| 00:00.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:08.58 | Godfather_ | Blackvel, file permisions solved :d |
| 00:13.06 | *** join/#asterisk dkirker (n=dkirker@1.sub-75-213-81.myvzw.com) |
| 00:13.33 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 00:17.15 | *** join/#asterisk rdude99 (n=rdude99@adsl-76-192-188-201.dsl.chcgil.sbcglobal.net) |
| 00:20.30 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:21.21 | rdude99 | When you have an AGI program as the first & only line of a dialplan context, what is the proper way to check that the line is active if the user "hangs up"? It seems my script just gets killed when they hang up. |
| 00:22.27 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 00:22.56 | rdude99 | And this messes me up because I have tasks executed in the bg that don't get killed |
| 00:25.05 | ChannelZ | It should send your script a SIGHUP |
| 00:25.22 | ChannelZ | so depending on what you're writing in, if you are able to do something on the SIGHUP besides just die.. |
| 00:25.37 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 00:26.20 | rdude99 | So it's safe to assume if I get SIGHUP == user hung up? |
| 00:26.31 | ChannelZ | yes |
| 00:26.31 | Blackvel | i am off to bed |
| 00:26.32 | Blackvel | n8 |
| 00:28.37 | rdude99 | okie dokie, thanks |
| 00:30.26 | Hypey | please help i'm getting this when i call my 0845 number |
| 00:30.30 | Hypey | *CLI> == Using SIP RTP CoS mark 5 |
| 00:30.30 | Hypey | [Nov 29 00:28:38] NOTICE[12240]: chan_sip.c:19034 handle_request_invite: Call from '' to extension 'arundelc' rejected because extension not found. |
| 00:31.11 | *** join/#asterisk corretico__ (n=laguilar@201.201.46.106) |
| 00:31.51 | [TK]D-Fender | Hypey: And just like the error says, * can't find an extension to match 'arundelc' in the context the call searches in. |
| 00:32.32 | Hypey | ive got this far with * and its nearly working just need to get it to ring |
| 00:33.35 | Hypey | any ideas on what i need to do ? |
| 00:34.04 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 00:34.37 | Hypey | ok i think ive worked it out lemme give that a go ;) |
| 00:35.41 | Hypey | ok ive got a context that exists and this in it exten => 1234,1,Dial(SIP/arundelc) |
| 00:36.23 | johnakabean | blackvel: leaving it at direct0000 would be ok but I don't think its included in your default context. you have context=yourcontext in sip.conf but have you made that context section and put include => contextofdirect |
| 00:37.10 | johnakabean | i know he went to bed but hopefully he'll scroll back like I do |
| 00:41.53 | Katty | hi |
| 00:47.11 | *** join/#asterisk centrex (n=pornstar@72.242.143.237) |
| 00:47.45 | centrex | question, I have an em tx line, and an em rx line, is there any way to hve them bridge together so what is received on one channel goes out the tx of the other? |
| 00:47.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:50.17 | ChannelZ | he can't scroll back if he wasn't in the channel |
| 00:50.55 | Katty | hi ChannelZ |
| 00:51.00 | ChannelZ | heya katty |
| 00:54.44 | ChannelZ | Hypey: Soooo call 1234 |
| 00:55.56 | ChannelZ | sweet I just busted my headset |
| 00:57.09 | Katty | :< |
| 00:57.34 | ChannelZ | cheap piece of crap anyway |
| 01:00.18 | Hypey | how do i stop asterisk from auto answering ? |
| 01:00.31 | Hypey | chan, sorted that mate thx |
| 01:01.39 | ChannelZ | Some dialplan functions will do an Answer() even if you hadn't previously |
| 01:01.59 | ChannelZ | like Playback() - since it doesn't make much sense to Playback something on a channel that hasn't been picked up |
| 01:04.37 | *** join/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 01:05.30 | *** part/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 01:06.15 | kkemp | is surprised Switchvox doesn't support not auto answering |
| 01:06.27 | centrex | I have a paging system that is always connected to two different t1 channels, one is em_rx, and the other em_tx. I want these bridged together so the audio coming in on one is always sent out the other |
| 01:06.38 | centrex | is that possible? |
| 01:11.56 | *** join/#asterisk dkirker (n=dkirker@194.sub-75-212-141.myvzw.com) |
| 01:18.20 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.138.161.dsl.dyn.forthnet.gr) |
| 01:28.39 | *** join/#asterisk jblack (n=jblack@pool-71-173-9-159.sctnpa.east.verizon.net) |
| 01:29.14 | Katty | hi jblack |
| 01:29.23 | jaytee | hi jblack |
| 01:29.28 | jaytee | hi Katty |
| 01:33.16 | Katty | hugs jaytee |
| 01:33.50 | jaytee | hugs Katty |
| 01:39.10 | *** join/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 01:40.04 | *** part/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 01:41.10 | torriem | So even if I don't want to receive SIP calls directly at my ipaddress, and only want to make and receive calls via a SIP service like SIPgate, I still have to forward the sip and rtp ports to my asterisk server which is behind NAT? If so, how is it that I can receive and make calls from a soft phone through sipgate without any special firewall configuration (presume it uses stun to traverse the nat). Also if I forward the sip por |
| 01:43.11 | johnakabean | ExecIf($["${TRUNKOUTCID:1:2}" != ""],Set,CALLERID(all) are we allowed to put two execute statements on that line? |
| 01:43.42 | johnakabean | or, a better question, does callerid(all) include callerid(ani) >? |
| 01:43.59 | jql | erm... |
| 01:44.42 | jql | I have some problems with that priority on principle |
| 01:52.43 | *** join/#asterisk BugKhaM (n=BugKhaM@125.25.224.70.adsl.dynamic.totbb.net) |
| 01:56.29 | johnakabean | exten => s,n(trunkcid),ExecIf($["${TRUNKOUTCID:1:2}" != ""],Set,CALLERID(all)=${TRUNKOUTCID}) |
| 01:56.29 | johnakabean | exten => s,n(trunkcid),ExecIf($["${TRUNKOUTCID:1:2}" != ""],Set,CALLERID(ani)=${CALLERID(number)}) |
| 01:56.32 | johnakabean | that works perfect |
| 01:56.45 | johnakabean | as long as the ani is below the other one |
| 01:58.40 | *** part/#asterisk centrex (n=pornstar@72.242.143.237) |
| 02:01.09 | Katty | do be do |
| 02:04.31 | *** join/#asterisk dkirker (n=dkirker@112.sub-70-212-183.myvzw.com) |
| 02:14.19 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 02:18.37 | ChannelZ | Foom! |
| 02:25.01 | *** join/#asterisk andres833 (n=andres83@186.80.122.105) |
| 02:25.20 | *** join/#asterisk dkirker_ (n=dkirker@141.sub-75-211-83.myvzw.com) |
| 02:27.11 | *** join/#asterisk Alagar (n=Administ@122.164.39.103) |
| 02:28.32 | *** join/#asterisk rdude99 (n=rdude99@adsl-76-192-188-201.dsl.chcgil.sbcglobal.net) |
| 02:39.18 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 02:39.56 | *** join/#asterisk sp0rkme (n=chatzill@firewall.dchsystem.com) |
| 02:40.43 | BugKhaM | can we have a SIP user registered with any password? |
| 02:40.51 | sp0rkme | hi there. ^_^ does anyone have a couple of spare moments to help with an ODBC issue? |
| 02:40.54 | drmessano | Sure |
| 02:41.09 | drmessano | sp0rkme: I did until you busted an emo smile |
| 02:41.23 | sp0rkme | aww. sorry. :P |
| 02:41.36 | drmessano | BugKhaM: No, only the password you set |
| 02:41.41 | *** join/#asterisk warden (n=balh@68.36.166.88) |
| 02:42.16 | warden | i am having a hardtime configuring my tdm400p w/ dahdi |
| 02:42.37 | warden | no hardware timing source found in /proc/dahdi, loading dahdi_dummy" |
| 02:42.46 | warden | is the error i am getting |
| 02:43.33 | sp0rkme | @drmessano I have unixODBC setup and it works. I'm having issues connecting it to Asterisk. The config file seem to be fine (but, there's not much documentation on res_odbc)... But Asterisk won't show any DSNs. |
| 02:45.13 | drmessano | Post your configs |
| 02:45.15 | Katty | watches some farscape |
| 02:46.07 | sp0rkme | [sip] |
| 02:46.08 | sp0rkme | enabled = yes |
| 02:46.10 | sp0rkme | dns => keobi_sip |
| 02:46.12 | sp0rkme | username => asterisk |
| 02:46.13 | sp0rkme | password => |
| 02:46.15 | sp0rkme | pre-connect => yes |
| 02:46.24 | sp0rkme | sorry, didn't mean for it to send that on all different lines. |
| 02:49.07 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 02:49.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:50.01 | drmessano | Pastebin would have been a lot better |
| 02:50.07 | sp0rkme | yes. |
| 02:50.17 | sp0rkme | but. i just realized i'm the stupidest person on the planet. |
| 02:50.32 | sp0rkme | notice the third line, it's "dns" instead of "dsn" |
| 02:50.56 | sp0rkme | sorry to waste your time. i'm going to go to bed now. |
| 02:51.07 | *** part/#asterisk sp0rkme (n=chatzill@firewall.dchsystem.com) |
| 02:51.13 | drmessano | ^_^ |
| 02:52.26 | ChannelZ | warden: if it's loading dahdi_dummy then your tdm isn't being found/configured |
| 02:53.56 | ChannelZ | is wctdm24xxp uncommented in /etc/dahdi/modules ? |
| 02:54.55 | [TK]D-Fender | Wrong module |
| 02:55.04 | [TK]D-Fender | TDM400P = wctdm |
| 02:55.45 | ChannelZ | ah yes I always get that one wrong |
| 03:01.30 | rdude99 | Hey I'm using dahdi with a "Wildcard X101P Clone" card connected to a standard phone line. I can't seem to get callerid info (agi_callerid_number returns 'unknown'). |
| 03:01.52 | rdude99 | Is there a way to get more info to see whether or not callerid info is coming in in the frst place? |
| 03:02.00 | rdude99 | I start asterisk with -cvvvvvvv |
| 03:02.25 | rdude99 | chan_dahdi.conf: http://pastebin.ca/1691931 | console log: http://pastebin.ca/1691935 |
| 03:03.56 | ChannelZ | Stick a NoOp(${CALLERID(all)}) in |
| 03:05.32 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 03:05.58 | rdude99 | Seems to return blank: -- Executing [s@ttip-pstn:1] NoOp("DAHDI/1-1", """ <>") in new stack |
| 03:07.23 | ChannelZ | I dunno about the x101p but generally if I say the channel has caller ID and it really doesn't I'll get an error in the console |
| 03:07.42 | ChannelZ | So if I had to guess I'd say you're getting CID but it just happens to be blank from whatever phone your calling from? |
| 03:08.44 | rdude99 | I'm just calling from my cell to my home line...pretty sure at least a number comes through (I know I won't see a name) |
| 03:09.13 | ChannelZ | although your "cidstart=bell" isn't correct (it should be 'ring' or 'polarity') |
| 03:10.14 | ChannelZ | not sure if that will cause it to malfunction tho |
| 03:12.16 | rdude99 | same result, blank cid |
| 03:12.36 | rdude99 | Must you answer the channel before callerid info is available? I'm guessing no, but just verifying. |
| 03:12.50 | ChannelZ | no |
| 03:17.41 | ChannelZ | Are you answering right away? |
| 03:18.12 | ChannelZ | I've heard of some carrier's CID being a bit 'slow', not transmitting until after the first ring |
| 03:18.49 | ChannelZ | I know I had to tweak this as a delay in my ATA... |
| 03:19.21 | rdude99 | I never answer; All my script does is check CHANNEL STATUS (which returns 4 [ringing]) for about 5 seconds then hangs up. |
| 03:19.58 | *** join/#asterisk voipmonk (n=voipmonk@67.204.61.239) |
| 03:20.08 | rdude99 | I set callerid=1234567 and 1234567 showed up in the log; so 'asreceived' has a problem |
| 03:20.35 | rdude99 | is there a dahdi debug file or command I could use to see if cid stuff is coming through on a low level? |
| 03:20.39 | ChannelZ | unless it's not receiving anything (busted card? who knows) |
| 03:21.21 | rdude99 | I hope it's not busted...bought it new. |
| 03:21.43 | rdude99 | Don't have anything else to compare it against unfortunately. |
| 03:21.53 | ChannelZ | well it's not officially supported IIRC |
| 03:24.01 | ChannelZ | http://www.voip-info.org/wiki/view/X100P+clone |
| 03:26.00 | rdude99 | I really hope it's not the card......only thing I've got & this has to be working by Monday, haha |
| 03:26.38 | drmessano | I've heard of some carrier's CID being a bit 'slow', not transmitting until after the first ring <-- Its always after the first ring in the US |
| 03:27.03 | kkemp | Test with a caller ID phone? |
| 03:27.43 | drmessano | rdude99: you should wait at least 8 seconds |
| 03:28.28 | rdude99 | 8 seconds? how so? I thought CID should be available soon after the phone is ringing |
| 03:28.38 | drmessano | Between the first and second ring |
| 03:28.48 | drmessano | So at least as long as the first ring |
| 03:29.34 | ChannelZ | my tdm800 doesn't even show the call on the console until after the first ring - I'm not even sure how you'd change it? |
| 03:30.07 | drmessano | Change what? |
| 03:30.23 | ChannelZ | a delay. |
| 03:30.25 | rdude99 | Hmmm...agi vars are sent as soon as the script is started....maybe I need to wait & then use the GET VAR agi command? |
| 03:33.46 | ChannelZ | Do a Wait(3) or something before your AGI see if it makes any difference |
| 03:36.52 | warden | channelz: i missed your post before |
| 03:37.01 | warden | yea it is loading i just get an error |
| 03:37.32 | warden | http://pastie.org/718784 |
| 03:37.42 | warden | i put everything up there |
| 03:38.01 | warden | check it out if you could and let me know what you think |
| 03:38.29 | rdude99 | Ehh....still no difference. (with an 8-second Wait() before my AGI() |
| 03:40.29 | ChannelZ | warden: hmm I dunno that's wierd. Personally I turn off all the other drivers than the one I need |
| 03:41.34 | warden | yes ive tried that and it yeilds the same result |
| 03:42.01 | ChannelZ | one thing that seems curious, why lspci is showing "Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface" |
| 03:42.28 | *** part/#asterisk torriem (n=torriem@c-67-186-209-155.hsd1.ut.comcast.net) |
| 03:42.40 | warden | yes im wondering if that is somehow the cause.. ive searched the net and i find people having issues but no solutiosn |
| 03:42.41 | ChannelZ | maybe that's just how that card ID's it's self |
| 03:43.05 | warden | ive had it work w/ zaptel but w/ dahdi i am not having any luck |
| 03:43.07 | ChannelZ | chipset or something |
| 03:43.49 | rdude99 | interesting, that's the same thing my lspci shows |
| 03:44.09 | warden | ok. but does yours work w/ dahdi? |
| 03:44.16 | warden | you have a tdm400p? |
| 03:44.24 | rdude99 | x101p |
| 03:44.45 | rdude99 | It works as far as basic dialing-out/in |
| 03:45.03 | rdude99 | but it's just a single line |
| 03:45.28 | warden | yea i cant even get that.. i cant get it to work at all |
| 03:47.47 | ChannelZ | no idea if this is applicable to you |
| 03:47.49 | ChannelZ | http://forums.whirlpool.net.au/forum-replies-archive.cfm/1067155.html |
| 03:48.30 | ChannelZ | potential kernel weirdness |
| 03:48.40 | warden | let me take a look |
| 03:49.34 | ChannelZ | does dahdi_cfg -vv give you anything interesting |
| 03:50.33 | Katty | http://foodiefriday.files.wordpress.com/2009/11/dsc02018.jpg <- i would do it. |
| 03:51.23 | ChannelZ | periodic table of cupcakes |
| 03:52.21 | Katty | mhmm |
| 03:53.24 | warden | let me try that |
| 03:53.49 | warden | 0 channels to configure |
| 03:55.19 | ChannelZ | dunno. you're on a trixbox? |
| 03:55.43 | warden | yes |
| 03:56.25 | ChannelZ | no clue. Need a newer version of dahdi, something goofy in the hardware/install... |
| 03:56.38 | warden | it specifically has to do w/ dahdi itself.. not trixbox i had the same issue w/ asterisk now |
| 03:56.56 | warden | let me check but i think its the latest version |
| 03:58.23 | warden | yea its 2.2.0 |
| 04:02.00 | ChannelZ | dunno what to say, for whatever reason the driver is not functioning |
| 04:05.30 | warden | ok.. thats a honest answer... lol just not sure where to go from here |
| 04:05.31 | ChannelZ | /proc/dahdi truly doesn't exist? |
| 04:06.06 | warden | it does |
| 04:06.22 | warden | only file in that directory is "1" |
| 04:06.28 | ChannelZ | cat 1 |
| 04:06.41 | ChannelZ | it probably only refers to dummy |
| 04:08.05 | warden | ye |
| 04:08.07 | warden | p |
| 04:08.09 | warden | Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: Linux26) 1" (MASTER) |
| 04:08.41 | ChannelZ | I don't know why the wctdm driver isn't finding your hardware. It's dahdi 2.2.0 and not 2.2.0.2? |
| 04:08.45 | *** join/#asterisk psilikon (n=joel@221-116.96-97.tampabay.res.rr.com) |
| 04:08.53 | warden | its 2.2.0 |
| 04:08.57 | ChannelZ | (the drivers, not the tools) |
| 04:09.11 | warden | hmm.. how do i check? |
| 04:09.58 | *** join/#asterisk andres833 (n=andres83@186.80.122.105) |
| 04:11.40 | ChannelZ | good question |
| 04:11.52 | warden | "/lib/modules/2.6.18-128.1.10.el5/dahdi/wctdm.ko" |
| 04:11.59 | warden | i guess that would be the verision number |
| 04:12.34 | ChannelZ | only of your kernel |
| 04:14.09 | ChannelZ | I don't really see anything in the changelog that seems relevant anyways.. at a loss |
| 04:15.14 | Katty | [TK]D-Fender: how do you pronounce apres? |
| 04:15.19 | Katty | [TK]D-Fender: ^- 'after' |
| 04:16.50 | ian6 | ah-pray |
| 04:16.57 | Katty | thanks |
| 04:17.14 | ian6 | ... that's not entirely accurate, but close enough. |
| 04:17.23 | ChannelZ | ah-choo! |
| 04:17.26 | Katty | could you be more accurate? |
| 04:17.59 | Katty | or how it's different than the above. |
| 04:18.12 | ian6 | not unless you can think of a word that sounds like pray but with more emphasis/roll on the r. |
| 04:18.42 | ian6 | ... let's call it emphasis, you shouldn't sound like a scot when you say it :P |
| 04:19.26 | ChannelZ | http://www.thefreedictionary.com/apres ? |
| 04:20.51 | Katty | sounds like the grand ole oprah with a bit of an accent |
| 04:21.19 | Katty | southern georga accent |
| 04:21.38 | Katty | ChannelZ: thank you |
| 04:23.09 | ChannelZ | sure |
| 04:33.52 | eppigy | NEIN |
| 04:34.08 | ChannelZ | alright imma going to go watch Benjamin Button so I can finally mail this thing back to netflix |
| 04:34.54 | spck | do yourself a favor and just mail it back |
| 04:35.25 | ChannelZ | heh really? it looks way too long |
| 04:35.32 | spck | seriously after you fall asleep and wake up with the credits rolling, you're going to be like holy shit that guy in irc was right |
| 04:36.02 | ChannelZ | :D I'll let you know |
| 04:36.24 | spck | fair warning |
| 04:37.56 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 04:39.30 | spck | http://bash.org/?909673 |
| 04:39.32 | spck | hehe |
| 04:45.04 | rdude99 | I somehow got some callerid error messages; this stuff mean anything : http://pastebin.ca/1692001 |
| 04:57.00 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
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| 05:32.57 | joako | I have an Asterisk machine behind a NAT router and constantly it looses a SIP registration. Registration for XXX timed out, trying again. If I do a SIP debug i see I send messages but there is no response back. How do I resolve this? IAX registration through the NAT works fine |
| 05:34.44 | *** join/#asterisk gooph (n=gooph@pool-71-96-244-205.dfw.dsl-w.verizon.net) |
| 05:47.44 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
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| 06:29.07 | raden | how much ram should asterisk have to handle 50 concurrent calls ? |
| 06:31.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:37.01 | ian6 | 3. |
| 06:40.49 | *** join/#asterisk wolvenar (n=wolv@97-112-199-158.frgo.qwest.net) |
| 06:45.32 | wolvenar | I am trying to set Alert-Info.. but cannnot quit figure it out.. this is what I am currently trying |
| 06:45.33 | wolvenar | exten = s,1,SIPAddHeader(Alert-Info: "<http://127.0.0.1/Bellcore-dr3>") |
| 06:45.43 | wolvenar | can someone tell me what I am doing wrong? |
| 06:48.32 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 06:58.39 | raden | ian6: ???? 3 GB ? |
| 07:10.17 | *** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 07:15.26 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 07:16.36 | ChannelZ | spck: well I didn't fall asleep, but I did watch the entire last half at 1.5X speed |
| 07:17.59 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-100.twcny.res.rr.com) |
| 07:19.13 | *** join/#asterisk thansen (n=thansen@c-76-27-87-49.hsd1.ut.comcast.net) |
| 07:20.09 | raden | how it going ChannelZ ? |
| 07:20.25 | ChannelZ | it's going |
| 07:20.55 | raden | how much ram is enough for asterisk ? |
| 07:21.22 | raden | and how many concurrent calls can a P4 2.8 Handle ? |
| 07:21.54 | ian6 | raden: I was just kidding. |
| 07:21.59 | ChannelZ | depends entirely on what it's doing |
| 07:22.06 | ian6 | raden: you would have to really, really underspec the machine to starve asterisk. |
| 07:22.26 | raden | P4 2.8 w/ 1 GB |
| 07:22.45 | raden | ChannelZ: using asterisk as a switch and for VM nothing else |
| 07:22.52 | ChannelZ | If it's not transcoding media I'm sure it could handle hundreds because it's not really doing anything |
| 07:22.53 | raden | and CDR to MYSQL |
| 07:23.10 | ian6 | even if it's transcoding, shrug. |
| 07:23.20 | raden | so 50 active calls should not be a problem ? |
| 07:23.44 | ChannelZ | I really have no idea because there are too many factors involved |
| 07:23.52 | raden | Have 2 identical spec machines will load balance customers 1/2 and 1/2 and use the other for each as secondary server |
| 07:23.53 | ChannelZ | Where are the calls coming from? |
| 07:24.04 | raden | ALL SIP |
| 07:24.12 | ChannelZ | all voip over the net? a T1? Analog? |
| 07:24.35 | raden | OC-192 w/ 10 MB connect |
| 07:24.46 | raden | All over net |
| 07:25.32 | *** join/#asterisk shinao1 (n=shinao1@41.219.192.180) |
| 07:25.45 | raden | ? ? ? |
| 07:25.52 | ChannelZ | precisely |
| 07:26.07 | raden | what ? |
| 07:26.18 | ChannelZ | ? ? ? |
| 07:26.36 | raden | shit thats only 56 calls per machine on 10MB connect |
| 07:26.41 | ChannelZ | I DONT KNOW. Give it a try, tell us what happens. |
| 07:26.43 | raden | so that will be my limit |
| 07:26.51 | raden | I will :) |
| 07:27.12 | raden | well the good thing is were starting small and growing we will only have about 30 customers to start next month |
| 07:27.22 | raden | with 120 active lines |
| 07:27.38 | raden | worse comes to worse pay another 75 a month add another 1U server |
| 07:27.48 | *** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 07:27.49 | ChannelZ | I have no idea how you're coming up with that math but ok |
| 07:27.54 | raden | I hate craming everything in a 1 U case |
| 07:29.08 | raden | 10 mbips * 1024 to get kbps correct ? 10240 / 180 KBPS for 711 bandwith bi-directionally = 56.88 |
| 07:29.11 | raden | am I wrong ? |
| 07:31.00 | raden | or would it just be the 90 k ???? |
| 07:31.43 | raden | ChannelZ: yopur right 113 Concurrent calls |
| 07:31.50 | raden | so thats my limit |
| 07:32.01 | ChannelZ | g711 is 64kbit |
| 07:33.45 | voipmonk | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 07:35.10 | *** join/#asterisk reber (n=reber@78.251.154.180) |
| 07:35.57 | raden | ChannelZ: dont forget overhead |
| 07:36.27 | ChannelZ | you don't forget it |
| 07:36.43 | ChannelZ | doesn't matter to me :) |
| 07:39.05 | drmessano | ~book |
| 07:39.06 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 07:45.51 | raden | lol :) |
| 07:58.13 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 08:09.13 | *** join/#asterisk Tim_Toady (n=moi@62.1.240.24.dsl.dyn.forthnet.gr) |
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| 08:23.13 | *** join/#asterisk sp0rkme (n=chatzill@firewall.dchsystem.com) |
| 08:23.25 | sp0rkme | hello everyone. :) |
| 08:23.50 | sp0rkme | is anyone available that can give me a bit of information about asterisk realtime? |
| 08:26.38 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 08:34.44 | ChannelZ | It's like asterisk, only in real time. |
| 08:42.36 | sp0rkme | are the extensions in a database retrieved at a dialplan reload? or when one does 'dialplan show' should it just say 'Alt. Switch => Realtime/'? |
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| 08:59.44 | *** mode/#asterisk [+o denon] by ChanServ |
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| 09:21.15 | joako | raden: In the real world you will see much higher bandwidth than what that calculator states... notice "SIP overhead disregarded".... it says 150 G729 channels = 6.92 Mbps but in reality I see almost double the bandwidth used in MRTG graphs |
| 09:21.33 | joako | The % should be less for G711u but you get the idea |
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| 10:21.46 | Godfather_ | hi |
| 10:22.38 | ChannelZ | belches |
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| 10:37.50 | oberon | hi |
| 10:38.11 | oberon | I have a couple of users who keep getting spam from hotmail/live servers |
| 10:38.51 | ChannelZ | which has what to do with asterisk? |
| 10:39.14 | oberon | nothing |
| 10:39.21 | oberon | I misclicked |
| 10:39.24 | oberon | sorry |
| 10:39.33 | ChannelZ | no worries |
| 10:39.34 | oberon | off I go to spamassassin channel |
| 10:39.48 | ChannelZ | :) |
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| 11:04.36 | ChannelZ | wanders off to bed |
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| 11:59.15 | datacompboy | Hi all! :) Any chance to run software echo canceller on SIP channel? It have excellent ping (1.5ms) and almost no jitter, connected to pstn provider. We have sometimes echo :( And provider refuses to run echo canceller on his side. Echo are copy of our-sending to client get back. Echo time is realtive small, near from "none", but really there and degrade sound quality. Is there any software canceller, that can be used in that case? |
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| 15:15.54 | hluesea | hello channel, is anyone answer my questions on here ? |
| 15:16.32 | Blackvel | isn't it sunday morning in US? i doubt ppl hanging around here at this time ;) |
| 15:16.50 | kaldemar | ~ask |
| 15:16.51 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:18.13 | Blackvel | i am not sure how I could should by context for directip inbound |
| 15:18.56 | hluesea | i am newbei in asterisk, all other persons like asking questions :) firstly i want to interconnecting 2 different asterisk sip server(my side OS is centos5.4) and I am not actually sure which opsions will be changed or done on sip.conf file |
| 15:19.25 | Blackvel | shall I use exten => _X. or pin it to 1-3 numbers? may I include exten => s,1,goto for mapping non-extensions to a standard number? |
| 15:19.52 | Blackvel | looks like my log told me that ppl called me as 000@host whereas i provided a quite different extension number |
| 15:19.59 | Blackvel | so what is best practice for that? |
| 15:21.03 | Blackvel | for my pstn isdn connection my gateway just pickups up well known msn extension calls (and ignores all others) |
| 15:28.01 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 15:31.15 | Blackvel | hluesea: sound pretty advanced to me :) |
| 15:31.28 | Blackvel | kaldemar: are you from the US? |
| 15:35.04 | kaldemar | Blackvel: no |
| 15:35.19 | kaldemar | ~book |
| 15:35.20 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:35.39 | Blackvel | i am wondering what are best ways to hook up with UK/USA nowadays |
| 15:35.40 | kaldemar | hluesea: ^^ take a look at that, is has an example with sip |
| 15:43.11 | hluesea | thank you |
| 15:44.13 | Godfather_ | Any ideas why i dont get the output from Monitor() to one file? I've installed also, and i'm getting no error on cli... and i'm using the m option. Monitor(wav,${filename},m) Here is some output if it can help http://pastebin.com/m71dbf18f |
| 15:44.46 | Godfather_ | hi Blackvel |
| 15:44.54 | Blackvel | oh hi |
| 15:45.26 | Blackvel | did you take your time for a break? :) |
| 15:45.37 | Godfather_ | *i've installed sox also |
| 15:46.09 | Godfather_ | Blackvel, err what? |
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| 16:05.59 | Katty | morning |
| 16:08.00 | raden | morning Katty |
| 16:09.09 | jaytee | morning |
| 16:10.03 | *** join/#asterisk Alystair (n=bob@CPE001d7315f431-CM00223a5f4fe5.cpe.net.cable.rogers.com) |
| 16:11.00 | Blackvel | evening :) |
| 16:11.10 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 16:11.28 | Blackvel | how do I specific in a goto the priority when I am using ,n, syntax? |
| 16:15.52 | heliosj | By using a label. |
| 16:15.53 | gr0mit | make sure you use n(label) and the exten => n,goto(label) |
| 16:16.05 | heliosj | Yeah, like that. :) |
| 16:18.02 | Blackvel | ahh found something: |
| 16:18.17 | Blackvel | Goto(extension,label) |
| 16:18.23 | Blackvel | looks like i want that ;) |
| 16:18.36 | Katty | context,extension,priority |
| 16:18.37 | Godfather_ | extension,label? |
| 16:18.55 | Katty | (from-pstn,s,1) |
| 16:18.58 | Godfather_ | extension,label you mean |
| 16:19.16 | Godfather_ | Blackvel, http://www.voip-info.org/wiki/index.php?page=Asterisk%20priorities |
| 16:19.26 | Blackvel | what US providers are a must have for asterisk / voip business? have been with fwd before. not sure if my iaxtel setup still works |
| 16:19.36 | Blackvel | http://www.voip-info.org/wiki/view/Asterisk+cmd+goto found that...was not aware...sorry |
| 16:19.44 | Katty | infobot: itsp |
| 16:19.45 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:19.52 | Katty | infobot: itsplist-us |
| 16:19.53 | infobot | from memory, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:20.29 | Blackvel | how do you provide a all time valid voip number in the US? |
| 16:20.36 | Blackvel | direct ip address? |
| 16:21.00 | Katty | i would recommend calling one of the itsps and asking them to explain exactly how it works. |
| 16:21.27 | Blackvel | ok, finished directip context coding. who wants to test? :) |
| 16:22.01 | Blackvel | is there a "replacement" for FWD? |
| 16:22.26 | Blackvel | most of the listed providers are commercial |
| 16:22.32 | Blackvel | if a company asks you about your number |
| 16:22.36 | Blackvel | what number do you give? |
| 16:22.45 | Katty | whatever number they can reach you at |
| 16:22.57 | Blackvel | direct ip? iaxtel? do you ask if they are connected to broadvoice and others? :) |
| 16:23.09 | Katty | you ask whatever you want to ask |
| 16:23.11 | Blackvel | so you only provide pstn numbers |
| 16:23.11 | Katty | it's your query |
| 16:23.48 | Blackvel | looks like there is no "easy" answer to that :) |
| 16:24.51 | Alystair | ~itsplist-ca |
| 16:24.51 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 16:26.42 | heliosj | Babytel is annoying, imho |
| 16:27.29 | Blackvel | off for 15 minutes |
| 16:28.11 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-208-82.dsl.lsan03.sbcglobal.net) |
| 16:28.12 | Alystair | heliosj: have any that you'd recommend that are not on that list? |
| 16:28.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:31.51 | levity | Alystair: i have had good results with acanac, they have poor technical support for asterisk though, you must be prepared to do google seaches |
| 16:33.35 | Alystair | I just want something to get VOIP service on my E71 |
| 16:33.38 | heliosj | I have an account with unlimitel... I can't say anything bad about them. |
| 16:33.49 | Alystair | hmmm |
| 16:34.13 | levity | if you pay twelve months in advance you can get unlimited north American calling $10 Canadian a month |
| 16:34.43 | Alystair | no way to trial any of these services eh |
| 16:35.04 | Alystair | just need to test if VOIP is even reasonable from my home location |
| 16:36.50 | raden | How many pass through calls a a P4 2.8 handle in G.711 and G.729 ? |
| 16:36.53 | *** join/#asterisk giancast (n=giancast@201.218.100.108) |
| 16:36.58 | giancast | Good morning |
| 16:37.58 | giancast | I have a question. |
| 16:38.12 | giancast | I follow your advices about the VOIP provider. |
| 16:38.32 | giancast | Now I have just installed a E-1 wich a friend make it work last week. |
| 16:38.59 | giancast | electrical supply went out 5 hours, and the E-1 went down. |
| 16:39.17 | giancast | How should I make it run, there is a command to make it run in the asterisk cli??? |
| 16:39.22 | giancast | I´m using ZAPTEL |
| 16:39.52 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:40.34 | giancast | anyone? |
| 16:44.17 | drmessano | Make what run? |
| 16:46.03 | tzafrir | giancast, what is the output of: cat /proc/zaptel/* |
| 16:47.18 | tzafrir | wonders if Nokia also makes a E1 |
| 16:50.35 | Blackvel | have you found a way to perm disable cli notice (sip_reg_timeout) messages? |
| 16:53.51 | drmessano | Who? |
| 16:54.41 | Blackvel | all |
| 16:54.41 | Blackvel | :) |
| 16:55.08 | drmessano | I dont have those, so no |
| 16:55.42 | giancast | how to make run the E-1 |
| 16:57.05 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 16:57.06 | giancast | I´m so lost |
| 16:57.21 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 16:57.31 | drmessano | Which distro, giancast? |
| 16:58.11 | giancast | Trixbox |
| 16:58.13 | florz | drmessano: he doesn't seem to be interested in solving his problem ... |
| 16:58.18 | jaytee | morning [TK]D-Fender |
| 16:58.54 | drmessano | giancast: Reboot.. The zaptel.exe should be in the startup folder in trixbox and run after reboot |
| 16:59.15 | jaytee | lol |
| 16:59.15 | giancast | just like that |
| 16:59.19 | drmessano | yes |
| 16:59.22 | drmessano | just like that |
| 16:59.25 | jaytee | zaptel.exe ahhaahahaaaaaa |
| 16:59.33 | giancast | zaptel.exe <-- exe in linux. |
| 16:59.40 | drmessano | Reboot |
| 16:59.47 | jaytee | Startup Folder rofl |
| 17:00.43 | drmessano | Zaptel should run on startup with Trixbox. If it doesnt, somebodys been tinkering, or the problem isnt Asterisk |
| 17:01.12 | jaytee | hooking up an E1 to a trixbox server sounds kinda like installing a NOS system and turbocharger on a 2-speed sitdown mower |
| 17:01.16 | drmessano | Ive already given you 110% more of an answer than one normally gets for being a trixbox user |
| 17:01.42 | giancast | sorry |
| 17:01.54 | giancast | I´m doing this /sbin/ztcfg -vvvv |
| 17:01.59 | giancast | but it says no such command |
| 17:02.23 | drmessano | Reboot.. If that doesnt work, try the trixbox forums |
| 17:02.29 | giancast | vim /etc/zaptel.conf |
| 17:02.34 | drmessano | Reboot.. If that doesnt work, try the trixbox forums |
| 17:02.45 | drmessano | Read at least 1 of those 4 |
| 17:02.56 | giancast | Thanks for your help. |
| 17:03.02 | [TK]D-Fender | Trixbox switched to DAHDI some time ago... |
| 17:03.06 | drmessano | No problem, have a great day |
| 17:03.18 | giancast | People never born knowing. |
| 17:03.40 | giancast | and DAHDI is better, less complicated? |
| 17:03.55 | [TK]D-Fender | giancast: Same thing, new verson & name |
| 17:05.13 | giancast | Thanks, and commands are same? |
| 17:06.27 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 17:23.54 | tzafrir | giancast, did you see the question I left you? |
| 17:25.34 | tzafrir | boots drmessano for suggesting those ideas |
| 17:26.25 | drmessano | Do you really want to start digging into the overscripted config of a trixbox? |
| 17:26.43 | tzafrir | No. But basic diagnostics is the same |
| 17:27.07 | tzafrir | If there's no ztcfg, it is likely to have dahdi. |
| 17:27.10 | drmessano | Likely any steps he takes to mitigate startup issues with Zaptel/Dahdi will break some scripted functionality or conflict |
| 17:27.23 | tzafrir | If so: what are the outputs of: dahdi_hardware; lsdahdi |
| 17:27.27 | [TK]D-Fender | [12:01]<jaytee>hooking up an E1 to a trixbox server sounds kinda like installing a NOS system and turbocharger on a 2-speed sitdown mower <- loved that episode of "Home Improvement" |
| 17:27.41 | jaytee | lol |
| 17:28.11 | tzafrir | giancast, see my question above |
| 17:28.21 | drmessano | Ive been through zaptel hell with Trixbox. Best to just reboot or run a yum update to fix |
| 17:28.41 | drmessano | But what do I know.. |
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| 18:49.12 | simplydrew | can anyone help me figure out getting a2billing installed? I'm at the point of installing mysql, but I keep getting access denied errors when matching the install script with the a2billing.conf file for the database information |
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| 19:01.43 | Katty | morning, again |
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| 19:02.43 | sun28 | morning |
| 19:03.41 | Katty | i got 12 hours of sleep last night |
| 19:05.58 | Blackvel | are you sure? not 12 hours party? :) |
| 19:06.28 | Katty | i would not feel this amazing had i been partying all night. |
| 19:08.14 | Blackvel | oh i hate it. writing my cv in english |
| 19:09.10 | *** join/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 19:09.10 | Blackvel | simplydrew: have you already setup mysql/user permissions hostbased? got me crazy the lasttime |
| 19:09.14 | *** part/#asterisk Corndawg_ (n=notreall@c-66-176-47-112.hsd1.fl.comcast.net) |
| 19:10.00 | simplydrew | Blackvel: I don't think so. I was questioning as to whether or not sql was already installed, but I can't find anything that was pre-existing. so I'm not sure |
| 19:10.58 | Katty | how do you not know if your own server has sql |
| 19:11.13 | simplydrew | Well I'm not the one that fully set this up |
| 19:11.56 | Katty | and why can't you contact him and ask for instructions |
| 19:12.17 | *** join/#asterisk kudos4421 (n=marco@c-98-200-120-247.hsd1.tx.comcast.net) |
| 19:12.34 | simplydrew | Not sure how to get in contact with him anymore |
| 19:12.43 | *** join/#asterisk debuggerboy (n=debugger@117.196.162.73) |
| 19:13.06 | kudos4421 | can someone explain what this debug message means? [Nov 29 13:10:15] DEBUG[27719]: app_macro.c:442 _macro_exec: Extension s, macroexten 603, priority 3 returned normally even though call was hung up |
| 19:13.33 | Katty | kudos4421: what does your macro do? |
| 19:13.56 | file | kudos4421: why are you asking? |
| 19:14.12 | Katty | hugs file |
| 19:14.22 | file | hugs Katty |
| 19:14.53 | kudos4421 | Katty/@file: my system gets into a loop when it rings multiple extensions to the point that it brings the system down. Control-C doesnt even close the CLI |
| 19:15.22 | Katty | that doesn't answer the question. either of them. |
| 19:16.00 | kudos4421 | Katty: the macro checks for call recodign |
| 19:16.04 | kudos4421 | recording |
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| 19:27.03 | ChannelZ | picks his nose |
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| 19:36.12 | giancast | I´m so frustrated. |
| 19:36.17 | giancast | my e-1 still not work |
| 19:38.12 | ChannelZ | hire an illegal |
| 19:39.24 | Katty | hehehehe |
| 19:39.58 | datacompboy | Hi all! :) Any chance to run software echo canceller on SIP channel? It have excellent ping (1.5ms) and almost no jitter, connected to pstn provider. We have sometimes echo :( And provider refuses to run echo canceller on his side. Echo are copy of our-sending to client get back. Echo time is realtive small, near from "none", but really there and degrade sound quality. Is there any software canceller, that can be used in that case? |
| 19:40.50 | Blackvel | connected to pstn provider |
| 19:40.55 | Blackvel | how? |
| 19:41.13 | Blackvel | gateway on your side? over internet to a sip provider which terminates to pstn? |
| 19:41.26 | datacompboy | Blackvel: mine asterisk <-sip-> provider <-pstn-> client |
| 19:42.02 | datacompboy | Blackvel: yes, over internet to a sip provider. ping is 1.5ms, no jitter. |
| 19:42.12 | Blackvel | and who hears echo? your asterisk sip phones/clients or the other party (pstn)? |
| 19:42.39 | datacompboy | Blackvel: asterisk hear echo on sip->pstn line. Echo caused by pstn hybrid |
| 19:42.53 | Blackvel | ah ok |
| 19:43.02 | Blackvel | that means other party doesnt cancel |
| 19:43.30 | Blackvel | i am not aware what your asterisk could do about (cancelling echo on sip) |
| 19:43.30 | datacompboy | Blackvel: it an echo, that easy removed with olsec, but on zap. But prov don't do that :( and i can't, since i have sip, not zap channel |
| 19:44.09 | Blackvel | i am not really uptodate for 1.4/1.6 but i know no echo canceller for sip |
| 19:44.25 | Blackvel | not sure about jitterbutter in 1.4/1.6 would change anything for you (probably not) |
| 19:44.36 | Blackvel | is it isdn pstn or analog pstn? |
| 19:45.06 | datacompboy | Blackvel: jitterbuffer doesn't help, since there no jitter:) and i have tried several jb configs. no luck. |
| 19:45.22 | datacompboy | Blackvel: eh... bad. it will be good to have echo canceller in Local channel, as with jitter buffer. |
| 19:45.30 | ChannelZ | you should find a new provider |
| 19:45.52 | Blackvel | some german it consulter tell me always, that isdn needs no echo canceller |
| 19:46.12 | Blackvel | so they go with isdn cards and bristuff/misdn and stuff like that with NO echo enabled (was not successful on MY side) |
| 19:46.18 | datacompboy | ChannelZ: yep, that we will do it prov do not fix problem tomorrow :) |
| 19:46.34 | Blackvel | personally its my opinion that a provider is crazy not to use HWEC (either isdn or analog) |
| 19:46.48 | datacompboy | Blackvel: yep, i know that too :D and I have enabled olsec in germany, since sometimes i HAVE echo, that need to be suppressed :) |
| 19:46.53 | Blackvel | since i switched to patton isdn bri (with hwec) echo cancel improved much |
| 19:47.27 | Blackvel | datacompboy: to be honest. in MY isdn setup no sw-ec helped (octasic..oslec...digium...) |
| 19:47.38 | Blackvel | maybe becoz of my epia via 1 gig cpu |
| 19:47.44 | Blackvel | i bought this patton isdn gw |
| 19:47.58 | Blackvel | with hw on and i only have 1-5% echo problems (inbound, not outbound) |
| 19:48.14 | drmessano | <PROTECTED> |
| 19:48.30 | Blackvel | how many calls / concurrent calls does your company have on your pstn provider? |
| 19:48.32 | drmessano | Echo is a function of analog hardware, not digital |
| 19:48.44 | datacompboy | Blackvel: we have relative old Digium cards, 400. Its without HW EC, so olsec used. |
| 19:49.01 | datacompboy | drmessano: read my 1st message -- echo caused by analog, but not removed before me. |
| 19:49.33 | Blackvel | drmessano: yes, heard that too. dunno why i had echo on isdn -> mobile phone / isdn pstn - isdn pbx - analog phones |
| 19:49.36 | drmessano | datacompboy: You dont have an echo problem, your provider does |
| 19:49.49 | datacompboy | Blackvel: here, where we got echo on sip, relatively small -- currently about 100 sessions per day, every call is 2-3 concurrent inbound/outbound calls |
| 19:49.50 | drmessano | datacompboy: You cant remove something thats not there |
| 19:50.06 | drmessano | Dump your provider if they can't interface to the PSTN properly |
| 19:50.10 | Blackvel | datacompboy: if you get a provider with HWEC you are fine :) would not even go into detail analysis for problems |
| 19:50.37 | datacompboy | drmessano: I _can_ remove echo, since i have all info in same state, as provider. just no ready software. |
| 19:51.26 | Blackvel | my personal believe: if voip is used in business it should be no matter of price |
| 19:52.21 | drmessano | datacompboy: No, actually, you can't.. You're talking about removing "echo" from a mixed digital signal.. You do NOT have access to both components needed in order to create a mix-minus and remove the "echo" |
| 19:53.03 | datacompboy | drmessano: Nope, i have both -- and what I send, and what I get. That's sip! I have both streams. |
| 19:53.19 | Blackvel | datacompboy: what sip hardware do you use? sip phones? or soft phones? once i started with this snom voip phone i got problems. with my isdn phones no problem at all |
| 19:53.50 | Blackvel | you could connect a hw card to your asterisk and use isdn phones...probably would eliminate hearing echo :) |
| 19:54.02 | datacompboy | Blackvel: i have echo on outgoing channel, while playback sound file to client. and i have on channel-in.au record louder copy of played file :) |
| 19:54.30 | drmessano | datacompboy: If the "echo" you are hearing is ONLY YOUR voice, then you have shitty phones.. if you had echo from the provider's PSTN connection, you would NOT be able to remove it, regardless of how many times you can tell me you have access to both ends |
| 19:54.50 | Blackvel | i know...its ridiculous suggesting that. but voip phones can introduce problems |
| 19:55.25 | drmessano | Its not ridiculous to suggest it.. there is an analog component to every voip phone. It's called the handset |
| 19:55.41 | florz | drmessano: why would one not be able to remove echo that originates from the PSTN behind a VoIP connection? |
| 19:55.44 | Blackvel | btw |
| 19:55.58 | Blackvel | snom 370 + standard headset = echo on my side (hearing me double) |
| 19:56.03 | drmessano | Shitty phones like grandstreams have lots of problems with "echo" |
| 19:56.05 | Blackvel | throwing away snom headset = fine |
| 19:57.00 | drmessano | florz: What are you going to use from the far end of a poorly coupled PSTN connection as your reference in EC? |
| 19:57.14 | Blackvel | get a 300 euro headset (bluetooth) and prolly that works better (got told by vendor where i bought snoms) |
| 19:57.26 | florz | Blackvel: I guess that phenomenon is called "sidetone" for lack of delay |
| 19:58.19 | Blackvel | florz: my feeling is that most EC can operate on the outbound, not on the inbound |
| 19:58.33 | drmessano | EC can operate on both ends |
| 19:58.41 | florz | drmessano: by "poorly coupled" you mean the hybrid mismatch? |
| 19:59.06 | Blackvel | so if other party does crap...you get crap (maybe its different for 10k HWEC. not sure about patton and sangoma) |
| 19:59.06 | drmessano | florz: yes. |
| 19:59.19 | *** join/#asterisk jpgneves (n=seven@78-130-56-32.net.novis.pt) |
| 19:59.23 | Blackvel | drmessano: my patton does not seem to be able to compensate echo for inbound...so....i do not trust for inbound ec |
| 19:59.34 | Blackvel | datacompboy: what sip hw? |
| 19:59.56 | florz | drmessano: well, if you have a fully digital connection to that hybrid's 4-wire interface, you basically have the exact same information an EC located with the hybrid has, no? |
| 20:00.45 | *** join/#asterisk datacompboy-reco (n=datacomp@l64-93-216.static.cn.ru) |
| 20:00.48 | drmessano | florz: No, you don't.. You have a mixed representation |
| 20:01.06 | datacompboy-reco | sorry, hard reconnect :D what someone did said? |
| 20:01.15 | florz | drmessano: what do you mean by "mixed representation"? |
| 20:01.43 | drmessano | florz: The difference between remixing the beatles catalog from a single track cassette or a 16 track master |
| 20:02.05 | florz | drmessano: well, an in concrete terms? |
| 20:02.05 | florz | +d |
| 20:02.16 | drmessano | ? |
| 20:02.52 | giancast | There is someone who can help me with a E-1 issue? |
| 20:03.00 | datacompboy-reco | giancast: what problem? |
| 20:03.08 | drmessano | giancast: Didnt we go over this earlier? |
| 20:03.34 | florz | drmessano: which signals are there "mixed in" in your VoIP data stream that's not mixed into the data stream you get from the hybrid directly? |
| 20:04.22 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 20:05.04 | drmessano | florz: You're still assuming we're using a digital signal.. EC must be applied to a sample of the analog signal to be effective. |
| 20:06.07 | florz | drmessano: you mean, like, ECs used by telcos are analog devices? |
| 20:06.21 | drmessano | Nope |
| 20:07.12 | drmessano | giancast: Stop messaging me |
| 20:08.10 | giancast | ok. |
| 20:08.13 | florz | drmessano: then, the "sample of the analog signal" is a "digital signal"? |
| 20:08.57 | datacompboy | florz: i think, i'm understand now what's wrong. We have not PCM samples, but G.711. |
| 20:09.05 | datacompboy | G711 are good codec, but it loozy. |
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| 20:09.29 | florz | "PCM samples" are "lossy", too |
| 20:09.31 | drmessano | datacompboy: SIP doesnt echo |
| 20:09.34 | datacompboy | Yep, agree: quality of echo cancel will be worst, than on hardware level. But it still possible. |
| 20:09.51 | datacompboy | drmessano: forget about SIP! Echo caused by ANALOG part of circuit |
| 20:10.05 | drmessano | If you hear echo on SIP, its the phones themselves or the provider |
| 20:10.19 | drmessano | So change out the phones or change providers |
| 20:10.26 | datacompboy | drmessano: ok, forget that thread completely. :) |
| 20:10.48 | datacompboy | florz: yes, but it 16 bit for all signal, linear. while G711 is non-linear, and only 8bit. |
| 20:11.38 | florz | datacompboy: no, "PCM samples" are not "16 bit" - actually, I'm not even sure about the "linear" part |
| 20:12.26 | florz | but anyhow, I don't think echo cancellers in telco networks are actually put before the compression happens? |
| 20:12.36 | datacompboy | florz: well, http://en.wikipedia.org/wiki/Pulse-code_modulation |
| 20:12.53 | datacompboy | florz: software canceller works before compression, in low-level drivers. |
| 20:14.46 | florz | datacompboy: that wikipedia article even explicitly mentions "A-law PCM", so PCM apparently doesn't imply "codepoints evenly distributed over voltage space" |
| 20:15.23 | florz | datacompboy: and I don't understand what you mean by "software canceller works before compression, in low-level drivers." |
| 20:16.55 | datacompboy | florz olsec etc are compiled into zaptel drivers, they not part of asterisk itself. |
| 20:17.23 | florz | datacompboy: well, yeah?!? |
| 20:17.46 | florz | zaptel usually handles compressed samples only |
| 20:18.00 | datacompboy | florz: http://www.rowetel.com/ucasterisk/oslec.html |
| 20:18.13 | florz | ISDN hardware, obviously, doesn't have any uncompressed samples |
| 20:19.28 | *** join/#asterisk steve745 (n=steve@pool-96-243-156-36.tampfl.fios.verizon.net) |
| 20:19.29 | datacompboy | florz: hmm... yes, you right. they operate over G.711 samples... |
| 20:19.32 | *** part/#asterisk steve745 (n=steve@pool-96-243-156-36.tampfl.fios.verizon.net) |
| 20:20.04 | *** join/#asterisk steve745 (n=steve@pool-96-243-156-36.tampfl.fios.verizon.net) |
| 20:20.56 | steve745 | am i in the right room for dial plan help |
| 20:21.27 | datacompboy | steve745: best room for dialplan help is room where you can edit your dialplan and reload it in asterisk. |
| 20:21.52 | steve745 | that really makes no sense |
| 20:21.57 | florz | and AFAIK the same is true for telco ECs, which is why you in principle could just as well do EC behind a SIP/RTP connection |
| 20:22.52 | datacompboy | florz: yep, that why i have started that question -- is there any software ready-to-use cancellers, that can be used over good-quality RTP stream. |
| 20:23.04 | lesouvage | steve745: I guess it is mend as a funny comment. Yes this is the room to be if you are in dialplan pain. |
| 20:23.15 | steve745 | ok |
| 20:23.41 | florz | actually, it's pretty sure that this is true for telco ECs - there wouldn't be much of a point to having multi-millisecond EC tails when it would have to handle the echo caused by some local loop only ... |
| 20:23.42 | lesouvage | steve745: don't take all the gurus here to serious ;-) |
| 20:24.22 | *** join/#asterisk brezular (n=brezular@adsl-d45.84-47-9.t-com.sk) |
| 20:24.25 | *** part/#asterisk debuggerboy (n=debugger@117.196.162.73) |
| 20:24.48 | florz | datacompboy: well, I don't know any - just that there shouldn't be any reason why it couldn't work in principle |
| 20:25.00 | steve745 | i'm trying to make an outbound call which i have don't with a call file and play message and when 1 is pressed i can get to dial external # ok biut want to send it to another sip ext when i put sip/100 for it to dial it hangs up the call |
| 20:26.40 | steve745 | <---- nueb to asterisk |
| 20:26.57 | datacompboy | steve745: post your dialplan to pastebin and show it |
| 20:27.01 | lesouvage | steve745: can you pastebin what you have now then we can go from there. |
| 20:27.17 | lesouvage | ~pastebin |
| 20:27.18 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:27.19 | steve745 | pastebin??? in channell |
| 20:27.44 | datacompboy | florz: in principe, yes. but in realty... i have no time for that work now :( faster to change provider, even if other provider requires telecom license to use it. :) |
| 20:28.26 | *** join/#asterisk jblack (n=jblack@pool-71-173-11-92.sctnpa.east.verizon.net) |
| 20:29.22 | steve745 | pasted |
| 20:29.38 | *** join/#asterisk shinao1 (n=shinao1@41.219.216.210) |
| 20:29.39 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) |
| 20:29.40 | lesouvage | steve745: can you show us the link |
| 20:30.05 | steve745 | http://pastebin.com/d7b1e5a44 |
| 20:30.26 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com) |
| 20:30.46 | florz | datacompboy: a somewhat hacky workaround could be to use some form of loopback ethernet devices and then use TDMoE |
| 20:32.14 | datacompboy | florz: yep, may be two cards will be faster... will see tomorrow, when support says something... i'm currently just finished complete flash player for asterisk-recorded files:) one problem left is playback for twin files (when -in and -out in two separate files) |
| 20:32.31 | lesouvage | steve745: you have a dialplan that should work but I don't think you need the timeout() lines. And you have onyone option in the menu (2). If an other option is pressed the dialplan will end. If ou add an i (illegal) extensions with a goto to the menu then you fix this. |
| 20:33.23 | lesouvage | steve745: and you need the recording for the voicemenu. |
| 20:33.40 | lesouvage | I'm in typo mode |
| 20:33.51 | steve745 | when i use sip/7275551212 it dial the external # but not the internal ext |
| 20:34.12 | steve745 | the recording is there |
| 20:35.16 | giancast | Yeahhhhhh |
| 20:35.19 | giancast | I did it |
| 20:35.21 | giancast | it´s upppp |
| 20:35.29 | giancast | I don´t know how I did it but it works |
| 20:35.35 | giancast | it´s running. |
| 20:35.41 | giancast | but is not really stable. |
| 20:35.46 | steve745 | thats what i told myself when i first got asterisk up |
| 20:35.59 | giancast | Can i post my zaptel and zapata |
| 20:36.09 | giancast | My E-1 is not really stable. |
| 20:36.15 | giancast | the alarm still in REC. |
| 20:36.40 | datacompboy | giancast: pastebin all info, do not write in channel. and pack your messages in one long instead of several short |
| 20:37.32 | lesouvage | steve745: Fom what I understand you want to add an option that gives the caller the option to enter a number that has to be called, is that correct? |
| 20:37.57 | steve745 | no to just press 1 to transfer |
| 20:38.31 | steve745 | system calls out plays recording and if called party wants more info press 1 to reach internal ext |
| 20:38.35 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation <--- |
| 20:38.53 | [TK]D-Fender | steve745: I do not see a "100" they can dial in your PB |
| 20:38.58 | drmessano | Nice paragraph that reiterates EC at the hybrid vs far end |
| 20:39.33 | *** join/#asterisk doubletoker (n=doubleto@adsl-17-234-27.jax.bellsouth.net) |
| 20:39.40 | lesouvage | justs add 1,1,Playback(<name of info file>) as the menu option 1 |
| 20:39.43 | giancast | with a paste bin you mean transfer a .txt file? |
| 20:39.54 | Qwell | wow, wtf did I miss Friday? |
| 20:40.20 | florz | drmessano: which part specifically do you mean? |
| 20:40.38 | datacompboy | drmessano: i know that. the only problem, if no packet loss, is jitter and timing. if no jitter (or corrected) it possible in theory with adaptive echo canceler. but, no code ready to use -- and that problem:) |
| 20:40.53 | datacompboy | good night everyone... there 2:40am :D |
| 20:41.07 | Blackvel | who has time (laters <1,5 hours) to test direct ip call and inbound/outbound voice? msg me |
| 20:42.01 | steve745 | will adding that playback line resolve the system hangup on the transfer |
| 20:42.02 | Qwell | infobot: log |
| 20:42.03 | infobot | [log] http://logs.nslu2-linux.org/livelogs/openmoko-cdevel/ for #openmoko-cdevel only. Maybe you meant "~logs" ? |
| 20:42.06 | Qwell | infobot: logs |
| 20:42.07 | infobot | All conversations are logged to http://ibot.rikers.org/channel, where "channel" is replaced by the URL-encoded channel name, such as %23freenode for #freenode. Lines starting with spaces are not logged. |
| 20:43.44 | lesouvage | steve745: it will play the message of your choice. You have to add an i extension like the way you add the t. something like i,1,Goto (s,5) so after choosing a non existing menu option the menu will be presented again. |
| 20:44.31 | [TK]D-Fender | steve745: What "transfer"? There is no 100 that you can dial in there. |
| 20:44.59 | steve745 | in where i'm lost i'm not sure what to do i'm a neub |
| 20:45.19 | [TK]D-Fender | steve745: You see how to make other exten in there.. did didn't MAKE ONE for anything you've discussed |
| 20:45.46 | [TK]D-Fender | steve745: exten => 100,1,NoOp(Holy crap do something) |
| 20:46.34 | lesouvage | [TK]D-Fender: I think steve745 tries to use the proper words but as he said, he is a newbe. |
| 20:47.33 | [TK]D-Fender | lesouvage: He didn't make an exten with a pattern to match what he wants to dial. And then there is the question about what he wants it to do. |
| 20:47.47 | Blackvel | steve745: did you do a tail -f /var/log/asterisk/messages while playing around? prolly it gives you the right idea what could be wrong (sometimes i miss that myself) |
| 20:48.12 | [TK]D-Fender | Blackvel: Whats wrong is there is no extension to matchw hat he's dialing. |
| 20:48.20 | [TK]D-Fender | Blackvel: And its been stated rather clearly |
| 20:48.29 | Blackvel | yes |
| 20:49.21 | Blackvel | i understand that. and prolly steve745 too. if he looked in console or messages he probably would have found out for himself (and i am only suggesting that becoz i missed that myself 1-2 times) |
| 20:49.56 | steve745 | is still kinda lost but getting closer to finding my way |
| 20:50.30 | steve745 | it makes good sense i'm just dumb and not too sure how i need to write the dial plan |
| 20:51.13 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:51.14 | [TK]D-Fender | steve745: exten => 100,1,Dial(SIP/steve) ; Wow, now when I dial 100 it doesn't hangup up and calls a SIP peer I defined! |
| 20:51.21 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 20:51.39 | [TK]D-Fender | steve745: Make an exten. Tel it step by step what you want it to do. |
| 20:51.52 | [TK]D-Fender | steve745: Its a simple as that. Get some sit-down time with the book... |
| 20:51.54 | [TK]D-Fender | ~book |
| 20:51.55 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:52.46 | *** join/#asterisk centrex (n=spectrum@209.168.156.25) |
| 20:53.05 | centrex | regarding em_w, can asterisk send a wink, or is it only capable of receiving? |
| 20:59.17 | *** join/#asterisk pyite_mac (n=dschreib@unaffiliated/pyite) |
| 20:59.26 | *** part/#asterisk pyite_mac (n=dschreib@unaffiliated/pyite) |
| 21:03.53 | steve745 | i'm never going to figure this out |
| 21:03.59 | warden | anyone having a hard time getting a tdm400p working w/ dahdi 2.2.0 |
| 21:04.31 | warden | i cant get dahdi to properly configure it |
| 21:06.06 | warden | if anyone wants to try to help me out here is a link showing the error i am getting |
| 21:06.07 | warden | http://pastie.org/718784 |
| 21:06.34 | lesouvage | steve745: yes you will! spend time trying to make a dialplan, read the book and try to do some stuff with asterisk again and within short time you will be able to make useful dialplans. |
| 21:07.47 | tzafrir | warden, what's the output of lsdahdi ? |
| 21:07.55 | [TK]D-Fender | steve745: I just gave you a single line to dial a device.... whats so hard about this? 1 more line lets you fall through to voicemail. Another line lets you jump base on time of day. Its the lowest and simplests for of programming. Tell it the # you're looking to match. Perform step 1, then 2, then 3, etc. 1 instruction at a time. |
| 21:08.56 | [TK]D-Fender | warden: Where do we see you even DEFINING your channels? |
| 21:09.36 | warden | yes it is |
| 21:09.58 | warden | umm shouldnt it pick up my card before defining my channels? |
| 21:11.03 | warden | No hardware timing source found in /proc/dahdi, loading dahdi_dummy |
| 21:11.36 | [TK]D-Fender | warden: Shouldn't what pick up your card? |
| 21:12.20 | [TK]D-Fender | warden: You are showing only the output of "dahdi show channels" effectively |
| 21:13.57 | warden | ok.. maybe i am not understanding correctly and i apologize for that.. but what does. No hardware timing source found in /proc/dahdi, loading dahdi_dummy |
| 21:14.08 | drmessano | I would think steve745 could figure this out faster than a lot of us |
| 21:14.11 | drmessano | I mean, he's got FIOS |
| 21:15.20 | warden | lol |
| 21:15.27 | *** part/#asterisk kkemp (n=kkemp@c-24-16-143-69.hsd1.wa.comcast.net) |
| 21:16.09 | *** join/#asterisk exitstate (n=xorl@xorl.xen.prgmr.com) |
| 21:16.27 | exitstate | quick question I have pbx_spool loaded, but it doesn't appear to be grabbing outgoing/*.call files? |
| 21:16.59 | [TK]D-Fender | drmessano: "going nowhere, mighty fast" |
| 21:17.21 | [TK]D-Fender | exitstate: Check your datestamps & eprmissions. Also how are you putting them there? |
| 21:17.23 | exitstate | i have vvvvvvvvr up and I don't even get a warning module show shows it's loaded |
| 21:17.37 | exitstate | [TK]D-Fender: using 'mv' |
| 21:18.06 | tzafrir | warden, what's the output of: dahdi_hardware; lsdahdi |
| 21:18.20 | exitstate | [TK]D-Fender: I don't even get warnings in the asterisk console which I would expect it to see it? |
| 21:18.34 | warden | pci:0000:00:0c.0 wctdm- e159:0001 Wildcard TDM400P REV E/F |
| 21:18.46 | [TK]D-Fender | exitstate: Check the timestamp. If its in the future * won't grab them |
| 21:19.05 | exitstate | in the past now |
| 21:19.42 | [TK]D-Fender | exitstate: check permissions |
| 21:20.01 | [TK]D-Fender | warden: rerun dahdi_cfg -vvvv |
| 21:20.42 | exitstate | [TK]D-Fender: owned by asterisk:asterisk 755 |
| 21:20.48 | exitstate | , perm 755 |
| 21:21.18 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 21:21.22 | DelphiWorld | hi |
| 21:21.27 | [TK]D-Fender | exitstate: mv them in, then out again, and make sure your CLI is on debug 10, verbose 10 |
| 21:21.32 | DelphiWorld | could anyone help me connect my nokia E65 to asterisk? |
| 21:22.03 | warden | [TK]D-Fender: ill have to work on this more when i get home later right im at work and its not letting me ssh to it right now ill run that whne i get home and work on it from there.. thanks for your help though |
| 21:23.10 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:23.50 | exitstate | [TK]D-Fender: yeah, it doesn't appear to be processing the spool at all |
| 21:25.54 | exitstate | with pbx_spool use count = 0 |
| 21:26.19 | hluesea | anyone could answer me ? i troubled on interconnecting another sip server without g729 codec. I use the gsm but other side is using g729 and i have to change my codec. In that point do I need g729 codec licencing ? and if yes, how much channel have i to ? [other sip server send me more than 5 channel at a same time] |
| 21:26.32 | *** join/#asterisk yoshx (n=yoshx@88-138-190-92.adslgp.cegetel.net) |
| 21:26.55 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:27.27 | tzafrir | warden, try: modprobe wctdm |
| 21:27.36 | warden | ok |
| 21:27.39 | warden | will do |
| 21:28.09 | warden | again it will be later on because right now i cant ssh to it for some reason (wife probably turned the thing off) :/ |
| 21:28.45 | tzafrir | warden, I have a feeling you simply don't have the modules installed. |
| 21:29.18 | tzafrir | What distribution is it? What kernel? |
| 21:29.20 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 21:31.01 | warden | its actually trixbox 2.8.0.2 |
| 21:31.10 | warden | i dont have the kernel version handy |
| 21:31.31 | warden | your probably right though.. i dont mess around w/ this to much.. last time i changed my configuration i still was using zaptel |
| 21:32.07 | [TK]D-Fender | hluesea: Yes, you need the licensed codc. |
| 21:32.07 | [TK]D-Fender | codec |
| 21:32.24 | hluesea | how much i need do you prefer any amount ? |
| 21:32.45 | hluesea | i am really new at this point and i am confused |
| 21:32.57 | exitstate | This is quite frustrating |
| 21:33.18 | [TK]D-Fender | hluesea: You need 1 license per simultaneous channel you intend to transcode |
| 21:33.55 | [TK]D-Fender | exitstate: Check your asterisk.conf to verify the folder its really loking in |
| 21:33.59 | warden | tzafrir: ill try the modprobe see what i get.. and ill see where to go from there.. at this point i cant do a whole lot since i ccon to the machine |
| 21:34.16 | warden | but i really appreciate your help |
| 21:34.22 | exitstate | astspooldir correct? |
| 21:34.37 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 21:35.12 | [TK]D-Fender | exitstate: yes |
| 21:35.31 | exitstate | /var/spool/asterisk |
| 21:35.44 | exitstate | it's the right dir, the modules loaded, nothing shows in debug/verbose (both set to 10 as request) |
| 21:36.26 | [TK]D-Fender | exitstate: reload the module from CLI and see if anything wierd happens |
| 21:36.30 | hluesea | TK D-Fender : thank you and |
| 21:37.02 | exitstate | Module 'pbx_spool.so' does not support reload |
| 21:37.36 | exitstate | ah got it |
| 21:40.05 | *** join/#asterisk yoshx (n=yoshx@88-138-190-92.adslgp.cegetel.net) |
| 21:40.34 | centrex | I have an em line that receives winks and responds back to them just fine, but when it is dialing out, it does not send a wink, it acts like a normal em immediate start line |
| 21:40.44 | centrex | Its on the latest 1.6.2 svn, and its configured for em_w |
| 21:40.48 | centrex | any ideas? |
| 21:48.24 | centrex | answer or be destroyed |
| 21:56.50 | ChannelZ | Perhaps you need some eyedrops |
| 21:58.20 | tzafrir | just noticed bash completes parameters for ./configure |
| 21:58.27 | Blackvel | ok |
| 21:58.39 | Blackvel | you can give me a direct ip call to test inbound/outbound voice (nat)? |
| 21:58.46 | Blackvel | who can... |
| 21:59.31 | tzafrir | ./configure --<tab><tab> |
| 22:05.49 | centrex | On this em line, when a call comes in one t1, and goes out the other t1 (both em), there is no ringing sent to the incoming caller. They call, it comes in from the pstn on t1 #1, then goes out to the second t1. The call goes through, but it doesn't ring. IF I put a ringing() application in the dialplan before it dials, then it rings once, then starts calling out the second t1 and stops |
| 22:05.54 | centrex | em_w I mean |
| 22:06.03 | centrex | Any way to fix that? |
| 22:06.10 | ChannelZ | Blackvel: I maybe can |
| 22:06.29 | centrex | Would it be related to the problem from before, where we're not sending a wink to the second t1? |
| 22:07.21 | ChannelZ | tzafrir: hmm nothing happens here -- bash for sure? |
| 22:07.41 | tzafrir | Do you have bash programmable completion? |
| 22:07.54 | ChannelZ | well apparently not :) |
| 22:08.15 | tzafrir | for instance, does 'cd' only complete names of directories? |
| 22:08.49 | tzafrir | or ssh complete files from remote hosts (if possible) ? |
| 22:08.54 | tzafrir | scp, actually |
| 22:10.51 | centrex | =( |
| 22:10.55 | *** join/#asterisk ticoit (n=ticoit@201.191.190.125) |
| 22:12.53 | ChannelZ | I've only ever used tab completion for the filesystem (files, dirs) |
| 22:19.33 | tzafrir | ChannelZ, try: . /etc/bash_completeion |
| 22:20.35 | ChannelZ | ahh. Interesting. |
| 22:22.00 | ChannelZ | there's an /etc/bash_completion.d too for expansions |
| 22:22.11 | ChannelZ | I wonder why this doesn't run by default |
| 22:24.07 | ChannelZ | I need to put this on a T-shirt: |
| 22:24.08 | ChannelZ | <datacompboy> G711 are good codec, but it loozy. |
| 22:32.43 | *** join/#asterisk unspin (n=unspin@96.49.129.159) |
| 22:33.15 | unspin | heya |
| 22:33.43 | unspin | could someone recommend an approach to debugging choppy audio with Asterisk 1.6? |
| 22:34.06 | unspin | it happen using the meetme conference module with SIP clients |
| 22:34.08 | unspin | s* |
| 22:34.32 | unspin | the audio kind of blips several times a second |
| 22:34.55 | unspin | to the point where its unusable |
| 22:35.20 | unspin | this is one a very new server, dual quad 2.4ghz Intel E5530 |
| 22:35.30 | unspin | running Centos 5.4 64 bit |
| 22:35.57 | unspin | no load whatsoever |
| 22:36.42 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 22:41.24 | *** part/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
| 22:44.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 22:51.56 | Blackvel | unspin: do you have zaptel installed? |
| 22:52.21 | *** join/#asterisk yoruk_ (n=yoruk@host45-243-dynamic.24-79-r.retail.telecomitalia.it) |
| 22:53.25 | Blackvel | unspin: as far was i know you need a timing device for working meetme!!! |
| 22:53.31 | Blackvel | as far was |
| 22:53.33 | Blackvel | as |
| 22:53.34 | Blackvel | damn |
| 22:53.38 | *** join/#asterisk wolfeyes (i=291175e1@gateway/web/freenode/x-vgvgxembgqerowmd) |
| 22:54.09 | wolfeyes | Is Mr Kroon in the house? |
| 22:57.02 | drmessano | He's out buying scones with Mr Basketknickers |
| 22:57.05 | drmessano | :( |
| 23:09.03 | *** join/#asterisk citywok (n=chatzill@c-71-231-179-213.hsd1.wa.comcast.net) |
| 23:10.09 | citywok | I've got a new issue with bandwidth.com sip provider. as soon as the call builds and Ack's, it sends a bye to bandwidth.com and the call terminates. |
| 23:10.23 | citywok | this is a new issue with 1.6.1.10 and the SVN tree. |
| 23:16.00 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 23:30.12 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 23:41.23 | citywok | ahh, known bug. ignoresdpversion=yes on the peer in sip.conf is the workaround. |
| 23:41.38 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) |
| 23:45.34 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 23:46.23 | *** join/#asterisk tzafrir (n=tzafrir@212.179.75.202) |
| 23:58.12 | Blackvel | cu |
