| 00:00.00 | ectospasm | but it's such an ugly hack |
| 00:00.08 | ManxPower | ectospasm: read up on it. It requires a Win32 box and a USB adapter. |
| 00:00.43 | ManxPower | I bet silvertip complains about me on the mailing lists. 8-) |
| 00:01.11 | ectospasm | I thought it required the Skype client for LInux, which required X Windows |
| 00:01.47 | ManxPower | perhaps it was changed. |
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| 00:06.01 | darkskiez | ManxPower: "You are doing the equiv of .." very good, made me laugh...maybe it was the whisky |
| 00:08.23 | ManxPower | ~zeeek |
| 00:08.24 | jbot | [zeeek] someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 00:08.27 | ManxPower | I always liked that one |
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| 00:24.55 | *** join/#asterisk Skty (n=nick@220-245-83-18.static.tpgi.com.au) |
| 00:28.02 | Skty | Morning all. Just a continuation to the question I asked last night. Basically I have a need to put an inbound caller on hold, call an external caller, then bridge the two calls effectivly. I can do this by putting the first caller into a meet me then transfer the second call into it. |
| 00:28.49 | Skty | The problem is I need to play a message to the party we called saying "You have an incoming call from X, press one to accept" then put them into the meet me. Easy enough - But what if they dont press one, I need a way to rip caller 1 back out of the meetme and throw them elsewhere |
| 00:28.53 | Skty | Any suggestions? :-) |
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| 00:47.33 | ManxPower | I guess it would depend on if your PSTN connection is PRI or VoIP or not. |
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| 00:52.04 | Skty | POTS. |
| 00:54.58 | ManxPower | Skty: POTS are considered answered as soon as Dialing is finished. |
| 00:58.33 | Skty | Thats fine - The question really is, is there a way from the dialplan to remove a caller from a meetme room? |
| 00:59.46 | ManxPower | forget about meetme. |
| 01:00.00 | ManxPower | set a music on hold class for the message. |
| 01:00.08 | ManxPower | use the M() option to Dial |
| 01:00.17 | ManxPower | and check the status of DIALRESULT |
| 01:01.45 | Skty | The problem is though - If the party we are calling does not answer it will go to voicemail, hence my need for some user interaction to accept the call. |
| 01:01.51 | *** part/#asterisk SpaceBassLaptop (n=ls@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 01:01.55 | Skty | (Ie. Voicemail on their mobile) |
| 01:05.00 | Skty | I guess if the party we are calling does not accept the call we can then use MeetMeAdmin to kick the first party out of the room, Id assume that the first caller will then just follow on in the dialplan |
| 01:06.25 | ManxPower | You need to timeout the dial before the call goes to voicemail |
| 01:06.46 | ManxPower | otherwise the voicemail will get a message " |
| 01:06.46 | ManxPower | Please press 1 to accept this call" |
| 01:07.05 | Skty | The time it takes for a call to go to voicemail however is dependant on the mobile carrier and totally out of my control. |
| 01:07.30 | ManxPower | Skty: I did not say it was going to be easy. |
| 01:08.15 | Skty | Nor did I think it would be, I walked into this project though assuming someone had written the ability to bridge two calls, apparently not yet :-) |
| 01:09.03 | ManxPower | Oh, I've done something similar with Asterisk. When someone gets a voicemail, a script generates a .call file then loops a prompt for the user to press a digit. |
| 01:11.23 | Skty | Yeah, I remember somewhere reading that FreePBX had put this in, however I was looking at a Trixbox install last night and couldn't see any mention of it. Even if I did I doubt it would be a very clean solution anyway (Sorry if your reading this Rob.) |
| 01:54.00 | famicon | eh |
| 01:54.14 | famicon | can anyone tell me what i would need to connect astrisk to the normal phone lines |
| 01:57.19 | De_Mon | famicon http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 01:57.43 | dan__t | hrm |
| 01:57.48 | dan__t | those Teliax guys are in Denver, right? |
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| 02:14.55 | deb_user | after upgrading to 1.4.11 from 1.4.0, everytime I try to dialout on a zap interface asterisk crashes with an ominous: segmentation fault. Anybody know what this means?? |
| 02:17.19 | De_Mon | heh |
| 02:17.47 | deb_user | De_Mon? |
| 02:19.30 | De_Mon | nothing, its just not much to go on |
| 02:19.39 | deb_user | De_Mon: I know |
| 02:20.13 | deb_user | but...this is a recent phenomenon, after upgrading zaptel driver and * |
| 02:20.22 | deb_user | there haven't been similar stories from other upgrades? |
| 02:20.34 | De_Mon | you're running debian unstable right? |
| 02:20.35 | deb_user | i notice the upgrade overwrote zaptel.conf |
| 02:20.41 | deb_user | De_Mon: nope, sarge |
| 02:21.04 | De_Mon | how did you install asterisk? |
| 02:21.23 | deb_user | i compiled from sourc |
| 02:21.24 | deb_user | i compiled from source |
| 02:23.13 | De_Mon | I installed source for 1.4.11 from unstable and did a dpkg-buildpackage and am not having any problems |
| 02:23.38 | deb_user | dpkg-buildpackage asterisk-1.4.11? |
| 02:23.47 | De_Mon | what do you see in /var/log/asterisk/messages |
| 02:24.59 | deb_user | oh geez |
| 02:25.06 | deb_user | how do I configure that? |
| 02:25.09 | deb_user | logger.conf? |
| 02:25.37 | deb_user | what a mess |
| 02:25.48 | deb_user | how do I purge my log file and start fresh? |
| 02:26.46 | De_Mon | beats me |
| 02:31.06 | deb_user | the command is logger rotate |
| 02:32.40 | deb_user | ERROR[2779] chan_zap.c: Unable to open channel 1: Device or resource busy |
| 02:32.41 | deb_user | here = 0, tmp->channel = 1, channel = 1 |
| 02:32.56 | deb_user | De_Mon: any insight on that error? busy? busy doing what? |
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| 02:56.50 | De_Mon | deb_user are you unable to unload the zaptel module and reload it? if asterisk isn't running it shouldn't be busy |
| 03:01.56 | deb_user | De_Mon: most diagnostics show the zaptel module checking out fine |
| 03:02.26 | deb_user | De_Mon: and I don't notice anything strange in the debug files, which I've jacked up to the highest level |
| 03:02.39 | deb_user | the system just crashes when I dial out on a zap channel |
| 03:02.48 | deb_user | let me check the other zap channels |
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| 03:05.20 | deb_user | bust...dialing out on zap channels crashes my * box |
| 03:05.24 | deb_user | weird |
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| 03:14.02 | De_Mon | so were you able to unload the module without asterisk running or not? |
| 03:14.11 | WhipsMcgee | what is a line appearance? |
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| 03:38.06 | deb_user | De_Mon: I downgraded to * 1.4.0 |
| 03:38.11 | deb_user | fixed the problem right up |
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| 03:44.20 | cavediver | Hi all. Simple question perhaps. But how do I get back to the shell after attaching the CLI with -r ? |
| 03:45.03 | cavediver | I've read the help but I can't seem to find any appropriate to use. |
| 03:46.58 | kiscokid | exit |
| 03:47.46 | cavediver | ah thanks. |
| 03:47.54 | cavediver | ctrl-c actually worked aswell. |
| 03:48.12 | cavediver | Didn't try that cause I thought it would just stop the deamon. |
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| 03:52.31 | JunK-Y | in the CLI, you can just type ! that will pompt u in the shell |
| 03:52.42 | JunK-Y | then do exit in the shell to return to the CLI |
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| 04:41.08 | Skty | Having a mental blank - Im putting a call on hold, I need to then execute a macro while said call is on hold, how can I make it start the macro :\ |
| 04:41.20 | Skty | Stupid long weekends + drinking = Death to brain. |
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| 04:57.36 | Skty | There has to be a way to do dialplan stuff in the background while a call is on hold.... I do it with the Alcatel systems constantly. This is annoying me. |
| 05:01.14 | Skty | Afaik I'm going to have to create a call file on the fly which then calls a macro, to execute a third call just to connect calls 1 and 2. It seems a little silly. |
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| 07:19.55 | jedaustin | I'm getting popping/clicking on a TDM400 FXO connection; I've made sure it's not sharing an IRQ, installed the newest zaptel/asterisk/etc (1.4). Could it be SATA drive causing it? |
| 07:20.09 | jedaustin | er connection=channel |
| 07:20.34 | jedaustin | I swapped the FXO module for another and it didn't change. |
| 07:29.23 | ectospasm | jedaustin: How many modules do you have on the card? |
| 07:30.12 | ectospasm | jedaustin: or do you have any unused module slots on the card? |
| 07:30.36 | jedaustin | ectospasm: 4, 1 FXO,3 fxs |
| 07:31.42 | ectospasm | Have you tried swapping the FXO with one of the FXS on the card (e.g. move FXO to slot 3 and move the FXS to slot 4( |
| 07:32.11 | jedaustin | I haven't tried that yet - worth a shot. Wouldn't it move the popping/clicking to the extension if that is the case? |
| 07:32.41 | ectospasm | depends on what's causing the pops/clicking |
| 07:33.42 | jedaustin | It's a Wildcard TDM400P REV I |
| 07:34.02 | jedaustin | I also have a REV F card but figured the newer one would be better |
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| 08:14.12 | sjobeck | hi all, anyway to set the Music on hold playlist to the exact order i want? i cant seem to figure what play order it uses when random is off. |
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| 09:38.43 | DarKnesS_WolF | mm 1.4 dosn't have ${TIMESTAMP} ? |
| 09:47.47 | mvanbaak | nope |
| 09:48.20 | mvanbaak | ${STRFTIME(${EPOCH},,%d-%m-%Y %H:%M:%S)} |
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| 10:01.03 | jozu | hi |
| 10:01.30 | jozu | i have a problem with outgoing calls |
| 10:01.40 | DarKnesS_WolF | damn it how can i add callerID from teh dialer ! |
| 10:02.14 | jozu | warning: chan_sip.c handle_response_invite received response: "forbidden" from "xxxx" |
| 10:02.40 | jozu | what this? |
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| 10:09.34 | DarKnesS_WolF | i'm 512 i'm calling 511 |
| 10:09.42 | DarKnesS_WolF | 511 = arg1 what dose 512 = ??? |
| 10:11.21 | Fluor_ | jozu, probably you need to authenticate with "xxxx" in order for it to handle your outgoing calls. |
| 10:13.36 | Fluor_ | DarKnesS_WolF, you can set your Outgoing CAllerID with 'Set(CALLERID(number)=512' in your extension just before you Dial().. |
| 10:16.27 | DarKnesS_WolF | Fluor_: yes but i want it to be dynamic |
| 10:16.41 | DarKnesS_WolF | cuz i have lots of extensions |
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| 10:18.12 | jozu | Fluor_ i put the Setcallerid option... is the secret pass or the sip number from the trunk? |
| 10:18.33 | Fluor_ | 'secret' is a password. |
| 10:20.11 | DarKnesS_WolF | Done |
| 10:20.29 | DarKnesS_WolF | ${CALLERID(NUM)} |
| 10:21.51 | Fluor_ | ${CALLERID(number)} would contain the CLI when in some incoming context, yes. |
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| 10:23.58 | jozu | i put in extension.conf, context default SetCallerID with pass or sip number but nothing |
| 10:24.07 | jozu | handle_response_invite "forbidden" |
| 10:25.36 | jozu | :? |
| 10:28.25 | jozu | whats the problem? |
| 10:29.29 | jozu | i put in SetCallerID my secret pass, my sip provider number and noting |
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| 10:41.44 | jozu | any idea, please? |
| 10:42.53 | tengulre | hi,all |
| 10:43.02 | tengulre | happy chinese day! |
| 10:49.15 | Fluor_ | jozu, you should put that information in your sip.conf in a line looking like 'register => user:pass@sipproviderip/context' |
| 10:50.57 | jozu | i already registered in sip |
| 10:51.36 | jozu | but.. i put register => user:pass@sipproviderip |
| 10:51.47 | jozu | no /context |
| 10:51.56 | jozu | is neccesary? |
| 10:55.12 | jozu | i put user:pass@sipproviderip/default |
| 10:55.25 | jozu | but.. more errors |
| 10:55.43 | jozu | set(callerid) error and the forbidden error |
| 10:57.47 | Fluor_ | http://www.voip-info.org/wiki/view/Talkin2ya follow that, but replace the necessary information. |
| 10:59.40 | Fluor_ | What does your Dial() call look like? (Should be someting like Dial(SIP/${ARG1}@yourprovider)) |
| 11:00.09 | Fluor_ | then you need to have a [yourprovider] section in your sip.conf |
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| 11:41.11 | jozu | Fluor_, i have the register line, i have the [provider] section in sip.conf |
| 11:41.23 | jozu | and i have the [default] in the extension.conf |
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| 11:42.08 | jozu | i tried multiple parameters, but nothing, response: "forbidden" |
| 11:43.41 | jozu | is a normal configuration |
| 11:45.05 | jozu | handle_response_invite "forbidden" from "mysip" |
| 11:45.17 | jozu | in all sips 101,100, |
| 11:46.28 | dongs | how about insecure=no |
| 11:46.30 | dongs | or =yes |
| 11:46.31 | dongs | or hwatever. |
| 11:46.58 | jozu | in users extensions? or in [provider] parameters? |
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| 11:48.15 | dongs | i think in extensions. |
| 11:48.19 | dongs | 101, 100 etc. |
| 11:51.51 | jozu | nothing, i put insecure=no in 100 but.... "forbidden" appear |
| 11:52.09 | jozu | i put insecure=no in [myprovider] at sip.conf.. and nothing |
| 11:57.12 | dongs | =yes |
| 11:57.13 | dongs | not no |
| 11:57.17 | dongs | sry. |
| 12:04.26 | jozu | same error, with insecure=yes and no |
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| 13:02.38 | dongs | i guess youre out of luck then |
| 13:20.46 | *** join/#asterisk kindor (n=arie@even.niet.nu) |
| 13:21.34 | kindor | how can i have asterisk call back the extension, where a call was put on hold if a person is holding for X seconds? |
| 13:25.52 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.52) |
| 13:40.52 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
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| 14:07.59 | *** join/#asterisk jozu (n=torrent@81.Red-88-5-190.staticIP.rima-tde.net) |
| 14:08.05 | jozu | hi |
| 14:11.22 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.52) |
| 14:12.01 | jozu | a question: i have a gsmgateway, when call it i give DISA for dmtf detection |
| 14:12.49 | jozu | its possible to call via gsm to the gsmgateway with the phonenumber+dmtf? |
| 14:12.52 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:15.45 | *** join/#asterisk _matt (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 14:16.42 | [TK]D-Fender | jozu, you can call anywhere your dialplan is configured to let you |
| 14:16.58 | Fluor_ | i don't understand exactly what you want to do. do you want asterisk to generate DTMF-tones? |
| 14:17.55 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 14:19.39 | jozu | i call the gsmgateway (6XXXXXX), the gsmgateway giveme second dial tone and y press (96xxxxxx) |
| 14:19.59 | jozu | its possible to call 6xxxxx+96xxxxxx from the mobile? |
| 14:20.35 | Fluor_ | maybe you can if you enter the number like 6xxxxx,,,96xxxxx |
| 14:20.46 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.33) |
| 14:20.56 | jozu | 3 ,,,? only? |
| 14:21.26 | Fluor_ | , adds a pause, i don't know how many you'll need ;-) |
| 14:21.58 | rob0 | My cat has four paws. So try ,,,, |
| 14:22.03 | jozu | mmm... cool |
| 14:22.05 | [TK]D-Fender | jozu, Show us how you are calling the gateway in your dialplan right now. |
| 14:22.07 | Fluor_ | lol :) |
| 14:22.20 | [TK]D-Fender | PASTEBIN it please |
| 14:22.21 | [TK]D-Fender | ~pb |
| 14:22.22 | jbot | A Pastebin is a place to paste your stuff 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 |
| 14:22.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 14:22.36 | jozu | but.. i put an IVR onto the gsmgateway... say 1 office 2 DISA... |
| 14:22.46 | jozu | and now? 6xxxxx,2,96xxxxx? |
| 14:24.36 | jozu | or 6xxxxx296xxxxx? |
| 14:27.09 | [TK]D-Fender | jozu, stop and show us what you've done in your dialplan so far so we can suggest something. |
| 14:28.37 | jozu | no dial plan, only 0|. to exit to sip voipprovider |
| 14:29.15 | jozu | i dont need dialplan... maybe NXXXXXXXXXXXX |
| 14:29.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:30.35 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.46.105.176) |
| 14:30.46 | [TK]D-Fender | jozu, So you haven't actually tried anything? |
| 14:31.52 | linagee | what the hell? www.grandstream.com is actually hosted on yahoo.com??? |
| 14:31.56 | linagee | grandstream-- |
| 14:31.58 | jozu | i tried all, but i dont know who's dtmf work in gsm |
| 14:32.23 | linagee | does it really suprise anybody? |
| 14:33.02 | [TK]D-Fender | jozu, that depends on your GATEWAY. How is it connected to *? |
| 14:33.24 | jozu | it's registered in a sip extension on the asterisk |
| 14:33.29 | linagee | if their host is yahoo, who are their programmers? children?? |
| 14:34.40 | [TK]D-Fender | jozu, Well try each of the 3 more popular DTMF modes and find out. |
| 14:35.06 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 14:35.58 | *** join/#asterisk xezz (n=phob@athedsl-18845.home.otenet.gr) |
| 14:36.10 | xezz | hello, suppose i have a ringing phone with extension 200, but im at another room with a phone/extension 205, can i pick up the call from 205 ? |
| 14:36.18 | xezz | by pressing some combination like #200 orsomething ? |
| 14:36.49 | *** join/#asterisk puga (n=Puga@201-68-26-194.dsl.telesp.net.br) |
| 14:37.19 | Fluor_ | afaik, that's not possible |
| 14:37.34 | Fluor_ | though there's some group pickup addon for SCCP phones |
| 14:38.02 | [TK]D-Fender | xezz, Sure, go look at "call pickup" on the WIKI for the list of ways. |
| 14:38.10 | Fluor_ | oh ;) |
| 14:38.23 | xezz | call pickup |
| 14:38.26 | xezz | okie thanx |
| 14:42.27 | *** join/#asterisk Corydon76-home (i=one@pdpc/supporter/bronze/Corydon76-home) |
| 14:42.27 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:43.04 | *** join/#asterisk Corydon76-dig (i=orange@pdpc/supporter/bronze/Corydon76-home) |
| 14:43.04 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 14:43.36 | linagee | does anyone have an aastra 9112i? |
| 14:45.04 | *** join/#asterisk jozu (n=torrent@81.Red-88-5-190.staticIP.rima-tde.net) |
| 14:47.55 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 14:48.00 | jozu | [TK]D-Fender.. when you call to check your voicemail, and voicemail say (press 1, for read mssgs, 2 for xxxx), and the voicemail have sip extension 200.... |
| 14:48.13 | jozu | you can call 200,1,2 ? |
| 14:48.47 | [TK]D-Fender | jozu, Voicemail doesnt' have an "extension". What are you talking about? You are making no sense at all |
| 14:51.59 | jozu | a moment.. please |
| 14:54.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:57.24 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 14:57.24 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 14:57.30 | khronos | Maybe jozu's talking about the default 8500 extension in the sample config. |
| 14:58.38 | [TK]D-Fender | khronos, never try being psychic with a schizo..... |
| 14:59.28 | [TK]D-Fender | khronos, And no... |
| 14:59.34 | linagee | snom 300 or aastra 9133i? (they both cost around the same) |
| 15:00.32 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:02.11 | [TK]D-Fender | linageeWhere are you located? |
| 15:03.14 | khronos | Hmm, k. |
| 15:04.50 | linagee | [TK]D-Fender: san diego |
| 15:05.12 | [TK]D-Fender | linagee : then NEITHER. Polycom IP320 kills both of the, |
| 15:05.17 | linagee | [TK]D-Fender: i'm trying to replace two gs POS phones that both went south after "upgrading" :( |
| 15:05.18 | [TK]D-Fender | them* |
| 15:05.39 | linagee | [TK]D-Fender: polycom based on where i'm located? huh? |
| 15:06.10 | [TK]D-Fender | linagee : Yes. In North America there is no point going with anything else. |
| 15:06.37 | linagee | [TK]D-Fender: polycom makes those cool little "three legged starfish" conference room devices, right? |
| 15:06.41 | [TK]D-Fender | linagee : in a lot of overseas places Polycom's import comes at a steep price. |
| 15:06.48 | [TK]D-Fender | linagee : Yes, those too. |
| 15:10.12 | [TK]D-Fender | linagee : www.telephonydepot.com |
| 15:10.35 | linagee | [TK]D-Fender: polycom soundpoint 330? |
| 15:10.56 | [TK]D-Fender | linagee : only if you need the pass-through |
| 15:11.50 | linagee | [TK]D-Fender: wow. they paint a gs barbietone-101 black and try to make it look professional. :( |
| 15:12.05 | linagee | it's like a piece of crap in a tuxedo |
| 15:12.09 | [TK]D-Fender | linagee : Who? |
| 15:12.13 | Qwell_ | heh, one of the gs guys came up to us at astricon |
| 15:12.20 | *** join/#asterisk puga (n=Puga@201-68-26-194.dsl.telesp.net.br) |
| 15:12.23 | Qwell_ | practically begging us to fix a SIP bug |
| 15:12.35 | Qwell_ | it was funny |
| 15:12.53 | Qwell_ | he was complaining because we were spamming the phones with NOTIFYs, and it was literally killing the phones |
| 15:12.59 | linagee | Qwell_: did you know their website is hosted off yahoo.com? |
| 15:13.01 | Qwell_ | they just couldn't handle the load (it wasn't that much) |
| 15:13.04 | Qwell_ | linagee: I did not |
| 15:13.11 | linagee | Qwell_: i was like, HUH? |
| 15:13.19 | Qwell_ | not yahoo.cn? |
| 15:13.30 | linagee | Qwell_: as i said before though, i'm guessing it doesn't suprise you guys much? |
| 15:13.39 | Qwell_ | nope |
| 15:13.41 | linagee | Qwell_: if their site is hosted on yahoo, where are their programmers hosted? children? |
| 15:13.55 | Qwell_ | chinese sweat shop |
| 15:13.59 | khronos | Day care. |
| 15:14.05 | Qwell_ | khronos: same thing |
| 15:14.36 | linagee | [TK]D-Fender: wow, that's awesome. that website beats ebay for polycom prices. :) |
| 15:15.05 | linagee | $87. interesting |
| 15:15.07 | [TK]D-Fender | linagee : many do. www.digiumcards.com |
| 15:15.20 | linagee | [TK]D-Fender: so i could have had something ten times better than a piece of crap for only $7 more? :( |
| 15:16.01 | [TK]D-Fender | linagee ; That'll teach you for shopping without good advisors |
| 15:16.28 | linagee | [TK]D-Fender: thanks! :) |
| 15:17.17 | linagee | ac adapter sold seperately? hrm |
| 15:19.31 | [TK]D-Fender | linagee : They are PoE native. IP 430/501 have the AC, 430 does PoE as well, 501 needs a special cable. |
| 15:21.42 | [TK]D-Fender | linagee : You know what happens when you cheap out.... |
| 15:23.56 | linagee | [TK]D-Fender: is there a PoE injector that has a 10/100 switch built in? |
| 15:24.29 | [TK]D-Fender | linagee : Not that I know of. Do you need a pass-through for PC? |
| 15:24.31 | linagee | er, like a 2 port switch |
| 15:24.41 | linagee | [TK]D-Fender: exactly. but i can see a reason to not use the pass through |
| 15:24.49 | linagee | [TK]D-Fender: what if someone wants to pull the phone over closer |
| 15:25.01 | linagee | then you have these three wires hanging off the back of the phone. lol |
| 15:25.07 | linagee | power/eth/eth |
| 15:25.49 | [TK]D-Fender | no cheap way for that. |
| 15:25.59 | linagee | [TK]D-Fender: if it's poe, it's just one cord |
| 15:26.05 | [TK]D-Fender | that implies a PoE switch, and they cost more than you want to think about. |
| 15:26.05 | linagee | hmmm |
| 15:26.34 | linagee | i might already have two wires coming to the room. *rubs chin* |
| 15:26.59 | styelz | the price diff between buying the adaptor and getting a poe switch isnt that much per port really |
| 15:27.13 | styelz | depending on the quantity |
| 15:28.34 | linagee | styelz: 1 |
| 15:29.04 | styelz | heh |
| 15:29.09 | [TK]D-Fender | styelz, Yeah, it is actually... |
| 15:29.16 | linagee | styelz: and it would have me curios really why the price difference is so close. you think it's by mistake? lol |
| 15:30.21 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
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| 15:48.47 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
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| 16:00.54 | *** join/#asterisk Op3r (n=Op3r@210.4.60.88) |
| 16:01.24 | Op3r | does anyone believe echo is possible using voip? |
| 16:01.59 | jbroome | what? |
| 16:04.41 | tzafrir | Op3r, echo can be generated by the handset. |
| 16:04.54 | Op3r | yep |
| 16:04.54 | tzafrir | Or can come from the remote party |
| 16:05.08 | tzafrir | There are some ways echo can slip in |
| 16:05.09 | Op3r | its on the equipment not on the asterisk side right? |
| 16:05.17 | tzafrir | right |
| 16:08.03 | Op3r | oh ok |
| 16:09.44 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com) |
| 16:09.50 | jozu | hi,toall |
| 16:10.31 | jozu | i have an voicemail check extension in sip 600 |
| 16:10.47 | mltlnx | hello... Does any one have an opinion on Digium PCI or PCI-E cards. Whats the benefit of having the PCI-E version? |
| 16:10.50 | jozu | it's possible to call them using numer+dtmf tones? |
| 16:11.12 | jozu | example: 600+1+1+2+3 |
| 16:12.07 | [TK]D-Fender | jozu, shat is "sip 600"? |
| 16:12.09 | jozu | the same operation want to put on my gsm-gateway extension |
| 16:12.25 | jozu | in number 600 |
| 16:12.36 | [TK]D-Fender | jozu, You are still making no sense at all |
| 16:12.59 | jozu | [TK]D-Fender i probe the ,,, the *, the #, and nothing |
| 16:13.07 | [TK]D-Fender | mltlnx, It has the benifit of working in a PCI-E slot where you don't have another choice. |
| 16:13.59 | [TK]D-Fender | jozu, I think you are in real need of either a clue or a translator.... |
| 16:14.14 | jozu | yes.. :( |
| 16:15.40 | jozu | it is possible to mark to my extension of the mailbox of separate voice and the tones dtmf that it requires? |
| 16:15.47 | jozu | translation ;) |
| 16:16.32 | *** join/#asterisk maxdoubt (n=max@user-38lm3s0.cable.mindspring.com) |
| 16:16.59 | maxdoubt | what's a good sip registrar? i'm trying to make calls to malawi (via diamondcard), but am not having much luck? |
| 16:17.05 | [TK]D-Fender | jozu, Sorry, nobody in here can help you due to the language barrier. |
| 16:17.39 | [TK]D-Fender | jozu, Get a friend to help you express your ideas. |
| 16:17.59 | mltlnx | TK: Ah I see....But if I have both a PCI and a PCI-E there is no benefit in having the PCI-E card? |
| 16:18.07 | maxdoubt | sorry, for being off-topic asterisk stuff... just didn't know where else to ask for help for voip stuff |
| 16:19.29 | *** join/#asterisk smultron (n=lukas@cpe-67-9-146-21.austin.res.rr.com) |
| 16:19.53 | [TK]D-Fender | mltlnx, I'd say its a negative move in terms of tranferrability. |
| 16:20.23 | [TK]D-Fender | maxdoubt, go check out the WIKI for a list or terminators |
| 16:20.44 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 16:20.55 | mltlnx | TK: Sorry. Im a bit slow....So you would do the PCI of PCI-E? |
| 16:22.25 | maxdoubt | [TK]D-Fender, what's the link to the wiki? google not helping much |
| 16:26.20 | [TK]D-Fender | ~wikis |
| 16:26.21 | jbot | wikis is, like, http://www.voip-info.org |
| 16:26.28 | [TK]D-Fender | mltlnx, PCI |
| 16:26.39 | mltlnx | Thanks! |
| 16:28.47 | rob0 | PCI-E has higher bandwidth and maybe runs at more cycles per second, but that might not be a factor for voice (depending how many channels and how busy they are, of course.) |
| 16:29.03 | mltlnx | Does anyone use a Sangoma a101 in a 1U enclosure? The specs say 2u form factor. But the guys at ASA computers think it will fit? |
| 16:29.51 | rob0 | sounds like a job for a hacksaw ;) |
| 16:30.39 | [TK]D-Fender | mltlnx, the A101 2U factor is for a VERTICAL mounting. |
| 16:30.42 | rob0 | My brother-in-law actually did force a 40-pin IDE cable onto a 50-pin SCSI hard drive once. |
| 16:31.12 | [TK]D-Fender | mltlnx, Most 2U servers use FULL-HEIGHT PCI clsot mounted HORIZONTALLY which makes this irrelevant |
| 16:31.40 | mltlnx | Ah so as long as it mounts horizonally I should be ok? |
| 16:31.54 | [TK]D-Fender | mltlnx, virtually guaranteed it does |
| 16:32.12 | mltlnx | thanks. good info! |
| 16:32.26 | jbroome | ask ASA, we've ordered from them before and have been very helpful |
| 16:32.54 | mltlnx | jbroome: thanks |
| 16:39.24 | jozu | another question more simply: i have chain_sip.c handle_respones_invite:Recivied "Forbidden" from "xxx" <sipxx> |
| 16:39.36 | jozu | what this? |
| 16:40.42 | *** join/#asterisk Lusid (i=Lusid@c-98-203-23-191.hsd1.fl.comcast.net) |
| 16:41.00 | Lusid | Hey everyone... it has been over two years since I've used Asterisk in a production project. I have a need now to do it again, and was wondering if anyone can suggest a good provider that support MANY simulataneous outbound PSTN calls at a decent rate? |
| 16:42.29 | jbroome | jozu: sounds like you have the wrong auth info on your sip client |
| 16:42.30 | Lusid | I actually need to be able to dial outbound to two different PSTN numbers, and then conference them together. But there will probably be 50+ of these happening at any one time, starting out. |
| 16:45.28 | [TK]D-Fender | Lusid, Not sure you'll want to trust VoIP over the internet for this... I'd suggest PRI.... |
| 16:46.13 | Lusid | Mmmmm... starting out, I need it to be as cheap as possible. |
| 16:46.52 | sjobeck | hi, all, hope all is going smoothly. I am struggling with something that I feel must be missing something. With random off, for music on hold, how can i control the playlist, how can i control the precise order the songs/files are played in. It doesnt seem to go alphanumeric, or timestamp, or size, or anything else I can readily discern. |
| 16:48.44 | sjobeck | i simply presumed it was alphanumeric, but doesnt seem to be (ie: 1song.mp3, 2song.mp3, 3song.mp3 nor does it do Asong.mp3, Bsong.mp3, Csong.mp3 either). |
| 17:02.26 | *** join/#asterisk hellop (n=hellop@cpe-66-91-197-100.hawaii.res.rr.com) |
| 17:03.04 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:03.42 | riddlebox | what would cause a zap fxo channel to ring one extra time after a person hangs up |
| 17:06.20 | hellop | like, you go to answer it during the last ring, but it's dead? |
| 17:07.58 | riddlebox | no, the called party answers then a second later it rings in my ear one more time |
| 17:10.13 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 17:11.19 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:14.16 | sjobeck | any one have any advice for me about playing my music on hold files in a specific order? |
| 17:14.53 | hellop | sjobeck, people are on hold so long they listen to multiple songs? |
| 17:15.08 | sjobeck | y, they are in queues, sometimes for long time |
| 17:15.15 | riddlebox | doesnt it play them in alphabetical order |
| 17:16.07 | *** join/#asterisk zumbush (n=ztriver@81.25.80.131) |
| 17:17.09 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:17.37 | zumbush | my voicemail .txt files is getting permission 0060 when created... stopping me from accessing with FreePBX/ARI..any ideas what im doin wrong? |
| 17:17.53 | zumbush | sry permission 006 |
| 17:18.25 | zumbush | and owner is asterisk:asterisk |
| 17:19.12 | zumbush | just recompiled with svn-branch of 1.4..but didnt fix it |
| 17:20.18 | Strom_M | what user are you when you build asterisk? |
| 17:20.25 | zumbush | root |
| 17:20.57 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:20.59 | Strom_M | and this is voicemail.conf you're talking about? |
| 17:21.16 | zumbush | the sound files is getting correct permissions but not the txt |
| 17:21.22 | zumbush | yea |
| 17:21.57 | zumbush | not the conf file.. the msg00x.txt files in /var/spool/asterisk/voicemail/default/INBOX |
| 17:22.02 | Strom_M | ah |
| 17:22.37 | zumbush | -------rw- 1 asterisk asterisk 272 30 sep 19.14 msg0000.txt |
| 17:22.51 | Strom_M | asterisk is running as user "asterisk"? |
| 17:22.54 | sjobeck | riddlebox: no, cant seem to discern the order at all, cant figure it out, but alphanumeric isnt it |
| 17:22.56 | zumbush | yes |
| 17:23.33 | zumbush | uhh.. or maybee as root |
| 17:23.39 | zumbush | hmm how do i check that |
| 17:23.51 | Strom_M | ps aux | grep asterisk |
| 17:24.56 | zumbush | root 23979 0.0 0.0 52768 716 pts/1 S 19:13 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 17:24.56 | zumbush | asterisk 24044 0.0 0.4 57828 9228 pts/1 Sl 19:13 0:00 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c |
| 17:25.32 | zumbush | so is it as root or asterisk..hmm |
| 17:25.43 | Strom_M | no |
| 17:25.58 | Strom_M | safe_asterisk is running as root; asterisk itself is running as 'asterisk' |
| 17:26.03 | zumbush | oki |
| 17:26.08 | zumbush | thx |
| 17:27.27 | zumbush | so any ideas whats wrong? |
| 17:29.36 | zumbush | the sound files in /var/spool/asterisk/voicemail/default/xxx is getting good permissions but the txt files is only getting 006 |
| 17:30.38 | zumbush | root 23979 0.0 0.0 52768 716 pts/1 S 19:13 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 17:30.38 | zumbush | asterisk 24044 0.0 0.4 57828 9228 pts/1 Sl 19:13 0:00 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c |
| 17:30.46 | zumbush | sry wrong post |
| 17:30.57 | zumbush | -------rw- 1 asterisk asterisk 272 30 sep 19.14 msg0000.txt |
| 17:30.57 | zumbush | -rwxr-xr-x 1 asterisk asterisk 92844 30 sep 19.14 msg0000.wav |
| 17:30.58 | zumbush | better |
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| 17:37.59 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com) |
| 17:38.15 | lonekazoo | any aa50 pros around? |
| 17:39.34 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:43.26 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 17:45.54 | *** join/#asterisk kkn088 (n=kikoun@84.4.61.127) |
| 17:46.19 | hellop | zumbush, thats weird.. |
| 17:51.03 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 17:51.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:53.03 | zumbush | yea wonder if it has anything to do with the umask settings for asterisk user and txt files |
| 17:53.07 | zumbush | reading up on it now |
| 17:57.29 | *** part/#asterisk r- (i=rob@rob.tc) |
| 17:58.52 | *** join/#asterisk smultron (n=lukas@cpe-67-9-146-21.austin.res.rr.com) |
| 18:06.06 | hellop | zumbush, you have your profile umask, but you can also have one in fstab |
| 18:06.41 | zumbush | ok |
| 18:07.39 | rob0 | fstab?? That only applies for a non-Unix FS, and it would be silly to use a non-Unix FS for asterisk. |
| 18:08.31 | zumbush | [root@obelix apps]# umask -p |
| 18:08.32 | zumbush | umask 0022 |
| 18:09.02 | hellop | rob0, fstab is what tells linux what partitions you have. it is not for non-Unix FS... |
| 18:09.05 | zumbush | strange that the voicemail txt files get 006 permissions |
| 18:09.14 | zumbush | rather it should be 600 |
| 18:09.52 | hellop | rob0, you mean a umask entry in fstab is for non-unix fs? |
| 18:10.54 | rob0 | hellop: read ""man mount", see "FILESYSTEM SPECIFIC MOUNT OPTIONS", and note that umask only appears with non-Unix filesystem types. |
| 18:12.49 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 18:13.55 | *** join/#asterisk jedaustin (n=chatzill@wsip-66-210-241-251.ph.ph.cox.net) |
| 18:15.11 | hellop | rob0, gotcha |
| 18:15.48 | hellop | thanks for the clarification, so I guess it's only in the user's profiles. |
| 18:17.06 | zumbush | is the umask universal or can it be different for e.g. asterisk user |
| 18:17.39 | hellop | How do you guys have your asterisk starting? |
| 18:17.52 | hellop | From a script in rc2.d? (debian) |
| 18:18.02 | hellop | zumbush, it's diff for each user |
| 18:18.29 | hellop | specified in each user's profile, which is created first time they login |
| 18:18.46 | zumbush | i start it with amportal script |
| 18:19.19 | zumbush | from freepbx |
| 18:19.25 | hellop | # the default umask is set in /etc/login.defs |
| 18:20.41 | zumbush | k |
| 18:22.43 | hellop | Well, any suggestions on how I should start asterisk? I usually just start it manually as root. |
| 18:26.31 | hellop | Should I, use bg/screen to start it from a start script? Or, make it run on a different tty? |
| 18:30.18 | hellop | zumbush, did you look in asterisk.conf yet? |
| 18:35.46 | zumbush | for what? |
| 18:36.10 | Op3r | have anyone ever experienced echo when using voip? |
| 18:38.01 | riddlebox | what would cause a zap fxo channel to ring one extra time after a person answers |
| 18:41.39 | [TK]D-Fender | Op3r, Apparently only you believe this. And why would that be? How about actually telling us the exact scenario you're dealing with? |
| 18:49.09 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:50.53 | *** join/#asterisk ManxPower (n=manxpowe@ip68-11-78-112.no.no.cox.net) |
| 18:53.00 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.65) |
| 18:54.00 | *** join/#asterisk fcmarc (n=fcmarc@modemcable144.37-37-24.mc.videotron.ca) |
| 18:56.25 | *** part/#asterisk fcmarc (n=fcmarc@modemcable144.37-37-24.mc.videotron.ca) |
| 18:59.26 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 18:59.50 | jedaustin | Op3r: ping the voip provider if it is higher than 100ms it could be your internet conneciton's latency |
| 19:00.35 | jingles | hola, * folks. I have need of helping. |
| 19:01.13 | jedaustin | jingles: just ask :) |
| 19:01.40 | *** join/#asterisk fcmarc (n=test@modemcable144.37-37-24.mc.videotron.ca) |
| 19:01.52 | jingles | can do. I've got asterisk up and running, I've compiled and installed the asterisk-addons. |
| 19:01.52 | jedaustin | Op3r: I found in some cases I get lower latency to the east coast than the west coast even though I'm in AZ. |
| 19:02.06 | jingles | mysql is up, running, accessible. however, when * is doing things, it isn't logging calls. |
| 19:02.35 | jingles | no errors or anything - just doesn't log calls. |
| 19:02.58 | jingles | I created a user, made sure it was 'granted' all the permissions on just the asterisk database - but still no cdr logging. |
| 19:03.03 | zumbush | fixed the filepermission issue with the voicemail .txt files with adding externnotify=/usr/bin/voicemail_chmod.sh in voicemail.conf |
| 19:03.08 | zumbush | that way i can access/see them in the ARI portal, but im still not getting any emails delievered |
| 19:03.08 | zumbush | maybe the app to send email of recieved voicemail is run before the external notify script |
| 19:03.18 | jedaustin | jingles: where are you looking? Mysql or /var/log/asterisk? |
| 19:03.30 | jingles | looking for what? |
| 19:03.46 | jedaustin | jingles: for the cdr log |
| 19:03.51 | jingles | I'm in mysql while * is running. select * from cdr; returns no rows, even after I can see the call has been made. |
| 19:04.18 | jedaustin | jingles: look in /var/log/asterisk to see if it's logging to a flat file |
| 19:04.24 | jingles | ok. |
| 19:04.38 | jingles | it's going to take a few minutes - have to go switch the prod server for the dev server. |
| 19:04.50 | jingles | since * won't actually *start* without the PRI plugged in. |
| 19:07.29 | jingles | the cdr is the last part I need before the system is 'production ready'. |
| 19:08.11 | jingles | it's strange though. sometimes, the T1 card gets config'd for span 3 - even though every config file says span 1. |
| 19:08.27 | jingles | and /etc/init.d/zaptel restart causes it to restart on span 1. |
| 19:08.34 | jingles | I think it's just insane. |
| 19:10.35 | jingles | ok. good call - the cdr *is* logging to cdr-csv files. |
| 19:10.38 | jingles | where's that set up? |
| 19:10.44 | jingles | I kinda don't want it doing that. |
| 19:11.18 | *** join/#asterisk mr-at (n=mr-at@i577A0B6C.versanet.de) |
| 19:12.35 | jedaustin | I'm not positive but I think cdr_mysql.conf |
| 19:12.54 | jingles | let me log in and see what that file looks like. |
| 19:13.04 | jingles | the docs on this part weren't clear, and sparse at best. |
| 19:13.31 | jedaustin | Yeah, there is a bit of learning curve :) |
| 19:14.28 | *** join/#asterisk mr-at (n=mr-at@i577A0B6C.versanet.de) |
| 19:14.56 | jingles | well, I've gotten past a lot of stuff - lots of digging for info. |
| 19:15.00 | jingles | but this bit's been a stumper. |
| 19:15.22 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:18.53 | jedaustin | freepbx takes care of a lot of it |
| 19:19.53 | ManxPower | freepbx has a learning curve MORE complex than Asterisk for anything but the most simple stuff |
| 19:20.14 | jedaustin | Not so bad :) |
| 19:20.56 | jingles | well, I'm missing something pretty simple - I can tell. I just don't know *where* the thing I'm missing is. |
| 19:21.14 | jingles | since it's logging to the csv files, I can only assume I got the addons installed right. |
| 19:21.37 | ManxPower | noload cdr_csv.so in /etc/asterisk/modules.conf Assuming you are not running trixbox |
| 19:22.03 | jingles | oh ho! |
| 19:22.07 | ManxPower | cdr_mysql.conf sets up CDR logging in MySQL |
| 19:22.46 | ManxPower | It might work in trixbox too, but trixbox screws with all of the other config files -- I assume it messes with /etc/asterisk/modules.conf too |
| 19:23.26 | jedaustin | ManxPower: I jumped off the Trixbox ship about a month ago. |
| 19:23.49 | ManxPower | most people do |
| 19:24.38 | jedaustin | I need something to standardize installs on though so now I use admins paradise. Once I upgrade all of the components and add a few more it's what I would build from scratch. |
| 19:24.57 | jingles | ok! so, the cdr_mysql.so file isn't found on my system. |
| 19:25.05 | ManxPower | I have dialplan design, AGI scripts, and macros for my standard install |
| 19:25.08 | jedaustin | Probably my next jump will be a script to build it all from scratch |
| 19:25.35 | jingles | so, what didn't get compiled? was there a switch I missed in the addons file? |
| 19:26.05 | riddlebox | what would cause a zap fxo channel to ring one extra time after a person answers |
| 19:26.08 | ManxPower | jingles: cdr_mysql is not part of Asterisk because of licensing issues. |
| 19:26.15 | ManxPower | look for asterisk -addons, it should be in there. |
| 19:27.01 | jingles | right. I'm in that directory now. |
| 19:27.07 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 19:27.12 | jingles | and it seems like the mysql stuff is *there* |
| 19:27.27 | jingles | I've compiled this directory a number of times. |
| 19:27.45 | ManxPower | when MySQL changed from LGPL to GPL it became incompatible with Digium's license |
| 19:28.06 | jedaustin | I'd love if more projects switched to postgresql |
| 19:28.35 | ManxPower | If Postgress is BSD or LGPL licensed I'll bet there is already a postgress logging driver in asterisk |
| 19:29.11 | ManxPower | There is an ./apps/app_sql_postgres.c |
| 19:29.17 | ManxPower | that's not for logging, of course. |
| 19:29.26 | ManxPower | most people seem to use ODBC for that sort of stuff. |
| 19:29.38 | styelz | jedaustin: you need mysql client dev files to compile cdr_mysql from addons |
| 19:29.41 | ManxPower | It's not like you need to support hundreds of transactions per min |
| 19:30.32 | Corydon76-dig | and even then, ODBC will usually work just fine, even with that kind of load |
| 19:31.06 | jingles | there we have it. missing the mysql dev libraries. |
| 19:31.40 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.153.79) |
| 19:33.09 | jingles | installing them right now. will recompile the addons. bet this starts working instantly. |
| 19:33.10 | jingles | ;) |
| 19:33.17 | zumbush | does this look like a recent svn branch --> Asterisk SVN-branch-1.4-r84135 ? |
| 19:33.35 | jingles | then I'll put the prod box back up, pull the special resolv.conf into my docs, and zappy! all done. |
| 19:33.38 | jingles | man! what a pain. |
| 19:36.14 | rob0 | hellop: No need for screen(1), although I used to do it that way. But you can start asterisk, then start a screen and open a console therein: "asterisk -rvvv". (Use "v"'s to taste.) |
| 19:38.33 | rob0 | hellop: I adjust the astctlgroup and astctlpermissions in the [files] section of asterisk.conf, so I can attach to a console from my user account. |
| 19:48.25 | hellop | rob0 hmm I'll look into it. |
| 19:48.34 | jingles | boo on ubuntu. |
| 19:48.54 | jingles | I guess I have to do a clean install of it *again* |
| 19:48.55 | hellop | rob0 I usually run ast as root, because anything else causes a bunch of errors... |
| 19:49.17 | hellop | after I gave the asterisk user write access to the log dirs, I got a bunch of zaptel related errors. |
| 19:49.34 | hellop | You're not supposed to run * as root right? |
| 19:50.14 | rob0 | I do. It's probably not the best way to do it, but I haven't suffered a disaster yet. |
| 19:50.52 | hellop | rob0, you don't need screen anyways, because you can just "asterisk -r" to bring it to current console right? |
| 19:51.10 | rob0 | Basically if you're not using privileged ports, and the asterisk user has permissions to the zaptel (if applicable) devices, you can be nonroot. |
| 19:51.21 | rob0 | correct |
| 19:51.38 | hellop | After you do -r how do you send it back to the ether? |
| 19:51.54 | Buhntz | exit |
| 19:52.00 | hellop | oh |
| 19:52.10 | hellop | always wanted to know that... |
| 19:52.16 | Buhntz | :) |
| 19:52.39 | Buhntz | the server keep runnin until you enter stop now |
| 20:02.17 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 20:03.48 | Op3r | oh sorry |
| 20:06.28 | Op3r | [TK]D-Fender: the reason why I ask a question about voip + echo is because I got into some pissing fight with a dude who claimed it is. I kept on telling him it is his headset. here's the exact exchange of words http://www.eflo.net/VICIDIALforum/viewtopic.php?t=4417 |
| 20:07.16 | Strom_M | well, duh, it's on the vicidial forums |
| 20:07.23 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 20:07.28 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:08.06 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 20:08.26 | *** join/#asterisk saftsack (n=saftsack@pD9E05E0A.dip.t-dialin.net) |
| 20:17.38 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:18.43 | *** join/#asterisk d00gster (n=doughant@69.156.106.185) |
| 20:21.42 | sjobeck | hi all, a good sunday to you. Is any one playing their music on hold music files in a SPECIFIC order, an exact pre-set order? I can not seem to get it to play mine in the order I need to. Random is off. There is no discernible order to their play order. |
| 20:21.58 | *** join/#asterisk saftsack (n=saftsack@pD9E05E0A.dip.t-dialin.net) |
| 20:22.19 | [TK]D-Fender | sjobeck, use a streaming source that you control for MoH. The internals & MPG123 options won't give you what you want |
| 20:22.54 | sjobeck | d-fender: ah, hmmm, ok, not bad. any suggestions? |
| 20:23.38 | sjobeck | I also wonder if I ought to create a bounty to add an option for this in native mode, since it would be quite helpful. I like native, nothing else to fuss with, you know, higher performance, etc, etc, etc, etc. |
| 20:25.09 | citats | sjobeck: if you add the files to the directory in the order you want them played it should work how you want |
| 20:25.25 | citats | sjobeck: depending on the filesystem in use i guess, but generally that should be the way it works |
| 20:27.20 | sjobeck | citats: thx. will try that, but ....... how weird is that? I'm not sure I luv the idea of having to remove all files, then add them back in again in the order I want, each time I need to tweak something, you know?, since rerecording a prompt in the sales queue or support queue happens from time to time, you know. I mean, I her ya, thanks, decent suggestion, but I'm just saying the community, I am betting, would appreciate the "orde |
| 20:27.58 | sjobeck | I thought it was alphanumeric, hoping it was alphanumeric, then all we all need to do is name the files: 1.mp3, 2.mp3, 3m.mp3, etc |
| 20:28.00 | citats | sjobeck: i suspect an alphanumerical sort option would be pretty easy to implement |
| 20:28.28 | sjobeck | then my users could do this themselves :) |
| 20:28.58 | sjobeck | d-fender: any suggestions on good easy solid external stream source? |
| 20:36.11 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 20:38.23 | delmar | Hi everyone. My server was rebooted last night and now Asterisk isn't working right. For every sip peer or device i'm getting errors like WARNING[27943]: chan_sip.c:1775 __sip_xmit: sip_xmit of 0x81a9a98 (len 514) to 123.123.123.123:5060 returned -2: Bad file descriptor. Anyone have any idea whats going on here? |
| 20:39.20 | *** join/#asterisk blq (n=Bl@dslb-088-064-152-071.pools.arcor-ip.net) |
| 20:45.56 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 20:50.12 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 20:50.41 | sjobeck | hey, all, whats best external source for streaming audio for MoH ? |
| 20:51.01 | sjobeck | or, how do you stream the local radio station in to your MoH ? |
| 20:51.28 | sjobeck | not via physical local headphone jack |
| 20:58.30 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-94-123-110.dsl.hstntx.swbell.net) |
| 20:58.45 | delmar | sweet. figured out my problem....... since this channel gets logged and ends up searchable the details are; |
| 20:59.50 | delmar | chan_sip.c:1775 __sip_xmit: sip_xmit of 0x81a9a98 (len 514) to 123.123.123.123:5060 returned -2: Bad file descriptor - caused by MythTV's Front end starting before Asterisk. MythTV has a Phone module which conflices. |
| 21:00.26 | delmar | now, if there had been a better message to indicate something else was bound to 5060 when Asterisk was trying to, I wouldnt have spent the last hour trying to figure it out. |
| 21:01.22 | delmar | but the cocky admin closed the ticket at http://bugs.digium.com/view.php?id=7381 saying "not an asterisk problem". I beg to differ... it IS an asterisk problem in that it lacks a descriptive enough error message. |
| 21:06.35 | dan__t | Agreed. |
| 21:08.33 | sjobeck | delmar: agreed. try to contact bug person to add that comment to the bug report, maybe someone can/will reopen it. |
| 21:08.48 | delmar | yeah. |
| 21:09.03 | delmar | there must be something else tho.. surely it would fail to bind 5060 and report that .. somewhere. |
| 21:09.16 | file | it does when chan_sip.so is loaded |
| 21:09.24 | delmar | file, yeah .. looking |
| 21:09.51 | delmar | but for most of us.. the console scrolls away for many pages and we never see that. |
| 21:10.12 | delmar | yep. bingo. here it is [Oct 1 09:49:52] WARNING[1014]: chan_sip.c:16947 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use |
| 21:11.35 | lonekazoo | anyone here familiar with the embeded OS on the asterisk appliance? |
| 21:11.37 | [TK]D-Fender | delmar, 1st guess : running a soft-phone on your server perhaps? |
| 21:12.00 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 21:19.06 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 21:22.17 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 21:22.58 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 21:24.42 | hrmphh | any idea why i might not be able to dial out my second zap group? http://pastebin.com/mf6ad651 = zapata.conf |
| 21:25.18 | hrmphh | [Sep 30 14:25:11] WARNING[4264]: channel.c:3099 ast_request: No channel type registered for '' |
| 21:25.24 | hrmphh | [Sep 30 14:25:11] WARNING[4264]: app_dial.c:1099 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented) |
| 21:28.48 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:31.07 | Strom_M | hrmphh: paste the relevant Dial() line in your extensions.conf |
| 21:31.21 | hrmphh | nm i made a typo sorry |
| 21:31.26 | hrmphh | T1RUNK instead of T1TRUNK :) |
| 21:31.33 | hrmphh | any idea how i set my outgoing callerid? |
| 21:31.43 | hrmphh | its just sending the 4 digit extension now (which i believe is default behavior) |
| 21:32.02 | Strom_M | no, it sends the caller ID of the originating channel |
| 21:32.16 | Strom_M | you can use the CALLERID() function |
| 21:32.36 | hrmphh | yeah sorry the 4 digit was the originating |
| 21:33.06 | hrmphh | context ael-trunkld { _91NXXNXXXXXX => { Dial(${T1TRUNK}/${EXTEN:${T1TRUNKMSD}},120,t); }; }; |
| 21:33.21 | hrmphh | so i should CallerId() before the Dial()? |
| 21:33.38 | Strom_M | CALLERID is a function, not an application |
| 21:33.44 | Strom_M | Dial is an application |
| 21:34.17 | hrmphh | k |
| 21:34.25 | hrmphh | some docs are saying to SetVar CALLERIDNUM instead |
| 21:34.38 | hrmphh | (http://www.voip-info.org/wiki/view/CallerID) |
| 21:34.40 | hrmphh | reading that now |
| 21:34.58 | Strom_M | sure, you can do that if you're using 1.0 and the year is 2004 |
| 21:35.07 | hrmphh | lol |
| 21:35.19 | hrmphh | which doc should i be checking? |
| 21:35.20 | Strom_M | but now you use the CALLERID() function, not any of the now-deprecated variables |
| 21:36.13 | hrmphh | k |
| 21:36.24 | hrmphh | do you recommend asteriskdocs.org for the most up to date documentation? |
| 21:36.48 | Strom_M | you could just use the CLI |
| 21:36.58 | Strom_M | "core show function CALLERID" |
| 21:36.59 | hrmphh | 'help' or something? |
| 21:37.00 | hrmphh | ah |
| 21:39.26 | *** join/#asterisk mihinomenest (i=X90s@66.255.220.17) |
| 21:41.15 | hrmphh | stupid question: how do you call functions (as opposed to applications)? |
| 21:41.45 | Strom_M | oh fine, i'll just tell you the answer |
| 21:41.54 | hrmphh | ah 'Set' |
| 21:41.55 | hrmphh | nm |
| 21:42.01 | Strom_M | before you run Dial, you want to Set(CALLERID(num)=3115552368) |
| 21:42.08 | hrmphh | http://www.asteriskguru.com/tutorials/calleridnum_function.html |
| 21:42.09 | hrmphh | found that |
| 21:42.13 | hrmphh | needed a good spot for docs |
| 21:42.19 | hrmphh | the pdf on digium's site is useless |
| 21:42.28 | Strom_M | have you read the book? |
| 21:43.00 | hrmphh | i have the asterisk ora book |
| 21:44.34 | Strom_M | have you read it though? :) |
| 21:45.57 | hrmphh | hah first few chapters |
| 21:46.01 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 21:46.06 | hrmphh | any idea how long i can make CALLERID(name)? |
| 21:46.15 | hrmphh | or even if telco will take it from pbx? |
| 21:46.37 | Strom_M | what kind of connection are you on? |
| 21:46.44 | Strom_M | where are you? |
| 21:46.49 | Strom_M | and which telco are you using? |
| 21:47.13 | hrmphh | T1 ISDN PRI |
| 21:47.16 | hrmphh | L3 |
| 21:47.19 | hrmphh | is the telco |
| 21:47.42 | Strom_M | and where are you |
| 21:47.46 | hrmphh | california |
| 21:48.31 | Strom_M | you /might/ be able to get away with it, but typically CNAM is looked up on the terminating end of the call |
| 21:48.40 | hrmphh | k |
| 21:48.41 | hrmphh | setting it anyway |
| 21:48.50 | hrmphh | i can try calling an analog of ours i guess |
| 21:48.55 | Strom_M | yeah |
| 21:49.09 | Strom_M | set it to something improbable, like BALLS INCORPORATED |
| 21:49.17 | Strom_M | or LOL HAI2U |
| 21:49.22 | hrmphh | dammit |
| 21:49.25 | hrmphh | "Unknown Name" |
| 21:49.27 | hrmphh | is what comes up |
| 21:49.33 | jingles | awesome. |
| 21:49.37 | hrmphh | on the AT&T POTS line |
| 21:49.37 | Strom_M | yep, then it's not settable |
| 21:49.38 | jingles | mysql stuff up and running. |
| 21:49.52 | Strom_M | again...typically CNAM is looked up on the terminating end of the call |
| 21:49.55 | hrmphh | yeah |
| 22:00.56 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-66-122-34-11.dsl.sntc01.pacbell.net) |
| 22:01.51 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
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| 22:04.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:04.39 | tzanger | hmm, wasn't atacomm the guy who was developing the quad-span that did transcoding en-route to the processor? |
| 22:06.00 | file | aye |
| 22:06.07 | tzanger | there he is |
| 22:06.12 | tzanger | so what's your favourite martini? |
| 22:06.18 | file | purple haze |
| 22:06.22 | tzanger | I never pictured you a martini type to begin with, but hey |
| 22:06.41 | file | what did you picture my type as being? :D |
| 22:08.05 | tzanger | well you're an easterner... I would have said the nastiest, grittiest whisky you can buy, but then I met you |
| 22:08.18 | file | meh |
| 22:08.29 | tzanger | after meeting you, I think I'd have imagined you with a snifter of really high-end scotch |
| 22:08.50 | tzanger | ... that, and an albino cat which you would stroke thoughtfully as you plotted to take over the world |
| 22:09.16 | tzanger | we should have a repo of this information somewhere |
| 22:09.28 | tzanger | asterisk developers, their pics and fave drinks/food/etc |
| 22:09.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:09.31 | tzanger | I wonder what we could mine out of that |
| 22:10.16 | file | get to work on that |
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| 22:11.36 | johnadsfsdfdf | anyone know how i can buy a vanity toll free number and port it to my voip provider? |
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| 22:11.59 | delmar | [TK]D-Fender, as I said, MythTV has a phone module. |
| 22:12.09 | delmar | all fixed now. |
| 22:13.50 | tzanger | file: done |
| 22:13.59 | tzanger | file: purple haze martini |
| 22:14.02 | tzanger | next? |
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| 22:16.39 | SplasPood | In asterisk 1.4 is there any way to prevent Dial(SIP/user1&SIP/user2) from spawning 1 bunch of CDR entries, 1 for every OK but not answered channel... ? Seems it used to only spawn 1 entry |
| 22:20.57 | SplasPood | it also seems to forget the CDR(accountcode) for these NO ANSWER records |
| 22:21.46 | puzzled | tzafrir_home: ping |
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| 22:34.54 | SplasPood | ahh... possible bug, found the report |
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| 22:39.01 | mltlnx | Any recommendations on simple 1U server for use with 15-20 users....I will be install a Sangoma T1 card in it. |
| 22:41.55 | hrmphh | good choice on the card |
| 22:41.59 | hrmphh | i just put in an a101d-x |
| 22:42.01 | hrmphh | its teh win |
| 22:42.08 | hrmphh | def get the echo cancel |
| 22:43.17 | mltlnx | hrmphh: any suggestions on a 1U? Im thinking supermicro. |
| 22:47.11 | [TK]D-Fender | mltlnx, good choice |
| 22:47.12 | jedaustin | Supermicro has a good rep in asterisk circles |
| 22:47.34 | mltlnx | Thanks guys! |
| 22:47.55 | jedaustin | You just have to make sure that 1) that the PCI card will fit in it and 2) that if it needs power for fxs that it has molex connectors |
| 22:48.14 | puzzled | jedaustin: recently there was someone that reported he had issues with a Digium card and a Supermicro board |
| 22:48.31 | jedaustin | Weird |
| 22:48.46 | puzzled | yes, card would not work. iirc he put it in a Dell and no problems |
| 22:49.02 | [TK]D-Fender | And to take that all in context : he's going to be using a SANGOMA card which does not NEED a molex. All null & void. |
| 22:49.06 | puzzled | so be careful when getting a Supermicro board for a Digium card |
| 22:49.16 | [TK]D-Fender | And tehre were a few SM boards that Digium's have issues with |
| 22:49.43 | [TK]D-Fender | puzzled, Better worded as be careful of ANY board you want to use with a Digium card. |
| 22:49.50 | jedaustin | Yeah |
| 22:50.13 | jedaustin | I've heard good things about Rhino cards too |
| 22:50.28 | jedaustin | I have two TDM400 cards.. constant tinkering :( |
| 22:50.47 | hrmphh | tdm400 = junk |
| 22:50.50 | hrmphh | we have one |
| 22:51.16 | jedaustin | I wish they had an automatic gain control circuit so that we didn't have to blindly monkey with rx/txgain |
| 22:51.36 | jedaustin | Anyone used the astribank? |
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| 22:52.32 | [TK]D-Fender | I've never heard anything conclusive of Rhino's PCI cards.... |
| 22:54.26 | jedaustin | Is there something iffy about using sata drives with asterisk? |
| 22:55.49 | jedaustin | I'm getting popping and clicking I didn't get when it had an IDE drive.. maybe not related, not sure. |
| 22:56.19 | hrmphh | popping and clicking = digium card |
| 22:56.25 | hrmphh | replace w/sangoma and you'll have no issue |
| 22:56.33 | hrmphh | we had same prob |
| 22:56.37 | jedaustin | Yeah it's a digium TDM400 card. Maybe I'll try the other one |
| 22:56.42 | hrmphh | dont bother |
| 22:56.48 | hrmphh | we've gone through several TDM400s |
| 22:56.51 | hrmphh | they're all worthless |
| 22:56.56 | jedaustin | Now I don't feel so bad :) |
| 22:56.59 | hrmphh | i actually emailed digium and they RMA'ed it for a nicer card |
| 22:57.00 | hrmphh | for free |
| 22:57.25 | jedaustin | That is good customer service |
| 22:57.40 | hrmphh | yeah well they were losing me as a hardware customer otherwise :) |
| 22:57.47 | jedaustin | Maybe Digium should buy Sangoma next :) |
| 22:58.01 | hrmphh | yeah they should scrap the TDM400 line at least for sure |
| 23:00.06 | jedaustin | I am definitely sick of tweaking it to get it to work; never worked completely right |
| 23:00.24 | jedaustin | Any issues with built in echo cancellation and faxes? |
| 23:01.38 | jedaustin | Which Sangoma card is equivalent to the TDM400? |
| 23:01.56 | mltlnx | a200 remora |
| 23:02.08 | [TK]D-Fender | "d" if you know whats good for you.... |
| 23:02.23 | mltlnx | that ec? |
| 23:02.34 | [TK]D-Fender | yup |
| 23:03.34 | mltlnx | Hey TK, what do you think of RAID 1 via md (native linux sw raid)? |
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| 23:04.10 | [TK]D-Fender | mltlnx, If you're going to do something, do it RIGHT. HARDWARE <- |
| 23:04.48 | mltlnx | i hear that....thanks |
| 23:07.19 | jedaustin | Would the a20101 have echo problems as bad as the digium TDM400? |
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| 23:10.48 | jedaustin | Wow! Almost twice as much for the A20101D as the A20101 |
| 23:11.12 | kiscokid | jedaustin: its worth it |
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| 23:11.21 | mltlnx | I second that! |
| 23:11.44 | kiscokid | unless you like echo |
| 23:13.35 | rob0 | unless you like echo |
| 23:13.49 | Qwell_ | jedaustin: You should try HPEC |
| 23:14.06 | jedaustin | Never heard of it :) |
| 23:14.13 | Qwell_ | is your tdm400p new? |
| 23:14.21 | jedaustin | No |
| 23:14.25 | Qwell_ | how old? |
| 23:14.34 | jedaustin | About 2 years.. never worked right |
| 23:14.47 | Qwell_ | tried calling support? |
| 23:15.05 | kiscokid | anyone know if the dialplan function sendurl() works with Polycom phones? |
| 23:15.07 | jedaustin | :) nope, I bought it from voip supply, does that matter? |
| 23:15.17 | Qwell_ | no |
| 23:15.41 | Qwell_ | when you call, ask them about HPEC |
| 23:17.30 | jedaustin | That new? they don't list it on their site |
| 23:18.31 | Qwell_ | it's on there |
| 23:18.39 | Qwell_ | it's a few months old |
| 23:18.52 | Qwell_ | http://www.digium.com/en/products/software/hpec.php |
| 23:19.09 | jedaustin | OH.. High Performance Echo Cancellation |
| 23:19.57 | jedaustin | Now I get it.. HPEC=software echo cancellation |
| 23:20.23 | jedaustin | How well does it work? |
| 23:21.02 | jedaustin | If it works well they should include it automatically |
| 23:21.31 | Qwell_ | it works very well |
| 23:21.46 | Qwell_ | it's not free - though, if you have a card within warranty, you can get it free |
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| 23:26.28 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com) |
| 23:26.28 | jedaustin | Only $10 (I'm only using one FXO channel, the rest is voip) |
| 23:27.03 | *** join/#asterisk ReDNeQ (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 23:27.15 | Qwell_ | jedaustin: yeah, but if it's within warranty, you won't have to pay a dime.. call sales tomorrow, and inquire about it :) |
| 23:27.20 | jedaustin | Would the HPEC addition fix the popping/clicking> |
| 23:28.10 | ReDNeQ | hello |
| 23:28.14 | jedaustin | ? |
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| 23:28.27 | Qwell_ | jedaustin: I believe so |
| 23:34.50 | WilliamK | anyone here ever used delegate for proxying stuff? |
| 23:39.32 | hrmphh | dont think so |
| 23:39.39 | hrmphh | popping/clicking is not echo |
| 23:39.51 | hrmphh | its incompatability w/mobo, timing issues, etc. |
| 23:40.07 | hrmphh | if you run that tool itll show you the %s and w/the digium cards its way off |
| 23:40.48 | jedaustin | Which tool? zttool? |
| 23:41.11 | hrmphh | yup |
| 23:43.27 | jedaustin | Sure you don't mean zttest? |
| 23:44.30 | jedaustin | zttool just shows the digium 400p card. I can do the loop test but it doesn't display anything useful. |
| 23:44.45 | jedaustin | zttest: Best: 100.000000 -- Worst: 99.987793 -- Average: 99.989583 |
| 23:45.18 | Qwell_ | jedaustin: call support tomorrow too. if you have any issues, especially as they relate to timing, they will be fixed |
| 23:45.31 | jedaustin | Will do |
| 23:47.24 | jedaustin | This board has a dual core pentium D processor |
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