| 00:00.36 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
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| 00:16.21 | rps2 | Greetings, once again. |
| 00:19.37 | *** part/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 00:19.49 | rps2 | Thanks to all of the fine folk here, I've got most of this beast working.....except incoming calls aren't answered. I don't see a ring indication in the logs either. |
| 00:22.31 | superbeef | rps2: ringing in via T1 or POTS? |
| 00:22.36 | rps2 | POTS. |
| 00:22.58 | rps2 | I do see "[Oct 8 16:56:30] WARNING[2391] pbx.c: Unable to register extension 's', priority 1 in 'DID_trunk_1_default', already in use" in the logs. |
| 00:23.16 | superbeef | hmm |
| 00:23.24 | superbeef | well i have setup a POTS yet, only t1, but i have to setup pots soon |
| 00:23.28 | superbeef | using DAHDI? |
| 00:23.32 | rps2 | Yup. |
| 00:23.53 | rps2 | Outgoing works a treat. Incoming just rings and rings. I have it set to send it to my SIP extension. |
| 00:23.54 | superbeef | you have your dahdi config posted anywyhere? |
| 00:24.10 | rps2 | Nope, but it's pretty stock. |
| 00:24.23 | superbeef | wanna throw it on pastebin real quick |
| 00:24.43 | rps2 | Sure. |
| 00:24.45 | superbeef | and your extensons.conf file that has that part of your dialplan |
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| 00:26.02 | rps2 | Dahdi config is at http://pastebin.ca/1605648 |
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| 00:27.09 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 00:27.25 | rps2 | extensions.conf is at http://pastebin.ca/1605650 |
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| 00:28.32 | superbeef | rps2 : what about chan_dahdi |
| 00:28.43 | superbeef | why 1-240 on line 18 |
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| 00:30.34 | rps2 | I don't know about the "1-240" bit. |
| 00:31.05 | rps2 | I think that's what one of the dahdi tools did. |
| 00:33.36 | rps2 | chan_dahdi is at http://pastebin.ca/1605658 |
| 00:36.00 | me|ong | wow |
| 00:36.06 | me|ong | i was about to ask the DUMBEST question |
| 00:36.17 | me|ong | till i realised i forgot to install build-essential package |
| 00:36.18 | me|ong | :S |
| 00:36.27 | rps2 | No, I have the corner on that market today, melong |
| 00:36.28 | *** join/#asterisk Shinsaku (n=Shinsaku@chello089076140236.chello.pl) |
| 00:36.32 | me|ong | baw nab' |
| 00:36.35 | me|ong | man* |
| 00:36.44 | me|ong | lol i couldn't figure out why ./configure wouldn't work |
| 00:36.50 | me|ong | had another box.. exact same build.. worked |
| 00:37.06 | me|ong | had it all typed.. just before hitting enter.. *light turned on* |
| 00:37.48 | me|ong | though i do have a small Q, anybody work with integrating this with a cisco 2600 series router? |
| 00:37.49 | me|ong | :) |
| 00:38.05 | me|ong | i'm going to tackle it but wouldn't mind knowing if anybody has done it yet |
| 00:43.11 | rps2 | No ideas, superbeef? |
| 00:43.19 | Sandheaver | me|ong: you can ask in #cisco |
| 00:43.41 | Sandheaver | oh, you're talking about allowing SIP ports through |
| 00:43.50 | superbeef | rps2: sorry spaced out hold on |
| 00:44.27 | me|ong | well.. kind of |
| 00:44.27 | me|ong | i want to run the cisco as the gateway.. and asterisk as the call manager |
| 00:44.44 | rps2 | superbeef: No worries. Just very confused. |
| 00:44.54 | me|ong | i'm experienced with networking etc.. but voice is new to me.. so i'm trying to learn ;) |
| 00:44.56 | *** join/#asterisk |Rain| (n=rain@ev.il.net) |
| 00:45.20 | superbeef | rps2, you have 3 FXS defined.. no FXO's for dialout |
| 00:45.23 | *** part/#asterisk |Rain| (n=rain@ev.il.net) |
| 00:45.24 | Sandheaver | me|ong: well look up what ports SIP and/or IAX2 uses and that's all you'll need to know |
| 00:45.26 | me|ong | i've read a bit on google, and after i get this server up I'll be trying to integrate |
| 00:45.50 | superbeef | and line 18 is saying enable echo cancelation for channels 1-240... and you probably have more like channels 1-3 |
| 00:46.42 | rps2 | Uh, dialout works fine. Those are FXO ports, which terminate the POTS lines. Should be good for incoming and outgoing. |
| 00:47.08 | rps2 | I'm trying to call into the system from a cell phone and have it answer. |
| 00:48.36 | superbeef | rps2: http://docs.tzafrir.org.il/dahdi-tools/ should they be fxoks instead of fxsks? |
| 00:49.50 | rps2 | No, they're FXO ports, but to terminate the central office lines, they use FXS signaling. |
| 00:50.01 | rps2 | (they have to simulate an analog phone). |
| 00:50.06 | rps2 | I know, it's confusing. |
| 00:50.54 | me|ong | that is... |
| 00:51.18 | me|ong | fxo is the in/out line.. but you're saying you're trying to set them up as fxs? |
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| 00:56.00 | superbeef | rps2: you're right, i got my wires crossed.. long day |
| 00:56.13 | rps2 | Whew! |
| 00:56.17 | superbeef | you should change line 18 to be 1-3 instead of 1-240 |
| 00:56.59 | superbeef | rsp2: i'm reconfused again... okay... so you have 3 POTS going into that box |
| 00:57.35 | rps2 | Well, three FXO ports, but only two lines right now. |
| 00:57.43 | rps2 | ports 1 and 2 |
| 00:57.57 | rps2 | Port 3 has nothing plugged in. |
| 00:57.59 | superbeef | this is in your dahdi config |
| 00:58.00 | superbeef | /home/vmware/RH71_D3_Server_5 |
| 00:58.02 | superbeef | no thats not |
| 00:58.07 | superbeef | this is |
| 00:58.08 | superbeef | fxsks =Â 1,2,3 |
| 00:58.47 | rps2 | Right. |
| 00:59.08 | rps2 | There are three ports, but only two are in use. |
| 00:59.39 | *** join/#asterisk Get_The_Fish (n=IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 00:59.57 | superbeef | so an FXO port is configured with fxsks and not fxoks? |
| 01:00.18 | rps2 | Yes, if you want it to simulate an analog phone for the phone company. |
| 01:00.48 | me|ong | so to receive a dial tone |
| 01:00.58 | *** join/#asterisk geneticx (n=geneticx@adsl-10-114-216.mia.bellsouth.net) |
| 01:01.36 | rps2 | Yup. You use an FXO port to connect to the central office and use fxs to receive a dial tone. |
| 01:02.00 | Get_The_Fish | hello all, I'm trying to use mixmonitor, so the user can dial *1 and start the monitor mid-call, but I cant find an example of the best way to write this into a dialplan...can someone give me a nudge/hint/link... |
| 01:02.38 | me|ong | you should be able to receive a dialtone over fxo though. |
| 01:02.57 | me|ong | or simulate it internally at least and still force calls out fxo |
| 01:03.22 | rps2 | You do, but the fxs signaling on the fxo port tells the phone company you're "acting" like a phone, not a central office. |
| 01:03.49 | me|ong | hmm |
| 01:04.12 | superbeef | ~fxsks |
| 01:04.24 | superbeef | denied by the chanbot |
| 01:04.26 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 01:04.31 | me|ong | lol |
| 01:04.31 | rps2 | http://pastebin.ca/1605710 |
| 01:04.47 | superbeef | http://docs.tzafrir.org.il/dahdi-tools/ section 2.4.3 |
| 01:05.02 | superbeef | fxsksChannel(s) are signalled using FXS Koolstart protocol. |
| 01:05.48 | rps2 | Read the pastebin. |
| 01:06.08 | superbeef | k reading |
| 01:06.47 | superbeef | okay i'm sold |
| 01:06.48 | superbeef | Ports are defined in the configuration by the signaling they use, as opposed to thephysical type of port they are. For instance, a physical FXO port will be defined in theconfiguration with FXS signaling, and an FXS port will be defined with FXO signaling.This can be confusing until you understand the reasons for it. |
| 01:06.50 | me|ong | yeah that's what i though. |
| 01:06.55 | me|ong | thought* |
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| 01:06.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:06.58 | superbeef | it still hurts your brain tho |
| 01:06.58 | rps2 | Yup. |
| 01:06.59 | superbeef | okay |
| 01:07.05 | rps2 | Uh, yeah, sorta. |
| 01:07.28 | superbeef | rps2: you need to change line 18 to echocanceler = mg2,1-2 |
| 01:07.37 | superbeef | so that it only tries to use 2 channels.. specifically the ones that are patched |
| 01:07.47 | rps2 | That'll screw it up? |
| 01:08.26 | me|ong | wait a min |
| 01:08.31 | me|ong | MGCP is supported by asterisk? |
| 01:08.41 | me|ong | ponders |
| 01:08.45 | superbeef | honestly i'm not sure how tolerant it is, but it's certainly referencing channels that don't exist |
| 01:08.47 | carrar | yeah |
| 01:09.10 | rps2 | Ok, I changed it. |
| 01:10.56 | superbeef | rps2: and now... everything magically.... the same probably |
| 01:11.16 | rps2 | Let me try. |
| 01:11.33 | *** join/#asterisk Shinsaku (n=Shinsaku@chello089076140236.chello.pl) |
| 01:11.38 | Get_The_Fish | (sorry to ask again) I'm trying to use mixmonitor, so the user can dial *1 and start the monitor mid-call, but I cant find an example of the best way to write this into a dialplan...can someone give me a nudge/hint/link... |
| 01:12.36 | rps2 | superbeef: Yup, still doesn't answer. I do see "[Oct 8 18:11:26] WARNING[4475] app_setcallerid.c: SetCallerPres is deprecated. Please use Set(CALLERPRES()=unavailable) instead." in the log when I call in, but that's it. |
| 01:13.13 | superbeef | rps2: anything you can do to enable for debugging? can you poste your dialplan |
| 01:13.15 | russellb | Get_The_Fish: see the 'x' and 'X' options to Dial() |
| 01:14.47 | rps2 | The dialplan is pretty gnarly. Actually, I have to chuck it in for the night and have a go at it again tomorrow. |
| 01:14.52 | me|ong | yuss |
| 01:14.54 | me|ong | up and running |
| 01:14.55 | Get_The_Fish | The man himself... thanks Russell |
| 01:15.12 | me|ong | :) time to configure... |
| 01:15.20 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 01:15.30 | me|ong | now does asterisk need to be configured with config files.. or can you use it like an IOS from network equipment? |
| 01:15.30 | russellb | Get_The_Fish: you're welcome, sir |
| 01:16.01 | russellb | me|ong: config files. you can do only very minor config from the CLI. |
| 01:16.13 | me|ong | awww shucks. that would have been cool :) |
| 01:16.24 | rps2 | Thanks for the help, chaps. I'll be back on it tomorrow. Right now, I have to take the lady out for her birthday. |
| 01:16.28 | superbeef | russellb: what's asterisk realtime for? |
| 01:16.38 | superbeef | rps2: sounds like more fun... good luck |
| 01:16.49 | rps2 | Thanks. Chat at you tomorrow. |
| 01:16.52 | russellb | superbeef: it's an interface for using configuration backends other than config files - databases primarily |
| 01:17.42 | superbeef | russellb: cool.... does asteriskGUI use it? My only experience is with freepbx which only feeds asterisk config files |
| 01:17.44 | *** join/#asterisk coppice (n=chatzill@148.162.17.210.dyn.pacific.net.hk) |
| 01:18.20 | Get_The_Fish | uh, Russell, did you per chance mean wW |
| 01:18.21 | russellb | superbeef: AsteriskGUI doesn't use it, either. That GUI modifies config files directly via the asterisk manager interface. |
| 01:18.28 | russellb | Get_The_Fish: it's possible. :-) |
| 01:18.42 | Get_The_Fish | yeah ok, that was it... had me confused for a sec :) |
| 01:18.58 | russellb | sorry ... i thought I looked, guess i can't read |
| 01:19.43 | Get_The_Fish | no worries, no worries, just saved me a headache so I'm thrilled... I forgot about that option was chasing my tail down the wrong rabbit hole :) |
| 01:20.02 | russellb | no prob |
| 01:27.56 | superbeef | is dahdi all you need ot configure a digium FXS card? or are there another level of drivers i need (i have a Digium Wildcare AEX800) |
| 01:28.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 01:35.03 | russellb | superbeef: just dahdi and asterisk. |
| 01:35.57 | superbeef | awesomeular |
| 01:37.46 | superbeef | http://www.youtube.com/watch?v=XDlg2JZXp4A&feature=channel |
| 01:37.52 | superbeef | sorry wrong window |
| 01:37.52 | superbeef | lol |
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| 02:20.43 | *** join/#asterisk prakash (n=prakash@207.101.224.194) |
| 02:21.31 | prakash | hi guys. where can I find a list of some big companies that use asterisk? |
| 02:22.18 | prakash | Our IT dept wants to use asterisk, but sales and finance are against it. |
| 02:23.13 | prakash | some big companies that use asterisk can convince them. |
| 02:23.19 | superbeef | lol |
| 02:23.46 | ChannelZ | Are they against it because they have no money for it or because they think it's crap? |
| 02:24.06 | prakash | we told them that we can get uspport form digium. |
| 02:24.45 | prakash | they have money. they are afraid that it will not be as stable as Cisco's of the world |
| 02:25.08 | superbeef | prakash: how big of a deployment |
| 02:25.09 | prakash | and that we will not have support if case of unknown problems. |
| 02:25.25 | prakash | we have 120 employees in the company. |
| 02:25.38 | superbeef | how many branches |
| 02:25.48 | prakash | about 40 of them in sales |
| 02:25.58 | superbeef | 40 branches,or extensions |
| 02:26.09 | prakash | we have 2 branches. 1 in CA and another in Idaho |
| 02:26.45 | superbeef | totally asteriskable |
| 02:26.47 | superbeef | http://www.voip-news.com/whitepaper/asterisk/ |
| 02:27.26 | prakash | thank you |
| 02:27.29 | superbeef | not exactly what you're looking for, but a start |
| 02:32.42 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
| 02:33.57 | prakash | it will enough if I can get digium's customers. I will try calling them tomorrow. |
| 02:34.08 | prakash | thanks for your help |
| 02:34.15 | superbeef | np good luck |
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| 03:00.33 | *** join/#asterisk digitalirony (n=digitali@shellium/member/digitalirony) |
| 03:01.15 | digitalirony | Looking for some info on how to convert time into unixtime, but inside of asterisk |
| 03:02.51 | p3nguin | ${STRFTIME(${EPOCH}||%Y%m%d-%:%M:%S)} gives me the current time, so I imagine you can play with that and get regular unix time. |
| 03:03.46 | p3nguin | I also use the "talking clock" by using this command SayUnixTime(||ABdY \'digits/at\' IMp) |
| 03:03.53 | digitalirony | well, the time that needs to be played isn't always going to be the current time |
| 03:04.02 | digitalirony | Yeah, thats what I am using |
| 03:04.12 | p3nguin | What time are you wanting to play? |
| 03:04.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:04.32 | p3nguin | Is there always going to be the same offset? |
| 03:04.41 | digitalirony | Well, basically call up 'clients' and tell them their appointment is at <time> |
| 03:05.15 | digitalirony | Where ASTTIME will hold a date and time like this "2009-09-18 18:5" |
| 03:09.56 | *** join/#asterisk tipsc (n=tipsc@adsl-074-186-104-155.sip.mia.bellsouth.net) |
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| 03:15.25 | digitalirony | what about the $SHELL |
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| 03:19.10 | jblack | So, I have this new android phone, which means t-mobile, which means fav-5. |
| 03:19.36 | jblack | Hqas anybody come up with an interesting way to take advantage of that? |
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| 03:58.30 | Kobaz | Octothorpe: # |
| 03:58.30 | jaytee | his friends just call him pound |
| 03:58.31 | Kobaz | heh |
| 03:58.31 | drmessano | has angered a Ubuntu |
| 03:58.31 | jaytee | oh noes! |
| 03:58.31 | superbeef | it can do that on occasion |
| 03:58.32 | drmessano | I'm trying out 9.10 LTS.. My wife wants to make the switch |
| 03:58.32 | drmessano | OH NOES |
| 03:58.32 | drmessano | Something about read errors and squashing an FS |
| 03:58.32 | drmessano | :( |
| 03:58.34 | ian6 | ... your wife is trying to convert you to linux? |
| 03:58.34 | ian6 | how does it feel to have her wearing the pants. |
| 03:58.34 | drmessano | No |
| 03:58.35 | carrar | no shit |
| 03:58.36 | drmessano | My wife is sick of Windows.. she wants a MAC, but has been happy with her Ubuntu trial |
| 03:59.45 | drmessano | Actually, she is sick of all the "greyware" going around |
| 03:59.46 | jaytee | give her Ubuntu with Compiz fully enabled and install Avant Window Navigator so she has a dockbar like the Mac. Then go to gnomelook.org and download the MacOSX icon set. |
| 03:59.58 | jaytee | problem solved |
| 04:00.18 | carrar | or just install OSX on your PC |
| 04:00.23 | drmessano | She actually likes the orange Ubuntuness better |
| 04:00.37 | drmessano | I wont have OSX under my roof |
| 04:00.44 | carrar | too bad |
| 04:00.45 | carrar | it's nice |
| 04:00.55 | drmessano | Not impressed |
| 04:01.18 | drmessano | Its fluffy *nix.. Can do that by picking something with a penguin on it too |
| 04:01.23 | carrar | it's a desktop |
| 04:02.06 | drmessano | I know what it is |
| 04:02.44 | drmessano | Damnit, i think I just burned a coaster |
| 04:05.35 | drmessano | I was running Centos 5.3 on my laptop.. and what I expected to happen just did.. I havent used my laptop in a month, ran an update, and the atheros card isn't working. I can't remember the hacky thing I needed to remember when getting a new kernel |
| 04:05.47 | drmessano | SAD FACE.. AKA Installing something else |
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| 04:26.21 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 04:29.04 | kimo_sabe | hey, is anyone on a digial line willing to do me a favor? I've getting some clicks and I'd like to get recorded to shove at the telco |
| 04:30.17 | superbeef | you just need some hold music or something? |
| 04:30.35 | kimo_sabe | preferably just silence |
| 04:30.55 | superbeef | how long |
| 04:31.01 | kimo_sabe | and a recording of the "silence" I'll be sending afterwards. |
| 04:31.26 | kimo_sabe | a few minutes would be plenty. |
| 04:31.43 | kimo_sabe | The clicks are on the remote side so I can't record it myself |
| 04:32.02 | superbeef | oh |
| 04:32.26 | superbeef | sorry short on tricks for that |
| 04:34.22 | kimo_sabe | just Record(/tmp/tw,ulaw,,300) off a T1 is all I'm after |
| 04:35.00 | superbeef | okay |
| 04:35.07 | superbeef | can i do the dial and the record command from the CLI |
| 04:38.51 | superbeef | kimo_sabe: my brain started working |
| 04:38.54 | superbeef | give me the # |
| 04:40.09 | kimo_sabe | I'm setting up a record on my side too, lemmi make sure I have this DID working |
| 04:44.57 | kimo_sabe | superbeef: 520-618-5423 |
| 04:45.21 | kimo_sabe | superbeef: that did will drop right into Record for 300s then die. |
| 04:46.35 | superbeef | k dialed |
| 04:47.05 | superbeef | yup static |
| 04:47.06 | superbeef | lol |
| 04:47.09 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 04:54.27 | kimo_sabe | bah, it overrote that recording on me |
| 04:54.41 | *** join/#asterisk lucasb (n=bussey@154.5.252.231) |
| 04:54.46 | kimo_sabe | can I get you to email me that recording? |
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| 04:55.45 | superbeef | yeah 1 sec |
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| 05:09.47 | kimo_sabe | superbeef: got it, thanks |
| 05:10.49 | superbeef | killer |
| 05:10.55 | Get_The_Fish | so, perhaps this is a stupid question, but when I use the originate AMI command, the callerid that I specify in the command is the one that shows on the phone's display on the first leg of the call, the "internal" phone... any way to change that, or am I doing something wrong here? |
| 05:11.15 | *** join/#asterisk bimbo (n=oso@200.92.164.140) |
| 05:14.59 | bimbo | hello, yesterday I was testing asterisk over the internet, this is more related to SIP than to asterisk but the other end was able to hear me while I wasn't, I'm not sure if the other end had a firewall or not (I had mine disabled), we had to use a STUN server and it worked only sometimes, anybody has an idea why this happened? |
| 05:15.14 | Get_The_Fish | NAT |
| 05:15.30 | carrar | ~sipnat |
| 05:15.31 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 05:28.23 | Get_The_Fish | not trying to spam the channel, but when I use the originate AMI command, the callerid that I specify in the command is the one that shows on the phone's display on the first leg of the call, the "internal" phone... any way to change that, or am I doing something wrong here? |
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| 06:33.11 | kaldemar | Get_The_Fish: how do you want it to work? that's what the definition is for. |
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| 06:39.43 | Get_The_Fish | kaldemar, I found some forum posts that shed some light into this some more. Apparently I am not the only one that has had this question/issue, and the answer is thats just the way it is |
| 06:40.55 | Get_The_Fish | http://lists.digium.com/pipermail/asterisk-dev/2008-November/035470.html |
| 06:41.10 | Get_The_Fish | http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg37098.html |
| 06:42.12 | Get_The_Fish | these two posters are explaining what I was asking better than I have, and it's been tabled already at least once by digium |
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| 07:29.27 | raden | morning |
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| 07:39.34 | ChannelZ | wanders off to bed |
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| 08:43.12 | MWE | hi there :) |
| 08:44.01 | MWE | Is it possible to track the channel which a call file will use or uses ? |
| 08:50.39 | kaldemar | what do you mean by tracking? |
| 08:50.50 | MWE | know which channel that it will use |
| 08:51.45 | MWE | or how yuo can stop a call file ringing the destination |
| 08:51.58 | kaldemar | you define the channel in the call file |
| 08:52.22 | MWE | thatÅ an dial SIP/phonenumber@dtmf |
| 08:52.34 | kaldemar | and you can hang up a channel with soft hangup |
| 08:52.36 | MWE | so how should I know which channel? |
| 08:52.55 | MWE | I start an application with the call file when the destination picks up.. |
| 08:53.58 | kaldemar | "core show channels concise" shows you all channels with the identifiers. |
| 08:56.25 | *** part/#asterisk digitalirony (n=digitali@shellium/member/digitalirony) |
| 08:57.52 | MWE | and is it possible to "kill" the channel by identifiers |
| 08:57.59 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 08:59.06 | kaldemar | as i said a few lines up, soft hangup |
| 08:59.10 | MEW`meeting | ok |
| 08:59.12 | MEW`meeting | ty |
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| 09:21.02 | freckle | Any idea why Set(Auth=${SIP_HEADER(Authorization)}) would not work on a REFER even though Authorization header exists, this works fine on INVITES just not on REFER |
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| 09:23.32 | xrmx__ | when i am already on call on a polycom ip650 and i get another call the first one is silenced, do you know how to avoid that? |
| 09:23.52 | yang | Argh, I have received Polycom phone package without a power supply ... |
| 09:24.23 | xrmx__ | yang, you bought a poe only package probably :) |
| 09:25.16 | yang | xrmx__: yes, its a PoE version and power version also (combined) |
| 09:25.30 | yang | i wonder if any of the network cards can produce PoE ? |
| 09:25.36 | yang | PCI cards |
| 09:25.53 | xrmx__ | yang, i don't think so |
| 09:28.17 | yang | that is probably the reason why this phone was discounted |
| 09:28.37 | xrmx__ | yang, which one? |
| 09:28.38 | yang | It might work on universal adapter, I just need to find out what voltage it uses |
| 09:28.45 | yang | Polycom IP 321 |
| 09:30.19 | yang | well the auction actually sas "without power outlet" but I saw it only now |
| 09:30.46 | xrmx__ | yang, polycom 32x are annyoing without the second ethernet port |
| 09:30.52 | yang | the PoE devices are usually quite expenssive , I think |
| 09:31.28 | yang | the switches I mean |
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| 10:02.21 | smtx | yang: quality is always not cheap |
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| 10:05.02 | bio-tty | in * v1.6.1.6 i bridge two sip legs with dial and have supportvideo yes, and both peers registered have h263 and alaw only. * goes out with only an audio media line to the callee. why? whats new? must i do a special dial? |
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| 10:29.45 | ast-thecode | Anyone of you can suggest a (free or commercial) T38 gateway for Asterisk that WORK?? |
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| 11:57.24 | ast-thecode | Anyone of you can suggest a (free or commercial) T38 gateway for Asterisk that WORK?? |
| 12:01.02 | tamiel | ast-thecode: callweaver probably meet your needs |
| 12:01.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:02.01 | bio-tty | in * v1.6.1.6 i bridge two sip legs with dial and have supportvideo yes, and both peers registered have h263 and alaw only. * goes out with only an audio media line to the callee. why? whats new? must i do a special dial? |
| 12:03.25 | [TK]D-Fender | bio-tty: I'm pretty sure Bridge does not cause any kind of reinvite but rather takes the audio from each sid and jsut passes them, transcoding where it must. |
| 12:03.31 | [TK]D-Fender | bio-tty: Therefor not supported |
| 12:05.58 | bio-tty | i mean, i use Dial (to do the bridge. didnt know about the bridge app( |
| 12:07.14 | bio-tty | combined - 0x8000c (ulaw|alaw|h263) is recognised for caller, but the response has audio only from * |
| 12:07.35 | bio-tty | audio and video is understood (core debug) |
| 12:10.03 | [TK]D-Fender | bio-tty: Do you also have "videosupport=yes" under [general] ? |
| 12:10.33 | bio-tty | when Dial is done, the channels i have are of type "0x80004, Rx: ACK" and "0x80004 Tx: BYE". so they are noted as ulaw+h263 and we said bye bye to one of them. |
| 12:10.54 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 12:10.54 | bio-tty | [TK]D-Fender: i have videsupport=yes under [general] _and_ under each peer |
| 12:11.12 | bio-tty | [TK]D-Fender: they are dynamic and both registered recognised as those peers |
| 12:12.33 | [TK]D-Fender | bio-tty: Pastebin a complete call from beginning to end including your sip.conf |
| 12:12.36 | [TK]D-Fender | ~pb |
| 12:12.37 | 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 |
| 12:12.40 | [TK]D-Fender | ^^^^^^^^^^ |
| 12:12.58 | bio-tty | i can see that the video media-line has been chopped off when * sends out the initial INVITE to the callee. * understands video but sends only this |
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| 12:17.19 | [TK]D-Fender | bio-tty: LINK please... |
| 12:18.21 | ast-thecode | Anyone of you can suggest a (free or commercial) T38 gateway for Asterisk that WORK?? |
| 12:18.40 | bio-tty | http://pastebin.com/d43eb462e |
| 12:18.51 | bio-tty | http://pastebin.com/d29fffcd5 |
| 12:19.44 | MEW`meeting | hi there, when I create a call file with as Channel: SIP/<phonenumber>@dtmfsay what should be the command that the softhangup will kill that call? just "softhang up phonenumbers@dtmfsay"? |
| 12:21.18 | [TK]D-Fender | MEW`meeting: No, go look at the channel name while its up |
| 12:21.50 | bio-tty | [TK]D-Fender: you see from the trace its pretty weird hmm? |
| 12:22.09 | [TK]D-Fender | MEW`meeting: it will look like "SIP/dtmfsay-ABCD" where ABCD is a 4 hex digit suffix added for uniqueness |
| 12:23.09 | MEW`meeting | <PROTECTED> |
| 12:23.24 | MEW`meeting | but |
| 12:23.58 | [TK]D-Fender | bio-tty: #368 : NO video |
| 12:24.00 | MEW`meeting | when I'm on a script that's created the call file how can I know in the script which channel it is used for that made callfile |
| 12:24.10 | [TK]D-Fender | bio-tty: and you did not specify your codecs clearly in each peer |
| 12:24.40 | [TK]D-Fender | MEW`meeting: What "script"? |
| 12:24.44 | MEW`meeting | AGI |
| 12:24.56 | MEW`meeting | php |
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| 12:25.26 | [TK]D-Fender | MEW`meeting: You won't know unless you track the new channel coming up and ID it with a value you set in the call file |
| 12:25.54 | bio-tty | [TK]D-Fender: ofcourse theres no video at #368 -- * didnt offer any |
| 12:26.14 | MEW`meeting | is there a special command for that new channel monitoring? |
| 12:26.27 | bio-tty | [TK]D-Fender: i didnt see any sample config documenting how video codecs are specified |
| 12:26.31 | kaldemar | MEW`meeting: as i told earlier, you get the whole channel name with "core show channels concise" |
| 12:26.37 | [TK]D-Fender | bio-tty: And you did not specify the codecs in your peers |
| 12:26.49 | [TK]D-Fender | bio-tty: "allow=h263" <- this is basic stuff. |
| 12:27.08 | bio-tty | [TK]D-Fender: as i dont do _any_ disallow=all i thought * is open for all |
| 12:27.21 | MEW`meeting | [TK]D-Fender, thnx kaldemar thnx, your solutions will be something for monday :) |
| 12:27.28 | [TK]D-Fender | bio-tty: don't think when you can do. |
| 12:27.46 | bio-tty | does |
| 12:28.07 | bio-tty | [TK]D-Fender: so i just uncomment my three nice lines in each peer then? |
| 12:28.29 | [TK]D-Fender | bio-tty: "disallow=all" the "allow=" for each codec you specifically want |
| 12:28.31 | robl^laptop | [TK]D-Fender: true! thinking is just a useless past time ;-) |
| 12:29.00 | [TK]D-Fender | robl^laptop: Yes, any signs of thought from you are definitely in the distant past ;) |
| 12:29.08 | bio-tty | [TK]D-Fender: did it and sip reload. same result |
| 12:29.40 | [TK]D-Fender | bio-tty: And I don't see your new configs and debug. Also please permanently strip all commented out lines from your sip.conf |
| 12:29.58 | bio-tty | [TK]D-Fender: okay |
| 12:33.02 | bio-tty | http://pastebin.com/d6ecd6a8c |
| 12:33.06 | bio-tty | http://pastebin.com/d2b563307 |
| 12:35.02 | bio-tty | [TK]D-Fender: seems to me that dial is stripping away the channel where it had the h263 |
| 12:35.47 | creativx | has anyone seen usb headsets with exterior noise cancellation? |
| 12:35.53 | bio-tty | [TK]D-Fender: s/channel/medialine/ |
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| 12:47.54 | bio-tty | [TK]D-Fender: maybe its just because parse_session_expires fails? |
| 12:50.34 | bio-tty | [TK]D-Fender: or related to the warning from process_sdp? |
| 12:51.09 | [TK]D-Fender | bio-tty: I don't see the video on the invite to "C" |
| 12:51.25 | bio-tty | [TK]D-Fender: and the problem is why * does not send that out |
| 12:51.40 | bio-tty | [TK]D-Fender: as it understood audio+video from the incoming leg |
| 12:51.56 | [TK]D-Fender | bio-tty: yes the incoming looks fine.. its the outgoing. |
| 12:52.13 | bio-tty | [TK]D-Fender: is there even more logging i can turn on? |
| 12:52.55 | [TK]D-Fender | bio-tty: nothing I'm aware of... SIP is what's responsible here and I don't see the codec being added... |
| 12:53.42 | bio-tty | [TK]D-Fender: and the follow-up result is that the two channels cant be bridged as their number of medialines differ i guess. |
| 12:53.46 | [TK]D-Fender | bio-tty: maybe restart * in case one change wasn't applied... |
| 12:54.06 | [TK]D-Fender | bio-tty: If that doesn't do it I'm not sure where this would ahve gone wrong... certainly doesn't look like your configs |
| 12:54.28 | bio-tty | [TK]D-Fender: no, not any longer |
| 12:54.43 | bio-tty | [TK]D-Fender: i will dig into this anyway. thanks for the help so far |
| 12:55.15 | [TK]D-Fender | bio-tty: You may want to give 1.6.0 branch a try just in case there is a newer bug introduced on this |
| 12:56.37 | bio-tty | [TK]D-Fender: it seems that there are bugs in the latest 1.4 and 1.6 -- on latest stable 1.4 the * crashed when i sent it the usual BYE |
| 12:59.23 | bio-tty | and * 1.6.1 isnt able to parse "Session-Expires: 500; refresher=uas" -- is that a legitimate behaviour? |
| 12:59.25 | [TK]D-Fender | bio-tty: OK well keep at it... it may be worth it to open a ticket on this |
| 12:59.55 | [TK]D-Fender | bio-tty: I'm not a true SIP expert and can't confirm all of those headers like that one. |
| 13:00.07 | bio-tty | [TK]D-Fender: i see that * itself has no blank after the ";" -- im not quite shure on the standard here |
| 13:00.53 | bio-tty | downloads 1.6.0 |
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| 13:04.44 | hesco | I'm using a GT-486 on my desk and am curious to know how to make call-waiting and three-way calling work. Any guidance to be found here? |
| 13:05.22 | [TK]D-Fender | hesco: Considered reading its manual? |
| 13:05.23 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 13:06.02 | hesco | yes, of course I have |
| 13:08.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.21 | [TK]D-Fender | hesco: Odds are it works the same as common analog with Flash for call waiting, and the same kind of normal procedure to do 3-way. |
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| 13:11.07 | hesco | with the flash button I have been able either to disconnect from the current call or to simply interrupt, but not change from the current call, but I've never been able to balance between the two calls. |
| 13:11.40 | hesco | I'm thinking there is some combination of configuration options I don't yet understand how to set |
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| 13:13.32 | [TK]D-Fender | hesco: 3-way = on call. Flash. Place 2nd call. Get answered. Talk. Press Flash. Now in 3-way |
| 13:14.56 | Baylink2 | hesco: you're trying to *alternate* instead of ending up in a conference? |
| 13:15.25 | hesco | thank you, yes, I've been using the feature for over twenty years now. just have not been able to manage with the HT-486 |
| 13:15.26 | bio-tty | [TK]D-Fender: i got the exact same thing with * v1.6.0.15 |
| 13:15.49 | [TK]D-Fender | bio-tty: I'd say put ti all up on the tracker... |
| 13:16.37 | hesco | I've tried to wind up on a conference call, but the phone would not cooperate. call-waiting same way. No use of flash switch seems to work as expected. |
| 13:17.52 | ManxPower-work | hesco: All those functions are totally done by the phone, not Asterisk. The only thing that might impact this is any call-limit/calllimit settings in sip.conf. If you are limiting the device to 1 call then you can't do things like three-way calls. |
| 13:18.28 | [TK]D-Fender | hesco: Or the features need to be enabled in thier config somewhere |
| 13:19.42 | hesco | yes, I'm again reviewing the settings in the web-accessible administrative interface and cannot imagine how this thing is misconfigured. |
| 13:20.03 | hesco | my iax provider is configured to permit multiple simultaneous calls. |
| 13:20.24 | Katty | has breakfast! |
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| 13:26.23 | Baylink2 | How the analog interface reacts to a switchhook flash and translates that into commands is pretty much entirely the province of the processor and code directly attached to that analog interface (IE: the ATA) |
| 13:28.27 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 13:28.52 | Katty | http://www.youtube.com/watch?v=DWCOYJg9ps4 <- Fun! |
| 13:29.13 | slowhand | tzafrir_laptop: after update dahdi-linux with the svn now, asterisk run. |
| 13:30.09 | bio-tty | [TK]D-Fender: the reason for the broken bridging was the Session-Expires header-field. ad-hoc fix was to set session-timers=refuse (it does however not refuse as is said in the comment of chan_sip.c) |
| 13:30.48 | bio-tty | [TK]D-Fender: so now its bridged, but the video line gets zero-ported by asterisk |
| 13:31.53 | *** join/#asterisk [netman] (n=netman@80.31.167.130) [NETSPLIT VICTIM] |
| 13:32.06 | [TK]D-Fender | bio-tty: Sounds like you've got some real ticket-worthy #'s there |
| 13:32.45 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-eqiahonbjptyvywg) |
| 13:39.43 | Arcopix | Hey guys, if I'm making a call via the Asterisk Manager API, how can I get the CallerID presentation to prohibited? |
| 13:40.46 | [TK]D-Fender | Arcopix: "core show application setcallerpres" |
| 13:42.31 | Arcopix | Nah... Thanks [TK]D-Fender , but that aint gonna work... Via the Asterisk Management API I'm putting the call to channel like SIP/sipcarrier/num, so I'm not executing any dialplan before the channel is up |
| 13:42.45 | [TK]D-Fender | Arcopix: then change that |
| 13:42.46 | Arcopix | So as you may see the setcallerpres is not gonna work for me |
| 13:42.59 | Arcopix | That's is not quite possible |
| 13:43.05 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) [NETSPLIT VICTIM] |
| 13:43.06 | [TK]D-Fender | Arcopix: it is entirely possible |
| 13:43.29 | [TK]D-Fender | Arcopix: You aren't choosing the right channel type. |
| 13:43.39 | Arcopix | So you are proposing me to change it to Local/exten@context, so I could execute dialplan |
| 13:43.50 | Arcopix | Already tried that, though there is a small problem |
| 13:43.52 | [TK]D-Fender | Arcopix: WOW... on the first try no less... |
| 13:44.21 | Arcopix | a2billing looks for channels with names Local-.... which does not exist after the call was bridged ... |
| 13:44.28 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 13:44.51 | [TK]D-Fender | Arcopix: So basically your problem is that a2billing is retarded.... you have our most heart-felt sympathies |
| 13:45.01 | Zeeek | Astricon minus 4 |
| 13:45.13 | Arcopix | I hate that shit, though my customer wants it... |
| 13:45.20 | Arcopix | What could I do? |
| 13:45.22 | kaldemar | stab it! |
| 13:45.28 | [TK]D-Fender | Arcopix: And you can force the Local channel not to beidge back it you specified the "/n" at the end |
| 13:45.48 | [TK]D-Fender | Arcopix: Local/exte@context/n |
| 13:46.07 | Arcopix | That could do the trick... Thanks, I'll test it out |
| 13:46.09 | Zeeek | anyone here going to Astricon? Speak up! I have gifts for you |
| 13:46.43 | [TK]D-Fender | Arcopix: multiple other options such as having * call itself via a peer and having that do the out-call |
| 13:47.22 | *** join/#asterisk slayer192 (n=mrbob@gw-sf.securax.net) |
| 13:50.01 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 13:51.30 | iksik | what is wrong when that kind of error occures: Auto fallthrough, channel 'SIP/kantczak-15b8a148' status is 'UNKNOWN' ? |
| 13:52.26 | [TK]D-Fender | ikthat isn't an error. You ran out of dialplan to execute |
| 13:52.30 | [TK]D-Fender | iksik: that isn't an error. You ran out of dialplan to execute |
| 13:53.01 | iksik | hmm |
| 13:53.11 | iksik | weird |
| 13:53.59 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 13:54.17 | iksik | this one is weird to: -- Executing Ringing("SIP/kantczak-15b8bb98", "") |
| 13:54.19 | iksik | :| |
| 13:54.34 | *** join/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 13:54.53 | iksik | this sip trunk is configured exactly like another (working) one... but from other operator |
| 13:55.28 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:56.03 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:56.07 | superbeef | I'm smashing my face here.... I don't understand why I'm getting timed out trying to write, bad file descriptor: see bottom of http://pastebin.ca/1607050 |
| 13:56.33 | [TK]D-Fender | iksik: And you're showing nothing |
| 13:56.40 | superbeef | I've gotta get this thing working enough to turn up a PRI to ship out by noon lol |
| 13:57.58 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 13:58.28 | [TK]D-Fender | superbeef: Well you're in FreePBX-Land and for all I know that AMI request is bad, etc... |
| 13:58.51 | [TK]D-Fender | superbeef: there is nothing telling in that error and no details to go along-with |
| 13:59.07 | superbeef | [TK]D-Fender: yeah i know... i'm trying to figure out how to get more info |
| 13:59.30 | superbeef | [TK]D-Fender: driving me nuts i did a build just like this a week ago |
| 13:59.40 | superbeef | [TK]D-Fender: and this nonsense didnt happen |
| 13:59.43 | [TK]D-Fender | superbeef: Well its jsut a warning so no crash-logs, etc |
| 14:01.12 | superbeef | [TK]D-Fender: hmmm its enough to keep freepbx from working |
| 14:02.02 | *** join/#asterisk d00gster (n=doughant@94.99.167.4) |
| 14:05.27 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:05.35 | *** join/#asterisk d00gster (n=doughant@94.99.167.4) |
| 14:08.05 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 14:11.40 | *** join/#asterisk craig-t (n=craig-t@196-210-175-188-tvwt-esr-2.dynamic.isadsl.co.za) |
| 14:12.41 | craig-t | new to asterisk, just confused about something. a call centre with ten agents, would they need 10 physical lines plugged into their asterisk box? |
| 14:13.07 | jde | only if they are using traditional phones |
| 14:13.21 | [TK]D-Fender | craig-t: No. You can have as many lines as you want and as many phones as you want of whatever type you want |
| 14:13.44 | jde | and that. |
| 14:14.13 | craig-t | ok, so would i be able to have 10 simultaneous calls over one analogue line? |
| 14:14.25 | [TK]D-Fender | craig-t: Of course not |
| 14:14.28 | ManxPower-work | craig-t: You need as many lines as you need simultaneous calls |
| 14:14.48 | ManxPower-work | But the number of extensions seldom has anything to do with how many simultaneous calls you need to support. |
| 14:14.53 | [TK]D-Fender | craig-t: An analog line is jsut as dumb with 8 as with you plugging a $10 phone from walmart into it |
| 14:15.02 | [TK]D-Fender | * |
| 14:15.02 | iksik | huh, got it - almost |
| 14:15.28 | iksik | He is looking for my cellphone CID not target ID :| |
| 14:15.31 | jde | craig-t, what kind of phones are the agents using |
| 14:16.40 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:17.30 | ManxPower-work | with the need for 10 calls, I'd look into PRI instead of analog |
| 14:17.49 | jde | yeah, at least for the trunk |
| 14:17.55 | iksik | hm, i'm calling from 666 to 111, and asterisk is trying to find 666 ... not 111 :| |
| 14:17.56 | iksik | why ? |
| 14:18.07 | jde | but if your agents are using analog phones you're going to need an FXS card with at least 10 ports on it |
| 14:18.20 | jde | or just get a modular one |
| 14:18.27 | [TK]D-Fender | iksik: You aren't showing us anything. |
| 14:18.37 | [TK]D-Fender | iksik: Why bother asking when you don't? |
| 14:19.19 | *** join/#asterisk coppice (n=chatzill@148.162.17.210.dyn.pacific.net.hk) |
| 14:19.44 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:19.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:20.14 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:20.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:22.31 | iksik | [TK]D-Fender, You don't tell me, what do You want to see... |
| 14:22.46 | iksik | didn't tell me? :| |
| 14:23.16 | [TK]D-Fender | iksik: Do i see your dialplan and the call? No. Seriously, how would you not know what to show? "Hi my dialplan isn't working right, why?". SHOW US |
| 14:23.58 | Naikrovek | show us the dialplan |
| 14:24.02 | iksik | [TK]D-Fender, cause there is no dialplan? there is only: Log ( DEBUG, Call from ${CALLERID(all)} to ${EXTEN} ) |
| 14:24.05 | iksik | but huh, I remember that |
| 14:24.17 | iksik | You don't like to wasting your time... but You're doing it all the time |
| 14:24.35 | ManxPower-work | iksik: You can't process calls without a dialplan. Pastebin the CLI output if a failed call. |
| 14:24.44 | ManxPower-work | s/if/of |
| 14:24.46 | iksik | I just did it above |
| 14:24.54 | Naikrovek | iksik: that's not the entire dialplan |
| 14:25.01 | Naikrovek | iksik: if it is, that's your problem |
| 14:25.17 | [TK]D-Fender | [10:17]<iksik>hm, i'm calling from 666 to 111, and asterisk is trying to find 666 ... not 111 :| <- this is a call. Why aren't you showing su the CALL? |
| 14:25.33 | [TK]D-Fender | iksik: And call processing = dialplan. So how is this NOT a dialplan problem? |
| 14:26.06 | iksik | [TK]D-Fender, "<iksik> hm, i'm calling from 666 to 111" was the effect of simple one line DIAL and watching the logfile of database |
| 14:26.09 | kaldemar | i smell a gui |
| 14:26.19 | Naikrovek | kaldemar: that was my thought as well |
| 14:26.28 | ManxPower-work | iksik: edit /etc/asterisk/logger.conf uncomment the line that says console, connect to Asterisk as "asterisk -rvvv" and try a call, pastebin the output. |
| 14:26.41 | [TK]D-Fender | iksik: This : [10:17]<iksik>hm, i'm calling from 666 to 111, and asterisk is trying to find 666 ... not 111 :| |
| 14:26.43 | [TK]D-Fender | has nothing to do with : [10:23]<iksik>[TK]D-Fender, cause there is no dialplan? there is only: Log ( DEBUG, Call from ${CALLERID(all)} to ${EXTEN} ) |
| 14:26.55 | iksik | omg... STOP WASTING YOUR TIME dude |
| 14:27.05 | iksik | :S |
| 14:27.21 | Naikrovek | he's not wasting his time, you're wasting yours by not showing us what we need to see to fix your problem |
| 14:27.22 | [TK]D-Fender | iksik: Who says * is loking for 666? youa ren't showing anything |
| 14:27.28 | iksik | I was changing it about 20 times while You was wasting your (and mine now to) time |
| 14:28.01 | [TK]D-Fender | iksik: yes, and you didn't tell us about the changes. didn't show us the earlier version, didn't show us your now current version. |
| 14:28.18 | kaldemar | iksik: if you don't want to be helped properly, please don't bother to ask. |
| 14:28.19 | Naikrovek | and we still haven't seen the actual problem either |
| 14:28.21 | [TK]D-Fender | ikYou have done absolutely nothing to help anyone help you. |
| 14:29.10 | ManxPower-work | iksik: It's easy to annoy [TK]D-Fender, but when you have others that think you're an idiot, maybe [TK]D-Fender isn't so off the mark. |
| 14:29.31 | shido6 | popcorn's done.. |
| 14:29.43 | [TK]D-Fender | ManxPower-work: Actually... how often am I ever off the mark even with repeated opposition? ;) |
| 14:29.51 | [TK]D-Fender | shido6: Sit back and enjoy the show! |
| 14:30.06 | ManxPower-work | Over the years many people on this channel have come up with good, tried and true troubleshooting methods. One of the most important of those methods is pastebin'ing the CLI output of a failed call. |
| 14:30.09 | SuPrSluG | will somebody quit ringin that bell i can't concentrate |
| 14:30.26 | iksik | ManxPower-work, http://pastebin.com/m258aebc6 |
| 14:30.32 | [TK]D-Fender | SuPrSluG: DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! DING! ding! |
| 14:30.50 | superbeef | does asterisk have an alergy to 0 byte config files? |
| 14:30.56 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 14:31.26 | Zeeek | VoIP Users Conference : Today is Friday! Join the conference in 90 minutes at http://VUC.me and get on #voip-users-conference - oh and don't forget, this is International Annoy [TK]D-Fender Day. Get out there and do your part! |
| 14:31.40 | ManxPower-work | iksik: you have something seriously screwed up. extension 1 is matching your call, not extension 666 |
| 14:31.43 | [TK]D-Fender | iksik: This is not a valid extension, and clearly you are executing the "s" exten. |
| 14:31.56 | [TK]D-Fender | ManxPower-work: his formatting he's showing isn't legal |
| 14:32.06 | [TK]D-Fender | ManxPower-work: Makes "s" look like a priority which it isn't |
| 14:32.11 | ManxPower-work | [TK]D-Fender: I see that now. It is impossible for the line to generate that log message. |
| 14:32.16 | iksik | ManxPower-work, as I said, I made about 20 changes from the time I was asking here first time |
| 14:32.28 | iksik | now it looks like this |
| 14:32.36 | [TK]D-Fender | iksik: So your exten is "s". i see nothing wrong with that. |
| 14:32.51 | SuPrSluG | I wouldn't even know where to begin. maybe FUBAR |
| 14:33.23 | ManxPower-work | iksik: exactly what is the problem? |
| 14:33.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:34.20 | iksik | ManxPower-work, exactly that what You see.... This "S" thing... There should be ID of my peer extension, right? |
| 14:34.21 | ManxPower-work | What *I* see is a call came in from your SIP account "kantczak" without a destination extension sent by your carrier, because of that the call matches exten => s,1,Log. |
| 14:34.50 | ManxPower-work | iksik: it would be the destination your carrier is sending or "s" if your carrier does not send a destination extensions. |
| 14:35.05 | kaldemar | there's only 1,s but no s,1 :) |
| 14:35.10 | [TK]D-Fender | ikNo, ${EXTEN} is the EXtensioN you are in in the dialplan, not a DEVICE |
| 14:35.19 | iksik | really? |
| 14:35.20 | ManxPower-work | kaldemar: it's obvious he did not paste his exten line |
| 14:35.22 | [TK]D-Fender | iksik: this is dialplan 101 |
| 14:35.29 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 14:36.02 | ManxPower-work | kaldemar: he just tried to (wrongly) retype it, wasting our time. |
| 14:36.03 | kaldemar | ManxPower-work: really? |
| 14:36.15 | iksik | ManxPower-work, cause there is no "line"... it's from database... |
| 14:36.16 | kaldemar | ManxPower-work: i wan't being too serious there. |
| 14:36.19 | ManxPower-work | kaldemar: yeah. extension lines begin with exten => noe with "1" |
| 14:36.24 | iksik | ;> |
| 14:36.38 | [TK]D-Fender | iksik: well you are in EXTEnSION "s" in the dialplan this has nothing to do witht he SIP device that si placing the call. |
| 14:36.56 | [TK]D-Fender | iksik: You don't seem to understand the nature of the most important variable in Asterisk |
| 14:36.59 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 14:37.11 | ManxPower-work | iksik: Have you read thru the Asterisk Book? |
| 14:37.20 | ManxPower-work | ~book |
| 14:37.21 | 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 |
| 14:38.00 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:38.04 | iksik | [TK]D-Fender, and this is Your problem ;-) You write about something, and You are sure, that You know the best.... But no... I've got 2 SIP trunks setup and they works perfectly... this one is configured in EXACTLY the same way, and it doesn't work... You know an answer? No? Mabe send You some logs from my operator? |
| 14:38.07 | iksik | o.O |
| 14:38.09 | ManxPower-work | The Asterisk Book is not perfect, but it will help you learn the basics. |
| 14:38.13 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:38.26 | ManxPower-work | iksik: I already gave you the answer. |
| 14:38.40 | [TK]D-Fender | iksik: Yes, I DO know the ansewr, and ManxPower told you it to your face already |
| 14:38.50 | iksik | ManxPower-work, I know, thank You... Now i'm answering to [TK]D-Fender... |
| 14:39.00 | ManxPower-work | twice, actually. |
| 14:39.07 | [TK]D-Fender | [10:34]<ManxPower-work>What *I* see is a call came in from your SIP account "kantczak" without a destination extension sent by your carrier, because of that the call matches exten => s,1,Log. |
| 14:39.29 | iksik | [TK]D-Fender, lol ;-) with nothing more information that I paste it here at start of this useless conversation... what is conclusion ? |
| 14:39.31 | iksik | ;-) |
| 14:39.37 | [TK]D-Fender | iksik: 1,s,LOG(DEBUG, Call from: ${EXTEN} / ${CALLERID(all)} <- ${EXTEN} = "s" |
| 14:40.21 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 14:40.23 | iksik | exten = s, right |
| 14:41.18 | ManxPower-work | iksik: do you understand what extension "s" is? |
| 14:41.30 | [TK]D-Fender | ManxPower-work: that isn't event he bigger issue |
| 14:41.41 | [TK]D-Fender | ManxPower-work: he doesn't understand waht ${EXTEN} holds. |
| 14:41.58 | Zeeek | different things for different people |
| 14:41.58 | ManxPower-work | one thing at a time |
| 14:42.10 | iksik | [TK]D-Fender, there IS a little difference... between: what ${EXTEN} SHOULD HOLDS, and what ${EXTEN} holds right now |
| 14:42.14 | ManxPower-work | Most n00bs don't understand "s" |
| 14:42.17 | iksik | do You see it? |
| 14:42.18 | iksik | ;> |
| 14:42.19 | [TK]D-Fender | ManxPower-work: "s" is nowhere near as relevant as far as literal name is concerned |
| 14:42.36 | [TK]D-Fender | iksik: WRONG |
| 14:42.40 | iksik | sure ;-) |
| 14:43.08 | [TK]D-Fender | iksik: EXTEN => s,1,LOG() <- Exten is "s" becuase thats the EXTEnsION you're on |
| 14:43.13 | ManxPower-work | iksik: so you understand that the reason the call is not matching what you want it to match is because your carrier is NOT SENDING THE NUMBER? |
| 14:43.20 | [TK]D-Fender | iksik: that is not a NUMEBR |
| 14:43.25 | iksik | omg :S |
| 14:43.25 | [TK]D-Fender | iksik: that is not a SIP PEER NAMENUMEBR |
| 14:43.38 | iksik | ok, You know better huh? ;-) |
| 14:43.41 | iksik | then explain |
| 14:43.48 | iksik | I've removed that line with s,... |
| 14:43.51 | ManxPower-work | Now if you want to know WHY the carrier is not sending the number...well I'd ask your carrier. |
| 14:43.59 | iksik | then from where is that : [Oct 9 18:28:01] NOTICE[5143]: chan_sip.c:18523 handle_request_invite: Call from 'kantczak' to extension 's' rejected because extension not found. |
| 14:44.00 | iksik | ? |
| 14:44.01 | iksik | ;> |
| 14:44.03 | [TK]D-Fender | ManxPower-work: No, we know why they're sending "s" :) |
| 14:44.07 | iksik | There is NO 's' |
| 14:44.20 | [TK]D-Fender | iksik: and? |
| 14:44.20 | ManxPower-work | iksik: That is exactly what should happen if you remove that line. |
| 14:44.36 | [TK]D-Fender | iksik: because they are looking for "s". And you don't even know why |
| 14:45.06 | ManxPower-work | [TK]D-Fender: can't we just wish him the best of luck and move on? |
| 14:45.30 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:45.37 | [TK]D-Fender | ManxPower-work: this equine is far from tenderized ;) |
| 14:45.42 | ManxPower-work | iksik: I wish you the best of luck in getting this fixed. |
| 14:45.51 | iksik | thanks... |
| 14:45.56 | [TK]D-Fender | iksik: Perhaps you should also learn how REGiSTER works |
| 14:46.09 | [TK]D-Fender | iksik: because THAT is why the call is going to "s". |
| 14:46.18 | iksik | no way |
| 14:46.22 | [TK]D-Fender | iksik: YES |
| 14:46.31 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:47.16 | [TK]D-Fender | iksik: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 14:47.25 | iksik | uhhh... |
| 14:47.47 | bio-tty | dial seem to have an issue with video medialines |
| 14:48.06 | bio-tty | wasnt so before. |
| 14:48.19 | bio-tty | only in the latest * versions |
| 14:48.24 | iksik | Do You understand, that I'm trying a WORKING configuration with other operators? With this one it doesn't works and it seems that general problem is that I've receive replaced CALLE ID with CALLER ID? |
| 14:49.26 | [TK]D-Fender | iksik: ${EXTEN} holds "s" because thats the exten in the dialplan that is processing. You don't seem to understand that about this variable. its content CHANGES. |
| 14:49.42 | [TK]D-Fender | iksik: AND you don't seem to have read the instructions that explains WHY that call landed on "s" in the first place. |
| 14:49.51 | iksik | there is no ${EXTEN} either ;> |
| 14:49.57 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 14:50.09 | [TK]D-Fender | [10:39]<[TK]D-Fender>iksik: 1,s,LOG(DEBUG, Call from: ${EXTEN} / ${CALLERID(all)} <- ${EXTEN} = "s" |
| 14:50.19 | iksik | <iksik> I've removed that line with s, |
| 14:50.25 | [TK]D-Fender | iksik: so? |
| 14:50.29 | iksik | so ? |
| 14:50.36 | [TK]D-Fender | iksik: You expected it to hold something it never should |
| 14:50.38 | iksik | there is no ${EXTEN} |
| 14:51.04 | [TK]D-Fender | iksik: And I told you what is causing the call to look for "s" |
| 14:51.08 | iksik | [TK]D-Fender I exptect that when I'm calling from 111 to 222, asterisk receive that 222 is called, not 111 |
| 14:51.21 | [TK]D-Fender | iksik: where do we see this call? |
| 14:52.40 | iksik | there is only that what I've paste here |
| 14:53.18 | [TK]D-Fender | iksik: then get an actual call to show because what you've shown makes no reference to 111 or 222 |
| 14:53.26 | [TK]D-Fender | iksik: one-liners like that are worthless |
| 14:53.55 | [TK]D-Fender | iksik: Pastebin an entire call. |
| 14:54.11 | iksik | o.O |
| 14:54.16 | iksik | http://pastebin.com/m258aebc6 |
| 14:54.20 | iksik | this is entire call |
| 14:54.53 | iksik | and when this call was happend, asterisk was trying to find my cellphone number in database... but it should'nt |
| 14:55.39 | [TK]D-Fender | iksik: there are no #'s in that at all |
| 14:55.55 | [TK]D-Fender | ikAnd no, it si not looking for a number in a database |
| 14:55.56 | iksik | hm? :| |
| 14:56.06 | [TK]D-Fender | iksik: We see * looking for the "s" exten for the call to land on |
| 14:56.15 | iksik | lol ;-) |
| 14:56.26 | [TK]D-Fender | iksik: And it does find it, and spits out 1 log entry |
| 14:56.48 | [TK]D-Fender | iksik: So you are processing the call and making 1 log enttry and not doing anything more. |
| 14:56.49 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 14:56.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:57.04 | iksik | [TK]D-Fender... no ;> |
| 14:57.13 | [TK]D-Fender | iksik: YES. We sii it in the debug |
| 14:57.22 | iksik | no |
| 14:57.26 | iksik | You don't ;-) |
| 14:57.32 | [TK]D-Fender | iksik: Call lands on s,1, LOG() and then HANGS UP because there are no more priorities to execute |
| 14:57.48 | [TK]D-Fender | -- Auto fallthrough, channel 'SIP/kantczak-15b98a70' status is 'UNKNOWN' <--- ran out of more stuff to do |
| 14:58.26 | iksik | and SELECT * FROM extensions WHERE 'name' = 'MYCELLPHONENUMBER' |
| 14:58.27 | iksik | ;> |
| 14:58.33 | *** join/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 14:58.53 | [TK]D-Fender | iksik: Youa ren't showing the entire call. |
| 14:59.48 | iksik | there is nothing more |
| 14:59.49 | [TK]D-Fender | [10:58]<iksik>and SELECT * FROM extensions WHERE 'name' = 'MYCELLPHONENUMBER' <- apparently there is |
| 14:59.49 | [TK]D-Fender | iksik: we can see what * is executing. |
| 14:59.49 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.03 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 15:00.14 | [TK]D-Fender | iksik: Its still landing on "s", Executing that log command and then hanging up because there is nothing more to do. |
| 15:00.27 | iksik | [TK]D-Fender, don't be rude... I was saying You about that ... ;> |
| 15:00.38 | iksik | I don't see any reason why You don' belive, what I said ;> |
| 15:00.51 | *** join/#asterisk ekacnet (n=ekacnet@ns1.eurocopter.ru) |
| 15:00.52 | [TK]D-Fender | iksik: Your pastebin shows us what is going on. |
| 15:00.54 | ekacnet | hello |
| 15:00.55 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:01.21 | [TK]D-Fender | iksik: And I've repeated it twice now |
| 15:01.34 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:01.44 | iksik | ok, then first and second configurations, are working with a little help of MAGIC |
| 15:01.50 | iksik | and this one don't want to |
| 15:01.50 | iksik | ;-) |
| 15:01.52 | iksik | right? ;-) |
| 15:01.58 | ekacnet | the function devstate is only accurate when the tested the device is called and not calling |
| 15:02.03 | iksik | bleh |
| 15:02.28 | ekacnet | ie if sip/100 is calling devstate(sip/100) will return not in_use |
| 15:02.33 | [TK]D-Fender | iksik: depends how you define "working". I see it entering the dialplan, making 1 log entry and hanging up. is that what you want? |
| 15:03.15 | [TK]D-Fender | ekacnet: pastebin your sip peer entry |
| 15:03.17 | [TK]D-Fender | ~pb |
| 15:03.18 | 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 |
| 15:03.22 | [TK]D-Fender | ^^^^ |
| 15:03.37 | ekacnet | at the opposite if sip/100 is called by sip/101 then devstate(sip/101) will either return "INUSE" or "RINGING" |
| 15:04.23 | Baylink2 | Hey, D-fender: why dont we change that IB entry to mention "asterisk.pastebin.com"? They provide "customized" ones; no sense not using it. |
| 15:04.32 | iksik | [TK]D-Fender... it works, cause it gets CALLEE ID and CALLER ID in proper places... this one doesn't |
| 15:05.23 | [TK]D-Fender | iksik: Callee (${EXTEN}) lands on "s" because of your REGISTER statement. And the callerID is already proper. |
| 15:05.46 | eppigy | hello |
| 15:05.52 | [TK]D-Fender | eppigy: YOU ARE DAVE!!!!!!!!!!!!! |
| 15:05.54 | Baylink2 | Hey |
| 15:06.02 | iksik | my register statement looks exactly the same like others ;< |
| 15:06.05 | Baylink2 | Isn't there already a bot that does that? ;-) |
| 15:06.09 | eppigy | [TK]D-Fender: CORRECT |
| 15:06.25 | [TK]D-Fender | iksik: No, it doesn't. |
| 15:06.42 | [TK]D-Fender | iksik: There is only 1 reason for the call to land on "s" the way it is |
| 15:06.50 | [TK]D-Fender | iksik: and the documentation tells you this |
| 15:06.57 | [TK]D-Fender | iksik: and I even linked you to it. |
| 15:07.07 | ekacnet | [TK]D-Fender: http://pastebin.com/m7b7fc79d |
| 15:07.37 | [TK]D-Fender | iksik: And then having that lack of understanding componded with the complete lack of understanding on how ${EXTEN} works is even worse |
| 15:08.01 | [TK]D-Fender | [11:03]<[TK]D-Fender>ekacnet: pastebin your sip peer entry <--------- |
| 15:09.01 | iksik | [TK]D-Fender, I admire you for your love to your own overvalued knowledge ;-) |
| 15:09.22 | iksik | cause these thing.... just come up from Your head ;-) |
| 15:09.26 | iksik | thins* |
| 15:09.29 | iksik | blah things* |
| 15:09.46 | [TK]D-Fender | iksik: No apparently undestanding how ${EXTEN} works is of GREAT value when using ASTERISK |
| 15:11.04 | ekacnet | [TK]D-Fender: you mean sip.conf entrires ? |
| 15:11.11 | [TK]D-Fender | iksik: you have shown sever problems in learning *, providing backup, and taking advice when given. |
| 15:11.15 | [TK]D-Fender | ekacnet: Yes |
| 15:11.33 | iksik | [TK]D-Fender, right... ;-) |
| 15:12.24 | ekacnet | [TK]D-Fender: http://pastebin.com/m110fabbb |
| 15:12.43 | iksik | [TK]D-Fender.. lol "providing backup" You are the best ;-) |
| 15:13.07 | iksik | ...fairy ;-) |
| 15:14.29 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 15:14.44 | [TK]D-Fender | ekacnet: "type=peer", "call-limit=99" |
| 15:15.02 | [TK]D-Fender | iksik: "fairy" huh? You really want to go there? |
| 15:15.48 | iksik | ;-) |
| 15:16.32 | ekacnet | [TK]D-Fender: sorry ? |
| 15:16.53 | [TK]D-Fender | ekacnet: make those changes and test |
| 15:19.08 | *** join/#asterisk ska (n=ska@cpe-70-124-73-96.austin.res.rr.com) |
| 15:23.20 | voipmonk | chokes on a kernel |
| 15:23.46 | voipmonk | mumbles something to a friend, "...yeah he said fairy." |
| 15:24.35 | ekacnet | [TK]D-Fender: no success still the same |
| 15:24.42 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 15:25.51 | [TK]D-Fender | ekacnet: please PB your new sip.conf & dialplan, CLI output of the call with "core show channels" and "core show hints" |
| 15:27.29 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 15:33.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:35.33 | superbeef | so i think my idiot error problem was because i made a seperate partition for /tmp and my permissions were 755 lol |
| 15:37.29 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 15:38.59 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 15:39.53 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:41.59 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:42.27 | Katty | hugs seanmh |
| 15:42.43 | seanmh | morning |
| 15:42.50 | seanmh | oh.. what were you waiting for? 1.6 support? |
| 15:43.40 | Katty | well, really. i was waiting for 5pm. |
| 15:44.02 | Katty | maybe my monday dr appt so i can find out why my ears are ringing. |
| 15:44.16 | Katty | but 1.6 support would be a good thing to know about. |
| 15:44.32 | seanmh | :D |
| 15:44.34 | seanmh | we have it |
| 15:44.38 | [TK]D-Fender | Katty: Lower the gain and point the mic farther away ;) |
| 15:44.49 | seanmh | can you e-mail isymphony-beta@i9technologies.com and I'll get you links |
| 15:45.42 | Katty | :> |
| 15:45.54 | *** join/#asterisk errotan (n=errotan@5403E4AD.catv.pool.telekom.hu) |
| 15:46.25 | *** join/#asterisk clintc (n=clintc@n128-227-249-6.xlate.ufl.edu) |
| 15:46.27 | Katty | noogies [TK]D-Fender |
| 15:47.55 | Katty | [TK]D-Fender: i've already spent an hour on the phone with 4 different offices getting all of my lab work, test results, and so forth together for my new GP |
| 15:48.31 | [TK]D-Fender | Katty: I'm sorry.... your neurosis is a "pre-existing condition" :p |
| 15:48.35 | Katty | seanmh: thank you sir. i will setup a 1.6 test box and give 'er a whirl. |
| 15:49.04 | seanmh | great |
| 15:49.04 | seanmh | we haven't had any bugs as of yet |
| 15:49.09 | Katty | [TK]D-Fender: i would think you'd be happy that i now have a GP who's primary focus is nuerotic patients ;P |
| 15:49.34 | Katty | seanmh: is the beta the 'free' or 'paid' version |
| 15:49.43 | [TK]D-Fender | Katty: No, This just gives you someone to run to with problems you should work out by yourself :) |
| 15:49.45 | p3nguin | superbeef: Does that mean you do know what the perms should have been and you can fix it? |
| 15:49.58 | seanmh | both |
| 15:50.08 | Katty | seanmh: just asks for a license? |
| 15:50.16 | Katty | i thinki have one of those somewher.e |
| 15:50.21 | Katty | in my email. once upon a time. |
| 15:50.23 | seanmh | I'll get you one. |
| 15:50.25 | Arcopix | bye |
| 15:50.29 | seanmh | oh you do? |
| 15:50.29 | *** part/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 15:50.34 | Katty | that's okay. i'm sure it's around here somewhere. |
| 15:50.35 | seanmh | if you have an old one I can modify it |
| 15:50.49 | Katty | well we're the people with the funky MAC address problem. |
| 15:51.08 | Katty | you've reset it eleventy billion times. |
| 15:51.19 | Katty | i probably have 20 emails with the license key in it |
| 15:51.20 | Katty | but! |
| 15:51.23 | eppigy | 8[] |
| 15:51.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:51.28 | Katty | it would be helpful if i had another key for testing |
| 15:51.31 | seanmh | :D |
| 15:51.35 | seanmh | Company name? |
| 15:51.42 | Katty | that way our production server will still work |
| 15:51.43 | eppigy | SLOSSIN INC |
| 15:51.47 | Katty | it's in my email signature |
| 15:52.01 | Katty | i can /query you if you have difficulting locating the account. |
| 15:52.05 | Katty | eppigy: hello, dear. |
| 15:52.56 | Katty | seanmh: i owe you cookies. |
| 15:53.15 | seanmh | emailed |
| 15:53.25 | Katty | seanmh: what kind do you guys want? |
| 15:53.25 | seanmh | :D |
| 15:53.35 | seanmh | no cookies necessary ;) |
| 15:53.39 | *** join/#asterisk Fs0L (n=Fs0L@136.223.19.17) |
| 15:53.43 | Katty | psh. |
| 15:54.20 | Katty | if you don't give me a good answer, i will just have to send chocolate chip. |
| 15:54.27 | *** join/#asterisk pyite_mac (n=dschreib@unaffiliated/pyite) |
| 15:55.14 | p3nguin | For bad answers, I would send some dough instead of cookies. Not good dough, though. Something like Snicker Doodle dough. Yuck! |
| 15:56.03 | Katty | seanmh: a charitable donation would also be an acceptable alternative. |
| 15:56.14 | Katty | seanmh: if that's the kind of cookies you'd prefer. |
| 15:56.23 | eppigy | Katty: hiya :> |
| 15:56.27 | Katty | eppigy: OHAI |
| 15:56.28 | seanmh | hah |
| 15:56.47 | Katty | seanmh: how about a nice letter to the president? |
| 15:57.02 | Katty | DEAR SIRS, sean is awesome. Love, Katty. |
| 15:57.11 | Katty | PS - he needs pay raise. |
| 15:58.48 | seanmh | haha |
| 15:59.00 | seanmh | I dunno if I could give myself a raise :D |
| 15:59.35 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:306e:2c75:b2c9:8871) |
| 15:59.36 | thehar | sneaks up behind pyite_mac |
| 16:00.15 | thehar | rides the google wave |
| 16:00.27 | *** join/#asterisk pittmodal (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 16:00.41 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 16:01.05 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 16:01.07 | Katty | seanmh: central SW or Coors rd? |
| 16:01.15 | *** part/#asterisk Fs0L (n=Fs0L@136.223.19.17) |
| 16:01.20 | seanmh | haha.. you're actually going to do it |
| 16:01.27 | Katty | taps fingers. |
| 16:01.30 | Katty | which road... |
| 16:01.34 | Katty | don't make me randomly pick one |
| 16:01.41 | seanmh | :D |
| 16:01.45 | Katty | I WILL DO EET |
| 16:02.04 | seanmh | we're located at the central address |
| 16:02.19 | Katty | k. |
| 16:02.20 | *** join/#asterisk anonymouz666 (n=anonymou@187-28-37-118.poolip.RJO.embratel.net.br) |
| 16:03.18 | Katty | and now what kind. |
| 16:03.37 | seanmh | SHRUG |
| 16:03.38 | seanmh | whoa |
| 16:03.38 | seanmh | caps |
| 16:03.44 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:03.46 | seanmh | it's really not needed ;) |
| 16:03.53 | Katty | i don't think i have a recipe for shrug cookies ^_- |
| 16:04.25 | seanmh | they're good |
| 16:04.28 | seanmh | you should try 'em |
| 16:05.54 | Chesther | Katty - send these: http://bit.ly/167DQ |
| 16:06.11 | ekacnet | [TK]D-Fender: I think I find: it's limitonpeer that is needed ! |
| 16:06.26 | Katty | bacon? ^_- |
| 16:06.40 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 16:07.25 | Katty | Chesther: do you work with i9technologies? |
| 16:07.33 | geneticx | are there any sip phones for less than $100 ? |
| 16:07.39 | Katty | geneticx: yes. |
| 16:07.50 | Katty | geneticx: just keep in mind you get what you pay for. |
| 16:07.51 | Chesther | Katty: Nope. Cornell University. |
| 16:07.55 | Katty | Chesther: ah, k. |
| 16:07.59 | coppice | geneticx: many |
| 16:08.13 | geneticx | any that you recommend? |
| 16:08.15 | Katty | geneticx: you might have a look at the polycom 320 or 330 |
| 16:08.30 | Katty | geneticx: they're around 100, good quality. |
| 16:08.34 | ManxPower-work | ~phones |
| 16:08.35 | 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. |
| 16:08.54 | Katty | geneticx: the 320 and 330 are the same, except the 330 has a 'line out' network jack. |
| 16:09.02 | Katty | geneticx: so you could connect a laptop or whatever to the phone. |
| 16:09.19 | Chesther | looks at the crappy Grandstream BugeTone on his desk and smiles. |
| 16:09.22 | geneticx | Katty: Ah sort of like a 2 port switch |
| 16:09.28 | Katty | yeah. |
| 16:09.37 | geneticx | nice |
| 16:09.42 | geneticx | Thanks for the input |
| 16:09.43 | Katty | yep, it's handy. |
| 16:09.53 | Katty | especially if you don't want to run any more cable drops |
| 16:10.09 | geneticx | that's exactly why I like it. |
| 16:10.18 | geneticx | =D |
| 16:10.23 | Katty | mhmm. |
| 16:10.24 | p3nguin | geneticx: You can get Linksys phones for around $80 on Amazon.com. |
| 16:10.49 | Katty | snom's are pretty cheap too. |
| 16:10.49 | Katty | i have one at the house. |
| 16:10.57 | Katty | it's okay for the house...but i wouldn't want to use it here at work. |
| 16:11.02 | geneticx | I like the SPA942 but I need a bulk of 10 and haven't found reasonable bulk prices yet |
| 16:11.16 | Katty | have you checked voip-supply.com? |
| 16:11.24 | Katty | they do bulk pricing |
| 16:11.33 | Katty | gotta call em tho |
| 16:11.36 | [TK]D-Fender | Katty: Of course their regular pricing sucks :) |
| 16:11.37 | geneticx | p3nguin: which model? |
| 16:11.57 | p3nguin | $82.24 for the SPA-921: http://www.amazon.com/Cisco-SPA921-1-line-1-port-Ethernet/dp/B000F16HX8/ref=sr_1_1?ie=UTF8&s=electronics&qid=1255104653&sr=8-1 |
| 16:12.00 | *** join/#asterisk baijum (n=baiju@122.167.84.9) |
| 16:12.17 | Katty | [TK]D-Fender: who do you like to order from? |
| 16:12.19 | geneticx | Katty: yeah I'll give it a shot |
| 16:12.29 | [TK]D-Fender | Katty: www.telephonydepot.com |
| 16:12.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-222-38-76.msy.bellsouth.net) |
| 16:12.42 | [TK]D-Fender | SPA-921 = bleh |
| 16:12.45 | Katty | looks |
| 16:12.56 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 16:12.57 | p3nguin | The 942 is $99.99. |
| 16:13.05 | geneticx | p3nguin: not bad, but I would like at least 2 line |
| 16:13.37 | Katty | geneticx: depending on how your server acts, 'line' support may be irrelivant |
| 16:13.54 | Katty | geneticx: a lot of people have their server configured to just grab the next free line in a group |
| 16:14.14 | [TK]D-Fender | geneticx: http://www.ipphone-warehouse.com/Polycom-Soundpoint-IP-321-2200-12360-025-p/2200-12360-025.htm |
| 16:14.16 | Katty | [TK]D-Fender: that website has some good pricing |
| 16:14.59 | [TK]D-Fender | Katty: TD is very good on the customer service side too |
| 16:15.13 | Katty | [TK]D-Fender: do you have a contact you'd like to share with me? |
| 16:15.37 | geneticx | Katty: yeah that's what im thinking of doing, but they don't need a second line to xfer calls though? if you have someone holding on line 1 ? |
| 16:15.39 | [TK]D-Fender | Katty: No rep... I just call it in with whoever answers. different people, same happy results |
| 16:16.05 | wcselby | argh! |
| 16:16.07 | wcselby | stupid computer |
| 16:16.21 | geneticx | yeah I gotta say I like the price of the Polycom 320 from telephonydepot |
| 16:16.38 | wcselby | won't let me download firefox, gets to 90% then Web Marshal blocks the download saying it's pornographic content |
| 16:16.51 | p3nguin | geneticx: Depends on how you handle it. You can "park" calls rather than putting them on regular "hold." |
| 16:16.52 | [TK]D-Fender | geneticx: 321/331. avoid the 320/330 |
| 16:17.24 | [TK]D-Fender | geneticx: And no, either can transfer jsut fine |
| 16:17.38 | [TK]D-Fender | geneticx: No need for parking, and your understanding of "lines" needs to be adjusted |
| 16:18.13 | p3nguin | How can you put "line 1" on hold and make another call if there's only a 1-line phone being used? |
| 16:18.24 | p3nguin | I'm interested in this. |
| 16:18.27 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:19.08 | p3nguin | For transferring, sure one line is probably enough. |
| 16:19.37 | p3nguin | Although I hate DTMF transfers, they are effective. |
| 16:20.50 | geneticx | [TK]D-Fender: would you care to adjust it? |
| 16:21.08 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 16:22.17 | [TK]D-Fender | geneticx: Lines = unique registrations. usually most peoploe use a signle registration per phone. This does not directly relate to the # of simultaneous calls it si cabale of handling |
| 16:23.03 | [TK]D-Fender | geneticx: Ploycom's use "line keys" which can be assigned to different registrations. Each can aslo support a varying # of calls before spanning to the next key even. |
| 16:23.54 | [TK]D-Fender | geneticx: An older IP 601 can have up to 6 regs on the base phone itself. Each line key is also capable of handling up to 8 calls meaning you could shuffle up to 24 on it before adding expansion modules |
| 16:24.28 | [TK]D-Fender | geneticx: On a smaller scale even a lowly IP 301 with 2 "lines" can handle 5 calls each which you could turn into 10 calls for 1 reg |
| 16:26.14 | *** join/#asterisk scalex000 (n=chatzill@158puntacana02.codetel.net.do) |
| 16:26.24 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 16:29.05 | geneticx | [TK]D-Fender: not sure if I follow your last lines... 1 line (unique registration) can handle 5 calls , but then how can I turn it into 10 calls for 1 registration if each can only handle 5? |
| 16:29.34 | verywiseman | what is package i need to run B410P with Asterisk 1.6? |
| 16:31.03 | wcselby | ~pb |
| 16:31.04 | 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 |
| 16:33.31 | eppigy | GIRL |
| 16:33.38 | Katty | hi |
| 16:33.43 | Katty | you called? |
| 16:34.16 | jaytee | TRABAJO |
| 16:34.24 | Katty | hugs jaytee |
| 16:34.30 | *** join/#asterisk garymc (n=garymc@host86-165-105-126.range86-165.btcentralplus.com) |
| 16:34.32 | Katty | jaytee: how are you recovering? |
| 16:34.33 | jaytee | hugs Katty |
| 16:34.47 | jaytee | Katty, you mean over the cat? |
| 16:34.49 | Katty | nods |
| 16:35.41 | jaytee | I'm doing ok. miss him alot but at least i know he's not in any pain anymore |
| 16:36.10 | [TK]D-Fender | geneticx: IP 301 = 2 line keys. each line key can shuffle 5 calls. you can have 1 reg span those 2 line keys. therefor you could configure it to handle 5 on the first, and spill to the 2nd for 5 more |
| 16:36.16 | [TK]D-Fender | geneticx: total = 10 |
| 16:36.16 | beek | Hello Katty , jaytee [TK]D-Fender |
| 16:36.25 | jaytee | afternoon beek |
| 16:36.26 | [TK]D-Fender | beek: Afternoon... |
| 16:38.16 | verywiseman | what is package i need to run B410P with Asterisk 1.6? |
| 16:39.43 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 16:39.47 | Kobaz | i don;t think i'm ever buying another audiocodes gateway again |
| 16:40.39 | ManxPower-work | Kobaz: excellent choice. |
| 16:40.48 | Kobaz | they are SUCH A PAIN to configure |
| 16:40.57 | Kobaz | every single audiocodes has taken me 2-3 days to configure |
| 16:40.59 | ManxPower-work | I'm an Adtran guy |
| 16:41.13 | [TK]D-Fender | verywiseman: DAHDI |
| 16:41.15 | Kobaz | even though i have a doc, i wrote to myself... on how to configure it to work with asterisk |
| 16:41.15 | Deeewayne | agrees with Kobaz |
| 16:41.29 | Baylink2 | I just use used Adtrans and Zhones on a T port. :-) |
| 16:41.30 | Kobaz | i'll run through the config |
| 16:41.39 | [TK]D-Fender | Kobaz: You should learn how to export the configs... makes mass provisioning almost instant |
| 16:41.47 | Kobaz | and, without fail... something won't work... |
| 16:41.50 | verywiseman | [TK]D-Fender, what about mISDN? |
| 16:41.52 | [TK]D-Fender | Kobaz: Learn once repeat fast |
| 16:41.56 | Kobaz | [TK]D-Fender: but the configs are always different for each gateway |
| 16:42.05 | Deeewayne | Kobaz, they used to take me a couple days to configure, but now I take the same exact steps every time and can configure them sort of quickly |
| 16:42.07 | Kobaz | [TK]D-Fender: like, i could import a config, and it just won't work |
| 16:42.10 | [TK]D-Fender | Kobaz: Not notmally ThAT different |
| 16:42.19 | Kobaz | each box i get has a different firmware it seems |
| 16:42.28 | Kobaz | and need slightly different options.. and it takes all day to figure out what those options are |
| 16:42.53 | Kobaz | i set up this new gateway exactly the same as all the others |
| 16:43.01 | Kobaz | and i keep getting 'can't find endpoint for phone number' |
| 16:43.17 | Kobaz | it's liukekjsdflsjadfjsadfhjsadfjhsdfa, the routing is fscking correct, everything else is correct, just work damnit |
| 16:43.26 | beek | I'm glad I'm not the only one who finds an Audiocodes configuration to be a work in patience. |
| 16:43.47 | Deeewayne | Kobaz, I also had problems configuring them by importing a config. I use cutecom now because it has command history |
| 16:44.06 | Deeewayne | I SAR and burn a lot too |
| 16:44.24 | [TK]D-Fender | beek: No... they ARE cryptic... |
| 16:44.32 | [TK]D-Fender | beek: Mediatrix's are far simpler |
| 16:45.02 | beek | [TK]D-Fender: I have an Audiocodes sitting on the shelf because I can't get the @#$%ing thing configured. |
| 16:45.30 | *** join/#asterisk rizwan (n=u2006231@121.52.144.100) |
| 16:45.40 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 16:45.44 | Deeewayne | beek, what are you trying to configure it for? |
| 16:45.45 | Kobaz | hah |
| 16:46.10 | beek | Deeewayne: All I wanted was a simple four-port gateway for four POTS lines. |
| 16:46.33 | Kobaz | beek: and four days later it still doesnt work |
| 16:46.46 | beek | Kobaz: nope. I gave up after two. |
| 16:46.48 | Kobaz | beek: i think we got shipped a bad gateway |
| 16:46.58 | Kobaz | beek: i can make calls out the audiocodes fxo |
| 16:47.01 | beek | Mine refuses to answer the inbound call. |
| 16:47.05 | Kobaz | but inbound calls.. it doesn't open up the audio path |
| 16:47.19 | Kobaz | it will pass the sip call to asterisk... asterisk picks up |
| 16:47.26 | Kobaz | as far as the fxo in concerned, it's still ringing |
| 16:47.42 | beek | Kobaz -- that's MY problem to a tee. |
| 16:47.47 | Kobaz | really? |
| 16:48.06 | Kobaz | and i have the *exact* same config, as another fxo gateway |
| 16:48.10 | Deeewayne | using MediaPacks ? |
| 16:48.11 | beek | Yes. |
| 16:48.16 | Kobaz | the other box has older firmware though |
| 16:48.25 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 16:48.26 | Kobaz | yeap, mp-114 |
| 16:48.36 | beek | Kobaz: Same model: mp-114 |
| 16:48.43 | geneticx | [TK]D-Fender: Ok makes sense now..thank you |
| 16:48.50 | Kobaz | think it's a bad firmware? |
| 16:48.51 | Deeewayne | I use mp114's w/ 5.4 firmware |
| 16:49.03 | beek | I'm going to check the firmware now. |
| 16:49.16 | Kobaz | i hagve 5.80A on this other box and it works fine |
| 16:49.45 | Deeewayne | there was a bug in 5.6 preventing us from using it |
| 16:49.48 | Kobaz | and i have another 5.00A and i'm having the 'can't find endpoint problem' |
| 16:50.56 | Deeewayne | so you can't get it register w/ asterisk? |
| 16:51.02 | Kobaz | it registers |
| 16:51.27 | Deeewayne | I don't know the cant find endpoint problem |
| 16:51.43 | hardwire | Deeewayne: did you spackle it into a wall? |
| 16:52.17 | Kobaz | the one that can't take inbound calls is 5.60A |
| 16:52.19 | Kobaz | i think o |
| 16:52.20 | Deeewayne | hardwire, no but I wanted to before I found the trick to configuring it |
| 16:52.28 | Kobaz | i think o'mm put 5.80 on it |
| 16:52.36 | Kobaz | beek: what firmware is your bad one? |
| 16:53.08 | Deeewayne | the trick for me was to learn how to configure it correctly once, then repeat on all others ... but you must have a beer in hand while you do it |
| 16:53.24 | beek | Kobaz: I'm checking now.... I don't remember the admin port. |
| 16:53.35 | beek | The box is plugged in and running, just not in use. |
| 16:56.07 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 16:56.08 | Kobaz | ack, what did i do with the firmware |
| 16:56.09 | beek | Kobaz: 5.40A.027.002 |
| 16:56.13 | Kobaz | k |
| 16:56.17 | Kobaz | weird |
| 16:56.27 | Kobaz | Deeewayne: what was the 5.60 problem you had? |
| 16:57.05 | beek | Mine registers fine, signals to Asterisk about an incoming call but then won't actually pick up the call and start the audio stream. |
| 16:57.07 | Deeewayne | it only affected CAMA, so you likely don't have that problem |
| 16:57.19 | Kobaz | ah |
| 16:57.37 | Deeewayne | my firmware: 5.40A.035.005 |
| 16:57.57 | beek | Deeewayne: and that works fine? |
| 16:58.06 | Deeewayne | yes |
| 16:58.32 | Kobaz | i think 5.60 is broken in other ways though |
| 16:58.44 | Kobaz | like this ringing in problem |
| 17:00.30 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 17:01.01 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 17:03.11 | beek | Deeewayne: What kind of beer do I need? Imported or domestic (US)? |
| 17:04.23 | *** join/#asterisk dexteruk (n=dexteruk@hst-4-6.cisbg.com) |
| 17:05.12 | Deeewayne | Imported works best, but I've been successful with Bud in a can |
| 17:05.17 | dexteruk | On asterisk when a sip connection is sent Congestion() it sends a 503 Service Unavailible I need code 34 |
| 17:05.38 | dexteruk | Is it posible to change these cause codes |
| 17:05.39 | Kobaz | dexteruk: SIPAddHeader() |
| 17:06.22 | ManxPower-work | I didn't think there was a 34 SIP cause code. |
| 17:06.56 | [TK]D-Fender | 34 is ISDN congestion |
| 17:07.10 | ManxPower-work | dexteruk: in 1.4 and later, I believe that Congestion() or Hangup() (I don't recall which one) allows you to give it a Q.931 cause code. |
| 17:07.23 | [TK]D-Fender | ManxPower-work: And some UA's will bitch because * gives a TRYING before reporting CONGESTION |
| 17:08.57 | Kobaz | ah hah |
| 17:09.05 | Kobaz | found the audiocodes firmware finally |
| 17:10.34 | ManxPower-work | [TK]D-Fender: it' |
| 17:10.38 | lucasb | Mornin... I have a question about Asterisk's extensions.conf dialplan.... anyone available? |
| 17:10.45 | *** join/#asterisk Wangster (n=johnlang@host-253.epicnet.ca) |
| 17:10.52 | ManxPower-work | [TK]D-Fender: it's still pretty stupid to randomly mix SIP and Q.931 cause codes when talking about it. |
| 17:10.53 | ManxPower-work | ~ask |
| 17:10.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:11.02 | lucasb | ManxPower-work: thanks |
| 17:11.10 | [TK]D-Fender | lucasb: Considering that the dialplan is 95% of *, sure.. |
| 17:11.47 | lucasb | [TK]D-Fender: Just checking to see if anyone was alive ;) |
| 17:11.59 | Wangster | I'm trying to get asterisk to write voicemail files as group rw. Apparently this is supported since 1.4 buy changing the umask at runtime. But I can't find any info on how to change the umask at runtime. Anyone know? |
| 17:12.03 | [TK]D-Fender | grrrrraaaaaahhhhhh barinzzzz |
| 17:12.04 | *** join/#asterisk rps2 (n=rick@adsl-99-74-144-118.dsl.lsan03.sbcglobal.net) |
| 17:12.30 | ManxPower-work | Wangster: "man umask" |
| 17:12.47 | lucasb | ~ask If I set a dialplan to match the incoming CLID and then change that CLID will it break the sequence? For instance: exten => _1NXXNXXXXXX/2505551111,1,Set(CALLERID(num)=2502751111) .... exten => _1NXXNXXXXXX/2505551111,2,Dial(SIP/${EXTEN}@127.0.0.1:42001) |
| 17:12.48 | ManxPower-work | or you can set the directory sgid of the asterisk user |
| 17:13.10 | lucasb | Oh, you don't need ~ask to ask a question |
| 17:13.13 | ManxPower-work | lucasb: try it and see, but I would expect it to. |
| 17:13.29 | lucasb | You expect it to break the sequence? |
| 17:13.38 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:13.41 | ManxPower-work | lucasb: yes. |
| 17:13.42 | rps2 | Greetings, once again. Hopefully, with your help, this will be my last foray into the "why the devil doesn't this work?" problems. |
| 17:13.50 | Wangster | ManxPower-work, i'm all up on umask but that has no effect when invoking asterisk via an init script (i've tried). |
| 17:14.23 | Wangster | ManxPower-work, but I hadn't thought of sgid. That sounds easier. |
| 17:14.24 | p3nguin | lucasb: Once you set the CID num to something else, it's different. |
| 17:14.40 | [TK]D-Fender | lucasb: I'd do a Goto() on the 1st where you match it |
| 17:14.43 | lucasb | So for the second sequence I should set the CLID to the updated one |
| 17:14.55 | [TK]D-Fender | lucAnd I agree with ManxPower-work in that I'd expect it to break |
| 17:15.43 | ManxPower-work | In my experience, the best thing to do for CID matches is match on the CID and use that to Goto the real part of the dialplan you want to process the call in. |
| 17:16.06 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 17:16.17 | pittmodal | hello all, I'm evaluating ways of transitioning my * server away from an external SIP provider and to either 1. reachable through an extension via my company's existing VOIP or 2. by getting a dedicated line and interfacing directly. There's a bunch of documentation on the latter but not the former - can anybody point me in direction to find out more? Google-foo is failing me. |
| 17:16.49 | [TK]D-Fender | pittmodal: What is your "existing company's VOIP" is not an external SIP provider? |
| 17:16.51 | p3nguin | lucasb: On the other hand, if your third line was "exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@127.0.0.1:42001)" then the new caller ID would not be tested against this exten. |
| 17:16.59 | [TK]D-Fender | if* |
| 17:17.32 | [TK]D-Fender | p3nguin: Depends if he already had other prios for each... |
| 17:17.36 | pittmodal | [TK]D-Fender: i meant our internal phone setup, i.e. dail my extension to ring my phone, or the asterisk extension to route to asterisk |
| 17:18.04 | p3nguin | [tk]d-fender: Matching extens and callerid num isn't too hard. Hopefully he can figure it out. |
| 17:18.11 | [TK]D-Fender | pittmodal: Sorry, that still sounds a little mixed up. please rephrase |
| 17:18.28 | [TK]D-Fender | p3nguin: His was a very minor tweak .... |
| 17:21.14 | pittmodal | [TK]D-Fender: well, I'd like to see how I take take an existing phone-jack - like the one that my desk phone uses - and plug it into the asterisk server so that I can reach it like I was calling any other extension. I know there's probably some things I have to learn about our existing setup, but was looking for more information about what hardware/configuration I would need to have it work... |
| 17:21.15 | pittmodal | ...on the asterisk side |
| 17:21.52 | pittmodal | not sure if that's any clearer :\ |
| 17:22.07 | ManxPower-work | pittmodal: So you want to connect an analog phone to the Asterisk server? |
| 17:22.21 | Kobaz | axeterisk |
| 17:22.28 | shido6 | easy enough |
| 17:22.39 | rps2 | Well, I've got mine working...except it doesn't answer incoming calls from the POTS lines. The rest of the lot works including dialing out via the POTS lines (thanks to the help I got here). |
| 17:22.56 | *** join/#asterisk Shinsaku (n=Shinsaku@chello089076140236.chello.pl) |
| 17:23.20 | [TK]D-Fender | pittmodal: You jsut need an FXO interface for *. So baskcailly you want * to act like a phone to your other PBX? |
| 17:23.22 | Wangster | ManxPower-work, bah.. sgid doesn't help the situation. Asterisk still creates the file without group write permission :| |
| 17:23.44 | ManxPower-work | Wangster: try the mailing list |
| 17:23.46 | ManxPower-work | ~mailinglist |
| 17:23.46 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 17:23.49 | [TK]D-Fender | rps2: pastebin your configs, and dialplan |
| 17:24.00 | p3nguin | After hooking my phone to the SPA-3102, the ringer is much more quiet than when it was plugged right into the wall jack. Is there any setting in it that I can use to turn up the ringer in the ATA? |
| 17:24.30 | Wangster | ya, the list. Just thought i'd check here first. |
| 17:24.34 | rps2 | [TK]D-Fender: I think pittmodal wants an FXS port...he wants * to act like a station, not a central office. |
| 17:24.34 | pittmodal | [TK]D-Fender: yeah, basically. I didn't know if it was as simple as getting a FXO since we use VOIP phones |
| 17:24.49 | Wangster | maybe I have to set the umask in "safe_asterisk" ? |
| 17:25.06 | beek | Deeewayne: Could I ask the favor of a copy of one of your working ini files? I'd love to compare yours to mine and see WTF may be my issue. |
| 17:25.11 | [TK]D-Fender | pittmodal: So you want * to be an analog extension of your existing PBX? |
| 17:25.20 | pittmodal | [TK]D-Fender: exactly |
| 17:25.27 | [TK]D-Fender | pittmodal: and the SIP phones you are referring to are connected to * correct? And not your other PBX. |
| 17:25.53 | Wangster | I see that they are setting ulimit in safe_asterisk so umask should work there as well. |
| 17:26.52 | pittmodal | [TK]D-Fender: correct. Currently my setup is reachable externally via a SIP provider (junction networks), but I want to transition away from ther |
| 17:26.58 | pittmodal | s/ther/that/ |
| 17:27.19 | [TK]D-Fender | pittmodal: Does * reg to Junction, or your other PBX? |
| 17:27.43 | pittmodal | [TK]D-Fender: right now to junction, but i want it to use my other PBX |
| 17:28.04 | [TK]D-Fender | pittmodal: Ok, Any reason in particular to keep the old PBX at all? |
| 17:28.17 | Wangster | ManxPower-work, yup, that works. umask in safe_asterisk. There should be a patch to that with the option commented out so it's easy to find. |
| 17:28.25 | pittmodal | [TK]D-Fender: it's running the rest of the companies phones ;) |
| 17:28.54 | [TK]D-Fender | pittmodal: so what is *'s purpose in this big picture? |
| 17:29.16 | Deeewayne | beek, http://pastebin.com/d59a89372 |
| 17:29.41 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:29.41 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:29.50 | pittmodal | [TK]D-Fender: it's going to function as a dictation platform. We do voice rec and I took the existing app_dictate application and ported it so that it feeds the audio to our recognized and returns text to the user via alternate means |
| 17:30.00 | beek | Deeewayne: Thanks! I'm going to do a little comparision here and see if I can figure out WTF my problem may be. |
| 17:31.04 | [TK]D-Fender | pittmodal: OK, so as an "application server". How many lines & phones on your old PBX and is there any strong reason that * can't or is not a desirable complete replacement? |
| 17:31.46 | lucasb | [TK]D-Fender: What do ya think of this: |
| 17:31.47 | lucasb | exten => _1NXXNXXXXXX,1,GotoIf($["${CALLERID(num)}" != "2505429800"]?skip) |
| 17:31.47 | lucasb | exten => _1NXXNXXXXXX,n,Set(CALLERID(num)=2502758833) |
| 17:31.47 | lucasb | exten => _1NXXNXXXXXX,n(skip),Dial(SIP/${EXTEN}@127.0.0.1:42001) |
| 17:32.03 | pittmodal | [TK]D-Fender: other than not wanting to rock the boat and increase the project scope, I don't think there's any reason not to use * |
| 17:32.04 | [TK]D-Fender | lucasb: Much nicer IMO |
| 17:32.21 | lucasb | It covers other scenarios that way... thanks for the help everyone |
| 17:32.24 | [TK]D-Fender | pittmodal: Ok, can you give me the summary of your other PBX... |
| 17:32.30 | Chesther | Is anyone aware of a current appliance along the lines of the Aastra CNX, which appears to be discontinued? |
| 17:34.04 | pittmodal | [TK]D-Fender: unfortunately, I'm waiting for more detailed specs from our IT, but it's a run of the mill small business (around 50 phones) PBX. Not doing anything particularly complicated |
| 17:34.06 | [TK]D-Fender | Chesther: Ever tried Asterisk? I hear its the Shiznit y0! |
| 17:34.22 | [TK]D-Fender | pittmodal: How many lines, and what kind? |
| 17:34.36 | [TK]D-Fender | pittmodal: And are you looking for only a single port into *? |
| 17:34.41 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 17:35.12 | Chesther | [TK]D-Fender: The CNX is/was asterisk based. I'm proposing building my own conference bridge, and the bossman is asking "can we just buy it?" |
| 17:35.13 | pittmodal | [TK]D-Fender: not sure on the lines, and yes - for now - only a single port into * |
| 17:35.46 | Qwell | Chesther: You mean something like switchvox? |
| 17:36.11 | Chesther | Well, I was thinking something that was optimized for conference bridge, rather than general-purpose PBX> |
| 17:36.22 | cusco | hi... |
| 17:36.28 | [TK]D-Fender | pittmodal: Ok, I'd recommend a Sangoma A200d or Digium TDM410P with 1 FXO module |
| 17:37.02 | [TK]D-Fender | pittmodal: Other cheaper options exist but have poorer disconnect supervision, and reduced audio quality which make impact your applications requirements |
| 17:37.02 | Kobaz | beek: hey |
| 17:37.37 | citywok | I'm trying to figure out which revision of the 1.6.0 tree has the bug i'm trying to find, and it works in 1.6.0.10, there is no .11, .12 wont compile, it crashes on compiling chan_sip. .15 has the bug, the calls dont complete properly, still waiting for build 13 to compile to test if it has something to do with build 12's failure to compile |
| 17:38.06 | [TK]D-Fender | citywok: What about 14 & 15? |
| 17:38.16 | [TK]D-Fender | citywok: You know... "current" :) |
| 17:38.20 | citywok | 15 has it. waiting for 13 to compile, if 13 works i'll compile 14 |
| 17:38.28 | cusco | this is what we have to outbound, right? - http://paste.debian.net/48644/ |
| 17:38.42 | cusco | now I would liek to adapt that to transfer a call |
| 17:38.45 | wcselby | astricontest - sounds cool |
| 17:38.45 | cusco | attended transfer |
| 17:39.23 | pittmodal | [TK]D-Fender: fortunately, I'm not sweating a few $100's of dollars. I was already looking at the 410s, but was looking for more information about setup and conf |
| 17:39.33 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 17:39.51 | [TK]D-Fender | cusco: I don't see what that has to do with "transferring". You are jsut dialing out... |
| 17:40.20 | [TK]D-Fender | pittmodal: Basic analog DAHDI interface doezens of guides out there, plus install support from Digium |
| 17:40.27 | [TK]D-Fender | pittmodal: And of we'll be here to help |
| 17:40.27 | cusco | [TK]D-Fender: yes that has nothign to do with it, I would like to hardcode a number to transfer out a call |
| 17:40.30 | citywok | [TK]D-Fender: build 13 works, so it's either build 14 or 15 that broke it, waiting for 14 to compile |
| 17:40.37 | cusco | so instead of using feature *8 |
| 17:40.49 | Kobaz | slaps beek |
| 17:40.50 | cusco | i would like to use atxfer with a hardcoded number |
| 17:41.11 | [TK]D-Fender | cusco: No wat to do this directly. |
| 17:41.12 | beek | looks around to see who hit him. |
| 17:41.13 | [TK]D-Fender | way* |
| 17:41.35 | beek | Kobaz -- sorry 'bout that. |
| 17:41.58 | Kobaz | beek: so i upgraded to 5.8, and incoming calls work now |
| 17:42.06 | *** join/#asterisk Gokee2_Extra (n=foo@24-113-159-168.wavecable.com) |
| 17:42.07 | Kobaz | beek: *but*... disconnect supervision no longer works |
| 17:42.08 | [TK]D-Fender | cusco: You could do some ugly stuff to hijack the call though. Applicationmap used to call a script. the script looks at the device calling it, and looks for the other call it is on and then uses AMI to Hijack the other party |
| 17:42.10 | cusco | would I be able to atxfer a call to 9876123456789 ? |
| 17:42.15 | beek | Really? |
| 17:42.25 | Kobaz | yeah |
| 17:42.27 | beek | Kobaz: Audiocodes giveth and Audiocode taketh away. |
| 17:42.28 | rps2 | When there's a break in the action, I'd like to see if someone can help me get this thing to answer my POTS lines. |
| 17:42.32 | pittmodal | [TK]D-Fender: thanks much for the recommendations. I'll go search around a bit for more info |
| 17:42.38 | cusco | that shounds complicated |
| 17:42.38 | Kobaz | beek: just like asterisk |
| 17:43.06 | [TK]D-Fender | cusco: It is, and its pretty mucha ll you've got if you want to make this a 1-touch solution directly under * |
| 17:43.08 | Kobaz | beek: i can't count how many times i'm upgraded asterisk to get a bug fix, and stuff that was working fine, starts breaking |
| 17:43.32 | beek | Kobaz: I haven't experienced that yet. |
| 17:43.41 | Kobaz | i think we're gonna start checking out freeswitch, and etc |
| 17:43.47 | Kobaz | asterisk just breaks too often :( |
| 17:43.49 | [TK]D-Fender | cusco: If you were using Aastra phones or another that can do in-call DTMF speed-dials, you sould do the *8+ ATXFER # all together. |
| 17:43.55 | cusco | [TK]D-Fender: it could be any sequence of numbers like 1111, as long as it was easy to type in the soft phone |
| 17:44.01 | [TK]D-Fender | cusco: But that isn't * doing the work |
| 17:44.44 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 17:45.28 | cusco | dtmf speed dial... |
| 17:45.55 | cusco | we have different OS's and different softphones even under the same OS, I don't think we can fit speed dial into it |
| 17:46.33 | cusco | I might be a slow learner, but could you point me in the right direction of that applicationmap |
| 17:46.37 | [TK]D-Fender | cusco: Yeah, supporting multiple devices makes this painful. |
| 17:46.52 | *** join/#asterisk seanmh (n=johndoe@67.41.13.74) |
| 17:46.54 | cusco | what is AMI? |
| 17:46.57 | [TK]D-Fender | cusco: Go look at the sample features.conf and there are some more practical examples on the WIKI |
| 17:47.08 | citywok | [TK]D-Fender: build 14 is the first build that breaks |
| 17:47.19 | cusco | I looked ad features.conf thats where I saw *8 |
| 17:47.29 | cusco | Im going to search the wiki |
| 17:47.30 | [TK]D-Fender | cusco: Bottom of the sample... |
| 17:47.32 | citywok | to file a bug, should i do what they ask and crank up to 4 all the logging, and submit a log with a call on build 13, and a call on build 14? what else do i need to submit it? |
| 17:47.40 | Gokee2_Extra | Hello all, I am configuring a Digium TDM410P with two FXO ports on channel one and two. I have got zapta working (I think) so now I am working on zapata.conf. I have been reading the 2nd edition of "Asterisk the future of telephony" and it has a sample config file for zapata.conf however the auto-generated zapata-channels.conf wants me to link to it in zapata.conf. If I do this linking what should I configure in zapata.conf? zapat |
| 17:47.40 | Gokee2_Extra | a-channels.conf has both lines going to context from-pstn I (think) I want the lines to go to separate contexts so I can do different things with the lines. |
| 17:48.52 | [TK]D-Fender | Gokee2_Extra: just absorb them into zapata.conf and pastebin it when you're done stripping all comments from your file first |
| 17:49.25 | citywok | and i'm not sure what section i should report this under, there are a bunch of chan_sip categories. i'm guessing interopability maybe? |
| 17:50.37 | Gokee2_Extra | [TK]D-Fender, Ok |
| 17:52.28 | Gokee2_Extra | Ok here http://pastebin.com/d56d73709 is what I have |
| 17:53.26 | cusco | ok in features.conf I can define the dtmf shortcut to call that app. now how do I create that app that will transfer to a hardcoded number? |
| 17:53.35 | [TK]D-Fender | Gokee2_Extra: Looks good... except you specified "channel => 1" twice instead of doing 2 |
| 17:54.16 | [TK]D-Fender | Gokee2_Extra: Not while you might want to treat incoming calls in separate contexts who might you want to use these lines when dialing out? |
| 17:54.19 | Gokee2_Extra | [TK]D-Fender, Ah right thanks |
| 17:54.51 | Gokee2_Extra | [TK]D-Fender, I want all users to have the choice of line one or two |
| 17:55.23 | [TK]D-Fender | cusco: THAT is the complicated part where you'll haev to probably do this in an AGI to grab the channel name of the spawned call to grab the device that called it and then hunt down the other call they are on, and then issue an AMI redirect or similar |
| 17:55.30 | [TK]D-Fender | cusco: which WILL be a PITA |
| 17:55.54 | cusco | ok I will get some help form my boss on that I guess |
| 17:55.55 | cusco | brb |
| 17:56.03 | [TK]D-Fender | cusco: then again.. doing this as a DTMF triggered feature was gong to be pain anyways. |
| 18:17.33 | Gokee2_Extra | anyone happen to know how to get a set of sample configuration files from Debian? |
| 18:18.39 | Gokee2_Extra | Nevermind I found it in asterisk-config |
| 18:18.44 | [TK]D-Fender | Gokee2_Extra: ok, ounds like you don't need to group your lines if you intend for the selection to be deliberate |
| 18:19.16 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:19.38 | ariel_ | afternoon everyone |
| 18:20.44 | Gokee2_Extra | [TK]D-Fender, Hmm the genzaptel thing did that automatically for me. Should I take it out of the file it tells me not to hand edit or is it ok there? |
| 18:21.13 | [TK]D-Fender | Gokee2_Extra: IMO, never run the gen script more than once |
| 18:21.27 | [TK]D-Fender | Gokee2_Extra: them absorb it and customize it yourself |
| 18:23.00 | *** join/#asterisk docelmo (n=chatzill@67-129-111-62.dia.static.qwest.net) |
| 18:23.04 | docelmo | Say can anyone tell me why an originate action from AMI wouldnt automatically kick off the call in asterisk even those the response was a success? |
| 18:23.56 | Gokee2_Extra | [TK]D-Fender, Ah I see, so would you get rid of zapata-channels.conf altogether then? |
| 18:24.44 | [TK]D-Fender | Gokee2_Extra: Yes |
| 18:24.55 | docelmo | TK any ideas? |
| 18:25.36 | rps2 | Stupid question. Does * skip non-existent steps in a context? E.g. if I have "s,1,blah" followed by "s,3,blahblah", does it go from 1 to 3 or will it choke without a step 2 in there? |
| 18:26.20 | [TK]D-Fender | rps2: No, your call will drop like a rock. |
| 18:26.29 | rps2 | Lovely. |
| 18:26.43 | rps2 | Ok, that explains part of my "why won't it answer the bloody phone". |
| 18:26.47 | [TK]D-Fender | docelmo: Nope |
| 18:26.52 | [TK]D-Fender | rps2: :) |
| 18:26.59 | ManxPower-work | rps2: people seldom used number'd priorities these days |
| 18:27.12 | [TK]D-Fender | always does so exclusively |
| 18:27.12 | ManxPower-work | (except for priority 1, of course) |
| 18:27.13 | docelmo | rps2: you can start with exten => 1,1,App then from there use 1,n,app |
| 18:27.21 | docelmo | the N will automatically number your priorities |
| 18:27.24 | ManxPower-work | [TK]D-Fender: Yeah, but you're weird. 8-) |
| 18:27.37 | rps2 | Manx: I'm aware of that, but the GUI seemed to put in two "s,1"s, then an "s,3" and it woudln't answer. |
| 18:27.43 | [TK]D-Fender | ManxPower-work: I'm also always right ;) |
| 18:27.54 | *** part/#asterisk pittmodal (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 18:28.10 | ManxPower-work | rps2: that sounds like a prettive massive bug for a GUI |
| 18:28.33 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp3-139.adsl.forthnet.gr) |
| 18:28.51 | rps2 | So, I have it answering the POTS lines by doing an "exten = s,1,Goto(default,6099,1)" and going to my extension, but my mic doesn't seem to be alive. |
| 18:29.14 | rps2 | I can answer, but nothing I say goes back to the other line. |
| 18:29.18 | superbeef | anybody have problems with IAX trunks droppiong between 1.2 and 1.4 boxes? |
| 18:30.20 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:30.24 | rps2 | s/line/end |
| 18:30.28 | *** join/#asterisk Moltar (n=Moltar@2620:0:d20:1:21b:63ff:fec8:db30) |
| 18:30.46 | citywok | [TK]D-Fender: it is a bug that they had alreayd fixed about 2 weeks ago with a patch https://issues.asterisk.org/view.php?id=16049 |
| 18:32.19 | Gokee2_Extra | Ok, I changed zapata.conf not to link to zapata-channels.conf at all and made a extensions.conf with echo for line two then called line two but asterisk did not pickup. zapata.conf http://pastebin.com/m69d83b9c Extensions.conf http://pastebin.com/d1ee7fe03 any idea why? |
| 18:32.42 | *** part/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 18:33.38 | *** join/#asterisk jeremyb (i=d8fe4bf3@gateway/web/freenode/x-lmbhvalbvymmyasj) |
| 18:34.04 | *** join/#asterisk pyite_mac (n=dschreib@unaffiliated/pyite) |
| 18:36.09 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:36.22 | [TK]D-Fender | rps2: line_one <-- you didn't make this context |
| 18:36.46 | [TK]D-Fender | citywok: Excellent :) |
| 18:36.59 | [TK]D-Fender | Gokee2_Extra: [ line_one] <-- you didn't make this context |
| 18:37.01 | [TK]D-Fender | rather |
| 18:37.57 | Gokee2_Extra | [TK]D-Fender, Do I need to make that one to make line two work? |
| 18:38.40 | [TK]D-Fender | Gokee2_Extra: Nope. PB the failed calla ttempt |
| 18:39.19 | Gokee2_Extra | [TK]D-Fender, PB? |
| 18:39.25 | [TK]D-Fender | ~pb |
| 18:39.26 | 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 |
| 18:40.00 | Gokee2_Extra | [TK]D-Fender, What do I paste? I did not see anything at all about my call from asterisk |
| 18:40.10 | wcselby | jaytee, you around? |
| 18:40.24 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 18:40.40 | [TK]D-Fender | Gokee2_Extra: at CLI do "core set debug 10" , "core set verbose 10", and call again |
| 18:41.02 | [TK]D-Fender | Gokee2_Extra: PB : "zap show channels" , " zap show status" |
| 18:42.37 | *** join/#asterisk titoyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 18:43.47 | titoyz | hi |
| 18:43.58 | *** part/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 18:44.06 | Gokee2_Extra | It also has a bunch of junk on startup http://pastebin.com/d774e744 And the two zap commands show only the dummy channel http://pastebin.com/d48d0f1df |
| 18:44.47 | jaytee | wcselby, I'm working on another computer in another room right now fixing a registry problem. Can ya wait about 15min? |
| 18:45.47 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 18:46.15 | wcselby | jaytee - yeah no worries, I was just wondering what that site was you posted debian examples to was |
| 18:46.15 | SuPrSluG | Gokee2_Extra: Where's context line_one in extensions.conf? |
| 18:46.42 | p3nguin | After hooking my phone to the SPA-3102, the ringer is much more quiet than when the phone was plugged right into the wall jack. Is there any setting in the ATA that I can use to turn up the ringer for the FXS port? |
| 18:46.47 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:46.58 | [TK]D-Fender | Gokee2_Extra: do "module reload chan_zap.so" and repeat |
| 18:47.17 | [TK]D-Fender | SuPrSluG: Already commented on that, don't worry |
| 18:47.20 | Gokee2_Extra | SuPrSluG, I went ahead and stuck it before line_two with another echo http://pastebin.com/d75323c3f |
| 18:47.32 | SuPrSluG | k |
| 18:48.04 | SuPrSluG | what's the output from ztcfg -vv ? |
| 18:48.30 | [TK]D-Fender | Gokee2_Extra: Acutally, stop * completely and do "ztcfg -vvvv" first, then restart & test |
| 18:48.44 | *** join/#asterisk jtodd (i=fh8yivhn@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 18:48.44 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:49.12 | Gokee2_Extra | [TK]D-Fender, The reload of chan_zap gave no output and everything looks the same |
| 18:49.39 | [TK]D-Fender | Gokee2_Extra: Umm... you never did show your /etc/zaptel.conf |
| 18:49.53 | *** join/#asterisk Shinsaku (n=Shinsaku@chello089076140236.chello.pl) |
| 18:49.53 | SuPrSluG | [TK]D-Fender: got LLDP working on polycom phones, auto vlan discovery. :-D |
| 18:49.57 | [TK]D-Fender | Gokee2_Extra: so go do that + my previous request |
| 18:49.57 | Gokee2_Extra | SuPrSluG, It shows two channels http://pastebin.com/de8f6c20 |
| 18:50.21 | wcselby | SuPrSluG - you got LLDP to work on polycoms? how? |
| 18:50.34 | [TK]D-Fender | SuPrSluG: I see your VLAN and raise you "I have my own damn physical subnet" :p |
| 18:50.39 | SuPrSluG | you need 3.2 firmware |
| 18:50.51 | SuPrSluG | show off |
| 18:50.51 | wcselby | ugh, of course you do. which the ip601 is not supported on |
| 18:50.56 | Gokee2_Extra | Zaptel.conf http://pastebin.com/d2a1be59b |
| 18:51.12 | SuPrSluG | no. unfortunately. |
| 18:51.29 | Katty | it really is shocking how many preservatives and additives they put into commercial food products. |
| 18:51.46 | [TK]D-Fender | Gokee2_Extra: Restart * and redo : "zap show channels" , " zap show status" |
| 18:52.34 | Gokee2_Extra | [TK]D-Fender, Ok, I did that and it still only shows the pseudo interface |
| 18:52.42 | SuPrSluG | Katty: only buy from the store perimeter. the stuff in the middle is toxic |
| 18:53.01 | [TK]D-Fender | Gokee2_Extra: Still? |
| 18:53.10 | Gokee2_Extra | [TK]D-Fender, Ya. still.... |
| 18:53.20 | [TK]D-Fender | Gokee2_Extra: Pastebin the results of "module reload chan_zap.so" |
| 18:53.25 | Katty | SuPrSluG: i was looking at the ingredient list of things like Crackers. |
| 18:53.38 | wcselby | SuPrSluG - is it just enabled by default in 3.2? also, what is needed to download 3.2 (support contract? etc) |
| 18:53.43 | Gokee2_Extra | [TK]D-Fender, That command outputs nothing |
| 18:53.54 | SuPrSluG | wcselby:yes |
| 18:54.04 | [TK]D-Fender | Gokee2_Extra: do unload first, then load |
| 18:54.20 | [TK]D-Fender | Gokee2_Extra: What ver of * exactly? |
| 18:54.24 | SuPrSluG | I think they have 3.2.1 on the public download site |
| 18:55.11 | ManxPower-work | polycom 3.2 is not available for general release yet. |
| 18:55.11 | SuPrSluG | yes it's there |
| 18:55.17 | ManxPower-work | At least it wasn't 2 days ago |
| 18:55.27 | SuPrSluG | http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip450.html?product_type=2&category=%2Fsupport%2Fvoice%2Fsoundpoint_ip%2Fsoundpoint_ip450.html%23document#document |
| 18:55.33 | wcselby | 3.2.1 is available, thanks SuPrSluG :) |
| 18:55.38 | [TK]D-Fender | ManxPower-work: Yes, it was... |
| 18:55.47 | [TK]D-Fender | ManxPower-work: I've had 3.2.1 since Oct 1st |
| 18:55.48 | Gokee2_Extra | [TK]D-Fender, I got "-- Unregistered channel -2" on unload nothing on load and asterisk 1.4.21.2~dfsg-3 |
| 18:55.54 | Katty | SuPrSluG: triscuits have 3 ingredients. Ritz has like 30 |
| 18:56.01 | ManxPower-work | 3.1.2 was there. |
| 18:56.03 | SuPrSluG | wcselby:do you have a switch that supports it |
| 18:56.21 | [TK]D-Fender | ManxPower-work: I've still got the ZIP's on my desktop with little date-stamps :) |
| 18:56.46 | wcselby | SuPrSluG - my main client moved to an all juniper network, which only has lldp, no cdp |
| 18:56.47 | SuPrSluG | SoundPoint IP, SoundStation IP and Polycom VVX SIP 3.2.1 [Split] [RECOMMENDED FOR HDX INTEGRATION WITH SOUNDSTATION IP 7000â SEE RELEASE NOTES] |
| 18:56.51 | wcselby | but their phones were all configured for cdp |
| 18:57.08 | robl^laptop | 3.2 is on the site. I downloaded it the ohter day |
| 18:57.10 | ManxPower-work | [TK]D-Fender: what are those date stamps? |
| 18:57.28 | SuPrSluG | looks like you have some billable time now :-D |
| 18:57.46 | [TK]D-Fender | ManxPower-work: created, modified & accessed on XP |
| 18:58.18 | ManxPower-work | [TK]D-Fender: we've been going back and forth with our polycom rep because we could not access 3.2.x |
| 18:58.20 | SuPrSluG | Katty: and Tiscuits rock |
| 18:58.30 | [TK]D-Fender | Gokee2_Extra: unload, try zap show channels" , " zap show status". then load. Repeat. Pastebin it all |
| 18:58.37 | wcselby | SuPrSluG indeed! :) we've actually been deploying some Cisco 7961's with LLDP support, but the lack of BLF / Presence on those has been disturbing to some users |
| 18:58.38 | SuPrSluG | Katty: my fav cracker |
| 18:58.38 | *** join/#asterisk sahX (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:58.46 | robl^laptop | ManxPower-work: I have 3.2.1 and got it from polycom's public site 3 days ago |
| 18:58.54 | [TK]D-Fender | ManxPower-work: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 18:59.01 | [TK]D-Fender | maxnlinks work fine |
| 18:59.53 | ManxPower-work | [TK]D-Fender: it's there NOW, it was not last week |
| 18:59.53 | [TK]D-Fender | ManxPower-work: thats where I got it from on Oct 1st..... |
| 18:59.53 | SuPrSluG | Katty: Not so good in soup though :-( |
| 18:59.58 | korcan | What causes these messages? |
| 19:00.00 | korcan | WARNING[19137]: udptl.c:807 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files |
| 19:00.00 | korcan | [Oct 9 14:53:02] ERROR[19137]: acl.c:472 ast_ouraddrfor: Cannot create socket |
| 19:00.04 | ManxPower-work | [TK]D-Fender: I'll request to be assigned a non-idiot rep from Polycom. |
| 19:00.10 | Gokee2_Extra | [TK]D-Fender, http://pastebin.com/d13f42704 |
| 19:00.15 | [TK]D-Fender | ManxPower-work: Wise :) |
| 19:00.53 | SuPrSluG | Gokee2_Extra: 2 channels to configure. shouldn't it say configure(d) ? |
| 19:01.10 | ManxPower-work | [TK]D-Fender: and you don't have an account on the PRC? |
| 19:01.18 | [TK]D-Fender | Gokee2_Extra: PB "ls -la /etc/asterisk" and "cat /etc/asterisk/zapata.conf". Adjust paths if necessary |
| 19:01.26 | [TK]D-Fender | ManxPower-work: Don't think so.... |
| 19:01.45 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 19:01.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:01.51 | [TK]D-Fender | ManxPower-work: I'm not up to date since the start of this year... my office is on all disco'd phones except 2 x IP 320's |
| 19:01.52 | wcselby | can anyone list the major diff's between polycom 650 and 670? |
| 19:02.01 | robl^laptop | wcselby: color |
| 19:02.01 | [TK]D-Fender | wcselby: polycom.com can. |
| 19:02.01 | wcselby | or point me to a list of differences? |
| 19:02.11 | [TK]D-Fender | wcselby: And I'll bet they are even obvious |
| 19:02.31 | wcselby | [TK]D-Fender - you're probably right, I just now started looking |
| 19:02.44 | Gokee2_Extra | [TK]D-Fender, http://pastebin.com/d290d787f |
| 19:03.13 | Gokee2_Extra | Does zapata.conf want a ending newline? |
| 19:03.40 | kfife | Anybody know if the new polycom firmware ILBC support includes wideband? |
| 19:03.45 | [TK]D-Fender | Gokee2_Extra: shouldn't impact 31 from showing up |
| 19:04.00 | Gokee2_Extra | ok |
| 19:04.19 | [TK]D-Fender | #1 |
| 19:05.03 | SuPrSluG | kfife: it's supports 13 and 15 k , so i don't think so. |
| 19:05.58 | ManxPower-work | WoW!! 26754: SoundPoint IP 320,321,330,331,450, 550, 560, 650, 670: Add support for |
| 19:05.58 | ManxPower-work | the iLBC codec in 3.2.x |
| 19:07.26 | Gokee2_Extra | [TK]D-Fender, Well I am gonna head down to the location. If you think of something I will probably be back in a few hours. My server is currently down so I can't stay connected. Thanks for all your help |
| 19:08.06 | [TK]D-Fender | ManxPower-work: Yup.... |
| 19:08.17 | kfife | SuPrSluG: thanks |
| 19:08.18 | kfife | http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_vvx_3_2_1_relnotes.pdf |
| 19:09.53 | [TK]D-Fender | ManxPower-work: IMO they should ahve added GSM 6.10 as well |
| 19:09.53 | [TK]D-Fender | *b00m* |
| 19:09.53 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 19:09.53 | SuPrSluG | usually it's a pretty boring read, this time it had a lot of nice suprises |
| 19:09.53 | [TK]D-Fender | Imagine how much more they could fit on once they trash the WEB config GUI :) |
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| 19:11.02 | *** mode/#asterisk [+ooo putnopvut russellb file] by irc.freenode.net |
| 19:11.04 | kfife | *unb00m* |
| 19:11.34 | kfife | [TK]D-Fender: An interface design masterpiece! |
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| 19:12.54 | kfife | I think that interface was someone's programming 101 "final exam" project |
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| 19:17.01 | SuPrSluG | http://www.hulu.com/labs/hulu-desktop-linux |
| 19:19.05 | [TK]D-Fender | finds that guy and FLUNKS HIM |
| 19:19.33 | jaytee | wcselby, you still here? |
| 19:20.34 | jaytee | Adobe Flash 10 on Windows is the suck! |
| 19:24.06 | Katty | pouts at the time |
| 19:25.35 | SuPrSluG | Katty: needs to read http://www.hplusmagazine.com/articles/neuro/perils-fds-fun-deficiency-syndrome |
| 19:28.06 | Katty | hmm. |
| 19:29.19 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 19:31.06 | dustybin | doesnt hug Katty |
| 19:31.55 | raden_work | afternoon Katty |
| 19:32.00 | raden_work | Naikrovek, how goes it bro |
| 19:32.56 | wcselby | jaytee - i'm here |
| 19:33.06 | wcselby | what was that site you were posting debian asterisk examples to? |
| 19:33.14 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 19:33.22 | jaytee | the site I referenced? it was www.voip-info.org |
| 19:33.30 | wcselby | no, maybe it wasn't you |
| 19:33.32 | Katty | hugs raden_work |
| 19:33.45 | wcselby | i thought you were posting config examples on an ubuntu / debian site |
| 19:33.47 | raden_work | gives katty a big hug |
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| 19:34.42 | Katty | wcselby: i have some debian/asterisk/3rd party examples on my blog |
| 19:34.47 | Katty | wcselby: maybe you were thinking of that? |
| 19:35.11 | ZPertee | does anyone have a suggestion on a cheap ata for a poor college student like myself who wants to play around with asterisk? |
| 19:35.16 | wcselby | Katty - I don't really think so, but maybe |
| 19:35.23 | Katty | k |
| 19:37.12 | [TK]D-Fender | Katty: I whole-heartedly agree.... |
| 19:37.19 | [TK]D-Fender | Katty: You need to read that article :) |
| 19:37.25 | Katty | raden_work: i did some gforce calculations on that ion rocket. |
| 19:37.33 | [TK]D-Fender | ZPertee: Linksys PAP2T |
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| 19:37.47 | Katty | raden_work: they really don't apply tho, because there's no gravity involved in space. |
| 19:37.51 | ZPertee | [TK]D-Fender, k. thanx |
| 19:38.13 | Katty | raden_work: there would only be gforce if you take the time it takes to get from ground lvl to the upper atmosphere. |
| 19:38.30 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 19:38.49 | *** join/#asterisk circut (n=circut@c-71-57-110-244.hsd1.il.comcast.net) |
| 19:38.56 | circut | hey all, got a bit of an issue here |
| 19:39.11 | Katty | raden_work: basically they'd hit 11km/sec to get out of earth's pull, and 8km/sec to get into orbit |
| 19:39.26 | Katty | raden_work: if they're smart, they will dock the rocket at say..the space station |
| 19:39.39 | circut | my users have POTS lines for their fax machines. Now sending faxes locally works just fine |
| 19:39.45 | Katty | raden_work: so they don't have to waist fuel on escape veloctiy, especially if they're going on a 39 day trip to mars :/ |
| 19:40.00 | circut | but when they try to fax something via long distance they cannot, because they need to enter a special 3-digit code to dial out |
| 19:40.08 | Katty | raden_work: there are other serious draw backs about the ability of a solar panel to keep that size of a craft powered once a reasonable amount of fuel is used. |
| 19:40.31 | raden_work | Katty, interesting you would still have a gforce from the weight of your body accelerating wouldn't you |
| 19:40.51 | *** part/#asterisk Baylink2 (n=jra@cerberus.vicimarketing.com) |
| 19:40.51 | circut | is there a way for me to enter this code via asterisk? |
| 19:41.10 | Katty | raden_work: for initial takeof you would |
| 19:41.15 | circut | im not too sure about it, since you need to dial the number, wait for a special tone, then enter the long distance code |
| 19:41.40 | Katty | raden_work: but there's no gravity (really) in space |
| 19:41.40 | raden_work | Katty, but even in space your still have weight and acceleration wouldnt that give you g-force ? |
| 19:41.46 | Katty | no |
| 19:41.52 | Katty | for gforce, you must have gravity |
| 19:41.58 | Katty | something must pull your weight |
| 19:42.04 | Katty | for you to have resistance |
| 19:42.20 | Katty | in space, you can poke a 32 ton blob and it will move |
| 19:42.36 | circut | only if your chuck norris |
| 19:42.41 | Katty | :P |
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| 19:43.29 | Katty | the body can't handle much more than 3 or 4 gs. |
| 19:43.44 | Katty | some people, like race car drives, can handle 5gs. |
| 19:44.52 | raden_work | Katty, ah yes very true you are right there needs to be resistance against a force to have gravitational force |
| 19:44.53 | circut | my gf can handle 5gs |
| 19:44.57 | circut | giggity |
| 19:45.01 | raden_work | Katty, sorry i finally got my brain back this morning :) |
| 19:45.32 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 19:45.33 | raden_work | 4 trips t0 my dentist and went to a diffrent one this morning and walla no more pain for me :) |
| 19:45.56 | Katty | nice. |
| 19:46.01 | mchou | raden_work: what did he give you? |
| 19:46.21 | raden_work | mchou, he took out the right tooth |
| 19:46.33 | raden_work | $940 later this week |
| 19:46.40 | raden_work | there went my pay for the week |
| 19:47.03 | mchou | raden_work: haha. the other dentist must have been incompetent |
| 19:47.04 | Katty | damn senate. |
| 19:47.10 | Katty | sighs |
| 19:47.35 | mchou | raden_work: no joke, next time g=take a vaction and get your dental work done in mexico |
| 19:47.47 | Katty | or at least canada. |
| 19:47.48 | raden_work | Katty, first dentist 20 years experience second one 5 years kinda odd really but I think i stick with this other dude now |
| 19:47.55 | ZPertee | what is the difference between a pstn pass thru port and a fxo port? http://www.grandstream.com/products/ht_series/ht486/ht486.html |
| 19:47.58 | Katty | it's hard to find a good dentist. |
| 19:48.02 | mchou | raden_work: better dentists, not to mention cheaper |
| 19:48.04 | Katty | i have a kids dentist. |
| 19:48.13 | Katty | he specializes in Scared Patients. |
| 19:48.14 | raden_work | Katty, i have state insurance i have to tell them i have no insurance to even get in around here otherwise there is waiting lists and all sorts of BS |
| 19:48.23 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 19:48.28 | Katty | ugah :< |
| 19:48.52 | raden_work | I have horrible dental anxiety this dude was awesome didnt feel a thing |
| 19:49.01 | Katty | that's excellent. |
| 19:49.17 | Katty | and i know how you feel. i'm thinking next time i go to the denist they just need to put me out |
| 19:50.29 | raden_work | Katty, have them put you on nitrous :) takes the edgy nerves off tell them youd like them to take things slow and you have dental anxiety and also tell them you would like your gums numbed before they actually give you any anesthetic |
| 19:51.09 | Katty | yeah i think i'll just stick with being out. |
| 19:51.17 | raden_work | my dentist visits are enjoyable |
| 19:51.18 | Katty | and not local. |
| 19:51.19 | Katty | i mean /out/ |
| 19:51.23 | [TK]D-Fender | ZPertee: Passthrough does not mean you can control the FXO via SIP. |
| 19:51.29 | raden_work | Katty, ever tried nitrous ? |
| 19:51.32 | mchou | I've got a question regarding IPKall and sip client (softphone). Trying a sanity check for my friend. Is it possible to connect softphone to IPKall directly (ie. not using a sip proxy, itsp, or *? |
| 19:51.33 | Katty | yes. |
| 19:51.39 | Katty | they put me on nitrous, to put me out |
| 19:51.40 | raden_work | it help ? |
| 19:51.47 | raden_work | hehehe |
| 19:51.56 | Katty | i woke up nauseous ;) |
| 19:51.59 | raden_work | you must have alot of anxiety eh ? |
| 19:51.59 | Katty | and the sky was red |
| 19:52.10 | Katty | well i had my wisdom teeth taken out |
| 19:52.17 | Katty | so, at the time, i was pretty out of it |
| 19:52.21 | Katty | and very scared. |
| 19:52.22 | raden_work | Katty, they probally ran it a lil to high when your toes start to tingled need to tell them to back it off |
| 19:52.50 | raden_work | it can make you very sick if too much of it gets in you |
| 19:52.53 | Katty | when i went in for surgery, they gased me before putting me out... |
| 19:52.54 | wcselby | i had four wisdom teeth removed along with two "extra" teeth - I ended up going to a dental surgeon who knocked me out with anesthesia |
| 19:53.01 | wcselby | i don't like going to the dentist |
| 19:53.13 | Katty | dont'think anyone does really :/ |
| 19:53.14 | raden_work | wcselby, i dont think anyone does |
| 19:53.21 | raden_work | Katty, ditto :P |
| 19:53.25 | wcselby | so I usually just don't go |
| 19:53.27 | wcselby | :P |
| 19:53.30 | raden_work | i dont mind it just finding a good one |
| 19:53.33 | Katty | i don't mind going to the dentist for xrays. |
| 19:53.37 | Katty | or cleaning, or whatever. |
| 19:53.39 | wcselby | i had perfect teeth as a kid...unfortunately that didn't last into adulthood |
| 19:53.54 | Katty | it's that acute fear of pain and suffering :P |
| 19:54.21 | raden_work | i should sue coca cola for not putting labels on there cans saying may cause tooth decay |
| 19:54.42 | mchou | raden_work: dude, that's common knowledge |
| 19:54.58 | raden_work | mchou, dude i know i was being sarcastic :P |
| 19:55.05 | mchou | raden_work: all that sugar + carbonic acid==bad teeth |
| 19:55.14 | Katty | carbonic acid? |
| 19:55.14 | Katty | lol |
| 19:55.20 | raden_work | uhuh |
| 19:55.26 | raden_work | stuff is good for cleaning :) |
| 19:55.39 | Katty | citric acid ;) |
| 19:56.06 | Katty | oh wow |
| 19:56.16 | Katty | this diet coke can says it has brominated vegetable oil as an ingredient |
| 19:56.24 | Katty | err diet mt dew |
| 19:56.30 | mchou | Katty: yup |
| 19:56.35 | Katty | creepy |
| 19:57.00 | mchou | that's Mt. Dew by definition |
| 19:57.05 | Katty | diet soda has several fatal flaws. |
| 19:57.07 | mchou | CREEPY |
| 19:57.09 | SuPrSluG | how does one brominate vegatable oil? |
| 19:57.17 | Katty | the worst, in my opinion, is that it makes you hungry. |
| 19:57.35 | SuPrSluG | is that a black thing? |
| 19:57.40 | Katty | SuPrSluG: bromine? |
| 19:57.47 | Katty | it's a member of the halogen family |
| 19:57.56 | Katty | kind of a dark brown liquidy element |
| 19:58.02 | mchou | SuPrSluG: is what a black thing? |
| 19:58.07 | [TK]D-Fender | .... |
| 19:58.10 | [TK]D-Fender | halogen is a GAS |
| 19:58.26 | Katty | yes. it is. |
| 19:58.31 | *** part/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 19:58.41 | SuPrSluG | american slang |
| 19:58.48 | *** join/#asterisk MrSeb (n=sebax75@87.253.113.240) |
| 19:58.55 | mchou | http://en.wikipedia.org/wiki/Brominated_vegetable_oil |
| 19:59.01 | mchou | read all about it |
| 19:59.10 | MrSeb | hi |
| 19:59.26 | Katty | it's nice to know i'm consuming halogens. |
| 19:59.37 | Katty | or at least, partially bonded bits of halogens. |
| 19:59.40 | mchou | lol |
| 19:59.50 | Chesther | Wow. Messes with thyroid function. |
| 19:59.52 | mchou | that wikipedia article is full of good news |
| 19:59.53 | Chesther | Good stuff. |
| 20:00.00 | *** join/#asterisk TheCompWiz (n=tmealey@wsip-68-109-200-102.mc.at.cox.net) |
| 20:00.07 | Katty | oh boy |
| 20:00.11 | Chesther | will stick to his coffee, kthnx. |
| 20:00.12 | Katty | depresses thyroid function |
| 20:00.20 | TheCompWiz | anyone know why rxgain would be ignored in chan_dahdi.conf? |
| 20:00.39 | mchou | Katty: get of Mt. Dew and you wont ever need to go to the doctor again |
| 20:00.43 | Katty | associated with brain damage, depression, memory loss, hallucinations, violent tendencies, seizures, cerebral atrophy, acute irritability, tremers... |
| 20:00.56 | *** join/#asterisk xpot-mobile (n=james@mx0.synergyconsultant.net) |
| 20:00.56 | mchou | Katty: exactly |
| 20:01.10 | Katty | let's use Neon in our dt mt dew instead! |
| 20:01.19 | [TK]D-Fender | TheCompWiz: Put it where it belongs and it won't be |
| 20:01.22 | Katty | maybe we'll glow! |
| 20:01.29 | MrSeb | someone can explain to me if is possible, having already a dialplan, define a new one, using the previous and adding only a number of preselection without rewrite all? Pratically, from one provider, I've added a new one, and this is for choose the provider to use |
| 20:01.36 | TheCompWiz | [TK]D-Fender: where does it belong if not chan_dahdi.conf? |
| 20:01.49 | Katty | "BVO has caused testicular damaged" |
| 20:02.08 | [TK]D-Fender | TheCompWiz: Show us what you've done. |
| 20:02.10 | raden_work | Katty, how u so smart ? |
| 20:02.22 | Katty | i'm not |
| 20:02.25 | Katty | i just like to read wikipedia. |
| 20:02.40 | Katty | BVO is banned int he use of soft drinks in India |
| 20:03.06 | TheCompWiz | [TK]D-Fender: took the "example" config... adjusted to match my trunk type... everything works as far as making calls & such... whatever I set rxgain to... the dahdi_monitor shows no changes. |
| 20:03.30 | [TK]D-Fender | TheCompWiz: and I asked to see backup.... |
| 20:03.36 | TheCompWiz | i.e. rxgain=0.0 and rxgain=20.0 are the same. |
| 20:03.38 | raden_work | wow |
| 20:04.08 | Katty | oh this is awesome! |
| 20:04.19 | Katty | aspartame breaks down into formaldehyde in the body |
| 20:05.43 | *** join/#asterisk brookshire (i=mbrooks@65.172.243.127) |
| 20:05.48 | circut | ggrrr |
| 20:05.56 | circut | why dont sending the DTMF tones work ... |
| 20:05.57 | TheCompWiz | [TK]D-Fender: just to make you happy: http://pastebin.com/d1e9d0b5b |
| 20:06.04 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 20:06.32 | [TK]D-Fender | TheCompWiz: Everything after channel => 1-10 gets IGNORED |
| 20:06.43 | TheCompWiz | why is that? |
| 20:06.51 | circut | [TK]D-Fender: hey dude long time no see |
| 20:06.59 | [TK]D-Fender | TheCompWiz: because thats how zaptel/DAHDI works. |
| 20:07.14 | [TK]D-Fender | TheCompWiz: You define your channel and the settings BEFORE IT apply |
| 20:07.20 | [TK]D-Fender | TheCompWiz: Welcome to Zaptel 101 |
| 20:07.34 | [TK]D-Fender | [16:01]<[TK]D-Fender>TheCompWiz: Put it where it belongs and it won't be |
| 20:07.35 | Katty | raden_work: In one case, a man who drank eight liters of Ruby Red Squirt daily had a reaction that caused his skin color to turn red and produced lesions diagnosed as bromoderma. <- from wikipedia. |
| 20:07.56 | Katty | 8 liter a day is a lot. |
| 20:08.30 | [TK]D-Fender | Katty: Cool... how many days did it take? |
| 20:09.13 | circut | [TK]D-Fender: you familiar with the following command?: Dial(DAHDI/g0/1234,,D(w4567) |
| 20:09.25 | Katty | [TK]D-Fender: the article says "Several Months" |
| 20:09.31 | circut | ive got that in my dialplan, but it doesnt appear to be waiting, or sending the digits |
| 20:09.39 | Katty | [TK]D-Fender: wikipedia references: http://content.nejm.org/cgi/content/short/348/19/1932 in footnote |
| 20:09.44 | [TK]D-Fender | Katty: 8 liters a day for several months. Holy shit |
| 20:10.10 | [TK]D-Fender | circut: and I don't see your actual dialplan or the call attempt. PASTEBIN is your friend |
| 20:10.12 | Katty | i'm surprised he didn't have other more serious problems |
| 20:10.13 | [TK]D-Fender | ~pb |
| 20:10.14 | 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 |
| 20:10.14 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:10.24 | Katty | more serious than bromoderma |
| 20:10.41 | circut | one minute |
| 20:10.53 | circut | it seems to be sending the dtmf codes |
| 20:10.58 | circut | but not for the call i need LOL |
| 20:12.20 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 20:13.11 | circut | http://pastebin.ca/1607954 |
| 20:13.25 | circut | its fairly straightforward, my phone is a SIP phone |
| 20:13.47 | circut | calling my cell phone which is in a different area code |
| 20:13.54 | [TK]D-Fender | circut: And the failed call attempt |
| 20:14.00 | circut | so i dial my cell, i hear the beep to enter the long distance code |
| 20:14.27 | circut | [TK]D-Fender: ? the call fails yes |
| 20:14.36 | [TK]D-Fender | circut: WHERE IS IT? |
| 20:14.55 | circut | after a couple seconds of hearing the beep i get: "36P4 Im sorry you have not dial enough digits for your call to be completed" |
| 20:15.05 | circut | i dont follow |
| 20:15.26 | [TK]D-Fender | [16:10]<[TK]D-Fender>circut: and I don't see your actual dialplan or the call attempt. PASTEBIN is your friend |
| 20:15.35 | [TK]D-Fender | circut: PASTEBIN the CLI OUTPUT of your call attempt |
| 20:15.40 | circut | ooh |
| 20:16.43 | *** join/#asterisk gardo (n=gardo@121.97.212.52) |
| 20:18.31 | circut | http://pastebin.ca/1607967 |
| 20:18.33 | circut | sorry abou tthat |
| 20:19.42 | circut | so its weired, because on legit outgoing calls i see it inserting the DTMF |
| 20:19.53 | *** part/#asterisk MrSeb (n=sebax75@87.253.113.240) |
| 20:20.19 | [TK]D-Fender | circut: What kind of DAHDI channel is that? |
| 20:20.20 | circut | the difference is this: |
| 20:21.05 | [TK]D-Fender | circut: And please show the EXACT output, and do not filter anything |
| 20:21.40 | circut | what i pasted you is exactly what i see |
| 20:21.57 | circut | on other calls i see this though |
| 20:21.58 | circut | <PROTECTED> |
| 20:21.58 | circut | <PROTECTED> |
| 20:22.47 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 20:23.57 | [TK]D-Fender | circut:I want to see the 2 complete call attempts masking nothing |
| 20:24.26 | [TK]D-Fender | circut: And jsut a side note for the PB you DID give me... they didn't answer <- |
| 20:24.44 | [TK]D-Fender | circut: So certainly no reason to enter digits |
| 20:25.14 | *** part/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 20:25.49 | circut | ok, so there is no answer |
| 20:25.56 | circut | but there is a tone that is sent across |
| 20:25.59 | circut | like a low beep |
| 20:26.08 | [TK]D-Fender | circut: Not answered. |
| 20:26.09 | circut | ive tryied putting our long distance code in front of the number to be dialed |
| 20:26.22 | circut | with a # to send it through |
| 20:26.47 | circut | but no go ;/ |
| 20:27.44 | circut | erm |
| 20:27.47 | circut | in front / behind |
| 20:28.22 | Katty | slings rubberband at [TK]D-Fender |
| 20:28.33 | TheCompWiz | [TK]D-Fender: thanks... I've never previously heard that the settings prior to the channel => was what applied. makes a lot more sense now. /bow |
| 20:29.45 | [TK]D-Fender | staples it to a 2x4 nails a clothes-peg to it, hooks another nail into the elastic, clips the head in the peg takes aim and FIRES |
| 20:30.52 | TheCompWiz | on a side note... is it uncommon to see rx = 0.7 & tx 9.2? ... seems like a huge difference between the two. |
| 20:31.06 | [TK]D-Fender | checkout time, later all |
| 20:31.10 | TheCompWiz | (just testing... it sounds quite a bit better...) |
| 20:32.37 | dustybin | slaps Katty with a noble peace prize |
| 20:34.16 | TheCompWiz | doubts Katty wants to associate with those who have recently received the Nobel Peace Prize. |
| 20:39.22 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 20:40.33 | Katty | TheCompWiz: what are you trying to say |
| 20:44.31 | TheCompWiz | just was messin' with my setup... and based on what the monitor tells me... it seems like the rx & tx gains are disproportionate. |
| 20:44.50 | TheCompWiz | I figured if rx was a bit off... tx wouldn't be that far off... but 9db seems like a lot of gain... |
| 20:45.15 | TheCompWiz | but it sounds right. |
| 20:47.14 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 20:48.29 | *** part/#asterisk TheCompWiz (n=tmealey@wsip-68-109-200-102.mc.at.cox.net) |
| 21:01.15 | fofware | hello, I want try conference in Asterisk 1.6, but MeetMe is not registered, that is something I forget in compilation time? |
| 21:04.03 | *** join/#asterisk [TK]D-Fender (n=joeblow@161.216.158.61) |
| 21:04.15 | fofware | yes, I see app_meetme depend of dahdi, I need it to enable conference |
| 21:04.49 | *** join/#asterisk Moltar (n=Moltar@2620:0:d20:1:21b:63ff:fec8:db30) |
| 21:04.59 | Katty | wraps [TK]D-Fender in copper and sticks him between two magnets. |
| 21:05.39 | [TK]D-Fender | Superconductor! |
| 21:06.18 | Katty | then we should connect the magnets to wire. |
| 21:06.21 | Katty | and stick the wire in water. |
| 21:06.29 | Katty | and then move to the MOON! |
| 21:07.08 | Katty | oh hey, did nasa launch that bomb yet? |
| 21:09.20 | *** join/#asterisk ergodicsum (n=ergudics@70.158.116.43) |
| 21:09.58 | ergodicsum | does anyone know what is the default text to speech engine installed in trixbox? |
| 21:10.22 | Katty | my guess would be festival. |
| 21:10.27 | Katty | but i would ask in #trixbox |
| 21:10.56 | [TK]D-Fender | errotan: should be festival |
| 21:12.28 | seanbright | Katty: yes they did |
| 21:12.37 | Katty | has found some CNN coverage to watch |
| 21:12.42 | Katty | seanbright: water? |
| 21:12.44 | seanbright | Katty: and it had no explosives on it, so i'm not sure it counts as a 'bomb' |
| 21:12.50 | Katty | okay so rocket |
| 21:12.51 | Katty | going very fast |
| 21:12.55 | Katty | to cause asplosey |
| 21:13.05 | Katty | and hopefully bits of ice to go everywhere? |
| 21:13.17 | seanbright | the visible plume was underwhelming |
| 21:13.25 | seanbright | but apparently they consider it successful |
| 21:13.35 | seanbright | nothing in terms of results yet |
| 21:13.39 | Katty | :< |
| 21:13.43 | seanbright | i'd imagine that will take a ridiculously long time |
| 21:13.47 | seanbright | because it's nasa |
| 21:13.50 | seanbright | and space. |
| 21:13.52 | Katty | probably. |
| 21:14.02 | Katty | it's a very expensive project. |
| 21:14.11 | Katty | but the results will be VERY interesting |
| 21:14.21 | seanbright | i'm not so sure. |
| 21:14.25 | seanbright | but |
| 21:14.27 | seanbright | shrugs |
| 21:14.40 | Katty | Bottled Moon Water(tm) |
| 21:16.09 | seanbright | we need to get this mars mission going |
| 21:16.40 | seanbright | i want to live to see it |
| 21:17.20 | Katty | there are still a lot of quirks with that ion engine |
| 21:17.43 | Katty | i'd like to see healthcare mission taken care of first |
| 21:18.40 | seanbright | meh |
| 21:19.47 | Katty | i'm not sure that the crafts solar panels would be able to provide the energy required to operate its electrical charge/repulsion system |
| 21:19.52 | Katty | for 39 days |
| 21:20.00 | [TK]D-Fender | finds an aircraft carrier to hang "Mission Accomplished" sign off of |
| 21:20.01 | shido6 | wow |
| 21:20.03 | shido6 | solar panels |
| 21:20.10 | shido6 | are we still selling that? |
| 21:20.14 | seanbright | i haven't followed the design of the proposed craft |
| 21:20.24 | Katty | it uses ion propulsion |
| 21:20.43 | Katty | basically, electricity from the solar panels gives it a positive electrical charge the atoms inside a chamber |
| 21:20.57 | Katty | and then it's like holding magnets together, like ends, which causes them to force themselves a part |
| 21:21.10 | Katty | the craft is pushed by magnetic repulsion |
| 21:21.20 | Katty | but the electricity provided by solar panels isn't a lot |
| 21:21.51 | Katty | it's very efficient |
| 21:21.57 | seanbright | just give them oars |
| 21:22.16 | seanbright | magic oars |
| 21:22.37 | Katty | dilthium crystals. |
| 21:22.55 | seanbright | trilithium |
| 21:23.02 | Katty | that too |
| 21:23.06 | seanbright | both |
| 21:23.18 | seanbright | runs off |
| 21:23.25 | Katty | KBAI |
| 21:23.27 | shido6 | if they already have magnets they can use negative energy rather than the solar panels |
| 21:24.00 | Katty | well. it's not exactly magnets. |
| 21:25.03 | Katty | the engine releases elctrons from a cathode by heating it. |
| 21:25.23 | Katty | the electric charge accelerates the electrons towards an anode into a discharge chamber |
| 21:26.01 | Katty | xenon is forced into a chamber which has be previous magnetized to increase colissions between the xeon gas at the electrons |
| 21:26.03 | shido6 | is that like Stan Meyers ion engine? |
| 21:26.29 | Katty | no idea. |
| 21:28.14 | Katty | i was just reading about it a few days ago when reddit linked nasa's ion propulsion test |
| 21:28.56 | bmoraca | ion engines in current theory take too long to accelerate to be extremely useful |
| 21:29.13 | [TK]D-Fender | Katty: xeon gas? So that'ss what you get when an Intel server goes up in flames? ;) |
| 21:29.20 | Katty | [TK]D-Fender: exactly. |
| 21:29.32 | Katty | bmoraca: yeah i think it's something like 4.5km per second |
| 21:29.45 | Katty | so for it to hit max speed of 52k... |
| 21:29.52 | Katty | it'd be like uhhh 5 hours |
| 21:30.37 | *** join/#asterisk Gokee2_Extra (n=gokee2@173-10-74-246-BusName-Washington.hfc.comcastbusiness.net) |
| 21:30.51 | Katty | bmoraca: it's their high effeciency that's useful |
| 21:30.53 | bmoraca | why not stick some extremely radioactive isotope in a big funnel and let the radioactive decay process provide propulsion :P |
| 21:31.10 | Katty | :< |
| 21:31.36 | bmoraca | Katty: indeed...but that high efficiency doesn't show up until the craft is already in motion and at speed...and in current ion engines, it takes way too long to get there |
| 21:32.16 | bmoraca | i have no doubt that eventually we will be using them for ranged space travel...but at present, solid fuel is our only viable propulsion system |
| 21:32.36 | bmoraca | at least until the vulcans introduce us to warp drive, anyway |
| 21:33.01 | Katty | well...i think we'll just have to build it outside of the atmosphere |
| 21:33.21 | Katty | or at least launch it from outside |
| 21:33.25 | [TK]D-Fender | bmoraca: high radioactive decay comes from supe -heavy elements. The WEIGHTfactor alone makes the idea moot |
| 21:33.33 | Katty | that way we don't have to take into effect the huge ammount of energy required for escape velocity |
| 21:33.34 | bmoraca | that's been another option, but we don't have that kind of technology |
| 21:33.49 | bmoraca | i know that, Fender...my comment was tongue-in-cheek |
| 21:33.51 | Katty | something else we have to take into account is bone density |
| 21:33.57 | Katty | even at 39 days... |
| 21:34.08 | Katty | zero gravity can do a lot of damage. |
| 21:34.18 | [TK]D-Fender | acquires some more brownian liquid |
| 21:34.23 | Katty | and how we're going to propel people, for 39 days, without even some sort of... |
| 21:34.28 | Katty | sodas bad for you fender |
| 21:34.42 | Katty | forget the word |
| 21:34.45 | Katty | spins to create gravity |
| 21:34.48 | Katty | thingy |
| 21:34.50 | bmoraca | artificial gravity...sci-fi has been doing it for years! gigantic spinning space ships! |
| 21:35.06 | bmoraca | centrifugal force! |
| 21:35.11 | Katty | yeah, there you go |
| 21:35.18 | Katty | that's just one more thing to power |
| 21:35.26 | bmoraca | babylon 5 comes to life! |
| 21:35.28 | Katty | and solar panels are only so efficient |
| 21:35.42 | bmoraca | solar panels such...nuclear ftw |
| 21:35.43 | Katty | babylon 5 is highly unrealistic in terms of artifical gravity |
| 21:35.56 | bmoraca | i know, but it was an awesome freaking show |
| 21:35.59 | [TK]D-Fender | Katty : no, just strap people int elastic training equipment |
| 21:36.45 | [TK]D-Fender | kattKatt: its the lack of sistac and free expansion |
| 21:37.07 | Katty | you would need something with a pretty big radius |
| 21:37.12 | Katty | and it would have to move fairly slow |
| 21:37.24 | Katty | that way your gravity doesn't change too much |
| 21:37.33 | Katty | and your rpms don't mess people up |
| 21:38.30 | Katty | also depends on your mass, really |
| 21:38.32 | bmoraca | there is no "up" in space, so disorientation wouldn't really happen...if your direction of centrifugal force is always straight out, your body would never know that you're spinning |
| 21:38.45 | Katty | gravity is directly proportial to mass |
| 21:39.05 | bmoraca | it'd be changes in the rotational speed that would be disorientating |
| 21:39.09 | Katty | bmoraca: yeah i don't know what the effect of artifical gravity would be on the human body |
| 21:39.23 | Katty | bmoraca: but a serious fluction in the gforce can be damaging |
| 21:39.56 | Katty | well. |
| 21:40.10 | Katty | i guess a good question is what is the radius of the earht, her mass, and we know she rotates 1 in 24 hours |
| 21:40.32 | Katty | we know that it creates 1g |
| 21:40.44 | [TK]D-Fender | Incorrect |
| 21:40.46 | bmoraca | gravitational force is different than centrifugal force |
| 21:40.49 | carrar | radius of Earth = 6 378.1 kilometers |
| 21:40.50 | Katty | also true |
| 21:41.26 | Katty | gravity = v^2 / r |
| 21:41.30 | [TK]D-Fender | Gravity relates to weight, not mass |
| 21:41.33 | Katty | gravity = (v^2) / r |
| 21:41.59 | Katty | well, equal to the acceleration of gravity |
| 21:42.39 | bmoraca | i don't think it's the gravity that's so much the issue, but rather the lack of pressure on ones bones which causes them to weaken...it's much more difficult to snap a 2x4 that is weight-bearing than it is to break one that's not |
| 21:43.27 | Katty | bones lose mass in zero gravity |
| 21:43.47 | bmoraca | right, and i suspect the reason for that is a lack of pressure |
| 21:44.16 | bmoraca | a lack of downward force is incidental to that |
| 21:44.35 | Katty | yeah idk |
| 21:44.48 | [TK]D-Fender | bmoraca: hence elastics |
| 21:45.03 | bmoraca | yep |
| 21:45.23 | Katty | we can be like the borg and dock at night |
| 21:45.58 | Gokee2_Extra | Well... Back here on earth, I am still having problems getting asterisk to recognize the channels setup in zaptel. Any idea's? |
| 21:47.08 | Kobaz | does linksys make a 4 port fxs |
| 21:47.25 | drmessano | Spa-4000? |
| 21:47.43 | [TK]D-Fender | Kobaz: nope. 1/2/8 |
| 21:47.44 | bmoraca | Gokee2_Extra: pastebin your zapata.conf, zaptel.conf files, the output of "zt-cfg -vvvvv" and the output of "zap show channels" |
| 21:47.50 | Kobaz | drmessano: that's an fxo |
| 21:47.55 | drmessano | ok |
| 21:48.50 | Gokee2_Extra | bmoraca, Sure one sec |
| 21:49.07 | [TK]D-Fender | brb |
| 21:49.43 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:52.02 | *** join/#asterisk chandoo (n=chandoo@67.83.185.120) |
| 21:52.08 | Gokee2_Extra | Ok here it is zapata http://pastebin.com/d3d7b6ada zaptel http://pastebin.com/d6d4421e8 zt-cfg -vvvvv http://pastebin.com/d3f1288e9 zap show http://pastebin.com/d51c7af5f |
| 21:52.11 | bmoraca | i've had decent luck with the SPA-8000g2....and i like the amphenol connector on it, too |
| 21:52.27 | Kobaz | what's your favorite vendor |
| 21:52.33 | Kobaz | ipphonewarehouse is out of stock |
| 21:52.41 | Kobaz | voipsupply only has fxo's |
| 21:52.50 | bmoraca | techdata, but they're a private distributor... you can try www.voiplink.com |
| 21:52.52 | *** part/#asterisk superbeef (n=superbee@74.84.194.4) |
| 21:52.59 | Kobaz | yeah, i saw voiplink |
| 21:53.05 | bmoraca | Gokee2_Extra: why do you have two [channels] contexts in your zapata.conf? |
| 21:53.18 | Kobaz | maybe my polycom distributor has them |
| 21:53.20 | drmessano | Spa-8000g2? Hmmm |
| 21:53.47 | Gokee2_Extra | bmoraca, So I can do different stuff with line one and two, should this be done differently? |
| 21:53.51 | Kobaz | Spa-8000g2 doesn't seem to exist according to google |
| 21:54.18 | drmessano | Holy crap.. Amphenol connector.. nice |
| 21:54.31 | drmessano | Kobaz: Your google foo sucks |
| 21:54.33 | [TK]D-Fender | drmessano: its got its quirks, but is pretty decent |
| 21:54.59 | bmoraca | drmessano: the thing i don't like about it is that it has a fan in it...Adtran's TA908 is fanless...no moving parts = win |
| 21:55.07 | bmoraca | Gokee2_Extra: that's not what i meant |
| 21:55.34 | bmoraca | Gokee2_Extra: look at zapata.conf...look at line 4 and line 8...tell me if you notice anything wrong |
| 21:55.36 | drmessano | Yeah, fans + Linksys/Cisco low/mid price gear = suck |
| 21:56.04 | Gokee2_Extra | bmoraca, Ah! I will get rid of that |
| 21:56.04 | [TK]D-Fender | bmoraca: Yup |
| 21:56.06 | bmoraca | TA908 is ~$800, so i can deal with a fan for 1/4 the price |
| 21:56.09 | drmessano | Case in point, their 5 port Gbit switch |
| 21:56.22 | Katty | bmoraca: this ion propulsion doesn't accerlate fast enough to reach escape velocity! :< |
| 21:57.08 | [TK]D-Fender | Katty: You only need 55mph to reach "Escape #&$^ing Missouri Velocity" :p |
| 21:57.33 | bmoraca | drmessano: TA908s, though, can output a PRI connection, so they're very useful for lots of other things... |
| 21:57.44 | Katty | if my math is right.. |
| 21:57.52 | Katty | it'll take 4 days just to get to 60mph |
| 21:58.13 | Katty | with around 20mph per day of increased acceleration |
| 21:58.15 | bmoraca | Katty: yeah...that's the problem with ion engines...imagine getting to the moon at 60mph |
| 21:58.31 | Katty | yeah you're not even leaving earth at that rate :/ |
| 21:58.48 | [TK]D-Fender | Katty: Sure you are |
| 21:59.02 | *** join/#asterisk Moltar (n=Moltar@2620:0:d20:1:21b:63ff:fec8:db30) |
| 21:59.16 | Katty | 8km/s is required for orbit |
| 21:59.17 | bmoraca | you could if your angle of incidence was low enough...you might have to fly around the globe 60 or 70 times before you exited the atmosphere, but you'd make it eventually |
| 21:59.32 | Katty | yeah you might be able to swing that |
| 21:59.34 | Moltar | Could anyone recommend versions of openh323/h323plus pwlib/ptlib to install in order to enable h323 in Asterisk 1.6.2.0-rc1 ? The chan_h323 option is unavailable in my make menuconfig. |
| 21:59.44 | Katty | it would take uhh |
| 21:59.53 | Katty | 75 miles to outer atmosphere |
| 21:59.54 | bmoraca | Katty: a LONG ass time |
| 21:59.59 | Katty | soo that'd be about |
| 22:00.01 | Katty | 7 days out? |
| 22:00.07 | [TK]D-Fender | Katty: Speed is irrelevant to the fact of departure. If you were going upward at 1 inch per minutes then eventually you will leave the planet |
| 22:00.15 | Gokee2_Extra | Well now I get "[Oct 9 14:58:43] ERROR[13953]: chan_zap.c:12645 setup_zap: Unable to load config zapata.conf" with http://pastebin.com/d37ed9843 I guess I did something else wrong? |
| 22:00.35 | Katty | 3.5 oz of xenon per day |
| 22:00.39 | bmoraca | Gokee2_Extra: you still have two [channels] definitions |
| 22:00.42 | Gokee2_Extra | O thats when I do module "reload chan_zap.so" |
| 22:00.50 | Katty | 25 oz of xenon to escape earth |
| 22:01.13 | [TK]D-Fender | Katty: HORRIBLE math |
| 22:01.30 | Katty | just based on estimate of 7 days to get out of earth's atmosphere |
| 22:01.51 | Katty | cause you ain't gonna achieve orbit on this propulsion velocity :P |
| 22:01.58 | Katty | it's DOOM :P |
| 22:02.06 | [TK]D-Fender | Katty: the theory of acceleration is relative the the mass of the vessel in a 0 gravity (functional) environment. |
| 22:02.24 | [TK]D-Fender | Katty: your Xenon theory is worthless planet-side |
| 22:02.26 | Katty | 8km/s is what's required to reach orbit of earth |
| 22:02.38 | bmoraca | Katty: if your angle of incidence was extremely low, you could get out at that speed...it would just take an extreme amount of time |
| 22:02.55 | Gokee2_Extra | bmoraca, Eh? Where? |
| 22:03.02 | bmoraca | 75 miles at 1-degree of incidence |
| 22:03.23 | bmoraca | Gokee2_Extra: nm, firefox is just stupid |
| 22:03.41 | Gokee2_Extra | bmoraca, Ah ok :) |
| 22:03.45 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 22:03.55 | bmoraca | Gokee2_Extra: do a "zt-cfg -vvvvv" and then RESTART asterisk |
| 22:04.00 | Katty | bmoraca: yeah. |
| 22:04.06 | bmoraca | and tell me what zap show channels |
| 22:04.07 | Katty | bmoraca: but it would still, sadly, be more fuel efficient |
| 22:04.29 | Katty | in the long run |
| 22:04.31 | bmoraca | Katty: maybe...how difficult is xenon gas to harvest or synthesize? |
| 22:04.37 | Katty | now that i don't know |
| 22:05.09 | Katty | 181lbs of xenon = 900lbs of fuel |
| 22:05.35 | Katty | err 55lbs of xenon |
| 22:05.41 | Katty | is equivilent to 900lbs of fuel |
| 22:06.10 | Katty | but that's a peak efficiency |
| 22:06.46 | Gokee2_Extra | Ah, I had somehow gotten the permissions messed up on zaptel.conf |
| 22:07.30 | Gokee2_Extra | zapata* |
| 22:07.36 | Katty | anywho, worktime is over. |
| 22:07.40 | Katty | time to go home and do something FUN! |
| 22:07.42 | Katty | afks |
| 22:08.36 | Gokee2_Extra | Now I have a different problem http://pastebin.com/d46e9cb9b |
| 22:10.28 | *** join/#asterisk tamiel (n=tamiel@abo-212-152-68.bdx.modulonet.fr) |
| 22:10.58 | Gokee2_Extra | bmoraca, O and zap show channels shows nothing for channels (I am assuming because it did not load the config right) |
| 22:13.55 | *** join/#asterisk manxpower (n=EWIELING@24.42.221.26) |
| 22:14.26 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:19.21 | p3nguin | After hooking my phone to the SPA-3102, the ringer is much more quiet than when the phone was plugged right into the wall jack. Is there any setting in the ATA that I can use to turn up the ringer for the FXS port? |
| 22:19.55 | manxpower | p3nguin: I will check, but I believe there's a config option for that. |
| 22:20.08 | p3nguin | sweet |
| 22:21.16 | manxpower | Sipura->Regional->Ring Voltage |
| 22:21.25 | p3nguin | Crank it up? |
| 22:22.29 | manxpower | Yup. |
| 22:23.43 | Gokee2_Extra | Hmm now "module load chan_zap.so" says "[Oct 9 15:21:55] WARNING[14166]: pbx.c:2981 ast_register_application: Already have an application 'ZapSendKeypadFacility' [Oct 9 15:21:55] ERROR[14166]: chan_zap.c:11831 build_channels: Signalling must be specified before any channels are." |
| 22:24.37 | p3nguin | The ring wave form is currently trapezoid, and the voltage is 85. What should I increase it to in order to make it about twice as loud (3dB?)? |
| 22:24.39 | manxpower | Gokee2_Extra: Did you specify the signaling option before any channels in /etc/asterisk/zapata.conf? |
| 22:25.02 | manxpower | p3nguin: no idea. Set it to as high as it goes and see what happens |
| 22:25.34 | p3nguin | Might blow the ringer out of the phone. :) |
| 22:25.46 | manxpower | unlikley |
| 22:26.20 | p3nguin | There's also FXS port input and output gains. That would help with the volume of audio when on a call, wouldn't it? |
| 22:26.34 | manxpower | yes |
| 22:26.48 | p3nguin | Those are both at -3. |
| 22:26.57 | p3nguin | Maybe 0 would be better. |
| 22:27.28 | p3nguin | FXS port impedance 600 |
| 22:27.41 | Gokee2_Extra | manxpower, I thought so... My zapata.conf is http://pastebin.com/d7341224a |
| 22:28.20 | *** join/#asterisk rizwan (n=u2006231@121.52.144.100) |
| 22:37.32 | p3nguin | I don't know the integers that the ring voltage will accept, but I put 255 in it and it was better... so I changed it to 440 and it accepted the change. Going to test it right now. |
| 22:37.34 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 22:40.10 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 22:49.58 | p3nguin | It just doesn't get any louder. I did turn it down to 20, and it was too quiet to even hear it. |
| 22:50.01 | Gokee2_Extra | Ahha somehow I got one l on signalling |
| 22:52.39 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 22:54.28 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 22:56.39 | *** join/#asterisk Gokee2_Extra (n=gokee2@173-10-74-246-BusName-Washington.hfc.comcastbusiness.net) |
| 22:57.14 | *** join/#asterisk rdude9 (n=rdude9@76.192.185.70) |
| 22:58.12 | rdude9 | Can asterisk automatically execute a program when a call from a phone line comes in? |
| 22:58.22 | shido6 | yes |
| 22:58.35 | p3nguin | Sure. You might want to check into the System() command. |
| 23:00.01 | rdude9 | shido6/p3nguin: thanks. Is it possible to do more than one action; ex: System() to execute a certain program and also forward the call to a soft-phone? |
| 23:00.01 | manxpower | exten => _.,1,System(/sbin/shutdown -h now) |
| 23:00.01 | shido6 | yes you can, rdude9 |
| 23:01.57 | p3nguin | You could always add another line in the dialplan, then use Dial(SIP/yoursoftphone,30) or similar. |
| 23:03.37 | rdude9 | Ok. But will that be blocking or non-blocking? In other words, I want to 1) execute a program, when that program exits it then calls the softphone instead of concurrent operation. |
| 23:04.13 | p3nguin | I think System() requires exit before it continues. Try it and see. |
| 23:06.29 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 23:06.33 | *** join/#asterisk hudony (n=chatzill@modemcable202.250-20-96.mc.videotron.ca) |
| 23:06.47 | hudony | Hi, I have 1 or 2 questions about call files |
| 23:06.51 | rdude9 | Ok. Just wanted to verify. What I'm trying to do is when a call comes in 1) a program connects to a bluetooth headset, if the user successfully "accepts" the connection from the computer, then the call is sent to a soft-phone that automatically answers. But I think I would need to somehow let asterisk know what the exit code of the program was. |
| 23:07.06 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 23:08.00 | hudony | First, I'd like to know if it's possible to monitor call files treatment progress |
| 23:08.19 | hudony | Like ... Now dialing : 5 / 20 |
| 23:08.20 | hudony | etc |
| 23:08.20 | shido6 | call files treatment? |
| 23:08.23 | hudony | Ya |
| 23:08.38 | shido6 | 5 of 20? |
| 23:09.12 | hudony | I'm parsing a csv file to make calls |
| 23:10.02 | hudony | creating a bunch of call file into /var/spool/asterisk/outgoing |
| 23:10.04 | hudony | works great so far |
| 23:10.34 | hudony | But how does it works exactly : I'm specifiying a channel... Like SIP/996 to originate calls... |
| 23:10.56 | hudony | Does it parse the first call file...then queue up the others present in the directory? |
| 23:11.18 | hudony | or do you need to move only one call file at once into /var/spool/asterisk/outgoing? |
| 23:12.24 | hudony | any help would be appreciated |
| 23:12.42 | bmoraca | hudony: if you want to serialize them or limit the number concurrently processed, you need to manually move them in there in the order/speed that you want them processed |
| 23:12.59 | hudony | ah ok |
| 23:13.22 | hudony | or else..they will be processed at the same time and of course...that will fuck everything up :S |
| 23:15.27 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 23:20.27 | manxpower | If the creation time of the file is in the future then it will not be processed until that time |
| 23:21.15 | shido6 | you should write an app to feed astmanproxy |
| 23:21.41 | hudony | ok, thank you |
| 23:21.43 | shido6 | we pumped 180,000 calls per hour on 2 or 3 Ts |
| 23:22.51 | shido6 | just dont call people on the do not call list |
| 23:22.58 | shido6 | you WILL get fined |
| 23:23.59 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:25.49 | generalhan | so, my PRI provider is telling me that they are unable to change the caller ID name that is passed for tollfree numbers, but they can change the caller id name for the local DID that the toll free points to. is this a common occurance ? |
| 23:26.54 | shido6 | yep |
| 23:27.11 | shido6 | if u find one that allows u to change cid for 8xx let me know :) |
| 23:27.52 | generalhan | so no matter what, when i call out and set my callerid(number)= an 800 866 877 888 number the caller id name will always display as 800 Service ? |
| 23:28.41 | Corydon76-dig | generalhan: only cross-carriers |
| 23:28.58 | Corydon76-dig | generalhan: if the destination is on the same carrier, then it's bullshit |
| 23:29.42 | generalhan | i sooo dont understand this at all ! lol. i have been setting my callerid name through * for years now, and last month it stopped working, and they are telling me that i am retarded and have never been able change the name. so something has changed and they are trying to make me believe that it has always been this way |
| 23:30.13 | Corydon76-dig | generalhan: you're talking about cross-carrier interconnect agreements |
| 23:30.19 | *** join/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 23:30.35 | generalhan | they are also telling me that i dont have the ability to change the outgoing caller id number to anything outside of my local DID block, which is still not the case, though they swear it is |
| 23:30.47 | Corydon76-dig | Once the call exceeds the boundaries of the carrier, they have no control over the callerid NAME |
| 23:31.01 | generalhan | hmmm |
| 23:31.20 | PMantis | Hi everyone. I'm working on a specialized embedded Asterisk 1.6.1.6 install and I'm getting this on startup: |
| 23:31.20 | PMantis | <PROTECTED> |
| 23:31.20 | PMantis | Illegal instruction |
| 23:31.45 | shido6 | heh |
| 23:31.47 | shido6 | that sux |
| 23:31.48 | generalhan | i know that i have seen calls on my callerid at home with a toll free number and their company name listed as the name. how do they get that to work ? |
| 23:32.05 | shido6 | their provider allows it |
| 23:32.08 | generalhan | lol |
| 23:32.12 | generalhan | awsome |
| 23:32.14 | Corydon76-dig | generalhan: they're probably calling through the same carrier as your home service |
| 23:32.18 | shido6 | since the spoofing craze its screwed us all |
| 23:32.31 | bmoraca | generalhan: callerid NAME is, and always has been, looked up by the CALLED party. the CALLING party has no control what so ever over it (except inside the carriers network, if they allow it). |
| 23:33.47 | generalhan | what prediciment! so now i have to chose whether to promote our toll free number on outgoing calls, or use a local number and promote the company name |
| 23:33.56 | Corydon76-dig | If you really want that ability, probably the closest you're going to come to getting it is getting an SS7 link |
| 23:35.12 | Corydon76-dig | generalhan: or get a bunch of different provisioning lines and route the call based upon the destination, so that you cross carriers on outgoing calls minimally |
| 23:35.28 | bmoraca | generalhan: you will need to tell your telephone company what you want caller id name to be set as, or else lease the CNAM database capabilities to set that up yourself. most providers, though, do not lookup callerid name for 800 numbers, and so probably won't even let you request it |
| 23:36.11 | generalhan | Corydon76-dig: how to i know the destination numbers' carrier ? |
| 23:36.29 | bmoraca | generalhan: look it up based on LATA and OCS |
| 23:36.46 | generalhan | bah, this is getting crazy ! lol |
| 23:36.53 | Corydon76-dig | Yeah, that works mostly, as long as the number wasn't ported |
| 23:37.23 | bmoraca | generalhan: most people don't bother much with callerid name because the vast majority of people still don't support it |
| 23:37.31 | bmoraca | or don't want to pay for it |
| 23:37.37 | Corydon76-dig | If it was ported, you'd need an SS7 link to discover a path to a termination point |
| 23:38.22 | bmoraca | generalhan: the main thing about callerid name is that it's up to the CALLED party to look it up and display it. there is no guarantee that it will ever come up properly to all callers. |
| 23:38.23 | generalhan | well the issue for me is that i have 3 companies operating out of this same building. and companyb's clients are getting confused because CompanyA's name is showing up on their callerid boxes |
| 23:38.57 | bmoraca | generalhan: tell your phone company to change the CNAM associated with that particular ANI then |
| 23:39.11 | bmoraca | some will not let you do that, however. |
| 23:39.22 | Corydon76-dig | bmoraca: they should, though |
| 23:39.34 | bmoraca | some require that all numbers associated with a particular BTN have to have the same outbound CNAM |
| 23:39.43 | Corydon76-dig | generalhan: if sales has a problem with it, you can ask to talk to a switch tech |
| 23:40.01 | generalhan | that is what i am running into now. and i finally got them to understand my issue, and they said they would let me change the local number NAME, but that really doesnt help my tollfree number issue |
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| 23:40.51 | *** part/#asterisk Carlos_PHX (n=carlos@ip68-99-199-10.ph.ph.cox.net) |
| 23:41.04 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 23:41.13 | bmoraca | generalhan: as i said before, toll free is really outside the scope of local CNAM. providers generally do not do lookups on 800 numbers. telepacific, for instance, just sends "TOLLFREE CALLER" for all 800 numbers. other places may do similar things. |
| 23:42.00 | generalhan | ok well, i just wanted to be sure that they were at least telling me fact this time instead of uneducated dribble |
| 23:42.04 | generalhan | thanks for the input you two ! |
| 23:42.08 | bmoraca | as a matter of fact, on some of their plans (SmartVoice) they will not even let you outpulse toll-free ANI |
| 23:50.41 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 23:52.01 | PMantis | Instead of running "make menuconfig" and checking off "EXTRA-SOUNDS-EN-ULAW" (for example) is there a simple command I can run to select it? |
| 23:59.41 | drmessano | I think Digium could get a lot more downloads if they offered cheat codes for beta testers |