| 00:00.12 | *** join/#asterisk weinerk (n=irc@bzq-79-177-57-154.red.bezeqint.net) |
| 00:00.29 | drmessano | Why would ebay be free? |
| 00:02.07 | Joel | ebay sucks for many reasons |
| 00:02.37 | weinerk | Please help - sip show peers = gives provider UNREACHABLE |
| 00:02.37 | weinerk | Some more details here: |
| 00:02.37 | weinerk | http://pastebin.com/m1aeaed56 |
| 00:07.40 | weinerk | ? |
| 00:10.24 | raden_work | drmessano, it should be free to list |
| 00:10.41 | raden_work | drmessano, can u imagine the amount a revenue you could produce if it was free to list |
| 00:11.02 | raden_work | drmessano, charge a reasonable 3% on completed listing ? |
| 00:11.07 | drmessano | How would THEY make money? |
| 00:11.12 | drmessano | .... |
| 00:11.22 | raden_work | drmessano, you have any idea how much ebay makes |
| 00:11.31 | drmessano | Of course I do |
| 00:11.44 | drmessano | and you're suggesting they stop charging |
| 00:11.54 | Joel | he's not suggesting they stop charging |
| 00:11.59 | raden_work | $367.2 million |
| 00:12.01 | Joel | he's saying they should only take a percent of a valid sale |
| 00:12.06 | raden_work | correct |
| 00:12.13 | Joel | raden_work, gross, what's the net? |
| 00:12.21 | raden_work | thats net |
| 00:12.24 | drmessano | Costs them money to allow you to list |
| 00:12.34 | Joel | raden_work, I highly doubt that's net. |
| 00:12.37 | raden_work | drmessano, they would make more though |
| 00:12.43 | raden_work | Joel, trust me its net !!!!!!!!!! |
| 00:12.54 | raden_work | Joel, I work with many investors |
| 00:12.57 | Joel | raden_work, do your extra exclamation marks serve as proof? |
| 00:13.08 | raden_work | eBay pulled in sales of $2.04bn, a 6.6 per cent drop from the same quarter last year |
| 00:13.18 | raden_work | there is there gross from last financial report i got |
| 00:13.25 | raden_work | 2.04 billion for Q4 last year |
| 00:13.41 | raden_work | actually sorry that Q2 reports |
| 00:14.06 | raden_work | Joel, not trying to argue just trying to make a point |
| 00:14.23 | Joel | raden_work, try in #ebaysucks maybe? |
| 00:14.38 | raden_work | sorry |
| 00:14.48 | raden_work | just seems unfair to alot of people |
| 00:15.07 | raden_work | there are things id like to sell but fear them not selling so i dont list im sure many people feel the same way |
| 00:15.19 | drmessano | Yeah, why should companies make money |
| 00:15.19 | drmessano | Those bastards |
| 00:15.19 | raden_work | now ebay owns skype as well there rates are going up |
| 00:15.25 | drmessano | No they dont |
| 00:15.59 | drmessano | They just sold all but a small percentage.. You don't get out much, do you? |
| 00:17.30 | raden_work | they aqquired all shares of skype for 2.5 billion in 2005 |
| 00:17.39 | raden_work | before that skype was private |
| 00:18.16 | drmessano | http://news.cnet.com/8301-1035_3-10322833-94.html |
| 00:18.21 | drmessano | Seriously.. |
| 00:18.37 | drmessano | Stop while you're ahead |
| 00:18.59 | Joel | he obviously sucks at the net |
| 00:19.06 | raden_work | drmessano, yes that was news sir |
| 00:19.23 | raden_work | Joel, just alot of other things todo dude |
| 00:19.38 | raden_work | dont keep up much with news these days watch my investments and thats it |
| 00:19.40 | drmessano | But yet he knows how much ebay netted last year |
| 00:19.56 | drmessano | Doesnt know they sold Skype, which was on the front page of the WSJ |
| 00:20.00 | raden_work | drmessano, yes cause i have financial consultants :P |
| 00:20.00 | Joel | drmessano, supposedly, no link to real evidence of course. |
| 00:20.12 | Joel | anyways stnr. |
| 00:20.13 | drmessano | Joel: Exclamation points? |
| 00:20.17 | raden_work | Joel, get a broker |
| 00:20.19 | Joel | drmessano, oh right, I forgot. |
| 00:20.37 | drmessano | raden_work: Get a browser |
| 00:20.40 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:20.48 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:20.50 | raden_work | why everyone have to be so mean all the time ? |
| 00:21.14 | Joel | I'm guessing it's just because you're an easy target? |
| 00:22.03 | raden_work | yea i guess so |
| 00:22.18 | carrar | Are you gonna cry? |
| 00:23.44 | raden_work | wow |
| 00:24.22 | raden_work | funny in real life no one would ever talk to me like that its funny the safety blanket a browser window gives |
| 00:24.33 | drmessano | Im not using a browser |
| 00:24.42 | raden_work | or a chat windows |
| 00:24.45 | raden_work | or a console |
| 00:24.56 | raden_work | have a good night guys |
| 00:25.27 | carrar | I think there is a highway sign with his name on it! |
| 00:25.39 | raden_work | lol |
| 00:26.07 | *** part/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 00:26.18 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:36.06 | *** join/#asterisk came0 (n=came0@93.147.121.70.cfl.res.rr.com) |
| 00:36.29 | el_critter | Hi, calls made from my polycom330 hangs at 50sec from answer. Here is a pastebin of 330's app.log if someone is kind to see... this is driving me crazy. http://pastebin.com/f3bf4e964 |
| 00:38.20 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 00:45.44 | *** join/#asterisk chendy (n=chatzill@58.250.11.251) |
| 00:48.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:56.18 | jblack | does anyone have a spare google voice invites? |
| 00:56.59 | Katty | no, but i have some swiss cheese |
| 00:57.04 | Nivex | mmm cheese |
| 00:57.08 | jblack | I like swiss cheese. |
| 00:57.13 | jblack | Sorry I missed you yesterday, Katty. |
| 00:58.26 | Katty | oh it's alright (= |
| 01:01.30 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 01:06.21 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 01:22.45 | *** part/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 01:23.21 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 01:24.16 | *** join/#asterisk Chesther (n=cam2@63-255-103-4.ip.mcleodusa.net) |
| 01:28.23 | mchou | umm....what's the difference between Zoiper Communicator and Zoiper 2.0? |
| 01:44.58 | *** join/#asterisk OrNix (n=ornix@91.151.249.47) |
| 01:47.11 | *** join/#asterisk Methose (n=Methose@c-66-56-6-74.hsd1.ga.comcast.net) |
| 01:50.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:50.20 | Methose | is anyone familiar with using sipgate to provide a DID to an asterisk system? |
| 01:51.15 | Methose | or is the general practice to use gizmo5? |
| 01:52.09 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 01:52.25 | sircolin | I have gizmo5 working here |
| 01:55.02 | *** join/#asterisk jtodd (i=hq84l3vt@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 01:55.02 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:55.26 | Methose | is it truly free and pretty reliable? |
| 01:55.30 | *** join/#asterisk coolthreads (n=coolthre@203.97.238.71) |
| 01:57.47 | sircolin | gizmo5 I find to be 98% reliable I use them only for sip 2 skype as there are cheaper options for outgoing calls and I pay'd $20 per year for that |
| 02:02.24 | Methose | that's what I'm setting up right now |
| 02:03.00 | Methose | I just configured sistosip with skype gateway and now I'm trying my hand a gtalk gvoice |
| 02:04.06 | Methose | waht's there to configure with gizmo5? is it pointing the number to the sip addy on the asterisk server through the gizmo web gui? |
| 02:04.39 | sircolin | yer it's very simple |
| 02:05.31 | sircolin | forward all to gizmo5 then forward gimzo5 to ringgroup@yourasterisk.com |
| 02:06.03 | sircolin | I think it's the same for gvoice too |
| 02:07.05 | Methose | yeah I must be missing something on the gizmo config, not seeing where to enter the extention@myasterisk.com |
| 02:07.21 | sircolin | 1 sec i'l check |
| 02:08.28 | Methose | psshh I'm blind, sorry I see it now 8-( |
| 02:08.46 | sircolin | cool |
| 02:08.59 | Methose | thanks tho |
| 02:22.00 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 02:29.06 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 02:29.29 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 02:32.20 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 02:51.45 | *** join/#asterisk voxter (n=voxter@32.176.35.124) |
| 02:54.54 | ChannelZ | Has anyone here gotten Google talk to work under asterisk? I have it running and logging in but don't get audio and haven't figured out why (although the gTalk client doesn't seem to have a dialpad so I guess I can only make it go directly to an extension) |
| 03:04.42 | *** join/#asterisk maxagaz (n=maxagaz@124.205.74.34) |
| 03:13.49 | *** join/#asterisk geneticx (n=geneticx@adsl-10-114-216.mia.bellsouth.net) |
| 03:16.58 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 03:17.47 | *** join/#asterisk coolthreads (n=coolthre@203.97.238.71) |
| 03:19.10 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) |
| 03:19.14 | *** part/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) |
| 03:20.44 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) |
| 03:26.58 | Damin | iCEBrkr is no l433t |
| 03:27.50 | iCEBrkr | Go drink |
| 03:27.51 | iCEBrkr | Drunky |
| 03:27.53 | iCEBrkr | :P |
| 03:28.13 | Damin | Sittin her w/ dayton, having a beer. |
| 03:28.24 | iCEBrkr | I spent the past 4hr migrating data |
| 03:28.33 | iCEBrkr | Updating software |
| 03:28.35 | voxter | I spent the past 4hrs migrating your mom |
| 03:28.36 | iCEBrkr | Reconfiguring crap |
| 03:28.38 | voxter | and upgrading your mom |
| 03:28.40 | voxter | and reconfiguring your mom |
| 03:28.45 | iCEBrkr | voxter: Sweet!! |
| 03:28.48 | voxter | SEVEN WAYS TO SUNDAY |
| 03:28.54 | Damin | I upgraded your mom last night. |
| 03:29.09 | iCEBrkr | voxter: She is cougar status... |
| 03:29.52 | iCEBrkr | Is still have shit to move over.. bleh.. Dexter time. |
| 03:29.55 | Damin | http://www.boingboing.net/2009/10/12/the-woman-who-cant-s.html |
| 03:31.22 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 03:31.33 | Damin | Test...i...cles |
| 03:31.55 | kn0x | okay heres the scenario, perhaps the is a -dev question... but I need to build asterisk with meetme support from inside a VSERVER |
| 03:32.06 | kn0x | I have dahdi running on the host machine |
| 03:32.08 | *** join/#asterisk ChannelZ (i=channelz@63.224.70.8) |
| 03:32.24 | kn0x | and have the /dev/dahdi.... devices available inside the guest. |
| 03:33.41 | kn0x | I followed some old instructions to trick asterisk with zaptel by copying libetonezone, zaptel.h and some others to the guest, but i couldnt find a dahdi.h and it configure isnt recognizing dahdi support |
| 03:40.11 | *** join/#asterisk bleger (n=ladorval@68.45.58.209) |
| 03:45.04 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 03:45.44 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 03:49.23 | *** join/#asterisk scalex000 (n=chatzill@76puntacana02.codetel.net.do) |
| 03:50.27 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 03:57.05 | *** join/#asterisk d00gster (n=doughant@94.97.29.222) |
| 04:18.27 | *** join/#asterisk soman (n=somnath@115.119.41.110) |
| 04:33.30 | *** join/#asterisk broken-iPod (i=62f38cac@gateway/web/freenode/x-byyvcivtttmjdzqw) |
| 04:34.26 | broken-iPod | Can I make Asterisk take the audio from a command line program (like mpg123 or espeak) and play it over a channel by running a command? |
| 04:35.13 | broken-iPod | Instead of invoking MP3Player(), can I use the audio from invoking "mpg123 /home/me/blah.mp3" |
| 04:38.28 | *** join/#asterisk dysinger (n=tim@75.85.135.191) |
| 04:41.30 | *** join/#asterisk Tim_Toady (n=moi@adsl45-72.kln.forthnet.gr) |
| 04:46.47 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-oyjtvyxnhfkzzrpy) |
| 04:57.06 | *** join/#asterisk e4 (n=e4@76.84.81.72) |
| 04:59.37 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 05:05.31 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 05:06.17 | *** join/#asterisk werdan7_ (n=w7@freenode/staff/wikimedia.werdan7) |
| 05:17.28 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:18.07 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 05:23.22 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:51.26 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 05:52.08 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 05:56.15 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.103) |
| 05:57.44 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 06:01.52 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 06:22.14 | *** join/#asterisk voxter (n=voxter@32.176.35.124) |
| 06:31.24 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 06:32.20 | *** join/#asterisk xrmx__ (n=rm@host27-26-dynamic.24-79-r.retail.telecomitalia.it) |
| 06:38.09 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 06:40.21 | *** join/#asterisk JunK-Y (n=junky@96.20.69.206) |
| 06:41.41 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:42.00 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 06:48.01 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:49.32 | *** join/#asterisk fiddur (n=fiddur@192.121.104.118) |
| 06:56.57 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 06:59.19 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 07:03.23 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 07:03.23 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:12.00 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:13.22 | *** join/#asterisk jmworx (n=jeval@mail.octasic.com) |
| 07:13.29 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:a1a7:afe7:72fa:32a) |
| 07:18.26 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:18.43 | *** join/#asterisk cuco (n=cuco@local.xorcom.com) |
| 07:19.07 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 07:35.12 | *** join/#asterisk smtx (n=smtx@80.153.133.87) |
| 07:39.24 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 07:43.40 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:44.37 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:48.23 | *** join/#asterisk Da-Geek (n=Da-Geek@garyan.gotadsl.co.uk) |
| 07:50.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:51.22 | *** join/#asterisk e4 (n=e4@cpe-76-84-81-72.neb.res.rr.com) |
| 07:52.04 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:52.24 | *** join/#asterisk CunningPike (n=CunningP@24.87.13.173) |
| 08:28.35 | *** join/#asterisk TJNII (n=TJNII@207.189.199.62) |
| 08:37.05 | *** join/#asterisk SPY```` (n=SPY@41.153.202.237) |
| 08:41.41 | *** join/#asterisk szallol (n=szallol@89.34.72.178) |
| 08:43.36 | *** join/#asterisk nikolam (n=nikola@dynamic-78-30-181-226.adsl.eunet.rs) |
| 08:44.02 | nikolam | hi, is it true that sterisk can`t work on solaris/opensolaris with SPARC? |
| 08:46.28 | SPY```` | yes correct |
| 08:53.22 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 08:55.27 | *** join/#asterisk maragani (n=rahul909@122.169.134.247) |
| 08:57.42 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 08:58.12 | *** join/#asterisk jgoo (n=secc@ppp58-8.adsl.forthnet.gr) |
| 08:58.15 | jgoo | I have a DIALSTATUS=CHANUNAVAIL |
| 08:58.21 | jgoo | misdn |
| 08:58.30 | jgoo | \Why does it get into this state? I tried misdn reload, and it didn't fix it |
| 08:58.48 | *** join/#asterisk TJNII (n=TJNII@207.189.199.62) |
| 08:59.19 | jgoo | misdn, port 1 is blocked, why? |
| 09:02.06 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 09:02.19 | Polysics | hello |
| 09:02.34 | Polysics | anyone is aware of a working flash-based phone implementation |
| 09:02.40 | Polysics | ? |
| 09:02.58 | Polysics | i've been trying to install red5phone but apparently half the project has disappeared |
| 09:03.47 | jgoo | I have a call, any CID any DID going to extension 100, when I call in, asterisk gets the call and does DIALSTATUS=CHANUNAVAIL |
| 09:05.24 | *** join/#asterisk Kchehab (n=kchehab@212.98.141.199) |
| 09:05.32 | Kchehab | hi all |
| 09:06.08 | Kchehab | i am trying to load Asterisk RealTime Sip |
| 09:08.50 | xrmx__ | does anybody have experience with call being misteriously dropped with asterisk and siemens hi path? |
| 09:09.58 | jgoo | This setup works, it just stopped today, earlier, could this be the phone freezing? There are no troubleshooting docs, just not getting CHANUNAVAIL - how can I reset it? |
| 09:09.59 | Kchehab | i add sippeers => mysql,general,sip_buddies to extconfig.conf |
| 09:10.20 | jgoo | xrmx__, what can you do with a seimens hi path? (and how much you pay for it) |
| 09:10.21 | Kchehab | and add sip_buddies to asterisk database |
| 09:10.40 | xrmx__ | jgoo, i have the asterisk box a customer have the siemens hi path :) |
| 09:11.38 | jgoo | xrmx__, aaah, I see. I have a customer whose idiot cousin made him by two fucking hi paths, one for his office, one for this home. He has one phone in his office, and one at home. He uses his mobile phones for all calls... wondering how much he wasted |
| 09:12.04 | jgoo | what causes chan unavailable? I just reset the damn phone that takes the call |
| 09:12.18 | jgoo | this is a working setup, that suddenly gives chan unavailable |
| 09:13.03 | jgoo | fucking phone are back up, I am seeing Extension Changed 119[ext-local] new state Idle for Notify User 100 |
| 09:13.16 | jgoo | what would that mean (I have 0 confidence I will fing documentation for this) |
| 09:13.30 | Gugge | jgoo, you could try to pastebin a verbose output while you make a call. then we could see whats wrong without guessing |
| 09:13.33 | jgoo | Why am I getting a new idle state for a user? and why is that happening when the system is coming back up |
| 09:14.01 | jgoo | and what would cause the system to not work, give CHANUNAVAIL AND have that symptom when it comes back - IE HOW can i stop this happening again? |
| 09:14.17 | Gugge | 1000 things, im not gonna guess |
| 09:16.23 | *** join/#asterisk jmkgreen (n=chatzill@84.12.140.56) |
| 09:17.33 | *** join/#asterisk Moz (n=me@81.179.238.144) |
| 09:17.36 | Moz | hi all |
| 09:18.03 | Kchehab | kaldemar hiii |
| 09:18.07 | jmkgreen | Is anyone here familiar with the message "[Channel]/[Peer] stopped sounds"? We're using IAX2 connecting to Gradwell and while the phone rings the audio is muted in one direction hence my query. |
| 09:18.09 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 09:18.33 | jmkgreen | This box was previously working fine - it has since been reinstalled and the old configs copied across |
| 09:18.34 | jgoo | Gugge, true, ok here is a pastebin |
| 09:18.34 | jgoo | http://pastebin.me/9ed3aab4e6f892f6c2c3e3c173290803 |
| 09:18.50 | Moz | does anyone know of any software that I can use with Asterisk to make it into an automated dialer? I want it to call specific telephone numbers with a recorded message for a set amount of time then hang up. Also known as "automated calling equipment" I think? |
| 09:18.52 | jmkgreen | we're puzzelled and Google doesn't provide anything but other people asking the same question :( |
| 09:19.22 | jgoo | The only thing is, everything seems normal until it says " == Everyone is busy/congested at this time (1:0/0/1)" |
| 09:19.45 | kaldemar | Kchehab: are you having a problem with that? |
| 09:19.51 | jgoo | Gugge, also " dialparties.agi: Extension 100 has ExtensionState: 4" looks weird |
| 09:20.17 | Gugge | jgoo, you expect me to use time on that pastebin ? |
| 09:20.33 | jgoo | Gugge, what is wrong with the pastebin? you mean the content, or that site? |
| 09:20.40 | Gugge | the content |
| 09:20.44 | jgoo | is there a syntax highlighting pastebin? |
| 09:20.57 | jgoo | Gugge, too verbose? not verbose enough? wrong content? |
| 09:20.59 | Gugge | correct linebreak would be a start |
| 09:21.11 | jgoo | you want SIP debug? Gugge , same linebreak as I have... |
| 09:21.23 | jgoo | Gugge, resize window? |
| 09:21.41 | Gugge | in my internet explorer its just one long line |
| 09:22.08 | jgoo | Gugge, yeah that pastebin has set wrapping, how crazy, I'll put onto another less insane site (and use safari, chrome or FF (in that order! :p )) |
| 09:23.19 | jgoo | Gugge, here : http://pastebin.com/d22a5d5aa |
| 09:23.29 | jgoo | with some highlighting, but I don't approve of the font weighting |
| 09:23.38 | Gugge | why do you dial "SIP/100&SIP/100&SIP/100" |
| 09:23.47 | Gugge | and try to set sip debug on too |
| 09:23.57 | Gugge | then we can see what the phone answers to that |
| 09:24.32 | Gugge | but dialing the same peer 3 times at the same time seems .... strange .... |
| 09:25.12 | jgoo | Gugge, well... the phone works now (I did misdn reload, didn't work... reload asterisk, don't think work, reboot device... working) |
| 09:25.15 | jgoo | SIP/100&SIP/100&SIP/100 << yeah, wtf |
| 09:25.27 | jgoo | How is that, I never ever, what is this? |
| 09:25.44 | Gugge | paste you extensions.conf .. and i dont have to guess that one either |
| 09:25.45 | jgoo | What even puts the ampersands in there? |
| 09:27.03 | Gugge | and why would it helt reloading misdn when its a call to SIP/100 that "fails" ? |
| 09:29.37 | kaldemar | jgoo: freepbx... |
| 09:30.26 | *** join/#asterisk jmworx___ (n=jeval@mail.octasic.com) |
| 09:30.38 | jgoo | Gugge, I didn't know this, I was testing an incoming call, last time it was stuck, for a different reason (I was just seeing if the same) an misdn reload fixed it |
| 09:31.05 | Gugge | stop guessing, and trace you way though the errors |
| 09:31.08 | jgoo | I hadn't debugged enough at that point, I just got a call that only the analogue systems were calling, then I saw asterisk was getting the call anyway |
| 09:31.10 | Gugge | thats how you fix things |
| 09:31.24 | jgoo | kaldemar, yeah, I guess so, stupid thing... |
| 09:31.58 | jgoo | Gugge, you are right, I am not so much guessing as trying to interpret things the best I can with my knowledge, but I know not to guess on thigns |
| 09:31.59 | Gugge | but if you enable sip debug, you can see why SIP/100 is busy |
| 09:32.41 | jgoo | Gugge, I rebooted it, and I think that stopped it now... I will have to wait until it happens again - is there any performance issues with keeping SIP debug on always? |
| 09:32.52 | jgoo | or can I set a trigger, if this happens, to enable sip debug? |
| 09:33.24 | jgoo | I mean... as a programmer... if an exception is thrown, it need to be handled, if it affects the user I'd better have an email pinging my phone within 30 seconds |
| 09:33.51 | jgoo | So, it'd be nice to maybe setup triggers, I think there are 'handle trunk errors' but not for sip calls right? |
| 09:34.33 | kaldemar | do it in the dialplan. |
| 09:35.10 | *** join/#asterisk Utopiah (n=libre@rps7452.ovh.net) |
| 09:35.22 | jgoo | yeah (I hate the dialplan syntax, really hate it) I guess I am going to have to swallow this pill of hatred and do it |
| 09:35.40 | jgoo | but I will say this, I am leanring the wrong way of doing dialplan syntax. Who designed this anyway? |
| 09:35.54 | Utopiah | anybody using Asterisk with french DID provider ippi.fr ? |
| 09:36.17 | kaldemar | you're definitely learning the wrong way if you start with freepbx. |
| 09:36.28 | jgoo | kaldemar, not really |
| 09:36.32 | jgoo | (wait for it) |
| 09:36.34 | Gugge | yes really |
| 09:36.45 | Gugge | start with an empty dialplan, and make it do what you want |
| 09:36.45 | jgoo | sure, it isn't the right way to learn the dial plan intricacies |
| 09:37.31 | jgoo | but, I wanted a 0 time install, reproducible system that would be easily configurable - this seemed like the way to do it - not saying it is the right way - because freepbx sucks, but the concept isn't invalid |
| 09:37.38 | jgoo | Gugge, you are right |
| 09:38.10 | jgoo | kaldemar, so while you are right, it doens't invalidate the process - also where are the docs for starting out with a blank dialplan and making your own? |
| 09:38.16 | jmkgreen | what's the usual suspect when audio only works in one direction (IAX2) ? |
| 09:38.26 | Gugge | but, if you use freepbx you should use what it gives you ... and stop trying to deal with the dialplan yourself |
| 09:38.32 | jgoo | I've gotten down to some leaf documents at every stage of this, and been shown lists of 'possible values'... not so much docs as a dictionary |
| 09:38.39 | Gugge | if you require something freepbx cant do ... dont use it |
| 09:39.00 | jgoo | Gugge, I've been wanted to write the dialplan myself anyway, but all I want is 'incoming call, ring this phone' |
| 09:39.17 | Utopiah | basically my question about ippi.fr is that they provide a DID but when I use their server they seem to only handle voice calls, not SMS, if I redirect the DID they provided me to my own instance of Asterisk, would it let the me handle SMS? (or is it somehow inherent to the DID?) |
| 09:39.25 | kaldemar | jgoo: the docs are in asterisk itself. in source package under doc/ and application and function documentation in CLI. |
| 09:39.34 | Gugge | jgoo, a 2 line dialplan can do that :) |
| 09:39.47 | jgoo | kaldemar, they are on the web? |
| 09:39.53 | kaldemar | jgoo: or, you can grab the book and learn it. |
| 09:40.10 | kaldemar | jgoo: of course they are. where do you think people get the packages? |
| 09:40.10 | jgoo | kaldemar, I've browsed the ~book !book or whatever |
| 09:40.31 | jgoo | there are 400 pages of "so then this guy flew a kite, and that is how we developed electricity, and chapter 11 how we came up with the name" |
| 09:40.34 | jgoo | seriously |
| 09:40.45 | jgoo | kaldemar, are they in html |
| 09:40.55 | jgoo | can I click and navigate and google through them |
| 09:41.07 | kaldemar | no. are you incapable or reading other than html documents? |
| 09:41.09 | jgoo | or are they expecting google to tar xvf the docs package and index it? |
| 09:41.13 | Gugge | no, you can download the source, and look in the doc dir |
| 09:41.17 | xrmx__ | if a send BYE to b and b respond with SIP/2.0 481 Call Leg/Transaction Does Not Exist does it mean that b fscked something? |
| 09:41.18 | jgoo | kaldemar, google reads WAY faster than me |
| 09:41.37 | kaldemar | http://svn.digium.com/svn/asterisk/tags/ <-- there, go look at your specific version |
| 09:41.47 | jgoo | I read a book a week, google reads a book a millisecond, I prefer to ask him where to find the pertinent parts :p |
| 09:41.53 | kaldemar | jgoo: google also gives you all kinds of invalid crap |
| 09:42.47 | kaldemar | core show applications, core show functions <-- those commands are your friends |
| 09:42.53 | jgoo | kaldemar, exactly, and http://svn.digium.com/svn/asterisk/tags/1.4.24/doc/ may be the reason... I am not sure this is even indexe |
| 09:43.03 | jgoo | kaldemar, I tell you, I was impressed by the cli |
| 09:43.30 | jgoo | I didn't realise there was nice tab completion and lists of commands, just playing around helps you learn a lot more - but where is that writte? |
| 09:43.36 | jgoo | in a doc in an svn repo? |
| 09:44.07 | Gugge | its in the book ..... |
| 09:44.13 | jgoo | I have a feeling asterisk docs should be: docs.asterisk.org/wiki/ and should start with "click here to find your distro and install asterisk", then "dialplan 101" then "using cli" |
| 09:44.23 | jgoo | Gugge, I'll read it |
| 09:44.42 | jgoo | but I'll tell you this, any book that has 11 chapters on how their history is just asking for trouble... ok, reading the book |
| 09:45.01 | Gugge | and then there is voip-info.org ... "asterisk cli" on google gives http://www.voip-info.org/wiki/view/Asterisk+CLI |
| 09:45.25 | jgoo | Gugge, kaldemar thanks guys - I know I am an asshole raging out -- s/onhow they got their name / their history /waht I wrote === fusking hell I need coffee |
| 09:45.54 | jgoo | It is just I didn't ever plan on learning this stuff... it isn't my job, I have 3 projects going on, a beach bar, and I am groking a frikking phone manual :p |
| 09:46.11 | jgoo | I love asterisk, I've written apps for it, albeit using fastagi / asterisk java |
| 09:46.17 | jgoo | the innards are like... innards and stuff |
| 09:47.10 | jgoo | I am just facepaling that subversion doesn't have a 'download dir as tar' feature |
| 09:47.54 | jgoo | Gugge, what would that 2 line extensions.conf look like? |
| 09:48.23 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 09:48.25 | Gugge | exten => _X!,1,Dial(SIP/100) | exten => _X!,2,Hangup() |
| 09:50.18 | jgoo | So, exten => _ (this is always the prefix, right?) X any, ! any number,1 (first priority), AppDial(Exten) | (pipe splits line?) second priority, AppHangup() |
| 09:51.17 | jgoo | Actually, I saw some docs that described what the _ was... I forget now... and http://svn.digium.com/svn/asterisk/tags/1.4.24/doc/extensions.txt doesn't have it |
| 09:51.45 | Gugge | | was just to avoid writing two lines here |
| 09:52.04 | jgoo | Gugge, aaah right, you should use \ |
| 09:52.17 | Gugge | or i could use \n .. or \r\n |
| 09:52.24 | Gugge | or i could write whatever the fuck i want |
| 09:52.29 | jgoo | no, that would imply a linebreak / return |
| 09:52.34 | jgoo | \ signifies a wrap |
| 09:52.40 | jgoo | =)))) yey, nerd fight! |
| 09:52.41 | Gugge | yep, and in the file here should be a linebreak |
| 09:52.54 | jgoo | Gugge, oh yeah, lol, fuck I suck now |
| 09:53.05 | Gugge | correct :P |
| 09:53.09 | jgoo | \r then, let's go osx style!! |
| 09:53.18 | jgoo | fuck windows and linux |
| 09:53.33 | jgoo | I think they changed that though, windows is \n, linux \r\n amirite? |
| 09:53.52 | Gugge | dont know, dont care .. my editors fix that for me |
| 09:53.59 | kaldemar | jgoo: _ indicates that the extension is a pattern, i.e. it contains wilcards like X, N, Z, ! or . |
| 09:54.22 | Gugge | http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns <- for _ X N Z ! . stuff |
| 09:54.33 | jgoo | yeah, that was it. Now I recall. |
| 09:54.49 | jgoo | _9|. |
| 09:57.51 | jgoo | So I want, any incoming, dial 100, then all other extensions to work. call holding and transfers, that isn't in extensions right? then I want some ring-groups - I need two phones to ring simultaneously, but just on one extension, then I want some short cuts to 10 external numbers - I can hack that up |
| 10:06.52 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 10:09.45 | *** join/#asterisk FalscherHase (n=1000@p5B2E266E.dip0.t-ipconnect.de) |
| 10:10.49 | *** join/#asterisk Ashura (n=ashura@89.119.206.194) |
| 10:11.46 | *** part/#asterisk Moz (n=me@81.179.238.144) |
| 10:16.54 | tzafrir_laptop | jgoo, actually '\' at the end of the line excapes the line-break |
| 10:17.22 | tzafrir_laptop | '\\' is originally from poetry notation, and also used in TeX |
| 10:18.10 | *** join/#asterisk Drasha (n=chatzill@195.113.224.147) |
| 10:19.43 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 10:22.13 | *** part/#asterisk smtx (n=smtx@80.153.133.87) |
| 10:25.37 | *** join/#asterisk adadelu (n=dempster@ada-bcn-fw01.adamoeurope.com) |
| 10:25.59 | Drasha | Hello, everyone. I have a question: In my extensions.conf I have something like this |
| 10:26.01 | Drasha | exten => _ZXXX,1,Dial(SIP/${EXTEN}) |
| 10:26.03 | Drasha | exten => _ZXXX,n,Dial(SIP/proxy/${EXTEN}) |
| 10:26.04 | Drasha | What I want to achieve is that the Asterisk tries to dial an extension registered to it and if the number is not registered, then call it through the proxy. It sorta works, but Asterisk does not skip the first line, because the number does not belong to it, rather it spends some 30 seconds dialing it before calling through the proxy. Is there a way to do it the way I want it? Thanks for an... |
| 10:26.06 | Drasha | ...answer. |
| 10:27.08 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 10:27.14 | adadelu | a small question regarding queues, and strategy. I currently have a linear strategy in 1.6.1.6. however the first extension is ignored. |
| 10:27.34 | FalscherHase | hi. i have a question about the AMI: why do i need write permissions on 'call' to query the channel status (Action: Status)? Shouldn't read permissions be enough? |
| 10:30.05 | jgoo | tzafrir_laptop, the more you know... \\ is new to me, just looks like an escaped \ |
| 10:32.50 | kaldemar | Drasha: set the peer as dynamic and check its ip address with func SIPPEER. if you get a valid ip (the phone is registered) dial it, otherwise through the proxy. |
| 10:33.30 | Drasha | many thanks kaldemar |
| 10:33.59 | kaldemar | Drasha: you probably even don't have to check the ip address, just set it as dynamic. |
| 10:34.29 | Drasha | they are dynamic |
| 10:36.34 | kaldemar | then show what happens when it dials, if there's no ip address, it should continue to the next priority immediately. |
| 10:36.57 | *** join/#asterisk weinerk (n=irc@bzq-79-179-45-120.red.bezeqint.net) |
| 10:36.58 | Kchehab | how to install asterisk to /etc/asterisk ,when i make install it located /etc/local/asterisk |
| 10:37.00 | *** join/#asterisk iksik (n=xk@livedata.pl) |
| 10:37.52 | kaldemar | that's just a configuration file dir. |
| 10:38.23 | Kchehab | yes --prefix=/usr ? |
| 10:39.01 | kaldemar | no. that prefix is for modules and binaries. |
| 10:39.17 | kaldemar | how are you installing asterisk? |
| 10:39.34 | Kchehab | ./configure male make install |
| 10:40.19 | TSM | im trying to work out the correct rxwink/rxflash/debounce settings for UK FXO, anyone have settings? |
| 10:41.38 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 10:45.02 | TSM | also do you happen to use a diffrent SIP secret for each extention? |
| 10:52.44 | kaldemar | Drasha: what does your sip.conf look like? use a pastebin to show it. |
| 10:55.31 | *** join/#asterisk Tim_Toady (n=moi@adsl45-72.kln.forthnet.gr) |
| 10:56.04 | Drasha | that .conf is rather large (system already in production), in which part are you interrested? |
| 10:56.49 | kaldemar | Drasha: [3066] |
| 10:57.12 | kaldemar | you get the error when dialing SIP/3066 |
| 10:57.23 | Drasha | that number is not in sip.conf - it is a number that is registered at proxy |
| 10:58.21 | Drasha | that's why I have problems with it - it tries 30 seconds to dial it locally and only after that it calls this number through a proxy |
| 10:58.38 | adadelu | nevermind, had to restart asterisk, a reload did'nt do the trick. |
| 11:06.56 | kaldemar | Drasha: SIP/3066 tries to dial a device defined in sip.conf by the name 3066. if you want to dial a number through a proxy, use SIP/3066@proxy or SIP/proxy/3066. |
| 11:15.36 | *** join/#asterisk kannan (i=kannan@121.245.29.139) |
| 11:17.03 | kannan | hello, I am not able to send or get fax . I have asterisk 1.4.x -> Audicodes FXS ATA -> Fax Machine . I have no built spandsp or enabled t.38 pass thru in asterisk. I think it is some settings to be done on the audiocodes, can anyone help? |
| 11:17.30 | kannan | the asterisk is connected to pstn with a single span Digium E-1 PRI card |
| 11:19.31 | jmkgreen | any ideas about calls via iax where the outbound leg is silent but the inbound leg has audio? This is usuing dahdi_dummy. I'm stumped. |
| 11:19.48 | jmkgreen | it's previously been working fine but the box had to be reinstalled |
| 11:20.05 | jmkgreen | the old configs were copied across and there have been no known firewall changes on our router |
| 11:20.26 | jmkgreen | the box itself has a source build of both asterisk 1.4 and dahdi |
| 11:21.25 | jmkgreen | the only clue is a log message when dialling saying that it is 'stopping sounds' |
| 11:23.01 | *** part/#asterisk Utopiah (n=libre@rps7452.ovh.net) |
| 11:26.01 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 11:26.10 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 11:26.29 | *** join/#asterisk nitin_ntfj (n=natloneh@63-255-103-4.ip.mcleodusa.net) |
| 11:29.21 | *** join/#asterisk gardo (n=gardo@121.97.179.176) |
| 11:32.28 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 11:38.03 | *** join/#asterisk Methose (n=Methose@38.101.237.250) |
| 11:40.29 | Drasha | kaldemar: sorry for the delay, I was having lunch. Anyway - the problem is that there are two sets of phones. IP phones, registered at Asterisk and analog ones registered at proxy. There is no way to distinguish them by number, so the idea was to try to call the number via Asterisk and if it doesn't work, then call via proxy. That is why the extensions.conf lines look like they do. |
| 11:43.18 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 11:43.21 | kaldemar | well you have to know which ones register to asterisk since you make configurations for them and seem to use numbers as device names. you must know which numbers to send to the proxy. |
| 11:43.54 | kaldemar | if not, there are ways to check if a device is defined, but dialing SIP/<number> is not one of them. |
| 11:45.49 | Drasha | Well yes, I know the registered ones, but doing an exhaustive enumeration in extensions.conf is unsupportable... Can you tell me which are the ways to check if the device is defined? You have told me about SIPPEER function. Is this it or are there some other? |
| 11:46.19 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 11:46.33 | kaldemar | function DEVICE_STATE might work for this, for example Dial(${IF($["${DEVICE_STATE(SIP/3066)}" = "INVALID"]?SIP/proxy/3066:SIP/3066)}) |
| 11:47.25 | Drasha | I will give it a try... it looks good. Thanks |
| 11:47.59 | kaldemar | i'm not sure if SIPPEER is useful for getting information on whether the peer is defined or not. |
| 11:48.47 | kaldemar | you can replace 3066 with ${EXTEN} if you're using patterns. |
| 11:51.19 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:53.16 | Drasha | Hmm.. I have a problem with Asterisk version. I am using 1.4.25.1. Wiki says that DEVICE_STATE() is available in form of a backport function DEVSTATE. I don't have it present and I am not eager to rebuild the Asterisk :-( |
| 11:54.18 | Drasha | and neither I have EXTENSION_STATE() or EXTSTATE() |
| 11:56.39 | fiddur | Drasha: If you want devstate for 1.4, I think you have to apply the backport... or upgrade :) |
| 11:57.32 | Drasha | yeah, I know... btw. is there a way to test the result of some asterisk function from cli? |
| 11:57.55 | Drasha | I am thinking of trying to use the SIPPEER function |
| 11:57.57 | *** join/#asterisk wonderworld (n=w@62.143.22.226) |
| 11:59.44 | *** join/#asterisk iksik (n=xk@94.23.93.206) |
| 11:59.47 | kaldemar | the functions are dialplan only. |
| 12:01.52 | Drasha | ok, I will test it with NoOp |
| 12:09.48 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 12:12.15 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.22 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:18.58 | *** join/#asterisk robl^laptop (n=robl@m4f5336d0.tmodns.net) |
| 12:27.10 | *** join/#asterisk jamesh1 (n=jhenders@66.111.111.254) |
| 12:28.25 | *** join/#asterisk dzastin (n=dzastin@dzastin-vm.ds.pg.gda.pl) |
| 12:29.53 | dzastin | hi, I have a problem with asterisk, spa-3102 and getting cid from pstn line - it works "sometimes" - can somebody help me with this? |
| 12:30.42 | jgoo | dzastin, put sip debug on the peer |
| 12:30.47 | jgoo | type that into the cli |
| 12:30.53 | jgoo | and then make a call until the callerid doesn't work |
| 12:31.10 | jgoo | also, set the 'answer' to 2 seconds, or change the time (I haven't got a 3102 I don't think) |
| 12:31.37 | Drasha | kaldemar: many thanks for your pointers, this expression did the trick - exten => _ZXXX,n,Dial(${IF($["${SIPPEER(${EXTEN}|context)}" = "users"]?SIP/${EXTEN}:SIP/proxy/${EXTEN})}) - because every phone registered at asterisk falls into "user" context |
| 12:31.45 | jgoo | dzastin, change the callerid type or localization of the device |
| 12:33.45 | dzastin | i have voip/pstn answer delay = 3 and this works sometimes - reloading asterisk or the gateway helps for 2-3 calls |
| 12:34.21 | dzastin | where can I change this callerid type and to what? |
| 12:35.18 | [TK]D-Fender | Drasha: Interesting approach.... however if you're looking to throw all non-local "extensions" to your proxy you don't need to compare to a fixed value like "users", you could get away with simply comparing as "non-blank" |
| 12:36.18 | *** join/#asterisk kazaa_lite (n=msaleem@94.193.98.124) |
| 12:36.32 | *** join/#asterisk gain_ (n=gain@host-78-12-243-216.cust-adsl.tiscali.it) |
| 12:37.34 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:37.59 | *** join/#asterisk Brooklyn (i=miracle@the.city.that.never-sleeps.us) |
| 12:38.45 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 12:39.34 | Drasha | d-fender: yes, you are right. That should do the trick as well and in a cleaner way... |
| 12:49.53 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 12:55.31 | jaytee | for some reason I can't setup an account on asterisk.org. I even tried my email addresses in case I'd setup an account and forgot and it just goes red. Is there some kind of review process for that site for new accounts? |
| 13:00.54 | *** part/#asterisk FalscherHase (n=1000@p5B2E266E.dip0.t-ipconnect.de) |
| 13:02.00 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 13:11.01 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 13:11.18 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:11.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:20.23 | *** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62) |
| 13:21.31 | *** join/#asterisk Khratos (n=khratos@190.166.103.144) |
| 13:32.32 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 13:32.51 | *** join/#asterisk knobo (n=user@88.87.57.60) |
| 13:33.18 | knobo | Does anyone have experience with running asterisk on vmware? |
| 13:33.47 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:36.09 | jaytee | knobo, I've done it in testing but not in a production environment. I know some people run it that way in production systems that only need SIP connectivity. |
| 13:41.33 | *** join/#asterisk garymc (n=garymc@81.134.0.102) |
| 13:42.04 | *** join/#asterisk jmkgreen (n=chatzill@fentech.gotadsl.co.uk) |
| 13:42.30 | Brooklyn | I normally configure Asterisk servers for The Netherlands but now I need to configure one for the US. Having some issues with the north american dialplan. In The Netherlands I make rules for cellphones, numbers you have to pay for and national/international traffic. |
| 13:42.43 | jmkgreen | hi does anyone know why my iax2 trunk would be suffering a one-way audio problem? |
| 13:42.52 | jmkgreen | it can receive audio but not send it |
| 13:43.05 | Brooklyn | What are the rules for the US? |
| 13:43.07 | eppigy | HELLO |
| 13:43.18 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 13:43.21 | Brooklyn | Cellphones start with ????XXXXX etc. |
| 13:43.46 | Brooklyn | If anyone could help me out I would REALLY appreciate that |
| 13:44.26 | Brooklyn | In The Netherlands, you dial a cellphone starting with 06 so I use this rule; |
| 13:44.27 | Brooklyn | exten=>_06XXXXXXXX,1,Noop(MOBILE-TRAFFIC) |
| 13:44.27 | Brooklyn | exten=>_06XXXXXXXX,n,Dial(DAHDI/G1/${EXTEN}) |
| 13:44.35 | Brooklyn | What should I do for a US Asterisk server |
| 13:45.06 | Gugge | I wish i lived in a country where mobile numbers were that easy to detect :) |
| 13:45.17 | Brooklyn | Hehe I was afraid it would be more difficult for the US :P |
| 13:45.18 | Gugge | here 8181xxxx is a mobile series, and 8282xxxx is a fixed series :P |
| 13:45.28 | Gugge | i have no idea how it is in the US |
| 13:45.32 | Brooklyn | Ok |
| 13:45.41 | Brooklyn | Americans around to help me out? |
| 13:45.50 | [TK]D-Fender | Brooklyn: there is no such thing as a "cellphone prefix" in North America |
| 13:46.16 | Brooklyn | Ok :P How can I detect someone calling to a cellphone or national number in Asterisk? |
| 13:46.44 | [TK]D-Fender | Brooklyn: What part of "impossible" did you not understand? |
| 13:46.51 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:47.02 | Gugge | "no prefix" doesnt always meen "impossible" :) |
| 13:47.17 | Brooklyn | So you're telling me in the US you do not know make a difference between calls to cellphones or landlines? |
| 13:47.19 | [TK]D-Fender | Gugge: Take this last one as a hint then :) |
| 13:47.30 | Gugge | [TK]D-Fender: sure :) |
| 13:47.31 | [TK]D-Fender | Brooklyn: correct |
| 13:48.15 | Brooklyn | So in the US you pay the same for calling to a cellphone and a landline? |
| 13:49.12 | [TK]D-Fender | Brooklyn: Yes |
| 13:49.23 | Brooklyn | Damn we suck in Holland :S |
| 13:49.31 | Brooklyn | Calling to a cellphone is way more expensive |
| 13:49.48 | Brooklyn | That explains why I make the cellphone exceptions ;) |
| 13:49.54 | _ShrikE | That is typical in many countries |
| 13:50.11 | _ShrikE | but not the us |
| 13:50.46 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:51.47 | Brooklyn | My other question regarding dialing rules. Dialing to a landline in the US always starts with a 0 ? |
| 13:51.56 | Brooklyn | Or do you dial the area code without a 0 |
| 13:53.07 | Brooklyn | So if I'm in New York and I need to dial someone in Main, would I dial 0207 or just 207? |
| 13:53.17 | _ShrikE | 207 or 1207 |
| 13:53.21 | [TK]D-Fender | Brooklyn: No, NEVER 0 |
| 13:53.23 | [TK]D-Fender | ~nanpa |
| 13:53.24 | infobot | from memory, nanpa is North American Numbering Plan Administration; the organization responsible for administering the integrated telephone numbering plan serving 19 North American countries. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.com/ |
| 13:53.26 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 13:53.28 | _ShrikE | 0207 would be an operator assisted call |
| 13:54.36 | *** join/#asterisk Daremonai (i=c1e3b8de@gateway/web/freenode/x-lmcsqkicvmoxrkmh) |
| 13:54.38 | Brooklyn | Okay, this is going to be easy then :) I only need to make a difference between US national numbers and international. International is everything starting with 00, it will go over the SIP line, everything else will be national and will be going through the normal network |
| 13:54.45 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:54.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:56.31 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:57.01 | [TK]D-Fender | Brooklyn: Int'l = 011 + county + dest |
| 13:57.06 | [TK]D-Fender | country codes* |
| 13:59.58 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:01.18 | Daremonai | Hello, can I use asterisk, with a normal modem, and my LAN card, to call people from my pc as well as receive calls on my pc? (or can that be done without the use of asterisk), either way, can you point me to a tutorial or something that will help me in doing so? |
| 14:01.31 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:02.31 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:03.12 | [TK]D-Fender | Daremonai: No. Normal "modem" are not usable with * |
| 14:03.30 | [TK]D-Fender | Daremonai: You need a supported FXO device. |
| 14:03.48 | Daremonai | [TK]D-Fender: hmmmm.... alright.. thanks |
| 14:04.02 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 14:04.12 | Daremonai | any idea how i could do what i said earlier with a normal modem? |
| 14:04.32 | [TK]D-Fender | Daremonai: No. Normal modems are not voice interfaces and ahve duplex issues, etc. |
| 14:04.50 | [TK]D-Fender | Daremonai: they are DATA devices. |
| 14:05.06 | Daremonai | [TK]D-Fender: oh, alright... thanks.. |
| 14:06.02 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 14:06.04 | Brooklyn | [TK]D-Fender for international calls, one should always start with 011 and not 00? |
| 14:06.19 | Brooklyn | 00 is a European thing I guess? |
| 14:06.21 | *** join/#asterisk Jumpie (i=n3rdz@76.100.241.4) |
| 14:06.26 | [TK]D-Fender | Brooklyn: Correct |
| 14:07.06 | Brooklyn | exten=>_011XXXXXXXXX.,n,Dial(SIP/${EXTEN}@SIPTRUNK) |
| 14:07.09 | Brooklyn | got it |
| 14:08.03 | *** join/#asterisk anonymouz666 (n=anonymou@187.28.37.118) |
| 14:08.15 | bn-7bc | [TK]D-Fender: hmm correct me if I'm wrong, bot does any country aotside the NANP use 011 as international prefix?? |
| 14:08.39 | [TK]D-Fender | bn-7bc: Sorry, I can't account for entire remainder of the planet... |
| 14:08.44 | Brooklyn | Most countries use 00 afaik? |
| 14:09.06 | *** join/#asterisk jde (n=jde_@65-122-116-130.dia.static.qwest.net) |
| 14:09.16 | Brooklyn | So the US uses 011 |
| 14:10.25 | *** join/#asterisk psilikon (n=psilikon@173.65.4.24) |
| 14:10.59 | Jumpie | fender..aw why not |
| 14:11.00 | Jumpie | lol |
| 14:11.44 | *** join/#asterisk retentiveboy (n=pdugas@67.211.6.18) |
| 14:14.01 | DMeloUK | anyone have exp with the aa50 ? |
| 14:18.04 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:18.40 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 14:19.23 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 14:21.58 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:21.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:22.53 | *** join/#asterisk moy (n=moy@74.12.134.3) |
| 14:26.31 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:26.31 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:29.34 | *** join/#asterisk coolthreads (n=coolthre@203-97-238-71.cable.telstraclear.net) |
| 14:35.32 | *** join/#asterisk mattblue (i=Matt@61.17.22.85) |
| 14:36.07 | mattblue | Hello all. :) |
| 14:36.57 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 14:37.40 | mattblue | I have been trying to do some work on Asterisk and am having trouble bonding/bridging two existing calls. |
| 14:38.03 | mattblue | Does someone have the time to point me in the right direction? |
| 14:38.20 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 14:38.53 | Naikrovek | mattblue: one moment |
| 14:39.46 | mattblue | Thanks Naikrovek. |
| 14:39.59 | Naikrovek | well i don't know the answer but someone will chime in soon |
| 14:40.01 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 14:40.06 | *** join/#asterisk hugorebelo (n=hugorebe@200.171.132.124) |
| 14:40.36 | mattblue | I will wait :) |
| 14:40.36 | Naikrovek | patience will serve you well in here this week i think |
| 14:40.53 | mattblue | Do you mean there is not too many folks around this week? |
| 14:40.57 | Naikrovek | Astricon is happening right now and many of the folks in here are busy schmoozing and networking |
| 14:41.22 | mattblue | Ah okay. Talk about timing! |
| 14:41.27 | Naikrovek | don't worry |
| 14:41.34 | Naikrovek | you'll get your answer in short order |
| 14:41.37 | Naikrovek | just not immediately |
| 14:42.07 | mattblue | Where is the conference being held at? |
| 14:42.18 | Naikrovek | glendale, AZ |
| 14:42.45 | mattblue | Great. Are you there as well? |
| 14:43.02 | mattblue | I will hang around and see if Google can help in the meantime. No luck so far. |
| 14:43.12 | Naikrovek | no, i'm in illinois |
| 14:43.54 | [TK]D-Fender | mattblue: before asking for directions you should show us where you ARE. |
| 14:45.13 | Naikrovek | yes show us what's not working |
| 14:48.34 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 14:49.34 | *** join/#asterisk xmnt (n=xmnt_@74.196.13.109) |
| 14:50.01 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 14:50.27 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 14:51.13 | mattblue | The work is back at my office, I am at home currently. Let me explain the scenario. Basically it is website where customers can book timeslots of consultants. When a customer books a time slot, a scheduled call will be sent from the phone system to that customer on the number they provide.So the integration is to insert a scheduled call into asterisk trixbox to intiate the call. |
| 14:51.28 | mattblue | I have been able to do the following things so far. |
| 14:51.42 | mattblue | Generate a call file with desired details and using this file make an automated call. Asterisk will play an automated message to the customer if he picks the call, otherwise disconnect.I am also able to record the calls. |
| 14:52.17 | mattblue | I am stumped at making asterisk bridge the customers call to the consultant; if the customer picks up. |
| 14:53.41 | [TK]D-Fender | mattblue: Your call file already dials out and then dumps them int he dialplan. So just DIAL your consultant. there is no "bridge 2 pre-existing calls" scenario |
| 14:53.57 | [TK]D-Fender | mattblue: the consultant isn't already on a call with your server. Youa re contacting them. |
| 14:55.51 | xmnt | Hi, I'm new to asterisk and have a question |
| 14:56.05 | shido6 | ask, xmnt |
| 14:56.07 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:56.21 | xmnt | I've read through the docs and am unsure what I need to setup a voip network |
| 14:56.30 | shido6 | oh dear |
| 14:56.49 | xmnt | I'm a linux admin so I have no problem w/ the technical side of things |
| 14:57.29 | shido6 | well - lets flesh out some details |
| 14:57.48 | shido6 | what do you have, what do you want to do , and what kind of budget are you working with ? :) |
| 14:57.54 | xmnt | but I'm not sure about how I obtain a number |
| 14:58.31 | xmnt | we'll right now I'm testing to see what's possible - really all we need is a system for several developers who all telecommute |
| 14:58.41 | shido6 | How many simultaneous callers do you expect on this one number? |
| 14:58.49 | xmnt | maybe 5 |
| 14:59.12 | shido6 | do these devs who telecommute already have ip phones or softphones or will you be providing them these resources? |
| 14:59.53 | xmnt | we've looked into getting ip phones - I have my own(a cisco) that I use for another company |
| 14:59.58 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:00.16 | shido6 | do you like your cisco ? |
| 15:00.21 | xmnt | but I've also toyed w/ the idea of using softphones initially |
| 15:00.45 | shido6 | I would recommend an ip phone with a cordless headset if our budget allows |
| 15:00.50 | shido6 | your |
| 15:00.51 | xmnt | yeah it works - i don't have any issues w/ sound cutting out like I did w/ softphones |
| 15:01.27 | shido6 | or a corded headset if you dont want them walking around a lot :) |
| 15:01.43 | shido6 | if you want them happy get a cordless headset for each of your ip phones |
| 15:01.50 | xmnt | it's a def. possibility - communication has become such a hurdle that we're at the point where whatever solves the problem is what we'll do |
| 15:02.03 | shido6 | hehehe |
| 15:02.31 | xmnt | skype is not working - and basically we're calling each other on cell phones |
| 15:02.42 | shido6 | yikes |
| 15:02.56 | shido6 | not all on a family plan ? :) |
| 15:03.02 | xmnt | haha, no |
| 15:03.21 | shido6 | maybe you should :) |
| 15:03.37 | shido6 | and add have each of the devs include the number you buy to their plan |
| 15:03.40 | xmnt | conferences are a mess - a mix of people on skype relaying info and people on their home phones etc... it's actually kind-of funny |
| 15:04.05 | shido6 | well asterisk can assist as you can own your own conferencing system |
| 15:04.07 | *** join/#asterisk Nafiux (n=IceChat7@189.226.72.140) |
| 15:04.23 | shido6 | and use such apps as appkonference <-------- go on, google it...... you know you want to |
| 15:04.27 | [TK]D-Fender | xmnt: So far all you need is a box to run * on and a connection with enough bandwidth for your calls |
| 15:04.54 | Jumpie | anybody use dim dim? |
| 15:05.02 | shido6 | reminds me of dim sum |
| 15:05.04 | Jumpie | are there hooks/mods that interface with asterisk? |
| 15:05.04 | xmnt | [TK]D-Fender, we've got both of those -- dedicated server w/ plenty of fast bandwidth and everyone uses broadband |
| 15:05.06 | Jumpie | lol |
| 15:05.16 | *** join/#asterisk viraptor (n=viraptor@79.135.98.151) |
| 15:05.17 | Jumpie | dim dim is like a whiteboard, web collaboration thing kinda like webex |
| 15:05.19 | shido6 | xmnt: then you're halfway there :) |
| 15:05.26 | [TK]D-Fender | xmnt: Ok, you're set then. For conferences, MeetMe will probably do jsut fine and is built-in |
| 15:06.01 | [TK]D-Fender | Jumpie: * does voice. Thats it. |
| 15:06.22 | [TK]D-Fender | Jumpie: Well... a certain amount of video as well.. |
| 15:06.32 | [TK]D-Fender | Jumpie: but no "interactive" services |
| 15:06.37 | viraptor | hi, is it safe to modify the astdb directly while asterisk is running? are the values read from the file every time they're requested? (i.e. no buffering) |
| 15:07.37 | [TK]D-Fender | viraptor: Yes, its safe |
| 15:07.37 | shido6 | thats why its there, viraptor :) |
| 15:07.37 | viraptor | [TK]D-Fender: cool, thanks |
| 15:07.37 | Jumpie | fender...no no.i nkow |
| 15:07.38 | *** join/#asterisk atz (n=atz@cpe-71-64-0-61.insight.res.rr.com) |
| 15:07.38 | Jumpie | im talkin 3rd party tie ins |
| 15:07.38 | Jumpie | like zimbra zimlets |
| 15:07.50 | xmnt | is the asterisk from the yum repos up to date or do you recommend building from source? |
| 15:08.00 | shido6 | build from source |
| 15:08.01 | [TK]D-Fender | xmnt: Source |
| 15:08.49 | mattblue | [TK]D-Fender : Sorry, just stepped out for a sec. So if there is already a call initiated with a number from the box, I can just DIAL another number and then connect them both? Has to be automated as well. |
| 15:09.54 | [TK]D-Fender | mattblue: .... your 1st person is in the DIALPLAN. You are doing whatever the hell you want from there. |
| 15:10.19 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 15:10.39 | *** part/#asterisk viraptor (n=viraptor@79.135.98.151) |
| 15:10.52 | [TK]D-Fender | mattblue: channel is in the dialplan, doesn't matter that * called them instead of them calling *. |
| 15:10.52 | mattblue | [TK]D-Fender : Thanks a lot mate. |
| 15:11.23 | mattblue | [TK]D-Fender : I will go check it out. Didn't seem to be working when i tried it, so i am sure I have been doing something wrong. |
| 15:13.11 | *** join/#asterisk e4 (n=e4@cpe-76-84-81-72.neb.res.rr.com) |
| 15:13.17 | *** join/#asterisk Khratos (n=khratos@190.166.103.151) |
| 15:14.16 | *** join/#asterisk jmkgreen (n=chatzill@fentech.gotadsl.co.uk) |
| 15:14.23 | Khratos | Good afternoon. |
| 15:14.40 | jmkgreen | I'm having real issues with a reinstalled Asterisk box |
| 15:14.50 | Khratos | Does someone knows if there's a limit in the length of the input stream that can be sent to AMI? |
| 15:14.53 | jmkgreen | IAX audio works but in one direction only |
| 15:15.15 | jmkgreen | I can't understand it |
| 15:15.58 | jmkgreen | asterisk logs show "stopped sounds" as the phone begins ringing |
| 15:16.06 | jmkgreen | but why I don't have any idea |
| 15:16.26 | jmkgreen | it was working fine before the O/S had to be reinstalled |
| 15:16.47 | Khratos | Have you checked if it is NAT or Firewall related? |
| 15:16.54 | Khratos | (The audio part) |
| 15:17.05 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 15:17.14 | jmkgreen | funny thing is that we're behind a NAT/firewall yet the audio coming back in is the direction that is it working. Outbound audio is the bit that's silent |
| 15:17.25 | jmkgreen | asterisk reckons it's playing the files even |
| 15:17.43 | jmkgreen | we have a second asterisk box behind the same nat/firewall and that's working fine |
| 15:18.43 | jmkgreen | I'm basically calling an office landline though gradwell. I cannot hear what our server is speaking, but I can hit the buttons on the handset and the server shows it hears the dtmf tones |
| 15:18.45 | Khratos | Have you tried making a test with a sip account to see if it is IAX oriented problem or networking in general? |
| 15:19.04 | jmkgreen | I can't use sip due to the firewalling |
| 15:19.17 | jmkgreen | thats why we always use iax within our office |
| 15:19.58 | shido6 | its not a sonicwall is it? |
| 15:20.04 | jmkgreen | I also know it's not our provider because our office lines use the same gradwell voip provider and they are working great |
| 15:20.37 | jmkgreen | nopoe |
| 15:20.55 | jmkgreen | (just checked that one) |
| 15:21.23 | jmkgreen | I've filed a support request with gradwell but I suspect they're avoiding calling me back as they probably found as little in google as I did |
| 15:22.07 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-159.se.biz.rr.com) |
| 15:22.09 | jmkgreen | I can see in the asterisk sources where this log message "stopped sounds" occurs - it's within dial() when it receives a frame that has a status of -1 |
| 15:22.48 | jmkgreen | quite why I'm getting such a frame is beyond me - as I say this machine was working fine before I was forced to reinstall (disk died) |
| 15:24.22 | Khratos | Do you have SELinux on that box? |
| 15:24.47 | jmkgreen | apparently (dpkg -l | grep selinux) |
| 15:24.55 | jmkgreen | ii libselinux1 2.0.65-5build1 SELinux shared libraries |
| 15:25.08 | Khratos | what does the command 'getenforce' (in the shell) gives you? |
| 15:25.26 | jmkgreen | no such command |
| 15:25.31 | jmkgreen | even as root |
| 15:26.07 | *** part/#asterisk dustybin (n=dustybin@thinkdebian.org) |
| 15:26.59 | Khratos | Are you sure that there isn't any iptables fancy rules applied on the box? |
| 15:28.16 | Jumpie | you know what i notice, alot of helps and guides that tell people how to lock down asterisk/trixbox, whatever and explain about changing passwords, almost always leave out 1 or 2 critical files thus breaking the system |
| 15:28.30 | Jumpie | and people scramble tryin to revert, and its not quite the same..i'm seeing this a lot |
| 15:29.58 | jmkgreen | Khratos: there are no iptables rules on this box at all |
| 15:30.24 | Khratos | Is it the same distro and version of the machine that is running without problems? |
| 15:30.25 | jmkgreen | I went to our in-house Asterisk guy who said it's either firewall or ports |
| 15:30.48 | jmkgreen | the firewall has been ruled out, and with IAX the port can't be to blame (at least I can't see how) |
| 15:31.08 | jmkgreen | Khratos: No, I'm now running Ubuntu 9.04 on this new box |
| 15:31.27 | jmkgreen | However, both dahdi and asterisk have been compiled from source as per the previous installation |
| 15:31.56 | shido6 | you ruled the firewall out by removing it completely from the network, yes? |
| 15:32.33 | jmkgreen | well I can't, it's part of the office infrastructure. As I said though the other asterisk box is also behind it and our phone system is working as normal |
| 15:32.51 | shido6 | did you point that ip to the new asterisk system and test? |
| 15:33.00 | jmkgreen | I'm told there have been no changes to it in weeks, and the previous installation was working within weeks ago |
| 15:33.36 | shido6 | change the ip of your new ast to the old ast and see if that works, if it does u have a rule not applied to the new ip on the old ip that you need to find |
| 15:33.39 | jmkgreen | I don't understand - the 'other' asterisk box is our phone system. This new one is purely a development IVR |
| 15:33.58 | jmkgreen | the new and old asterisk boxes are the same physical hardware with the same dhcp ip address |
| 15:34.26 | jmkgreen | other than the ubuntu distro and minute versions of asterisk (possibly) there should be no changes |
| 15:34.31 | jmkgreen | er minor versions |
| 15:34.34 | shido6 | hehe |
| 15:34.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:34.45 | jmkgreen | I even did a tcpdump |
| 15:35.15 | jmkgreen | there's clearly audio back from gradwell but there's nothing being sent to gradwell except IAX control packets |
| 15:35.47 | jmkgreen | that's the odd thing - if it were firewall/nat the cause I'd expect the problem reversed |
| 15:35.58 | [TK]D-Fender | jmkgreen: notransfer=yes <------- |
| 15:35.58 | Khratos | :/ In a desperate intent to debug this, would take the new box to a place with internet and without firewall, and test |
| 15:36.29 | jmkgreen | Khratos: appreciate that, but that particular test requires significant effort :-) |
| 15:37.17 | Khratos | [TK]D-Fender, may I ask you something? |
| 15:37.39 | [TK]D-Fender | Khratos: You just did. I may even permit ANOTHER question :) |
| 15:38.41 | *** join/#asterisk Nafiux (n=IceChat7@189.226.72.140) |
| 15:38.41 | jmkgreen | [TK]D-Fender: No change |
| 15:38.48 | Khratos | :) Ok. Look, Sending this ( http://slackware-es.com/ami-input.txt ) to AMI via Telnet and/or PHP script results in no error message, but voicemail.conf not getting complete written |
| 15:38.48 | Nugget | telnet is eeeeeeevil! |
| 15:39.10 | jmkgreen | <PROTECTED> |
| 15:39.18 | Khratos | Do you have any Idea of what could be causing that? (Tried pastebin, but the content of the file did not pass their antispam filter) |
| 15:40.03 | Khratos | On CLI and AMI everything seems to be Ok, but the file gets partially written, not all entries added. |
| 15:40.38 | Nafiux | phones |
| 15:40.43 | [TK]D-Fender | Khratos: Never used that option |
| 15:40.52 | Nafiux | ~phones |
| 15:40.53 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 15:41.11 | *** join/#asterisk timeshell (n=chatzill@142.46.193.194) |
| 15:41.32 | jmkgreen | scratches his head |
| 15:41.35 | tzafrir_laptop | Nugget, most of the time when people mention 'telnet' they refer to the manager interface or so |
| 15:41.36 | jmkgreen | ... some more |
| 15:41.46 | Khratos | It works with less entries, as a charm. but when I try to append more entries, it does not work |
| 15:41.59 | [TK]D-Fender | tzafrir_laptop: Silly wabbit |
| 15:42.05 | tzafrir_laptop | I suggest you update you 'evil' text to refer to 'use ssl' or whatever if you actually care about it |
| 15:48.13 | IBC_jkenney | Has anyone noticed any memory leak with 1.6 where it keeps resources tied up |
| 15:48.42 | [TK]D-Fender | IBC_jkenney: Which 1.6? |
| 15:48.42 | [TK]D-Fender | IBC_jkenney: Only about a few dozen releases so far... |
| 15:48.55 | IBC_jkenney | 1.6.1.4 |
| 15:49.05 | IBC_jkenney | Yeah but this one is pretty bad |
| 15:49.14 | [TK]D-Fender | IBC_jkenney: Already 2 behiind.... |
| 15:49.23 | IBC_jkenney | Really |
| 15:49.39 | [TK]D-Fender | IBC_jkenney: You should try reading the topic occasioanlly |
| 15:50.35 | IBC_jkenney | anything to be afraid of with a upgrade to the new one |
| 15:52.03 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-41-58.dhcp.mrqt.mi.charter.com) |
| 15:52.28 | pmhaddad | are there any known issues with compiling asterisk-addons 1.6.1.1 on Lenny? |
| 15:52.55 | jmkgreen | are the issues.asterisk.org logins the same as for forums.asterisk.org ? |
| 15:52.58 | pmhaddad | its crashing out on make when it tries to compile chan_ooh323 |
| 15:54.34 | Jumpie | lawl |
| 15:56.39 | pmhaddad | http://pastebin.com/m598c3ed9 if anyone wants to take a look |
| 15:56.40 | pmhaddad | thanks |
| 15:57.07 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 16:04.32 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:09.29 | DigitalFlux-AFK | i need to know the traffic rate of the g729 codec that asterisk is using .. |
| 16:09.37 | DigitalFlux-AFK | can any body help me out with that ? |
| 16:10.04 | *** part/#asterisk bzing2 (n=dr105@194.66.208.236) |
| 16:11.50 | brad_mssw | DigitalFlux-AFK: as always, search on voip-info.org, and you'll come up with useful things like : http://www.voip-info.org/wiki/view/Bandwidth+consumption |
| 16:15.59 | DigitalFlux-AFK | brad_mssw: Checking.. Thanks |
| 16:16.15 | pmhaddad | never mind, just disabled chan_h323 with make menuselect |
| 16:17.10 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 16:17.25 | Dovid | hi. i just had an argument with some one on Asterisk+NAT |
| 16:17.44 | DavidR2008 | I have a gxp 2000 phone that doesn't send the correct touch tones to an asterisk 1.6.2.0-rc3 server using a Multi-Purpose Key during a call. I've upgraded the phone to the latest firmware and that seemed to make the problem worse. It was working correctly while be used with an asterisk 1.4.23.1 server last week. Any help would be greatly appricated. |
| 16:17.55 | Dovid | if Asterisk is on a public IP and you have two end users that are behind NAT, Asterisk needs to be in the media path ? |
| 16:19.01 | *** join/#asterisk Gugge (n=Guggeman@vlan2.dlxhosting.dk) |
| 16:21.30 | shido6 | why did you upgrade, DavidR2008 ? |
| 16:21.59 | Qwell | Dovid: unless the devices have a way to traverse the NAT |
| 16:22.07 | Qwell | ie; direct port forwarding |
| 16:22.46 | DavidR2008 | I was on here yesterday with the same problem and that was one of the steps [TK]D-Fender wanted me to try |
| 16:27.44 | [TK]D-Fender | 's other suggesting was to make a bonfire out of them... |
| 16:27.59 | *** join/#asterisk errotan (n=errotan@5403E441.catv.pool.telekom.hu) |
| 16:28.06 | DavidR2008 | I'm well aware :-D |
| 16:28.14 | DavidR2008 | not really an option for me. |
| 16:29.32 | Qwell | oh, it's an option. it's just one you refuse to take. |
| 16:29.46 | IBC_jkenney | anyone in here play with openldap and the spa962 using them for the directory? |
| 16:29.57 | DavidR2008 | I'm not allowed to burn company property without approval ;-) |
| 16:30.08 | Qwell | DavidR2008: so get approval ;) |
| 16:30.28 | Qwell | There are even people here who might pay to see a video of it on youtube afterwards. |
| 16:30.57 | DavidR2008 | do you think I could get enough money to pay for replacement phones? |
| 16:31.00 | IBC_jkenney | <========== do you take cc's ? |
| 16:31.38 | Qwell | DavidR2008: reminds me of what a friend of mine did.. |
| 16:31.58 | raden_work | DavidR2008, change the tone settings |
| 16:32.03 | Qwell | DavidR2008: Apple wouldn't replace his macbook under warranty, so he took a sledge hammer and a camcorder... |
| 16:32.25 | Qwell | posted it on youtube (where it got a few hundred thousand views) and emailed it to Steve Jobs. |
| 16:32.31 | Qwell | had a new macbook the following week |
| 16:32.36 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:32.50 | Qwell | it was amusing |
| 16:32.57 | raden_work | Qwell, is that how we want to teach our children to resolve issues ? |
| 16:33.00 | raden_work | lol |
| 16:33.01 | Qwell | totally my idea too. he owes me a macbook. |
| 16:33.20 | Qwell | raden_work: no, but damned if it wasn't funny |
| 16:33.35 | raden_work | Qwell, i bet you dont have a link do you ? |
| 16:33.38 | raden_work | id love to see that |
| 16:33.39 | Qwell | I do |
| 16:33.45 | Qwell | http://consumerist.com/consumer/above-and-beyond/apple-gives-macbook-smasher-a-new-macbook-274740.php |
| 16:33.48 | Qwell | link with story! |
| 16:34.33 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 16:35.55 | Qwell | actually, you'll have to click the "Previously" link |
| 16:38.21 | Jumpie | lol magic moisture cloud |
| 16:38.57 | raden_work | wow 1/2 million hits |
| 16:39.24 | raden_work | sweet |
| 16:39.39 | raden_work | if i do that with my crap blackberrys i wonder if they send me new ones |
| 16:39.42 | Dovid | Qwell: How do other devices do it ? |
| 16:40.01 | Qwell | Dovid: the devices themselves don't. you have to configure your network to make it work |
| 16:40.05 | Qwell | raden_work: doubt it |
| 16:40.09 | *** join/#asterisk outtolunc (n=me@66.218.53.172) |
| 16:40.33 | Dovid | Qwell: So if I had an SBC then "technically" there should be no NAT issues |
| 16:40.36 | raden_work | Qwell, i know |
| 16:40.43 | Jumpie | lol all the dogs started barking |
| 16:40.44 | raden_work | I gave up went back to samsung phones |
| 16:40.50 | Jumpie | rookie with that sledgehammer |
| 16:41.29 | raden_work | DavidR2008, you get it worked out yet ? |
| 16:41.38 | raden_work | Naikrovek, morning bro |
| 16:41.39 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 16:41.42 | Naikrovek | howdy |
| 16:42.22 | raden_work | crap need to reboot new kernel |
| 16:42.23 | raden_work | brb |
| 16:42.39 | DavidR2008 | raden_work, no, I changed it from RFC2833 and now the number keys send tones, but the MPK buttons don't send anything. I figure I'll contact grandstream and see what they say |
| 16:42.48 | DavidR2008 | *from RC2833 to INFO |
| 16:46.53 | *** join/#asterisk raden_work (n=jon@69.179.99.17) |
| 16:47.05 | DavidR2008 | brb |
| 16:49.23 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 16:50.57 | nny | anyone have a favorite device for a remote voip phone to FXO setup? Basically one client is gonna be in a box miles away, connecting over VoIP to the main. Want to have one or two of the SIP channels talking to an ATA that acts as a SIP to FXO converter. Not sure if there is something in the SPA line that does this without trying to be a full blown pbx |
| 16:51.26 | [TK]D-Fender | nny: SPA-3102 |
| 16:52.16 | nny | [TK]D-Fender: beautiful thanks |
| 16:52.21 | hardwire | nny: a box? |
| 16:52.38 | nny | yeah lol, that's what they call it |
| 16:52.44 | nny | it's a ticket booth for sunset cruises |
| 16:52.47 | hardwire | is it cardboard? |
| 16:52.48 | hardwire | oh. |
| 16:53.01 | nny | heh |
| 16:53.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:53.28 | hardwire | you can hardly go wrong with sipura |
| 16:54.55 | robl^laptop | sipura though was bought out by linksys, which was bought by Cisco ;-) |
| 16:56.21 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:56.49 | nny | yeah always nice when we quote linksys phones (most common) and I can just call em cisco now |
| 17:00.09 | p3nguin | raden_work: I guess no one ever taught you about kexec, huh? |
| 17:02.40 | raden_work | p3nguin, ? |
| 17:02.50 | p3nguin | (1142.23) <raden_work> crap need to reboot new kernel |
| 17:03.03 | raden_work | p3nguin, no |
| 17:03.07 | raden_work | ill look into that |
| 17:03.16 | raden_work | p3nguin, you have APC software working correct ? |
| 17:03.22 | p3nguin | Yes. |
| 17:04.21 | raden_work | im going to try to get mine working again today |
| 17:06.35 | *** join/#asterisk baijum (n=baiju@122.166.147.40) |
| 17:08.25 | *** join/#asterisk metfan2007 (n=metfan20@201.103.56.246) |
| 17:09.03 | p3nguin | There's really nothing difficult about it. |
| 17:10.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:19.24 | *** join/#asterisk buttons840 (n=buttons8@63.230.20.246) |
| 17:24.30 | *** join/#asterisk lwh (n=lwh@66.212.165.145.tor.pathcom.com) |
| 17:26.54 | *** join/#asterisk voxter (n=voxter@166.183.62.157) |
| 17:27.09 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
| 17:28.03 | *** join/#asterisk DavidR2008 (n=chatzill@208.34.240.2) |
| 17:28.44 | buttons840 | [TK]D-Fender, here is the CLI logs from the WaitForSilence problem I'm having. Maximum verbose, level 10 debug. Notice line 118 and line 133, how WaitForSilence behaves differently. The first time there is no indications of how long silence has been, and the 2nd time there are, obviously, hundreds of indications of silence. Wonder why the difference in behavior? |
| 17:28.44 | kaii | can somebody think of a way to playback a file to a caller and simultaneously call an extension? (while the file is playing back) |
| 17:29.33 | buttons840 | kaii, there is hold music, but that's all i know of, i'm don't know much though |
| 17:29.35 | CcRnp | you can use a macro on Dail function so that you can plaback a file and call a extension at the same time |
| 17:29.52 | [TK]D-Fender | CcRnp: Nope. |
| 17:30.06 | CcRnp | ? |
| 17:30.16 | [TK]D-Fender | kaii: Indeed your only real option is to use an Moh Class with a folder with only 1 file in it which yuo preplace prior to the outcall. |
| 17:30.31 | [TK]D-Fender | ccMacro doesn't get run until after they ANSWER |
| 17:30.33 | *** join/#asterisk jayrod422 (n=jarrod@204.15.220.206) |
| 17:30.35 | kaii | buttons840: well, the announcement would start at a random point when in inculde it in MOH since not every user has an indivudial MOH stream, they all share the same "radio playback" and jump in the stream at a random point |
| 17:31.04 | [TK]D-Fender | kaii: Specify another class in the DIAL |
| 17:31.17 | *** join/#asterisk grEvenX (n=even@cC0FD00C3.dhcp.bluecom.no) |
| 17:31.32 | kaii | [TK]D-Fender: i do not understand. does this start a new MOH process for each caller? |
| 17:31.59 | [TK]D-Fender | kaii: IIRC it'll start from the beginning for each... |
| 17:32.12 | kaii | i will give that a try, thx |
| 17:34.48 | DMeloUK | speaking of moh - how do I adjust the moh volume on the aa50 appliance? |
| 17:35.38 | kaii | [TK]D-Fender: my more complicated idea was using local channel and inband progress.. but i'm really unsure if this will for all channels, or work at all. |
| 17:36.18 | [TK]D-Fender | kaii: Nope.... local channel involves having an active channel thats answered |
| 17:36.35 | [TK]D-Fender | kaii: Other option involves recoding app_dial. |
| 17:36.44 | kaii | 1,Answer() 2,Dial(SIP/exten&Local/playback), in playback do: 1,Progress 2,Playback(announcement,n) |
| 17:36.55 | [TK]D-Fender | kaii: Which frankly your request warrants and is share by many. |
| 17:37.08 | [TK]D-Fender | kaii: Won't work |
| 17:37.10 | kaii | [TK]D-Fender: playback involes having an active channel too, unless your telco lets you send inband |
| 17:37.35 | kaii | mh why are you 100% sure it won't work? explain pls |
| 17:37.39 | [TK]D-Fender | kaii: app_dial playing the audio to you is one thing... using a LOCAL channel like that also runs that call in parallel, not in series |
| 17:37.46 | [TK]D-Fender | kaii: very not happening.. |
| 17:37.56 | *** join/#asterisk baijum (n=baiju@122.166.147.40) |
| 17:38.31 | kaii | [TK]D-Fender: thats what i want, call the playback and the exten in parallel |
| 17:38.45 | [TK]D-Fender | kaii: No you misunderstand... they are not MERged |
| 17:39.04 | kaii | mh i really do not understand |
| 17:39.43 | p3nguin | If I set a variable in context1, can I retrieve the value of it in context2? |
| 17:39.44 | [TK]D-Fender | kaii: 2,Dial(SIP/exten&Local/playback) <-- in this case, the Local channel will answer immediately and the SIP call will DROP <- |
| 17:39.57 | [TK]D-Fender | p3nguin: * dialplan doesn't have "scope" |
| 17:40.08 | [TK]D-Fender | p3nguin: So yes |
| 17:41.22 | kaii | [TK]D-Fender: but the local channel only contains "progress()" and "playback(,noanswer)", so why should it answer immediately? |
| 17:42.08 | p3nguin | And at what point would the variable be destroyed? When the call ends? |
| 17:42.39 | [TK]D-Fender | kaii: * doesn't pipe multiple progress indications together so you can get early media fromt he local while the other rings "silently". this just isn't going to happen.... |
| 17:43.04 | [TK]D-Fender | p3nguin: Variables are local to the channel and copied per inheritance rules into channels it spawns |
| 17:43.24 | kaii | mh |
| 17:44.41 | *** join/#asterisk cherva (n=cherva@78.128.16.162) |
| 17:45.32 | kaii | [TK]D-Fender: so it is because local channel progress is not passed to the calling channel |
| 17:46.10 | cherva | can someone help me debug an inbound calls problem ? I have all trunks registered and I can make outbound calls, but when I call from outside I get "The subscriber rejected your call" message.... |
| 17:46.15 | [TK]D-Fender | kaii: Dial will only bridge if it is considered answered |
| 17:46.35 | [TK]D-Fender | cherva: pastebin your outbound call attemp with SIP DEBUG enabled |
| 17:46.37 | [TK]D-Fender | ~pb |
| 17:46.37 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:46.39 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 17:46.54 | p3nguin | In context1, I use exten => _NXX.,2,Set(LASTCID=${CALLERID(num)}) to attempt to capture the caller's phone number into the LASTCID variable. In context2, I wanted to use exten => *69,1,SayDigits(${LASTCID}) so that I can dial *69 and hear the last number who called. This obviously does not work, so what is a better approach? |
| 17:48.42 | [TK]D-Fender | p3nguin: Won't work because as I said, its limited to the CHANNEL. When that first call hangs up the variables are GONE |
| 17:48.53 | [TK]D-Fender | p3nguin: Your call to *69 is a completely new call |
| 17:49.08 | kaii | [TK]D-Fender: but when i dial from SIP/85 to DAHDi/g1/somenonexistent number (assumed dial options r,R are not used) i can hear a friendly voice telling me foo, without the call being answered ... so dial bridges in that case, without the call being answered |
| 17:49.19 | p3nguin | [tk]d-fender: Yeah, so is there an approach that will actually work? |
| 17:49.37 | p3nguin | set a global variable, maybe? |
| 17:49.37 | [TK]D-Fender | p3nguin: "core show function DB" |
| 17:49.42 | kaii | p3nguin: |
| 17:50.13 | p3nguin | [tk]d-fender: Does this require additional database software? |
| 17:50.18 | [TK]D-Fender | p3nguin: No |
| 17:50.20 | kaii | p3nguin: you could store it in a global variable, yes. but across asterisk restart these are lost, too. better use DB, as fender pointed out |
| 17:50.42 | *** join/#asterisk kimitaka (n=swiceje@cpe-075-180-228-106.ec.res.rr.com) |
| 17:50.48 | p3nguin | With no additional software required, I am going to attempt to figure it out. |
| 17:51.10 | kimitaka | Can someone recommend me a free sip client for iphone/ipod touch? |
| 17:52.14 | kaii | [TK]D-Fender: that friendly voice on the PSTN makes me believe it would be possible, if chan_local would work with progress indication |
| 17:52.47 | CcRnp | 1.Sip-trunk inbound call === >extention200 |
| 17:52.48 | CcRnp | 2. extention200 ===>calls===>extention201 |
| 17:52.48 | CcRnp | call answered by 201 |
| 17:52.48 | CcRnp | 200 transfer SIP call to 201 |
| 17:52.48 | CcRnp | Now my question : |
| 17:52.48 | CcRnp | Is there a way to show the original caller ID of the incoming SIP call at destination extension, while transferring the call to another local extension with "Attended transfer"? if not can i atlest keep the track of calls, like |
| 17:52.52 | CcRnp | incomming call = <sip-inboundcall number> |
| 17:52.54 | CcRnp | answered = extension 200 |
| 17:52.56 | [TK]D-Fender | kaii: nCertainly not in PARALLEL with another call with "&" <- |
| 17:52.58 | CcRnp | transfered to = extension 201 |
| 17:53.00 | CcRnp | call hanguped by = extension 201 |
| 17:53.02 | CcRnp | can anyone help me out with my problem |
| 17:53.07 | [TK]D-Fender | CcRnp: PASteBIN |
| 17:53.10 | [TK]D-Fender | ~pb |
| 17:53.10 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:53.24 | CcRnp | opps sorry ! |
| 17:53.44 | [TK]D-Fender | CcRnp: and NO, an attended transfer will not show the original callerid. that is not its function. |
| 17:53.45 | kaii | [TK]D-Fender: ok, now i understood. sry for being so stubborn ^^ |
| 17:53.54 | p3nguin | Set(DB(callerid/last)=${CALLERID(num)}) |
| 17:54.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:54.52 | cherva | [TK]D-Fender: http://pastebin.com/m2ebad308 |
| 17:55.55 | [TK]D-Fender | cherva: in [nexcom2] add "insecure=port,invite" |
| 17:56.02 | [TK]D-Fender | p3nguin: better |
| 17:56.04 | kaii | [TK]D-Fender: doesnt chan_sip in 1.6.1 support sending callerid updates via SIP INFO? |
| 17:56.33 | [TK]D-Fender | kaii: Nothing I'm aware of. |
| 17:56.50 | p3nguin | After I set the entry in the DB in context1, do I then need to read the value into a variable within context2 first, then I can use SayDigits(${myvariable})... or can I use SayDigits to read the value from the DB directly? |
| 17:57.07 | kaii | CcRnp: see option "calleridupdate=yes" in your sip.conf ... worked for me |
| 17:57.24 | kaii | [TK]D-Fender: it really works |
| 17:57.31 | *** join/#asterisk gardo (n=gardo@121.97.138.207) |
| 17:57.33 | CcRnp | thanks i will check it out ! |
| 17:58.21 | kaii | CcRnp: just looked after it and found out it was part of the bristuff patchset |
| 17:58.33 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:58.34 | p3nguin | and can I read entries of the DB in the console? |
| 17:59.11 | kaii | CcRnp: there are patchsets out there providing this feature (via said option), but these are not part of the main distribution and may be badly maintained |
| 17:59.25 | cherva | [TK]D-Fender: it made a difference now I get this http://pastebin.com/d418045e1 and Check operator servicies on the display of my GSM |
| 17:59.31 | [TK]D-Fender | kaii: Possible they've added that and I'd never heard of it.. |
| 17:59.53 | kaii | [TK]D-Fender: just looked, its a patchset |
| 17:59.54 | CcRnp | where can i get those patchsets ? |
| 18:00.10 | [TK]D-Fender | cherva: -- Executing [s@macro-dial:7] Dial("SIP/nexcom2-09b514e8", "SIP/029624989,"",tr") in new stack <-- you are dialing an invalid device here |
| 18:00.28 | [TK]D-Fender | cherva: * is now answering the call and you have a FreePBX configuration issue with your inbound route |
| 18:00.29 | kaii | [TK]D-Fender: see http://svnview.digium.com/svn/asterisk/team/oej/calleridupdate/ .. will not work with all vendors, though. |
| 18:00.44 | [TK]D-Fender | cherva: And FreePBX is NOT supported here, please continue in their channel for support |
| 18:00.47 | [TK]D-Fender | ~freepbx |
| 18:00.48 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:01.02 | [TK]D-Fender | kaii: I consider that disclaimer a given.... |
| 18:01.38 | kaii | [TK]D-Fender: what do you mean? |
| 18:01.43 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:01.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:01.55 | [TK]D-Fender | kaii: "will not work with all vendors, though." |
| 18:02.14 | p3nguin | I can read it on the console: database get callerid last |
| 18:02.45 | kaii | [TK]D-Fender: as said it works well for us with snom phones and aastra (never tested myself with aastra) |
| 18:03.47 | CcRnp | howl can i use the original callerid to transfered calls for call recording ? |
| 18:06.04 | *** join/#asterisk baijum (n=baiju@122.166.147.40) |
| 18:06.28 | CcRnp | any idea? |
| 18:06.43 | kaii | [TK]D-Fender: Dial(exten,,m(announcement_class)) jumps in at random, by the way. |
| 18:08.58 | [TK]D-Fender | kaii: I believe there is a "random" flag you can set in the call... haven't fiddled with this much... |
| 18:09.29 | [TK]D-Fender | kaii: Though as I mentioned, this is a VERY popular idea that should get a feature request issued, then merged for it... |
| 18:09.42 | p3nguin | exten => *69,1,SayDigits(${DB(callerid/last)}) |
| 18:09.47 | p3nguin | works well. |
| 18:10.20 | [TK]D-Fender | CcRnp: odds are the recording has already started with a fixed name before the call is actually transfered and you can't change the name in the middle |
| 18:12.11 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 18:12.49 | *** join/#asterisk shido6 (n=shido6@74.132.202.71) |
| 18:17.33 | xmnt | o.k, so i got asterisk installed w/ freepbx and i've got it up and running - I'm showing all o.k. for system status |
| 18:18.00 | ManxPower-work | ~freePBX |
| 18:18.01 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:18.17 | xmnt | now this may seem like a silly question but how do I use it - i'm assuming something like ekiga can handle it |
| 18:18.27 | carrar | ~freePBX |
| 18:18.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:18.30 | CcRnp | 1.inbound call => Extension 200 => Answers ; recording occurs between extension 200 and inbound call 2.extension 200 => holds => incomming call ;; 3.extension 200 => calls => extension 201;; recording occurs between extension 200 and extension 201;;4.extension 200 => transfers => inblound call to => extension 201;;;;;now my question is how can i recorded the transfered call with its original callerid in db. |
| 18:18.43 | xmnt | carrar, thanks |
| 18:18.49 | buttons840 | I've confirmed my problem with WFS is related to my to my sip lines (sip protocol, qwest ipld carrier, dsl lines), when i call locally (another phone in the office) WFS works, but when calling over these sip lines, WFS works sporatically. |
| 18:22.27 | kaii | CcRnp: in DB, you can not. this is a design flaw in CDR, because it was invented when features like attended transfer where still to invent. |
| 18:23.20 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gw.teamgleim.com) |
| 18:23.41 | kaii | CcRnp: there is work in progress in asterisk development to create another way to store such information |
| 18:24.41 | leif[astricon] | CDR data? It's called CEL |
| 18:24.43 | leif[astricon] | ~cel |
| 18:24.44 | infobot | [cel] Channel Event Logging |
| 18:25.25 | kaii | the new thing, right |
| 18:26.06 | CcRnp | is it possible to record it on my own custom db , usgin agi script ? |
| 18:26.27 | kaii | no |
| 18:32.09 | *** join/#asterisk rps2 (n=rick@adsl-99-74-144-118.dsl.lsan03.sbcglobal.net) |
| 18:32.24 | rps2 | Greetings! |
| 18:32.40 | CcRnp | thank you guys ! i will work out with these features and let you know if i came up with anythine |
| 18:32.44 | CcRnp | *anything |
| 18:32.45 | buttons840 | http://pastie.org/654923 here is my dial plan and CLI is anyone care to look. When the first call to WaitForSilence does not work, but the second does. During this call I made constant noise until it was apparent from the CLI (spamming listening indicators) that WFS was listening, then i was quiet and confirmed that WFS works properly on it's second call. The dial plan context is 5 lines, if anyone wants to look. |
| 18:33.45 | rps2 | First off, a HUGE shout out to everyone who helped me get this thing sorted. It works a treat except for the directory--which never works the first time you enter data, but does work the second time. |
| 18:34.23 | rps2 | I did find a somewhat nasty bug in the GUI, though. Where would I report that? I don't see a bugzilla for *. |
| 18:34.36 | [TK]D-Fender | ~mantis |
| 18:34.37 | infobot | extra, extra, read all about it, mantis is at http://bugs.digium.com |
| 18:34.43 | [TK]D-Fender | ^^^^^^ |
| 18:34.45 | rps2 | Ah. |
| 18:34.51 | rps2 | Ok. |
| 18:35.29 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:36.04 | *** join/#asterisk voxter (n=voxter@166.183.62.157) |
| 18:40.03 | Katty | stares sleepily |
| 18:42.34 | buttons840 | something about background causes WFS to go from broken to working. dial plan is WFS; Background; WFS first WFS doesn't work, 2nd does... |
| 18:44.06 | kaii | CcRnp: http://www.venturevoip.com/news.php?rssid=2011 |
| 18:45.12 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 18:46.20 | CcRnp | thank you ! appriciated ! |
| 18:47.39 | kaii | CcRnp: consider this is (=should be) always up to date with head development, so better test the hell out of it before using it in production |
| 18:47.47 | rps2 | Ok, bug reported. |
| 18:48.13 | rps2 | Any idea why the directory has issues with the first attempt but is successful on the second? |
| 18:49.54 | CcRnp | sure ! i will test before i use it in production ! |
| 18:50.41 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:56.39 | *** join/#asterisk btsteve (n=tstevens@204.10.20.30) |
| 19:00.19 | *** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62) |
| 19:07.47 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 19:07.47 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:08.57 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-71-143.client.mchsi.com) |
| 19:15.28 | *** join/#asterisk muddd (n=fubard@c-98-224-33-102.hsd1.fl.comcast.net) |
| 19:17.04 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp12-200.adsl.forthnet.gr) |
| 19:23.54 | Katty | is it too early to listen to christmas music? |
| 19:24.13 | raden_work | YES !!!!! |
| 19:24.16 | Katty | :< |
| 19:24.18 | Katty | k |
| 19:24.22 | raden_work | can we get through November first |
| 19:24.32 | Katty | :< |
| 19:24.45 | Katty | i'd throw a snowball at you, if i had one |
| 19:25.00 | raden_work | well you have no snow soooo no music ok ! |
| 19:25.12 | Katty | <PROTECTED> |
| 19:25.45 | Katty | how about just one? :> |
| 19:27.27 | Katty | the dollar store has these cute little plastic snowflakes...they're about 12" |
| 19:27.40 | Katty | i'm gonna hang them from my deck :> |
| 19:27.54 | Qwell | Katty: still 2 holidays left |
| 19:28.19 | Katty | http://farm3.static.flickr.com/2576/4006638340_f8a6b82769_b.jpg <- squirrel feeder. |
| 19:29.10 | Katty | it's a little fuzzy. i took it from inside the house so i wouldn't disturb the critter. |
| 19:29.26 | Naikrovek | squirrels are underrated |
| 19:29.36 | Naikrovek | i kept several as part-time pets as a kid |
| 19:29.48 | Naikrovek | they were free to come and go as they pleased, but when they showed up i fed them |
| 19:30.06 | *** join/#asterisk errotan (n=errotan@5403E521.catv.pool.telekom.hu) |
| 19:30.23 | Katty | i' |
| 19:30.31 | Katty | i'd like to have a window feeder for them (= |
| 19:30.47 | Katty | and the birdies too, of course. |
| 19:31.01 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 19:31.04 | Katty | hi fender |
| 19:31.39 | [TK]D-Fender | Katty: Mew... |
| 19:31.57 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:33.42 | Katty | [TK]D-Fender: how're you dear |
| 19:34.01 | [TK]D-Fender | Katty: Getting by... wish it was Friday... |
| 19:34.20 | Qwell | [TK]D-Fender: bit early in the week to be wishing that |
| 19:34.47 | [TK]D-Fender | Qwell: Life sucks, but rarely swallows... |
| 19:35.10 | [TK]D-Fender | is happy about the latest wave of additions to his music book.. |
| 19:36.11 | Katty | hugs [TK]D-Fender |
| 19:37.34 | *** join/#asterisk jshriver (n=jshriver@cblmdm24-53-165-86.buckeyecom.net) |
| 19:37.35 | jshriver | greetings |
| 19:37.41 | jshriver | does asterisk keep an error log? |
| 19:38.02 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:40.09 | mbrevda | jshriver: depends on your setup; try /var/log/astrisk/full |
| 19:41.31 | mbrevda | Im having a problem with a gateway which is sending some funky sip stuff. Here is the log, I dont know exactly what going on, but basicly its send to caller id as the TO, which is being rejected, but then its trying again as something else |
| 19:41.32 | jshriver | ty |
| 19:41.33 | mbrevda | http://pastie.org/655028 |
| 19:42.20 | *** join/#asterisk tamiel (n=tamiel@abo-212-152-68.bdx.modulonet.fr) |
| 19:43.04 | *** join/#asterisk gardo (n=gardo@110.55.38.214) |
| 19:43.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:43.22 | *** part/#asterisk jshriver (n=jshriver@cblmdm24-53-165-86.buckeyecom.net) |
| 19:45.46 | [TK]D-Fender | mbrevda: Found peer '5003' for '6461112222' from 192.168.0.166:5060 <--- doesn't look like something good for it to land on. |
| 19:46.00 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:48.05 | mbrevda | [TK]D-Fender: what do you mean? |
| 19:48.37 | [TK]D-Fender | mbrevda: Should calls from that device match with 5003? |
| 19:50.14 | mbrevda | [TK]D-Fender: here's the thing: the device registers with three seperate numbers (for three different fxo lines). But technicaly, it shouldnt. The confusing part is that the callerid endes up being 5001 - not 5003 (and the problem is that it shouldnt be either - it should be 6461112222) |
| 19:50.36 | mbrevda | it shouldnt=match |
| 19:57.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:58.54 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 20:03.51 | muddd | Is the TDM800P the card that I am supposed to connect my analog phones to? I've been reading up on asterisk all day and trying to figure it out before I plan on using it. |
| 20:05.38 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 20:05.45 | *** join/#asterisk Khratos (n=khratos@190.166.103.151) |
| 20:07.42 | mbrevda | what does: 'No user '5551236049' in SIP users list' mean? |
| 20:07.51 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:08.48 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 20:09.12 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-26-97.dsl.lsan03.sbcglobal.net) |
| 20:10.20 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 20:16.19 | [TK]D-Fender | muddd: You already have this card? |
| 20:16.44 | *** join/#asterisk donnib (n=Mihai@0x555281d0.adsl.cybercity.dk) |
| 20:17.03 | donnib | can somebody help me with registering a sip device to my server ? |
| 20:17.19 | donnib | i keep getting 401 Unauthorized even though i know that the password is correct and the username |
| 20:17.30 | donnib | the account has been tested on xlite |
| 20:17.42 | donnib | i am using a Zyxel 2602 SIP adaptor |
| 20:19.12 | donnib | anyone ? |
| 20:20.40 | mchou | pastebin sip trace |
| 20:20.48 | muddd | [TK]D-Fender, no I haven't bought anything yet. I'm not educated on phone systems, just computer networking and such. I'm about to start a business and might want to save some money by doing the asterisk thing myself. |
| 20:21.22 | mchou | donnib: nobody is gonna be able to proceed without the pertinent info |
| 20:21.37 | [TK]D-Fender | muddd: What kind of lines & phones are you considering? |
| 20:22.48 | donnib | mchou: http://pastebin.com/d97762c4 |
| 20:23.17 | mchou | muddd: yeah, inquiring minds want to know |
| 20:23.29 | muddd | [TK]D-Fender, I'm trying to keep costs down so I guess analog phones, using comcast's digital voice as my provider |
| 20:23.47 | mchou | umm |
| 20:24.06 | [TK]D-Fender | muddd: And how do you connect to comcast's service? |
| 20:24.11 | muddd | unless you guys have much better suggestions. |
| 20:25.35 | donnib | so any ideas ? |
| 20:25.35 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 20:25.49 | ayeso | Trying to decide if I should use the manager API or .call files for outbound auto dialing... any advice? |
| 20:27.39 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:27.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:29.56 | lesouvage | donnib: I assume that the Asterisk server and the device are on the same lan? |
| 20:30.17 | donnib | yes |
| 20:30.23 | lesouvage | donnib: are you sure you don't have a space in front or at the and of the secret? |
| 20:30.27 | donnib | nope |
| 20:30.48 | donnib | have typed the password many times |
| 20:30.49 | donnib | i have tried copy paste |
| 20:30.53 | lesouvage | donnib: is it a softphone? |
| 20:30.59 | donnib | no |
| 20:31.06 | donnib | it´s a hardware router/SIP device |
| 20:32.16 | kaldemar | the device isn't even trying to authenticate |
| 20:32.35 | lesouvage | donnib: and the IP of the device is 192.168.1.1 ? (see the contact field) |
| 20:32.45 | donnib | yes |
| 20:32.57 | donnib | it´s my main router and SIP device |
| 20:33.04 | donnib | and the server is 192.168.1.10 |
| 20:34.18 | donnib | it's really weird |
| 20:34.55 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 20:34.55 | *** join/#asterisk dwayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk RypPn (i=TuMbL@188-220-36-226.zone11.bethere.co.uk) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) [NETSPLIT VICTIM] |
| 20:34.55 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) [NETSPLIT VICTIM] |
| 20:34.56 | *** join/#asterisk errotan (n=errotan@5403E521.catv.pool.telekom.hu) [NETSPLIT VICTIM] |
| 20:34.56 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) [NETSPLIT VICTIM] |
| 20:35.50 | lesouvage | WWW-Authenticate: Digest algorithm=MD5 is the MD5 stuff put in place ? |
| 20:36.03 | *** join/#asterisk w-m- (i=w@freenode/staff/wikipedia.wimt) |
| 20:36.17 | *** join/#asterisk muddd (n=fubard@c-98-224-33-102.hsd1.fl.comcast.net) |
| 20:36.29 | *** join/#asterisk voxter (n=voxter@166.183.62.157) |
| 20:36.31 | donnib | lesouvage: not sure i can follow u here |
| 20:36.40 | donnib | lesouvage: what do you mean ? |
| 20:36.44 | muddd | whew. well i'm sure what i was saying last is totally lost |
| 20:37.12 | muddd | dangit |
| 20:37.34 | mchou | muddd: yup, lost in ether |
| 20:38.02 | donnib | lesouvage: what do you mean by MD5 has ? do i have to do something ? |
| 20:38.16 | lesouvage | donnib: wait a moment please |
| 20:38.21 | donnib | oki |
| 20:38.50 | muddd | I want to learn about office phone systems, can anyone recommend a book or a readme/howto/tutorial on the internet? I'm thinking of starting a biz and would like to be cheap and do the office phone system myself. |
| 20:39.22 | *** join/#asterisk Defraz (n=tim@99-204-167-244.pools.spcsdns.net) |
| 20:40.08 | Joel | ~book |
| 20:40.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:41.05 | mchou | there's no way I'd recommend that book |
| 20:41.26 | Joel | yeah, it does require technical knowledge. |
| 20:41.45 | mchou | no, tech knowledge has nothing to do with it |
| 20:41.47 | muddd | I've been reading about asterisk all day... but there are some things I am unclear on |
| 20:42.16 | muddd | I am a computer tech though, not a phone guy :( |
| 20:42.39 | mchou | it's just not suitable for someone who wants to understand deployment options and scenarios |
| 20:43.04 | mchou | that book is way down in the weeds |
| 20:43.11 | muddd | I'll explain my situation: |
| 20:43.22 | lesouvage | donnib: if you go to sip.conf and change md5 in Digest and do a reload does it still not register. (be aware that the secret is now passing the network without any security) |
| 20:43.38 | mchou | when say 10k ft. level overvies would be more suitable |
| 20:43.46 | donnib | ok, let me try |
| 20:44.32 | mchou | overview* |
| 20:44.55 | muddd | starting a biz, 2 people at first, up to 8 max, want to use asterisk, wondering about the TDM800P (digium) as the "output" (fxs?) to some analog phones in the office |
| 20:45.27 | Qwell | muddd: fxs for phones, yes. |
| 20:45.38 | mchou | muddd: one phone # to the whole org or everyone has individual phone #? |
| 20:46.10 | muddd | from what I gather, everyone phone will have to have its own phone number, for outbound calls, but an 800 number for all inbound |
| 20:46.23 | mchou | ok |
| 20:46.26 | voxter | Qwell: you arent here this year, boo! |
| 20:46.40 | Qwell | voxter: I'm boycotting due to lack of tater tots, apparently |
| 20:46.49 | Qwell | s/, apparently//, because that's totally the reason |
| 20:47.02 | muddd | I read about using skype and asterisk, but would I be able to use analog phones with that? |
| 20:47.04 | mchou | muddd: put skype out of your mind |
| 20:47.05 | voxter | god damnit, now i want tater tots |
| 20:47.10 | muddd | I'm not sure how Comcast business phones (digital voice, they say its not voip) connects |
| 20:47.12 | voxter | although i learned that there is an IN n out near by |
| 20:47.15 | voxter | <3 |
| 20:47.18 | Qwell | muddd: yes |
| 20:47.18 | muddd | ok thank you about skype |
| 20:47.29 | Qwell | muddd: ignore what he said about putting skype out of your mind. |
| 20:47.34 | muddd | lol |
| 20:47.35 | Qwell | skype works fine with Asterisk now. |
| 20:47.35 | muddd | ok |
| 20:47.37 | Qwell | ~skypeforasterisk |
| 20:47.38 | infobot | skypeforasterisk is, like, a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 20:47.45 | donnib | lesouvage: any chance u know where the digest setting is in freepbx ? i know it's a long shot but anyway.... |
| 20:48.24 | Qwell | voxter: bleh |
| 20:48.57 | Qwell | I wonder if they have in n out in vegas... |
| 20:49.26 | Joel | Qwell, yes |
| 20:49.28 | Qwell | sweet |
| 20:49.34 | voxter | they do yes |
| 20:49.38 | Qwell | makes reservations for April |
| 20:49.43 | *** join/#asterisk wcselby (n=wcselby@204.15.220.206) |
| 20:49.54 | muddd | Will the people on analog phones be able to transfer their calls to other people in the office? |
| 20:49.58 | Qwell | Sahara better be close to the strip |
| 20:50.07 | mchou | muddd: yup |
| 20:50.10 | muddd | If there is a book I can buy to answer all my questions and teach me more I would gladly buy it. |
| 20:50.17 | muddd | and not pester you people :D |
| 20:50.22 | muddd | ty mchou |
| 20:50.35 | wcselby | mudd |
| 20:50.38 | wcselby | ~book |
| 20:50.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:50.45 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:50.59 | mchou | wcselby: dude. sombeody pionted to the book already |
| 20:51.05 | mchou | pointed* |
| 20:51.10 | muddd | mchou wouldn't recommend that |
| 20:51.11 | wcselby | :) sorry I just joined the chan |
| 20:51.18 | Qwell | muddd: everybody else in here would :) |
| 20:51.22 | Joel | muddd, this is covered in the book I just dropped the url for, shrug |
| 20:51.36 | Joel | muddd, so? mchou doesn't know what he's talking about |
| 20:51.41 | muddd | I'm willing to buy more than one book lol |
| 20:51.45 | lesouvage | donnib: I'm not familiar with freepbx. If you go to the extesion in the webinterface there must be an option to set the kid of authentication and for testing you could try it with an option that might me something like "plain text" |
| 20:51.47 | Joel | this book is free |
| 20:51.52 | Joel | I'm not sure why you are scared of it |
| 20:52.07 | wcselby | lesouvage: try asking in #freepbx |
| 20:52.07 | mchou | there's nothing to be scared of |
| 20:52.15 | Joel | Qwell, the in'n'out is 'glitzy' too, which makes it that much better |
| 20:52.17 | donnib | thx will try |
| 20:52.22 | wcselby | ~freepbx |
| 20:52.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:52.32 | Qwell | Joel: walking distance from the strip? |
| 20:52.38 | Joel | Qwell, It may be sacrilegious of me but I've only had in'n'out once since being back, and that was more then enough. |
| 20:53.04 | drmessano | The book is kinda crappy.. It has not one chapter on Bicycle Repair.. All ASTERISK, ASTERISK, ASTERISK.. |
| 20:53.05 | muddd | the pdf and html links are dead |
| 20:53.12 | lesouvage | wcselby: I'm trying to help donnib |
| 20:53.15 | drmessano | Click the book |
| 20:53.36 | donnib | wcselby: i know. sorry |
| 20:53.37 | Joel | Qwell, yes |
| 20:53.39 | wcselby | lesouvage: sorry |
| 20:53.43 | Joel | Qwell, other side of the from nyny |
| 20:53.46 | Joel | 15 |
| 20:53.54 | lesouvage | wcselby: don't mind ;-) |
| 20:53.55 | Joel | four blocks? five? |
| 20:54.06 | Qwell | vegas blocks. so like 12 miles |
| 20:54.18 | Joel | nah, it's not that bad |
| 20:54.23 | drmessano | If you click the PDF link on the second URL, you need to WAIT for it to DOWNLOAD.. it is a BOOK, afterall |
| 20:54.26 | Joel | sides, cabs are everywhere |
| 20:54.43 | Qwell | Joel: not gonna spend $20 on a ...nevermind. yes I would |
| 20:54.53 | drmessano | Not on a sloppy one, anyway |
| 20:55.03 | muddd | ah |
| 20:55.11 | Joel | Qwell, it'd be like $5 cab ride from nyny max |
| 20:55.28 | muddd | oh hell its working now, sorry |
| 20:55.41 | *** join/#asterisk jayrod422 (n=jarrod@204.15.220.206) |
| 20:55.48 | drmessano | You're forgiven, newb |
| 20:56.00 | muddd | ty |
| 20:56.14 | Joel | what's in vegas in the spring qwell? |
| 20:57.00 | Joel | they asked me if I wanted to hit whatever's going on now |
| 20:57.05 | Joel | but I said seen one, seen em all. |
| 20:57.36 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:57.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:58.24 | DavidR2008 | does anyone know where astricon 2010 is going to be held? I can't find any info in the intarwebs |
| 21:00.46 | leifmadsen | it probably haven't been decided yet |
| 21:02.46 | drmessano | Augusta, GA <-- My vote |
| 21:02.51 | drmessano | At the James Brown Arena |
| 21:03.01 | wcselby | lol leif[astricon] - are you in a session now or are you outside somewhere? |
| 21:03.07 | drmessano | Seats 4k, should be enuff room for us nerds |
| 21:03.39 | drmessano | Except maybe the portly ones, but there's plenty of "tailgate" space |
| 21:04.10 | *** part/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 21:04.11 | leif[astricon] | wcselby: I'm in the codezone |
| 21:04.20 | *** join/#asterisk CrazyTux[w] (n=Administ@216-110-94-230.static.twtelecom.net) |
| 21:04.33 | leif[astricon] | I did my talk this morning |
| 21:10.03 | wcselby | leif[astricon]: i was there :) |
| 21:10.37 | leif[astricon] | nice` |
| 21:11.01 | leif[astricon] | ! |
| 21:11.01 | leif[astricon] | I have moved to a couch in the code zone, and am attempting to get some doc stuff in order |
| 21:12.20 | wcselby | cool, I'm in the Xen session - it's really good so far |
| 21:13.33 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:15.29 | *** join/#asterisk RypPn (i=TuMbL@188-220-36-226.zone11.bethere.co.uk) |
| 21:16.45 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 21:19.40 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:19.58 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-71-143.client.mchsi.com) |
| 21:23.18 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 21:34.14 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 21:41.51 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) |
| 21:48.01 | atz | i've been trying all day to get ODBC working in asterisk... have my odbc.ini and odbcinst.ini files setup correctly, i think |
| 21:48.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:48.09 | atz | echo "select 1" | isql -v asterisk-connector # that works |
| 21:48.29 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 21:49.07 | CrazyTux[w] | Say I want to send a call to a remote PSTN destination Dial(SIP/XXXX) upon call "ANSWER" I wish to play a message such as "You've received a call from XYZ, press 1 to accept" upon pressing 1 I want the call to "bridge" and connect |
| 21:49.17 | CrazyTux[w] | Is there any information online as far as completing this process and or some guidance |
| 21:49.24 | atz | but inside asterisk command-line, odbc show still comes back empty. i've been going off the asterisk (starfish) book, and echo "select 1" | isql -v asterisk-connector |
| 21:49.42 | atz | sorry... book and http://climbing-the-hill.blogspot.com/2008/04/asterisk-realtime-architecture.html |
| 21:55.48 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:56.58 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 22:17.35 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 22:21.28 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 22:29.04 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 22:29.49 | *** join/#asterisk leif[astricon] (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:29.49 | *** mode/#asterisk [+o leif[astricon]] by ChanServ |
| 22:37.36 | *** join/#asterisk voxter (n=voxter@166.183.74.98) |
| 22:40.23 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 22:43.42 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 22:48.11 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 22:49.59 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:50.00 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:52.26 | CcRnp | can anyone help me ! i want to use video on asterisk and it seems like i dont have H.263 codec install on my system |
| 22:52.34 | CcRnp | where can i download this codec |
| 22:53.44 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 22:56.59 | Katty | hi :> |
| 22:57.33 | *** join/#asterisk AeroCloud (n=aero@ip98-165-113-125.ph.ph.cox.net) |
| 22:59.02 | CcRnp | anyone here can help me with asterisk with video ? |
| 22:59.35 | raden_work | hi Katty |
| 23:00.17 | Katty | herro raden |
| 23:00.54 | raden_work | awe i your herro :) |
| 23:01.11 | *** join/#asterisk tuxcrafter (n=jelle@84-245-3-195.dsl.cambrium.nl) |
| 23:01.14 | tuxcrafter | hi all |
| 23:02.06 | tuxcrafter | i am on the asterisk web gui and i am searching for the option to forward all calls for number xxx xxxx xxxx to external number xxx xxxx xxxx like with *21* |
| 23:02.19 | *** join/#asterisk CcRNP (n=shishir@208.179.165.18) |
| 23:02.22 | tuxcrafter | where can i do this or find documentation/ |
| 23:02.43 | raden_work | tuxcrafter, come back when [TK]-fender around hes a expert at forwarding :) |
| 23:03.07 | CcRNP | asterisk video support please |
| 23:03.21 | raden_work | Dial 4 |
| 23:04.31 | CcRNP | how to enable video support in asterisk ? |
| 23:05.01 | raden_work | http://www.voip-info.org/wiki/view/Asterisk+video |
| 23:05.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:06.05 | tuxcrafter | raden_work: ok :D is it that hard to do then ? or should i make some speical incomming call rules? |
| 23:06.22 | tuxcrafter | raden_work: i just feel stupid i cant find it in the webgui |
| 23:06.41 | raden_work | astDB |
| 23:07.00 | CcRNP | i went through http://www.voip-info.org/wiki/view/Asterisk+video already, i am concern about codec , do i need to install the h.263 codec on my system or just enabling videosupport=yes in sip.conf will enable the video support |
| 23:11.53 | *** part/#asterisk DMeloUK (n=Dominic_@64.129.94.138) |
| 23:15.30 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:15.41 | *** join/#asterisk Chesther (n=cam2@63-255-103-4.ip.mcleodusa.net) |
| 23:19.30 | *** join/#asterisk voxter (n=voxter@32.176.226.95) |
| 23:31.48 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 23:34.42 | p3nguin | What is the significance of the 101 priority? |
| 23:52.14 | *** join/#asterisk voxter (n=voxter@166.128.186.193) |
| 23:53.50 | Joel | p3nguin, failure. |
| 23:56.01 | p3nguin | I've seen reference to n+101 before, too. Does that mean I need to calculate the value of the last priority, then add 101 to it when writing my dialplan? |
| 23:56.25 | Joel | it used to be that a failure would jump 101 priorities. |
| 23:56.41 | Joel | you shouldn't be using priorities at all now |
| 23:56.58 | Joel | I believe labels are the preferred way to go. |
| 23:58.23 | p3nguin | hmm |
