| 00:04.47 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 00:05.15 | MrTelephone | How do you choose a ring cadence with the Dial() cmd? |
| 00:05.31 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:05.58 | MrTelephone | Yes I already tried to look.. is the only way to do it through the ALERT_INFO in the sip header? |
| 00:06.13 | thx2000 | Thats the only way I know to do it |
| 00:12.38 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:12.38 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) Addons 1.2.8, 1.4.4 (Oct. 16, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php |
| 00:13.19 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 00:13.51 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 00:19.42 | *** part/#asterisk killfill (n=killfill@pc-164-134-45-190.cm.vtr.net) |
| 00:27.04 | k31th | anyone alive? |
| 00:27.07 | peanut- | no |
| 00:27.35 | *** part/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 00:27.59 | *** join/#asterisk anujsingh (n=anuj@59.94.131.126) |
| 00:28.47 | anujsingh | hello |
| 00:29.02 | k31th | olleh |
| 00:29.25 | anujsingh | is there someway that i can record and retrive conversation of any user? |
| 00:30.17 | anujsingh | retriving conversation by dialing extension numbe of specific user? |
| 00:30.36 | anujsingh | similar to VoiceMail application? |
| 00:30.48 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 00:31.36 | grimsy | anujsingh: have you looked at http://www.voip-info.org/wiki-Asterisk+record+calls ? |
| 00:32.19 | grimsy | also, depending on the local laws, you'll probably need peoples consent before doing it |
| 00:32.56 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:33.34 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 00:42.37 | anujsingh | thanks grimsy i am looking threw the link you given |
| 00:43.06 | grimsy | np. haven't actually looked at it too hard myself, so not sure how much help it'll be |
| 00:46.53 | *** join/#asterisk javb (n=javb@190.80.224.20) |
| 00:47.20 | javb | i`m getting "Asterisk died with code 1. Automatically restarting Asterisk", any ideas? |
| 00:59.40 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:01.00 | anujsingh | grimsy what about dialing an extension , similar to Voicmail application and if it can play the recorded file of the particular extension/user |
| 01:01.18 | grimsy | no idea sorry |
| 01:02.29 | *** join/#asterisk abritton (n=abritton@207-255-210-182-dhcp.cbe.md.atlanticbb.net) |
| 01:03.15 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 01:06.52 | *** join/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 01:07.04 | jsoftw | Im having dramas with one sip trunk not getting thru the firewall properly. |
| 01:07.37 | jsoftw | I have an almost identical trunk which uses the exact same firewall rules and it works fine. But this other trunk just aint playing the game. |
| 01:13.23 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:20.16 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 01:20.29 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-132-60.neo.res.rr.com) |
| 01:23.15 | *** join/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
| 01:25.09 | *** join/#asterisk axscode (n=axscode@103.211.111.218.klj04-home.tm.net.my) |
| 01:30.30 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.244) |
| 01:31.13 | *** join/#asterisk dlynes (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 01:41.12 | Micc | does asterisk 1.4 show actionID with new channel events? |
| 01:41.35 | Micc | or have some other way to associate originate with a channel? |
| 01:43.51 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:45.43 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:08.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:13.21 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 02:13.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:16.29 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:16.29 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 02:24.31 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 02:27.40 | javb | A Sangoma A200D, with 2 FXO Modules says, "(4 ports)" does it means that with 2 FXO ports i can get 4 analog lines? |
| 02:28.24 | russellb | i'd be happy to answer questions about digium cards :-p |
| 02:30.56 | [TK]D-Fender | javb, No. It means that each module IS 2 ports |
| 02:31.44 | javb | [TK]D-Fender: ? |
| 02:31.47 | [TK]D-Fender | javb, You just inverted your sentence and tried calling "modules" ports half way through. |
| 02:35.48 | blitzrage | russellb: !!! |
| 02:37.04 | russellb | blitzrage: greetings sir |
| 02:37.25 | blitzrage | russellb: how goes this fine evening? |
| 02:42.44 | russellb | it goes ... well bye :) |
| 02:44.49 | russellb | [TK]D-Fender: is #asterisk attached to your hip? |
| 02:45.26 | [TK]D-Fender | yeah, but a chick on the phone now :) |
| 02:45.44 | russellb | heh, cool. |
| 02:47.18 | Falkon_MX-5 | woohoo, got my hardphones |
| 02:47.46 | Falkon_MX-5 | all the ringtones on the polycom 501 suck |
| 02:47.57 | JerJer | moo |
| 02:49.04 | russellb | mooooooo |
| 02:49.13 | *** join/#asterisk CVirus (n=GoD@196.205.193.87) |
| 02:49.13 | Falkon_MX-5 | who wants a mustache ride? |
| 02:49.21 | russellb | i do, i do! |
| 02:49.43 | russellb | i mean ... not really |
| 02:50.29 | Falkon_MX-5 | okay, I have working phones, now I need to set up a termination with vitelity |
| 02:51.24 | Falkon_MX-5 | for a home phone, would you use a main menu and extensions? It just seems kind of weird |
| 02:52.43 | *** join/#asterisk beek (n=klinebl@pool-72-94-31-84.phlapa.fios.verizon.net) |
| 02:52.52 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 02:59.42 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:06.10 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 03:06.18 | JerJer | press 9 to hear the toilet flush |
| 03:07.43 | Falkon_MX-5 | I think I'm just going to have it route all calls to this phone |
| 03:08.00 | Falkon_MX-5 | so, common voicemail |
| 03:08.03 | *** join/#asterisk Op3r (n=edwin@121.97.179.227) |
| 03:14.03 | Op3r | can anyone point to the direction of setting up caller id on a tdm2400p? |
| 03:14.17 | russellb | there shouldn't be any real setup required |
| 03:14.33 | russellb | other than the "callerid" option in /etc/asterisk/zapata.conf |
| 03:14.39 | russellb | for FXO, callerid=asreceived |
| 03:14.58 | russellb | for FXS, callerid="Russell Bryant" <(256) 428-6000> |
| 03:16.02 | Op3r | but Im trying to do it on the extensions.conf like exten => _1NXXNXXXXXX,n,Set(CALLERID(number)=6062663474) |
| 03:16.07 | Op3r | :( |
| 03:16.37 | russellb | you can't send callerid over an analog line |
| 03:16.57 | Op3r | yay |
| 03:17.21 | russellb | assuming that's what you're trying to do ... |
| 03:17.25 | Op3r | oh god I've been doing this all wrong |
| 03:17.35 | Op3r | :( |
| 03:17.45 | Op3r | ok let me set it up on zapata.conf |
| 03:19.48 | Op3r | pardon my stupidity because I've been up for 25 hours now whats the difference between fxo and fxs? fxo is for channelbanks or so and fxs is for outgoing line right? |
| 03:19.56 | Qwell | ~fxsfxo |
| 03:19.57 | jbot | somebody said fxsfxo was an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 03:20.36 | Op3r | OK |
| 03:20.37 | Op3r | thanks |
| 03:20.51 | Op3r | even though I cant process that information at the moment but thanks |
| 03:26.33 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:30.57 | Falkon_MX-5 | I need to get my pbx to route an incoming call straight to a certain phone. It seems to be doing its job, but it's not ringing when it routes the call. What can I do here? |
| 03:31.15 | Falkon_MX-5 | exten => s,1,Answer() |
| 03:31.15 | Falkon_MX-5 | exten => s,2,Dial(SIP/42,20) |
| 03:35.01 | *** join/#asterisk TokyoJimu (n=TokyoJim@cpe-76-88-61-165.san.res.rr.com) |
| 03:35.56 | TokyoJimu | The Caller ID asterisk sends to my phones via SIP is always one hour behind. Wy? |
| 03:37.20 | russellb | because your phone isn't set up properly |
| 03:37.27 | russellb | SIP requires that time is sent in GMT |
| 03:37.53 | russellb | so either the timezone setting on your phone is wrong, or the clock on your pc is not set correctly |
| 03:39.14 | TokyoJimu | russellb: I'll have to check my ATA-186 box then. |
| 03:42.38 | [TK]D-Fender | Falkon_MX-5, Who isn't ringing? |
| 03:45.03 | Falkon_MX-5 | [TK]D-Fender: when I call in, it automatically answers on the console and dials the appropriate extension, but while the extension is rining, the person on the phone hears nothing |
| 03:45.12 | TokyoJimu | russellb: Thanks. Fixed. Though it seems that the ATA-186 doesn't support Summer Time, so I'll be changing it manually again in a week or so. |
| 03:45.28 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:45.36 | [TK]D-Fender | Falkon_MX-5, Sounds like you don't have an indication zone set |
| 03:45.47 | russellb | TokyoJimu: you're welcome :) |
| 03:46.27 | Falkon_MX-5 | hmm? |
| 03:46.58 | *** join/#asterisk blq (n=Bl@dslb-088-066-229-151.pools.arcor-ip.net) |
| 03:46.58 | Falkon_MX-5 | [TK]D-Fender: What I want it to do on an incoming call is ring at a specific phone |
| 03:47.39 | [TK]D-Fender | Falkon_MX-5, Sounds lie its already doing that, its just that they don't HEAR that its ringing. |
| 03:48.03 | [TK]D-Fender | Falkon_MX-5, Make sure to set your tone zone & load zone if this a zaptel channel. |
| 03:49.12 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 04:02.45 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 04:03.01 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 04:04.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:08.53 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
| 04:10.41 | fetcher | can anything be done w/ Asterisk 1.2 to help smooth out incoming SIP jitter from a particular peer? |
| 04:11.10 | fetcher | I know 1.4 has a new SIP jitter-buffer |
| 04:11.38 | syle | upgrade i;m sure 1.2 won;t be supported very long |
| 04:12.46 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 04:13.00 | Flauto | hello |
| 04:13.27 | Flauto | [Oct 25 23:09:54] WARNING[3194]: chan_zap.c:11121 process_zap: Ignoring signalling |
| 04:13.27 | Flauto | [Oct 25 23:09:54] ERROR[3194]: chan_zap.c:10472 build_channels: Unable to reconfigure channel '3' |
| 04:13.27 | Flauto | [Oct 25 23:09:54] WARNING[3194]: chan_zap.c:11437 reload: Reload of chan_zap.so is unsuccessful! |
| 04:13.32 | Flauto | i got this message |
| 04:13.39 | Flauto | would anyone to help out |
| 04:14.01 | Flauto | i have a x100p which is channel 5 |
| 04:14.28 | Flauto | i also have a old tdm card wich has two fxo and fxs ports |
| 04:14.45 | Flauto | 3 is one of the fxo ports |
| 04:15.09 | fetcher | Flauto: I've seen that only with Sangoma cards ... it seemed not to cause any problems (ports worked despite the warnings) |
| 04:15.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@70.240.164.157) |
| 04:16.14 | Flauto | in 1.4 do i still need to modprobe after makeinstall zaptel? |
| 04:16.52 | Flauto | i have problems with the fxs ports as well, there is no dial tone |
| 04:17.23 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:18.13 | Flauto | fetcher, for tdm card, do i have to modprobe wctdm and something else? |
| 04:18.21 | Flauto | like for the fxs ports |
| 04:19.22 | Flauto | if i want to replace this tdm card with a two fxo ports one, what is the cheap but good one to get |
| 04:20.07 | *** join/#asterisk BBHoss_Work (n=hoss@216.186.235.254) |
| 04:23.00 | *** join/#asterisk iamamoron (n=kerlnel2@202.137.121.84) |
| 04:23.02 | iamamoron | hi there |
| 04:23.15 | iamamoron | any ideas on how to do call forwarding in asterisk 1.2? |
| 04:23.28 | phix | hmmm, some phone numbers I type in are coming back as unavailable, even though they should match a pattenr |
| 04:23.34 | iamamoron | most of the samples are done in lower version |
| 04:23.48 | iamamoron | any ideaS |
| 04:23.49 | iamamoron | ? |
| 04:24.00 | phix | eg, _13XXXX and _1[389]00XXXXXX |
| 04:24.04 | HarryR | uh, the samples should work for 1.2... |
| 04:24.34 | iamamoron | nop |
| 04:24.44 | iamamoron | DBput is deprecated |
| 04:25.02 | iamamoron | it is replaced by Set(.........) |
| 04:25.11 | iamamoron | anybody else? |
| 04:25.12 | [TK]D-Fender | iamamoron, Go read the docs on what changed.... |
| 04:25.35 | iamamoron | any ideas on how to implement call forwarding in asterisk-1.2 |
| 04:25.41 | [TK]D-Fender | iamamoron, And you do it the same way with the proper functions.... |
| 04:25.59 | iamamoron | its been two days [TK]D-Fender |
| 04:26.12 | iamamoron | thats why i landed here |
| 04:26.20 | iamamoron | maybe somebody have implemented |
| 04:26.51 | iamamoron | googling doest keep me healthy |
| 04:27.01 | [TK]D-Fender | iamamoron, 2 days? You've been in here plenty before and are only now switching to 1.2 and even thats at EOL for 1.4 |
| 04:27.03 | iamamoron | on finding info thats why I alnded here |
| 04:27.20 | [TK]D-Fender | iamamoron, Go read the DOCS folder with your source tarball |
| 04:27.55 | *** join/#asterisk abritton (n=abritton@207-255-210-182-dhcp.cbe.md.atlanticbb.net) |
| 04:28.02 | iamamoron | what I mean doing the call forwarding |
| 04:28.12 | [TK]D-Fender | phix, PASTEBIN your CLI output & code. |
| 04:28.24 | iamamoron | anybody else? |
| 04:28.43 | [TK]D-Fender | iamamoron, "show function DB" <-------- |
| 04:28.44 | iamamoron | thank you [TK]D-Fender but honestly what you are suggesting doesnt help sorry |
| 04:28.52 | phix | [TK]D-Fender: ok |
| 04:29.05 | [TK]D-Fender | iamamoron, Sure it helps. Reading tells you what functions are current. |
| 04:29.27 | iamamoron | my problem is implementation |
| 04:29.32 | iamamoron | anybody else? |
| 04:29.42 | iamamoron | with a working 1.2 |
| 04:29.46 | [TK]D-Fender | iamamoron, But you don't want to work at it, you want someone to just hand you a pile of finished code |
| 04:29.55 | iamamoron | nop I am |
| 04:30.10 | iamamoron | I just want to ask who have experience |
| 04:30.22 | [TK]D-Fender | iamamoron, plenty of us, including myself. |
| 04:30.36 | iamamoron | sorry [TK]D-Fender] for the second time you are not helping hope you understand |
| 04:30.59 | [TK]D-Fender | iamamoron, in each exten you check a value in something like AstDB to see where we should actually be directing the call and go there. |
| 04:30.59 | phix | ~pastebin |
| 04:31.01 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 04:31.24 | [TK]D-Fender | iamamoron, If there is any kind of misunderstanding its because your questions are poor. |
| 04:31.50 | phix | [TK]D-Fender: hmmm it is telling me the circuit is busy, but no one was on the phone at the time I placed the call |
| 04:32.04 | [TK]D-Fender | phix.... PASTEBIN IT. |
| 04:32.24 | [TK]D-Fender | phix, Circuit busy sure doesn't sound like a pattern match failure... |
| 04:32.29 | phix | [TK]D-Fender: http://rafb.net/p/3mBrcp78.html |
| 04:32.54 | iamamoron | nop it is not poor [TK]D-Fender its just that you dont get it |
| 04:32.59 | phix | [TK]D-Fender: It gives the same error message on the phone (unavailable) as a pattern match failure |
| 04:33.03 | [TK]D-Fender | phix, So wheres the pattern match error? |
| 04:33.03 | iamamoron | heop you understand |
| 04:33.17 | phix | hmm I should change the default error messages |
| 04:33.38 | phix | [TK]D-Fender: There isn't one, It is a busy circuit :/ but no one was using the phone at the time the call was placed |
| 04:33.47 | phix | (no one except the user placing the call obviously) |
| 04:34.06 | [TK]D-Fender | iamamoron, You aren't asking anything meaningful at this point and I have aready given you a description of implementation and pointed you to getting instructions on the key function to help you do it. |
| 04:34.12 | JT | iamamoron: the questions are poor |
| 04:35.25 | [TK]D-Fender | phix, that call status message is very vague. you should always do tests like this with sip debug enabled so you can see better result codes as to what happened. |
| 04:35.28 | iamamoron | ok here |
| 04:35.49 | iamamoron | any online sample you can point that has sample dial plan for call forwarding using 1.2 |
| 04:36.15 | iamamoron | so that I can read and study |
| 04:36.34 | iamamoron | because it seems that I am kinda in the grey hole in call forwarding |
| 04:36.34 | JT | phix: yes, you are sending your numbers INCORRECTLY |
| 04:36.49 | JT | phix: you need to send them in international number format to pennytel |
| 04:37.01 | JT | so not 131313 |
| 04:37.08 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 04:37.53 | [TK]D-Fender | iamamoron, well you could have saved all this time and just checked the WIKI.... not like it would have taken more than 1 minute to find samples. |
| 04:39.06 | *** join/#asterisk marmsu (n=marmsu@66-207-221-226.beanfield.net) |
| 04:39.10 | iamamoron | huh |
| 04:40.09 | marmsu | does anyone know if it's possible for Asterisk to continue executing the dialplan if the calling party disconnects during a Dial()? it seems to terminate the script if that happens |
| 04:40.40 | marmsu | the 'g' flag seems to do the trick if the called party hangs up, which is nice, but doesn't cover the case if the calling party hangs up |
| 04:40.46 | [TK]D-Fender | marmsu, use the "h" Asterisk Standard Extension |
| 04:41.30 | phix | [TK]D-Fender: ok |
| 04:41.32 | phix | JT: hmmm |
| 04:42.04 | phix | JT: so 001161131313 ? or 61131313 ? |
| 04:42.52 | phix | JT: hmmm local calls work without prepending a 61${AREACODE}${PHONENUMBER} |
| 04:43.04 | JT | that's strange |
| 04:43.13 | *** join/#asterisk MacWinner (n=chatman@70-100-130-167.dsl1-fairport.roc.ny.frontiernet.net) |
| 04:43.17 | JT | 61131313 is what it should be |
| 04:43.20 | JT | same for everything |
| 04:43.24 | phix | ok, I will try that |
| 04:43.25 | JT | as per pennytel documentation |
| 04:43.32 | phix | where is this documentation? |
| 04:43.37 | JT | pennytel.com |
| 04:43.47 | phix | JT: hmmm yeah where abouts? I can't find shit on that site |
| 04:44.08 | phix | JT: for calling overseas I obviously don't prepend 61 ay :) |
| 04:45.44 | phix | hmmm, will appending 61 infront of a number work on a PSTN line? |
| 04:46.57 | JT | what |
| 04:47.01 | JT | they're all PSTN lines |
| 04:47.07 | JT | pennytel is for calling the PSTN |
| 04:47.32 | JT | if you are calling overseas, you append the appropriate country code |
| 04:47.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:48.32 | phix | exten => _0NXXXXXXXX,1,Macro(dialout,${COUNTRYCODE}${EXTEN}:1) ; Mobile / STD. |
| 04:48.40 | *** join/#asterisk marc7 (n=marc@S010600131024913b.vc.shawcable.net) |
| 04:48.50 | phix | JT: no, I mean if I call out of a Zap device |
| 04:48.56 | phix | onto PSTN (land line) |
| 04:49.13 | [TK]D-Fender | phix, Almost.. |
| 04:49.17 | phix | JT: you have your AU extension.conf handy for DC? :) |
| 04:49.19 | phix | DC = DCC |
| 04:49.32 | phix | :1 goes within the ${} ? |
| 04:50.04 | [TK]D-Fender | phix, For starters |
| 04:50.08 | phix | My dialout macro will call through ZAP if SIP provider fails |
| 04:50.10 | marmsu | [TK]D-Fender: thanks, your suggestion to use the "h" extension works well |
| 04:50.22 | [TK]D-Fender | phix, then there is the matter of the FIXED country code. Only use them for 1 country? |
| 04:50.28 | marmsu | [TK]D-Fender: one thing though, is it possible to obtain the original ${EXTEN} from here? |
| 04:50.41 | phix | [TK]D-Fender: that match that pattern yes |
| 04:50.45 | [TK]D-Fender | marmsu, You're welcome. You need a combination to handle a lot of this stuff |
| 04:51.01 | phix | [TK]D-Fender: out of country numbers will need a 0011 infront which will match my other pattern |
| 04:51.10 | phix | exten => _0011X.,1,Macro(dialout,${EXTEN}) ; International calls. |
| 04:51.18 | marmsu | a combination? |
| 04:51.19 | [TK]D-Fender | marmsu, What I'd advise : Copy it to a channel var and pull it. |
| 04:51.34 | marmsu | ahh .. interesting. I'll look into that, thanks. |
| 04:52.07 | MacWinner | hi, could someone suggest a good voip termination provider? looking for something reliable, reputable and cheap.. not necessarily the cheapest |
| 04:52.19 | phix | JT? [TK]D-Fender? |
| 04:52.47 | [TK]D-Fender | MacWinner, For where? |
| 04:53.11 | MacWinner | USA, UK, SE Asia countries |
| 04:53.42 | [TK]D-Fender | MacWinner, Might be better off doing a "best of breed" setup with 1 provider for wach region |
| 04:54.28 | MacWinner | yah, didn't think of that |
| 04:54.36 | MacWinner | thanks! |
| 04:54.52 | MacWinner | was looking for 1 provider for simplicity |
| 04:55.16 | MacWinner | i'm a bit new to asterisk so not sure how complicated setting up multiple providers would be |
| 04:55.56 | JT | phix: no, i don't think prepending 61 will work on a normal line |
| 04:55.58 | [TK]D-Fender | MacWinner, I've run 3-4 in a single system. And sure it may be a bit more to set up but usage can be completely transparent base on how you make your dialplan. |
| 04:56.11 | JT | phix: check back in 20mins re dialplan for AU |
| 04:56.38 | MacWinner | cool, can u reco a few that you used? |
| 04:59.08 | iamamoron | any links for call forwarding please? |
| 04:59.13 | iamamoron | with sample dial plans? |
| 04:59.32 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:00.09 | [TK]D-Fender | iamamoron, ... |
| 05:00.11 | [TK]D-Fender | ~wikis |
| 05:00.11 | jbot | it has been said that wikis is http://www.voip-info.org |
| 05:00.33 | iamamoron | there are log there |
| 05:00.39 | iamamoron | please try to be specific |
| 05:00.44 | *** join/#asterisk thx2000 (n=the@netblock-208-127-150-56.dslextreme.com) |
| 05:00.45 | iamamoron | http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
| 05:00.49 | iamamoron | this is not working |
| 05:01.10 | [TK]D-Fender | iamamoron, Yeah, that looks good... |
| 05:01.59 | ido | what do you folks think of trixbox CE (the free one) as a medium sized business/call center platform? just a single trixbox, not multiple ones...to be handling about 10-20 simultaneous calls |
| 05:02.26 | thx2000 | ido: I've implemented a few and I like it |
| 05:02.44 | ido | no problems with scalability or features? |
| 05:03.03 | thx2000 | It's a heck of a lot easier than implementing a feature set from scratch... |
| 05:03.28 | thx2000 | And you still have the flexibility of creating your own features should you need to |
| 05:04.23 | ido | the one thing i'm trying to do is integrate the phone system (and recorded phone conversations) into a custom ticket system, and i plan to use a speech recognition library of some sort to get the "gist" of the conversation into the ticket, with an attached mp3 recording of the call |
| 05:04.31 | [TK]D-Fender | ido, if the FreePBX cookie-cutter setup fits your needs then "whatever". |
| 05:04.44 | ido | trixbox is freepbx? |
| 05:04.52 | [TK]D-Fender | ~trixbox |
| 05:04.52 | jbot | well, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 05:04.56 | ido | i'm still deciding what platform to use for the voice part |
| 05:05.12 | phix | JT: ok |
| 05:05.21 | [TK]D-Fender | ido, What otehr platforms being considered? |
| 05:05.22 | ido | oh ok i see for some reason i thought trixbox was directly asterisk based...i'll ask any trixbox questions in their channel in the future. |
| 05:05.28 | ido | asterisk, straight up, is another choice |
| 05:05.55 | phix | JT: hmmm, well that isn't good, my dialout macro will need to be able to fall back to dialing out using PSTN line (via ZAP) :/ |
| 05:05.56 | [TK]D-Fender | ido, Trixbox is a full distro with * which is CONFIGURED by FreePBX to pump out cookie cutter configs. |
| 05:06.06 | ido | the configuration isn't a problem -- i'm quite happy with complicated configuration files thanks to my background with sendmail (anyone remember .cf files before the days of m4?) |
| 05:06.36 | ido | is there already a library or plugin or free / open source software that handles speech recognition for phone calls? |
| 05:06.47 | thx2000 | ido: ARI will do something fairly similar to what you're describing |
| 05:06.53 | ido | (is anyone else doing this kind of speech to text conversion for conversations?) |
| 05:07.00 | [TK]D-Fender | ido, yes, there is Sphinx (iffy but free), and 1 other... |
| 05:07.08 | [TK]D-Fender | ido, 1 sec on that... |
| 05:07.22 | [TK]D-Fender | ido, Lumenvox. |
| 05:07.24 | ido | sphinx was the one i was planning on using |
| 05:07.30 | ido | i'll take a look at both |
| 05:07.37 | [TK]D-Fender | ido, "affordable" and rpetty decent from the few accoutns I've heard of it. |
| 05:07.51 | ido | well, this'd need to be open source |
| 05:07.53 | ido | i think |
| 05:08.32 | [TK]D-Fender | ido, I would think "workable" would take precedence :) |
| 05:09.37 | ido | well, the conversation doesn't need to be perfectly transcribed, and i've already got a spam filter -esque script in place to filter messages (and hopefully transcripts too) into the correct category. |
| 05:09.54 | ido | the transcription would be mainly for categorization, the audio recording is for actually responding to the ticket |
| 05:10.04 | MacWinner | have you guys ever heard of the "Dove Foundation" ? I got a call cold call from there.. the automated system they used is so lifelike. it fooled me for a long time |
| 05:10.04 | MacWinner | i actually that i was talking to a real person |
| 05:10.13 | [TK]D-Fender | ido, Oh you want REAL voice recognition for PROSE? |
| 05:10.37 | ido | yes, i suppose? |
| 05:10.48 | ido | it doesn't need to be that real |
| 05:10.53 | JT | phix: what isn't any good? |
| 05:11.06 | [TK]D-Fender | ido, Oh, then screw Sphinx. From what I hear it's spotty enough with a multiple-choice w/ limited dictionary :) |
| 05:11.18 | ido | haha |
| 05:11.23 | ido | i'm screwed aren't i haha |
| 05:11.29 | [TK]D-Fender | ido, Just giving it to you raw & uncut. |
| 05:11.41 | [TK]D-Fender | ido, No, just not "free" :) |
| 05:11.42 | ido | that's how i prefer it fender |
| 05:11.46 | ido | uncut |
| 05:11.53 | ido | i mean |
| 05:12.01 | ido | i was responding to the first of those two |
| 05:12.11 | ido | http://linux.slashdot.org/article.pl?sid=06/10/10/1953216&from=rss <-- slashdot, so biased :( |
| 05:12.17 | thx2000 | I can't imagine theres a really good solution out there...Seems like theres a handful of pretty big guns who still haven't gotten vr right |
| 05:12.50 | ido | well, dragon naturally speaking seems to do a fantastic job? |
| 05:13.08 | [TK]D-Fender | ido, No clue on integration capabilities with * though. |
| 05:13.12 | JT | apparently it's really hard for a cumputer to interpret human voice |
| 05:13.32 | ido | [TK]D-Fender: i can handle the integration...if it's got a C api, it can be integrated ;) |
| 05:13.36 | ido | jt: yes. |
| 05:13.37 | [TK]D-Fender | ido, About a decade ago I was working on a VB front-end to use Dragon for my home automation AI :) |
| 05:13.43 | thx2000 | I'll admit its been a while since i've used any of the stuff out there...but even after reading into it for an hour or so it was still about useless |
| 05:13.56 | dlynes_laptop | ido: dragon dictate is actually quite bad, if you want something on a server |
| 05:14.14 | dlynes_laptop | ido: dragon dictate is only really good at interpreting one specific voice; it's not adaptive |
| 05:14.24 | ido | okay. |
| 05:14.29 | dlynes_laptop | ido: and even then, it's voice training files get corrupted regularly |
| 05:14.36 | dlynes_laptop | s/it's/its/ |
| 05:14.49 | ido | haha |
| 05:14.53 | ido | jbot ftw |
| 05:14.54 | jbot | i heard ftw is wtf backwards, or for the win |
| 05:14.55 | ido | oh wait |
| 05:14.57 | ido | nvm |
| 05:15.03 | ido | okay |
| 05:15.43 | ido | i think you folks have convinced me to just let the person on the line summarize the ticket instead of transcription :( |
| 05:15.45 | dlynes_laptop | ido: I used to use it regularly because I used to teach people with physical handicaps how to use computers; some of them were quadraplegics, and so they had to use speech recognition software...dragon dictate was pretty much the best one, and even then it wasn't that good |
| 05:16.02 | [TK]D-Fender | ido, just like my IQ... I though they wrote 181 but I had the results upside-down... |
| 05:16.37 | ido | [TK]D-Fender: it must really be 181 because that one flew right over me |
| 05:19.11 | [TK]D-Fender | whee! |
| 05:20.13 | [TK]D-Fender | ok, bed-time... |
| 05:20.18 | [TK]D-Fender | later all |
| 05:24.59 | MacWinner | have any of you used sellvoip.net? |
| 05:25.27 | MacWinner | i think they have been around for a while. i remember looking at them 2 years ago |
| 05:26.29 | iamamoron | any links for call forwarding please? |
| 05:27.05 | grimsy | iamamoron: http://www.google.com.au/search?complete=1&hl=en&q=asterisk+call+forwarding&btnG=Google+Search&meta= |
| 05:27.24 | MacWinner | exgn also looks cheap |
| 05:28.09 | iamamoron | but i use softphone like x-lite |
| 05:28.15 | iamamoron | would it be possible? |
| 05:28.19 | JT | zomg |
| 05:29.14 | iamamoron | when i push flush |
| 05:29.22 | iamamoron | it will go to line 2 |
| 05:33.01 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 05:33.07 | iamamoron | any ideas? |
| 05:34.15 | JT | flush? |
| 05:35.08 | iamamoron | flash i mean |
| 05:35.10 | iamamoron | sorry |
| 05:35.33 | iamamoron | I am suspecting my xlite is problem dialplan is ok |
| 05:35.37 | iamamoron | any ideas? |
| 05:35.51 | iamamoron | anybody have tried to forward a call? |
| 05:35.56 | iamamoron | using xlite |
| 05:37.23 | iamamoron | any ideas? |
| 05:38.05 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 05:39.32 | iamamoron | any ideas? |
| 05:43.49 | JT | iamamoron: stop repeating? |
| 05:43.50 | JT | iamamoron: stop repeating? |
| 05:44.04 | kaldemar | iamamoron: are you actually asking somthing? |
| 05:44.55 | kaldemar | uncle google will give you plenty of links for call forwarding. |
| 05:45.11 | *** join/#asterisk techie (n=techie@adsl-76-214-5-75.dsl.lsan03.sbcglobal.net) |
| 05:46.36 | iamamoron | yes I am also there but cant fined perfect for what i want |
| 05:46.45 | iamamoron | maybe somebody could shed a light |
| 05:46.53 | phix | JT: Does this look about right? --> http://rafb.net/p/Jkr0ij90.html |
| 05:48.12 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:48.57 | JT | phix: not very precise and emergency calls will fail |
| 05:49.02 | phix | also, what are some ways increase priority on SIP traffic? (Asterisk is running on Debian Etch Linux, which is also a router as well) |
| 05:49.09 | grimsy | iamamoron: rtfm http://www.counterpath.com/assets/files/191/X-Lite3.0_UserGuide.pdf |
| 05:49.24 | phix | JT: hmmm, ok, I will wait 10 min or so for your example then :) unless you know of a site that has them up already |
| 05:49.41 | phix | JT: yes I know I havn't put 000 in there |
| 05:51.17 | phix | so far I have used tc to proiritise upload traffic, but I havn't realy found a good way to prioritise download traffic (short from dropping packets) |
| 05:52.31 | phix | JT: How do you overcome this? calls are a bit flaky when people in my LAN are uploading or downloading (they are not maxing out the bandwidth though which is ADSL2 - 4000/872) |
| 05:52.34 | JT | http://rafb.net/p/bZYALx88.html |
| 05:52.50 | JT | 4000, sound like ADSL 1.5 ;) |
| 05:53.44 | phix | why do 13 numbers go through PSTN line? |
| 05:53.59 | phix | JT: yeah well i am 2.something Kms away from exchange :( sucks to be me |
| 05:54.31 | JT | phix: it doesn't |
| 05:54.34 | phix | well actually ADSL1 has a max bandwith of 8000 kbps, so I am getting ADSL0.5 speeds :) |
| 05:54.35 | JT | well |
| 05:54.42 | JT | it does, but that's commented out |
| 05:54.43 | JT | clearly |
| 05:54.49 | JT | i use dial macros |
| 05:54.49 | phix | oh ok :) |
| 05:55.00 | phix | can I see your dial macro? |
| 05:55.10 | JT | and variables that set the priority or each call type |
| 05:55.12 | JT | of |
| 05:55.40 | phix | [2-37-8]? couldn't that be written as [2378] ? |
| 05:55.50 | phix | ok |
| 05:56.04 | JT | that could |
| 05:56.13 | phix | so it will kill off a call if an emergency number is called? |
| 05:56.46 | JT | what? |
| 05:57.01 | phix | your dialmacro |
| 05:57.20 | phix | so you do not allow 1900 numbers? |
| 05:57.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:57.55 | phix | a 13 number is the same as a 1300 number? |
| 05:59.18 | JT | except they're different lengths |
| 05:59.21 | JT | and costs |
| 05:59.22 | JT | to buy |
| 05:59.59 | *** part/#asterisk techie (n=techie@adsl-76-214-5-75.dsl.lsan03.sbcglobal.net) |
| 06:00.40 | JT | http://rafb.net/p/S4G8yF62.html |
| 06:00.47 | JT | and no 1900 numbers |
| 06:00.56 | JT | most voip providers don't support them, anyway |
| 06:03.01 | phix | ok |
| 06:03.28 | *** join/#asterisk DrCron (n=rszasz@c-24-5-134-158.hsd1.ca.comcast.net) |
| 06:04.17 | JT | and if you want to allow them it should be _190XXXXXXX |
| 06:04.20 | JT | i think |
| 06:04.39 | phix | yeah |
| 06:05.17 | phix | hmmm, I will just play a custom message explaining it will be routed through to PSTN line and high charges will apply |
| 06:05.44 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:05.46 | phix | Congestion? |
| 06:06.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:09.55 | phix | hmmmm |
| 06:11.27 | phix | do most VoIP providers support 000 ? |
| 06:11.29 | iamamoron | any ideas on cal forwarding on xlite |
| 06:11.40 | iamamoron | my client is xlite |
| 06:12.30 | phix | I use that too |
| 06:12.35 | phix | but I am starting to dislike it |
| 06:12.46 | phix | I would like a LDAP addressbook or similar |
| 06:12.46 | iamamoron | what i mean i need to flash the caller and transfer to another extension |
| 06:13.15 | JT | you can't use flash on SIP, end of story. |
| 06:13.40 | JT | phix: the Congestion application is well documented |
| 06:13.50 | JT | a lot of providers support 000 |
| 06:13.57 | phix | ok |
| 06:15.30 | *** join/#asterisk ManxPower (n=manxpowe@254.sub-70-221-93.myvzw.com) |
| 06:15.40 | grimsy | iamamoron: omfg rtfm |
| 06:15.42 | grimsy | and stfu |
| 06:21.14 | phix | hmmm 1XXXXXXX are local national calls right? (a national number that is charged at local call rate) |
| 06:22.26 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:23.40 | JT | i think you mean _13XXXX or _1300XXXXXX |
| 06:23.44 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:31.25 | phix | hmmm, maybe |
| 06:32.18 | phix | 0198 333 713 |
| 06:32.22 | phix | 01 even |
| 06:33.43 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 06:34.39 | JT | 0198 is for isp dialup |
| 06:34.51 | phix | yep |
| 06:35.03 | phix | I don't need to worry about that ay |
| 06:35.27 | phix | you would think there would be some document that explains the telephone numbering format used in Australia :) |
| 06:37.43 | grimsy | phix: i found this a while ago - on ebay of all places -- http://reviews.ebay.com/Australian-Telephone-Prefix-Numbering-Plan_W0QQugidZ10000000003244647 |
| 06:37.48 | grimsy | found it useful |
| 06:37.58 | grimsy | not sure how accurate it is though |
| 06:39.38 | grimsy | and just came across this on wikipedia http://en.wikipedia.org/wiki/+61 |
| 06:44.04 | phix | grimsy: thanx :) |
| 06:44.13 | grimsy | np :D |
| 06:44.16 | phix | ah good ol' wiki |
| 06:44.44 | phix | ummm is there a document that lists and explains the usage of all applications available in extensions.conf? |
| 06:44.52 | *** join/#asterisk axscode (n=axscode@230.57.49.60.klj02-home.tm.net.my) |
| 06:45.36 | grimsy | have you looked through the sample extensions.conf file from svn? |
| 06:46.41 | grimsy | file is at http://svn.digium.com/svn/asterisk/trunk/configs/extensions.conf.sample |
| 06:54.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:56.53 | phix | no, I never new about it |
| 06:56.57 | phix | <PROTECTED> |
| 06:59.35 | *** join/#asterisk techie (n=techie@adsl-76-214-5-75.dsl.lsan03.sbcglobal.net) |
| 06:59.49 | *** part/#asterisk techie (n=techie@adsl-76-214-5-75.dsl.lsan03.sbcglobal.net) |
| 06:59.56 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:01.41 | JT | what the hell |
| 07:01.52 | JT | grimsy: |
| 07:01.54 | JT | phix: |
| 07:02.00 | JT | there IS a proper document |
| 07:02.06 | JT | the national numbering plan |
| 07:02.09 | grimsy | where? |
| 07:02.09 | JT | it's what i used |
| 07:02.35 | grimsy | i'd love to see it |
| 07:02.40 | grimsy | couldn't find it when i was looking |
| 07:02.48 | *** part/#asterisk beek (n=klinebl@pool-72-94-31-84.phlapa.fios.verizon.net) |
| 07:03.45 | *** join/#asterisk BeeBuu (n=chatzill@125.95.28.86) |
| 07:03.58 | BeeBuu | hello,all. |
| 07:04.26 | JT | http://www.comlaw.gov.au/ComLaw/Legislation/LegislativeInstrumentCompilation1.nsf/0/B69E1A2BE874F1F2CA2572EB0021DE3B?OpenDocument |
| 07:04.31 | JT | grimsy: quite easy to find really :P |
| 07:04.35 | BeeBuu | is AMI better than AGI? |
| 07:04.46 | JT | BeeBuu: they do totally different things |
| 07:04.55 | grimsy | just found the link as you sent it JT thanks |
| 07:04.56 | grimsy | :D |
| 07:05.29 | agx | BeeBuu, don't think so, AMI si mostly to see events and i should say is not a good output format |
| 07:08.35 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.237) |
| 07:12.45 | BeeBuu | OH? |
| 07:13.10 | JT | BeeBuu: they're completely different |
| 07:13.14 | JT | read up on them |
| 07:14.06 | BeeBuu | em,it beyond me |
| 07:14.33 | BeeBuu | am i a nut? |
| 07:16.22 | JT | ~thebook |
| 07:16.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 07:16.25 | phix | JT: ok :) another question, is there a document for asterisk that tells me all of the applications available in a dialplan and their usage? Eg, Playback(), Record(), Dial() etc..? |
| 07:16.53 | *** join/#asterisk ming_zym (n=ming_zym@125.120.41.166) |
| 07:16.54 | BeeBuu | Thank you very much,JT |
| 07:16.58 | JT | phix: the book mentions most |
| 07:17.13 | JT | and in asterisk... show application <blah> |
| 07:22.33 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 07:27.34 | BeeBuu | JT:no full introduction? |
| 07:27.46 | JT | BeeBuu: ? |
| 07:28.16 | BeeBuu | the question of phix |
| 07:28.34 | BeeBuu | is there a document for asterisk that tells me all of the applications available in a dialplan and their usage? Eg, Playback(), Record(), Dial() etc..? |
| 07:31.00 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:31.25 | BeeBuu | "It can also be used for a wide variety of applications,such as automated dialers and click-to-call systems" |
| 07:32.22 | BeeBuu | so i can control * to do what i want to do? |
| 07:32.51 | BeeBuu | without AGI |
| 07:33.26 | phix | JT: ok |
| 07:36.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-240-164-157.dsl.hstntx.swbell.net) |
| 07:38.31 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-220-104.sd.sd.cox.net) |
| 07:38.50 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 07:38.55 | sahafeez | hello, i have checked the wiki but i have not found an answer |
| 07:39.29 | sahafeez | i have a 1.2.x system. polycom phones. digium card with a pri. i have a call tomorrow that i need to make and i need it recored. |
| 07:40.02 | sahafeez | the monitor cmd seems to be for inbound to an extention. can someone point me at the right thing. |
| 07:40.22 | *** part/#asterisk ming_zym (n=ming_zym@125.120.41.166) |
| 07:44.15 | sahafeez | if i am using automon and have exten => _NXXXXXX,2,Dial(${TRUNK}/${EXTEN},,o) currently |
| 07:44.43 | phix | JT: the mentions how to create unavailable messages? or examples the different types of errors messages that can be customised? |
| 07:44.51 | sahafeez | do i add wW to the end? so it is ,,owW) |
| 07:45.49 | *** join/#asterisk munmun (n=mun_mun@202.122.127.252) |
| 07:46.43 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 07:46.44 | phix | I would like to know what that means too |
| 07:46.55 | phix | the letters at the end of a dial app |
| 07:47.23 | JT | ... |
| 07:47.29 | JT | show application dial |
| 07:53.06 | Mw3 | in zap show status i get very high irq for my te110p (1628523 after 2 days uptime). and the line quality is very bad. what can cause this? |
| 07:55.17 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
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| 08:05.51 | phix | ok |
| 08:10.29 | phix | that is very helpful, thank you JT |
| 08:12.31 | Shaun2222 | will gotoif work if i dont specify a =, like a true/false type deal... example: GotoIf($["${SCREEN_FILE}"]?2:4) |
| 08:15.35 | *** join/#asterisk LukinoVoip (n=LukinoVo@host15-224-static.57-82-b.business.telecomitalia.it) |
| 08:17.41 | kaldemar | Shaun2222: if SCREEN_FILE is always 1 or 0, it might work. try it. |
| 08:18.29 | kaldemar | Shaun2222: are you trying to test if SCREEN_FILE is defined or what? |
| 08:23.28 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.57) |
| 08:25.45 | LukinoVoip | Hi all, i have a setup like this: TENOVIS PBX -> (PRI) -> PHILIPS PBX -> (PRI) -> AST with TE410P.It works fine except of TENOVIS vs AST calls where i get a Q.931 protocol error with cause 101 (Wrong call state).Any Ideas? |
| 08:26.07 | LukinoVoip | http://asterisk.pastebin.ca/750144 |
| 08:26.15 | LukinoVoip | :s |
| 08:29.25 | *** join/#asterisk lukketto (n=lukketto@host192-27-dynamic.7-87-r.retail.telecomitalia.it) |
| 08:32.18 | *** join/#asterisk jfg (i=56d4e44c@gateway/web/cgi-irc/ircatwork.com/x-6f08cabf4d8a2d69) |
| 08:32.21 | jfg | heelo |
| 08:32.53 | jfg | I need support on TC400P card |
| 08:35.21 | Uatec | call digium |
| 08:36.58 | *** join/#asterisk jfg (i=56d4e44c@gateway/web/cgi-irc/ircatwork.com/x-61110f451c4e534c) |
| 08:37.34 | jfg | Join our live IRC Asterisk® chat channel to receive useful information, advice and troubleshooting help. |
| 08:37.42 | jfg | so, i'm here |
| 08:37.50 | BeeBuu | ~PBX |
| 08:37.51 | jbot | i heard pbx is a Private Branch eXchange |
| 08:39.12 | jfg | does anyone try to use a TC400P under debian ? |
| 08:39.27 | jfg | I have some problems at boot |
| 08:40.36 | jfg | the modules are loaded very slowly.. so slowly that my system don't boot ! |
| 08:45.11 | Shaun2222 | i'm drawing a blank, whats teh difference between exten => 100 and exten => _100 |
| 08:48.41 | kaldemar | Shaun2222: _ indicates that the extension is a pattern. |
| 08:48.49 | Shaun2222 | ah, thats right. |
| 08:49.06 | Shaun2222 | i use paterns all the time i dont think i ever have anything hard set :) |
| 08:49.26 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:49.50 | kaldemar | but as such, 100 and _100 both match to 100 only. so no difference. |
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| 08:56.11 | GaryH-Work | Is that "Builtins" or "Butlins" ?? :-) |
| 08:57.14 | *** part/#asterisk GaryH-Work (n=chatzill@2001:618:42d:101:213:72ff:fecf:8262) |
| 09:00.02 | Daviey | eww IPv6 address |
| 09:07.45 | Uatec | what? |
| 09:08.01 | Daviey | eww IPv6 address |
| 09:08.08 | Uatec | what IPv6 address? |
| 09:08.16 | Daviey | GaryH-Work's |
| 09:08.39 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 09:09.14 | Uatec | oh yes, ipv6 |
| 09:09.17 | Uatec | it's cool |
| 09:09.26 | Uatec | you can abreviate ipv6 |
| 09:10.37 | Daviey | ipv6 is butt ugly |
| 09:12.56 | Uatec | butt ugly like your face |
| 09:12.57 | Uatec | :P |
| 09:14.08 | Daviey | :`( |
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| 09:41.45 | penguinFunk | very intellectual conversion about some complex issues there |
| 09:41.54 | penguinFunk | .. |
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| 09:45.44 | xheliox | penguinFunk: ipv6 good. rain bad. me hungry. |
| 09:50.03 | penguinFunk | :P |
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| 10:13.37 | pif | hi, I can't hear and be heard on a call received through my agent |
| 10:13.53 | pif | known problem? |
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| 10:21.58 | RockHound | good day |
| 10:23.22 | RockHound | what can be the cause of all playback from the pbx stuttering? for example voicemail prompts and wait music... |
| 10:23.39 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 10:30.53 | Guggemand | to little cpu |
| 10:30.55 | Guggemand | bad timing |
| 10:30.58 | Guggemand | something like that |
| 10:31.38 | axscode | is there an apt-get install asterisk with mysql support? |
| 10:33.03 | RockHound | Guggemand: hmm ... I had asterisknow running which did the play back fine, but have decided to setup my own install on the same hardware .. and now it studders ... so I can eliminate the CPU I guess |
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| 10:54.16 | k31th | woo |
| 10:55.24 | k31th | !book |
| 10:55.29 | k31th | ~book |
| 10:55.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
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| 10:59.56 | xheliox | 2nd edition, eh? |
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| 11:00.56 | Blueneon | i've just bought a new phone and plugged it into my asterisk box, and the phone is doing something really dodgy |
| 11:01.10 | Blueneon | its picking up and hanging the line every 2 secs |
| 11:01.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:01.25 | Blueneon | any ideas |
| 11:02.13 | michael-i | I'm constantly getting ~0:46 long single tone voicemails coming in from my analog provider. The hangup is being detected in other calls but voicemail seems to still have a problem. Any ideas on how to filter or prevent these "constant-tone" recordings? |
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| 11:23.09 | roxlu | hi |
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| 11:24.33 | lilalinux | is it possible to have line-breaks in an extension? |
| 11:25.14 | roxlu | Does someone have this phone: Atcom AT-320PD ? |
| 11:25.39 | k31th | no |
| 11:25.51 | roxlu | ... or is this a 'good' phone? |
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| 11:32.10 | k31th | I have recorded a voice prompt in wa form, I am trying to get asterisk to play it |
| 11:32.29 | k31th | however i get |
| 11:32.43 | k31th | File /var/lib/asterisk/sounds/custom/enter-ext-of-person.wav does not exist in any format |
| 11:33.03 | k31th | Unable to open /var/lib/asterisk/sounds/custom/enter-ext-of-person.wav (format 0x4 (ulaw)): No such file or directory |
| 11:39.30 | michael-i | how are you calling that in the dialplan? |
| 11:43.36 | michael-i | k31th: ^ |
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| 12:08.37 | dwclarkNU | i know this is a little off topic, but does anyone else use iCall for a provider and are experienceing outages? |
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| 12:12.26 | McDouglas | hi |
| 12:12.33 | k31th | michael-i: yeah |
| 12:12.44 | McDouglas | can someone tell me what is the command to make asterisk repeat which buttons i pressed on the phone? |
| 12:13.07 | michael-i | k31th: just wondering if your prompt was working |
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| 12:20.55 | blitzrage | McDouglas: logger.conf, on the console => line, add 'dtmf' (without the quotes" |
| 12:21.10 | blitzrage | then 'logger reload' from the *CLI> |
| 12:22.56 | dwclarkNU | any opinions on voicepulse as a provider? |
| 12:24.34 | McDouglas | hmm, if i press a button i get this: http://pastebin.com/d5b671b8f |
| 12:24.45 | McDouglas | but why dont i get something simmilar if i press the R button? |
| 12:25.05 | McDouglas | (i want to figure out why transfering with the R not working :\) |
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| 12:25.43 | blitzrage | the R button? |
| 12:25.48 | blitzrage | R is not a DTMF digit |
| 12:26.01 | McDouglas | well, that explains it then :P |
| 12:26.09 | McDouglas | but why transfer is not working then? |
| 12:26.34 | blitzrage | I have no idea -- you haven't given enough information |
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| 12:26.45 | McDouglas | well, lets see, this is a panasonic dect phone, with a dvg ata attached to it |
| 12:26.53 | blitzrage | SIP? |
| 12:27.01 | blitzrage | Analog -> SIP? |
| 12:27.06 | blitzrage | and what the heck is an "R" button? |
| 12:27.10 | McDouglas | in the old analog system i could transfer a call by pressing the R button on the phone and dial the extension |
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| 12:27.30 | McDouglas | its a button with a letter R on it |
| 12:27.40 | McDouglas | its used to transfer in the old system |
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| 12:27.46 | blitzrage | that isn't gonna work because you need to use the asterisk transfer, which is probably enabled via the featuremap (features.conf) |
| 12:27.55 | McDouglas | yes |
| 12:28.07 | McDouglas | if i use the button i specified in features.conf =(X) |
| 12:28.09 | blitzrage | McDouglas: it won't "just work" on Asterisk though -- I have no idea what that R button is trying to signal... |
| 12:28.11 | McDouglas | the transfer works |
| 12:28.16 | blitzrage | there you go |
| 12:28.18 | McDouglas | but |
| 12:28.21 | McDouglas | if i use x-lite |
| 12:28.26 | McDouglas | or the linksys spa921 |
| 12:28.35 | McDouglas | i can transfer a call with the "transfer" button |
| 12:28.39 | McDouglas | and dont have to use X |
| 12:28.42 | blitzrage | right -- that is a SIP transfer |
| 12:28.49 | McDouglas | hmm |
| 12:28.56 | blitzrage | those are the 2 ways you can do transfer |
| 12:29.00 | McDouglas | so i cant initiate a sip transfer with an analog phone+ata ? |
| 12:29.29 | blitzrage | either via DTMF in the featuremap (used for analog phones), or SIP phones use the Transfer button, because there is a SIP message that gets sent to Asterisk telling it to do a transfer (which analog phones obviously can't do) |
| 12:29.45 | blitzrage | did you not just say you did it with the DTMF method? |
| 12:29.47 | McDouglas | well, i understand what you are saying |
| 12:29.51 | McDouglas | yes |
| 12:29.55 | McDouglas | with the dtmf method it works |
| 12:30.00 | blitzrage | that's how you do it with an analog phone + ata |
| 12:30.04 | McDouglas | :\ |
| 12:30.30 | McDouglas | it always sucks when users have to be retrained |
| 12:30.37 | blitzrage | unless your ATA will somehow catch the DTMF and send the transfer message via SIP |
| 12:30.49 | blitzrage | but I wouldn't know because I never use ATAs |
| 12:30.50 | McDouglas | looks like it cant do that |
| 12:31.03 | McDouglas | well, thanks for the info anyway |
| 12:31.06 | blitzrage | np |
| 12:31.06 | McDouglas | have to go now |
| 12:35.11 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 12:35.29 | agallo | When i've canreinvite!=no i cannot used dtmf to enable special functions from res_features; is there a way to block them (disabling for a single peer when globally enabled) ? |
| 12:36.15 | blitzrage | agallo: do you mean canreinvite=yes? there is no !=no |
| 12:37.08 | agallo | blitzrage, if RTP goes thru * the autopark features work; if RTP goes peer-2-peer it does not work; i've autoparking enabled for all... i want to disable it for a single peer that is using canreinvite=update,nonat :) |
| 12:37.10 | blitzrage | and if the media is reinvited and you're using rfc2833, then the DTMF will not be going through asterisk - but will be going to the end points directly, thus Asterisk will never see the DTMF |
| 12:37.15 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.178.16) |
| 12:37.29 | blitzrage | if the DTMF is in the media, how is asterisk supposed to see it? |
| 12:37.48 | agallo | blitzrage, ah, should i changed DTMF signalling then? to inband? |
| 12:37.50 | blitzrage | the only way I think is to use dtmfmode=info so the DTMF is passed in SIP messages |
| 12:37.56 | blitzrage | inband is still in the media |
| 12:38.06 | blitzrage | INFO is in the SIP signalling |
| 12:38.16 | agallo | uhm... dtmfmode=info does not work with most phones... i'll give it a try in any case TY! :) |
| 12:38.47 | blitzrage | agallo: I didn't say it'd work -- I said that was the only way to make DTMF not sit in the media stream |
| 12:39.14 | agallo | blitzrage, yes you've been usefull i'm doing a test right now :-P |
| 12:39.19 | blitzrage | heh |
| 12:39.40 | blitzrage | 2 dreams destroyed this morning while giving a useful answer... I'm on a roll :) |
| 12:40.41 | agallo | blitzrage lol |
| 12:40.58 | agallo | blitzrage, not working, bugstream 2000 does not handle dtmf via info :) |
| 12:42.04 | agallo | then i'm just going to not put Dial Options for enabling features if the phone has canreinvite enabled :) |
| 12:42.17 | blitzrage | good call :) |
| 12:54.56 | agallo | just noticed that incoming sip calls from ixnetworx.com randomly get rejected with "chan_sip.c: Sending fake auth rejection for user "39123456789"; any idea? |
| 12:54.59 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:55.40 | Falkon_MX-5 | I'm looking through the wiki, but I can't find how to forward a call and keep the origin information so it displays the correct caller ID on the destination phone |
| 12:55.49 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:56.36 | Falkon_MX-5 | what I want asterisk to do is when it gets an incoming call, forward it to my SIP phone. I have it set up to do that, but it doesn't show the origin on the phone, only that the asterisk box is calling |
| 12:57.50 | Falkon_MX-5 | nm, I'm an idiot. It is forwarding hte origin information |
| 12:59.39 | *** join/#asterisk socken23 (n=azrael@ip-213-189-154-029.fix.magnet.ch) |
| 13:00.10 | socken23 | Hi all! I'm constantly getting the following messages after I unplugged / plugged two cables to my junghanns cards: '1 received TEI check request for TEI = 0'. Any idea what's going wrong here? |
| 13:00.47 | agallo | Falkon_MX-5, you got a snom and sets a wrong display mode? |
| 13:01.36 | Falkon_MX-5 | agallo: nah, I Just thought it was displaying "asterisk" instead of the call origin, but it's not. I just tested it again. When I get an incoming from my IAX termination, will it show the origin number? |
| 13:02.01 | Falkon_MX-5 | if I tell the asterisk console, to dial my extension, then it says "asterisk" |
| 13:04.16 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 13:04.42 | *** part/#asterisk dwclarkNU (n=Administ@c-76-19-196-67.hsd1.ma.comcast.net) |
| 13:04.47 | *** join/#asterisk vyamba (n=chatzill@194.42.96.206) |
| 13:05.21 | Falkon_MX-5 | if anyone has an answer to that question, msg it to me. I gotta go. |
| 13:10.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.29 | agallo | Falkon_MX-5, well "asterisk" should happen when you don't have a callerid, like in anonymous call; now i don't recall 'because i've lots of scripts now and i don't use dialplan; but could also be that you forgot to set callerid for your calling phone |
| 13:13.47 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 13:14.04 | rpyne | sippeers => mysql,asterisk,sip_buddies and my res_mysql.conf has been edited. My Realtime Voicemail works fine. I have created my SIP table in MySQL and have inserted test entries, but my SIP entries are not registering. Any ideas are welcomed. Thanks. (I left the channel last night and just returning) |
| 13:14.31 | rpyne | Hello everyone. I am currently setting up asterisk RealTime Sip. I have edited my extconfig.conf sipusers => mysql,asterisk,sip_buddies |
| 13:14.37 | rpyne | sippeers => mysql,asterisk,sip_buddies and my res_mysql.conf has been edited. My Realtime Voicemail works fine. I have created my SIP table in MySQL and have inserted test entries, but my SIP entries are not registering. Any ideas are welcomed. Thanks. (I left the channel last night and just returning) |
| 13:17.26 | *** join/#asterisk axscode (n=axscode@60.53.85.142) |
| 13:18.04 | vyamba | hello everyone. i am setting up Asterisk with MFC/R2 and Digium TE411P card. |
| 13:20.05 | *** join/#asterisk tdi (n=tdi@gvf90.internetdsl.tpnet.pl) |
| 13:20.11 | vyamba | the line is up, but having problems with making and receiving calls. the log shows "Make call failed - Blocked" and "Exiting with DIALSTATUS=CHANUNAVAIL" |
| 13:20.11 | vyamba | when dialing out |
| 13:20.13 | tdi | hi |
| 13:20.44 | vyamba | when calling the line from PSTN, i get message that "all the lines to the exchange dialed are busy" |
| 13:20.52 | vyamba | could anyone assist. thank you |
| 13:23.44 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:25.50 | *** join/#asterisk BeeBuu (n=chatzill@125.95.199.137) |
| 13:26.21 | BeeBuu | where is the AMI documents? |
| 13:26.23 | BeeBuu | ~AMI |
| 13:26.24 | jbot | it has been said that ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 13:26.29 | [TK]D-Fender | ~book |
| 13:26.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:26.31 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 13:27.15 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:28.42 | *** join/#asterisk bintut (n=bintut@cm211.gamma177.maxonline.com.sg) |
| 13:30.08 | tdi | got a auto caller implemented, manager makes originate to Local channel in some context, in this local channel runs agi that Dials some number, when you answer that call, agi in second context is run and plays somthing. Normal Local channel usage. The problem is when you get something else than ANSWER, ex CANCEL or BUSY. Then agi dials again, the called phone (cell) rings, i answer but the asterisk does not come to the second agi in local to play th |
| 13:32.15 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 13:32.29 | rantsh | good morning everyone |
| 13:33.17 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 13:33.36 | tdi | good morning |
| 13:33.38 | rantsh | can asterisk be aware of time? I want to redirect an extension to a different one from 6pm to 12pm, is this possible??? |
| 13:34.08 | tdi | with some agi or even without yes |
| 13:34.09 | tzafrir_home | rantsh, core show application gotoiftime |
| 13:34.17 | BeeBuu | sure |
| 13:34.22 | tdi | take time to variable, check and redirect |
| 13:34.28 | tdi | ohh |
| 13:34.30 | tdi | nvm |
| 13:34.41 | tdi | is it possible to check status of originate call ? |
| 13:34.41 | rantsh | uuuu |
| 13:35.12 | tzafrir_home | tdi, hmmm... try to see what channel it has generated, and check it. |
| 13:35.25 | tzafrir_home | Not sure how to check that in a bullet-proof way, though |
| 13:35.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:35.39 | tdi | yes i can run even listener |
| 13:35.40 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 13:35.47 | tdi | but it won't be detailed i guess |
| 13:35.49 | *** join/#asterisk coppice (n=chatzill@232.196.17.210.dyn.pacific.net.hk) |
| 13:36.40 | tdi | got to overcome these locals somehow |
| 13:37.33 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.244) |
| 13:37.48 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:38.07 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 13:42.55 | tdi | quit |
| 13:52.13 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-b8875c4bd040d92e) |
| 13:56.09 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 13:56.09 | penguinFunk | gawd its last thing on a friday |
| 13:56.11 | sigmounte | hi ! |
| 13:56.30 | penguinFunk | must get out of work soon or i'll go round the bend |
| 13:56.34 | penguinFunk | :| |
| 13:57.04 | penguinFunk | hi sigmounte |
| 13:57.30 | sigmounte | Asterisk 1.4.2 , using monitor in queue , but files are not joined even using monitor-join = yes in queues.conf , any idea why ? |
| 13:57.34 | sigmounte | hi penguinFunk ! |
| 13:57.42 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 13:58.12 | penguinFunk | you restarted asterisk after modifying .conf files? |
| 13:58.12 | ussrback | Anybody knows how to set up Voicemail with mysql database? |
| 13:58.24 | penguinFunk | ussrback: there is a nice guide online |
| 13:58.33 | sigmounte | penguinFunk, yes :) |
| 13:58.38 | sigmounte | kill and restart :) |
| 13:58.48 | penguinFunk | restart now |
| 13:58.52 | penguinFunk | in cli |
| 13:58.56 | penguinFunk | ussrback: http://www.voip-info.org/wiki/view/Asterisk+CDR+csv+mysql+import |
| 13:59.47 | ussrback | CDR? i need voicemail |
| 13:59.53 | *** join/#asterisk stunsch (n=stunsch@113.Red-81-38-217.dynamicIP.rima-tde.net) |
| 14:01.20 | sigmounte | penguinFunk, i've restarted , tried again , and files are still not joined 1193410026.0-out.wav 1193410026.0-in.wav ;'( |
| 14:02.21 | stunsch | I'm looking for a free or reasonably priced softphone that can do assisted transfers with the click of a button. Any ideas? |
| 14:04.33 | [TK]D-Fender | sigmounte: Go make sure you have SOX installed |
| 14:05.22 | sigmounte | yep SoX Version 13.0.0 |
| 14:06.11 | socken23 | If I enter 'pri show span 1' I see 'Type: CPE (PtMP)' but I have a P-P connection. How can I change that?? |
| 14:06.42 | socken23 | I changed zapata.conf, signalling=bri_cpe.. thought that should do the trick!? |
| 14:06.44 | ai-a[wrk] | Is connecting line presentation implemented yet ? as in when a call's transfered the display of the phone updates when the call is finally transfered ? |
| 14:07.13 | tzafrir_home | socken23, bristuffed zaptel or misdn? |
| 14:07.23 | socken23 | bristuffed |
| 14:07.25 | tzafrir_home | zaptel |
| 14:07.30 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:07.32 | tzafrir_home | you need to restart asterisk |
| 14:07.41 | socken23 | isn't a 'reload' sufficient?? |
| 14:07.49 | tzafrir_home | signalling of a channel only takes effect on restart |
| 14:07.55 | socken23 | aha ;-) |
| 14:07.58 | socken23 | good to know |
| 14:08.13 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:08.20 | socken23 | But the signalling=bri_cpe part is a good idea!? |
| 14:08.51 | tzafrir_home | if you need ptp BRI? yes |
| 14:09.02 | socken23 | yes, we need that ;-) |
| 14:09.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:09.45 | socken23 | Mhh...didn't change anything |
| 14:10.03 | *** join/#asterisk Darthclue (n=e054502@fw149.nisd.net) |
| 14:11.16 | socken23 | And I still get a lot of '1 received TEI check request for TEI = 0' messages..... |
| 14:11.41 | *** join/#asterisk ToTo (n=ToTo@209.8.41.76) |
| 14:12.59 | ToTo | hi all |
| 14:13.05 | socken23 | ah wait a sec... Just realized there's another config file |
| 14:13.19 | *** join/#asterisk mkl1525 (n=mkl1525@i59F74D20.versanet.de) |
| 14:14.05 | socken23 | mh....still no change...darn |
| 14:14.09 | mkl1525 | Hi (1.2), did anyone experience * crashing when a queue agent does an attended transfer to a phone that isn't in the queue? |
| 14:14.27 | De_Mon | yes |
| 14:14.34 | De_Mon | all sorts of problems with agents |
| 14:16.18 | mocker | Woo, Sangoma is sending me schwag to give away at my user group! |
| 14:17.35 | tzafrir_home | socken23, you get that message? |
| 14:18.14 | socken23 | yes, still the same message |
| 14:18.20 | grandpapadot | http://yro.slashdot.org/yro/07/10/26/136217.shtml |
| 14:18.27 | tzafrir_home | socken23, could you please pastebin your zapata.conf (and every file in #include-s)? |
| 14:19.44 | sigmounte | ok finaly joined file for monitor are working , but how can i change the filename format ? |
| 14:20.13 | grandpapadot | sigmounte: What are you trying to do (jj)? |
| 14:20.25 | socken23 | tzafrir_home: yes, wait a sec |
| 14:21.19 | grandpapadot | brb |
| 14:21.26 | sigmounte | grandpapadot, i want to change the "1193410606.2.wav" to something like "1193410606.%phonenumber%.wav |
| 14:25.19 | *** join/#asterisk karleeto (n=karl@private.franklincomputer.net) |
| 14:28.08 | karleeto | anyone in here experienced with aastra phones, could you please join #aastra? |
| 14:28.16 | karleeto | how is everyone this morning? |
| 14:28.21 | karleeto | good, i hope |
| 14:28.23 | karleeto | ! |
| 14:28.58 | *** join/#asterisk huey23 (n=huey23@64.192.209.132) |
| 14:29.57 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:30.08 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 14:30.09 | socken23 | tzafrif_home: OK, took a little longer. Here's the zapata.conf itself: http://pastebin.com/m3f689c55 |
| 14:30.31 | huey23 | [TK]D-Fender: i am pretty sure you dont like me because i don't look at the "blatenly obvious" but could you check something for me? |
| 14:31.11 | socken23 | tzafrif_home: and here's zapata-BRI-HFC.conf: http://pastebin.com/m44cdcd6 |
| 14:34.48 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 14:35.02 | [T]ank | anyone here an NFAS expert? |
| 14:36.20 | socken23 | tzafrir_home: any idea?? |
| 14:36.54 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:37.14 | coppice | what is there about NFAS that one could meaningfully be an expert on? :-\ |
| 14:37.42 | jameswf | I am an expert on powdered doughnuts |
| 14:38.08 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:38.15 | ai-a[wrk] | NFAS (National Field Archery Society) |
| 14:38.18 | coppice | no, no. too wide an area. is your key expertise the powder or the doughnut? |
| 14:38.24 | tzafrir_home | socken23, anything in zapata-auto.conf ? zapata_additional.conf? |
| 14:38.32 | *** join/#asterisk saftsack (n=saftsack@pD9E07FB9.dip.t-dialin.net) |
| 14:38.38 | tzafrir_home | Could you please paste them all to a single pastebin post? |
| 14:39.15 | [T]ank | coppice: I have a client with a creative set up on their t1s. I dont think I am getting it configured right. |
| 14:39.20 | socken23 | ah sorry.. no, only in zapata-auto.conf 'callerid=asreceived", the rest is empty |
| 14:39.26 | Darthclue | coppice, that is still too wide. please break down your expertise based on ingredients. |
| 14:39.37 | k31th | tzafrir_home: hows it going |
| 14:39.47 | jameswf | dude so I left arizona for a year and they shut down all the krispy kreme's here, this leads me to the conclusion my fat arse was there sole income :) |
| 14:39.52 | [T]ank | they have 4 PRIs... dchannel on only trunks 1 and 3 |
| 14:40.22 | coppice | I'm a doughnut consultant. nuts with enough dough you can hire me |
| 14:40.29 | [T]ank | so... because i have configured everything to have a dchan... I get a lot of stuff like: Ring requested on unconfigured channel 1/4 span 1 |
| 14:41.00 | jameswf | I think a node just pooped |
| 14:41.04 | jameswf | :)) |
| 14:41.04 | [T]ank | :-D |
| 14:41.08 | [T]ank | only for most |
| 14:41.41 | Darthclue | krispy kreme expanded too quickly into some areas. we used to have 2, now only have one and it is no longer convenient to stop at so i haven't had them in quite some time. |
| 14:42.07 | socken23 | tzafrir_home: So, any idea? Otherwise I'm leaving for the weekend :-D |
| 14:42.18 | coppice | Darthclue: your doughnuts certs will be lapsing |
| 14:43.16 | tzafrir_home | socken23, you have "signalling=bri_cpe_ptmp" |
| 14:43.28 | Darthclue | not a problem there. a client called on saturday and laughed throughout the entire call. the nuts have recertified my dough. |
| 14:43.33 | tzafrir_home | where did you set signalling=bri_cpe"? |
| 14:43.42 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:44.02 | *** join/#asterisk karleeto (n=karl@private.franklincomputer.net) [NETSPLIT VICTIM] |
| 14:44.16 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) [NETSPLIT VICTIM] |
| 14:44.40 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 14:45.28 | socken23 | tzafrir_home: in the first file, zapata.conf |
| 14:46.12 | socken23 | Ah, it's also in zapata-BRI-HFC.conf... but I commented that line out |
| 14:46.27 | tzafrir_home | but the last "signalling" line before the "channel" line decides |
| 14:47.09 | socken23 | right, but that's commented out now.. uploaded the version before the change |
| 14:47.29 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:47.31 | *** join/#asterisk ngsheaufeng (n=ngsheauf@211.24.186.35) |
| 14:47.59 | tzafrir_home | yet another good time to mention http://bugs.digium.com/9503 (separate sections in zapata.conf) |
| 14:48.33 | ngsheaufeng | greetings from malaysia. i need some help with a mysterious issues i have with TDM400P FXO module. appreciate if any advance user can assist |
| 14:49.28 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 14:49.30 | tzafrir_home | some of use primitive users might even help you |
| 14:49.48 | tzafrir_home | but this will only happen once you ask your question :-) |
| 14:50.01 | ngsheaufeng | i would like to know if TDM400P work with line that is using 24VDC instead of standard 48VDC? |
| 14:50.05 | socken23 | HAHA!!! It works again! |
| 14:50.15 | socken23 | checked all settings, restarted, perfect |
| 14:50.21 | socken23 | that means: WEEK END! :-) |
| 14:50.27 | socken23 | thanks for your help |
| 14:51.05 | *** join/#asterisk dwclarkNU (n=Administ@c-76-19-196-67.hsd1.ma.comcast.net) |
| 14:51.20 | tzafrir_home | 48V? Doesn't the card use 12V? |
| 14:51.21 | dwclarkNU | does softecho support echo cancellation over sip/iax trunks? |
| 14:51.25 | ngsheaufeng | we have set up asterisk 1.4 & confirm the PSTN is working with a direct line from a telco. |
| 14:51.27 | blitzrage | socken23: niiiice... weekend! |
| 14:51.43 | ngsheaufeng | i'm refering to the off hook voltage |
| 14:51.49 | ngsheaufeng | on hook rather |
| 14:51.49 | coppice | dwclarkNU: you can't echo cancel over VoIP paths |
| 14:53.01 | ngsheaufeng | let me explain my problem... |
| 14:53.44 | ngsheaufeng | when we connect to a direct line from telco... everything works, calling in and out to external number. |
| 14:54.08 | ngsheaufeng | but our telco recently introduce a new package, a Centrex-based solution |
| 14:54.11 | *** part/#asterisk huey23 (n=huey23@64.192.209.132) |
| 14:54.15 | sigmounte | ${TIMESTAMP}-${CALLERIDNUM})-${UNIQUEID} just give me --.wav , normal ? :) |
| 14:54.34 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 14:54.38 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 14:54.41 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) [NETSPLIT VICTIM] |
| 14:54.46 | ngsheaufeng | whenver we plug in that Centrex-based line, the line goes off hook immediately and external caller gest a busy tone all the time, even when the Asterks PC is off |
| 14:54.53 | socken23 | ok, see you guys! |
| 14:54.53 | Katty | blitzrage: blitzrage blitzrage do you know any commercial freepbx peoples. |
| 14:54.53 | Katty | blitzrage: who are nice. |
| 14:54.53 | *** part/#asterisk socken23 (n=azrael@ip-213-189-154-029.fix.magnet.ch) |
| 14:54.55 | Katty | blitzrage: and like muffins. |
| 14:55.03 | Katty | blitzrage: nice is a requirement. muffins optional. |
| 14:55.06 | jeremy_g | is it possible to make asterisk accept anonymous incoming calls |
| 14:55.14 | ngsheaufeng | yes it can |
| 14:55.17 | Katty | jeremy_g: iax can |
| 14:55.20 | jeremy_g | ngsheaufeng:how? |
| 14:55.22 | jeremy_g | sip calls |
| 14:55.23 | jeremy_g | not iax |
| 14:55.32 | Katty | i think you just make an anon sip account |
| 14:55.42 | Katty | if i remember correctly, there was an example commented out in the original sip.conf |
| 14:56.24 | *** join/#asterisk compiledkernel (n=compiled@65.202.115.2) |
| 14:57.05 | jeremy_g | Katty:it has to be without this anon account |
| 14:57.07 | ngsheaufeng | we found out that the standard on hook voltage should be 48V however the on hook voltage supplied via this Centrex line is at 24V. |
| 14:57.16 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:57.22 | Katty | oh |
| 14:57.29 | Katty | ^_- |
| 14:57.31 | Katty | i dunno then. |
| 14:57.37 | jeremy_g | ngsheaufeng:Centerex saving on the volts as well |
| 14:57.38 | jeremy_g | hehe |
| 14:58.13 | jeremy_g | ngsheaufeng:so is there a solution? anonymous asterisk |
| 14:58.17 | ngsheaufeng | that causes the problem... TDM400P remain offhook and line goes busy when Centrex is used |
| 14:58.50 | jeremy_g | like i just want to give my sip proxy address over the internet so that users can just make sip calls without requiring any authentication. |
| 14:58.51 | *** join/#asterisk DarkFlib (n=root@host90-152-23-30.ipv4.regusnet.com) |
| 14:59.30 | sigmounte | the ${} variables don't look to work in my extentions.conf , any idea ? |
| 15:00.02 | De_Mon | sigmounte you'll have to give us an example |
| 15:00.11 | DarkFlib | quick question... is there any way to use realtime extensions without having to specify the contexts in extensions.conf (either the real file or a static db setup)? |
| 15:00.16 | agx | anyone has an idea why i'm getting tri-plicated DTMF when calling my misdn IVR ? |
| 15:00.33 | sigmounte | De_Mon, /mnt/stockage/monitor/${TIMESTAMP}-${CALLERIDNUM(num)} send me a file like this : -.wav |
| 15:01.05 | De_Mon | sigmounte well ${TIMESTAMP} is a custom variable, does it exist? CALLERIDNUM() simply doesn't exist |
| 15:01.30 | DarkFlib | sigmounte: are you thinking of ${EPOCH} instead? |
| 15:01.49 | blitzrage | Katty: haha... sorry, I know of no such beasts |
| 15:02.01 | Katty | blitzrage: that's probably a good thing. |
| 15:02.09 | Katty | blitzrage: maybe i just need a blanket and muffin instead. |
| 15:02.11 | jeremy_g | anon-sip calls <-- i wanna know more about it |
| 15:02.14 | sigmounte | DarkFlib, ${EPOCH} ?? |
| 15:02.24 | blitzrage | Katty: perhaps... I could go for a muffin... I'm starving |
| 15:02.30 | Katty | blitzrage: ok! |
| 15:02.51 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) [NETSPLIT VICTIM] |
| 15:02.51 | *** mode/#asterisk [+o file] by ChanServ |
| 15:02.57 | Katty | oh |
| 15:02.58 | Katty | look |
| 15:02.59 | Katty | it's file |
| 15:02.59 | blitzrage | stop talking about file now |
| 15:03.09 | Katty | k |
| 15:03.18 | blitzrage | he gets violent |
| 15:03.18 | De_Mon | oh katty, remember this: http://www.dallasnews.com/sharedcontent/dws/news/localnews/stories/101607dnmetburglar.1767f026d.html |
| 15:03.28 | jeremy_g | but i cant believe file is such a man as you guys put it |
| 15:03.40 | Katty | De_Mon: i'm using irssi via a command prompt in windows... |
| 15:03.44 | blitzrage | he is a wolf in sheeps clothing |
| 15:03.46 | Katty | De_Mon: i'm not even going to try to copy and paste that. |
| 15:03.47 | De_Mon | I told you I would find the story a few days ago and got sidetracked |
| 15:03.49 | Katty | De_Mon: give me a refresh |
| 15:04.01 | De_Mon | oh katty, remember this: http://www.dallasnews.com/sharedcontent/dws/news/localnews/stories/101607dnmetburglar.1767f026d.html |
| 15:04.09 | DarkFlib | jeremy_g: basicly I don't want to have to do a reload if at all possible, its likely the config will grow pretty big... so having to specify a new context and reload is a pain... any ideas? |
| 15:04.11 | Katty | blitzrage: i know. |
| 15:04.17 | Katty | blitzrage: he wouldn't share his orange juice with me |
| 15:04.21 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 15:04.22 | blitzrage | that bastard! |
| 15:04.25 | De_Mon | http://tinyurl.com/24qh2u |
| 15:04.26 | Katty | hi Juggie |
| 15:04.29 | De_Mon | go with that |
| 15:04.31 | Juggie | silly netsplit |
| 15:04.39 | Katty | Juggie the candian. |
| 15:04.43 | jeremy_g | heheh |
| 15:04.47 | Katty | canadian. |
| 15:04.51 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 15:05.00 | Katty | can uh de un |
| 15:05.04 | blitzrage | <-- also Canadian |
| 15:05.09 | jeremy_g | DarkFlib:then why do you want a static ara for? |
| 15:05.12 | blitzrage | <-- getting screwed by the crappy USD |
| 15:05.13 | jeremy_g | it involves reloads |
| 15:05.15 | Katty | :< |
| 15:05.20 | Katty | blitzrage: i wanna move to canada. |
| 15:05.22 | file | moo |
| 15:05.24 | jeremy_g | haha |
| 15:05.27 | Katty | blitzrage: us is...eating my soul |
| 15:05.36 | blitzrage | Katty: I could use a room mate in Toronto |
| 15:05.46 | Katty | blitzrage: i dun think ryan would like that |
| 15:05.55 | De_Mon | take him with you |
| 15:06.12 | blitzrage | Ryan Seacrest?! |
| 15:06.13 | Katty | significant other |
| 15:06.19 | Katty | who would be moving with me |
| 15:06.19 | De_Mon | lol |
| 15:06.36 | Katty | just depends on the person. |
| 15:06.40 | blitzrage | fair enough |
| 15:06.42 | DarkFlib | jeremy_g: since we need to specify new contexts and to my knowledge dynamic RT doesn't support the use of contexts that aren't already specified in the extensions.conf |
| 15:06.45 | Katty | but i've dated girls before, you're right |
| 15:06.55 | blitzrage | oh ok, so I'm not totally insane |
| 15:07.02 | Katty | no, no you are totally insane. |
| 15:07.08 | ngsheaufeng | Does TDM400P FXO module work with Centrex line which uses 24VDC instead of 48VDC as on hook voltage? |
| 15:07.10 | blitzrage | oh good -- I was scared for a minute |
| 15:07.18 | Katty | blitzrage: but i still love you |
| 15:07.20 | Juggie | blitzrage, your still insane. |
| 15:07.21 | jeremy_g | DarkFlib:how do you plan to add the new context if you dont want to change the config files |
| 15:07.25 | blitzrage | Juggie: you're* |
| 15:07.32 | blitzrage | Katty: ok sweet -- good to know! |
| 15:07.35 | Juggie | and i have the pictures to prove it :) |
| 15:07.41 | blitzrage | Juggie: :D |
| 15:07.45 | Katty | blitzrage: i've read that you have to live in canada for 4 years before you apply for citizenship |
| 15:07.51 | Katty | blitzrage: and pass a Look At Canada test. |
| 15:08.00 | Katty | blitzrage: which i presume is full of pictures and shiny maps. |
| 15:08.07 | blitzrage | Katty: I think that is true -- my friend Seung-Hoon Song just got citizenship |
| 15:08.09 | DarkFlib | jeremy_g: thats why I'm asking... I noticed a couple of patches to allow bypass switches and similar hacks but I was wondering if there is a proper way to do it |
| 15:08.09 | Katty | blitzrage: but i didn't really see a Skilled Work requirement. |
| 15:08.21 | Katty | blitzrage: so... i just gotta live there and pass a test? |
| 15:08.21 | rpyne | exit |
| 15:08.24 | blitzrage | Katty: someone sent me a link the other day for that... so it does exist |
| 15:08.30 | blitzrage | Katty: pretty much! |
| 15:08.32 | Katty | blitzrage: that's crazy mad. |
| 15:08.36 | *** join/#asterisk ManxPower (n=manxpowe@82.sub-70-221-74.myvzw.com) |
| 15:08.38 | blitzrage | Katty: and you can keep US citizenship I htink too |
| 15:08.39 | Katty | blitzrage: i should just file for refuge status |
| 15:08.40 | blitzrage | not sure.... |
| 15:08.45 | blitzrage | Katty: lol -- totally |
| 15:08.48 | Katty | heeelp bush eating my soul |
| 15:08.54 | irule | is it possible to hav extension numbers into an array and handle them in a macro? |
| 15:09.04 | blitzrage | irule: depends what you mean by an array |
| 15:09.14 | Katty | irule: i could but then onto an array. |
| 15:09.16 | jeremy_g | DarkFlib:with static ara, you would still have to reload |
| 15:09.17 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:09.18 | Katty | irule: a redundant array |
| 15:09.26 | Katty | irule: of disks. |
| 15:09.30 | Katty | irule: perhaps mirror? :P |
| 15:09.31 | blitzrage | there isn't really a true array in the dialplan... but you can simulate it |
| 15:09.37 | DarkFlib | jeremy_g: yes I'm aware of that... |
| 15:09.47 | De_Mon | ARRAY() isn't a true array? |
| 15:09.55 | jeremy_g | blitzrage:is canada any good? i have a chance to immigrate to it quite easily. |
| 15:10.00 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:10.04 | jeremy_g | its damn cold i heard |
| 15:10.05 | blitzrage | no... Canada is crappy -- stay home |
| 15:10.10 | irule | katty are you a beautyful blonde? or hale barry type? |
| 15:10.16 | Juggie | haha |
| 15:10.19 | Juggie | yes, stay the hell out |
| 15:10.21 | Corydon76-dig | No, ARRAY() is not a true array |
| 15:10.25 | Katty | uhm. |
| 15:10.36 | Katty | irule: if i say hairy and ugly, will you leave me alone? |
| 15:10.37 | *** part/#asterisk ngsheaufeng (n=ngsheauf@211.24.186.35) |
| 15:10.39 | Corydon76-dig | It's a receiver struct |
| 15:10.43 | jameswf | or a mullet wearing truck driver |
| 15:10.45 | irule | lol |
| 15:10.48 | blitzrage | Corydon76-dig: correct :) But I simulate it with a While() loop and VARNAME_X where X is the position of the array |
| 15:10.58 | Juggie | we like you guys better when you stay south of us |
| 15:11.03 | jeremy_g | blitzrage:yeahs that what i thought already |
| 15:11.03 | irule | Katty just wondering |
| 15:11.12 | blitzrage | jeremy_g: and it's freezing and never sunny |
| 15:11.15 | blitzrage | all year round |
| 15:11.28 | Corydon76-dig | blitzrage: what happens when you want to remove an element in the middle of the array? :-P |
| 15:11.34 | Juggie | jeremy_g, ya, its constantally at freezing... if it gets warm my igloo will melt. |
| 15:11.38 | blitzrage | Corydon76-dig: ummm... you don't :) |
| 15:11.39 | Corydon76-dig | or at the beginning, even? |
| 15:11.46 | DarkFlib | blitzrage: could be worse... might be wet and never sunny all year round... like UK |
| 15:11.52 | blitzrage | DarkFlib: true true |
| 15:11.53 | jeremy_g | Juggie:hehe |
| 15:12.08 | blitzrage | Juggie: and the snow snakes get really angry |
| 15:12.35 | jeremy_g | Juggie:yeah, i have some relatives in tronto. nice igloos they got |
| 15:12.44 | blitzrage | my igloo is on the 16th floor |
| 15:12.46 | blitzrage | :) |
| 15:12.46 | jameswf | I have been in telephony for 10 years havent met a hot female tech ever in those 10 years |
| 15:12.46 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:12.55 | De_Mon | wow xbox division posts Q1 profit |
| 15:13.08 | jeremy_g | jameswf:you should watch the astricon pics featuring katty |
| 15:13.08 | De_Mon | did hell just freeze over? |
| 15:13.17 | De_Mon | jeremy_g link? |
| 15:13.35 | jeremy_g | De_Mon: "Now, I can't do it for you." |
| 15:13.36 | DarkFlib | I met one at VON:Sweden... she wanted to meet up in VON:Italy, shame I wasn't going... :( |
| 15:13.41 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 15:14.03 | jameswf | normaly the hot ones are in sales....hmmm |
| 15:14.11 | jeremy_g | DarkFlib: was she working for the pulvermedia :D |
| 15:14.12 | jameswf | scared her off :) |
| 15:14.27 | DarkFlib | nope... although thats rather suprising I think |
| 15:14.52 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 15:15.03 | jeremy_g | jameswf: beauty, brains, personallity,availability (pick any two) :D |
| 15:15.12 | jameswf | ok since she isnt in here, I have had the mispleasure of dealing with her... she isnt that bright |
| 15:15.14 | blitzrage | lates all, I'm off to find a halloween costume for the bar tonight |
| 15:15.34 | jameswf | but most "installers" arent |
| 15:15.49 | jameswf | damn windows people polutingth gene pool |
| 15:16.00 | DarkFlib | anyway... I need to find a solution to my problem... so later all... |
| 15:16.30 | jeremy_g | asterisk still cant take anonymous incoming calls. please update me if thats wrong |
| 15:16.37 | jameswf | solutions are $2.50, correct solutions are gravely more expensive |
| 15:16.53 | Corydon76-dig | "dumb looks are free" |
| 15:17.00 | jeremy_g | jameswf: is there a dot in the 250 there |
| 15:17.02 | jameswf | all day :) |
| 15:17.04 | jeremy_g | :D |
| 15:17.45 | DarkFlib | $2.50? thats less than a half a pint of beer.... :P |
| 15:17.46 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 15:17.58 | jameswf | unless you drink bum beer |
| 15:18.00 | jameswf | :) |
| 15:18.08 | jameswf | steele reserve mmmmmm |
| 15:18.12 | Corydon76-dig | One of my grade school teachers had a sign on her desk... "Answers - $1. Answers that require some thought - $2. Correct answers - $4. Dumb looks - Free." |
| 15:18.18 | *** join/#asterisk codec (n=codec@iglu.paranoid-penguin.de) |
| 15:18.20 | billybongo | how can i play a sound file from a blob in mysql? |
| 15:18.24 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:18.30 | dijungal | hey |
| 15:18.49 | dijungal | is there a cli command with will show me the amt. of available agents? |
| 15:18.57 | jameswf | billy, a recorded sound or a sound representation of the data |
| 15:19.14 | dijungal | available to take a call, meaning not on pause and logged in |
| 15:19.15 | DarkFlib | When I went to .ro the other year, I was drinking Aurora beer for 0.20EUR/pint.... |
| 15:19.21 | Corydon76-dig | dijungal: 'queue show <name>' |
| 15:19.30 | *** join/#asterisk linuxmaniac (n=maniac@193.153.218.212) |
| 15:19.38 | DarkFlib | now that was bad... but after the first few you didn't really care |
| 15:19.48 | dijungal | queue show is not a command |
| 15:20.14 | Corydon76-dig | Are you using 1.4? |
| 15:20.22 | dijungal | 1.2.19 |
| 15:20.38 | Corydon76-dig | Well, there's one problem. 'show queue <name>' |
| 15:21.21 | dijungal | yep |
| 15:21.26 | dijungal | i guess that's my option |
| 15:21.27 | dijungal | ok thanks |
| 15:21.39 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 15:22.00 | billybongo | jameswf a recorded sound |
| 15:23.08 | billybongo | so i record a sound, upload to mysql and play it in * |
| 15:24.20 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 15:24.28 | jameswf | you save the sound to /path and add path to database |
| 15:25.04 | brodiem | Guys..anyone know what the best way might be to launch an application triggered by a queue agent answering a call (or when they hangup, whichever is easier)? It would be a help if it could also utilize a previously set variable from the dial plan |
| 15:25.12 | jeremy_g | um using 1.2 |
| 15:25.19 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:25.45 | billybongo | jameswf, i realise i could do it that way, but I'd rather have the sound itself in mysql because I might need to play it from a number of different asterisks |
| 15:26.59 | billybongo | I wonder if the voicemail functions are available at a lower level |
| 15:28.21 | *** join/#asterisk cl4sh (n=cl4sh@syf.ds.pg.gda.pl) |
| 15:31.51 | *** join/#asterisk afrosheen (n=cj@207.71.49.137) |
| 15:32.26 | afrosheen | one of my users is having his bank do a callback on his DID here...when he picks up it asks him to hit # to verify. Hitting # tells asterisk he wants to do a transfer. Workarounds? |
| 15:35.05 | afrosheen | boy is it ever dead in here today |
| 15:35.20 | *** join/#asterisk Egonis (n=roman@207.245.216.9) |
| 15:35.35 | Egonis | How can I test an outgoing channel from a T1 from the Asterisk CLI? |
| 15:35.45 | ManxPower | afrosheen: stop using t/T |
| 15:35.53 | Egonis | i.e. I want to Dial(Zap/G1/Number) |
| 15:35.55 | agx | anyone has a tips about duplicated DTMF when calling from mobile --> ISDN ? |
| 15:36.11 | afrosheen | ManxPower, thanks, I'll take a look at that |
| 15:36.18 | ManxPower | afrosheen: at least the "T", which allows the caller to transfer the call. Do you really want to do that? |
| 15:36.30 | *** join/#asterisk minkus (n=minkus@doc-24-32-190-194.bh.wv.cebridge.net) |
| 15:36.31 | ManxPower | agx: relaxdtmf=no |
| 15:36.37 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 15:36.44 | lirakis | do i need to restart asterisk for new MOH files to take effect? |
| 15:36.52 | ManxPower | Egonis: only if you have a soundcard and alsa-dev installed when you built Asterisk |
| 15:37.08 | ManxPower | lirakis: mp3 MoH or other format? |
| 15:37.20 | Egonis | ManxPower: I am doing this remotely, still matters, right? |
| 15:37.38 | lirakis | ManxPower: i am converting the MOHmp3 files to ulaw |
| 15:37.45 | ManxPower | Egonis: kind of hard to test if if you are not at the console. |
| 15:37.52 | Egonis | ManxPower: Well, I'm SSH'ed |
| 15:38.11 | ManxPower | lirakis: I suspect non-mp3 MoH will update with just a reload |
| 15:38.12 | Egonis | ManxPower: What if I called into the T1, and had it automatically dial out through another channel on the same span? |
| 15:38.32 | ManxPower | Egonis: The soundcard will be used for the call, if you are remote, you can't speak into the microphone or listen to the speakers. |
| 15:38.41 | *** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
| 15:38.41 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-132-60.neo.res.rr.com) |
| 15:38.42 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 15:38.42 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 15:38.42 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 15:38.55 | lirakis | ManxPower: oaky.. so .. if i want to stop using the mp3MOH files .. and switch to the ulaw files .. ? |
| 15:39.12 | afrosheen | ManxPower, thanks for the tip, I changed the outgoing dial string from trw to rw, should be good now :) |
| 15:39.29 | ManxPower | afrosheen: you don't have any clue what those parameters do, do you? |
| 15:39.38 | Egonis | In zapata.conf, should trunkgroups precede channels? I tried specifying channels in a span into a group, but 'group show channels' returns nothing |
| 15:39.51 | afrosheen | ManxPower, actually I do, but this is the first time this issue has sprung up in years of usage |
| 15:39.55 | agx | ManxPower, its already relaxdtmf=no; btw its on incoming calls not outgoing :) |
| 15:40.29 | ManxPower | agx: OK. You may want to play with the rxgain then. up it by 2db first, if that doesn't work, lower it by 2db |
| 15:40.55 | ManxPower | afrosheen: So you don't really know how many people may have used your PBX to make free calls. |
| 15:41.30 | afrosheen | ManxPower, I never had a Trw, it has always been trw, so yeah, I don't know because the number is likely to be zero |
| 15:41.56 | ManxPower | t can cause problems as well, depending on the direction of the Dial. |
| 15:42.09 | afrosheen | mmm hmm, I see what you tried to do there |
| 15:42.12 | ManxPower | also with "r" you could hear a ringing, followed by a busy under some situations. |
| 15:42.15 | agx | ManxPower, good idea :) |
| 15:43.53 | agx | ManxPower, found, seems that with faxdetect=yes in misdn.conf there is this problem... |
| 15:44.11 | ManxPower | agx: Ah. I never, ever use faxdetect |
| 15:44.31 | ManxPower | I use NVFaxDetect actually |
| 15:45.07 | agx | me too but wanted to give it a try since it does not force you to answer the channel :) |
| 15:46.12 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 15:47.08 | *** join/#asterisk blq (n=Bl@dslb-088-066-243-015.pools.arcor-ip.net) |
| 15:47.33 | tzafrir_home | Egonis, the order between [trunkgroups] and [channels] is irrelevant |
| 15:48.30 | ai-a[wrk] | Is connecting line presentation implemented yet ? as in when a call's transfered the display of the phone updates when the call is finally transfered ? |
| 15:48.58 | *** join/#asterisk lawrencej (n=lawrence@s142-179-213-37.ab.hsia.telus.net) |
| 15:49.06 | lawrencej | can someone help me ? |
| 15:49.16 | lawrencej | I'm not sure if that is possible |
| 15:49.25 | lawrencej | ? |
| 15:49.41 | ai-a[wrk] | ask something, else its impossible. |
| 15:49.42 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 15:49.48 | lawrencej | help ? |
| 15:49.57 | ai-a[wrk] | nope, cant help you. |
| 15:50.03 | lawrencej | :( |
| 15:50.10 | tzafrir_home | ~ask |
| 15:50.10 | jbot | 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. |
| 15:51.20 | *** part/#asterisk Egonis (n=roman@207.245.216.9) |
| 15:51.35 | lawrencej | Ok I just purchased a a tdm 2400 with a voice echo mod and 16 analog lines. when I make a call thew the box every 5 sec thier is a noise a click like someone is hanging up. Any Ideas? |
| 15:51.54 | ai-a[wrk] | DarkFlib :) |
| 15:52.02 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 15:52.07 | DarkFlib | you're following me... |
| 15:52.08 | DarkFlib | :P |
| 15:52.10 | ai-a[wrk] | Heh. |
| 15:52.39 | ai-a[wrk] | you know about this presentation of phone number when transfered? |
| 15:52.59 | tzafrir_home | lawrencej, a call from where to where? |
| 15:53.05 | DarkFlib | transfered to what kinda phone? |
| 15:53.33 | lawrencej | my cell to the box which is forwards to my soft phone on my laptop |
| 15:53.52 | DarkFlib | so you want to keep the original callerid? |
| 15:53.55 | lawrencej | I have setup one box with a tdm 400 no problems |
| 15:54.02 | agx | ManxPower, where did you tooked NV..Detect? yourself or there is an extra package with other GPL apps? |
| 15:54.29 | tzafrir_home | lawrencej, is the tdm400 card still in that box? |
| 15:54.35 | lawrencej | no |
| 15:54.41 | lawrencej | this is a new box |
| 15:54.48 | tzafrir_home | is ztdummy loaded? |
| 15:55.00 | lawrencej | let me check |
| 15:55.12 | tzafrir_home | lsmod | grep ^zaptel |
| 15:55.16 | *** join/#asterisk pepse (n=pepse@71-223-117-66.phnx.qwest.net) |
| 15:56.11 | *** join/#asterisk wunderkin (n=wunderki@ip72-208-16-104.ph.ph.cox.net) |
| 15:56.41 | lawrencej | zaptel 186532 59 zttranscode,ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm24xxp |
| 15:57.01 | lawrencej | yes ? |
| 15:58.35 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 15:59.11 | *** part/#asterisk dwclarkNU (n=Administ@c-76-19-196-67.hsd1.ma.comcast.net) |
| 16:00.42 | lawrencej | The weird thing is it happens the exact same time over and over again no mater if it phone to soft phone. I don't think it is network related because i conect to my 400 box over a vpn and the audio is fine. So i Believe i can remove the laptop soft phone out of the problem |
| 16:01.00 | lawrencej | tzafrir_home ? |
| 16:01.22 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 16:02.30 | tzafrir_home | lawrencej, is /proc/zaptel/1 a ztdummy span? |
| 16:03.06 | tzafrir_home | if so: try: rmmod ztdummy |
| 16:03.07 | lawrencej | ok now your over my head |
| 16:03.20 | tzafrir_home | and see if there's any improvement |
| 16:03.42 | lawrencej | how to i find out if ztdummy is a span ? |
| 16:04.39 | tzafrir_home | cat /proc/zaptel/1 . Is it a short file that mentions ZTDUMMY? |
| 16:06.13 | lawrencej | does not say anything about ztdummy |
| 16:06.22 | lawrencej | but it does say IRQ misses: 1085355 |
| 16:06.26 | lawrencej | at the top |
| 16:06.36 | lawrencej | right after span 1 |
| 16:06.51 | lawrencej | Span 1: WCTDM/0 "Wildcard TDM2400P Board 1" |
| 16:06.51 | lawrencej | <PROTECTED> |
| 16:07.02 | lawrencej | then it lists all the channels |
| 16:08.29 | Zeeek | Folks in a few moments, the weekly voIP conference, mostly about asterisk is happening. http://x2z.eu/voip for info |
| 16:08.32 | lawrencej | should i still try to rmmod ztdummy ? |
| 16:10.52 | Zeeek | .. |
| 16:11.27 | lawrencej | tzafrir_home ? |
| 16:12.15 | Zeeek | I'll shut up in 1sec, but if you are interested in the voIP Users Conference, /join #voip-users-conference - ok that's all :) |
| 16:12.23 | ManxPower | lawrencej: if you have that many IRQ misses, you have some massivly totally major super weird problems |
| 16:12.32 | tzafrir_home | lawrencej, irq misses are not my field. I guess it's tiem to contact Digium support |
| 16:13.41 | lawrencej | eh ? |
| 16:14.12 | lawrencej | so you think the problem with irq ? |
| 16:14.59 | Corydon76-dig | There's probably something else on your system (like a RAID card) which is eating interrupts |
| 16:15.02 | JT | it says irq misses, what do you think |
| 16:16.33 | ManxPower | The IRQ misses problem must be fixed before you do anything else with Asterisk. |
| 16:19.22 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 16:19.58 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:20.25 | *** join/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
| 16:20.39 | ct2clay | does anyone know of any free ACD managers for trixbox CE? |
| 16:21.14 | Qwell | ct2clay: see topic |
| 16:21.32 | *** join/#asterisk thx2000 (n=the@netblock-208-127-150-56.dslextreme.com) |
| 16:21.49 | ct2clay | where? |
| 16:21.54 | Qwell | /topic |
| 16:21.55 | ct2clay | oh |
| 16:21.56 | ct2clay | sorry |
| 16:24.25 | anonymouz666 | codefreeze: my CDR got crazy when using SIP Refer messages ;) |
| 16:25.02 | grandpapadot | Hey guys, I have an 8-core server running a asterisk 1.2.24, it processes about 100 calls a minute or so doing some transcoding. We're not having any issues, but I noticed that only CPU0 gets the bulk of the threads. Is the threading model in 1.4 the same as in 1.2? |
| 16:26.02 | grandpapadot | CPU0 will hover at about 10-20% while the other cores never really get above 3%. |
| 16:30.30 | TrentCreek | no answers at the web site? |
| 16:33.07 | TrentCreek | I have never heard of an 8 core CPU? I though only 4 existed and Intel suppose to come out witha 6 core next year |
| 16:33.39 | grandpapadot | Quad dual-cores or dual-quad cores, dunno, hosted |
| 16:34.05 | agx | grandpapadot, you should consider providing some CPU cycle for seti@home too :-P |
| 16:34.22 | grandpapadot | On a production system? uhm, no |
| 16:34.36 | grandpapadot | Unless the alien's are subscribing to our service .. |
| 16:34.46 | TrentCreek | I have seen some poeple providing with 16 CPU systems. |
| 16:34.57 | grandpapadot | But the latency would be really low as the best we could hope for would be packet radio.. |
| 16:35.00 | codefreeze | anonymouz666: not surprised. file a bug. I'll try to do something about it when I'm done with zaptel transfers. Just one warning: if the phone does something on its own, and doesn't tell asterisk, there's nothing I'll be able to do. |
| 16:35.45 | grandpapadot | Then there's the whole inter-world licensing thing, I don't think the GPL has a clause for that... |
| 16:36.16 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 16:36.24 | anonymouz666 | codefreeze: where I can get the CDR presentation that you did for Astricon? |
| 16:36.56 | codefreeze | anonymouz666: have you checked the astricon web site? If it's not there, I'll kick the astricon maintainers. |
| 16:38.15 | anonymouz666 | yes I am blind then. Can't find |
| 16:39.16 | *** join/#asterisk davefisher (n=davefish@c-66-30-12-107.hsd1.ma.comcast.net) |
| 16:41.16 | *** join/#asterisk mainproc (n=kvirc@87.69.28.50.cable.012.net.il) |
| 16:41.21 | mainproc | hello all |
| 16:42.04 | mainproc | any1 here? |
| 16:43.16 | ai-a[wrk] | mainproc: what do you think ? |
| 16:43.30 | *** join/#asterisk pepo-- (n=pepOSX@190.72.31.122) |
| 16:43.35 | mainproc | great |
| 16:43.45 | mainproc | can u give me some help with setting trunks? |
| 16:44.10 | *** join/#asterisk mkl1525 (n=mkl1525@212.204.47.147) |
| 16:44.14 | anonymouz666 | codefreeze: and in this case, the refer message hits asterisk. refer-to and referred-by |
| 16:44.49 | mkl1525 | Hi, is there some watchdog tool that restarts * when it crashed? |
| 16:45.01 | anonymouz666 | but the behaviour is very strange. |
| 16:45.06 | ai-a[wrk] | mainproc: trunks ? as in extensions ? as in set up how ? |
| 16:45.14 | anonymouz666 | sometimes I got two legs recorded, sometimes no... |
| 16:46.04 | ManxPower | mainproc: there is no such thing as a trunk in Asterisk. |
| 16:46.35 | ai-a[wrk] | He just pm'ed me. |
| 16:46.37 | ai-a[wrk] | grrr. |
| 16:47.02 | codefreeze | anonymouz666: OK, then I should be able to do something. I've been noticing timing related effects on CDR's in my hacking... so either your situation is timing related, or it's just two different but similar things giving vastly different results. |
| 16:47.28 | mainproc | there is on freepbx |
| 16:47.52 | Qwell | mainproc: no there isn't |
| 16:48.07 | Qwell | they CALL it a trunk, but it most definitely is not one |
| 16:48.14 | mainproc | ok |
| 16:48.19 | mainproc | can u help me setup one? |
| 16:48.46 | ai-a[wrk] | a trunk belongs to an elephant. |
| 16:49.00 | ai-a[wrk] | mainproc: how do you want help ? |
| 16:49.16 | mainproc | just help me setup one trunk |
| 16:49.18 | mainproc | can u? |
| 16:49.26 | ai-a[wrk] | want us to come around and plug in your devices ? identify them on your network ? set up their sip config ? write context to handle internal dialing? |
| 16:49.39 | ai-a[wrk] | mainproc: have you read the asterisk book ? |
| 16:49.58 | mainproc | yeah, but i'd like some1 to do the 1st with me |
| 16:50.13 | ai-a[wrk] | ok, well i dont get paid for this so i will leave you with the book. |
| 16:50.30 | anonymouz666 | codefreeze: XXX -> calls YYY. INVITE->200OK one leg here. After YYY send a refer to asterisk with 'refer-to: ZZZ' and 'referred-by: XXX' (another call here). |
| 16:50.38 | ai-a[wrk] | the first 5 pages in the book set up two ext with a context and explain how to do it. |
| 16:50.39 | anonymouz666 | and then you hangup. |
| 16:51.36 | anonymouz666 | cdr is like: something we have XXX to YYY. and then XXX to ZZZ. sometimes just XXX to ZZZ. |
| 16:52.49 | mainproc | ok |
| 16:52.58 | mainproc | where can i get those sip trunks? |
| 16:53.06 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 16:54.14 | xai | Can I reload asterisk config files from the console, or do i need to restart * ? |
| 16:54.14 | *** join/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 16:54.25 | codefreeze | anonymouz666: make sure to put all that detail into the bug report! ok? |
| 16:54.33 | HarryR`Work | xai, yes, reload :) |
| 16:54.51 | ai-a[wrk] | mainproc: STOP saying trunk. |
| 16:55.03 | ai-a[wrk] | mainproc: read the book and follow its example.. |
| 16:55.08 | ai-a[wrk] | what do you want from us ? |
| 16:55.19 | ai-a[wrk] | you want someone to do it for you, you need to pay for it. |
| 16:55.28 | mainproc | are there free sip proxies that will let me call pstn phones? |
| 16:55.41 | ai-a[wrk] | mainproc: google.com |
| 16:55.49 | geminidomino | Does asterisk 1.4.13 include the nvfaxdetect and spandsp for fax-to-email setups, or are those still addons? |
| 16:55.51 | mainproc | what to look? |
| 16:56.46 | *** part/#asterisk mkl1525 (n=mkl1525@212.204.47.147) |
| 16:57.15 | ai-a[wrk] | mainproc: i give in. |
| 16:57.25 | mainproc | good |
| 16:57.29 | mainproc | u didn't help anyway |
| 16:57.35 | *** join/#asterisk blq (n=no@dslb-088-066-234-029.pools.arcor-ip.net) |
| 17:00.21 | geminidomino | Do I still need to install nvfaxdetect and spandsp? All the info I can find that says so is pretty old, and freepbx seems to have options for them built in, thus my confusion |
| 17:00.35 | ManxPower | geminidomino: no and it never will. |
| 17:01.00 | geminidomino | ManxPower: Ok. Thanks. That could explain why I can't get anything working. :) |
| 17:01.25 | anonymouz666 | codefreeze: OK! what about the presentation? |
| 17:07.24 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 17:09.11 | *** join/#asterisk roadies (n=chrislro@stun.cmf.nrl.navy.mil) |
| 17:10.10 | roadies | It seems I'm missing all the latest sound files for my Asterisk Appliance, is there a URL pointing to a atr file of all the gsm files? |
| 17:12.26 | xai | HarryR: thanks.. does reload wipe out any removed configurs that I delete? |
| 17:13.38 | mainproc | is there any free sip to pstn so i can test? |
| 17:15.32 | xai | For some reason the 2nd edtion orielly book doesnt have a synopsis of the console commands |
| 17:15.56 | Qwell | xai: type help, that'll give you everything |
| 17:16.04 | Qwell | (yes, seriously) |
| 17:16.56 | xai | ok, thanks. |
| 17:17.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:20.01 | codefreeze | anonymouz666: the presentation isn't in the astricon web site? |
| 17:20.14 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-235-73.lv.lv.cox.net) |
| 17:22.02 | anonymouz666 | codefreeze: I can't find. |
| 17:25.10 | *** join/#asterisk Op3r (n=edwin@121.97.179.227) |
| 17:26.18 | codec | anyone running a junghanns duogsm card with 2 gsm/umts sims? |
| 17:26.44 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:27.58 | irule | ive seen the asterisk appliance, and atom's IP04, are there any other models out there that deliver * in small factor and low price? |
| 17:29.48 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-81-172.bstnma.east.verizon.net) |
| 17:32.41 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 17:33.47 | Yourname`` | Hi. So when I say "show modules" and it lists all enabled and working modules. I would like to disable a few modules, would it impact resources? |
| 17:34.32 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:34.36 | generalhan | hey all ! |
| 17:35.01 | generalhan | is there anyway that i can dial an Agent, rather than an Extension from the dialplan? |
| 17:36.43 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:38.53 | [T]ank | I am still trying to figure out a strange issue. I am getting error messages on my asterisk CLI> that say that the call recordings are failing. However I am seeing the call recordings. Here is what I am seeing: http://pastebin.ca/750627 |
| 17:38.53 | [T]ank | Any help would be appreciated. |
| 17:40.50 | *** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
| 17:41.12 | Darthclue | [T]ank, Verify that you have sox and nice installed. |
| 17:42.19 | Darthclue | generalhan, depends on how you have agents configured. take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+Agent+channels |
| 17:42.58 | Falkon_MX-5 | This is my first time setting up a termination. I have a section in extensions.conf given to me by vitelity that is supposed to define outgoing local calls. It works if I dial as if I were calling long distance, but it doesn't work for local calls. |
| 17:43.13 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:43.43 | Falkon_MX-5 | The line is exten => _011.,1,Dial(IAX2/falkon@vitel-outbound/${EXTEN}) | this is not working correctly. |
| 17:44.53 | Maliuta | Falkon_MX-5: you have an entry for them in the iax.conf? |
| 17:44.58 | Falkon_MX-5 | yes |
| 17:45.07 | Falkon_MX-5 | it works if I dial 1-xxx-xxxx |
| 17:45.16 | Yourname`` | So weird. All other Asterisk installations doing the SAME thing use about 10-12% CPU, and then there is this one install that is around 25% CPU usage. And I can't figure out why. No transcoding being done.. what else could it be? |
| 17:45.18 | Falkon_MX-5 | I just don't know how to make local hcalls |
| 17:46.04 | generalhan | Darthclue: thanks for the reply ... I should have just tried it before i asked though ... cause it seems to work just fine by using Dial(Agent/555) just like it does with SIP/555 |
| 17:46.33 | Darthclue | generalhan, yvw |
| 17:47.05 | *** part/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 17:47.15 | generalhan | now if i can just figure out a magical way to seperate Agent recordings by Agent number ill be all set ! |
| 17:47.33 | *** join/#asterisk gardo (n=gardo@121.97.104.75) |
| 17:47.57 | generalhan | its terrible that i can use ${DATE} and ${EXTEN} in agents.conf |
| 17:48.30 | Maliuta | Falkon_MX-5: you should be using the reference from your iax.conf: exten => _011.,1,Dial(IAX2/<name in iasx.conf>/${EXTEN}) |
| 17:48.54 | Maliuta | Falkon_MX-5: and you should know that it will send exactly the numbers you dial |
| 17:49.12 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 17:49.13 | generalhan | s/can/cant/ |
| 17:50.22 | Falkon_MX-5 | Maliuta: How do I separate internal lines from external so asterisk doesn't think I just dialed a bad extension? |
| 17:51.25 | Maliuta | Falkon_MX-5: that doesn't make sense to me. do you mean inbound from outbound calls? |
| 17:52.02 | Falkon_MX-5 | no, internal extensions from outgoing calls |
| 17:52.22 | [TK]D-Fender | Falkon_MX-5: Thats what CONTEXTS are for. Go re-read chapter 5 of.... THE BOOK. |
| 17:52.24 | [TK]D-Fender | ~book |
| 17:52.25 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:52.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:52.41 | Falkon_MX-5 | it wouldn't use the context [outgoing] without me including it from default. It wouldn't do it |
| 17:53.15 | *** join/#asterisk Joe_CoT (i=joe_cot@ubuntu/member/joeterranova) |
| 17:53.26 | Joe_CoT | hey, having some problems with System Recordings. Xorcom updated asterisk on me, and now my ivr message no longer works. I'm having trouble finding out where it should be on the system, and asterisk doesn't seem to be returning any errors while playing it. ideas? |
| 17:53.28 | Darthclue | generalhan, are you using chan_agent to manage your agents? |
| 17:54.14 | *** join/#asterisk trippsss (n=ss@ASA-ParksLuttrell.phonoscope.com) |
| 17:55.19 | Maliuta | Falkon_MX-5: I use contexts and macros -> http://pastebin.com/d4f848997 |
| 17:55.33 | Maliuta | I onbly removed a few numbers from it |
| 17:55.51 | Falkon_MX-5 | thanks, that will help |
| 17:56.46 | Maliuta | not the prettiest out there, but it works |
| 17:57.40 | Falkon_MX-5 | Maliuta: yours is doing exactly what I want mine to do with the routing. |
| 17:57.56 | *** join/#asterisk becks` (n=flux_@171-173.5-85.cust.bluewin.ch) |
| 17:58.09 | [TK]D-Fender | Joe_CoT: Sorry, GUI's aren't supported here |
| 17:58.40 | Joe_CoT | [TK]D-Fender, i'm not asking you to support a gui. My main question is if there's any way to debug why asterisk isn't playing a sound file |
| 17:58.46 | becks` | hi, slightly offtopic... somebody knows if the VoIP Testing Consortium? have they ever released their test cases to the public? |
| 17:59.06 | [TK]D-Fender | Joe_CoT: Then pastebin smething useful for us to examine. |
| 17:59.08 | [TK]D-Fender | ~pb |
| 17:59.09 | 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 |
| 17:59.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 17:59.12 | ManxPower | Joe_CoT: not with a GUI there isn't, because the GUI screws up the dialplan so badly it is virtually impossible to troubleshoot. |
| 17:59.24 | ManxPower | ~gui |
| 17:59.24 | jbot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 17:59.34 | ManxPower | ~trixbox |
| 17:59.35 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 17:59.43 | Joe_CoT | k. bye. |
| 17:59.44 | *** part/#asterisk Joe_CoT (i=joe_cot@ubuntu/member/joeterranova) |
| 17:59.44 | zerohalo | 603 is up... |
| 18:00.03 | ManxPower | They seem to unhappy when they realize this is the wrong channel for them. |
| 18:04.23 | *** part/#asterisk roadies (n=chrislro@stun.cmf.nrl.navy.mil) |
| 18:05.13 | Falkon_MX-5 | Maliuta: may I msg you a question about your paste? |
| 18:05.30 | *** join/#asterisk horsesgofaster (n=dcantera@pool-72-82-218-77.cmdnnj.east.verizon.net) |
| 18:06.16 | [T]ank | Darthclue: Sox and Nice are in fact installed |
| 18:07.28 | Yourname`` | Hi. What causes Asterisk to increase CPU load. What factors lead to it? |
| 18:09.45 | denon | Yourname``: transcoding |
| 18:09.58 | irule | ~help |
| 18:10.29 | irule | how may I take a look at the full list of ~subjects to ask the bot? |
| 18:11.42 | [T]ank | so if i have sox and nice installed and am still getting an error of res_monitor.c:318 ast_monitor_stop: Execute of ( nice -n 19 soxmix.... failed. What else can I look at? |
| 18:11.53 | [T]ank | the recordings are happening. |
| 18:12.46 | thx2000 | Anyone have any experience w/ the Grandstream GXW gateways? |
| 18:13.00 | Falkon_MX-5 | here's a stupid question, how do I dial a country code with my handset? |
| 18:13.26 | [TK]D-Fender | ~gs |
| 18:13.27 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:13.27 | Yourname`` | denon: Checked transcoding, and that too is the same. |
| 18:13.45 | Yourname`` | denon: Is that the only thing though? I checked number of modules loaded, bla bla.. and it's all the same again. |
| 18:13.53 | syle | duh |
| 18:13.58 | denon | I don't understand what you're trying to find out |
| 18:14.05 | denon | call volume is going to change cpu usage |
| 18:14.08 | denon | even the type of call |
| 18:14.14 | thx2000 | jbot: have u used an audiocodes gateway? |
| 18:14.44 | Yourname`` | denon: Yeah, but it's not doing too much. Infact it's almost the same as every other install. |
| 18:14.48 | Yourname`` | Same hardware, etc. |
| 18:15.09 | denon | and cpu is doing what? |
| 18:15.40 | [TK]D-Fender | thx2000: Stop talking to the bot... its too funny for this hour of the day! |
| 18:15.47 | thx2000 | lol |
| 18:17.01 | [TK]D-Fender | thx2000: I've used one once and several others here have as well |
| 18:17.40 | thx2000 | Do they suck any less? |
| 18:17.59 | [TK]D-Fender | thx2000: Substantially. |
| 18:18.02 | Yourname`` | denon: I think I just found it. CPU mhz in the 'offending' installation is 1000, whereas everywhere else it's 2400 mhz. |
| 18:18.18 | [TK]D-Fender | thx2000: AudioCodes IS difficult to learn at the start but once you understand them they are very powerful |
| 18:18.56 | Falkon_MX-5 | All I know is the Grandstream IAXY deals are a hell of a lot cheaper than digium's |
| 18:19.09 | Qwell | ~cheap |
| 18:19.09 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 18:19.20 | Qwell | grandstream == crap |
| 18:19.25 | Falkon_MX-5 | noted |
| 18:19.38 | Yourname`` | denon: Thank you :0 |
| 18:20.07 | thx2000 | I was actually pretty pleased w/ the grandstream gateway until i realized full duplex went out the window w/ "echo cancelation" enabled... |
| 18:20.16 | thx2000 | echo cancellation on a grandstream device = half duplex |
| 18:23.23 | anonymouz666 | grandstream is crap but sells 1 million phones/year |
| 18:23.44 | thx2000 | ...people like crap |
| 18:24.21 | trippsss | anyone work with mediant T38 FXS devices (MP-112)? i'm about to set up a FXS for faxing over IAX2 trunk to another * box and then out a mediant 1000 FXS ports... |
| 18:24.24 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:25.21 | *** join/#asterisk ReD-MaN (i=daemon@172-220.static.golden.net) |
| 18:25.55 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:26.27 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 18:26.30 | trippsss | thx2000: i set up a mediant 1000 with PRI card and FXS ports card as SIP g/w for *. royal pain in the ass to set up but works pretty solid now |
| 18:28.13 | thx2000 | trippsss: I'm just lookin for somethin to work w/ POTS lines. I'd probably stick to some kind of zap device on that type of install |
| 18:28.17 | *** join/#asterisk southtel (n=slester@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 18:28.52 | *** join/#asterisk asdx (n=diego@adsl-151-246.click.com.py) |
| 18:29.27 | southtel | Does anyone know if there are special settings that need to be set to allow a PRI to reveive the dialed/send DID? |
| 18:29.56 | southtel | We just converted from an E&M channel bank to a PRI, and now we're not getting the DID digits. |
| 18:30.05 | [TK]D-Fender | trippsss: Don't think thats going to work.... |
| 18:30.07 | *** part/#asterisk horsesgofaster (n=dcantera@pool-72-82-218-77.cmdnnj.east.verizon.net) |
| 18:30.49 | [TK]D-Fender | trippsss: * only does T.38 PASSTHROUGH. going to IAX... DEAD. There is no T.38 translation and if you think it'd survive regardless, well.... |
| 18:30.53 | [TK]D-Fender | ~wglwat |
| 18:30.54 | jbot | wglwat is probably well, good luck with all that |
| 18:31.29 | [TK]D-Fender | southtel: You must. a call coming in from PRI LANDS on the DID. Thats the exten that gets dialed in * |
| 18:31.42 | trippsss | [TK]D-Fender: how about going from MP-112 T38 straight to mediant on the other end and out that way bypassing * entirely? |
| 18:31.57 | [TK]D-Fender | trippsss: That I presume would work fine. |
| 18:33.07 | *** join/#asterisk Braxus (n=hikaru@66.147.214.164) |
| 18:35.19 | *** join/#asterisk SirWhit (n=sirjames@blk-11-12-158.eastlink.ca) |
| 18:35.52 | southtel | D-Fender: I thought so as well, but when I do a catchall _X. and then noop($EXTEN), I get nothing. |
| 18:35.56 | sigmounte | anyone know a good wireless voip phone who can sync his phonebook over ethernet ? |
| 18:36.13 | [TK]D-Fender | southtel: pastebin it at verbose 10 including your code. |
| 18:36.21 | southtel | Sure. |
| 18:36.24 | [TK]D-Fender | (CLI output that is) |
| 18:37.46 | SirWhit | is this the place to ask an asterisk dimensioning question? |
| 18:39.14 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 18:39.38 | *** join/#asterisk CVirus (n=GoD@82.201.174.66) |
| 18:39.56 | *** join/#asterisk iamthelostboy (n=np@125-236-212-46.adsl.xtra.co.nz) |
| 18:40.10 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:41.13 | [TK]D-Fender | SirWhit: Maybe. |
| 18:41.45 | southtel | D-Fender; It may be because I have immediate=yes in zapata.conf. |
| 18:42.04 | southtel | But when I set it to no, we got a "double ring" when dialing out. |
| 18:42.14 | southtel | (still workng on that pastebin, though) |
| 18:42.41 | [TK]D-Fender | southtel: Pastebin your zaptel & zapata and your dialout string |
| 18:43.33 | iamthelostboy | hi all, im having a problem with a digium tdm400p, under asterisk 1.4.13. Card is configured, and i think everything looks ok under zttool, zapata.conf is configured as per docs, but i get no channels when I do a core show channels, and the phone rings out when i call it. im pretty new to all this so Im probably missing something fairly obvious.. |
| 18:44.11 | iamthelostboy | im only trying to get incoming calls working right now.. will deal with outgoing later |
| 18:47.08 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 18:48.41 | iamthelostboy | i only have a single fxo module at the moment. does it matter what port the module is in on the card? |
| 18:48.59 | [TK]D-Fender | Iampastebin your zapata.conf & dialplan |
| 18:49.40 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:49.40 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:51.44 | southtel | Pastebin -- http://asterisk.pastebin.ca/750728 |
| 18:52.56 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 18:54.12 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 19:00.51 | tzafrir_home | iamthelostboy, zap show channels . Does it hsow anything besides "psedudo"? If not, something is wrong |
| 19:01.13 | tzafrir_home | iamthelostboy, it sure matters what port |
| 19:02.10 | iamthelostboy | asterisk 1.4 doesnt have the zap command |
| 19:02.13 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 19:02.22 | iamthelostboy | not that i can find, making life a little difficult |
| 19:02.43 | iamthelostboy | ive tried the fxo on ports 1,2 and now 4, changing the configs as i go |
| 19:03.47 | *** join/#asterisk saftsack (n=saftsack@pD9E07FB9.dip.t-dialin.net) |
| 19:03.54 | [TK]D-Fender | southtel: Get rid of that immediate=yes |
| 19:04.19 | [TK]D-Fender | iamthelostboy: Ok, I'm guessing you didn't completely recompile * AFTER installing Zaptel... |
| 19:04.25 | southtel | Roger that....any thoughts on the double ring they're getting when they dial out? |
| 19:04.30 | [TK]D-Fender | iamthelostboy: This is required so taht chan_zap can get built. |
| 19:04.38 | [TK]D-Fender | southtel: Show me :) |
| 19:05.10 | [TK]D-Fender | southtel: a 3 port PRI? |
| 19:05.16 | *** join/#asterisk ReD-MaN (i=daemon@172-220.static.golden.net) |
| 19:05.21 | [TK]D-Fender | southtel: (channel that is).... Awefully small.. |
| 19:05.58 | iamthelostboy | when i first installed it all, i did install zaptel first, and I have installed asterisk again since |
| 19:07.04 | [TK]D-Fender | iamthelostboy: do "load chan_zap.so" |
| 19:07.07 | sigmounte | sorry , but how many hard voip phone support remote phonebook ? |
| 19:08.13 | iamthelostboy | i see.. ill recompile... |
| 19:08.20 | Idle | I am trying to get this notifier for asterisk working... is it possible to run a dialplan application from the CLI? |
| 19:08.54 | [TK]D-Fender | Idle: Not so directly, but you can originate a local channelt o an exten that'll do the job. |
| 19:09.05 | irule | where is the zapata and libpri for the latest 1.4.13? |
| 19:09.13 | Idle | :S |
| 19:09.27 | [TK]D-Fender | irule: www.asterisk.rog |
| 19:09.32 | [TK]D-Fender | irule: www.asterisk.org |
| 19:09.33 | blitzrage | .rog! |
| 19:10.29 | tzafrir_home | what versions do you have? |
| 19:10.36 | southtel | D-Fender: Yeah, just 3 ports |
| 19:11.02 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 19:11.26 | southtel | D-Fender: That worked beautifully...no "double ring" like they mentioned earlier. |
| 19:11.30 | CoffeeIV | is there a way to do fax detection on an IAX2 trunk ? |
| 19:11.34 | [TK]D-Fender | southtel: Usually the cost of going PRI is high for that few ports. Got a special deal? |
| 19:12.02 | southtel | Yeah, multi-site deal. |
| 19:12.10 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.244) |
| 19:12.20 | [TK]D-Fender | southtel: Cool |
| 19:12.38 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.68) |
| 19:14.10 | southtel | IIRC, that location is a "flex" PRI, in combo with their data. |
| 19:15.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:16.23 | iamthelostboy | excellent.. looks like its compiled chan_zap.so this time round |
| 19:17.51 | [TK]D-Fender | southtel: AH, IAD like Cisco 2630? |
| 19:18.01 | mocker | ~book |
| 19:18.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 19:18.42 | [TK]D-Fender | iamthelostboy: Good to hear |
| 19:21.08 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:21.45 | iamthelostboy | i get a few errors now when i try to do a load chan_zap.so |
| 19:22.30 | iamthelostboy | unable to specify channel 4: no such device or address |
| 19:23.47 | iamthelostboy | the fxo is on port 4, in zaptel.conf i have fxsks=4 and in zapata.conf i have channel => 4 |
| 19:25.44 | [TK]D-Fender | iamthelostboy: What makes you think its on port 4? |
| 19:25.47 | *** join/#asterisk CRiS1S (n=jeroeng@dsl.intellit.nl) |
| 19:26.26 | iamthelostboy | thats the plug i plugged the module into physically, and thats the led thats on the back of it |
| 19:27.15 | [TK]D-Fender | iamthelostboy: pastebin "dmesg" to be sure, and make sure to reload your cards module & test ztcfg-vvvv |
| 19:29.45 | iamthelostboy | right, im definitely a little new.. and stillnot quite understanding things. after a ztcfg it came up and i could start the channel in asterisk |
| 19:29.46 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 19:29.50 | ZaVoid | morning all |
| 19:29.55 | ZaVoid | has anyone ever done comparisons |
| 19:30.14 | iamthelostboy | i was under the impression that it would be configured if it looked ok in zttool |
| 19:30.18 | ZaVoid | between capacity on asterisk using g.711 vs g.729 (not doing any transcodings) |
| 19:31.20 | Strom_C | ZaVoid: what, just g.729 passthrough? |
| 19:31.30 | ZaVoid | yeah Strom_C |
| 19:31.34 | ZaVoid | and g.711 pass through |
| 19:31.37 | Qwell | the differences would be pretty small |
| 19:31.39 | ZaVoid | or should i say vs |
| 19:32.02 | ZaVoid | yeah but the rtp traffic is signifigantly more right? |
| 19:32.03 | Qwell | your disk I/O will go down if you're reading/writing a lot of files |
| 19:32.07 | CRiS1S | anyone know what might be the problem when i call my voipnumber (from outside) and pick up the phone it drops after a few seconds? |
| 19:32.07 | [TK]D-Fender | ZaVoid: I'd say practically... NONE :p |
| 19:32.12 | ZaVoid | sup fender |
| 19:32.12 | Qwell | your network I/O will too, but... |
| 19:32.15 | ZaVoid | really ya think so? |
| 19:32.26 | ZaVoid | yeah i know network will but tehy are on gigabit ethernet |
| 19:32.26 | Qwell | it's still the same number of packets, unless you use packetization |
| 19:32.35 | [TK]D-Fender | ZaVoid: well.... you're only talking packet size, no processing... |
| 19:32.35 | southtel | D-Fender -- I don't know the specific device at that location, but yeah...we have several locations with that sort of setup...makes PRI/T1 a reasonable propisiton for smaller businesses. |
| 19:32.46 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 19:33.14 | [TK]D-Fender | southtel: I've seen a few odd telcos that seriously push aggressive hybrid products like that. |
| 19:33.30 | [TK]D-Fender | ZaVoid: Getting by.... |
| 19:33.50 | Falkon_MX-5 | [TK]D-Fender: Vitelity gives me 011 to call international. Do you know what the format of the number should be? |
| 19:33.51 | Qwell | ZaVoid: the only real reason to use g729 is for bandwidth savings on a WAN |
| 19:34.07 | ZaVoid | well that and most of my carriers don't want g711 or g723? |
| 19:34.12 | [TK]D-Fender | Falkon_MX-5: Perhaps you should read their FAQ's or ask them... |
| 19:34.25 | Qwell | ZaVoid: you just said it was all gbit, so...that would assume it's all internal traffic, right? |
| 19:34.36 | [TK]D-Fender | ZaVoid: G.711 is heavy, and G.723 way to patent encumbered |
| 19:34.43 | Falkon_MX-5 | [TK]D-Fender: How do you enter a country code on a standard phone? |
| 19:34.58 | Strom_C | Falkon_MX-5: north america? |
| 19:35.04 | Strom_C | 011 + country code + number |
| 19:35.05 | [TK]D-Fender | Falkon_MX-5: Whats YOUR ide of "standard"? You keep asking dangerously generic questions.... |
| 19:35.17 | Strom_C | on mobile phone: +country code... |
| 19:35.26 | [TK]D-Fender | Falkon_MX-5: And even leave me to guess where' you're calling TO. |
| 19:35.37 | [TK]D-Fender | Falkon_MX-5: or FROM. |
| 19:35.38 | Strom_C | Falkon_MX-5: where are you calling from? |
| 19:35.51 | [TK]D-Fender | load chan_clueless.so |
| 19:35.55 | Falkon_MX-5 | Strom_C: and there's the point. Thank you |
| 19:35.58 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 19:36.05 | ZaVoid | yeah Qwell mostly |
| 19:36.20 | Strom_C | Falkon_MX-5: where are you calling from? |
| 19:36.25 | ZaVoid | brb |
| 19:36.26 | Falkon_MX-5 | US |
| 19:36.34 | Strom_C | Falkon_MX-5: wireline phone or mobile phone? |
| 19:36.47 | Wonka | .oO( from a car phone, maybe? *duck* ) |
| 19:36.54 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:36.57 | Strom_C | Wonka: that's still mobile :) |
| 19:37.03 | Falkon_MX-5 | handset. vitelity gave me 011, I just didn't know what the rest of it was |
| 19:37.10 | Wonka | Strom_C: i didn't deny that :) |
| 19:37.11 | Strom_C | "handset"? |
| 19:37.23 | Falkon_MX-5 | not mobile |
| 19:37.27 | Strom_C | wireline then |
| 19:37.31 | Falkon_MX-5 | I know how to do it with mobile |
| 19:37.35 | Strom_C | so yes, 011 followed by the country code and the number |
| 19:37.43 | ReDNeQ | why would asterisk not load any modules on reboot |
| 19:37.44 | Falkon_MX-5 | thanks |
| 19:37.46 | Strom_C | just the same way it's been for at least thirty years now |
| 19:38.03 | Falkon_MX-5 | I've never called another country |
| 19:38.06 | Falkon_MX-5 | except with mobile |
| 19:38.08 | [TK]D-Fender | Strom_C: You and your newfangled telecom! :p |
| 19:38.32 | Falkon_MX-5 | now, I can, because it's 1.5 cents/min to call my brother in china |
| 19:39.45 | *** join/#asterisk Bob_Loblaw (n=michaelc@fwsdo.projectdesign.com) |
| 19:41.29 | iamthelostboy | does anyone in here manage the forums? ive been trying to join up and it wont let me |
| 19:41.48 | *** join/#asterisk ToTo (n=ToTo@host166-92-dynamic.56-82-r.retail.telecomitalia.it) |
| 19:42.15 | iamthelostboy | no, thats not right, it will let me, it just wont tell me my password |
| 19:51.35 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 19:53.11 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 19:53.27 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-235-73.lv.lv.cox.net) |
| 19:55.27 | ussrback | i have problem with res_odbc. i have libtool 1.5.6-4.EL4.1,libtool 1.5.6-4.EL4.1,unixODBC-devel 2.2.9-1 and unixODBC 2.2.9-1 installed. but when i make make menuselect i have XXX before res_odbc. XXX 13. res_odbc |
| 19:55.40 | ussrback | what shoud i install to fix it? |
| 19:59.23 | lawrencej | hey i got a clicking noise going on with my 2400 card can someone help ? |
| 20:03.13 | russellb | lawrencej: please contact support@digium.com |
| 20:03.23 | ussrback | i have problem with res_odbc. i have libtool 1.5.6-4.EL4.1,libtool 1.5.6-4.EL4.1,unixODBC-devel 2.2.9-1 and unixODBC 2.2.9-1 installed. but when i make make menuselect i have XXX before res_odbc. XXX 13. res_odbc |
| 20:03.29 | lawrencej | :( |
| 20:03.59 | russellb | lawrencej: that is their job and they are experts in helping you solve these issues ... |
| 20:04.12 | russellb | ussrback: re-run the configure script |
| 20:04.48 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 20:06.05 | *** part/#asterisk myiagy (n=myiagy@189.34.7.113) |
| 20:06.57 | fakhir | can anyone clue me in to why my did context seems to not want to work for me |
| 20:06.57 | fakhir | http://pastebin.com/m29400ae9 |
| 20:06.57 | fakhir | i want calls from a particular number to go to a voice menu but all other calls to go to a ring group |
| 20:06.57 | fakhir | problem is even when i call from that number the call gets sent to the ring group |
| 20:12.39 | zerohalo | remove the 1 |
| 20:13.14 | ZaVoid | back |
| 20:13.17 | fakhir | so just |
| 20:13.17 | fakhir | [DID_trunk_1] |
| 20:13.18 | fakhir | include = default |
| 20:13.18 | fakhir | exten = _19145140281,1,Goto(voicemenu-0,s,1) |
| 20:13.18 | fakhir | exten = _X.,1,Goto(ringroups-1,s,1) |
| 20:13.20 | fakhir | ? |
| 20:14.02 | ZaVoid | [TK]D-Fender: i was just trying to figure if anyone has run an asterisk with 100+ concurrent g.711 sessions in pass through..i know it can do it with g.729 |
| 20:14.12 | ZaVoid | but sounds like it can handle it |
| 20:15.39 | zerohalo | fakhir: exten = _9145140281,1... or does your DID provider deliver calls w/ a 1 prefix? |
| 20:16.24 | fakhir | not sure ill try without |
| 20:18.57 | fakhir | nope did not work |
| 20:21.53 | *** join/#asterisk blq (n=no@dslb-088-067-016-063.pools.arcor-ip.net) |
| 20:25.09 | mocker | Anyone know if the Polycom 601 automatically includes the power cable, or is it some crap where it's an addon? |
| 20:25.21 | mocker | Buying from voipsupply |
| 20:25.41 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:27.02 | kaldemar | fakhir: exten => _X./<from-number>,1,Goto(... and a line without the /<from-number> |
| 20:27.08 | Alowishus | mocker: I'd call them to be sure, but generally any PoE capable Polycoms are packaged without an AC adapter |
| 20:27.55 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.1) |
| 20:28.32 | coolfreecode | hello =) |
| 20:29.40 | southtel | exit |
| 20:29.44 | southtel | Whoops... |
| 20:29.51 | southtel | This isn't a terminal. |
| 20:32.07 | coolfreecode | what is a adit 600? |
| 20:34.36 | _ShrikE | a channel bank and more |
| 20:34.48 | _ShrikE | depending on the options you have in it |
| 20:35.45 | coolfreecode | i need digium card? |
| 20:38.19 | J4k3 | lots of people make pretty much everything digium makes |
| 20:40.02 | *** join/#asterisk moellerdk (n=chatzill@4806ds1-hl.1.fullrate.dk) |
| 20:41.56 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 20:49.44 | *** join/#asterisk beek (n=klinebl@pool-72-94-31-84.phlapa.fios.verizon.net) |
| 20:55.07 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:59.23 | mcab | Alowishus: no, I think it's just the 320/330s that don't have an AC adapter, everything else should... |
| 21:07.05 | tristanbob | what is the best softphone? commercial and open source |
| 21:07.08 | *** join/#asterisk blq (n=no@dslb-088-065-171-205.pools.arcor-ip.net) |
| 21:08.05 | Shaun2222 | whats with these notices... NOTICE[13389]: cdr.c:434 ast_cdr_free: CDR on channel 'IAX2/307-9' not posted |
| 21:08.12 | brodiem | Guys..anyone know what the best way might be to launch an application triggered by a queue agent answering a call (or when they hangup, whichever is easier)? It would be a help if it could also utilize a previously set variable from the dial plan |
| 21:08.44 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 21:09.18 | brodiem | tristanbob: counterpath.com |
| 21:09.48 | tristanbob | ahhh they make xlite! |
| 21:10.01 | ManxPower | ~doc |
| 21:10.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 21:10.03 | brodiem | indeed |
| 21:10.53 | ussrback | how can i store voicemail user ids and mailboxes in database? |
| 21:14.38 | *** join/#asterisk ^Migs^ (n=Migs@75-162-149-227.slkc.qwest.net) |
| 21:16.26 | [TK]D-Fender | brodiem, Depends how you call your agents. |
| 21:17.03 | ^Migs^ | I'm looking to install Asterisk in a new school. Could the system be used as an overhead intercom in multiple rooms? |
| 21:17.26 | Qwell | ^Migs^: sure |
| 21:17.26 | ^Migs^ | I'm looking through the feature list on http://www.asterisk.org/support/features and I'm not entirely sure what some of those mean |
| 21:17.39 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 21:17.45 | [TK]D-Fender | ^Migs^, go read THE BOOK, and you'll have a much better understanding. |
| 21:17.48 | [TK]D-Fender | ~book |
| 21:17.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:17.48 | Mercestes | I can't create my forum account. :( |
| 21:18.05 | Mercestes | ~docs |
| 21:18.05 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 21:18.06 | Corydon76-dig | The typical platform for Intercom is to use a PA system connected to an ATA device |
| 21:18.44 | ussrback | [TK]D-Fender: do u have any link for storing voicemail users in database |
| 21:19.06 | Mercestes | ussrback: google asterisk rta |
| 21:19.17 | Mercestes | follow the links at the bottom. |
| 21:19.30 | Shaun2222 | [TK]D-Fender: i get NOTICES all teh time about CDR not posted... these normal or do i need to do somthing |
| 21:20.06 | ^Migs^ | cool, thanks |
| 21:20.16 | [TK]D-Fender | I think a whole bunch of you need to stop directing wuestions to people not already talking specifically about what you're working on. |
| 21:20.22 | Mercestes | Shaun2222, on broken calls you will get a cdr not posted which is a normal message. Are you getting any cdrs at all? |
| 21:20.49 | Mercestes | [TK]D-Fender, how do I match an extension if it doesn't exist as a phone? |
| 21:20.52 | Mercestes | >.> |
| 21:20.56 | Shaun2222 | ya cdr works, but my dial() dials a bunch of phones waiting for one of them to respond. |
| 21:21.09 | Shaun2222 | i get the not posted on the ones that dont answer from the looks of it |
| 21:21.23 | Mercestes | Shaun2222, I'm not 100% on the new cdr messaging, but it could be generated by the phones *not* answering the call. |
| 21:21.52 | Mercestes | IT's a new level of verbosity to annoy [TK]D-Fender with questions about it. |
| 21:22.19 | Mercestes | same with the "starting simple switch on" and "ntp skew" messages. |
| 21:22.39 | ^Migs^ | wtf, you want me to read that whole book? |
| 21:22.57 | Mercestes | ^Migs^, alternatively, you could hire a consultant to do it for you. |
| 21:23.13 | Shaun2222 | lol |
| 21:24.18 | Mercestes | Gah, and the crowd goes silent. |
| 21:24.36 | ^Migs^ | I'm silent because I'm reading the book :) |
| 21:24.36 | brodiem | [TK]D-Fender: well ultimately I'll need two solutions - currently using queues and the agent channel driver. The other doesn't use the agnet channel driver, which I probably won't have a problem doing what I need there |
| 21:25.08 | Mercestes | ^Migs^, excellent! :) |
| 21:25.38 | Darthclue | ^Migs^, you'll be much happier if you understand what you are working with. The book should help you to understand it. And if you still choose to hire a consultant, you'll have a better understanding of what he/she/it says. |
| 21:26.29 | Mercestes | "hiring consultants" is a mythological concept. |
| 21:27.13 | Darthclue | Which part of that is mythological? |
| 21:28.13 | Mercestes | no one hires consultants. |
| 21:28.47 | Mercestes | It is more likely they'll hire a "sysadmin" to maintain the web, firewall, ftp, email, security, network and lightbulbs and then dump the phone system on you as well. |
| 21:29.22 | Mercestes | but that whole "pay asterisk people to do the smart work" for me is only a theory, never proven into law. |
| 21:29.23 | file | "well, the phone system runs on Linux and since you know Linux you are an expert on this!" |
| 21:29.30 | Mercestes | file: exactly. |
| 21:29.57 | Mercestes | Closest I got was being promised to get paid after I shutdown the phone system of this guy that promised to pay me to begin with. |
| 21:30.14 | HarryR | Mercestes, more like, since you're used to Linux and can easily learn new programs why not learn Asterisk too? |
| 21:30.25 | HarryR | imo sysadmins have some of the widest ranges of experience in the industry |
| 21:30.32 | Mercestes | HarryR, Since it runs in linux then...it's part of linux so you should know it. |
| 21:30.55 | Mercestes | Agreed. |
| 21:30.58 | HarryR | Mercestes, not really, because you're a sysadmin your mindset allows you to adapt to new stuff easier |
| 21:31.16 | Mercestes | Well, we should specify, LINUX sysadmins have the widest range of experiences ever. |
| 21:31.20 | HarryR | imo some of the senior level sysadmins are some of the most flexible people I know |
| 21:31.27 | Mercestes | Windows sysadmins get to specialize in one program. IIS, Exchange, Active Directory. |
| 21:31.34 | HarryR | really their developers, but rarely have to delve into code |
| 21:31.48 | Mercestes | Yea...flexible....there was this one chick...she could so touch the back of her head with her ankle. |
| 21:31.57 | HarryR | uh |
| 21:32.00 | HarryR | I can do that! |
| 21:32.04 | Mercestes | really? |
| 21:32.06 | HarryR | yes |
| 21:32.13 | Mercestes | oh... |
| 21:32.15 | *** join/#asterisk destructure (n=kwatz@66.193.229.254) |
| 21:32.21 | Mercestes | sorry..I can't talk to you...I'm banned from that other channel. =/ |
| 21:32.25 | Darthclue | and troubleshoot the pbx, web server, and database server at the same time? |
| 21:32.46 | Mercestes | Darthclue, with her toes even. |
| 21:32.51 | HarryR | Darthclue, what... our guys do that all the time |
| 21:34.02 | Shaun2222 | ok in 1.4 the default format for music on hold is wav now... |
| 21:34.29 | Shaun2222 | is there a specific format the wav file should be when converting the mp3/ |
| 21:35.18 | Mercestes | Shaun2222, google asterisk mp3 to gsm |
| 21:35.43 | Mercestes | It has the bit rates for wav, mp3, gsm, and a few other sox magic goodies. |
| 21:35.53 | Shaun2222 | also by using the default wav format now and not having to use format_mp3 are my customers going to stop hearing the firt bit of a song everytime they are put on hold.. |
| 21:35.56 | Shaun2222 | thats really annoying |
| 21:36.40 | Shaun2222 | i would rather the music just play constantly and whenever a person is placed on hold they just hear whats playing... |
| 21:36.49 | coolfreecode | who knows how to work a bank channel ADIT 600 |
| 21:37.22 | ^Migs^ | Another question. Can't seem to find the answer; maybe I just missed it. Do you actually have to the Digium Interface card to be able to use Asterisk? |
| 21:37.34 | Qwell | ^Migs^: nope |
| 21:37.39 | *** part/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
| 21:37.55 | Mercestes | ^Migs^, I run asterisk on my wrt54gl router. |
| 21:38.20 | coolfreecode | ADIT 600 + Asterisk help |
| 21:38.20 | Mercestes | so....no |
| 21:38.33 | ^Migs^ | do you need ANY special equipment besides a standard server? |
| 21:38.39 | Qwell | ^Migs^: phones |
| 21:38.42 | Mercestes | coolfreecode, I'm pretty sure it's a "channel bank" and not a bank channel |
| 21:38.47 | Mercestes | you don't even really need to have phones... |
| 21:38.48 | Qwell | maybe a network |
| 21:38.52 | Mercestes | .....or a standard server.. |
| 21:38.52 | Qwell | Mercestes: yeah.. |
| 21:38.55 | ^Migs^ | well, obviously |
| 21:39.06 | Mercestes | What part of "I run asterisk on my linksys router" evaded you? |
| 21:39.31 | Mercestes | You just need linux and enough space to hold it and viola... |
| 21:39.33 | ^Migs^ | I recognized that as a linksys model, but I'm not sure everyone knew that |
| 21:40.02 | Darthclue | If want to 'use' the software, you'll need a network, some type of device capable of running the software, and some type of phone. |
| 21:40.24 | Darthclue | s/If want/If you want |
| 21:40.31 | Mercestes | Which basicaly means, a PC, a nice roll over cable, and a couple free soft phones. |
| 21:40.54 | Mercestes | But that's the really geeky equivalent to two tin cans and some string. |
| 21:40.59 | Darthclue | or a wrt54gl :) |
| 21:41.20 | Mercestes | Darthclue, lol, yea, PC and network bundled into one little.....piece of trash. |
| 21:41.31 | Mercestes | I think I can pass maybe one whole call on it before I smoke it. |
| 21:42.01 | Mercestes | It spends more of it's time keeping my PC from getting hacked.... |
| 21:42.08 | Darthclue | it's still usable. may be a one time use only, but still usable. |
| 21:42.10 | Mercestes | ...which reminds me..I need to swap it out for an OpenBSD box. |
| 21:43.07 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 21:44.32 | Mercestes | ^Migs^, realistically, you will need some form of interface to the "world" which is where the real cost starts happenign. |
| 21:44.40 | ^Migs^ | obviouslly |
| 21:45.00 | ^Migs^ | I was thinking that the traditional telephone line would just run directly to the server room, then Asterisk takes over from there |
| 21:45.07 | Mercestes | If you wish to terminate VoIP you will need an Inet connection and an ITSP. If you wish to terminate analog, then an Fxo card, adn if you wish to terminate PRI then you need a T1 card. |
| 21:45.13 | ^Migs^ | then we get VoIP phones and VoIP intercoms, and anything else we want |
| 21:45.24 | Mercestes | I'm a little confused. |
| 21:45.29 | ^Migs^ | okay, that made no sense to me. I'll read the book first. |
| 21:45.50 | Mercestes | "telephone line" appeared to be stated singular, but "phones" and "intercoms" seemed to be plural. |
| 21:45.57 | *** part/#asterisk Bob_LobLaw (n=michaelc@fwsdo.projectdesign.com) |
| 21:46.09 | Mercestes | Am I to understand you have 1 phone line and 2 or more phones? |
| 21:46.50 | ^Migs^ | yeah, I figured I could set it up so everyone would just have an extension, then the phones and intercoms could be plugged into network jacks in the wall, without any traditional phone lines running through the building except to the server room |
| 21:47.03 | ^Migs^ | the school isn't even built yet |
| 21:47.13 | ^Migs^ | so right now we have nothing. We're trying to plan it. |
| 21:47.29 | Mercestes | I should point out that a "phone line" singular will handle one phone call, singular. |
| 21:47.54 | Mercestes | and to handle plural phone calls you will need phone lines plural or atleast 1 T1-PRI. |
| 21:48.02 | Shaun2222 | sweet... my music on hold dir should be a few gb's now... :/ |
| 21:48.04 | ^Migs^ | is Asterisk actually going to be cost-effective? |
| 21:48.14 | Mercestes | In a word, absolutely. |
| 21:48.19 | Shaun2222 | playing a wav is probably less taxing on the cpu i suppose. |
| 21:48.42 | Mercestes | but, it doesn't magically make a single phone line handle multiple calls. |
| 21:48.47 | ^Migs^ | ok |
| 21:49.09 | Mercestes | anything more than 4 simultanious calls probably warrants a PRI, which handles 23 at once. |
| 21:50.16 | Mercestes | anything from 1 -4 simultanious calls can be handled by individual analog lines (telephone lines) plugged into an FXO card. |
| 21:50.52 | Mercestes | Your "cost effectiveness" is realized in that regardless of how many users you add or what features you setup, you don't have to pay a per user per year licensing fee to anyone, unless....you want to. |
| 21:51.49 | Mercestes | "cheap" is in relative to a traditional PBX, not your home telephone service, however. It's not more cost effective than an answering machine if you are using it for an answering machine. |
| 21:52.15 | Mercestes | but some people try... |
| 21:52.22 | Strom_C | ...and fail miserably |
| 21:52.29 | Mercestes | Exactly. |
| 21:57.06 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 21:57.52 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-81-172.bstnma.east.verizon.net) |
| 22:02.27 | ussrback | anybody has configured voicemail realtime? |
| 22:02.56 | Mercestes | It's buggy. |
| 22:03.03 | Mercestes | that's the best thign I have to say about it. |
| 22:03.06 | coolfreecode | anybody has configured a ADIT 600? |
| 22:03.07 | Mercestes | but i've done it before. |
| 22:03.24 | Mercestes | I loathe to say that because that invariably means I'll get bombarded with questions on how to make it work. =/ |
| 22:03.27 | ussrback | Mercestes: what do u mean? |
| 22:03.50 | Mercestes | ussrback, Whcih part would you like me to elaborate on? |
| 22:05.03 | ussrback | ok i have database where user box id and contexts are stored. i want to use this data for asterisk |
| 22:05.09 | Mercestes | coolfreecode, from here on out, you should say "I am trying to interface an ADIT 600 with Asterisk using the MGCP protocol and it's not working. Can anyone help me?" |
| 22:05.34 | coolfreecode | I am trying to interface an ADIT 600 with Asterisk using the MGCP protocol and it's not working. Can anyone help me? |
| 22:05.40 | Mercestes | :) |
| 22:05.56 | *** join/#asterisk blq (i=no@dslb-088-064-144-148.pools.arcor-ip.net) |
| 22:10.50 | coolfreecode | I am trying to interface an ADIT 600 with Asterisk using the MGCP protocol and it's not working. Can anyone help me? |
| 22:11.08 | ussrback | [Oct 26 15:04:13] ERROR[31788]: res_config_odbc.c:130 realtime_odbc: No database handle available with the name of 'asterisktemp' (check res_odbc.conf) |
| 22:11.08 | ussrback | [Oct 26 15:04:13] WARNING[31788]: app_voicemail.c:2806 leave_voicemail: No entry in voicemail config file for '000001 |
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| 22:11.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:11.44 | ussrback | how can i fix this? |
| 22:14.27 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:21.33 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:26.09 | Shaun2222 | other than a stream is there any music on hold player that will just constantly play and not always play the beggining of the same song? |
| 22:26.26 | HarryR | Shaun2222, shoutcast? |
| 22:26.57 | *** join/#asterisk NoCarrier (n=John@unaffiliated/badpacket) |
| 22:27.48 | Shaun2222 | other than stream...... |
| 22:28.23 | Qwell | Shaun2222: what you described is pretty much the definition of a stream |
| 22:28.37 | Qwell | so maybe you mean non-networked stream? |
| 22:28.50 | Shaun2222 | ya. |
| 22:29.01 | Qwell | then...shoutcast |
| 22:29.22 | Shaun2222 | i suppose i could install a local shoutcast server.. |
| 22:29.34 | Shaun2222 | then what use streamplayer to play it? |
| 22:29.39 | Qwell | sure |
| 22:30.00 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:32.15 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:33.11 | *** join/#asterisk Farris1 (n=jrush@gateway.wiquest.com) |
| 22:33.13 | Farris1 | Is there a way to disable the name record confirmation when you enable join/leave announce on a conference? |
| 22:34.05 | Shaun2222 | whats the ideal way to play a stream, would be nice to have it pull and use a playlist.pls so it gets a list of servers.. |
| 22:42.33 | Shaun2222 | hmm, streamplayer doesnt seam to be able to connect to remote shoutcast servers, do some net stream filter based on "client"? |
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| 22:48.02 | chancleta | Good night from Spain... |
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| 22:49.21 | k31th | Evening |
| 22:50.09 | Maliuta | mo'ning |
| 22:51.40 | *** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
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| 22:57.59 | coolfreecode | I am trying to interface an ADIT 600 with Asterisk using the MGCP protocol and it's not working. Can anyone help me? |
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| 23:08.08 | Rawplayer | anyone in here integrated asterisk into active directory(ldap)? |
| 23:09.45 | TJNII | Rawplayer: No, but if you do I'd be interested in hearing how you did it |
| 23:10.11 | TJNII | I played with ldap a few years ago and I found I had to roll my own interface code most the time |
| 23:12.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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| 23:13.04 | docelmo | Rawplayer why would you want to use AD? |
| 23:13.31 | docelmo | Microsoft's implementation of LDAP sucks and is a pain in the ass to make work correctly. |
| 23:14.09 | Rawplayer | quiet a challenge then ;p |
| 23:14.40 | docelmo | Yes.. I find it simpler to just integrate windows via TAPI or some click to call application to asterisk |
| 23:19.07 | k31th | docelmo: what c2c app you use? |
| 23:19.17 | k31th | I want to put some thing on our intranet. |
| 23:27.37 | docelmo | I wrote my own |
| 23:28.49 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 23:28.57 | docelmo | The way my app works is to accept TAPI requests or URL links to perform a call.. it makes the desk phone ring then once its picked up it will start ringing on the other end |
| 23:29.01 | jjshoe | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+QueueStatus thoughts on the same thing with the ability to specify a specific queue? |
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| 23:34.56 | *** mode/#asterisk [+o d3wayne] by ChanServ |
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| 23:44.06 | k31th | docelmo: kool, php ? |
| 23:45.54 | *** join/#asterisk rnovotny22 (n=ro085181@h460dca42.area2.spcsdns.net) |
| 23:53.25 | Shaun2222 | great stream works with madplay but now it's rediculisly loud |
| 23:55.42 | *** join/#asterisk jgomo3 (n=jgomez@200.72.223.2) |
| 23:55.44 | k31th | exten => _9.,1,Dial(IAX2/office:blahh@iax-out.netzensolutions.com/${EXTEN:1}) should work fine right? if i dial 9 then the number that should work fine? |
| 23:56.04 | jgomo3 | does some body had programed with jiaxclient? |
| 23:56.19 | k31th | if i try and dial it i get rejected because extension not found. |
| 23:56.56 | remmo | we have implemented jiaxclient |
| 23:57.15 | jgomo3 | why not hard code the 9 extent? you know: exten => 9, ......... |
| 23:57.50 | k31th | jgomo3: tbh im a total n00b at this in the past i have always useda gui to config asterisk... |