn't answer |
14:08.36 | *** join/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za) |
14:08.49 | Stephnie | when someone calls me ....or at my number...right? |
14:09.12 | iCEBrkr | Stephnie: If someone calls you and your line is busy or you don't answer, you have Asterisk jump to VoiceMail() |
14:09.28 | iCEBrkr | It'll prompt the caller to leave a message depending on your extensions status |
14:10.02 | Stephnie | ok ..what about if I call friend and want to play MP3 at his/her voice mail??? |
14:10.06 | iCEBrkr | Damnit!!!! How do you tell Meetup.com to stop sending you daily updates?!?! |
14:10.13 | Stephnie | my friend* |
14:10.16 | iCEBrkr | Stephnie: Huh? |
14:10.28 | Supercross | hello everyone |
14:10.37 | Stephnie | what about if I call my friend and want to play MP3/WAV at her voice mail??? |
14:10.41 | Stephnie | got it now??? |
14:10.54 | iCEBrkr | So you call someone, and then what? |
14:11.03 | iCEBrkr | You want to stream in mp3's while you're on the call? |
14:11.14 | Stephnie | YES!!!! MP3 or WAV... |
14:11.36 | I-MOD | wav, not that hard....mp3, harder |
14:11.44 | Stephnie | ok I got for WAV |
14:11.45 | iCEBrkr | There's a way to do it, but I'm not sure how.. You have to do it from outside of Asterisk I believe. |
14:11.47 | Stephnie | go* |
14:11.57 | *** mode/#asterisk [+o drumkilla] by ChanServ |
14:12.34 | Stephnie | I-MOD have something to say ... |
14:12.37 | Stephnie | :) |
14:13.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:13.30 | Stephnie | I-MOD ?? |
14:13.52 | I-MOD | 1 sec, reading backlog |
14:14.32 | Stephnie | ok.. |
14:15.11 | mutilator | has anyone ever run into problems with a |
14:15.15 | mutilator | Zoom X5 GS Ver 2.0.1-00 ADSL modem |
14:15.19 | mutilator | and sip? |
14:15.25 | mutilator | i keep getting sipura 2002 ata's |
14:15.36 | I-MOD | iCEBrkr: what Stephnie wants is for asterisk to detect whether or not she has been sent to voicemail on someone else's box and if she has hit voicemail, play a pre-recorded message into it |
14:15.38 | iCEBrkr | I wish I had that problem. |
14:15.39 | mutilator | and when i plug them into the zoom modem it locks up the modem |
14:15.52 | mutilator | soon as i unplug the ata it works perfect |
14:16.08 | mutilator | this has happened with 6 different atas and different modems |
14:16.18 | iCEBrkr | I-MOD: Sounds like Telemarketing spam to me. |
14:16.31 | I-MOD | :) |
14:16.46 | Flyboy-SR22 | mutilator: what is the problem..? |
14:16.48 | *** join/#asterisk RevK (n=RevK@flawless.1ec.aaisp.net.uk) |
14:16.58 | mutilator | the dsl modems lockup with i plug in ata's |
14:16.59 | iCEBrkr | mutilator: umm, does the modem have RJ45 ports for extra network devices or something??? |
14:17.08 | mutilator | yea it's has 4 ports |
14:17.09 | Stephnie | I-MOD: no |
14:17.10 | fugitivo | I-MOD: app_amd (answering machine detection) |
14:17.12 | RevK | I have a dumn question on asterisk's handling of jitter... |
14:17.14 | Flyboy-SR22 | mutilator: I had to replace my sisters Zoom modem several time due to problems with the hardware, but it only effected * when it ws down !! |
14:17.33 | iCEBrkr | fugitivo: You realize there's no traces of app_amd on the wiki or anywhere else, right? |
14:17.34 | fugitivo | I-MOD: it's not 100% accurate |
14:17.40 | fugitivo | iCEBrkr: right |
14:17.44 | Stephnie | I-MOD: example: I call my friend and want to play MP3/WAV at her voice mail??? |
14:17.53 | fugitivo | iCEBrkr: do you want the module? ;) |
14:18.01 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-169.nas28.salt-lake-city1.ut.us.da.qwest.net) |
14:18.01 | iCEBrkr | fugitivo: :P Duh! |
14:18.06 | Flyboy-SR22 | mutilator: Wow - haven't seen that problem...its the only device you are plugging into the modem..? have you tried a switch instead..? |
14:18.26 | mutilator | there have been 1 & 2 pc's also plugged in |
14:18.33 | mutilator | but the problem was resolved down to plugging in the ata |
14:18.45 | mutilator | 6 different times now |
14:18.46 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
14:18.47 | mutilator | different locations |
14:18.56 | mutilator | different ata's and different modems |
14:18.58 | iCEBrkr | mutilator: You got a hub? Plug the ATA's into a hub instead of into the modem |
14:19.03 | fugitivo | iCEBrkr: where do i send it? |
14:19.10 | iCEBrkr | fugitivo: icebrkr@cyberdyne.org |
14:19.12 | Flyboy-SR22 | mutilator - so basically you plus the ata into a switch on your network and it kills the modem..? |
14:19.12 | mutilator | i'll try it at the next customer with that problem |
14:19.13 | iCEBrkr | that'll work |
14:19.56 | RevK | WHilst asterisk may cope with jitter, does it cope well with packet reordering at all? |
14:20.40 | mutilator | i was thinking maybe a firmware problem with the modem |
14:20.48 | mutilator | because i ran into it on one customer |
14:20.56 | mutilator | wehre they tried the new msn voice chat, and that uses SIP |
14:21.01 | mutilator | and it locked their dsl modem |
14:21.11 | Stephnie | I-MOD: still reading log? |
14:21.11 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
14:21.12 | *** join/#asterisk Seedy (n=Seedy@65.200.153.2) |
14:21.37 | *** join/#asterisk gr0mit (n=w10277@206.41.25.138) |
14:21.51 | RevK | We are seeing signs that iax does not cope with packet reordering... I am wondering if this is expected... |
14:22.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
14:24.47 | *** join/#asterisk danzig (n=chatzill@ruc-kj-013.ruc.dk) |
14:25.09 | danzig | EHLO all! |
14:25.37 | danzig | Anyone know about any progress with a skype channel for *? |
14:26.40 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
14:26.50 | cuco | danzig, this will not happen. skype does not like that. |
14:27.16 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
14:27.28 | fourcheeze | danzig: how could that work? |
14:27.33 | iCEBrkr | cuco: You'd almost think they'd want some sort of inter-connect... |
14:27.42 | danzig | Yes, they just published their specifications! (for people to make hardware phones, but if one can make a HW, one can also make a SW) |
14:27.48 | fourcheeze | more likely skype have to join the sip world eventually |
14:27.51 | Dr-Linux | any open source voice recognation program work with asterisk? |
14:28.05 | danzig | dr>>Festival |
14:28.06 | bweschke | Dr-Linux: sphinx would be your best bet there |
14:28.06 | iCEBrkr | Dr-Linux: Sphinx Supposedly.. |
14:28.06 | fourcheeze | danzig: url? |
14:28.17 | iCEBrkr | danzig: Festival isn't voice recognition |
14:28.21 | *** join/#asterisk svenna_ (n=svenna@p548D23B9.dip0.t-ipconnect.de) |
14:28.23 | Err | RevK: define "does not cope" - does it drop connections, or simply ignore the late packets? |
14:28.26 | danzig | sorry |
14:28.29 | synthetiq | any asterisk agi people here who use perl? |
14:28.34 | iCEBrkr | Festival is text to speech |
14:28.37 | cuco | danzig, you cannot connect skype to another vpio network, as far as i understood, this is part of the eula |
14:29.01 | RevK | Err, it asks as if the packets were dropped, i.e. audio break up. |
14:29.10 | Err | I would expect that |
14:29.19 | danzig | yup, I got mixed up with text to speech. Will just go find skype URL |
14:29.34 | Err | ...unless the late packets are still in time to be played, which is doubtful if they're behind later packets :-) |
14:29.39 | RevK | We have 40ms packets, and around 60ms random jitter independantly on two channels, so it is possible for packets to arrive one out of order |
14:29.49 | Dr-Linux | Sphinx is an open source? and good to use with asterisk? |
14:30.02 | Err | do you have a jitter buffer sufficient to allow delayed packets to still be in time to be played? |
14:30.18 | RevK | What I wanted to confirm is that an out of order packet would be held to be played if the jitter buffer is at a sensible level, and the later out of order packet correct slotted in if in time. |
14:30.26 | clive- | use the jitterbuffer |
14:30.32 | RevK | Also, if the late arrival of an out of order packet caused the jitter buffer time to push up to allow for more. |
14:30.34 | bweschke | Dr-Linux: yes - it's open source - is it good to use with Asterisk, well it's probably the best you can find w/OpenSource tools. |
14:30.50 | RevK | The jitter buffer settings are set up in iax.conf, anything else I need to know about setting up jitterbuffers? |
14:30.58 | Err | I wouldn't expect the jitter buffer to auto-tune by default |
14:31.18 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
14:31.38 | RevK | Yes, but tuning for variable jitter without reordering is not quite the same as allowing for packet reordering, and I am wondering if the code copes with that. I guess I can go and read the code. |
14:32.12 | iCEBrkr | I really wish I could build a 'TellMe' type system wit Sphinx. :P |
14:32.28 | Err | no, packet reordering should be the same, really |
14:32.31 | Dr-Linux | bweschke: Thanks, could you please suggest Sphinx link or any user manual for this? |
14:32.37 | RevK | I would hope so... |
14:32.41 | Err | jitter is jitter |
14:32.56 | RevK | Practical testing suggests not |
14:33.16 | Err | sufficient jitter must cause reordering, unless your delay monotonically increases forever :-) |
14:33.17 | iCEBrkr | Dr-Linux: Getting Sphinx to integrate into Asterisk == PITA |
14:33.19 | SibRw0rk | does the topic mean that the new version of asterisk is a memory leak, or fixes a memory leak? |
14:33.34 | RevK | Err, jitter is jitter. I can see how one could code for jitter without reordering quite easily and break when there is reordering. |
14:33.35 | Err | (jitter or drops, one, of course) |
14:33.45 | Dr-Linux | iCEBrkr: PITA ? :S |
14:33.59 | RevK | But reorder looks like a drop until the out of order packet arrives... |
14:34.00 | iCEBrkr | Dr-Linux: Pain in the ass |
14:34.09 | RevK | I'll have to read the code won't I... oh well. |
14:34.11 | Dr-Linux | Opsss |
14:34.17 | RevK | ffff |
14:34.24 | iCEBrkr | SibRw0rk: I read that as having fixed a memory leak |
14:34.25 | bweschke | Dr-Linux: ya - it's not a real user friendly process - but there are folks that have gotten it working |
14:34.52 | iCEBrkr | Getting Sphinx to work alone was a chore |
14:34.55 | Dr-Linux | hhm.. |
14:35.19 | Dr-Linux | you mean, using voice recognition system is not a good approach with asterisk yet? |
14:35.36 | *** join/#asterisk saftsack (n=lottc@p54A7EC24.dip.t-dialin.net) |
14:35.47 | iCEBrkr | Dr-Linux: There's just no real good voice recognition software for linux |
14:35.48 | bweschke | Dr-Linux: you can do it - it's just going to take alot more work than, say, configuring asterisk by itself does now. |
14:35.52 | iCEBrkr | Dr-Linux: that's free, of course |
14:35.56 | saftsack | _Sam--: hi |
14:36.04 | bweschke | there is a company working on a low cost solution for it |
14:36.19 | bweschke | within a couple months I think they plan to have something available |
14:36.59 | DarkFlibble | its not really that complex... |
14:37.15 | saftsack | is it importan to upgrade to asterisk 1.2.4? |
14:37.26 | DarkFlibble | you just need to use Fourier Tranforms and patten matching... |
14:37.32 | fugitivo | iCEBrkr: yes, loquendo is good |
14:37.37 | RevK | Looks like code tries to handle out of order - I'll read more |
14:37.45 | fugitivo | commercial software, obviously |
14:37.46 | DarkFlibble | which a limited set of patterns to match it should be pretty acurate... |
14:38.09 | DarkFlibble | freeform speech recognition tho... thats a different matter |
14:38.18 | iCEBrkr | fugitivo: Is that what TellMe uses? |
14:38.39 | fugitivo | iCEBrkr: i don't know, www.loquendo.com, they have asr, tts, and a lot of voice software |
14:38.51 | iCEBrkr | Oh, I think I've been there |
14:39.07 | fugitivo | i saw it working, it's wonderful |
14:39.22 | *** join/#asterisk ivanfm (n=ivanfm@201-1-164-43.dsl.telesp.net.br) |
14:39.59 | Money5ack | guys ? |
14:40.21 | Money5ack | i've got some curios compiling failures in chan_sip when i try to compile t38 support in |
14:41.15 | Money5ack | is here somebody who can help me a little bit ? |
14:41.28 | clive- | money I never knew t38 worked with asterisk yet |
14:41.35 | SibRw0rk | when i call out - i get a shit load of time out warnings then get a 603 declied |
14:41.57 | danzig | I dunno... Maybe I got hopefiull too fast - slashdot har "MSNBC has a look at some of the interesting gadgets that will be available for purchase now that Skype has published instructions on how to build the service into phones", but I cannnot offhand find the actual published specifications... |
14:42.07 | SibRw0rk | http://pastebin.com/532134 |
14:43.16 | Dr-Linux | hhm.. |
14:43.45 | Dr-Linux | http://turnkey-solution.com/asterisk-sphinx.html << here is a guide for Sphinx |
14:43.57 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
14:44.01 | Money5ack | clive: on the bugtracker list are some people who got t38 support into asterisk and its running, i had running it to for a while without know it and without any changes in sourcecode... |
14:44.19 | iCEBrkr | Dr-Linux: Yea, good luck with that :P |
14:44.33 | Dr-Linux | iCEBrkr: could you tell me please what makes it bad? PITA ? |
14:44.35 | SibRw0rk | iCEBrkr: http://pastebin.com/532134 - help? |
14:44.58 | sivana | is there a list of conf files that can safely be deleted? |
14:45.03 | Money5ack | now i had to reinstall that machine and there is no t38 support compiled in so i downloaded the t38_bits work from steveu and patched all files.. |
14:45.14 | Money5ack | now i have some big failures in chan_sip |
14:45.20 | *** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
14:45.20 | iCEBrkr | Dr-Linux: Getting Asterisk to talk to Sphinx in general.. It uses AGI() and file descriptor 3? Which is proprietary to Asterisk |
14:45.54 | Dr-Linux | yeah, it uses AGI() |
14:46.04 | iCEBrkr | Dr-Linux: I've gotten Sphine to recognize simple words.. UP, DOWN, LEFT, 1, 2, 3, YES, NO type stuff. |
14:46.17 | iCEBrkr | Dr-Linux: But I'll be damned if I could get it to control Asterisk |
14:46.48 | iCEBrkr | SibRw0rk: What's the issue? |
14:46.57 | elg | can I log in agents from the CLI? |
14:47.17 | elg | as in callback login |
14:47.21 | SibRw0rk | iCEBrkr: http://pastebin.com/532134 when i make a phone call i get these warnings, and then a 603 Declined |
14:47.37 | iCEBrkr | SibRw0rk: I'm not much of a network guy. |
14:48.03 | iCEBrkr | AFK |
14:48.08 | fugitivo | is ok a dual xeon 2gb ram for 100 g711 simultaneous calls? |
14:48.08 | SibRw0rk | i don't understand why i get a everyone is busy/congested @ this time - when i'm the only one on the network |
14:48.10 | SibRw0rk | argh |
14:48.14 | SibRw0rk | i have to go onsite anyway |
14:48.15 | SibRw0rk | bbl |
14:48.18 | junbug | hmm, any majors cost barries FCC etc.. if I wanna provide outbound voip access to a couple of clients on my colo box |
14:48.34 | RevK | We were not setting minjitterbuffer, I wonder if that is a factor - I'll test |
14:48.40 | brettnem | junbug: not besides the normal 911 requirements |
14:49.25 | junbug | brettnem: oh i have to provide that? even tho they have a land line |
14:50.16 | brettnem | my understanding is that if you are providing telephone service, you better provide 911 as well. |
14:51.04 | saftsack | fugitivo: 100 simaltaneous calls. wow :> |
14:51.09 | brettnem | bah |
14:51.33 | junbug | brettnem: *sigh* alrighty then ... oh well i can do like inphonex.com they charge $25 per 911 dial |
14:51.34 | danzig | fug> if you are not transcoding (translating from one protocol to another) calls, that box is much more than necesarry. If you are translating all the calls to another codec, it may not be enough. |
14:51.46 | fugitivo | no transcoding |
14:51.58 | brettnem | junbug: they charge $25 per 911 call? What a scam |
14:52.41 | wunderkin | fugitivo: a single xeon could easily handle that, actually im sure something smaller could also.. im doing a lot more than that on a single xeon |
14:53.00 | fugitivo | wunderkin: what do you do with your box? |
14:53.16 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
14:53.17 | saftsack | you guys a crazy. howto connect 100 telehones on asterisk? with a pri card? |
14:53.21 | danzig | I run 170 users on a pIII 750 Mhz 256 Mb ram - 10 simulatnious calls = 0.00 load. But it depends a lot on what u are doing - playing music, speech, conferences, transcoding, all take a lot more |
14:53.48 | fugitivo | saftsack: 4 E1, ip phones or softphones in my case |
14:53.52 | danzig | saft>> ethernet, if they are IP phones :-) |
14:54.02 | saftsack | fugitivo: :) |
14:54.16 | wunderkin | fugitivo: i tested 4 t1 doing backgrounddetect and connecting to a remote pgsql server |
14:54.26 | brettnem | when I get > 100 registrations.. asterisk starts acting funny.. leaving open file decriptors, etc.. no transcoding.. |
14:54.27 | saftsack | a e1 one port card is cheaper than a bri 8 port card. why? ^^ |
14:54.29 | fugitivo | wunderkin: with a single xeon? |
14:54.33 | wunderkin | ya |
14:54.47 | fugitivo | brettnem: sip registrations? |
14:54.56 | brettnem | fugitivo: yes |
14:55.08 | fugitivo | brettnem: 1.2.x? |
14:55.11 | coppice | xeons are like breasts. nature intends for them to come in pairs |
14:55.25 | brettnem | fugitivo: peers.. actually on both 1.0 and 1.2.x |
14:55.44 | fugitivo | brettnem: hmmm |
14:55.45 | brettnem | and who said coppice couldn't tell jokes? |
14:55.56 | fugitivo | coppice: for 100 simultaneous g711 calls, dual or single? |
14:56.38 | coppice | for just G.711 a single current xeon should do |
14:56.39 | wunderkin | fugitivo: im doing all ulaw, on a single xeon 3.0 2mb cache |
14:56.46 | wunderkin | yeah |
14:57.04 | fugitivo | great, the problem is if they want to start using voip providers |
14:57.13 | fugitivo | i'll tell them to buy a dual motherboard |
14:58.56 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
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15:00.28 | danzig | brettne>> I have 350 registrations all the time, and * i rock solid - never reboot. But I have no hardware in the box - all connected over ethernet. My impression is that zaptel and other hw drivers have more memory leack etc than * itself |
15:01.03 | brettnem | danzig: I'm not using any zaptel or any other hardware |
15:01.21 | fugitivo | danzig: what hardware for the server? |
15:01.37 | brettnem | single P4, 1 Gb mem |
15:01.53 | danzig | brett>> hmmm... well,all I can say is I am on * 1.0.9 on Debian, works great fo me... |
15:02.13 | brettnem | it'll be a long time before I ever say that asterisk works well for me.. heh |
15:02.30 | *** part/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za) |
15:02.32 | danzig | fug>> trash bamboo pIII 733 Mhz. Not to be recommended, but works fine. I have a backup. |
15:03.01 | fugitivo | danzig: g711?? |
15:03.16 | fugitivo | don't tell me you use g729 for 300 users with that box |
15:03.36 | danzig | um.. ALAW - that is g711A, is it not? I can't remember... |
15:04.08 | gaupe | danzig: that's right |
15:04.20 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
15:04.35 | danzig | no, I use ALAW (occansionally ULAW), there are 170 users, but that have 170 phones and 150 SIP trunks to various providers |
15:04.49 | fugitivo | danzig: are you using channel banks or gateways? |
15:04.50 | *** join/#asterisk ComPuTeR (n=NAZAN___@85.107.169.248) |
15:04.54 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
15:05.34 | oej | ~seen cresl1n |
15:05.36 | jbot | cresl1n <n=matt@gateway.digium.com> was last seen on IRC in channel #asterisk, 3d 13h 7m 13s ago, saying: 'nmsclera: DS0s on the PRI'. |
15:05.37 | danzig | neither - all phones are ethernet attatched (Grandstream GXP 2000), all trunks are SIP all the way out to the provider. |
15:06.04 | fugitivo | danzig: you use g711 for the trunks? |
15:06.22 | fugitivo | how many simultaneous calls? |
15:06.22 | *** join/#asterisk neon_kl (i=neon_kl@218.208.240.171) |
15:06.25 | danzig | fug>> 711a, yes |
15:07.13 | danzig | fug>> Very seldom see over 10 - i.e. "20" from asterisks point of view |
15:09.29 | danzig | fug>> we bought an Athlon 64 server for it, but saw that it was overkill, så haven't been bothered to move from the test box yet - but these people are students in a dorm - we don't lose 100000$/hour if the phones don't work |
15:12.34 | *** join/#asterisk CaViCcHi (n=matteo@81.208.84.216) |
15:12.44 | CaViCcHi | HI |
15:13.24 | Nugget | gros glandeur! |
15:13.38 | CaViCcHi | can someone help me with a function? |
15:13.39 | JunK-Y | nugget: hjeheh |
15:13.48 | CaViCcHi | IFTIME function |
15:14.26 | CaViCcHi | in ael |
15:14.47 | CaViCcHi | extensions.ael I have an extension... i use... |
15:14.50 | CaViCcHi | IFTIME(9:00-13:00|mon-fri|*|*?goto s|work); |
15:15.26 | CaViCcHi | it answers back... WARNING[38235]: pbx.c:1690 pbx_extension_helper: No application 'IFTIME' for extension ... |
15:15.42 | CaViCcHi | and hangs up |
15:16.36 | CaViCcHi | Any Help? |
15:16.43 | JunK-Y | what it does if u do show function IFTIME ? |
15:16.53 | CaViCcHi | it shows me how to use |
15:16.56 | CaViCcHi | so its compiled in |
15:17.15 | JunK-Y | show me 1 line before and after. |
15:17.19 | CaViCcHi | -= Info about function 'IFTIME' =-... |
15:17.37 | CaViCcHi | Temporal Conditional: Returns the data following '?' if true else the data following ':' |
15:17.46 | CaViCcHi | [Description] |
15:17.46 | CaViCcHi | Not available |
15:17.51 | *** join/#asterisk sevard (n=kynan@198.174.233.25) |
15:17.56 | JunK-Y | i mean in ur AEL |
15:18.01 | CaViCcHi | oh ok |
15:18.21 | sevard | Can some one please point me to a document explaining modules, how to load them, where to download them, what each module does. I can't find any documents and would like to learn this. |
15:18.27 | *** join/#asterisk P0L0 (n=n0n3@140.Red-83-58-255.dynamicIP.rima-tde.net) |
15:18.30 | danzig | Ca>> What * version? Can it be that that function is only available in a newer version? |
15:18.37 | CaViCcHi | inizio: |
15:18.37 | CaViCcHi | <PROTECTED> |
15:18.42 | CaViCcHi | <PROTECTED> |
15:18.42 | CaViCcHi | nolavoro: |
15:19.19 | CaViCcHi | Asterisk 1.2.1 |
15:19.31 | danzig | thats not the problem then |
15:20.06 | DannyF | damn, internet is broken today :/ |
15:20.41 | CaViCcHi | yes... but I'm following instructions... i cant understand why |
15:21.17 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
15:21.43 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
15:22.21 | RevK | IFTIME is not a conditional command, it is a function. What did you think goto s|lavora evaluates too exactly? |
15:22.38 | saftsack | hi i have asterisk + hylafax. are some hylafax experts here? |
15:22.49 | CaViCcHi | it just evaluates? |
15:22.49 | saftsack | because i want to know, howto print with hylafax |
15:22.55 | RevK | You want GotoIfTime |
15:23.03 | saftsack | also that all outgoing faxes are printed with hylafax |
15:23.14 | RevK | functions do evaluate to a value... What did you think Temporal Conditional: Returns the data following '?' if true else the data following ':' means? |
15:23.20 | RevK | It "returns a value", not "does a command" |
15:23.32 | RevK | Hope that helps |
15:23.38 | CaViCcHi | It helps a lot... |
15:23.46 | CaViCcHi | i work too much :P i need vacancy |
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15:29.41 | *** join/#asterisk apardo (n=apardo@87.218.45.104) |
15:29.57 | wunderkin | gone fishing |
15:30.05 | *** join/#asterisk b_52FREE (n=b_52FREE@adsl-212-18-192-81.adsl.iam.net.ma) |
15:30.06 | danzig | I think he meant holiday. Vacance in french. Or something. |
15:31.09 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
15:31.16 | [TK]D-Fender | well.. that was neat... |
15:31.25 | b_52FREE | hi |
15:32.42 | CaViCcHi | yea |
15:32.51 | CaViCcHi | just a Zlang :P |
15:33.15 | CaViCcHi | another nice question could be... how can i retrieve # from a call? |
15:33.33 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
15:34.42 | wunderkin | what number? |
15:34.54 | CaViCcHi | Phone number |
15:34.59 | CaViCcHi | caller |
15:35.18 | CaViCcHi | of Mr. I'm calling your asterisk |
15:35.54 | wunderkin | ${CALLERID(num)} |
15:36.47 | CaViCcHi | Here I am in italy... and sometimes u need that ur ISP allows you to see the # |
15:37.15 | CaViCcHi | i'll try anyway |
15:37.32 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-38lc0f9.dialup.mindspring.com) |
15:40.52 | Nugget | http://colo.slacker.com/stuff/italy.swf |
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15:46.23 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
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15:47.12 | [Atlas] | Nice! |
15:50.56 | ast_freak | Does 1.2.3 have a memory leak? I thought I saw some conversation about that yesterday. |
15:51.03 | ast_freak | Does 1.2.4 have a memory leak? I thought I saw some conversation about that yesterday. |
15:51.08 | ast_freak | :^) |
15:51.29 | [Atlas] | i thought i saw a conversation yesterday about that too but i think it was but .3 |
15:51.38 | [Atlas] | if i remember correctly |
15:52.00 | ast_freak | Wierd, I could have sworn it was 1.2.4. |
15:52.08 | sevard | <PROTECTED> |
15:52.14 | mzo_ | argh, a 1.2.4 out? already? |
15:52.14 | [Atlas] | can i give you my word as a spaniard? |
15:52.23 | [Atlas] | ;p |
15:53.13 | [Atlas] | ast realtime via odbc is in 1.2.x stable right? no need to cvs co? |
15:53.56 | ast_freak | lol |
15:54.01 | ast_freak | no good |
15:54.03 | ast_freak | :^) |
15:54.07 | [Atlas] | LOL |
15:55.30 | mdave | sevard, exten => 3 refers to the '3' being dialed |
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15:55.57 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
15:56.00 | mdave | exten => s,3 refers to nothing being dialed, and the '3' refers to the order in which the exten => specifications are referred to |
15:56.06 | mdave | aka 'priority' |
15:56.26 | sevard | mdave: thank you very much. is there a 'current time' module? |
15:56.42 | mdave | no idea |
15:56.56 | mdave | but see http://voip-info.org/wiki/view/Asterisk+config+extensions.conf |
15:57.08 | mdave | for more details on the syntax for extensions.conf |
15:57.26 | sevard | awesome, is there a module respository? |
15:57.44 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
15:57.52 | mdave | im not even sure what a module is |
15:57.54 | mdave | :P |
15:58.09 | sevard | like the wake-up call module |
15:58.15 | stack_ | Is it possible to have a queue with members that are external phone numbers or voicemail boxes? |
15:58.28 | sevard | i believe it works after placing the module in the extensions.conf file but I think the time on my server is screwed up |
15:59.22 | rob0 | <== just got back from watching http://colo.slacker.com/stuff/italy.swf and LOL :) |
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16:00.29 | RoyK | ka-ding |
16:00.32 | RoyK | anyone here from uk? |
16:00.37 | RoyK | gr0mit: ping |
16:00.56 | *** join/#asterisk ComPuTeR (n=TrgirL_@85.107.169.248) |
16:02.34 | fourcheeze | RoyK: me me |
16:02.58 | *** join/#asterisk rmorris (n=rmorris@d221-85-117.commercial.cgocable.net) |
16:02.59 | RoyK | fourcheeze: do you know a good place to place a colo? |
16:03.02 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
16:03.09 | fourcheeze | aha yes |
16:03.11 | fourcheeze | well |
16:03.15 | fourcheeze | what kind of place are you after? |
16:03.32 | RoyK | we prolly need a PRI and 1-4U worth rackspace |
16:03.49 | Dr-Linux | anybody is using sphinx voice recognition with asterisk? |
16:04.03 | fourcheeze | if RevK is awake I've a feeling that's his line, if he's the RevK that I think he is |
16:04.30 | *** join/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net) |
16:04.36 | diclophis | howdy all |
16:04.41 | diclophis | ... so what is up with SMS? |
16:04.41 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
16:04.53 | diclophis | is that the same thing as sending an email to 1231231234@vmail.net ? |
16:05.11 | fourcheeze | RoyK: sent you a PM |
16:06.45 | sevard | why doesn't this work? |
16:06.46 | sevard | exten => 556,1,Answer |
16:06.47 | sevard | exten => 556,2,Playback,current-time |
16:07.05 | diclophis | IIRC the syntax for playback is Playback(filename) |
16:07.15 | rmorris | I have * 1.2.x my x-lite phone works fine for logging into voice mail, but it seems * does not hear the tones from my hardware phone |
16:07.38 | RoyK | fourcheeze: ping? |
16:07.40 | sevard | diclophis: so current-time would not spit out the current time? |
16:07.45 | [Atlas] | so , if im reading the doxygen for asterisk realtime correctly, you still have to configure some things by hand in the config files? |
16:07.47 | *** join/#asterisk b_52FREE (n=b_52FREE@adsl-212-18-192-81.adsl.iam.net.ma) |
16:07.48 | fourcheeze | RoyK: not getting anything from you |
16:07.50 | rmorris | AT-320 phone sip |
16:08.00 | fourcheeze | RoyK: you on jabber? |
16:08.08 | diclophis | there might be a variable in the channel for the current time |
16:08.16 | diclophis | then you would need to do something like SayTime |
16:08.17 | diclophis | perhaps? |
16:08.25 | diclophis | but IIRC playback is only for files |
16:08.40 | RoyK | fourcheeze: wtf? hm... |
16:08.50 | RoyK | fourcheeze: only msn and ichat |
16:08.56 | fourcheeze | ok, I can do msn |
16:09.12 | RoyK | roy@karlsbakk.net |
16:09.13 | sevard | diclophis: Do you know where I could pull up a document that would have the correct syntax for that? |
16:09.32 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-227-29.claranet.co.uk) |
16:10.05 | sevard | diclophis: I'm looking at http://www.voip-info.org/wiki/view/say+time but to tell you the truth I don't fully understand it |
16:11.25 | *** join/#asterisk leopardus (n=leopardu@217.22.179.69) |
16:11.40 | rmorris | anyone? ideas why asterisk would not understand/hear tones correctly? |
16:11.43 | diclophis | do youi know anything about unixtime? |
16:11.51 | sevard | diclophis: I do not. |
16:12.08 | diclophis | the "time" variable needs to be an integer... that represents the number of seconds that have elapsed from the epoch |
16:12.10 | sevard | exec date ;) |
16:12.19 | sevard | I knew that. Yes. |
16:12.32 | leopardus | hello : which is the best, and simple to install, softphone for linux? |
16:12.48 | sevard | leopardus: if you're using X then the xlite phone from xten.com is simple. |
16:12.54 | diclophis | so now you need to find the current time somehow |
16:13.06 | diclophis | i would imagine there has to be a channel var |
16:13.23 | sevard | SAY DATETIME <time> <escape digits> [format] [timezone] |
16:13.31 | leopardus | sevard : xlite acts funny on linux |
16:13.46 | sevard | leopardus: I haven't experienced 'funnyness' |
16:14.42 | leopardus | sevard : I'm running xlite on the same server as asterisk |
16:14.42 | leopardus | sevard : should that be a problem? |
16:14.50 | sevard | I have yet to try but i imagine not. |
16:14.51 | diclophis | try SAY TIME ${EPOCH} |
16:15.46 | sevard | diclophis: I get the same thing. It connects but gives me dead air. I did 'reload'. |
16:16.03 | leopardus | sevard : I'm downloading lipz4 ,,,,?? |
16:16.06 | sevard | #time extension |
16:16.06 | sevard | exten => 556,1,Answer |
16:16.06 | sevard | exten => 556,2,SAY TIME ${EPOCH} |
16:16.12 | rmorris | just got off the phone with the phone support people. They say (surprise) this is an asterisk problem not a phone problem |
16:16.13 | sevard | leopardus: what?! |
16:16.18 | *** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com) |
16:16.37 | leopardus | sevard : I'm going to try lipz4 |
16:16.39 | JunK-Y | sevard: SAY TIME is an agi command. |
16:16.47 | diclophis | ah... |
16:17.00 | diclophis | is there a way of doing that in the dialplan? |
16:17.10 | JunK-Y | show application SayUnixTime |
16:17.13 | leopardus | sevard : going for tea, maybe I catch you again, bye |
16:17.40 | sevard | SayUnixTime([unixtime][|[timezone][|format]]) |
16:18.19 | *** join/#asterisk crich1999 (n=crich@p54BFC4D9.dip0.t-ipconnect.de) |
16:18.36 | diclophis | that should doit |
16:18.37 | ast_freak | Jan 31 09:18:18 DEBUG[20337] pbx_spool.c: Delaying retry since we're currently running '`, |
16:18.37 | ast_freak | <PROTECTED> |
16:18.41 | sevard | Awesome! |
16:18.45 | ast_freak | What's up with that? |
16:18.47 | stack_ | Is it possible to have a queue with members that are external phone numbers or voicemail boxes? |
16:18.58 | sevard | exten => 556,1,Answer |
16:18.58 | sevard | exten => 556,2,SayUnixTime |
16:19.03 | sevard | works perfectly |
16:19.04 | JunK-Y | sevard: * is awesome :) |
16:19.14 | diclophis | ha, i knew it was something easy |
16:19.21 | sevard | very much so, each day i get my hands dirtier with this and it surprises me even more |
16:19.36 | sevard | there is a weather module also, no? |
16:19.37 | diclophis | so now back to my question.. what is the most reliable way to sen hpone mail? |
16:20.01 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
16:20.11 | rob0 | sevard: that's my extension 8463 :) |
16:20.11 | NDT | Anyone have a copy of the file that was on the wiki with this description? --> PHP ZAP GUI: A single php page to view calls taking place on ZAP channel (copper line). Has the option to hang up call. Very Basic. |
16:20.36 | NDT | dead link now |
16:20.49 | sevard | rob0: can you copy and paste to me (perhaps in /msg) your 8463 part in your extensions.conf ? |
16:20.55 | rob0 | sure |
16:21.07 | sevard | thanks man |
16:22.08 | _Paulo_ | Steve Underwood is the man! |
16:22.31 | *** join/#asterisk beraldi (n=beraldi@gw-telecorp.telecorp.com.br) |
16:23.09 | sevard | rob0: oh, i thougt you meant weather :P |
16:23.23 | *** part/#asterisk CaViCcHi (n=matteo@81.208.84.216) |
16:24.38 | *** join/#asterisk infobox (n=dpizarro@200.31.105.196) |
16:25.54 | *** part/#asterisk beraldi (n=beraldi@gw-telecorp.telecorp.com.br) |
16:26.11 | *** join/#asterisk matteo (n=matteo@81.208.84.216) |
16:26.27 | *** join/#asterisk serg_b (n=sergey@9i.ru) |
16:26.40 | badboyz | is there a way to make asterisks playback a message to the caller, while the dialplan is still executing? |
16:26.53 | Nugget | background() |
16:26.56 | badboyz | background() doesnt do it .. background waits till its done playing before moving on |
16:27.32 | wunderkin | what else do you want to do badboyz |
16:27.52 | badboyz | wunderkin: i want the caller to hear a sound file, while the dialplan is still executing |
16:28.01 | wunderkin | what else is it executing |
16:28.10 | iCEBrkr | badboyz: you can't spawn 'threads' |
16:28.20 | iCEBrkr | badboyz: Everything executes top-down |
16:28.25 | badboyz | lets say they are hearing a greeting, while the dialplan is trying to find someone to answer the call |
16:28.33 | *** join/#asterisk j4m3s_ (n=j4m3s@user-24-214-119-188.knology.net) |
16:28.49 | wunderkin | badboyz so you are doing a findme |
16:28.53 | iCEBrkr | badboyz: You'll have to dump them in a queue or park their call while the dialplan is running around |
16:29.20 | mdave | anyone have any non-US test numbers? I just got fwdOUT setup and working, would like to give it a spin |
16:29.46 | badboyz | wunderkin: yes, but w/ audio playing during the findme process |
16:29.51 | iCEBrkr | mdave: FWD completed their PSTN inter-connect?? |
16:30.14 | wunderkin | i know there is a findme app in mantis |
16:30.17 | mdave | er, eh? no this is the community line-sharing thing |
16:30.25 | mdave | fwdout.net |
16:30.32 | iCEBrkr | oh so it works like DUNDi? |
16:30.40 | mdave | ive seen that, but dont know what it is |
16:30.40 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
16:30.43 | *** join/#asterisk Krill (n=majestic@210-84-11-13.dyn.iinet.net.au) |
16:30.57 | mdave | although something in fwdout mentions it will 'use dundi' if it doesnt have a route |
16:31.23 | mdave | where to find inf on DUNDi? |
16:31.34 | mdave | nvm |
16:31.41 | Dr-Linux | anybody is using sphinx voice recognition with asterisk? |
16:31.48 | iCEBrkr | Dr-Linux: LOL |
16:31.58 | iCEBrkr | mdave: Yea, it appears to be a lot like DUNDi |
16:32.36 | mdave | dundi.com appears to have a low of theory and 'why its great' type info, but seems to be lacking any detailed info on how it works or how to use it |
16:33.11 | mdave | ugh. and their press release is a MSword file |
16:33.22 | iCEBrkr | yeah and their whitepaper is a PDF |
16:33.25 | iCEBrkr | my machine is about to die |
16:33.32 | iCEBrkr | junk workstation here at the office :( |
16:33.33 | Dr-Linux | iCEBrkr: heh ;) |
16:33.33 | mdave | pdf isnt as reprehensible as word |
16:33.38 | Assid | sup iCEBrkr |
16:33.38 | mdave | annoying at times |
16:33.45 | iCEBrkr | It's sad when you have better hardware at home than at work. |
16:34.12 | iCEBrkr | mdave: Yea, well, I'm already in .NET and spawning a copy of PDFreader is thrashing my system |
16:34.12 | mdave | but regardless, there seems to be no tech info there.. any idea where to find info that gets to the point? |
16:34.37 | Dr-Linux | iCEBrkr: i insatalled everything for sphinx, but have a question |
16:34.38 | Dr-Linux | [root@I2C-PBX root]# ./sphinx-netclient.pl /var/lib/asterisk/sounds/thanks-for-using.gsm |
16:34.39 | Dr-Linux | Result: YES |
16:35.04 | serg_b | how can i move both legs of bridged call into meetme ? for example by assigning dynamic feature ? |
16:35.07 | iCEBrkr | mdave: The whitepaper describes it a bit better |
16:35.22 | iCEBrkr | Dr-Linux: I'm headed to lunch.. and it's been almost a year since I've tinkered with Sphinx |
16:35.22 | infobox | hi |
16:35.50 | rmorris | Anyone know why mpg123 would be running in the background all the time? |
16:36.08 | Dr-Linux | iCEBrkr: oky :S |
16:36.50 | mdave | rmorris, it sits waiting, paused, to play music on hold |
16:37.01 | rmorris | thx ! |
16:37.03 | wunderkin | Dr-Linux, im still fighting to get sphinx3 working properly |
16:37.05 | mdave | if you dont need or want any music on hold |
16:37.06 | sevard | Can anyone help me with this wake-up module? It seems to be working but not making making the correct call |
16:37.09 | wunderkin | in batch mode |
16:37.09 | mdave | edit musiconhold.conf |
16:37.22 | sevard | when somebody requests a wakeup it's put into a file in /var/spool/asterisk/wakeups/ |
16:37.34 | sevard | in the asterisk manager it says: |
16:37.39 | sevard | <PROTECTED> |
16:37.45 | sevard | but the phone doesn't recieve a call |
16:38.39 | Dr-Linux | wunderkin: did you try sphinux2 ? |
16:38.52 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
16:38.58 | *** join/#asterisk fulgas (n=fulgas@209.8.233.254) |
16:39.46 | *** join/#asterisk groogs (n=greg@d226-27-136.home.cgocable.net) |
16:42.15 | wunderkin | yeah, my problem is just getting it configured and working with the dictionary i want |
16:43.30 | rmorris | do I need to place the voicemail setting in both internal and external contexts? |
16:44.14 | rmorris | ie. voicemail works if I dial in zaptel, but not in the internal context |
16:45.06 | matteo | mm |
16:45.06 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
16:49.02 | sevard | whiskey tango foxtrot! |
16:49.03 | sevard | awesome |
16:50.11 | mover | any help needed for a notify problem |
16:50.39 | mover | mwi on realtime works with rtcache right? |
16:50.56 | rmorris | so back to my #1 question ... are there any settings that tell asterisk how to listen for tones? |
16:52.37 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
16:53.38 | pointer | is there a way to check to see if a file exists from within the dialplan? STAT() doesn't exist in the code but is in the wiki and EXISTS() appears to look in the DB or something along those lines |
16:53.49 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
16:55.24 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
16:55.42 | *** join/#asterisk pb_ (n=pb@2002:5246:d929:1:20e:2eff:fe2d:60bf) |
16:56.10 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
16:56.26 | znoG | anyone know how to view active Zap calls? |
16:57.25 | synthetiq | zap show channels |
16:57.38 | pointer | no, that shows all zap channels |
16:57.40 | synthetiq | err |
16:58.06 | pointer | show channels |
16:58.11 | pointer | but that shows all of them |
16:58.55 | wasim | asterisk -rx 'show channels' | grep Zap |
17:00.15 | *** join/#asterisk bigjb (n=nbigjb@195.60.10.114) |
17:00.30 | znoG | no, i want to view active calls to the Zap chans |
17:00.50 | znoG | ah show channels is good |
17:00.54 | pointer | any ideas on the file exists problem? |
17:01.39 | mover | anyone here expert of sip protocol and asterisk? |
17:01.57 | jontow | hmm, whats the consensus on FreeBSD 6.0-REL and asterisk 1.2+? |
17:02.06 | jontow | usable for VoIP only, in a very reliable fashion? |
17:04.24 | *** join/#asterisk masonf_ (n=masonf@dungle.vineyard.net) |
17:05.36 | znoG | hrm, is it normal to see this: |
17:05.37 | znoG | Zap/8-1 s@macro-llamar:100 Busy Congestion() |
17:05.42 | znoG | in show channels |
17:05.48 | znoG | they don't seem to disappear either |
17:07.08 | wasim | znoG: that means it hasn't detected calling channel hangup, put a congestion(60) or something |
17:07.22 | znoG | ok |
17:07.24 | masonf_ | can I make the hold button on my polycomm soundpoint play asterisk music on hold? |
17:07.24 | *** join/#asterisk matteo (n=matteo@81.208.84.216) |
17:08.49 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
17:09.11 | znoG | wasim: its actually showing channels that are not active anymore, such a call that took place earlier, shows "state" as Up, but i've confirmed the call has already taken place. |
17:09.15 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
17:11.52 | jaike | masonf: it does for me...i didnt have to make any special configuration |
17:12.05 | copantl | any body know how to convert a v400p from t1 to e1? |
17:13.11 | wasim | copantl: jumper it |
17:13.15 | mdave | how does one find out the available formats for the record and/or monitor file? one would think either of the doc pages on those would mention, but they dont seem to |
17:13.19 | wasim | copantl: or hardwire it in the driver |
17:13.53 | wasim | Use 'show file formats' to see the available formats on your system |
17:14.06 | mdave | aha |
17:14.07 | mdave | thank you |
17:14.18 | jaike | mdave: wav, wav49, gsm |
17:14.45 | copantl | wasim: dont have any jumper |
17:14.50 | diclophis | so... yea, whats up with phone mail? |
17:14.53 | diclophis | and Sms? |
17:15.03 | wasim | copantl: v400? |
17:15.15 | mdave | there a page anywhere comparing size/quality for those? |
17:15.20 | copantl | v400 t/e1 |
17:15.37 | copantl | varion right |
17:15.38 | copantl | ' |
17:15.43 | wasim | right, haven't used it |
17:16.29 | *** join/#asterisk _zebras (n=chris@62.69.89.38) |
17:16.48 | copantl | and www.zapatatelephony.org is down |
17:17.23 | wasim | its on the tor2, so those were manufactured either t1 or e1 |
17:17.28 | wasim | not switchable ... |
17:17.44 | jaike | mdave: i suggest wav49, small size. can be played on windows media player |
17:17.55 | _zebras | I'm running asterisk at home, when I simulate an external incoming call it works great, rings the right group etc - but when I ring from outside the line is permanently engaged (fxo connected to pstn) - any ideas how I can start to troubleshoot this.. |
17:18.52 | copantl | any body have a V400p card? |
17:19.51 | mdave | windows media player is irrlevent |
17:19.54 | mdave | this is for archival |
17:20.16 | mdave | i dont run windows anywhere for any reason |
17:20.57 | *** join/#asterisk QbY (n=Kelvin@adsl-068-209-210-253.sip.cha.bellsouth.net) |
17:22.29 | copantl | wasim: do you think is not switchable? |
17:22.47 | miztic | is there a way to make asterisk wait for a dialtone before sending digits with Dial() ? I'm having an odd problem where local calls don't work 80% of the time, its like the exchange isn't getting the first digit |
17:23.05 | znoG | this is weird, some calls that are made don't seem to close the channel when they've finished. I presume it's because Asterisk is not detecting a hangup, which is strange. |
17:23.21 | mdave | i dont need full wav/pcm quality, but id like slightly better than wav49/gsm |
17:24.00 | miztic | i'd even settle for inserting a delay between opening the line and dialing the number, inserting "," into the dialed number doesn't appear to help any |
17:24.07 | mdave | id like to find a comparison/overview of all the formats |
17:24.36 | mdave | anyway, i dont have time now |
17:24.45 | jaike | i think youll have to do some tests |
17:25.06 | mdave | i suppose |
17:25.10 | mdave | thanks |
17:25.18 | mdave | :) |
17:25.20 | mdave | im off |
17:25.46 | SibRw0rk | AH WTF! |
17:25.50 | SibRw0rk | i can't figure out what this error means |
17:26.00 | diclophis | so does anyone know anything about phone mail? |
17:26.26 | SibRw0rk | http://pastebin.com/532370 |
17:26.28 | SibRw0rk | anyone pleas3 |
17:26.29 | wasim | miztic: w is wait |
17:26.34 | miztic | aha |
17:26.37 | miztic | let me try that |
17:26.38 | miztic | thanks |
17:26.40 | SibRw0rk | what does Warning[315] entail? |
17:27.08 | wasim | dial(zap/1/wwwww9484858) |
17:27.17 | miztic | gotcha, doing that now |
17:27.20 | miztic | appreciate it :) |
17:27.39 | miztic | bingo |
17:27.39 | miztic | great |
17:27.44 | miztic | lets see if that helps |
17:28.47 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
17:31.20 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
17:32.50 | miztic | i think that's done it, thanks a lot |
17:32.59 | *** join/#asterisk roulduke_ (i=5sbgb7qg@p508D21A8.dip0.t-ipconnect.de) |
17:33.12 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net) |
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17:34.44 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net) |
17:37.19 | pauldy | I haatee commminggg back ttto aaaa blueeee channeeeel window without thhhat faintest idddea of whooo said my nnname or wwwhat the connntext wasss |
17:38.07 | znoG | thats why you use irssi and do /lastlog pauldy |
17:38.20 | rob0 | It was me who said that pauldy, but I was just joking. |
17:39.17 | pauldy | xchat isfor me cause I'm guid like that |
17:39.47 | *** join/#asterisk kio (n=kio@195-11.customer.cloud9.net) |
17:39.53 | *** join/#asterisk basty (n=basty@212.218.65.235) |
17:40.00 | basty | Hi |
17:40.54 | pauldy | your one letter off |
17:41.00 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
17:41.03 | iCEBrkr | pauldy: Yea, but with irssi, you can use screen which allows you to attach and then reattach console sessions from different terminals. :P |
17:41.28 | *** join/#asterisk samueltc (n=samuel@levinux.UQAR.UQUEBEC.CA) |
17:41.35 | samueltc | hi |
17:41.48 | samueltc | how I can track an Originate [call] in the manager api? |
17:41.58 | wasim | uniquecallid |
17:42.04 | samueltc | actually, I set a random callerid |
17:42.13 | samueltc | yes but how I known the uniquecallid |
17:42.19 | wasim | its a var |
17:42.24 | Assid | umm.. is it possible to set the timezone that will be used in gotoiftime ? |
17:42.35 | iCEBrkr | samueltc: Track it by the extension |
17:42.38 | pauldy | iCEBrkr, I know but then I couldn't bitch and how much fun would that be |
17:42.46 | samueltc | I known, but when I originate the call, I don;t known the uniquecallid |
17:43.35 | samueltc | iCEBrkr: the extensions is not present in all events |
17:43.59 | iCEBrkr | samueltc: The ones you care about it is.. |
17:44.42 | iCEBrkr | samueltc: I've already started work on a soon to be professional grade call manager. I've worked extenstively with the manage port. The stuff you care about is trackable by extension |
17:44.43 | samueltc | I care about Newchannel and the extension is not present |
17:44.47 | iCEBrkr | or well.. Channel |
17:44.59 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
17:45.02 | iCEBrkr | samueltc: I think you're going about it the wrong way then. |
17:45.09 | basty | I have a little Problem with setting up dialplans on Asterisk. We have a main Asterisk-Server, that is connected to a SIP-Provider. The SIP-Provider requests calls in a string like: Dial (SIP/11<callingnumber>*20*0049{EXTEN:1}@sip.provider... If I insert a Number into the field "Dialingno" it sends this number for external calls. Now I am trying to figure out a way on how to make this more flexible. Means a customer connects with an other Asterisk to a SIP |
17:45.29 | *** join/#asterisk FastJack (i=fastjack@p5091F6F3.dip.t-dialin.net) |
17:45.54 | *** part/#asterisk gr0mit (n=w10277@206.41.25.138) |
17:46.01 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
17:46.19 | Dr-Linux | wunderkin: did you try sphinux2 ? |
17:46.28 | samueltc | iCEBrkr: Newchannel give me the state of the channel, but there is no extension in that event |
17:46.42 | iCEBrkr | samueltc: Why are you looking for Newchannel? |
17:46.49 | samueltc | 'ringing' |
17:47.00 | Dr-Linux | iCEBrkr: welcome back :P |
17:47.12 | iCEBrkr | samueltc: I suspect you're trying to do the same deal with like desktop manager/callerID type app? |
17:47.31 | samueltc | iCEBrkr: no |
17:47.54 | Nugget | http://www.echostore.com/skype-usb-to-rj11-adaptor.html <-- has anyone here used one of these (or equivalent) as a fairly lame way to link asterisk and skype? |
17:48.11 | Nugget | I'm thinking that box, plus an FXO might let me route to skype |
17:48.25 | iCEBrkr | Nugget: haha, maybe |
17:48.43 | *** join/#asterisk RoyK (n=roy@ti211310a080-2037.bb.online.no) |
17:48.47 | Nugget | it'd be ugly, for sure, but it might just work. |
17:49.03 | _zebras | where do I set the initial delay before asterisk answers the call, whether I direct incoming pstn direct to extensions or groups it still seems to ring 2-3 times before performing the incoming rules... |
17:49.09 | samueltc | If I could get the uniqueid of my call, that would work..., actualy I'm Originate the call with a uniqueid in the callerid field, then with that field I known the uniqueid |
17:49.10 | iCEBrkr | Nugget: No uglier than those cellphone cradles |
17:49.15 | Nugget | yeah |
17:49.31 | austinnichols101 | samueltc: but you still have to have your PC on |
17:49.39 | austinnichols101 | running skype |
17:49.42 | samueltc | austinnichols101: my pc? |
17:49.53 | iCEBrkr | Nugget: Shit, $60?? |
17:49.58 | samueltc | i'm not talkinga about skype... |
17:50.09 | Nugget | I think I'm going to order one, just to play with. |
17:50.17 | Nugget | cheap enough that if it doesn't work I won't mind |
17:50.22 | austinnichols101 | sorry: meant nugget |
17:50.22 | iCEBrkr | Nugget: Go for it, lemme know how it works out :P Then I'll have to get one too |
17:50.43 | iCEBrkr | samueltc: I'm lost as to what the end result of you're trying to do is... |
17:50.51 | Katty | so. i got the xrays. |
17:50.58 | Katty | and the dentist even said i had gorgeous teeth. |
17:51.08 | *** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com) |
17:51.24 | Katty | bad news is, i'm going in for surgery :< |
17:51.57 | _Sam-- | he liked your teeth so much he wants to pull a few out and keep them as his own! |
17:52.07 | samueltc | iCEBrkr: it's pretty simple, I originate call on the manager API, and then I track that call. But the problem is, once the call originated, I have no way to track it because I don;t known the uniqueid. |
17:52.08 | Katty | he can't do it. |
17:52.13 | Katty | it will require an oral surgeon |
17:52.20 | Katty | all 4 of them must go, he says. |
17:52.23 | Katty | curse of the little people. |
17:52.24 | iCEBrkr | samueltc: Even if you set ActionID or anything like that?? |
17:52.39 | iCEBrkr | samueltc: I guess ActionID is more for when you're waiting for a response back from a manage command.. hrrrm. |
17:52.42 | samueltc | iCEBrkr: yes, actionID don't give me the uniqueid |
17:53.16 | Netgeeks | Katty: Wisdom teeth? |
17:53.29 | SwK[Work] | anyone get a situation where chan_zap.c:1583 zt_set_hook: zt hook failed: Device or resource busy is filling the screen and the message log |
17:53.37 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
17:53.48 | samueltc | iCEBrkr: actualy i'm generating a random number in the callerid field of my Originate command |
17:54.04 | samueltc | then the callerid is present in all event I need to track |
17:54.09 | Katty | Netgeeks: yes. |
17:54.17 | JunK-Y | Nugget: let me know how it works, that will be a nice gift to brother-in-law. |
17:55.09 | iCEBrkr | samueltc: If you know the channel or extension/phone number you're doing the originate on----hrmm shit |
17:55.16 | iCEBrkr | Ok, I'm kinda stumped |
17:55.25 | Nugget | I just ordered one. I'll let everyone know how it works out |
17:55.25 | Netgeeks | Katty: Well, the good news is that you will most likely get the fun gas... I did |
17:55.29 | iCEBrkr | Cuz with my call manager, I know the originating extension number and I track that.. |
17:55.34 | Nugget | Holding off on buying the FXO, though. :) |
17:55.59 | iCEBrkr | Nugget: Well, if it works the way the device claims, then it's quite obvious you could use it to interface with Asterisk |
17:56.05 | iCEBrkr | errr, wait. |
17:56.05 | samueltc | iCEBrkr: if 2 people call the same extensions? |
17:56.05 | Netgeeks | I had all 4 removed at the same time, and it took me out of commission for about 3 days |
17:56.06 | iCEBrkr | Maybe not |
17:56.08 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
17:56.25 | iCEBrkr | samueltc: Wouldn't matter, cuz you have the Originating extension/number |
17:56.25 | *** join/#asterisk fugitivo (n=ajf@201.255.176.83) |
17:56.36 | samueltc | ok |
17:56.44 | iCEBrkr | Nugget: That USB thing is gonna try to literally dial, which means it'll want a dialtone |
17:57.03 | Netgeeks | Katty, I don't remember any significant pain. |
17:57.15 | Netgeeks | some soreness, dull ache kind |
17:57.15 | Nugget | yeah, let me clarify. I've got one FXO I use for POTS right now. I'll use it for testing and if it works out I'll add a second FXO. |
17:58.04 | iCEBrkr | Nugget: I might be confused.. But those cheap-o Intel voice-modems, they're considered FXO or FXS? I continually get those confused/backwards |
17:58.14 | Nugget | FXO |
17:58.17 | justinu | FXO |
17:58.19 | iCEBrkr | ok |
17:58.22 | Nugget | FXO plugs into a dial tone. FXS makes a dial tone. |
17:58.28 | Katty | Netgeeks: :> |
17:58.32 | fugitivo | ~fxofxs |
17:58.33 | jbot | rumour has it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
17:58.33 | iCEBrkr | Nugget: ok, ok ok.. gotcha |
17:58.34 | junbug | any good external fxo's these days |
17:58.37 | Katty | now /thats/ what i wanna hear! |
17:58.46 | justinu | junbug: spa-3000 |
17:58.54 | iCEBrkr | Nugget: So, how's that USB device going to work with an FXO card if the FXO card doesn't provide dialtone? |
17:59.09 | justinu | katty: i had 4 teeth pulled when I was a teen to make room for the wisdom teeth |
17:59.10 | Nugget | it only needs a dialtone if you want it to be able to route calls out over PSTN. |
17:59.10 | junbug | the 3000 is fxo ... lemme check it out |
17:59.19 | Nugget | I just plan to use it as a gateway to skype |
17:59.21 | Katty | justinu: they're going in after mine. |
17:59.27 | Katty | Netgeeks: how do they put you out? |
17:59.30 | iCEBrkr | Nugget: Oh.. outbound from Asterisk to Skype |
17:59.32 | Nugget | so it's acting as an FXS, so to speek. |
17:59.36 | Nugget | or inbound. |
17:59.39 | justinu | Katty: when's d-day? |
17:59.45 | Katty | justinu: feb 17th |
17:59.47 | Nugget | but no PSTN connection at all. I already have asterisk for that |
17:59.47 | Netgeeks | Katty: I got Gas to start and Sodium Pent after the gas had me goofy |
17:59.48 | iCEBrkr | Nugget: You wouldn't be able to take inbound Skype calls |
17:59.54 | Nugget | it says it can. |
17:59.57 | Katty | Netgeeks: was it an iv? |
18:00.15 | justinu | Katty: boo, plenty of time to dwell on it too :( |
18:00.18 | iCEBrkr | Nugget: But how? The USB device is going to dial a phone number to connect the two legs of the call. |
18:00.22 | Nugget | no. |
18:00.35 | iCEBrkr | Nugget: It'll ring a phone? |
18:00.36 | Nugget | unless your definition of "phone number" is a lot looser than mine |
18:00.39 | Nugget | yes, it will |
18:00.42 | Katty | justinu: i'm not all that worried about the surgery, to be honest. |
18:00.43 | iCEBrkr | oh well, cool |
18:00.45 | Netgeeks | Katty I don't know. the gas had me soo goofy, I can't remember ANY shots, IV's etc. I came to in a dentist's chair with no IV in me |
18:00.59 | Katty | justinu: i was more worried about the financial issue... |
18:01.09 | Katty | Netgeeks: oh, goodly. |
18:01.12 | Nugget | It's exactly like an SPA-3000, except for skype instead of PSTN. |
18:01.14 | iCEBrkr | Nugget: Hrrm. Damnit, now I want one of these, just for fun |
18:01.15 | *** join/#asterisk MatsK (n=mk@cC30123C5.inet.catch.no) |
18:01.19 | jpablo | hey people, I'm having a problem with my pri, when i dial a number trougth it and the number doesn't exits it gives a busy tone imediatly, instead of passing the providers error message. |
18:01.23 | samueltc | iCEBrkr: http://pastebin.ca/39276 I can match the uniqueID with the callerid on line 14 |
18:01.24 | Netgeeks | Heya Drumkilla! |
18:01.29 | rob0 | <== had oral surgery in about 1968 |
18:01.42 | jpablo | any idea how can i hear the prividers error message? the people in the call center really need it |
18:01.55 | Katty | Netgeeks: will i remember much? |
18:02.01 | justinu | Katty: there is that too, isn't there... |
18:02.05 | samueltc | then I knonw that the phone is ringing on line 31 |
18:02.24 | Katty | justinu: yeah, but my dad volutneered to put it all on a credit card for me to pay back. |
18:02.37 | Katty | justinu: so i'm not all freaked out about billing anymore (= |
18:02.38 | Netgeeks | Katty: I had dreams that I was talking to my Teeth (they were like 5 feet tall and walked) and we were reminiscing the good times before they left... thats all I remember |
18:02.45 | Netgeeks | Katty: Like I said, gas is good |
18:02.46 | justinu | w00t, what would we do with out parents |
18:02.49 | iCEBrkr | samueltc: See, I match all my stuff on Channel |
18:02.55 | Katty | Netgeeks: so you dreamed...during the operation?! |
18:03.02 | Netgeeks | Katty: yes |
18:03.02 | Katty | Netgeeks: you're not like completely /out/? |
18:03.05 | Katty | oh dear. |
18:03.08 | Katty | this isn't good |
18:03.14 | iCEBrkr | samueltc: Cuz I'm only looking for events that are directed towards me, or my desktop call manager |
18:03.17 | iCEBrkr | hrrrm |
18:03.20 | Netgeeks | Katty: I had no recollection of the surgury itself |
18:03.31 | Zeeek | Katty it's less good than rice milk |
18:03.34 | justinu | NO2 will put you on the rings of saturn |
18:03.34 | Netgeeks | Katty: Like a real dream that you wake up and wonder where you are |
18:03.48 | justinu | a completely different plane |
18:04.10 | samueltc | iCEBrkr: but i'll probably originate call to the same channel (IAX2/out-gw00/14185551212) |
18:04.14 | Katty | Netgeeks: hmmmmmmmmmmmmm... |
18:04.15 | Zeeek | movies are lousy on planes |
18:04.21 | Katty | Netgeeks: were you not scared of the effects of the drugs? |
18:04.24 | samueltc | initiate more than one call on the same channel |
18:04.27 | *** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
18:04.27 | *** join/#asterisk [ToTo] (n=ToTo@host72-146.pool872.interbusiness.it) |
18:04.30 | iCEBrkr | samueltc: That's external. All your extensions will be SIP |
18:04.38 | justinu | the drugs are the best part! |
18:04.39 | iCEBrkr | Ohhh |
18:04.40 | iCEBrkr | ok |
18:04.42 | iCEBrkr | hrrm. |
18:04.54 | Zeeek | Katty get a general anasthetic |
18:04.57 | Err | Katty: it's called "twilight" anesthesia around here - you won't remember it at all (I've had it done several times) |
18:05.05 | Netgeeks | Katty: No, I was too scared of the whole surgery event to worry about the drigs |
18:05.05 | Zeeek | then even if you die, it won't matter |
18:05.08 | Netgeeks | drugs |
18:05.12 | Katty | Netgeeks: oh, ok |
18:05.17 | Katty | Zeeek: :<< |
18:05.27 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
18:05.37 | Zeeek | Is it more than like wisdom teeth pulling? |
18:05.39 | Katty | Netgeeks: describe procedure please |
18:05.43 | samueltc | If I could just update the callerid (after the match on the newcallerid event) in the manager API that would probably work... |
18:05.51 | brettnem | Err: Do they IV you for twilight? |
18:06.01 | justinu | nah, it's just gas |
18:06.06 | justinu | nawwws |
18:06.07 | Err | brettnem: it's just gas |
18:06.15 | Err | IVs are if they put your completely under, AFAIK |
18:06.18 | Netgeeks | All I can say is that I don't recall any un-pleasantness associated with the event... I remember going to the oral surgeon, getting the gas and having a finger pulse/oxy monitor I kept trying to push off as the gas took hold |
18:06.26 | _Sam-- | Katty: you need someone to hold you and tell you its going to be fine....because it will, its a minor procedure |
18:06.26 | brettnem | hmm.. last time I had NO2, I remembered everything.. I suppose I didn't have enough |
18:06.54 | Netgeeks | Katty: I remember the dream as clear as it was 2 minutes ago, and waking up in a pretty empty room in a reclining dentist chair with a dental assistant |
18:07.32 | Netgeeks | Katty: I went home and lived on jello/milkshakes, yogurt, etc. for the weekend and was back to work like tuesday (surgury was on friday) |
18:07.39 | Zeeek | I woke up in a waiting room with three other people holding bloody kleenexes just like the one I was holding! |
18:07.40 | brettnem | I remember my dentist trying to calm me down by telling me how much better it is being at the dentist than the orthodontist |
18:07.41 | justinu | Netgeeks: one question... was she hot? |
18:07.44 | iCEBrkr | LOL @ rGd's signoff |
18:08.02 | iCEBrkr | justinu: ^5 |
18:08.15 | SwK[Work] | http://pastebin.ca/39277 |
18:08.17 | SwK[Work] | h0 h0 h0 |
18:09.03 | Netgeeks | justinu: I honestly don't remember.... if the drugs do anything like a few beers, then I don't see how she couldn't have been hot |
18:09.10 | justinu | lol |
18:09.13 | Zeeek | heh |
18:09.50 | Dr-Linux | hi justinu ;) |
18:10.23 | justinu | hey there |
18:10.36 | _Sam-- | whats up Dr-Linux...you get your project finished? |
18:11.27 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net) |
18:11.37 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net) |
18:11.40 | Dr-Linux | _Sam--: we didn't try to start it yet. they java guy was on leave :) |
18:11.41 | *** part/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
18:11.46 | Netgeeks | Katty: When are you getting the procedure done? Any idea yet? |
18:12.07 | _Sam-- | did you ever figure out why the one read() never worked right? and gotoif |
18:12.37 | _Sam-- | that was still bothering me |
18:12.51 | _Sam-- | where gotoif was going to the wrong priority..remember? |
18:13.17 | Dr-Linux | _Sam--: i just figured out that the problem is with the script. not with dialplan |
18:13.50 | _Sam-- | did you find out what values the script/agi is wanting you to give it from the dialplan? |
18:14.07 | Dr-Linux | _Sam--: dialplan is fine, |
18:14.12 | justinu | Netgeeks: she said feb 17th ;) |
18:14.20 | _Sam-- | but the dialplan has to talk to the agi and give it values |
18:14.34 | Dr-Linux | yes |
18:14.47 | _Sam-- | you know what values the agi needs to get? |
18:14.50 | Dr-Linux | but what if the agi script was wrong ? |
18:15.22 | _Sam-- | if the agi script was wrong then it wouldnt connect to the informix database |
18:15.23 | Assid | hrmm.. is there a way to set the owner of voicemails to something else besides root? |
18:15.30 | Dr-Linux | _Sam--: do you think dialplan was wrong? |
18:15.57 | _Sam-- | Dr-Linux: i feel really stupid, but i couldnt figure out why the gotoif was going to the wrong priority |
18:16.05 | _Sam-- | so im not sure exactly what is/was wrong |
18:16.10 | _Sam-- | because i looked at it like 50 times |
18:16.17 | *** join/#asterisk CoderCR (n=creyna@ip21.farheap.net) |
18:16.38 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
18:17.02 | *** part/#asterisk CoderCR (n=creyna@ip21.farheap.net) |
18:17.03 | _Sam-- | but you tihnk it is because the agi was returning bad values |
18:17.05 | _Sam-- | ? |
18:17.07 | Dr-Linux | _Sam--: no thats not Read() or gotoif problem, thats script problem |
18:17.13 | Dr-Linux | yess |
18:17.24 | Assid | i set the owners to daemon:daemon in the asterisk.conf |
18:17.29 | Dr-Linux | _Sam--: agi always return 0 |
18:17.29 | Assid | but it still doesnt do that |
18:17.30 | _Sam-- | ok, that does make sense, but we SAW what the agi was returning...it was returning 0 |
18:17.37 | _Sam-- | and we said if 0, goto 20 |
18:17.40 | _Sam-- | but it went to 30 |
18:18.03 | tekati | I have the ability to switch providers to get me a little better upload speed from 412 to a meg. The new link is wireless and doing tests it will be a difference to my SIP provider from 85ms to 125ms. How will that effect performance? Anyone know? |
18:18.06 | Dr-Linux | _Sam--: but always 0, if i hit wrong i never saw anything else except 0 |
18:18.53 | _Sam-- | but even when it was 0, the gotoif didnt go to the right place i thought |
18:18.58 | _Sam-- | forget that it never returns 1 |
18:19.05 | _Sam-- | even when it returned 0, it didnt go to the right place |
18:19.08 | badboyz | anyone messed w/ the d option for the Dial commmand? its where a person can DTMF during the call -- i cant find more documentation but i would like it while its ringing, they could punch 1 and it goes to VM |
18:19.22 | Assid | hey iCEBrkr you around? |
18:19.48 | Assid | any clue on the file owner ship? |
18:19.56 | iCEBrkr | Assid: For? |
18:20.11 | Dr-Linux | _Sam--: i hate AGI's one thing, it doesn't show error output, it always show returning 0 with any thing |
18:20.31 | Assid | voicemails |
18:20.37 | Assid | they keep doing as root:root |
18:20.46 | Assid | i tried setting the ownership in asterisk.conf |
18:20.48 | Dr-Linux | _Sam--: as i'm playing with another thing since 4 hours, AGI returns 0 |
18:20.49 | Assid | but it doesnt help |
18:20.49 | _Sam-- | what about /var/log/asterisk/my_agi.log |
18:21.00 | Dr-Linux | ooo ic |
18:21.47 | iCEBrkr | Assid: Apparently you're running Asterisk as root.... |
18:22.16 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
18:23.04 | _Sam-- | Dr-Linux: do you have any experience with embedded linuces on SBC computers or MINI ITX? |
18:23.23 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
18:23.47 | *** join/#asterisk muzzz (n=chatzill@218.111.66.117) |
18:24.07 | fugitivo | i do with mini itx |
18:24.14 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
18:24.17 | _Sam-- | epia? |
18:24.20 | fugitivo | yes |
18:24.30 | _Sam-- | how many concurrent calls can a newer 1.3ghz epia handle? |
18:24.37 | _Sam-- | w/ say 512ram |
18:24.38 | justinu | embedded meaning what? no hard drive? |
18:24.47 | _Sam-- | embedded into a diskonchips or something |
18:25.05 | fugitivo | _Sam--: what kind of calls, codec, etc? |
18:25.06 | _Sam-- | embedded into the actual sbc or something as well |
18:25.24 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
18:25.24 | _Sam-- | fugitivo: no PSTN, just SIP --> IAX Provider or something |
18:25.33 | _Sam-- | using U |
18:25.57 | fugitivo | g711? |
18:26.01 | _Sam-- | yep |
18:26.28 | `lyme | Dial failed due to CHANUNAVAIL |
18:26.33 | `lyme | what all could that mean? |
18:27.18 | *** join/#asterisk dijit0_ (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net) |
18:27.46 | dijit0_ | would anyone happen to have any idea that when i call someone the caller id shows up as 19999991234 ?? thats nothing close to what i have set, lol |
18:28.18 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
18:29.27 | junbug | dijit0_: some providers dont support call id forwarding ... |
18:29.30 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
18:30.06 | dijit0_ | it USED to work though |
18:30.17 | dijit0_ | just recently, me and my friend havce the same problem all of a sudden |
18:30.24 | *** join/#asterisk razu (n=razu@ip59.cab62.mus.starman.ee) |
18:31.48 | dijit0_ | grr! when i call through idefisk and connect to nufone or wahtever, it works fine |
18:32.04 | dijit0_ | bypassing asterisk, that is... |
18:36.02 | dijit0_ | ugh! now its working... but why did it stop in the first place... |
18:36.12 | mzo_ | it's called a 'feature' |
18:36.21 | *** part/#asterisk mhnoyes__ (n=mhnoyes@user-2ivfmv1.dialup.mindspring.com) |
18:37.28 | dijit0_ | meaning it works half the time? lol |
18:37.40 | mzo_ | yes! |
18:37.42 | mzo_ | a feature! ) |
18:38.45 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
18:38.54 | dijit0_ | heh, know of any voip providers that allow you to receive callerid WITH NAME? that asterisk can work with |
18:39.07 | MattH | if someone is registering their asterisk server to mine with sip (yes I know maybe we should be using iax) what is causing this error when I try to send a call to them? |
18:39.07 | MattH | Jan 31 12:36:28 NOTICE[3716] chan_sip.c: Failed to authenticate user "+15703232166" <sip:+15703232166@63.174.244.172>;ta g=as7facadd4 |
18:39.12 | _Sam-- | dijit0_: teliax is doing caller id with name, works well |
18:39.25 | DarkFlibble | dijit0_ its generally an additional service... about 0.2c/ call |
18:39.59 | rajiv|work | _Sam--: is teliax's network fixed ? |
18:39.59 | _Sam-- | im sure there other places too, but teliax does all our origination so i dont know any others |
18:40.04 | _Sam-- | rajiv: is it ever? |
18:40.20 | dijit0_ | i c... ok thx |
18:40.33 | rajiv|work | _Sam--: can you recommend them? |
18:40.39 | _Sam-- | i dont have any problems with teliax's routing...but i have noticed for days 1-3% packet loss at their router. |
18:41.04 | _Sam-- | rajiv: i think it depends on your expectations...i could recommend them, but if you're not happy with them, then my recommendation doesnt matter :) |
18:41.08 | FuriousGeorge | is it asterisk or eyebeam that is playing the double "boop" callwaiting sound? if its the former, can i stop it? |
18:41.10 | _Sam-- | they work reasonably well for me. |
18:41.16 | zamsler | hmm |
18:41.16 | _Sam-- | i do have backup accounts, just in case. |
18:41.26 | zamsler | what kind of prices are you guys looking for ? |
18:41.54 | FuriousGeorge | well, i want to stop the sound, not the call from being "answerable" by that extension |
18:42.10 | FuriousGeorge | the booping drowns out the conversation for local party |
18:42.26 | _Sam-- | darwin_35: say something |
18:42.41 | _Sam-- | when are you going to fix the 1-3% packet loss at your router |
18:43.11 | justinu | heh, 1-3% loss isn't all that bad |
18:43.21 | _Sam-- | i notice it |
18:43.23 | _Sam-- | on calls |
18:43.24 | FuriousGeorge | how do we test packet loss at the router it |
18:43.34 | justinu | ulaw? |
18:43.37 | _Sam-- | yeah |
18:43.46 | [av]bani | yay its sam |
18:43.48 | justinu | must be the gxp not doing any PLC |
18:43.50 | _Sam-- | FuriousGeorge: something like mtr works |
18:44.33 | _Sam-- | hey their bani...how goes it? |
18:45.13 | [av]bani | your gxp2000 comment on the wiki got nuked somehow |
18:45.35 | _Sam-- | i tried |
18:45.35 | zamsler | hooooooah |
18:45.36 | justinu | congressional staffers screwing with the voip-info.org wiki too?? :P |
18:45.42 | _Sam-- | lol |
18:46.18 | [av]bani | i think it got nuked in an edit |
18:46.29 | [av]bani | someone else must have been editing at the same time |
18:46.58 | jhiver | ~seen p0lar |
18:47.00 | jbot | p0lar <~p0lar@64.254.225.62> was last seen on IRC in channel #asterisk, 318d 22h 10m 6s ago, saying: 'time to order..hehe'. |
18:47.22 | [av]bani | _Sam--: the only phones you have are gxp-2000 ? |
18:47.47 | zamsler | lol |
18:47.54 | _Sam-- | yep, and a few of those utstarcomm wifi phones |
18:48.09 | [av]bani | how are the utstarcomm? |
18:48.18 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
18:48.18 | [av]bani | i'm assuming they are equally cheap |
18:48.25 | _Sam-- | i guess since i dont have anything to compare them to....i would say they work ok |
18:48.30 | malverian[work] | Anyone know of some good festival voices (eg.. better sounding than the mbrola ones) |
18:48.46 | _Sam-- | i just dont know what else i would compare em to...i got my wife a plantronics USB wireless headset... |
18:48.57 | _Sam-- | i think the plantronics headset through a softphone works better , sound quality wise |
18:49.03 | [av]bani | compare them to the gxp2000's |
18:49.41 | _Sam-- | the gxp2000s sounds much better |
18:50.03 | _Sam-- | the utstarcomm gets some 'clipping' type sounds, like there is too much gain on someplace |
18:50.17 | _Sam-- | i have to turn the volume down pretty low for it to go away |
18:50.32 | _Sam-- | but all in all, i have 3 guys in a warehouse with the f1000s , and they never complain |
18:50.46 | _Sam-- | sometimes they have to reboot them because calls will stop working |
18:51.01 | justinu | my lusers would complain non-stop about that. |
18:51.15 | justinu | non-fucking-stop |
18:51.20 | _Sam-- | we are talking about once a week |
18:51.22 | _Sam-- | not once a day |
18:51.25 | justinu | doesn't matter |
18:51.26 | _Sam-- | maybe even less than once a week |
18:51.36 | rajiv|work | _Sam--: i want origination that works. i want my phone to ring when calls come in, and the quality to be acceptable. i'm using gizmo now and surprisingly it works. |
18:51.44 | *** part/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br) |
18:51.52 | _Sam-- | see, it helps too that i am the owner of the place, because when people complain, i can always tell them , dont like it, get your own phone, or theres the door |
18:51.57 | justinu | cool |
18:52.04 | hypnox | any dundi experts here? I am running a local dundi 'cloud'. Numbers are advertised and looked up okay, but the IP of the destination is always set to 127.0.0.1 - even though i use ${IPADDR} in dundi.conf as recommended |
18:52.30 | _Sam-- | rajiv: i think you could find alot of companies that would be able to provide reliably what you're looking for |
18:52.35 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
18:52.37 | _Sam-- | i am anxious to give asterlink a shot |
18:52.57 | _Sam-- | their server(s) are the closest ones that ive seen route-wise to me |
18:53.39 | *** join/#asterisk INOT|Bewildered (n=thehouse@80-195-138-8.cable.ubr07.uddi.blueyonder.co.uk) |
18:54.35 | trixter | they are a good bunch of guys |
18:54.48 | dijit0_ | this caller id problem seems to be related to whether i set it in idefisk or not... i dont want the user to be able to set it, but if not specified, it doesnt work for some reason |
18:55.03 | [av]bani | _Sam--: who do you use for origination? |
18:55.09 | _Sam-- | teliax currently |
18:55.13 | badboyz | so if you set a timeout to a s,1,Dial(SIP/200,30) -- where does it go when it times out? +101 ? |
18:55.28 | [av]bani | do your callers complain about choppiness? |
18:55.45 | [av]bani | i call out and people say its choppy, but i hear them fine |
18:56.48 | _Sam-- | no, i dont get many complaints at all. i am like 55ms away from the server i connect to. when i listen in on sales calls they always sound good, and i never hear the customer complain... |
18:56.50 | justinu | that's upstream problems |
18:57.03 | [av]bani | i dont get problems with junction networks, and theyre farther away |
18:57.13 | [av]bani | we have a ds3 |
18:57.17 | [av]bani | and its not loaded |
18:57.43 | _Sam-- | i am on a good ISP who is homed on three or 4 different backbones, i think that helps me out often |
18:57.47 | [av]bani | teliax is 30ms away and jn is 90ms |
18:57.52 | _Sam-- | because my path to teliax goes all the way over cogent |
18:57.54 | [av]bani | we're multihomed too |
18:57.58 | _Sam-- | no interconnects or anything |
18:58.11 | [av]bani | teliax has issues, jn doesnt even though jn is 3x as far |
18:58.22 | [av]bani | so i'm inclined to assume teliax has inbound issues |
18:58.28 | jpablo | hey people, I'm having a problem with my pri, when i dial a number trougth it and the number doesn't exists it gives a busy tone imediatly, instead of passing the providers error message, any idea how can i pass the error message audio back to my extensions ? |
18:58.36 | [av]bani | since our outbound goes over the same pipes |
18:59.04 | _Sam-- | brad_mssw was complaining about the same thing with teliax, but i think the problem is related to the route you take to teliax |
18:59.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:59.16 | [av]bani | _Sam--: i tried different routes, same results |
18:59.53 | _Sam-- | do you connect to voip-co3.teliax or voip-co2? |
18:59.58 | [av]bani | co3 |
19:00.06 | _Sam-- | for some reason co2 is like 1 hop closer |
19:00.10 | [av]bani | i hear them fine, callers say i sound choppy |
19:00.17 | _Sam-- | i connect to co2 |
19:00.29 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
19:01.13 | MRH2 | anyone use polycom phones and sync to a ntp server? |
19:01.20 | _Sam-- | bani if you want the guys IM info from teliax let me know, he is usually pretty helpful, which is one of the reasons ive stayed around. |
19:01.46 | Zeeek | MRH2 ya |
19:01.52 | _Sam-- | if he realizes that you know possibly as much as he does, he does tend to listen and try to help |
19:01.59 | _Sam-- | otherwise he will tell you to reboot your computer and send you on your way |
19:02.27 | MRH2 | mine are consistently 10 secs slower - do you get the same? |
19:02.38 | [av]bani | _Sam--: cool, icq? |
19:02.41 | Zeeek | no I kept getting on ehour diff! |
19:02.47 | Zeeek | one hour |
19:02.48 | _Sam-- | aolim, will message name |
19:03.20 | Zeeek | MRH2 I'd say you're not syncing with the server at all |
19:03.47 | _Sam-- | [av]bani: what are you doing with a DS3 over there? |
19:03.52 | brettnem | Hey, anyone know why I'd get a bunch of stuck SIP channels with method REFER? |
19:03.56 | _Sam-- | maybe you could talk teliax into putting a server on it :) |
19:03.59 | MRH2 | definately am ;) |
19:04.08 | [av]bani | _Sam--: isp |
19:04.16 | [av]bani | we resell dsl |
19:04.16 | Zeeek | the phone is running free though? |
19:04.27 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
19:04.28 | MRH2 | running free? |
19:04.32 | _Sam-- | nice, i owned an ISP from 94-2002, sold it off |
19:04.36 | [av]bani | :) |
19:04.40 | _Sam-- | who babysits your servers? :) |
19:04.41 | [av]bani | 94-02, wow |
19:04.43 | Zeeek | you can't have sync and be 10 secs off! |
19:04.46 | [av]bani | i do :() |
19:05.08 | [av]bani | we're replacing our inhouse POS phone system with * for PBX |
19:05.10 | _Sam-- | we have the new verizon fiber down here that is killing off the dsl resellers |
19:05.14 | _Sam-- | verizon FIOS |
19:05.18 | [av]bani | trying to figure out a good solution for PSTN |
19:05.21 | _Sam-- | i dont think they can resell it |
19:05.37 | [av]bani | teliax would work if they werent choppy |
19:05.48 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
19:05.58 | rajiv|work | [av]bani: maybe you need QoS on your upload? |
19:06.04 | _Sam-- | i use teliax to run my current biz...we do over 7 million a year in telephone sales...and teliax does work ok |
19:06.09 | [av]bani | i also tried iax, same |
19:06.16 | _Sam-- | telephone sales = mail orders sales, not sales of telephones :) |
19:06.23 | [av]bani | :) |
19:06.27 | [av]bani | mail orders of what? |
19:06.39 | _Sam-- | we sell motorcycle stuff for sportbikes |
19:06.42 | brettnem | here's a pastebin of those STUCK SIP channels in REFER method: http://pastebin.ca/39286 |
19:06.46 | brettnem | Any ideas anyone? |
19:06.47 | iCEBrkr | _Sam--: SELL ME STUFF |
19:06.53 | [av]bani | :) |
19:06.53 | _Sam-- | anything you want, at cost |
19:06.56 | _Sam-- | www.kneedraggers.com |
19:07.02 | iCEBrkr | _Sam--: Oh you're one of THOSE guys? |
19:07.06 | *** join/#asterisk DarkFlibb (n=darkflib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com) |
19:07.07 | iCEBrkr | :-/ |
19:07.11 | _Sam-- | which guys? |
19:07.17 | [av]bani | nice website |
19:07.21 | [av]bani | impressive |
19:07.28 | iCEBrkr | _Sam--: A few of the kneedragger.com guys used to hang out in #Motorcycles on EFNet |
19:07.30 | MRH2 | if the ntp is down i get flashing date/time of 1st jan - if ntp gets a successful response time is 10secs out |
19:07.35 | _Sam-- | that was me, ice. |
19:07.36 | iCEBrkr | This was years ago tho.. I'm not sure now |
19:07.39 | iCEBrkr | hahaha |
19:07.40 | [av]bani | heh |
19:07.41 | _Sam-- | i still have friends on efnet #motorcycles |
19:07.44 | _Sam-- | ryanb |
19:07.48 | brettnem | I have around 83 stuck sip channels.. anyone have any clues? |
19:07.52 | iCEBrkr | _Sam--: The internet is too small :P |
19:07.53 | [av]bani | the owner gets to loaf around on irc all day <3 |
19:07.57 | Damin | iCEBrkr: What's up bitch? :) |
19:07.59 | Zeeek | MRH2 never heard anything like that (not that that helps you, sorry) |
19:08.00 | stack_ | how does the "include" statement work with regards to the time options? If I have a set of holiday rules and then a daytime rule and the holiday was on a weekday, wouldn't it include both the holiday rule and the daytime rule? |
19:08.02 | iCEBrkr | Damin: Nap time!!!! |
19:08.09 | _Sam-- | iCEBrkr: that is funny shit...do you still hang out there? |
19:08.14 | Damin | iCEBrkr: Is that a sanctioned work event? :) |
19:08.17 | MRH2 | ok r urs in sych exactly? |
19:08.20 | denon | _Sam--: your "You may be interested in" feature is broke |
19:08.27 | brettnem | argh |
19:08.28 | iCEBrkr | _Sam--: Seriously tho, I got a '89 gixxer.. I need plastics!! not fiberglass shit :P |
19:08.30 | [av]bani | 'hacker safe' eg safe for hackers! |
19:08.36 | _Sam-- | i had a falling out with someone on that channel because i wouldnt give them a deal....the guy was 'desmo' i think |
19:08.39 | iCEBrkr | Damin: If I were President, it would be! |
19:08.48 | _Sam-- | denon: give me a page |
19:08.57 | denon | _Sam--: http://www.kneedraggers.com/list/1.2 - click "Goodridge speed bleaders" |
19:09.01 | iCEBrkr | _Sam--: LOL I remember Desmo.. |
19:09.04 | denon | I assume its an old product or something |
19:09.11 | justinu | desmodromic? |
19:09.12 | iCEBrkr | _Sam--: I was supposed to meet up with him at Daytona Bikeweek |
19:09.14 | brettnem | argh |
19:09.14 | denon | but you need to take it out of both tables ;) |
19:09.20 | brettnem | anyone want to talk about asterisk? :) |
19:09.26 | Zeeek | MRH2 like I said, it was off for an hour no matter what I did on the web interface. Rebooting without an ftp server didn't help, only with ftp and XML did it straighten up |
19:09.26 | MRH2 | what firmware are you on Zeeek? |
19:09.35 | _Sam-- | iCEBrkr: if you make it down there...look me up :) |
19:09.38 | _Sam-- | we race there every year |
19:09.39 | jhiver | nite' all |
19:09.41 | Zeeek | 1.4 something, it's a ip500 |
19:10.06 | iCEBrkr | _Sam--: I'm in Tampa/St. Pete.. if I get my bike together, I'll be at Bikeweek. |
19:10.08 | _Sam-- | denon: thanks...i will have the web guy fix it |
19:10.08 | brettnem | Anyone know anything about STUCK SIP CHANNELS? :) http://pastebin.ca/39286 |
19:10.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net) |
19:10.10 | denon | _Sam--: also http://www.kneedraggers.com/details/Brembo_Radial_Brake_Master_Cylinders--12-BREMBO-1.html - alt tag in the wrong place |
19:10.16 | iCEBrkr | _Sam--: hopefully this year, it won't be a 3hr shower.... |
19:10.39 | denon | also .. heh .. well, just tell your web guy to get his act together ;) |
19:10.45 | iCEBrkr | denon: haha |
19:10.47 | _Sam-- | iCEBrkr: nice, when im at the don ceaser next time i'll look you up :) |
19:10.57 | iCEBrkr | _Sam--: Rich bastard. |
19:10.58 | _Sam-- | but as far as plastics for your 89...that is a tougher proposition |
19:11.04 | iCEBrkr | _Sam--: LOL |
19:11.19 | iCEBrkr | I know.. I know. |
19:11.32 | iCEBrkr | I only need 3 more.. and a lower is the toughest to find |
19:11.45 | MRH2 | wll my config files are sorted fine |
19:11.49 | _Sam-- | maybe you should step up to the 21st century with a 6 year old bike :) |
19:11.54 | iCEBrkr | lol |
19:11.55 | _Sam-- | an 00 R6 would be fine :) |
19:12.01 | iCEBrkr | <-- Gixxer dude |
19:12.11 | iCEBrkr | tho, I find the R6 & R1 pretty sexy |
19:12.14 | _Sam-- | i bleed blue, we run a yamaha race team |
19:12.17 | _Sam-- | they give us bikes and money |
19:12.21 | MRH2 | i mean they are fine even adjust for daylight savings |
19:12.22 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net) |
19:12.30 | _Sam-- | but ive always been an R6 guy |
19:12.36 | iCEBrkr | _Sam--: I have a handful of friend who race CCS |
19:12.42 | _Sam-- | i probably know em |
19:12.50 | _Sam-- | in the florida region? |
19:12.52 | iCEBrkr | Yeah |
19:12.52 | fugitivo | ok |
19:12.55 | iCEBrkr | Tampa Track Junkies |
19:13.00 | fugitivo | enough of bikes |
19:13.03 | _Sam-- | sorry |
19:13.03 | iCEBrkr | fugitivo: lol |
19:13.04 | fugitivo | now let's talk about trains |
19:13.07 | Zeeek | MRH2 actually I probably never looked to see if there was 10 sec diff |
19:13.11 | _Sam-- | its like asterisk...addicting. |
19:13.12 | iCEBrkr | CHOOCHOOO |
19:13.12 | fugitivo | trains and monkeys |
19:13.28 | iCEBrkr | fugitivo: Whatever man, I got VoIP on my bike! |
19:13.29 | _Sam-- | if you really ever need anything i will be glad to hook ya up, ice. |
19:13.37 | fugitivo | iCEBrkr: 1337! |
19:13.38 | _Sam-- | and tell those losers on #motorcycles that i really miss them :) |
19:13.51 | iCEBrkr | _Sam--: Cool stuff!! I'll be in the market for a new helmet |
19:13.56 | iCEBrkr | Oh, I don't hang out there anymore |
19:13.58 | iCEBrkr | :) |
19:14.06 | _Sam-- | i am friends too with D3scart3s |
19:14.12 | _Sam-- | from #motorcycles, you remember him? |
19:14.14 | iCEBrkr | I remember him too |
19:14.21 | _Sam-- | we raced together at ccs a few times |
19:14.24 | _Sam-- | him and mrcrash too |
19:14.24 | justinu | what do you guys ride? |
19:14.51 | _Sam-- | i have 2 yamaha R6s (600cc) and 1 yamaha R1 (1000cc), and a yamaha 450 dirt bike |
19:14.51 | MRH2 | ok :) |
19:15.00 | justinu | cool |
19:15.02 | synthetiq | a huffy. |
19:15.04 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/cpg142/thumbnails.php?album=50 |
19:15.06 | _Sam-- | do you ride at all? |
19:15.06 | iCEBrkr | synthetiq: LOL |
19:15.14 | justinu | yeah |
19:15.16 | iCEBrkr | ^^^^^^^^^^ URL to my bike pics! |
19:15.20 | justinu | i used to have a gsxr600 |
19:15.24 | justinu | and a gsxr1000 |
19:15.26 | iCEBrkr | It's old skewl, but it's clean! |
19:15.34 | fugitivo | _Sam--: i like supercross bikes |
19:15.56 | _Sam-- | fugitivo: hell yeah, they are great if you are jumping 3000 feet in the air |
19:16.12 | fugitivo | :) |
19:17.31 | justinu | i had this for a while also: http://justinu.smugmug.com/photos/31062641-O.jpg |
19:18.01 | iCEBrkr | justinu: Oh fuck a duc :P |
19:18.08 | justinu | heh |
19:18.11 | iCEBrkr | :P |
19:18.56 | Katty | let's leave the poor ducks alone. |
19:19.01 | [av]bani | <_Sam--> to show my appreciation for #asterisk, free bikes for everyone in the channel |
19:19.10 | iCEBrkr | _Sam--: Man, you rock! |
19:19.11 | _Sam-- | lol!! |
19:19.15 | [av]bani | yay! |
19:19.22 | _Sam-- | justinu: too bad bennie b. sux anymore |
19:19.25 | [TK]D-Fender | ... oh the joy of the donorcycle.... |
19:19.29 | iCEBrkr | [av]bani: The only deal is, they're orange with Asterisk stars on them. |
19:19.32 | [av]bani | haha |
19:19.35 | mzo_ | i'd still ride it |
19:19.36 | Katty | [TK]D-Fender: i'll donor your cycle in a minute. |
19:19.37 | [av]bani | iCEBrkr: no problem! |
19:19.39 | iCEBrkr | lol |
19:19.45 | _Sam-- | lol |
19:19.51 | [TK]D-Fender | Katty: Mew. |
19:19.52 | _Sam-- | this is one my current bikes |
19:19.54 | _Sam-- | http://www.kneedraggers.com/racer6/images/DSC00086.jpg |
19:19.57 | Katty | [TK]D-Fender: mew :< |
19:20.18 | [TK]D-Fender | Katty: Actually saying what said backwards sounds like some sort of proposition ;) |
19:20.20 | [av]bani | i want a hello kitty cycle |
19:20.35 | iCEBrkr | _Sam--: Yea, I have to admit, it looks nice |
19:20.42 | fugitivo | justinu: you have a ducati??? |
19:21.05 | justinu | i did |
19:21.12 | doug | mmm |
19:21.12 | justinu | i sold my bikes |
19:21.14 | *** part/#asterisk doug (i=doug@zaxxon.telerama.com) |
19:21.21 | iCEBrkr | Fine, fine. I'll link directly to my bike too ... http://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/dsc00137.jpg |
19:21.48 | _Sam-- | this is some race team pictures from 05: |
19:21.49 | _Sam-- | http://www.kneedraggers.com/gallery/ |
19:22.14 | justinu | my 600: http://justinu.smugmug.com/photos/3171203-O.jpg |
19:22.27 | _Sam-- | iCEBrkr: damn, you got the 20th anniversary edition? :) jk |
19:22.31 | iCEBrkr | LOL |
19:22.35 | iCEBrkr | _Sam--: The original! |
19:22.42 | _Sam-- | thats a cool bike. |
19:22.59 | iCEBrkr | _Sam--: I actually would love to have the 20th anniversary edition one so I could park them next to each other :P |
19:23.07 | _Sam-- | iCEBrkr: my brother has one |
19:23.16 | _Sam-- | justinu: that is a nice picture...where is that from? |
19:23.32 | iCEBrkr | I'd park'm like this... |
19:23.32 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/DSC01963.JPG |
19:23.39 | iCEBrkr | warning, HUGE pic |
19:23.41 | _Sam-- | <looks like you need a rear tire, justin> |
19:24.06 | justinu | on the 600? |
19:24.06 | fugitivo | how much is a bike like that? |
19:24.26 | _Sam-- | 8000-11,000 USD |
19:24.36 | _Sam-- | the racebikes are like 20 with the work we do |
19:24.41 | *** join/#asterisk pbd (n=plancomm@12.144.118.36) |
19:24.44 | fugitivo | that's cheap |
19:24.52 | _Sam-- | the bikes are the cheap part |
19:24.53 | pbd | Greetings, all. |
19:25.03 | fugitivo | i'm sure i can't get a bike like that for less than 15k here |
19:25.20 | _Sam-- | fugitivo: then someone is ripping you off. |
19:25.29 | _Sam-- | the retail price of the units is like 8000-11000 USD |
19:25.35 | _Sam-- | except for the ducati that justin showed |
19:25.40 | fugitivo | _Sam--: yes, the goverment with taxes :) |
19:25.57 | iCEBrkr | I just want one of each... |
19:26.08 | _Sam-- | iCEBrkr: where was the last picture from? |
19:26.09 | iCEBrkr | 1000RR, Ducati 999, etc. |
19:26.14 | pbd | Quick question, and yeah, I'll take lumps for not knowing this off the top of my head.. but to those of you who have installed the 729 codec- how did you decide between the i386, i586, and i686 versions? 'arch', or should I know something I'm blind to at the moment. |
19:26.27 | fugitivo | pbd: wrong channel, this is #bikes |
19:26.37 | iCEBrkr | _Sam--: We have Quaker Steak and Lube ( it's a bar/resturant ) which holds a weekly bikenight |
19:26.42 | iCEBrkr | fugitivo: lol |
19:26.55 | fugitivo | #monkeys-and-trains soon |
19:27.06 | justinu | i think i still like the duc 998 better |
19:27.09 | pbd | Woa. Interesting. Client tells me I'm in asterisk users, shows the right person list.. but isn't. Sorry for the intrusion. |
19:27.27 | fugitivo | pbd: do you like bikes? |
19:27.41 | *** join/#asterisk saftsack (n=lottc@p54A7EC24.dip.t-dialin.net) |
19:27.45 | _Sam-- | lol |
19:27.52 | saftsack | _Sam--: hi |
19:27.52 | iCEBrkr | _Sam--: The bike scene is HUGE down here in Tampa.. I'm sure I could send a lot of business your way |
19:27.57 | _Sam-- | its funny that we (many of us) have the same interests with the bikes |
19:27.58 | _Sam-- | and phones |
19:28.04 | saftsack | i have a little question with hylafax. it runs now :) |
19:28.06 | _Sam-- | hey there sacky |
19:28.09 | fugitivo | and trains |
19:28.15 | iCEBrkr | and monkeys |
19:28.17 | saftsack | but howto print every fax which is sent? |
19:28.21 | iCEBrkr | MOOSE PENIS |
19:28.24 | saftsack | i thought on notify? |
19:28.31 | _Sam-- | saftsack: i use cypheus for that |
19:28.38 | _Sam-- | cypheus is a client that runs on a windows pc |
19:28.39 | fugitivo | saftsack: what bike model/brand? |
19:28.50 | saftsack | cypheus is a client for faxing |
19:28.54 | saftsack | or? |
19:29.03 | _Sam-- | yes, for faxing, and for retrieving and printing faxes |
19:29.14 | _Sam-- | it submits faxes from the windows pc to the hylafax server |
19:29.17 | saftsack | i mean the server as internal which should print all sent faxes for archiving them |
19:29.27 | _Sam-- | it does archive them on its own |
19:29.28 | *** join/#asterisk sthw45ywyw5 (n=sthw45yw@38.136.42.2) |
19:29.28 | iCEBrkr | I really should start looking into faxing with Asterisk |
19:29.36 | saftsack | _Sam--: yes i know in doneq |
19:29.37 | iCEBrkr | So I can have a 'fax machine' |
19:29.39 | fugitivo | i use spandsp |
19:29.46 | _Sam-- | <PROTECTED> |
19:29.50 | iCEBrkr | Tho, I don't see that happening since I don't have a land-line |
19:29.53 | iCEBrkr | :( |
19:29.59 | saftsack | _Sam--: yes but i want to have them in paperform |
19:30.03 | saftsack | also printed |
19:30.15 | fugitivo | saftsack: using lpd? |
19:30.22 | _Sam-- | like i said, i use cypheus for that....so that is all i know. |
19:30.24 | saftsack | i use cups |
19:30.28 | _Sam-- | cypheus prints every incoming fax for me |
19:30.31 | *** part/#asterisk pbd (n=plancomm@12.144.118.36) |
19:30.48 | saftsack | _Sam--: is cypheus running on your server? |
19:30.52 | fugitivo | saftsack: you could use some kind of script with lp, i don't know how hylafax works |
19:31.06 | _Sam-- | cypheus runs on the windows PC and connects to the hylafax server , like POP3 email |
19:31.08 | sthw45ywyw5 | someone help. I am trying to call from Zap to Polycom 301 phone. The polycom phone rings , I pick up, but neiter side can hear the other speak. |
19:31.09 | _Sam-- | same type thing. |
19:31.10 | saftsack | fugitivo: i want to do exact this one ;) |
19:31.16 | jpablo | hey people my asterisk installationg is missing vm-youhaveno.gsm, but asterisk needs it, any idea where can i get it ? |
19:31.31 | fugitivo | jpablo: www.asterisk.org |
19:31.43 | saftsack | _Sam--: do you know the notify script? |
19:31.52 | _Sam-- | saftsack: im sorry, no |
19:32.11 | jpablo | fugitivo, where exactly ? it isn't in the asterisk tarball nor asterisk-sound tarball. |
19:32.16 | dijit0 | does anyone know how to force the caller id in asterisk whether idefisk has set it or not? |
19:32.26 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
19:32.33 | saftsack | do you know another method to execute a command directly after sending a fax? |
19:33.01 | *** join/#asterisk mutil (i=WebChat@i.think.napoleon.dynamiteblows.com) |
19:33.05 | fugitivo | jpablo: what asterisk version? |
19:33.15 | mutil | hey all how goes it |
19:33.26 | jaike | jpablo: try downloading old asterisk-sounds |
19:33.35 | mut | question.. whats the point in qualify? |
19:33.41 | _Sam-- | saftsack: did you check out all the info on hylafax.org? |
19:33.48 | saftsack | yes |
19:33.49 | mut | what does it do other than give me incorrect 'pings' |
19:33.53 | _Sam-- | i think it tells you there how to do it |
19:33.55 | jpablo | fugitivo, i'm running svn 1.2 |
19:33.57 | saftsack | i found a faq for this point but just for incoming |
19:34.04 | _Sam-- | http://www.hylafax.org/content/Automatically_print_incoming_faxes |
19:34.06 | saftsack | but theres no outgoing help |
19:34.14 | _Sam-- | oh, i thought you meant just incoming. |
19:34.16 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
19:34.20 | saftsack | no |
19:34.25 | saftsack | i mean just outgoing ;) |
19:34.42 | saftsack | so i wanted to use the notify script for this |
19:35.05 | _Sam-- | why do you need hard copy of every outgoing fax? |
19:35.13 | _Sam-- | it it all stored and accessible if you need it later |
19:35.20 | _Sam-- | then you can print manually the ones you need |
19:35.34 | saftsack | ok |
19:35.40 | saftsack | but my archive folder is empty |
19:35.45 | saftsack | i can find them in docq |
19:35.52 | fugitivo | that's true |
19:35.52 | _Sam-- | you need to setup a cron script to archive |
19:36.02 | _Sam-- | #Fax Stuff |
19:36.02 | _Sam-- | 00 07 * * * /usr/local/sbin/faxcron -rcv 45 > /dev/null |
19:36.02 | _Sam-- | 15 07 * * * /usr/local/sbin/faxqclean -aA > /dev/null |
19:36.08 | fugitivo | why you use a faxserver if you need to print all faxes? buy a fax machine for that :) |
19:36.30 | saftsack | what is a fax machine? ^^ do you mean the hardware faxes? |
19:37.16 | *** join/#asterisk Seedy (n=Seedy@65.200.153.2) |
19:37.38 | Seedy | Hello fellow phone nerds! |
19:37.47 | fugitivo | saftsack: yes |
19:37.53 | Netgeeks | Hi Seedy, did you get everything figured out last night? |
19:38.21 | Seedy | Netgeeks: Nope! But I'm understanding things more and more. |
19:38.37 | Seedy | And I have a few more basic questions I can't find answers too |
19:39.08 | Netgeeks | well, ask away, someone here is likely to answer in some format |
19:39.19 | mut | so does anyone know what sip qualify does? |
19:40.14 | justinu | it "pings" the UA |
19:40.17 | justinu | with a sip options |
19:41.10 | mut | yea but for what purpose |
19:41.21 | mut | because i can ping it normally and get like.. 30ms replies |
19:41.23 | saftsack | _Sam--: faxcron is just for incoming, right? |
19:41.46 | mut | but the sip 'ping' bounces from 40ms to 300ms to timing out (3000ms) |
19:41.48 | MattH | on asterisk box A I get "Got SIP response 481 "Call Leg Does Not Exist" back from " and on asterisk box B I get " Failed to authenticate user "+15703232166" <sip:+15703232166@63.174.244.172>;tag=as03ed0edb |
19:41.48 | MattH | " even though I have it setup as a peer and user any thoughts? |
19:42.22 | Netgeeks | mut: it isn't an ICMP ping.. it's a SIP ping, meaning it just doesn't test to see if the network stack on your system in running |
19:42.26 | Seedy | My setup is an Asterisk Box that is accepting incoming calls from an external analog phone. I'm trying to figure out if IAX or SIP would work better with this (From what I've read IAX is better, but I don't think it will work with external phone lines) |
19:42.49 | fourcheeze | MattH: I bet there is a user of that name somewhere in your config |
19:42.49 | mut | Netgeeks: yea.. and? |
19:42.53 | [TK]D-Fender | Seedy : either can work just fine |
19:42.59 | Netgeeks | it checks to see that the far end is accepting SIP messages and waits for a response, so the round trip time includes the time it takes for the pinged system to process the sip message and respond |
19:43.16 | mut | so even though a normal ping on the ata gives 30ms |
19:43.18 | Netgeeks | so not only do you check that the system is there and alive, you also check that it's talking SIP and understanding it |
19:43.22 | mut | this sip ping gives 200ms |
19:43.27 | mut | whats that telling me? |
19:43.46 | MattH | fourcheeze, there isn't though that "user" is my callerid |
19:43.47 | mut | sometimes its unreachable alltogether |
19:43.54 | mut | but i can still log into the ata and configure it |
19:44.03 | mut | and it says it's registered, it has dialtone, it can make calls |
19:44.17 | Netgeeks | right, if you can ping an ATA at the exact same time you do a sip options (qualify) message, and the ping time is 30 ms, and the sip round trip is 230ms, then 200 ms was spent by the ata processing the sip message and responding |
19:44.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-170.dslextreme.com) |
19:44.49 | *** join/#asterisk brock05 (n=admin@c220-239-93-31.rochd1.qld.optusnet.com.au) |
19:44.51 | Seedy | So I'm having trouble with receiving DTMF digits. Asterisk doesn't seem to recognize them at all. I am using SIP right now and was wondering if using IAX might fix this problem. Or is that a dumb idea. |
19:44.57 | jaike | mut: try replacing patch cables...happened once with me |
19:45.08 | fourcheeze | MattH: that's strange. That error normally means that either something shold have authenticated which didn't, or shouldn't have which did |
19:45.09 | mut | this happens to a lot of people though.. |
19:46.19 | jpablo | Seedy, there are three ways to send dtmf, both ends must be configured to use the same method |
19:47.36 | Seedy | jpablo: What are these three methods? Inband, rfcxxxx |
19:47.42 | *** join/#asterisk los415 (n=los415@64.201.109.62) |
19:48.04 | [av]bani | Seedy: try changing your dtmf mode, try rfc2833 |
19:48.33 | [av]bani | my guess is you're using inband |
19:48.41 | MattH | fourcheeze, right ... it's odd.. I can't figure it out :) hehe still trying... thinking maybe peer/user statements are messed up on the client end |
19:48.45 | [av]bani | with sip, that's almost always wrong :) |
19:49.22 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
19:49.33 | fourcheeze | MattH: yeah, do you have something as a peer in the sending box? |
19:49.38 | jpablo | Seedy, inbad, rfc?? and sip info |
19:50.00 | Seedy | Thanks guys, I'll try using rfc2833 |
19:50.12 | `lyme | why would a FXS be reporting as unavilable, when i know its not in use? |
19:50.24 | MattH | fourcheeze-away, it's setup as a friend in the setuping box. |
19:50.39 | jpablo | `lyme, they somethings get "stuck", yeah that sucks. |
19:51.22 | `lyme | well, im still in the initial setup phases.. and they ahve never reported themselves as being available. Im baffeled. |
19:52.13 | `lyme | i have a 4 port digium card, only 1 line is plugged in, and the outbound route checks through all those ports, and they all come back unavail, even the one that i know is good... |
19:52.28 | `lyme | is there a manual way to set them as avail? |
19:52.54 | austinnichols101 | what's a good pc for asterisk for a home system? Price isn't as much of a concern as low power, footprint, etc. |
19:53.17 | _Sam-- | austinnichols101: build a mini-itx solid state machine |
19:53.22 | fugitivo | `lyme: /etc/zaptel.conf and /etc/asterisk/zapata.conf ? |
19:53.30 | Seedy | If I am using a router, does that mean I am behind a NAT? |
19:53.44 | *** join/#asterisk flashn253 (i=flashnet@Darkstar.AceShells.com) |
19:53.44 | austinnichols101 | seedy: not necessarily |
19:54.04 | austinnichols101 | seedy: what's the ip address on your local machine? |
19:54.42 | fugitivo | how does a person learn asterisk without knowledge of networking basics? |
19:54.59 | `lyme | fugitivo: they are there, and configured, and will take incomming calls just fine.... its just the outbound :( |
19:55.25 | `lyme | i set the channels with 1 avail connection each, and even tried bumping that to 50, and it still didnt work :( |
19:55.28 | fugitivo | `lyme: pastebin your outbound part of extensions.conf |
19:55.29 | _Sam-- | fugitivo: thats like saying how do you learn physics if you dont know basic math |
19:55.34 | Seedy | austinnichols101: How do you mean? Other computers can access me with 192.168.0.x but my external ip is something else |
19:55.43 | mzo_ | you learn physics by dropping heavy things off tall things. |
19:55.44 | I-MOD | nat |
19:55.52 | fugitivo | lol mzo_ |
19:56.13 | _Sam-- | i guess if you didnt know basic networking, you would learn it really fast when you are trying to setup your asterisk box. |
19:56.18 | austinnichols101 | seedy: go to http://www.whatismyip.com and check your addy. That shows what you look like when you're accessing the internet |
19:56.25 | *** join/#asterisk SocialD (n=SocialD@CPE0040f45b3a28-CM00407b85d7bb.cpe.net.cable.rogers.com) |
19:56.27 | _Sam-- | or at least, you would learn just enough to configure your ip and default routes :) |
19:56.28 | mzo_ | nat is like being a super secret agent who has to carry messages to the free people outside without divulging what you're carrying. :P |
19:56.37 | *** join/#asterisk trek (n=rsi@lns-bzn-35-82-250-207-194.adsl.proxad.net) |
19:56.40 | SocialD | Hey |
19:56.48 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
19:56.50 | austinnichols101 | seedy: so the router is doing the translation between your local (private) address and the public address |
19:57.33 | SocialD | I just installed NetBSD 3: |
19:57.43 | SocialD | And I want to run Asterisk on it |
19:57.45 | fugitivo | SocialD: does zaptel work on that? |
19:57.49 | Seedy | austinnichols101: Yeah... I'm trying to figure out if the NAT is causing my DTMF problems |
19:57.53 | SocialD | Havn't tryed yet |
19:57.57 | brock05 | Hi I have one SIP account that keeps coming up with Bad Auth. All the other SIP accounts are working fine and everything in the account that comes up with bad auth looks ok. Does anyone know what might be doing it ? |
19:58.06 | austinnichols101 | seedy: you're definitely behind nat |
19:58.15 | Err | Seedy: if you're getting a connection, NAT is not the issue :-) |
19:58.29 | Seedy | Err: Thanks, that is what I needed to know |
19:58.38 | fugitivo | Err: you can have a connection, but not audio, and nat could be the problem |
19:58.40 | Err | if the audio and control channels are both working, NAT isn't in the way |
19:58.45 | SocialD | Can you guy's give me a sec, im gona run downstairs to my pentium 1 I wana run Asterisk on.. when I try to do a 'make install' I get a nasty GCC compilation error, 4 minutes be right back |
19:58.47 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
19:59.05 | SocialD | <NetBSD 3> |
19:59.28 | Seedy | So I'm sure that incoming audio is working fine. But outgoing audio I am unsure of. Could this be a nat issue? |
19:59.38 | fugitivo | Seedy: yes |
19:59.42 | Err | yes |
19:59.42 | jaike | one way audio? |
19:59.43 | I-MOD | it could be you need to update your asterisk |
19:59.58 | fugitivo | Seedy: what asterisk version? |
20:00.15 | jaike | lol |
20:00.16 | Seedy | I-MOD: I am using 1.2.3 |
20:00.26 | I-MOD | awww....so much for an easy fix |
20:00.30 | fugitivo | I-MOD: i think the bug was 2-way no audio |
20:00.34 | Seedy | I am not sure if it is one way audio |
20:01.08 | Seedy | The problem is I am not interfacing with another phone, just the server. So I don't know how to test if the server is getting my audio |
20:01.21 | Seedy | I guess I could try and record some audio to a file... |
20:01.29 | I-MOD | Monitor() |
20:01.43 | Err | or Echo() |
20:01.46 | fugitivo | `lyme: are you using asterisk2home? |
20:01.47 | I-MOD | that too |
20:01.52 | fugitivo | asterisk@home |
20:01.53 | `lyme | sadly, yes. |
20:01.57 | fugitivo | well |
20:02.06 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
20:02.07 | fugitivo | ~amp |
20:02.09 | jbot | amp is, like, NOT supported here! people using it should join #amportal |
20:02.09 | Seedy | Err: Echo doesn't work for sure |
20:02.21 | fugitivo | `lyme: trying to debug it is a pain in the ass |
20:02.30 | `lyme | ah... |
20:02.33 | `lyme | thaNKS ANYWAYS |
20:02.51 | Err | heh, if Echo doesn't work, you don't have two-way audio |
20:03.08 | *** join/#asterisk miketaht (n=mtaht@67-127-179-114.ded.pacbell.net) |
20:03.33 | austinnichols101 | aah has echo function: *43 |
20:04.31 | Seedy | Hmmmm.... Ok, I've got enough info to try out some more tests. Thanks again! |
20:06.21 | `lyme | #amportal is a quiet place. LOL |
20:06.38 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
20:08.15 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
20:08.43 | *** part/#asterisk brock05 (n=admin@c220-239-93-31.rochd1.qld.optusnet.com.au) |
20:11.51 | *** join/#asterisk dijit0_ (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net) |
20:13.45 | SocialD | Ok im back ! |
20:13.46 | SocialD | make: stopped in /AsteriskHAX/asterisk-1.2.0-beta1 |
20:14.09 | SocialD | Thats what I get when I do a make install in my NetBSD 3, box |
20:14.23 | iCEBrkr | SocialD: That's what you get for running NetBSD |
20:14.25 | iCEBrkr | :P |
20:14.33 | SocialD | I tried DSL before |
20:15.12 | SocialD | I spent 4 hours trying to get GCC work.. that failed, I got help from people in #damnsmalllinux .. they told me to install g++ |
20:15.12 | iCEBrkr | SocialD: and CentOS, Fedora, or Debian isn't an option? |
20:15.41 | SocialD | My harddrive can only hold 2 gigs |
20:15.54 | iCEBrkr | Minimal install. |
20:15.55 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@62.240.252.22) |
20:15.59 | SocialD | It's a pentium 150MHz |
20:15.59 | [TK]D-Fender | SocialD : Slackware is for yoU!!!! |
20:16.01 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
20:16.08 | [TK]D-Fender | ;) |
20:16.09 | stack_ | in extensions.conf, "include" lets you specify a time for the include. If I have an include for a holiday and an include for normal operations, and the holiday falls on a normal operation day, do both includes get used? |
20:16.15 | SocialD | Oh yeah I got that distro round here somewhere |
20:16.33 | iCEBrkr | SocialD: I'm not even sure if that's enough horse power to do all the codec/transcoding in Asterisk :) |
20:16.36 | fugitivo | SocialD: did you try lfs? (linuxfromscratch) |
20:16.49 | SocialD | I've gotten asterisk on there before |
20:17.01 | SocialD | I just don't know if its gona do anything else then CLI> |
20:17.45 | [TK]D-Fender | Yeah, if you're hard-core Linux, LFS would be perfectly fine choice... |
20:17.46 | stack_ | if you have the time |
20:17.46 | SocialD | Im not that hard core sorry |
20:17.46 | [TK]D-Fender | Slackware is jsut sort of an easy way out. |
20:17.46 | fugitivo | you don't need much time |
20:17.46 | fugitivo | less than gentoo |
20:17.46 | SocialD | Alright i'll go install it, be back in an hour if im still alive |
20:17.49 | fugitivo | because you don't install crap :) |
20:18.42 | [av]bani | http://www.voip-info.org/tiki-index.php?page=AstLinux \o/ ? |
20:19.32 | justnulling | what is this error Auto-congesting call due to slow response? |
20:19.33 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
20:19.50 | *** join/#asterisk pb__ (n=pb@cpc3-cmbg6-5-1-cust33.cmbg.cable.ntl.com) |
20:21.32 | *** join/#asterisk fdgfd (n=fdgfd@adsl-ull-16-220.42-151.net24.it) |
20:21.37 | fdgfd | hello! |
20:22.12 | fdgfd | Someone tried to put asterisk on embedded device such an Access Point? |
20:22.29 | [av]bani | yes |
20:22.55 | iCEBrkr | I'm not sure why you'd want to do that, other than geek factor |
20:22.59 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
20:23.06 | [av]bani | iCEBrkr: solid state pbx |
20:23.08 | _Sam-- | reliability |
20:23.16 | [av]bani | also very cheap |
20:23.16 | _Sam-- | if you are just using a home setup, its fine |
20:23.20 | [av]bani | yep |
20:23.24 | _Sam-- | people put it on the linksys wrt54g |
20:23.28 | fdgfd | yes |
20:23.33 | [av]bani | its good for 4-6 lines, especially if you use reinvite |
20:23.34 | fdgfd | I mean wrt54g |
20:23.50 | iCEBrkr | Yeah cuz 100mhz is OOO so much horse power to run an office on |
20:23.56 | _Sam-- | who said office? |
20:23.59 | fdgfd | I've heard it today |
20:23.59 | _Sam-- | nobody but you |
20:24.02 | [av]bani | iCEBrkr: you dont need ghz to run signaling |
20:24.18 | iCEBrkr | LOL |
20:24.19 | [av]bani | fdgfd: www.openwrt.org |
20:24.20 | dijit0_ | does anyone else hear use idefisk and knows why the caller id will not send right if one isn't specified in idefisk itself? |
20:24.24 | iCEBrkr | You guys crack me up |
20:24.35 | _Sam-- | dijit0_: ask ZOA |
20:24.40 | _Sam-- | er he's not here, he wrote it |
20:25.05 | dijit0_ | argh@! ok, thx |
20:25.12 | _Sam-- | fgd: another fun project would be to build your own solid state asterisk machine |
20:25.15 | fugitivo | what are the "watchers" when doing show hints? |
20:25.15 | fdgfd | avbani: yes, I know it. I would like to know if It can be usefull (for you) to have a network of AP (WRT) with asterisk |
20:25.25 | iCEBrkr | So explain to me something... |
20:25.26 | harryvv | I wonder how much faster a hardware based asterisk solution would be that over software. Say by the time a extention is pressed to the time the phone rings. |
20:25.27 | [av]bani | iCEBrkr: as long as you know the limitations, its fine. no transcoding, use reinvite where possible. works fine |
20:25.27 | _Sam-- | using something like mini-itx or a something |
20:25.28 | *** join/#asterisk batphone (n=will@69.15.174.114) |
20:25.28 | stack_ | in extensions.conf, "include" lets you specify a time for the include. If I have an include for a holiday and an include for normal operations, and the holiday falls on a normal operation day, do both includes get used? |
20:25.39 | [av]bani | _Sam--: i wanna try a gumstix :D |
20:25.44 | fdgfd | wait, I draw :-) |
20:25.47 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
20:25.47 | batphone | anyone in here have luck using IAXYs over a VSAT link for FAX? |
20:25.49 | _Sam-- | ive been looking into some sbc's |
20:25.50 | iCEBrkr | If a 3Ghz machine has issues with timing and such.. How well is a shitty router CPU going to handle that?? |
20:26.04 | [av]bani | iCEBrkr: oh yeah, no meetme :) |
20:26.08 | iCEBrkr | lol |
20:26.09 | harryvv | There is a asterisk firmware based solution on A ZORCOM BOX. |
20:26.10 | iCEBrkr | :( |
20:26.14 | dijit0_ | i am not sure whether my issue is asterisk or idefisk... but when i set a custom callerid in my iax.conf, it has no effect unless a callerid is set in idefisk, which can be any callerid, THEN the one i have in my iax.conf will work |
20:26.21 | [av]bani | iCEBrkr: home answering machine usually doesnt need meetme... |
20:26.23 | _Sam-- | that xorcom box is nothing |
20:26.30 | _Sam-- | my solid state boxes are nicer than that thing |
20:26.31 | harryvv | sam, have you used it? |
20:26.36 | _Sam-- | no i saw the specs last night on it |
20:26.38 | iCEBrkr | [av]bani: oh, you silly guys thing so small |
20:26.42 | [av]bani | iCEBrkr: you might as well ask "well how does cisco do it with a 200mhz mips" |
20:26.49 | [av]bani | the answer: very carefully |
20:26.52 | trixter | isnt the4 xorcom basically a rip of thevoipconnection.coms vs1, including the name |
20:26.58 | iCEBrkr | [av]bani: um, a cisco router doesn't do half the shit asterisk does. |
20:27.01 | _Sam-- | http://www.xorcom.com/ts-1/features.html |
20:27.08 | harryvv | sam, have done performance test on yours? |
20:27.13 | [av]bani | iCEBrkr: for routing sip, it does |
20:27.14 | iCEBrkr | [av]bani: Which is why they got away with 386 CPUs for such a long time |
20:27.34 | fdgfd | please look here : http://www.frascati1.org/asterisk.jpg |
20:27.38 | [av]bani | iCEBrkr: the only idfference is ciscos dont do pbx functions,b ut they route sip the same basically -- they're a gateway |
20:27.42 | _Sam-- | harryvv: no i havent...but for the same price as the xorcom running a via 1gh, i run a celeron 2.8ghz w/ 1g ram |
20:27.49 | [av]bani | iCEBrkr: so they do signaling, transcoding, etc. |
20:27.59 | harryvv | I see |
20:28.07 | iCEBrkr | [av]bani: All I'm saying is I hear about all these time sensitive and quality issues even on a beefy machine, and now you're going to tell me that some PoS home networking router is a solution?? |
20:28.08 | [av]bani | so again, the answer is: 'very carefully'. know your limitations |
20:28.14 | harryvv | sam, and for the storage you use a flash card? |
20:28.19 | [av]bani | iCEBrkr: yep |
20:28.20 | fugitivo | where is the documentation about "watchers"? |
20:28.27 | iCEBrkr | [av]bani: Limitations.. 1 phone call... |
20:28.29 | [av]bani | iCEBrkr: it's _a_ solution, and works very well |
20:28.32 | [av]bani | iCEBrkr: hardly |
20:28.37 | iCEBrkr | [av]bani: Duct tape is a solution... |
20:28.40 | [av]bani | iCEBrkr: read up on "reinvite" |
20:28.45 | iCEBrkr | Doesn't mean it's the RIGHT solution |
20:28.56 | _Sam-- | yes, a 1G cf flsah card runs the main os and asterisk with a GUI and many utilities.../var runs from ram so hardly any writes to the CF card, and i backup with rsync all the critical data to a USB thumb drive |
20:29.01 | [av]bani | and get back to me when you're done reading |
20:29.24 | iCEBrkr | uh |
20:29.25 | _Sam-- | if i used mini itx it wouldnt be much bigger than 4 packs of cigarettes or so |
20:29.26 | iCEBrkr | sure dude. |
20:29.29 | s34n | The Dial command should allow me to dial more than one number/device separated by &, true? |
20:29.30 | harryvv | sam, thats interesting :) how fast does it boot up? |
20:29.42 | znoG | does asterisk support distinctive ring in the SIP protocol? (apparently a certain tag can be set to let ATAs know its a diff ring) |
20:29.42 | _Sam-- | harryvv: its takes about three minutes because it has to load /var into ram |
20:29.49 | harryvv | I see |
20:29.50 | [av]bani | _Sam--: www.gumstix.org :D~~~~~~ |
20:29.56 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
20:29.58 | znoG | never mind |
20:30.00 | fdgfd | It's usefull for you to have WRT with asterisk in this topology?? http://www.frascati1.org/asterisk.jpg |
20:30.01 | harryvv | About the same time as a standars asterisk box |
20:30.07 | [av]bani | znoG: distincitve ring is dependent on the endpoints |
20:30.11 | _Sam-- | its a bit slower for sure |
20:30.11 | Drew___ | mv /me /dev/bed |
20:30.17 | trek | i'am lost between hisax, midsn and zaphfc for hfc cards. What do you advise ? |
20:30.25 | _Sam-- | it takes about 1 minute to load the 120 megs of /var into ram |
20:30.28 | harryvv | sam, and power consumption? |
20:30.31 | _Sam-- | maybe a bit less |
20:30.44 | [av]bani | fdgfd: should be fine, just remember you can't do any transcoding |
20:31.01 | *** join/#asterisk Dusty_ (n=Dusty@64.89.118.139) |
20:31.02 | _Sam-- | it uses maybe 50 watts max, i dont have a way to test it. |
20:31.10 | _Sam-- | but the only moving parts are the chassis fans |
20:31.21 | _Sam-- | so i cant imagine it draws very much |
20:31.23 | iCEBrkr | _Sam--: Sounds like this system called ShoreTel :P |
20:31.27 | harryvv | sam, borrow a Digiital volt meter with amp clamp. |
20:31.45 | _Sam-- | i want to build the next one on a mini itx |
20:31.51 | _Sam-- | ive built three using micro atx |
20:31.51 | batphone | anyone in here have luck using IAXYs over a VSAT link for FAX? |
20:31.53 | [av]bani | _Sam--: gumstix uses like <1W :) |
20:32.01 | [av]bani | you can run it off AA batteries |
20:32.05 | _Sam-- | power consumption wasnt THAT high on my list |
20:32.09 | [av]bani | :D |
20:32.12 | [av]bani | its cool though |
20:32.21 | *** part/#asterisk Dusty_ (n=Dusty@64.89.118.139) |
20:32.24 | fdgfd | avbani: I don't need transcoding but I need "nat" function of asterisk (as an SBC) is UP. So how many call do you think a WRT and a true server can hold (to balance the network) |
20:32.30 | Err | heh, I'm not sure that most people know someone with a multimeter and amp clamp well enough to borrow one |
20:32.42 | _Sam-- | i have a multimeter but know amp clamp |
20:32.42 | [av]bani | fdgfd: probably maximum of 3-4 |
20:32.47 | [av]bani | for wrt |
20:32.49 | _Sam-- | know = no |
20:32.51 | fdgfd | ok |
20:32.52 | [av]bani | for 'true server' probably 100's |
20:32.53 | Err | yes, me too |
20:32.58 | fdgfd | and a 3ghz server? |
20:33.05 | *** join/#asterisk _dusty (n=Dusty@64.89.118.139) |
20:33.05 | [av]bani | 3ghz? 100's probably |
20:33.19 | fdgfd | ok and, sorry, the last question |
20:33.21 | [av]bani | i think someone tried the other day and hit 500 |
20:33.24 | iCEBrkr | _Sam--: Have you looked into Soekris |
20:33.31 | fdgfd | if I want "101" |
20:33.32 | _Sam-- | ice; yeah, the CPU power is too limited |
20:33.35 | [av]bani | iCEBrkr: soekris is rather limited cpu |
20:33.41 | fdgfd | (more than one server :P ) |
20:33.43 | iCEBrkr | Compared to? |
20:33.43 | [av]bani | iCEBrkr: a gumstix is fater! |
20:33.49 | _Sam-- | compared to other small computers |
20:33.49 | iCEBrkr | Really? |
20:33.50 | [av]bani | faster |
20:33.53 | [av]bani | yep |
20:33.56 | fdgfd | I must do something like DNS load balancing? |
20:33.57 | [av]bani | its pretty slow actually |
20:34.01 | iCEBrkr | [av]bani: You called the gumstick fat! LOL |
20:34.07 | fugitivo | god, asterisk even runs on a nokia 770 |
20:34.10 | [av]bani | soekris is mainly intended for APs |
20:34.14 | _Sam-- | there are some cool SBCs |
20:34.18 | fugitivo | and it's a crappy cpu |
20:34.21 | _Sam-- | that could do asterisk in about the size of a pack of smokes |
20:34.26 | _Sam-- | and be at least 1/2 decent |
20:34.31 | fugitivo | (www.nokia.com/770) |
20:34.34 | *** join/#asterisk Simon-_ (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb) |
20:34.55 | harryvv | who here has had the most reliable * system running a year strait? |
20:35.06 | [av]bani | http://www.gumstix.com/spexboards.html |
20:35.17 | _Sam-- | los415:34:36 up 329 days, 3:08, 3 users, load average: 0.77, 0.77, 0.83 |
20:35.20 | iCEBrkr | _Sam--: I wonder if you could run something like that and rsync changes from a 'master' asterisk server. |
20:35.22 | _Sam-- | not quite a year yet |
20:35.35 | fugitivo | _Sam--: show uptime inside the CLI |
20:35.37 | _Sam-- | iCEBrkr: you wouldnt need rsync if you use sql replication and realtime |
20:35.56 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
20:36.09 | iCEBrkr | _Sam--: Wouldn't that be a bit of traffic? |
20:36.10 | *** join/#asterisk saftsack (n=saftsack@p54A7CB78.dip.t-dialin.net) |
20:36.15 | [av]bani | fdgfd: its unlikely given 1.2.3 and such :) |
20:36.16 | malverian[work] | Wow.. the Festival OGI voice models are really nice |
20:36.23 | malverian[work] | Like.. suprisingly good. |
20:36.29 | iCEBrkr | malverian[work]: You found some good voices? |
20:36.30 | _Sam-- | my solid state boxes run sql / realtime fine |
20:36.39 | malverian[work] | iCEBrkr, Yeah. |
20:36.49 | malverian[work] | Listen to the demos here (non-commercial only .. like mbrola) |
20:36.59 | [av]bani | _Sam--: you're running from ram, mainly :) of course they run fine |
20:37.00 | iCEBrkr | _Sam--: What kinda call volume can it handle tho? |
20:37.09 | malverian[work] | iCEBrkr, http://cslu.cse.ogi.edu/demos/ttsdemos.htm |
20:37.43 | fdgfd | <[av]bani>: sorry I don't understand ! :( I mean: If I need more power, asterisk give me something to make 2 server asterisk work as it was one? (scalability) |
20:37.48 | _Sam-- | iCEBrkr: i wouldnt want to route a quad pri card over it, but the ones ive built (all custom) could easily do 2 PRI |
20:38.00 | _Sam-- | it depends on the customer needs |
20:38.16 | [av]bani | fdgfd: not really, asterisk doesnt do clustering yet :)) |
20:38.22 | _Sam-- | but my lowest grade hardware so far has been celeron 2.6ghz / 512ram |
20:38.42 | iCEBrkr | _Sam--: There's a system my friend sells called ShoreTel. They have 'switches' that route calls apparently.. They're all solid state like your setup |
20:38.54 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
20:39.02 | iCEBrkr | _Sam--: I'm just trying to find a way to duplicate that |
20:39.06 | [av]bani | as long as you dont transcode, you shouldnt need much cpu at all. its just routing packets |
20:39.07 | fdgfd | avbani: thank you for your precious help! |
20:39.09 | *** join/#asterisk Insanity5 (n=feaw@ip68-111-5-23.sv.om.cox.net) |
20:39.19 | [av]bani | use reinvite as much as possible and youre not even routing anymore, just signaling |
20:39.53 | iCEBrkr | [av]bani: I thought you coudln't reinvite behind NAT? |
20:40.05 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
20:40.24 | _Sam-- | iCEBrkr: i dont know, but it seems like shoretel is just a pbx type system |
20:40.32 | _Sam-- | im not sure what they do that yo couldnt do with * |
20:40.38 | iCEBrkr | _Sam--: It is, but it's using SIP phones now and it's all VoIP |
20:40.44 | _Sam-- | same here |
20:40.48 | iCEBrkr | Exactly. |
20:41.05 | _Sam-- | i could sell a unit with fxo though |
20:41.08 | _Sam-- | if you wanted one :) |
20:41.13 | iCEBrkr | From what I saw, Asterisk is pretty damn close to what ShoreTel can do.. |
20:41.35 | iCEBrkr | But ShoreTel has a 'master' machine which tells the 'switches' how to work. |
20:42.03 | *** join/#asterisk areski (n=areski@36.Red-83-55-99.dynamicIP.rima-tde.net) |
20:42.09 | _Sam-- | how is that any different than , say, registering your remote asterisks to your main asterisk server? |
20:42.13 | iCEBrkr | I don't know much more about it than that, so it could be doing something really simple like [av]bani keeps saying about just using reinvite and routing |
20:42.18 | justinu | asterisk seems to hit a bandwidth wall of about 15mbit/s when "routing" rtp |
20:42.20 | justinu | no transcoding |
20:42.36 | iCEBrkr | _Sam--: Apparently the main ShoreTel box could go down and the switches still work. |
20:42.36 | harryvv | Who has used a voip whosale provider that has been nearly flawless in performance over the last year? |
20:42.52 | *** join/#asterisk [ToTo] (n=ToTo@host72-146.pool872.interbusiness.it) |
20:43.15 | _Sam-- | in that way, it does seem like using mysql replication would work |
20:43.20 | Insanity5 | How would I record phone calls only to-from an extension? I know how to record all incoming/outoging calls. |
20:43.25 | _Sam-- | so that if the main * went down, the remotes would still have configs |
20:43.34 | iCEBrkr | _Sam--: Yea, that's what I was getting at with the whole rsync idea.. |
20:43.40 | iCEBrkr | hrrrm |
20:43.51 | _Sam-- | i use replication for a bunch of other things...never thought it about it for that |
20:44.04 | iCEBrkr | There ya go |
20:44.25 | _Sam-- | when i get up to 3 million minutes and get on level 3 with servers in 2 places, you will have solved one of my problems :) |
20:44.29 | harryvv | I guess the idea of a ALMOST reliabile whosale voip provider is a myth |
20:44.36 | QbY | anyone know of a good example of asterisk and ser? |
20:44.52 | iCEBrkr | _Sam--: haha cool |
20:45.12 | iCEBrkr | _Sam--: My friend just likes to diss opensource/linux projects.. So I need ammo once in awhile. |
20:45.24 | harryvv | Sam, are you a installer, end user provider or both? |
20:45.31 | _Sam-- | iCEBrkr: what do you do down there in tampa? |
20:45.36 | saftsack | _Sam--, if i want to write a post to the hylafax mailing list |
20:45.37 | justinu | iCEBrkr: what the hell? is he some kind of windoze luser? |
20:45.42 | iCEBrkr | _Sam--: I told him what I'm doing with Asterisk and he's trying to sell me a ShoreTel + SDK solution.. :-/ |
20:45.49 | saftsack | i have just to write a mail to hylafax-users@hylafax.org or? |
20:45.51 | iCEBrkr | justinu: Exactly. |
20:46.01 | iCEBrkr | _Sam--: I'm a code monkey. |
20:46.03 | _Sam-- | harryvv: at this point i am nothing more than a wannabe.....i do some small office installs and resell some service. (voip isnt my main business endeavoer - yet) |
20:46.04 | Insanity5 | iCEBrkr - IT works fine. You pay the $$$, I won't :) |
20:46.05 | justinu | and you call him a friend? yikes |
20:46.33 | iCEBrkr | justinu: He's actually more of a jock-Microsoft loser. |
20:46.40 | justinu | lol, wrong |
20:46.50 | iCEBrkr | justinu: He jumped on the bandwagon way back when.. When he heard computers will make you money |
20:46.50 | Insanity5 | iCEBrkr - Some people say the comemrcial products are bad; I don't pretend to think so. I just say they're an very bad value :) |
20:47.01 | harryvv | iCEBrkr down there boy... I worked at micrsoft |
20:47.03 | justinu | heh |
20:47.12 | iCEBrkr | justinu: So, he's not even a computer geek, he just knows the stuff he learns without any of the technical knowledge of how/why it works that way |
20:47.23 | [TK]D-Fender | GET HIM!!!!!! |
20:47.25 | iCEBrkr | harryvv: sucks to be you |
20:47.52 | harryvv | iCEBrkr yea? It was a good income stream for me then :) but its all a permatemp enviroment. |
20:47.55 | iCEBrkr | justinu: Of course I call him my friend. We drink and throw darts and ride motorcycles!! |
20:47.57 | justinu | yeah - i've known people like that |
20:47.59 | Insanity5 | Ok guys, how do I "Monitor" calls originating from a single extension only? |
20:48.08 | iCEBrkr | harryvv: Hey, whatever pays the bills!! |
20:48.15 | saftsack | _Sam--, i dunno how2 write to the list, sry :( |
20:48.17 | _Sam-- | iCEBrkr: what does he ride? :) |
20:48.19 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
20:48.20 | harryvv | iCEBrkr Thats ALL that I care. |
20:48.23 | iCEBrkr | _Sam--: One guess :P |
20:48.35 | [TK]D-Fender | Insanity5 : Just shove the monitor only on secions of your dialplan limited to that phone. |
20:48.42 | harryvv | To many arogant IT people find out thay will get layed off then feel like a ass. |
20:48.42 | _Sam-- | hmm...if he is a money guy, goes with corporate, non-opensource... |
20:48.49 | _Sam-- | i have to say a honda cbr1000rr |
20:48.52 | iCEBrkr | _Sam--: A Harley... at the moment.. But he's been eye'n a 'Busa |
20:48.52 | iCEBrkr | LOL |
20:48.58 | Netgeeks | I don't know alot about shoretel, but look at http://www.netgeeks.net It is an asterisk based system that provides N+1 node clustering and expansion by adding a box or boxes |
20:49.27 | Insanity5 | [TK]D-Fender - I guess I don't know how to do that. All I have is a [default] with setcallerid and dial. I can monitor there, but it catches everyone. |
20:49.56 | [TK]D-Fender | Insanity5 : You need to make a seperate context or something then, or seperate the functionality with CALLERID checks, etc. |
20:50.00 | iCEBrkr | Netgeeks: Shameless plug |
20:50.01 | harryvv | Heard of a IT manager that had this arogance by saying to the employees at the company"Im a GOD, I am the network administrator" in this one company..he got layed off 3 weeks after starting |
20:50.26 | Insanity5 | [TK]D-Fender - Yikes. I understand how to seperate with incoming (simply put under the incoming extension), but outgoing, where would I start? |
20:50.43 | Insanity5 | harryvv - Arrogant network admins are the worst. Laid off or fired? |
20:50.47 | iCEBrkr | harryvv: Those kind of people typically don't know their ass from a hole in the ground... |
20:50.48 | Netgeeks | iCEBrkr: shameless yes, plug maybe, answer to the guy who was looking for options, definately |
20:50.53 | [TK]D-Fender | Insanity5 : Make a whole new context jsut for that phone and shove the monitor stuff in there. |
20:50.55 | iCEBrkr | Netgeeks: lol |
20:51.20 | [av]bani | http://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html |
20:51.23 | [TK]D-Fender | killall -9 iCEBrkr |
20:51.30 | [TK]D-Fender | :O |
20:51.36 | synthetiq | anyone know how many call set ups per second asterisk can handle? |
20:51.40 | harryvv | Insanity5 in this one case fired. I knew a few years ago I would spend time in the #windows channel of efnet. There were a few of them there. |
20:51.43 | Netgeeks | I'm not ready to just shamelessly plug the system, as it's still in beta testing and the integrated billing system is still a week or two away from beta testing quality |
20:52.06 | Insanity5 | harryvv - I've actually tried to get help in there once wih complex windows problems. What a waste of time. |
20:52.18 | harryvv | Insanity5 did thay put you down? |
20:52.27 | Insanity5 | harryvv - No community support, whatsoever. The best you can hope for it newsgroups |
20:52.34 | harryvv | I know |
20:52.38 | Insanity5 | harryvv - Smart-ass one-off we don't want to help remarks. |
20:52.46 | harryvv | yup |
20:52.48 | harryvv | I know |
20:52.51 | [av]bani | synthetiq: i heard someone the other day saying they had 50/sec or so |
20:52.55 | Insanity5 | harryvv - And it was a very complex issue. filters with file replication services cross forest over a one way trust. |
20:53.01 | Insanity5 | harryvv - Yuck, any way you look at it :) |
20:53.07 | harryvv | I even said I worked at microsoft at that time and was kicked off the #windows channel. |
20:53.12 | synthetiq | but SER can only do 20 |
20:53.29 | synthetiq | and SER has less overhead |
20:53.36 | harryvv | then asterisk? |
20:53.37 | Insanity5 | harryvv - The MVP's in the newgroups are usually pretty helpful though. |
20:53.45 | harryvv | Ahh |
20:54.25 | Insanity5 | [TK]D-Fender - where would I start for outbound filtering? i made the new context, but how to I direct my outgoing calls to go through that, without saying, implementing a long-distance access code like system. |
20:54.35 | harryvv | Insanity5 so you do alot of asterisk support on the side? |
20:54.44 | [TK]D-Fender | what are you filtering for? |
20:54.48 | Insanity5 | harryvv - Nope. No production asterisk on my company :( |
20:54.54 | harryvv | ahh |
20:54.55 | Insanity5 | [TK]D-Fender - origination from a certain SIP device. |
20:54.57 | *** join/#asterisk arcy (n=arcanum@ppp45-adsl-90.ath.forthnet.gr) |
20:54.58 | harryvv | why not! |
20:54.59 | harryvv | :) |
20:55.02 | Insanity5 | harryvv - My home asterisk box, lol |
20:55.08 | harryvv | k |
20:55.13 | Insanity5 | hardwire - they pay too much money for a commercial solutioon. |
20:55.17 | Insanity5 | err, that was for harry |
20:55.23 | [TK]D-Fender | Insanity5 : Well rigth before your dial lines maybe shove in some GOTOIF's based on the callerid of the phone originating the call... |
20:56.15 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
20:56.15 | harryvv | Insanity5 how much does your company pay? |
20:56.25 | Insanity5 | harryvv - I don't even know. |
20:56.33 | *** join/#asterisk infobox (n=dpizarro@libra.infostar.com.pe) |
20:56.42 | Insanity5 | harryvv - I do love the software, and it is something asterisk is missing, big time. |
20:56.45 | Insanity5 | harry - http://www.sipcenter.com/sip.nsf/html/Sponsors+Interactive+Intelligence |
20:56.46 | justinu | one billion dollars |
20:56.55 | *** join/#asterisk MatsK (n=mk@cC30123C5.inet.catch.no) |
20:56.58 | Insanity5 | The outlook integration, transfer, record, conference, etc, from the PC works great. |
20:57.03 | infobox | hello |
20:57.06 | infobox | Please |
20:57.27 | Insanity5 | infobox - hi :) |
20:57.28 | hypnox | hmm, dundi looked-up calls seem to be using GSM codec only, i am not sure where to override this. dundi.conf ? |
20:57.28 | infobox | I have a big problem making a ISDN/FXS gateway |
20:57.42 | Insanity5 | [TK]D-Fender - I guess if then's would work. I can see that getting real messy though :). |
20:57.47 | infobox | hi Insanity |
20:57.56 | harryvv | Insanity5 would have any kind of voip domain name registered by now but 99.9% of any domain i type in...even some strange names are used. |
20:57.57 | Insanity5 | hello :) |
20:58.12 | _Sam-- | bkw_ you around? |
20:58.14 | Insanity5 | hardwire - What do you mean? |
20:58.16 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
20:58.18 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
20:58.19 | kpettit | lo |
20:58.27 | kpettit | Another day, another version of asterisk :) |
20:58.27 | infobox | I am using TDM2406B and TE110P boards |
20:58.38 | hardwire | quit it |
20:58.41 | hardwire | tab tab |
20:58.45 | hardwire | thats all I gotta say |
20:58.47 | *** join/#asterisk sevard (n=kynan@198.174.233.25) |
20:58.54 | harryvv | infobox how has those cards been working out for you |
20:59.08 | fugitivo | infobox: how does the tdm2400 perform? |
20:59.38 | sevard | grr I can't figure this out at all. asteriskcdrdb is freaking empty. |
21:00.06 | infobox | I can not configure them successfully |
21:00.10 | kpettit | Anybody have any experice transfering calls with AMI? |
21:00.24 | infobox | I am using a PIV 2.66 Ghz with 1GB rAM |
21:00.32 | fugitivo | infobox: why not? |
21:01.14 | Insanity5 | harryvv - So, are there any good win32 front ends for asterisk? It should be able to conference, transfer, record, outlook integration with voicemail, etc. And it shouldn't be some clunky Java thing that is a pain to set up :). |
21:01.32 | Insanity5 | harryvv - I kind of doubt that it's possible without a proprietary hook-in to asterisk though :(. |
21:01.37 | znoG | if only there was a "remote" way to reboot an ATA |
21:01.38 | infobox | fugitivo: the zaptel cannot load |
21:01.46 | Netgeeks | there was that .NET app someone wrote.... I foget the name |
21:02.16 | sevard | why the heck is my asteriskcdrdb empty graaaa |
21:02.18 | Netgeeks | IPSwitchBoard |
21:02.24 | Netgeeks | search for it |
21:02.46 | infobox | I put the TDM240X boards on PCI's number 2 and three |
21:03.11 | infobox | I connected them to power connectors |
21:03.18 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
21:03.53 | infobox | mayeb someone has exprience using both models of boards (TDM2406 and TE110P) ain the same pc ??? |
21:05.07 | libila | anyone know what the cause of this is? ZT_CHANCONFIG failed on channel 1: No such device or address (6) I googled and someone moved their PCI card from the top slot (next to agp) to a different one and that error stopped, I tried that but it didn't work. |
21:05.19 | Insanity5 | Netgeeks - That's pretty far on the clunky side last time I poked around with it :P |
21:05.47 | Insanity5 | Netgeeks - Required lots of custom seutp anyways. |
21:06.15 | infobox | has anyone intent to built a ISDN E1/30FXS gateway? |
21:06.16 | Netgeeks | Insanity: I never played with it myself, so I can't say for sure, just remembered it was a windows app that supposably did alot of nice stuff with an asterisk install |
21:07.07 | Insanity5 | Netgeeks - It will, but it's got a little ways to go. It's probably help if asterisk had a method of hooking into it other than SIP for stuff like that. |
21:11.53 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
21:12.11 | QbY | wow.. Asterisk is EVERYWHERE.. |
21:12.28 | Netgeeks | Oh, how I love cascading style sheets..... *puke* |
21:12.40 | QbY | I just talked to a wholesale origination and termination provider.. and they are using Asterisk too.. interesting.. |
21:13.00 | *** join/#asterisk Defraz_ (i=t0tal@tim.mychoice.cc) |
21:13.14 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
21:13.24 | *** join/#asterisk wunderkin- (i=kev@ip68-226-113-228.ph.ph.cox.net) |
21:13.52 | _Sam-- | QbY: who did you talk to |
21:13.55 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-3-0-cust141.bagu.broadband.ntl.com) |
21:14.01 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
21:14.07 | QbY | i3 Networks |
21:14.19 | _Sam-- | havent heard of em yet |
21:14.22 | *** join/#asterisk Little-L_ (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk) |
21:14.46 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net) |
21:15.02 | QbY | they are a Master Reseller and VAR for Level 3 |
21:15.18 | _Sam-- | there are only a few real level 3 partners |
21:15.22 | _Sam-- | like 8x8 and delathree |
21:15.42 | _Sam-- | deltathree rather |
21:15.51 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
21:15.56 | iCEBrkr | _Sam--: Damn you.. Now I wanna build a solidstate asterisk box like you. |
21:16.05 | _Sam-- | iCEBrkr: do it, i dont have any patents...yet :) |
21:16.09 | iCEBrkr | hehe |
21:16.15 | _Sam-- | QbY: they are not listed as an master reseller |
21:16.19 | _Sam-- | http://www.level3.com/userimages/dotcom/microsites/MasterResellers/ |
21:16.29 | iCEBrkr | _Sam--: I wanna build it in a 1U |
21:16.46 | _Sam-- | i found some nice 1u mini itx cases |
21:16.55 | QbY | i3 Networks is listed as a Master |
21:17.07 | _Sam-- | give me a url |
21:17.15 | QbY | the one you just sent me |
21:17.22 | _Sam-- | oh hah! |
21:17.25 | QbY | i loaded it and see i3networks, listed just below Delta Three |
21:17.28 | _Sam-- | guess it must be close to quitting time |
21:17.33 | _Sam-- | you are indeed correct! |
21:17.35 | QbY | hehehe |
21:17.39 | [av]bani | _Sam--: i'm still looking for nano-itx ... |
21:17.53 | _Sam-- | ravenpi: i thought there is one or two boards out |
21:18.27 | _Sam-- | QbY: i called lvel3 yesterday to find out what you need in order to get to resell their service |
21:18.34 | justinu | money |
21:18.35 | _Sam-- | i am only about 2.9 million minutes short |
21:18.37 | justinu | like everything |
21:18.43 | [av]bani | i have yet to find anyone selling any nano-itx boards, only cases(!) |
21:18.51 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-74-138.aubnin.fios.verizon.net) |
21:19.06 | Netgeeks | Sam: 3 million minutes required? |
21:19.08 | _Sam-- | justinu: who else besides l3 provides that type of reseller stuff? |
21:19.10 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
21:19.17 | _Sam-- | netgeeks 3mil for termination, 2.5 mil for orig |
21:19.25 | justinu | i think global crossing |
21:19.25 | _Sam-- | roughly 50K a month a commitment |
21:19.50 | Netgeeks | XO as well |
21:19.57 | QbY | XO doesn't do origination and termination |
21:20.23 | _Sam-- | i think with my current volume i am stuck reselling other 2nd tier services like deltathree and 8x8 |
21:20.27 | justinu | we're pretty stoked with level3's quality so far |
21:20.37 | justinu | sam: try tmccom.com |
21:20.39 | Netgeeks | XO doesn't do orig and term, or XO doesn't do both? |
21:20.48 | QbY | neither |
21:20.54 | QbY | per my last agent meeting |
21:20.57 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
21:21.12 | _Sam-- | ty for the advice |
21:21.45 | Netgeeks | hrm, wonder if they have a hidden product. One of the companies I do consulting work for has a XO contract for origination |
21:22.04 | Martincit0 | Advice pls: Is there any way to run a system call (i.e. perl script) and catch the returning value (from the dialplan)? Thanks |
21:22.05 | znoG | ahh would be sooo neat to send some command to a Sipura/Linksys unit to force reboot |
21:22.26 | Netgeeks | hrm, maybe res_perl |
21:22.37 | Martincit0 | thanks |
21:22.59 | Netgeeks | is that even in stable? res_perl? or just head? |
21:23.02 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
21:23.25 | sevard | What wakeup agi does everyone use? |
21:23.39 | _Sam-- | i use the agi on my cell phone to make some noise and wake me up |
21:23.46 | _Sam-- | its called a clock |
21:23.58 | libila | I'm getting an error when doing ztcfg, could someone take a look at it? http://tinyurl.com/csh3v |
21:24.06 | Assid | is there a way to set a timezone manually for a call? |
21:24.18 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
21:24.18 | _Sam-- | <sorry couldnt help it> you make it seem like everyone is doing wake up calls (maybe they are and i dont know it) |
21:24.27 | *** join/#asterisk PSC (n=chatzill@gateway.imtco.com) |
21:24.29 | sevard | _Sam--: nice. |
21:24.48 | _Sam-- | you saw the wake up scripts under the "Tips and Tricks" section of the wiki? |
21:24.50 | sevard | I do have a working wakeup php script but currently it plays "Music on hold" for the wakeup call, which.. isn't so great. |
21:24.57 | sevard | _Sam--: yes, there are a few. I'm trying them out. |
21:25.46 | znoG | sevard: my wakeup script swears at me in the morning :) |
21:25.53 | _Sam-- | iCEBrkr: when you build your solid state box and want a CF image to run on it, let me know. |
21:26.04 | _Sam-- | i am a bit slacking with it, i still have to get it to kpettit |
21:26.21 | *** part/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net) |
21:26.23 | sevard | i really like another wakeup AIG that the tips and tricks suggests but unfortunatly it doeesn't like to call the user, i watch the call being placed but it doesn't go through |
21:26.39 | _Sam-- | my image has apache, asterisk, mysql 5, x windows w/ fluxbox, mozilla, and some other stuff that i forget right now |
21:26.40 | iCEBrkr | _Sam--: It's not going to be any time soon. I'm in the middle of restoring my bike and well, having too hobbies is a bit expensive :P |
21:26.57 | *** join/#asterisk PSC (n=chatzill@gateway.imtco.com) |
21:27.02 | _Sam-- | but its fully featured, even for development asterisk work |
21:27.08 | iCEBrkr | _Sam--: I'm thinking I won't have a GUI/Desktop. It's going to be more of an appliance |
21:27.37 | _Sam-- | for the extra space required to have a real webbrowser, and that enviroment,...and with how cheap CF is...its well worth it! |
21:27.59 | _Sam-- | my stuff is still an appliance...but if it had a problem..and i needed to go onsite, i would be self contained |
21:28.03 | iCEBrkr | _Sam--: But why? My 'solution' is to have a 'switch' much like a hub or smart switch |
21:28.07 | iCEBrkr | No need for an interface. |
21:28.08 | _Sam-- | with a web browser, and all the other tools i need to fix it |
21:28.18 | iCEBrkr | maybe a web server on there to serve up web pages for configuration stuff |
21:28.21 | [av]bani | _Sam--: x windows? O NOES DONT YOU KNOW X KILLS ASTERISK |
21:28.26 | _Sam-- | no need for an interface, until you need one |
21:28.42 | iCEBrkr | _Sam--: Who needs prety when you have bash? :P |
21:28.43 | _Sam-- | it doesnt startx unless you log in :) |
21:28.47 | iCEBrkr | s/prety/pretty |
21:29.05 | _Sam-- | sometimes you need a web browser or GAIM |
21:29.10 | iCEBrkr | lol |
21:29.18 | kpettit | _Sam--, that's alri8ght man. I'm keeping busy with all these Asterisk updates |
21:29.24 | *** join/#asterisk signaleleven (n=evan@lion.ragga-jungle.com) |
21:29.32 | iCEBrkr | _Sam--: You shouldn't be surfing the web from your Asterisk box!! :) |
21:29.35 | _Sam-- | there is no downside to having those tools available |
21:29.46 | iCEBrkr | _Sam--: lynx works just fine. |
21:29.48 | _Sam-- | i surf the web when im doing development asterisk work all the time |
21:29.51 | dlynes | people actually use X on an asterisk box? :) |
21:29.59 | _Sam-- | lynx doesnt work jsut fine. |
21:30.01 | SocialD | How do I set up asterisk, so it detects what port my modem is on, so I can dial into my house |
21:30.03 | iCEBrkr | lynx + wget. What more do you need? |
21:30.09 | _Sam-- | not anymore, with the complexity of webpages |
21:30.24 | dlynes | sam: links...does frames and javascript |
21:30.30 | kpettit | Just got my gentoo build read for 1.2.4 and zaptel 1.2.3 |
21:30.34 | iCEBrkr | _Sam--: Well, the end result wouldn't have the overhead of a GUI on it.. |
21:30.39 | kpettit | man I love ebuild's |
21:30.45 | _Sam-- | i have no overhead of a gui except disk space |
21:30.47 | synthetiq | sam needs flash and graphics to give you a siezure |
21:30.49 | _Sam-- | which is CF card space |
21:30.53 | [TK]D-Fender | I run X, KDE, Samba, Apache, FTP, and plenty more on my server and it seems to work just fine.... |
21:30.54 | _Sam-- | which is cheap. |
21:30.57 | iCEBrkr | _Sam--: even then, I surf from my work station, and cut-n-paste URLs into wget |
21:31.02 | Netgeeks | ah, i hate gentoo ;0 |
21:31.06 | [av]bani | i usually just take fedora and cut it down to barebones |
21:31.07 | _Sam-- | iCEBrkr: these are devices that deployed at customer locations |
21:31.12 | _Sam-- | when its breaks, and when i show... |
21:31.17 | _Sam-- | i can be fully self contai9nted on my appliance |
21:31.21 | _Sam-- | without needing to use their pcs |
21:31.32 | iCEBrkr | _Sam--: If it breaks at a client location, you take a new one out there and bring the old one back. |
21:31.34 | [av]bani | thinking of trying a cut down ubuntu though |
21:31.35 | fugitivo | [av]bani: isn't easier to lfs insted of cutting down fedora? |
21:31.45 | _Sam-- | if it breaks, i bring a new CF card out there,r eally. |
21:31.50 | _Sam-- | but i can still have tools at my disposal |
21:31.51 | _Sam-- | to fix it |
21:31.54 | iCEBrkr | _Sam--: There ya go :P |
21:31.56 | [av]bani | fugitivo: not really, it takes more work to 'build up' lfs than it does to cut down fedora |
21:32.15 | _Sam-- | there is no downside to having the tools available, if you have the space |
21:32.15 | fugitivo | [av]bani: hmmm |
21:32.16 | iCEBrkr | _Sam--: Thoughts of network devices can't be wrong. |
21:32.16 | _Sam-- | that is my personal opinion |
21:32.20 | [TK]D-Fender | And then we tried Slackware and it was JJUUUUSSSSTTT right! |
21:32.25 | fugitivo | [av]bani: i think it's impossible to cut down fedora :) |
21:32.25 | _Sam-- | i have a 1G card, might as well use the space |
21:32.32 | iCEBrkr | errr Thousands of network devices... |
21:32.37 | iCEBrkr | If I could type, I'd be dangerous |
21:32.48 | _Sam-- | running asterisk is a bit different then running say an 8 port hub. |
21:32.48 | [av]bani | fugitivo: no, i got it trimmed down to <512m |
21:32.55 | [av]bani | still a bit large though |
21:33.07 | [av]bani | i'm thinking ubuntu shold be able to trim down more |
21:33.11 | iCEBrkr | _Sam--: Why tho? Why's it gotta be different? The idea is to plug the shit in and it just works.. |
21:33.15 | tzanger | 512M? |
21:33.18 | tzanger | I have slackware in 52M |
21:33.30 | [av]bani | yeah but ... slackware ... :( |
21:33.30 | _Sam-- | the idea behind a solid state asterisk box is not plug and play |
21:33.30 | signaleleven | when I call Dial from an AGI application if the called party hangs up then I get ANSWER for dialstatus but if the caller hangs up then I get noresponse... how can I get a ANSWER (and more importantly DIALEDTIME) when the caller hangs up? |
21:33.32 | tzanger | [av]bani: slackware is teh win |
21:33.34 | _Sam-- | the idea behind it is plug and run for a while. |
21:33.37 | fugitivo | [av]bani: i have a custom lfs + asterisk + a lot of stuff with <120mb |
21:33.37 | signaleleven | anyone know? |
21:33.39 | [av]bani | tzanger: bring on those floppes! |
21:33.43 | [av]bani | floppies |
21:33.48 | fugitivo | [av]bani: took only one day of work |
21:33.51 | tzanger | [av]bani: oh please |
21:34.02 | [av]bani | tzanger: my first linux was slackware... |
21:34.11 | tzanger | [av]bani: my linux is still slackware |
21:34.11 | [av]bani | downloaded over 14.4k modem |
21:34.16 | fugitivo | i think everybody's first linux was a slackware |
21:34.16 | [av]bani | i moved on :) |
21:34.17 | tzanger | I've tried all the others, slackware just works for me |
21:34.37 | iCEBrkr | I really need to try LFS |
21:34.47 | fugitivo | [av]bani: same here, no packages at all, that was good :) |
21:34.55 | tzanger | lfs was good if you have a specific need in mind (I was making CF-based firewalls) but for a general distro it's got no advantage |
21:35.05 | fugitivo | agreed |
21:35.11 | fugitivo | i use it for my asterisk servers only |
21:35.11 | iCEBrkr | tzanger: Yea, for sure |
21:35.14 | fugitivo | CF |
21:35.23 | [av]bani | theres still use for auto-updating and package management, even on tiny embedded systems |
21:35.27 | [av]bani | (eg, openwrt, ipkg) |
21:35.30 | tzanger | I just use slackware ofr 'em. I can pare it down for what I need, and it just goes |
21:35.30 | *** join/#asterisk clive- (n=pirch@dsl-165-149-246.telkomadsl.co.za) |
21:35.38 | tzanger | for the embedded stuff I'll be rolling my own but that's hardly a distro then :-) |
21:35.38 | [av]bani | actually a mangled openwrt for x86 would be almost idea |
21:35.51 | [av]bani | since you can get openwrt into 4mb :) |
21:35.56 | *** join/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net) |
21:36.15 | [av]bani | and it has a package manger, and nice build system |
21:36.21 | fugitivo | i see (and talking with my experience) harder to cut down a distro than building a linux from scratch |
21:36.24 | *** part/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net) |
21:36.33 | tzanger | I agree with that |
21:36.45 | *** join/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net) |
21:36.47 | _Sam-- | im not sure about that |
21:36.57 | _Sam-- | it was easier for me to start with debian base than from scratch |
21:37.07 | _Sam-- | adn add/remove packages that were needed using synaptic |
21:37.28 | _Sam-- | i started from scratch |
21:37.30 | fugitivo | _Sam--: how many mb did you reach? |
21:37.43 | _Sam-- | how many COULD i have done it in? |
21:37.47 | _Sam-- | or how many am i using? |
21:37.47 | fugitivo | the problem is that a base distro still have a lot of crap |
21:37.57 | iCEBrkr | _Sam--: So you use debian as your model until you figured out exactly what was needed? |
21:38.00 | _Sam-- | CF cards are cheap...i dont care if i have to buy 512 vs. 256 |
21:38.29 | _Sam-- | no, debian is the base. |
21:38.33 | _Sam-- | you can still apt-get stuff |
21:38.38 | iCEBrkr | Ahh |
21:38.51 | fugitivo | but you don't really know what you have in your system |
21:39.06 | _Sam-- | in terms of what? |
21:39.12 | _Sam-- | i know which packages and libraries are installed |
21:41.11 | libila | anyone know what ZT_ChANCONFIG failed on channel 1 means? http://tinyurl.com/csh3v is the complete error. |
21:41.16 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
21:42.03 | signaleleven | anyone know why dial returns noreponse from an AGI app when the caller hangs up but not when the callee does? |
21:42.26 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
21:43.52 | clive- | signaleleven, is that like for astcc ? |
21:44.03 | iCEBrkr | <VOICE TYPE="Cartman"> Screw you guys, I'm going home </VOICE> |
21:44.12 | signaleleven | clive-: no, it's a custom AGI |
21:44.42 | signaleleven | clive-: but similar purpose, if the callee hangs up then everything's cool, if the caller does then I get noresponse for all the Dial vars |
21:44.43 | mzo_ | ii wish that would actually work |
21:44.50 | clive- | signaleleven are you writing it in perl or php ? |
21:44.57 | signaleleven | clive-: perl |
21:45.02 | sevard | i'm editing the crap out of wakeup.php |
21:45.08 | sevard | i just changed the menus so they're more helpful |
21:45.09 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
21:45.27 | sevard | what would be better than 'music on hold' which is generally soft and not very much like WAKE UP, DUDE for a wake up call? |
21:45.38 | fugitivo | sevard: publicity |
21:45.45 | *** join/#asterisk PSC (n=chatzill@gateway.imtco.com) |
21:45.51 | sevard | fugitivo: ..what? |
21:45.54 | signaleleven | clive- and the strange thing is that on the console when the caller hangs up you get a message saying that it exited non-zero |
21:46.20 | clive- | signaleleven, im no expert, but did you try "deadagi" |
21:46.35 | fugitivo | sevard: promotions, advertising, etc |
21:46.48 | signaleleven | clive-: no, I saw reference to it but haven't played with it yet... I'll give it a try. |
21:46.55 | sevard | fugitivo: would you be very happy if you got an advertisement on your wakeup call? |
21:47.02 | clive- | I think that may solve it for you |
21:47.13 | fugitivo | sevard: no, but if I run the hotel i will |
21:47.28 | AgilixSupport | Ok ... newbie question: what is VoicePet? Is it just an old .tar.gz of asterisk with some install scripts? Anything unique? |
21:47.41 | signaleleven | clive-: indeed you're right... thanks :) |
21:48.02 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
21:48.13 | [av]bani | [TK]D-Fender: around? |
21:48.20 | sevard | fugitivo: I guess a custom mp3 would be the best. This is your schedualed wake up call. After you've gotten ready for the day why not try our breakfast menu in the lobby. The weekly specials are only $4.00 and are better than your sister. |
21:48.35 | justinu | 4 dollars? that can't be a hotel |
21:48.38 | justinu | it'd hae to be 40 |
21:48.39 | sevard | heh |
21:48.44 | fugitivo | sevard: that's great |
21:49.17 | fugitivo | sevard: with feng shui music the hotel will increment sales |
21:49.20 | sevard | I wouldn't be exactly sure on how to play a custom mp3, what would I change this to |
21:49.21 | sevard | $parm_application = 'MusicOnHold'; |
21:49.42 | sevard | $parm_application = 'mpg123 /path/to/file.mp3'; ? |
21:50.17 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
21:52.20 | [TK]D-Fender | [av]bani : yup |
21:52.27 | [av]bani | [TK]D-Fender: you have a sangoma 104d right? |
21:52.31 | [TK]D-Fender | yup |
21:52.42 | [av]bani | with the hardware EC? |
21:52.45 | [TK]D-Fender | yup |
21:52.50 | [av]bani | how is the EC |
21:52.57 | [TK]D-Fender | <- man of many words... or was it 1 word many times...? |
21:53.12 | [TK]D-Fender | [av]bani : *0* echo. period. Ever. |
21:53.17 | [av]bani | yay? |
21:53.20 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
21:53.20 | [TK]D-Fender | nigh-Godly. |
21:53.33 | [av]bani | so the a200 should be good since it uses the same EC |
21:53.49 | [av]bani | of course, just because it works good on digital doesnt mean it will work well (or at all) on analogue :) |
21:54.05 | [TK]D-Fender | YUP! A200 = TMD400/2400 killer in vast majority of scenarios IMO. |
21:54.29 | *** join/#asterisk KrIS83 (n=kris@p549B2226.dip0.t-ipconnect.de) |
21:54.38 | [TK]D-Fender | [av]bani : The A200 is build on their AFT card and uses the same DSP. Basically its like an A104d + Channel bank all in one |
21:54.42 | [av]bani | a104d: G.168-2002 EC with 128ms tail |
21:54.48 | [av]bani | spa-3000: G.165 EC with 8ms tail |
21:54.50 | [av]bani | heh... |
21:55.00 | [TK]D-Fender | Thats G.168-2002 at 100% on all densities... |
21:55.07 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
21:55.09 | [av]bani | yes... 'godly ec' |
21:55.18 | [TK]D-Fender | Look at the degrade rate on the TDM2400.... |
21:55.21 | [av]bani | how much did the 104d cost? |
21:56.01 | [TK]D-Fender | Me? *0*. My reseller burned some of his rep on our troubles with 2 x TE405P's which we ditched for the A104d very soon after its release. |
21:56.04 | _Sam-- | sorry to chime in...is the 104d a 4 port FXO? |
21:56.13 | [av]bani | _Sam--: 4 port PRI |
21:56.15 | [TK]D-Fender | _Sam-- : No, a 4 Port T1 |
21:56.19 | _Sam-- | damn i see |
21:56.30 | [TK]D-Fender | _Sam-- : You'd be thinking of their new A200 series |
21:56.38 | [TK]D-Fender | 2-24 port |
21:56.43 | [av]bani | [TK]D-Fender: not his fault though! i bet he doesnt sell te405p's anymore though :) |
21:56.58 | [TK]D-Fender | [av]bani : Actually yes so far.... |
21:57.06 | [av]bani | big mistake :) |
21:57.10 | [TK]D-Fender | We just had so many problems with them that I don't care where they end up :) |
21:57.43 | [TK]D-Fender | Best part is I'm only really using 1 port on it :) |
21:58.04 | [av]bani | :) |
21:58.16 | [av]bani | sangoma doesnt seem to have any EC cards < 4 ports |
21:58.28 | [av]bani | a 2 port EC T1 would be nice |
21:59.12 | [TK]D-Fender | [av]bani : coming out within 3 months from what I'd been told on the inside... |
21:59.19 | [TK]D-Fender | :) |
21:59.28 | libila | is it bad to have wctdm as number 22 on the list of interrupts? (cat /proc/interrupts) |
21:59.35 | [av]bani | they should just sell a generic EC card so people with digiums can use them |
21:59.54 | _Sam-- | i never realize echo was a big problem on PRIs? |
22:00.08 | [av]bani | _Sam--: usually isnt, because PRI cards usually have EC on them |
22:00.23 | [TK]D-Fender | [av]bani : I has to be tightly integrated to be effective. Extra latency would be problematic. Their design is fairly elegent and its YGWYPF |
22:00.26 | _Sam-- | i had a PRI here with digium no EC card...never had echo problems |
22:00.41 | [av]bani | _Sam--: you're lucky then |
22:00.46 | _Sam-- | but most PRIs EC is recommended? |
22:00.51 | [av]bani | yes |
22:00.59 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
22:01.01 | clive- | did everyone forget that no EC cards for asterisk existed until like a few months ago |
22:01.03 | tzanger | _Sam--: no, EC is recommeneded for any VOIP application. |
22:01.04 | [TK]D-Fender | _Sam-- : Always. Anything that his the PSTN should have EC |
22:01.24 | _Sam-- | what does the EC within the phones accomplish? |
22:01.25 | tzanger | personally I really love the MG2 canceller but on large systems you need a lot of horsepower to run EC on all the channels |
22:01.36 | *** join/#asterisk RoyK (n=roy@193.80-202-93.nextgentel.com) |
22:01.47 | tzanger | _Sam--: VOIP is inherently (much) more latent than circuit-switched networks |
22:01.52 | tzanger | and echo is caused by latency |
22:02.06 | h3x | nah |
22:02.13 | h3x | echo is caused by the analog crap on the other end |
22:02.13 | h3x | heh |
22:02.15 | _Sam-- | i am 100% voip, and i dont have many echo problems |
22:02.39 | h3x | _Sam-- i had a PRI here with digium no EC card...never had echo problems |
22:02.40 | [av]bani | you should have _zero_ |
22:02.42 | h3x | _Sam-- i am 100% voip, and i dont have many echo problems |
22:02.43 | _Sam-- | i wouldnt go so far as to say I NEVER have echo problems, but they are really rare |
22:02.46 | h3x | that aint 100% voip? |
22:02.48 | rajiv|work | what can you do if you do not have enough side tone on a sip phone? |
22:03.04 | [av]bani | _Sam--: a proper setup should have zero |
22:03.15 | [av]bani | rajiv|work: yell at the vendor |
22:03.30 | _Sam-- | h3x: 100% voip would be (in my own opinion) if i didnt use the digium card / PRI and routed all my traffic to a remote gateway |
22:03.36 | _Sam-- | that is what i do now, and what i call 100% voip |
22:03.52 | h3x | ok well your remote gateway turns it into a pstn line |
22:04.01 | h3x | 100% voip would be a voip phone to a voip phone |
22:04.34 | rajiv|work | i need new phones |
22:05.04 | _Sam-- | bani, so what does the EC in the phones do? |
22:05.43 | mzo_ | it stops errors between the user and the phone. |
22:05.58 | tzanger | h3x: well yes, but analog will exist on any system since your mouth and ears aren't modular. |
22:05.58 | [av]bani | _Sam--: kills echo from local (handset, speakerphone) |
22:06.22 | h3x | i have digital ears!!! |
22:06.31 | tzanger | [av]bani: a "proper" setup can still have hideous echo |
22:06.37 | tzanger | hence the insidiousness of the situation |
22:07.10 | h3x | it aint the analog speaker/mic that causes echo (usually) |
22:07.16 | tzanger | PSTN hopoff requires echo cancellation to be 100% effective. You can get lucky, and most people do, but without echo cancellation you're screwed |
22:07.28 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
22:07.36 | _Sam-- | how would you do echo canc if you hand off to a remote gatway? |
22:07.40 | h3x | its the... what do they call it |
22:07.42 | h3x | talk bridge? |
22:07.51 | tzanger | h3x: well that's exactly what echo is created from (the mic picking up the speaker, or the hybrid reflecting too much energy back) but what makes it NOTICEABLE is the latency |
22:08.02 | h3x | the analog shit in a analog phone that mixes your own voice into the speaker |
22:08.08 | h3x | yes!!! |
22:08.10 | h3x | HYBRID! |
22:08.10 | tzanger | that's why a telephone call can sound perfectly fine but use an X100P or other FXO module and suddenly it's hideously echoey |
22:08.35 | tzanger | PRI and digital (VOIP) circuitry does not CAUSE echo. but it doesn't mean you won't have it |
22:08.36 | bkw__ | fadsf |
22:08.50 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
22:08.52 | tzanger | bkw__: I agree |
22:09.07 | tzanger | bkw__: but how are we going to get congress to approve? |
22:09.46 | s34n | do changes in voicemail.conf require some action to become effective? |
22:09.46 | _Sam-- | hey bkw...asterlink doesnt do local numbers for origination? |
22:09.58 | _Sam-- | the only things i saw were 8** |
22:10.18 | [av]bani | _Sam--: EC needs to be done at the point where PSTN handoff occurs, so with a remote gateway the remote gateway would do the EC |
22:10.32 | [av]bani | _Sam--: EC is done at endpoints. origination and termination |
22:10.55 | [av]bani | _Sam--: so, your voip phone would have EC and the PSTN gw would have EC |
22:10.56 | _Sam-- | i see...thank you for explaining...maybe that is why my calls are ok, maybe teliax actually has EC (doubt it) |
22:11.01 | [av]bani | teliax has EC |
22:11.08 | [av]bani | a reasonable one too |
22:11.16 | [av]bani | junction networks has a less good one |
22:11.32 | tzanger | it's good to echo cancel on the PSTN side too instead of in software |
22:11.44 | tzanger | because a lot of the delay occurs at the PCI level |
22:11.54 | mut | is MARK2 & aggresive better than kb1? |
22:12.00 | tzanger | so if you can kill the echo before it hits the PCI bus, that means the PCI latency isn't going to make what is left all that bad |
22:12.06 | tzanger | mut: I hate agressive |
22:12.16 | _Sam-- | echo canc all works on like sine waves and stuff? |
22:12.21 | tzanger | mut: I have found that MG2 is better than KB1 which it enhances, which in turn is an enhanced MARK2 |
22:12.35 | _Sam-- | im just curious what it does / how it works in a really small nutshell |
22:12.45 | _Sam-- | it like does phase shifts or something? |
22:12.46 | mut | i tried MG2 |
22:12.51 | mut | but it just didn't get rid of the echo |
22:12.54 | mut | i dunno what to do about it |
22:13.14 | [av]bani | _Sam--: EC is an art :) |
22:13.47 | [av]bani | _Sam--: it keeps a buffer of outgoing audio, and looks for the returning echo, and applies an out-of-phase signal to cancel it out. |
22:13.49 | [Atlas] | Cheap PoE switches/hubs? |
22:13.53 | *** part/#asterisk jaike (n=a@203.131.137.76) |
22:14.03 | [av]bani | _Sam--: but it's hard because the delay can vary, and the level of the returning signal can vary |
22:14.06 | _Sam-- | hmmm so it is basically a phase shift? |
22:14.10 | h3x | !PoE |
22:14.11 | h3x | er |
22:14.13 | h3x | ~PoE |
22:14.14 | jbot | [poe] Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt |
22:14.17 | [av]bani | sorta |
22:14.18 | h3x | dammit |
22:14.20 | h3x | what the hell |
22:14.32 | h3x | jbot is from #perl !? hahahah |
22:14.35 | _Sam-- | i see the concept |
22:14.39 | [Atlas] | LOL |
22:14.51 | h3x | i remember jbot |
22:15.12 | _Sam-- | what frequency does the EC sample at? |
22:15.30 | h3x | its a time shift not a frequency |
22:15.46 | [Atlas] | ~sub |
22:15.56 | h3x | ~CPAN |
22:15.57 | jbot | methinks cpan is the Comprehensive Perl Archive Network; the store of all Perl modules, or 'The World', or reasonably evil, or 'Reasonably Evil', or "perl -MCPAN -e shell" |
22:15.57 | _Sam-- | but to know if its out of line (echoing) it seems it would need to sample to know how far skewed it is |
22:16.19 | jlewis | I've got something weird going on with 2 customer cvs-stable boxes |
22:16.20 | h3x | this is gonna be funny |
22:16.22 | h3x | ~python |
22:16.23 | jbot | it has been said that python is Available at http://www.python.org Python is an interpreted, interactive, object-oriented programming language. It is often compared to Tcl, Perl, Scheme or Java. |
22:16.27 | [av]bani | _Sam--: it keeps a buffer of the audio it sends out... so it already has it sampled |
22:16.33 | _Sam-- | i see isee |
22:16.36 | jlewis | about the same time this afternoon, both started doing chan_zap.c:8015 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
22:16.55 | [Atlas] | ahh lame perl bot didnt bash python! |
22:17.05 | h3x | yeah |
22:17.13 | h3x | maybe coz one of the CREATORS of perl writes python! hahaha |
22:17.17 | jlewis | interesting note...this error says nothing about which PRI its talking about...one of the servers has 2 |
22:17.20 | [Atlas] | ~wheresthegooldoldcoderflamewars |
22:17.34 | h3x | jlewis: Uhm, do you have the asterisk boxes connected back to back? |
22:17.50 | sevard | alright, if anyone knows php are they able to toss me some quick help? |
22:18.13 | sevard | in wakeup.php i changed $parm_application = 'MusicOnHold'; to $parm_application = 'MP3Player(/var/lib/asterisk/Norah_Jones-Come_Away_With_Me-Dont_Know_Why.mp3)'; |
22:18.16 | sevard | and it just doesn't work. |
22:18.20 | [Atlas] | sevard: depending on the value of the know variable i might be able to |
22:18.20 | jlewis | no...one connects to some kind of call center PBX, the other to a cisco voip router |
22:18.29 | jlewis | both have worked for months |
22:18.29 | RoyK | hej |
22:18.42 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
22:18.57 | sevard | i know the file will play because if i add this to extensions.conf it works exten => 557,1,MP3Player(/var/lib/asterisk/Norah_Jones-Come_Away_With_Me-Dont_Know_Why.mp3) |
22:19.13 | _Sam-- | im telling NOrah Jones |
22:19.20 | sevard | it's legal! |
22:19.26 | _Sam-- | lol ok ! |
22:19.38 | sevard | and just for testing :\ it's the only mp3 i had available :P |
22:19.46 | _Sam-- | you know i am just kidding. |
22:19.49 | sevard | i do know |
22:19.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
22:19.59 | sevard | but damnit, i can't figure out why it's not playing |
22:20.30 | _Sam-- | did you check /var/log/asterisk/my_agi.log |
22:20.31 | [Atlas] | Asterisk does not care for norah,, try some nofx |
22:20.38 | [Atlas] | :D |
22:21.05 | sevard | i don't have a /var/log/asterisk/my_agi.log |
22:21.09 | _Sam-- | i guess that could be specific to me only...sinc ei think i tell mine to log |
22:21.34 | _Sam-- | $stdlog = fopen("/var/log/asterisk/my_agi.log", "a"); |
22:21.44 | sevard | you're putting that where? |
22:21.51 | _Sam-- | $in = fopen("php://stdin","r"); |
22:21.51 | _Sam-- | $stdout = fopen('php://stdout', 'w'); |
22:21.51 | _Sam-- | $stdlog = fopen("/var/log/asterisk/my_agi.log", "a"); |
22:21.57 | RoyK | ~pb |
22:21.58 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
22:22.04 | _Sam-- | bah 3 lines, no pb |
22:22.07 | sevard | heh |
22:22.18 | sevard | _Sam--: still confused on where that's going |
22:22.24 | harryvv | sam perl? |
22:22.29 | _Sam-- | that is the top of my php |
22:22.33 | sevard | putting. |
22:22.54 | harryvv | sam, do you know what that same command is in c? |
22:23.00 | _Sam-- | sevard: sending PM |
22:23.12 | harryvv | function i mean :) |
22:23.22 | _Sam-- | to log your agi? |
22:24.10 | [av]bani | _Sam--: how many gxp2k's you got? |
22:24.19 | _Sam-- | between my place and the offices i setup, maybe 50 |
22:24.23 | [av]bani | :O |
22:25.31 | _Sam-- | its the only hard phone ive ever seen or worked with :/ |
22:25.35 | [av]bani | :/ |
22:25.42 | _Sam-- | makes selling it easy |
22:25.48 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
22:26.19 | _Sam-- | the last place i brought brochures for a bunch of phones, but mainly the sales im getting are all really price sensitive |
22:26.26 | _Sam-- | that is why they are calling me to begin with... |
22:26.40 | _Sam-- | so when i show them a $150 phone or an $100 phone...they are always picking the cheapest |
22:26.50 | _Sam-- | i brought brochures and tried to show them nicer phones |
22:26.52 | _Sam-- | but they all want cheap |
22:27.07 | [av]bani | gxp is certainly cheap |
22:27.29 | mut | soo |
22:27.35 | mut | show them really expensive |
22:27.37 | mut | then expensive |
22:27.42 | _Sam-- | hah that is not a bad idea |
22:27.44 | trixter | the gxp2000 is supposed to get updatable lcd scrollys soon so you can have arbitrary messages scroll on them |
22:27.56 | trixter | the beta firmware isnt that bad either, added some features that should have been there all along |
22:28.05 | mut | like "Get back to work you slacker" |
22:28.08 | mut | scrolling all day? |
22:28.11 | _Sam-- | but my problem is if i price the whole setup (including phones) out of their reach or come in with a package that is too much dollars, then i get no sale |
22:28.18 | [av]bani | trixter: lcd scrollys? |
22:28.33 | RoyK | oh well... Iran is making WMDs. Bush is out there again :) |
22:28.37 | trixter | arbitrary messages that you can send to the hpnoes to display on the lcd |
22:28.43 | [av]bani | wheres that from? |
22:28.53 | _Sam-- | i hope bush gets some military action in iran so my oil options go through the roof |
22:28.58 | _Sam-- | just what i was banking on |
22:29.03 | trixter | I spoke to a contact at grandstream today |
22:29.08 | _Sam-- | and knowing bush, he will, because he has oil options too :) |
22:29.29 | trixter | and knowing the democrats they will vote for it becuase they too have oil options |
22:29.34 | trixter | remember john kerry has far more money than bush |
22:29.44 | trixter | something about heinz ketchup |
22:29.48 | trixter | $700M right there |
22:29.54 | rob0 | oh yeah That's his wife |
22:30.03 | trixter | yeah as in they are married thus its johns too |
22:30.07 | [av]bani | the bush family is a mega conglomerate, not just gwb |
22:30.25 | trixter | odd that the beureau of prisons uses heinz isnt it? |
22:30.25 | cpm | It doesn't matter, The fix is in. Mark Warner, 2008. Done deal. enjoy it. |
22:30.28 | trixter | captive audience |
22:30.32 | [av]bani | the whole family has huge oil corporations |
22:30.41 | justinu | hey, i like heinz |
22:30.45 | trixter | many democrats do too |
22:30.48 | trixter | that is my point |
22:30.50 | trixter | its not one sided |
22:31.03 | _Sam-- | i am the furthest thing from a bush supporter...but ive found the way so if you cant beat em join em...and oil has paid off handsomely |
22:31.04 | rob0 | I don't think there is anyone at the national political level in *any* major country who has any honor left. |
22:31.12 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
22:31.26 | cpm | what rob0 says |
22:31.51 | cpm | throw all the bums out. All of 'em (except Sen Byrd, he's my hero) |
22:32.02 | _Sam-- | until people have a means to initiate real change themselves, then politics will always be a sham, like organized religion |
22:32.04 | mzo_ | hah, the cia office of clandestine services was recrutiing on campus today. |
22:32.15 | justinu | byrd is great! |
22:32.18 | cpm | cool, did you get an interview? |
22:32.20 | _Sam-- | <sorry to all organized religion zealots> |
22:32.30 | cpm | No offense taken :) |
22:32.32 | [av]bani | _Sam--: heretic! |
22:32.48 | [Atlas] | <-- scarred for life ^_^ |
22:33.11 | _Sam-- | but because the common person's voice is still never heard in politics, how will that system ever work? people instead of getting more into politics and trying to effect change now just bury their heads in the sand because they know its useless |
22:33.25 | Seldon1975 | Bush's atrocity of a government has made Americans hated all over the world |
22:33.28 | *** join/#asterisk Zodiacal (n=hehehe@bdsl.66.14.242.199.gte.net) |
22:33.44 | Psykick | you guys know what I believe ...... |
22:33.50 | Psykick | we'd all be better off without money |
22:33.50 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
22:34.06 | [Atlas] | wow im quite the outcast here :D |
22:34.07 | _Sam-- | Psykick: come work for me? :) |
22:34.08 | [av]bani | Psykick: give me yours. problem solved. |
22:34.10 | rob0 | Psykick: I've tried that. In fact I am trying that as we speak. :) |
22:34.29 | mzo_ | I vote for whatever party gets me lots of sex and gives me a free macintosh, and working software. :P |
22:34.47 | Psykick | it helps having it ..... but if we really want to effect a positive change for all ..... money must not be the sole point of living |
22:35.24 | [Atlas] | money and mans ego are the fountainhead of human progres |
22:35.28 | _Sam-- | i dont see money as the issue...some people work hard for it and get it ...and some people are lazy bums and drink 40s of milwaukees best all day and dont get any. |
22:35.49 | [av]bani | _Sam-- sits and IRCs all day |
22:35.50 | [Atlas] | progress* |
22:35.51 | Seldon1975 | democracy is flawed |
22:35.53 | _Sam-- | i really do see people who work hard getting financially rewarded though |
22:35.54 | Psykick | poverty is an induced form for lack of money |
22:36.15 | [Atlas] | laziness and lack of education induces laziness |
22:36.18 | _Sam-- | and i dont think that is wrong, it gives people motivation to work harder |
22:36.19 | Psykick | democracy isn't flawed .... it's the people that are supposed to cause effect from that democracy |
22:36.52 | Psykick | and what usually causes no effect in the areas where change is needed most is money |
22:36.56 | Seldon1975 | Psykick: that's just it; democracy is inextricable from the flaws of everyday people |
22:37.03 | _Sam-- | i think instead of having elected officials that the public in this day and age should able to vote outright on any issues instead of having a "representative" of the public vote on our behalf |
22:37.14 | [av]bani | corporation = person, thats the major flaw |
22:37.15 | [Atlas] | money doesnt do that -- lack of moral fiber does that |
22:37.22 | Psykick | to a degree Sam .... |
22:37.25 | Seldon1975 | bani: true dat |
22:37.40 | Psykick | the biggest problem is that not all people share the same opinion or views |
22:37.55 | [av]bani | Psykick: yeah, it would be far simpler if everyone just STFU and obeyed |
22:37.55 | sevard | BRASS MONKEY |
22:37.59 | sevard | YEAHG!!! |
22:38.02 | _Sam-- | lol |
22:38.04 | sevard | that makes me happy when things work |
22:38.23 | Psykick | and in an extreme case ..... the wrong changes can be invoked |
22:38.29 | [Atlas] | http://www.atlasshrugged.tv/speech.htm ---- an interesting not often thought of Point of view for money |
22:38.40 | Seldon1975 | Psykick: the problem is that the average man is too short-sighted to see whats good for him |
22:38.51 | Psykick | too true seldon |
22:39.05 | [Atlas] | Seldon1975: that is a very dangerous path of logic |
22:39.06 | [av]bani | whatever happened to #asterisk ? |
22:39.19 | _Sam-- | Seldon1975: what makes you think you know whats better for the average man than he does? |
22:39.24 | _Sam-- | that is what politicians think |
22:39.25 | Seldon1975 | [Atlas]: perhaps, but it can't be denied |
22:39.29 | _Sam-- | "i know whats good for america" |
22:39.37 | [av]bani | #asterisk renamed to #politics :< |
22:39.37 | [Atlas] | that tends to lead to lack of individual freedom because some appointed person "knows better" |
22:39.38 | Psykick | I believe alot of what is wrong with this world is that we allow ourselves to be influenced .... many of us say that we aren't but .... if you look at it from all viewpoints .... we are |
22:40.03 | Seldon1975 | _Sam--: I am not proposing myself as a solution! |
22:40.06 | [av]bani | oh well someone wake me up when the pseudophilosophy is over |
22:40.12 | _Sam-- | lol |
22:40.23 | _Sam-- | only so much SIP you can speak before it turns to something else :) |
22:40.47 | Psykick | prejudice ... hate crimes ... all because of influence |
22:40.48 | [Atlas] | money is the only means man can deal with another for trade without including force |
22:40.50 | _Sam-- | we've had motorcycles, SIP, and politics today...a broad array of topics. |
22:40.58 | Seldon1975 | we're too enamoured with Western science/philosophy |
22:41.04 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
22:41.09 | Zodiacal | anyone know of a way to speed up dialing with my digum tdm400p? it seems like it takes about 5 seconds from the time Dial() is issued to the time my trunk acctualy gets opened and starts ringing.. |
22:41.24 | cpm | Seldon1975, or not enamoured enough. You pick. |
22:41.39 | rpm | what variable is the current date stored in? ${DATE} ? |
22:41.39 | Seldon1975 | cpm: I pick 'too mcuh' |
22:41.44 | [Atlas] | Seldon1975: there is no honor in being the poor of ethiopia or romania |
22:41.47 | Seldon1975 | much* |
22:41.48 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
22:41.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net) |
22:41.54 | [av]bani | hmm someone should make RJ11 EC's |
22:42.10 | Seldon1975 | [Atlas]: irrelevent comment |
22:42.12 | Psykick | I believe there are groups or organizations of people who influence the world for the worse |
22:42.17 | Psykick | in order to get what they want |
22:42.25 | _Sam-- | they are called terrorist |
22:42.26 | _Sam-- | s |
22:42.39 | [Atlas] | Seldon1975: no it wasnt lol it is western culture that make our "poor" the richest people in the world |
22:43.08 | Psykick | sam: what if those terrorists were paid to do what they're doing in order to distract us from a much more hideous problem |
22:43.11 | [Atlas] | i would so much rather live here and be poor with 2 microwaves a tv and 3 dvd players than not be able to feed my family somewhere else |
22:43.15 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
22:43.23 | [Atlas] | oh god i have to stop |
22:43.25 | [Atlas] | sorry guys |
22:43.29 | Seldon1975 | [Atlas]: still irrelevent - is being 'rich' by material standards what's best for us? look at society, look at whats happening to the planet |
22:43.51 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net) |
22:43.56 | Psykick | I agree seldon |
22:43.59 | cpm | Seldon1975, good essay for you to read someday: http://www.michaelmoser.org/books/book_engine.htm |
22:44.09 | Seldon1975 | this is the hottest year in the last hundred |
22:44.19 | _Sam-- | im not sure how us being any less 'wealthy' would make us a 'better' society in the USA |
22:44.27 | _Sam-- | would we be more compassionate? |
22:44.36 | Seldon1975 | _Sam--: geez I hope so |
22:44.52 | Psykick | sam: you look at those that don't |
22:45.00 | cpm | compassionate hasn't hurt a lot, more would be a nice thing to try for a change. |
22:45.03 | *** part/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co) |
22:45.11 | Psykick | sam: do you see them killing each other ... for money ... for a better way of living |
22:45.16 | *** join/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co) |
22:45.29 | Seldon1975 | _Sam--: you could hardly be less compassionate |
22:45.43 | cpm | <PROTECTED> |
22:45.44 | [Atlas] | Seldon1975: watching my family starve becuase there is no way for me to get ahead,, watching my children murdered and my wife raped at the hands of a brutal govnmt is alot less attractive for me |
22:45.49 | Psykick | I believe that if they had money they would only use as much is necessary to survive |
22:46.06 | _Sam-- | im the least compassionate person you could ever meet...but i would argue money has nothing to do with it...its because ive worked hard and im not very compassionate for people who havent worked as hard. |
22:46.14 | [Atlas] | and as far as killing eachother for money... capitalist societies are by far the least violent,, that is a fact |
22:46.30 | Psykick | sam: compassion is not something that is influenced by money |
22:46.33 | Seldon1975 | [Atlas] erm... the alternative to squandering the worlds resources isnt starving! |
22:46.36 | Psykick | its a matter of the heart |
22:46.56 | Seldon1975 | [Atlas] you dont think we can all subside without material wealth? |
22:47.10 | *** join/#asterisk girllinux_21 (n=ircap8@126.Red-83-56-107.dynamicIP.rima-tde.net) |
22:47.13 | _Sam-- | i see the poor people in american society, and i have no compassion, or heart...because im like "that f'er could have studied as hard as i did and got a good job too" |
22:47.21 | [Atlas] | Seldon1975: the fact is no country in history has found your enlightened way -- the closest we have ever came as the human race to peacable comfortable life is through capitalism |
22:47.24 | cpm | I think we would subside without material weath. |
22:47.29 | Psykick | sam: what's not to say that they didn't |
22:47.35 | Seldon1975 | [Atlas] bullshit |
22:47.50 | Seldon1975 | [Atlas]: throughout history there are heaps of examples |
22:48.02 | Psykick | sam: and that their job was made redundant .... partner left them ... world ... family all turned their backs on them |
22:48.03 | _Sam-- | Psykick: most of the poor are poor because they are lazy! <sorry that is a HUGE generalization and not exactly my point>...but that is how i often feel |
22:48.03 | [Atlas] | go for it |
22:48.14 | [Atlas] | agree with sam |
22:48.18 | Seldon1975 | [Atlas]: for a time, the romans, the greeks, the chinese |
22:49.04 | [Atlas] | ill leave you to stufy some history to find out how wildly erroneous of a statement that ws |
22:49.04 | [Atlas] | was |
22:49.08 | [Atlas] | study* |
22:49.09 | *** join/#asterisk rene- (i=rene@201.144.60.114) |
22:49.13 | *** part/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co) |
22:49.29 | Seldon1975 | [Atlas]: ill leave you to study it and learn |
22:49.29 | Seldon1975 | [Atlas]: the truth |
22:50.10 | [Atlas] | Seldon1975: i am not much of a programmer or asterisk hacker,, but one thing i do know is world history -- especially in classical times |
22:50.31 | Seldon1975 | [Atlas]: I'm not saying they were like that throught all of history; but they enjoyed times of peaceful prosperity |
22:50.32 | Psykick | sam: I agree to a certain degree |
22:50.52 | rene- | hello |
22:50.53 | Psykick | sam: most have not been exposed to encouraging parents .... or loving parents for that matter |
22:51.04 | [Atlas] | and by the way so there is no confusion rome was a republic built on money much like america today the only difference was the lack of democratic influence |
22:51.18 | shmaltz | silly: |
22:51.19 | [Atlas] | when it turned into a dictatorship for all intents and purposes the people were abused |
22:51.20 | shmaltz | http://ask.yahoo.com/20060131.html |
22:51.30 | Seldon1975 | [Atlas]: ok I'm talking about before that |
22:52.07 | cpm | what we have now is a corportacracy, a capitalist fuedalism. Not exactly a republic. |
22:52.45 | Seldon1975 | [Atlas]: do you really think that society is at it's peak? |
22:52.53 | Seldon1975 | [Atlas]: no, infact it's falling apart |
22:52.58 | _Sam-- | good conversation, i would love to stay and continue, but ive already stayed at work an extra to debate...its quittin time ...i will think of some good arguments on the way home :) |
22:53.08 | cpm | g'night |
22:53.10 | Seldon1975 | hehe |
22:53.11 | Seldon1975 | ciao |
22:54.04 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
22:54.04 | [Atlas] | Seldon1975: at its peak -- technologically yes,, as far as peace goes we have alot of things to get over in the world i dont believe money is one of them |
22:54.05 | cpm | Seldon1975, it is certainly at a cusp, that's pretty hard to argue. Either we outgrow this feudalism AGAIN, or we go back through it all AGAIN. No argument. |
22:54.53 | *** join/#asterisk RoyK (n=roy@193.80-202-93.nextgentel.com) |
22:55.14 | [Atlas] | all of the major bloodbathes of the world were not influenced by money ,, but by the forcing of power, through religion, ideals and such |
22:55.26 | _Sam-- | they (the govt) needs to setup asterisk servers for voting...then when they have big issues, we just vote through asterisk :) |
22:55.33 | [Atlas] | the fact is there will always be someone in power |
22:55.40 | _Sam-- | then we could vote on the issues instead of leaving it to our representatives |
22:55.56 | [Atlas] | money is the only means to get there -- for the most part -- without force |
22:56.09 | Psykick | [Atlas]: quite a few that we don't know about ..... and I don't believe that they are apart of any government |
22:56.10 | lesouvage | Does asterisk support/make use of dual core processor power on the motherboard. |
22:56.19 | [Atlas] | what we lack is moral fiber and honesty -- that mean is easily corrupted with or without money |
22:56.32 | [Atlas] | there always needs to be a medium of exchange if not money then what? |
22:56.34 | Psykick | lesouvage: that's a linux thing not asterisk .... someone correct me if I'm wrong |
22:56.38 | [Atlas] | religion, virtue, faith? |
22:56.39 | mzo_ | i thought this was #history? |
22:57.18 | austinnichols101 | set ramblingOffTopicConversation = on |
22:57.23 | [Atlas] | lesouvage: Asterisk runs quite well on my dualcore bro :) |
22:57.31 | austinnichols101 | nice |
22:57.36 | lesouvage | Psykick: as far as I now it depends on the application but I may be wrong. |
22:57.38 | Psykick | [Atlas]: where are you based |
22:57.49 | [Atlas] | idaho |
22:57.58 | [Atlas] | for now :D |
22:58.17 | Seldon1975 | Atlas: do you contend that the Iraq war is not about money? |
22:58.25 | justinu | you're THE ho? |
22:58.31 | justinu | quite an honor |
22:58.32 | mzo_ | i thought they were fighting Nod? |
22:58.43 | lesouvage | Atlas: and both cores are used by Asterisk. |
22:58.46 | Seldon1975 | Atlas: because if you do we might as well stop talking |
22:58.50 | [Atlas] | Seldon1975: im no bush lover |
22:58.51 | mzo_ | asterisk does smp fine? |
22:59.03 | [Atlas] | no asterisk is not using both cores |
22:59.08 | mzo_ | it doesn't? |
22:59.09 | cpm | Seldon1975, Not directly, it's about empire. |
22:59.13 | [Atlas] | but..... os and other processes are |
22:59.20 | mzo_ | i have an smp box, but i never bothered to check if it did or not |
22:59.24 | [Atlas] | there was a *drastic* improvement |
23:00.02 | bigjb | did you recieve my message Psykick ? |
23:00.03 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
23:00.03 | Seldon1975 | [Atlas]: you said "all of the major bloodbathes of the world were not influenced by money" were you ignoring the Iraq conflict or have you not heard of it? |
23:00.32 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
23:00.33 | Seldon1975 | cpm: my ass |
23:00.37 | Seldon1975 | cpm: its about Oil |
23:00.41 | Seldon1975 | cpm: = money |
23:00.43 | [Atlas] | Seldon1975: you ahve asked me a question i take an unusual stance on,, i believe that may be a big motivating factor yes,, but i do believe we should be there --- not for money or WMDs or anything like that but saddam did need dethroned as it were |
23:00.48 | mzo_ | is there some way to tell if asterisk is doing somthing with smp? some show (something) command? |
23:01.03 | cpm | Naw, it's about empire. Oil is also a big part, as is money, but principally, it's about control. |
23:01.14 | [Atlas] | Seldon1975: i also believe the war goes much deeper than that |
23:01.20 | Seldon1975 | forget saddam, forget WMDs; the biggest motivation for Bush to go to Iraq was Oil == money |
23:01.33 | Seldon1975 | Im trying not to rant |
23:01.38 | Seldon1975 | but it's so undeniable |
23:02.05 | Psykick | bigjb: yes thank you .... |
23:02.08 | lesouvage | Atlas: thanks for the info, buying a mb with a Intel 775 800 serie Dual Core seems to be a good idea. |
23:02.25 | cpm | Seldon1975,http://www.newamericancentury.org/ |
23:02.34 | [Atlas] | lesouvage: no prob bro :) |
23:02.43 | cpm | Come back and argue it again once you have read the manifesto. |
23:03.07 | mzo_ | that website is scary |
23:03.09 | [Atlas] | seldon1975: that is not a bloodbath on the scale of the major one of which i was talking |
23:03.14 | mzo_ | i always thought it was fictional for a movie plot |
23:03.15 | cpm | It's about empire. Yes, oil is a part, money is a part, but in the end, it's control and empire building. |
23:03.24 | [Atlas] | indeed their hatred for us is as great as we want power |
23:03.28 | bigjb | am waiting for word from avaya but ive had a play today and am pretty sure he was right |
23:03.30 | mzo_ | ...I find your lack of faith disturbing... |
23:03.37 | [Atlas] | if they showed us love we would not be over there |
23:03.59 | Seldon1975 | Atlas: that is such an ignorant statement I just cannot bring myself to argue any more |
23:04.01 | [Atlas] | agree with cpm for the most part |
23:04.15 | justinu | PNAC called "a new pearl harbor" an "the opportunity of ages" |
23:04.17 | [Atlas] | Seldon that was as great a copout as i have ever heard :) |
23:04.19 | justinu | that was in the 90s |
23:04.29 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
23:04.30 | Seldon1975 | theres just no point |
23:04.45 | cpm | Big purges of the last 100 years, Stalin, Hitler, Pol Pot, nothing money related there. All about empire, control, power. |
23:04.58 | justinu | "the process of transformation, even if it brings revolutionary change, is likely to be a long one, absent some catastrophic and catalyzing event -- like a new Pearl Harbor." |
23:05.00 | Seldon1975 | "if they showed us love we would not be over there" thats going in my book of all time American ignorances |
23:05.00 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
23:05.12 | [Atlas] | let us not forget the crusades and dark ages |
23:05.19 | sevard | bbl |
23:05.22 | *** part/#asterisk sevard (n=kynan@198.174.233.25) |
23:05.29 | cpm | [Atlas], yeah, again, pretty much the same thing. |
23:05.32 | [Atlas] | Seldon1975: you misunderstood |
23:05.36 | [Atlas] | my statement |
23:05.51 | [Atlas] | would the people stand for it if they were not in fear? |
23:05.59 | [Atlas] | would anyone stand for it? |
23:06.04 | [Atlas] | no. |
23:06.17 | Seldon1975 | Atlas: do you have any idea why terrorists (not Iraquis mind you) attacked America? |
23:06.31 | cpm | justinu, yeah, that's a pretty chilling statement, esp in retrospect, yeah? |
23:06.37 | [Atlas] | but throughout the conflict -- we have gone everywhere form as a country being 50/50 divided on us being there give or take 20 on each side |
23:06.42 | Seldon1975 | Atlas: was it because they just hate capitalism that much? |
23:06.43 | Psykick | everything in this world is about control |
23:06.54 | justinu | cpm - agreed. |
23:06.55 | [Atlas] | Seldon1975: i believe i understand that topic better than you |
23:07.01 | [Atlas] | no |
23:07.04 | [Atlas] | but that is part |
23:07.07 | Psykick | start losing it ... start a hate crime ..... fly a plane into a building killing thousands ... |
23:07.13 | Seldon1975 | Atlas: ok then; please explain why terrorists attacked merica |
23:07.18 | Seldon1975 | Atlas: I'm all ears |
23:07.18 | [Atlas] | it has alot to do with our support for israel |
23:07.22 | mzo_ | weird, so asterisk isn't smp. |
23:07.31 | justinu | merica.... merica.... fuck ya!! |
23:07.39 | cpm | It has a huge amount to do with our support for Israel, |
23:07.46 | [Atlas] | but it also has much to do with their hatred for our wasy of life |
23:07.52 | *** join/#asterisk dfroe (n=chatzill@dslb-084-056-227-185.pools.arcor-ip.net) |
23:07.56 | justinu | what I want to know is why we've never seen any surveillence footage of the so-called hijackers |
23:07.56 | bigjb | can anyone tell me why is it that im able to connect via sip to asterisk from home via nat port forwarding, but once it connects the call it fails to make a outgoing channel? |
23:07.58 | Seldon1975 | Atlas: oh my god |
23:08.03 | Seldon1975 | Atlas: pls stop |
23:08.17 | [Atlas] | Seldon1975: you can not believe ther might be a few reasons? |
23:08.23 | Seldon1975 | "it also has much to do with their hatred for our wasy of life" |
23:08.37 | cpm | folks who don't exactly hate us outright have been warning us since before the clerical revolution in Iran, toppling our puppet tyrant, that our support of Israel really pisses them off to no end. |
23:09.09 | Psykick | I don't agree either seldon |
23:09.18 | Seldon1975 | don't agree with what? |
23:09.26 | Psykick | <Seldon1975> "it also has much to do with their hatred for our wasy of life" |
23:09.33 | rob0 | Sure ... America's way of life is to exterminate or assimilate anyone who's different |
23:09.36 | Psykick | it has absolutely nothing to do with it |
23:09.45 | Seldon1975 | what has nothing to do with what? |
23:09.47 | masked | i just had mormans come to my door who will probably end up suicide bombing themselves in the name of whats good, thats enough relgion and polotics for me for one day |
23:09.55 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
23:09.58 | ReD-MaN | terrorism is nothing but a bunch of cowards thinking they are in control and can bully people around anyways |
23:09.59 | mzo_ | mormons :P |
23:10.01 | masked | so now can you pipe it and take it to a voip conference? |
23:10.04 | Psykick | the terrorists and the choices they made we're not over lifestyle |
23:10.09 | [Atlas] | ok i have to be done -- i have to do some work |
23:10.13 | Seldon1975 | ok |
23:10.18 | Seldon1975 | nice chattin' :} |
23:10.23 | masked | yeh it wont be long before born again christians start terrorism for their beliefs |
23:10.25 | masked | well... |
23:10.38 | [Atlas] | Seldon1975: even though i disagree with you ,, no hard feelings :) |
23:10.40 | masked | its a bit late for that really isn't it, bush already kills people in the name of god |
23:10.44 | [Atlas] | take care |
23:10.53 | Psykick | see ya atlas |
23:10.54 | Seldon1975 | [Atlas]: take care mate |
23:11.01 | cpm | I have to go also, Seldon1975, [Atlas], all, thanks for the chat! |
23:11.09 | Seldon1975 | cpm: you too |
23:11.13 | cpm | g'night. |
23:11.15 | [Atlas] | yeah thanks guys :) made my day fun hehe |
23:11.19 | Seldon1975 | heh |
23:11.20 | Seldon1975 | ciao |
23:12.27 | Psykick | masked: I don't believe that for a sec |
23:12.41 | Psykick | masked: I believe someone will get too greedy |
23:12.54 | Psykick | then all hell will break loose |
23:14.25 | Psykick | I better go as well |
23:14.33 | Psykick | later guys ... thanks for the chit chat |
23:14.54 | *** join/#asterisk oatis (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net) |
23:15.35 | oatis | Hi, my asterisk server is behind my router at the office and I would like to use it from home too... what ports do I need to forward in the router at the office to the asterisk server so I can do this? |
23:15.40 | oatis | Im using SIP |
23:15.53 | *** part/#asterisk clive- (n=pirch@dsl-165-149-246.telkomadsl.co.za) |
23:16.11 | dfroe | Hi, I read http://www.voip-info.org/wiki/view/GXP-2000 and found it very interesting. The new features like an adressbook are great for my phone! Does anyone of you know the (main) author of this site so I can contact him for further details? |
23:18.54 | austinnichols101 | oatis: nat at the office? nat at home? |
23:19.04 | rene- | how can one queue be spanned across multiple asterisk boxen? |
23:19.19 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
23:19.38 | RoyK | anyone here that knows why digium hasn't included PLC in codec_g729? |
23:19.54 | rob0 | I think the wiki pages all have an edit history available. |
23:19.54 | austinnichols101 | oatis: and what do you want to use at home - 1 phone, multiple phones? |
23:20.17 | RoyK | rene-: it can't |
23:20.48 | masked | i have a problem installing zaptel it doesn't put the module entries into /etc/modules.conf so it fails anyone know how to fix this? |
23:20.49 | *** join/#asterisk santiago (n=santiago@63.245.86.182) |
23:20.56 | RoyK | rene-: i beleive zoa is working on a commercial project on that subject, but standard asterisk doesn't support it |
23:21.21 | shmaltz | idiot: |
23:21.22 | shmaltz | http://www.breitbart.com/news/2006/01/31/D8FFUHC01.html |
23:21.48 | rene- | i saw something about it on *-biz i believ |
23:22.13 | rene- | im on a 6E1 + 300 user setup i dont think one box will suffice |
23:22.29 | shmaltz | rene, why not? |
23:22.31 | tronix | masked: don't need to put zaptel in modules.conf |
23:22.33 | mzo_ | one machine can handle 300 users? |
23:22.40 | shmaltz | lol |
23:22.57 | rene- | i wont use compression for calls but i will be doing gsm recording of calls |
23:23.16 | *** join/#asterisk sindy_84_ (n=ircap8@126.Red-83-56-107.dynamicIP.rima-tde.net) |
23:23.17 | ptiggerdine | SCSI disks u must. |
23:23.32 | rene- | ptiggerdine |
23:23.49 | mzo_ | so that grandstream disk is the best bargain these days? |
23:23.50 | rene- | i was thinking of having a realtime data in a separate mysql box |
23:24.04 | rene- | i could rig that with scsi and offload recording via nfs to that |
23:24.05 | mzo_ | hah, er, phone, not disk, that scsi thing got to me. :P |
23:24.11 | tronix | masked: start asterisk with -vvvvc option and see if it mentions chan_zap.so loading |
23:24.14 | ptiggerdine | nfs is slow. |
23:24.33 | rene- | how can i offload the complexities of recording to a third machine |
23:24.34 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.107.102.Dial1.Chicago1.Level3.net) |
23:24.44 | ptiggerdine | one storge box connected to the two systems would work better. |
23:24.45 | *** join/#asterisk lucasjb (n=lucas@mail.stabat.com) |
23:24.49 | litage | to install h323 support, do you just run 'make' from within the asterisk-addons directory, or do you need to specifically run 'make' within asterisk-addons/asterisk-ooh323c/ ? |
23:25.11 | tronix | litage: not sure but you could do 'make -n install' from both dirs and see which looks sane |
23:25.12 | ptiggerdine | good luck compiling it. |
23:25.22 | brockj49464 | is there an IRC channel for AAH? |
23:25.25 | tronix | litage: -n pretends to install but doesn't actually do it, so you find out what it'd have installed |
23:25.27 | masked | tronix: ok, make install failed, but asterisk still loads that module |
23:25.38 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.92) |
23:25.52 | masked | tronix: so that should be fine? |
23:26.05 | litage | thanks tronix |
23:26.07 | rene- | tigger: do yo mean something like the apple xserve raid? something fiber based? |
23:26.14 | tronix | masked: on the asterisk side, think so, yes. on kernel side, will want to check with 'lsmod | grep zaptel' |
23:26.28 | tronix | ditto for whatever wcxxx module you use too |
23:26.32 | ptiggerdine | rene-, not I'm talking SCSI here. |
23:26.32 | lucasjb | Hiyas, I have a problem with my Asterisk system. When I've got one SIP user connected testing extensions.conf, everything seems to work fine, but if I have two users, as soon as they system wants to play an audio file, my CPU usage goes from 99% idle to 99% system. The process events/0 is taking all system resources. Where should I start looking to fix this problem? |
23:26.52 | ptiggerdine | F/C it going to be over kill |
23:27.01 | ptiggerdine | but NFS is going to be very slow. |
23:27.18 | hardwire | wtf is libzap? |
23:27.28 | masked | tronix well its not loaded |
23:27.57 | rene- | lucas you must certainly have an mp3 player issue |
23:27.59 | tronix | masked: okay. what happens if you do 'modprobe zaptel' |
23:28.10 | rene- | try to upgrade your mpg123 imple,entation |
23:28.11 | masked | tronix module not found |
23:28.26 | tronix | masked: you've got two parts of zaptel stuff: one is kernel side, one is asterisk side. you've got asterisk side ok, but need kernel side |
23:28.31 | tronix | kernel side talks with the actual hardware |
23:28.34 | masked | tronix ill show u the install error |
23:28.49 | lucasjb | rene-, Thanks I'll try that. |
23:28.50 | tronix | masked: could you use www.pastebin.com? |
23:28.54 | masked | yep |
23:28.58 | tronix | sweet |
23:29.32 | masked | http://pastebin.com/532995 |
23:29.42 | masked | ok the error is at the end of line 3 |
23:29.50 | *** join/#asterisk sevarrd (n=kynan@198.174.233.25) |
23:30.03 | sevarrd | is there such a thing as a Clean Room / Dirty Room module for *? |
23:30.09 | dlynes | does anyone know of a way to get a timing device working under freebsd for musiconhold, if you're using a pure software solution for Asterisk? |
23:30.16 | masked | where it says echo "alias etc etc" >> (modules.conf should be here) ; |
23:30.59 | masked | sevarrd: asterisk will clean my room? |
23:31.10 | masked | you mean to say all this time i've been cleaning my room without the need to? |
23:31.19 | masked | omgbbq |
23:31.21 | tronix | masked: i'm thinking a null variable is causing your install failure. should be easy to work around. what distro you using? |
23:31.31 | sevarrd | no, hotels use clean room / dirty room systems where a maid calls in and reports a room clean and ready for sale |
23:31.37 | rene- | tigger: is this possible? asteriskbox-scsiadapter-scsicable-scsiadapter-scsidrives-mysqlbox |
23:31.42 | sevarrd | and it's omgwtfbbq :P |
23:31.45 | tronix | :-) |
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23:32.20 | znoG | sevarrd: you could do one in Perl (asterisk->AGI) |
23:32.22 | sevarrd | I was searching and didn't find one already written for *, has anyone heard of one? |
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23:33.06 | dfgas | is there a howto on installing from source not iso |
23:33.40 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-169.nas28.salt-lake-city1.ut.us.da.qwest.net) |
23:33.44 | rene- | so do you people think one top of the line dual xeon box or dual Opteron can handle 6xE1 with 300 sip users? |
23:33.47 | masked | sevarrd you could make something simple like that with a dialplan |
23:33.57 | sevarrd | masked: ? |
23:34.00 | masked | tronix im using LFS 6.0 |
23:34.11 | infobox | hello has anybody had problems with INtel 915 chipset and TE110P? |
23:34.15 | tronix | masked: ahh-ha. hm. what hardware card do you use? |
23:34.25 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.92) |
23:34.30 | hardwire | anybody have a good process for retrieving recorded calls (via snom 3x0 record) |
23:34.31 | masked | tronix i just installed a digium x100p, yet to use it |
23:34.36 | [av]bani | rene-: 6 x E1 is only 180 channels... |
23:35.03 | tronix | masked: ok. you're doing zaptel installation by doing a source build, right? what version? |
23:35.09 | masked | sevarrd, i assume they pick up the room phone, dial an extension then indicate whether it's clean or not? |
23:35.12 | _Sam-- | sevarrd: to do what you want should be really easy |
23:35.24 | masked | tronix 1.2.3 |
23:35.26 | _Sam-- | but will need some custom programming most likely |
23:35.26 | sevarrd | _Sam--: how would one go about that |
23:35.33 | rene- | [av]bani: agreed but doesnt converting from zap to sip (g711) count? |
23:35.33 | _Sam-- | i personally would keep an sql table |
23:35.48 | _Sam-- | and when they (the maids) call in, i would update that table |
23:35.51 | [av]bani | rene-: probably not much |
23:36.23 | sevarrd | _Sam--: then maybe paste that table to a html document |
23:36.30 | masked | sevarrd have something like, they dial an extension, get prompted for the room number, then idicate 1 for clean 2 for dirty, and that gets logged to the database |
23:36.35 | _Sam-- | or have your hotel be able to query against that table |
23:36.55 | tronix | masked: ok. let me look at something .. brb |
23:37.07 | masked | the table doesn't have to be pasted sevarrd, you will query it like _Sam-- said |
23:37.12 | _Sam-- | like you could have a webpage using php that would show that status of any rooms |
23:37.13 | masked | tronix thanks mate |
23:37.14 | sevarrd | wow, i'm not looking forward to trying to program that. I know _nothing_ about sql |
23:37.15 | rene- | [av]bani: i would need call recording for 75% of calls would that impact performance very badly? |
23:37.38 | _Sam-- | it will need some custom programming from what it sounds |
23:37.53 | masked | yeah shouldn't be too hard tho |
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23:38.20 | masked | just paypal donate the right ppl and you'll have it done in no time :) |
23:38.23 | _Sam-- | there are usually at least a dozen ways to do the same thing |
23:38.32 | _Sam-- | but that is my first thought on how i would do it |
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23:39.24 | _Sam-- | if the phones are voip and connect to the same asterisk server they wouldnt even need to do anything else except call the server when its clean |
23:39.25 | _Sam-- | and hang up |
23:39.34 | _Sam-- | the server would know what room it is |
23:39.36 | masked | _Sam--, sevarrd, considering each room has an extension/callerid, it could be as simple and dialing an extension and hanging up |
23:39.47 | masked | exactly. |
23:40.06 | rene- | i should buy opteron just to stay safe |
23:40.21 | masked | tronix it did it in 1.2.2 aswell, i was trying to install it for the ztdummy module |
23:40.31 | ptiggerdine | rene-, buy something that from the major players. |
23:40.37 | ptiggerdine | IBM, HP, DELL |
23:40.44 | masked | tronix just fyi |
23:41.05 | sevarrd | masked: most def |
23:41.11 | tronix | masked: ah, good info. still checking, btw. |
23:41.54 | sevarrd | it looks like i'm going to have to learn mySQL and php |
23:41.56 | _Sam-- | sevarrd: how do you handle voice mails after the customer checks out? |
23:41.58 | sevarrd | because I know neither. |
23:42.08 | _Sam-- | like if the customer had voice mail, then they check out...do you have something to get rid of it? |
23:42.15 | sevarrd | _Sam--: currently I don't, this is just a test server sitting in my room |
23:42.20 | [av]bani | whats a good WinCE softphone? |
23:42.38 | _Sam-- | im always interested to hear how specific industries use asterisk... |
23:42.44 | _Sam-- | hearing about the hotel is good |
23:42.49 | [av]bani | _Sam--: ILECs use it for... phones! |
23:42.54 | [av]bani | !!! |
23:42.54 | tronix | masked: what kernel do you use? 2.4 or 2.6? also, think I see your installation problem |
23:43.05 | masked | tronix 2.6 |
23:43.07 | sevarrd | _Sam--: I suppose i'd have to look into that |
23:43.17 | _Sam-- | but now after just listening to sevarrd i would feel pretty comfortable talking to hotels about their phones |
23:43.25 | tronix | masked: Makefile in zaptel-1.2.3 has a part that tries to figure out MODCONF variable -- which file to dump the alias entries in |
23:43.41 | tronix | masked: and from your pb output, looks like it couldn't figure out MODCONF location and left it empty |
23:43.45 | [av]bani | _Sam--: most hotel phone systems are unbelievably shit |
23:43.49 | tronix | masked: which causes odd errors that you saw |
23:44.04 | masked | so, export that variable? |
23:44.11 | sevarrd | _Sam--: you sound excited, want to teach me? :) |
23:44.22 | tronix | masked: could hack an entry like: MODCONF=/path/to/somewhere after that big MODCONF section |
23:44.28 | tronix | masked: just to get stuff going. |
23:44.45 | tronix | masked: but you're going to need to figure out which file/dir your distro uses for modules definitions |
23:45.00 | masked | tronix yeah thats no drams |
23:45.04 | masked | thanks mate. |
23:45.07 | litage | compiling and installing ooh323c that comes in asterisk-addons produces chan_ooh323.so , but if i compile a version of ooh323c that i downloaded, chan_ooh323.so isn't created. how are you supposed to do this? |
23:45.08 | tronix | splendid |
23:45.12 | _Sam-- | sevarrd what are some other things specific to hotels that you have to worry about? |
23:45.27 | _Sam-- | wake up call, room clean / not clean, voicemail, calling cards |
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23:45.59 | sevarrd | that sounds pretty much it besides maybe a head maid going around and overriding room clean entries |
23:46.33 | _Sam-- | not that it matters one bit, im just curious how many rooms are you talking about? |
23:47.17 | sevarrd | not sure yet, it's hypothetical, i wanted to help mom and pop places with ~50 rooms |
23:47.18 | masked | tronix ok that worked. |
23:47.28 | tronix | great |
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23:47.43 | _Sam-- | what are you going to use for phones / fxs? existing phones with ata? |
23:47.47 | masked | tronix one thing tho |
23:47.52 | _Sam-- | im just wondering what your strategy / plan is |
23:48.01 | sevarrd | _Sam--: I don't have one yet |
23:48.03 | tronix | masked: what's that? |
23:48.13 | masked | tronix it added module entries in like install wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg |
23:48.18 | masked | as well as aliases |
23:48.23 | sevarrd | well, i've looked at serveral ATAs |
23:48.29 | [av]bani | _Sam--: sounds like a job for powerline ethernet/wifi and ATAs |
23:48.29 | _Sam-- | so its your goal to sell these type setups to hotels? but you dont have a hotel that wants it right now? |
23:48.44 | [av]bani | or a DSLAM and CPEs |
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23:49.07 | tronix | masked: aye. problematic? |
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23:49.21 | _Sam-- | if all the phones run back to a phone closest, couldnt you use some type channel bank there? |
23:49.22 | tronix | masked: don't think it'll load at next boot |
23:49.27 | tronix | masked: for the stuff you don't use |
23:49.29 | justinu | how much do DSLAMs run? |
23:49.36 | [av]bani | depends on the dslam |
23:49.42 | [av]bani | $1500 - $150,000 |
23:49.43 | Qwell[] | $12.50, black market |
23:49.55 | tronix | :-) |
23:49.56 | [av]bani | how fast/far you wanna go? |
23:49.57 | mzo_ | i don't think you get warranty on the black marketz |
23:49.59 | justinu | Qwell[]: i'll meet you in the parking lot tonight |
23:50.05 | justinu | [av]bani: no idea, just kinda curious |
23:50.10 | masked | tronix my concern is are those entries supposed to be there? |
23:50.14 | [av]bani | justinu: ebay + dslam |
23:50.18 | masked | tronix or were they placed there by mistake? |
23:50.24 | austinnichols101 | good, fast, cheap - pick any two you want. |
23:50.24 | tronix | Qwell[]: you forgot the key part -- make sure it isn't Vonage-locked. :-) |
23:50.28 | Aldo | hi |
23:50.36 | Qwell[] | vonage-locked dslam? |
23:50.37 | [av]bani | of course if you dont want nifty phones, a channel bank in the office would suffice |
23:50.41 | [av]bani | 50 FXS... |
23:50.41 | Aldo | I have an server asterisk 1.12 |
23:50.47 | _Sam-- | yeah thats what im thinking |
23:50.50 | _Sam-- | easy sell |
23:50.51 | Qwell[] | Aldo 1.12? |
23:50.52 | _Sam-- | existing phones |
23:50.54 | tronix | masked: it just dumps a pile of stuff in by default |
23:50.57 | tronix | masked: so that's normal |
23:51.01 | masked | oh ok cool |
23:51.01 | [av]bani | just no nifty new phone features |
23:51.05 | Aldo | v1.1.2 |
23:51.07 | tronix | masked: makes it easier to throw in different cards in the future |
23:51.13 | Qwell[] | Aldo: No such thing |
23:51.16 | masked | righto |
23:51.18 | [av]bani | hmm one thing, hotels usually use phones locked down to legacy PBX |
23:51.23 | _Sam-- | when was the last time you stayed at a hotel with a phone that had an LCD display? |
23:51.29 | [av]bani | with the custom MWI and stuff |
23:51.34 | rpm | is it illegal to record phone conversations for personal use? im not a business..? |
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23:51.49 | Qwell[] | rpm: Do you have permission? Are you in a state that requires it? |
23:51.50 | [av]bani | rpm: depends on your jurisdiction and who is calling |
23:51.52 | Aldo | It was configuraded but I don't Hear nothing |
23:51.54 | Qwell[] | Go see a lawyer... |
23:51.54 | tronix | rpm: depends on your local/state|provincial|regional laws |
23:51.58 | rpm | i am in canada, im calling dell. |
23:52.07 | [av]bani | ah canada, go hog wild then |
23:52.10 | [av]bani | no law there |
23:52.11 | _Sam-- | lol |
23:52.19 | austinnichols101 | you can listen in for QA purposes and then you're allowed to record if you hear something illegal |
23:52.21 | Aldo | I think the problem are the codecs |
23:52.23 | masked | tronix maybe i should reboot.... but when i modprobe zaptel i get this zaptel: Unknown symbol crc_ccitt_table |
23:52.31 | Qwell[] | Aldo: What version of *? |
23:52.35 | Qwell[] | 1.1.2 is not valid |
23:52.35 | tronix | masked: ah think that's easy fix. no reboot needed btw. hang on |
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23:52.38 | [av]bani | if they threaten you tell them they can go shove their head up a mooses butt |
23:52.48 | Qwell[] | Do you mean 1.2.2? If so, upgrade |
23:52.56 | Aldo | 1.2.2 |
23:52.57 | hardwire | crap |
23:53.00 | Aldo | v 1.2.2 |
23:53.01 | Qwell[] | go upgrade |
23:53.02 | masked | tronix k |
23:53.03 | _Sam-- | [av]bani: if you needed 50 fxs what would you buy? |
23:53.05 | hardwire | what are the upsides to 1.2.3 vs 1.2.1 :( |
23:53.05 | Qwell[] | come back when you're done |
23:53.06 | QbY | anyone know how to reset a Sipura SPA-2000 when you don't have the password?? |
23:53.08 | *** join/#asterisk sevardz (n=kynan@198.174.233.25) |
23:53.09 | hardwire | well 1.2.4 |
23:53.11 | sevardz | connection. :P |
23:53.11 | tronix | masked: look at README. has solution |
23:53.14 | Qwell[] | hardwire: less memory leakage |
23:53.21 | hardwire | Ihate how this many stable releases just like.. happened |
23:53.27 | [av]bani | QbY: pick up fxo, pres **** and then 73738 |
23:53.32 | hardwire | and their changelogs leave something to be desired |
23:53.35 | Qwell[] | hardwire: "release" versions |
23:53.38 | tronix | masked: it's in the 'Brief F.A.Q.' section at the end. e-z fix tho |
23:53.49 | [av]bani | QbY: err FXS |
23:53.52 | QbY | [av]bani -- Its asking for a password |
23:53.52 | hardwire | Qwell[]: you would think release meant stable :) |
23:54.15 | [av]bani | QbY: nice, its locked then. you might be able to provision it via tftp |
23:54.18 | [av]bani | or dhcp |
23:54.24 | brockj49464 | Question about SPA2100: How do you find out how long the flash button on a phone is? Is there a way to log hook switch? |
23:54.30 | QbY | you got any instructions for that? |
23:54.39 | [av]bani | QbY: have you got a dhcp server? |
23:54.42 | _Sam-- | who else makes a 24port fxs channel bank besides rhino? |
23:54.44 | QbY | yeah |
23:54.55 | Qwell[] | _Sam--: can get an adit |
23:54.58 | znoG | is there any difference between doing a quick tap on the hangup switch and pressing FLASH? |
23:55.04 | [av]bani | qby just a mo |
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23:55.07 | justinu | adtran, mainstreet |
23:55.08 | Qwell[] | znoG: not really |
23:55.11 | _Sam-- | ty. |
23:55.11 | masked | tronix breif faq is where? |
23:55.14 | justinu | a bunch of people |
23:55.34 | tronix | masked: end of README file in zaptel-1.2.3 directory |
23:55.36 | justinu | fxs channel banks have been around for 30+ years |
23:55.43 | justinu | maybe 40 |
23:56.25 | _Sam-- | most play fine with asterisk? |
23:56.43 | masked | tronix oh sorry missed u saying that |
23:56.51 | tronix | masked: ha, no worries, mate |
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