| 00:01.04 | rue_mohr | maybe if I use speed dials instead of line keys, and the incomming calls are distinguished by caller id |
| 00:01.09 | rue_mohr | ????? |
| 00:01.11 | grimko | seanbright, should I disable the skinny module in the Global or Modules section of the modules.conf file ? |
| 00:01.47 | seanbright | grimko: ... |
| 00:02.13 | mosty | rue_mohr, one sip account per phone per line? |
| 00:02.13 | seanbright | if you were going to disable a "module" |
| 00:02.30 | seanbright | would you put that in the "global" section or the "modules" section? |
| 00:02.44 | rue_mohr | mosty, already there, BUT I cant make the sip phone cntact asterisk when they jut hit the line key |
| 00:02.44 | seanbright | let common sense prevail |
| 00:02.55 | grimko | well, according to you command, it will end up at the end of the file, that it to sayn the global section :D |
| 00:03.09 | seanbright | you have a global section in your modules.conf? |
| 00:03.14 | grimko | absolutely |
| 00:03.20 | seanbright | pastebin your modules.conf |
| 00:03.22 | seanbright | i have to see this |
| 00:03.30 | mosty | rue_mohr, what do you want to happen when you hit the line key? |
| 00:03.42 | grimko | the >> command can be dangerous, thats why I prefer to open files and edit them myself ;) |
| 00:03.52 | rue_mohr | I need it to open an asterisk connecdtion where I have it immediatly dial the zaptel channel |
| 00:03.56 | seanbright | no... the > command can be dangerous... >> is easy |
| 00:04.06 | seanbright | i'd rather append than clobber |
| 00:04.13 | seanbright | grimko: pastebin your modules.conf |
| 00:04.14 | seanbright | ~pb |
| 00:04.15 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 00:04.24 | mosty | rue_mohr, you want to ring a FXS port/device? |
| 00:04.41 | grimko | I know, anyway, adding thing to a file without opening it is never the best choice |
| 00:04.43 | rue_mohr | mosty, yes, |
| 00:04.51 | grimko | well thx very much anyway, that should do th trick seanbright |
| 00:04.56 | seanbright | grimko: pastebin your modules.conf |
| 00:04.58 | seanbright | ~p |
| 00:04.59 | jbot | well, p is q and not q |
| 00:05.00 | seanbright | ~pb |
| 00:05.01 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 00:05.03 | rue_mohr | but the phone dosn't send out for a connection till its got some digits |
| 00:05.12 | rue_mohr | (polycom 601) |
| 00:05.15 | seanbright | i want to see the [global] section |
| 00:05.22 | mosty | rue_mohr, just use speed dials |
| 00:05.41 | rue_mohr | yes, but |
| 00:05.46 | grimko | its empty |
| 00:05.58 | seanbright | you can just delete it |
| 00:06.00 | seanbright | it's never read |
| 00:06.04 | rue_mohr | how do I deal with incomming calls, cant do it by button, would have to be by caller id |
| 00:06.20 | seanbright | [modules] is the only relevant section in that file |
| 00:07.20 | rue_mohr | mosty, besides, I cant, these polycom 601 phones dont have enough buttons |
| 00:07.38 | rue_mohr | they have 6 line buttons, thats it |
| 00:08.05 | grimko | it says : Module names listed in "global" section will have symbols globally |
| 00:08.05 | grimko | ; exported to modules loaded after them. |
| 00:08.32 | seanbright | what version of asterisk is this? |
| 00:08.35 | grimko | Ubuntu distro for your information |
| 00:08.48 | mosty | grimko, maybe you can add a breakout box for more line keys- i don't know the 601's. or get a phone with lots of line keys like the snom320 and up |
| 00:10.17 | grimko | asterisk 1.4.21 |
| 00:10.17 | mosty | rue_mohr, rather |
| 00:10.18 | seanbright | grimko: well the docs are wrong, at least according to the code. |
| 00:10.20 | seanbright | grimko: unless ubuntu does some magic on their own |
| 00:10.31 | jaytee | you can buy sidecars for the Polycom Soundpoints that add more keys |
| 00:10.34 | rue_mohr | no, we cant spend anymore, and these stupid polycom phones take up enough of a desk already |
| 00:10.36 | rue_mohr | grrr |
| 00:10.39 | jaytee | not sure if the 601 supports them |
| 00:10.47 | grimko | I don't think they got something different than debian on asterisk... strange |
| 00:11.57 | ipguy | why worldn't my softphone ring if my sip phones and softphone are both signed into the same extension ? |
| 00:12.03 | rue_mohr | I could capture digits and spill, but people are going to get really pissed off if they have to dial a whole number to find out the line is busy |
| 00:12.19 | rue_mohr | but I will have blf |
| 00:12.22 | rue_mohr | hmm |
| 00:12.23 | grimko | seanbright, gota go. thanks for your help |
| 00:12.35 | mosty | ipguy, because only the last device to register on a sip account will get those calls |
| 00:13.35 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-1b2834353ab9eed7) |
| 00:13.47 | seanbright | jeebus |
| 00:13.57 | seanbright | they backport apps from 1.6 into their 1.4 packages |
| 00:13.58 | seanbright | yuck |
| 00:14.25 | ipguy | mosty: last ? that would be the softphone then. |
| 00:15.00 | mosty | ipguy, do a sip show peers in asterisk- that will tell you what asterisk thinks registered last |
| 00:15.19 | grimko | well then last question |
| 00:15.35 | grimko | I use a dedicated server where asterisk runs |
| 00:15.55 | grimko | its connected to a SIP network, and to clients on a VPN |
| 00:16.20 | grimko | do I have to use bindaddr 0.0.0.0 to register to the registrar ? |
| 00:16.53 | grimko | can't I just listen on my vpn network ? |
| 00:17.09 | rue_mohr | if I look at this as network of 34 phones... |
| 00:17.31 | rue_mohr | where picking up a set causes the others to be dialed automaticaly |
| 00:18.09 | rue_mohr | creek, no paddle, same problem |
| 00:20.59 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:21.29 | *** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com) |
| 00:22.45 | Zippoman | Does any one know of any way I could accomplish text to speech on asterisk without playing it through the microphone just directly on the box? Also does anyone know of any way to accomplish speech to text where it turns the persons words they are speaking into text? Is this even Possible? |
| 00:23.13 | NovceGuru | aha switchvox bashes hosted solutions in one of their webinars but offers it on thier site |
| 00:24.51 | murdock_ut | I'm having a bit of a problem with an IVR and 1.6. The user presses 1 and it is supposed to dial an extension. That works except the user does not hear any ringing. |
| 00:25.53 | jaytee | did you add an r for ringing indication in the Dial app options? |
| 00:26.14 | murdock_ut | jaytee: Tried that. It didn't do anything. |
| 00:27.31 | mosty | is the destination ringing? |
| 00:28.20 | murdock_ut | Yes. |
| 00:28.27 | murdock_ut | Here is the pastebin: http://pastebin.ca/1303944 |
| 00:28.41 | rue_mohr | arg |
| 00:28.48 | rue_mohr | k, well, getting closer |
| 00:30.08 | *** join/#asterisk klapzin (n=klapzin@200.230.21.51) |
| 00:30.29 | klapzin | anyone can help me with ael ? |
| 00:31.44 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 00:32.29 | klapzin | i have one problem with ael+mysql |
| 00:32.35 | klapzin | someone ? |
| 00:32.50 | codefreeze-lap | what can we do for you? |
| 00:33.35 | klapzin | codefreeze-lap, ok i have one problema with ael+mysql, can i chat with you in pvt ? |
| 00:35.52 | murdock_ut | Any ideas on my lack of ringing? |
| 00:36.12 | NovceGuru | listen to louder music |
| 00:36.19 | murdock_ut | That works. |
| 00:36.21 | murdock_ut | :) |
| 00:36.59 | NovceGuru | looks pretty basic, but i'm no dialplan guru |
| 00:37.26 | NovceGuru | murdock_ut: what's the network topology of all involved devices? |
| 00:38.23 | murdock_ut | I have been testing my dialplan with a phone that accessing the * box across a vpn. |
| 00:38.56 | NovceGuru | what type of vpn? |
| 00:39.36 | murdock_ut | ipsec |
| 00:39.40 | NovceGuru | can you test the dialplan with an extension on the same network as the * box? |
| 00:40.13 | murdock_ut | I get ringing when I call an extension directly. |
| 00:40.26 | NovceGuru | hm |
| 00:40.42 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
| 00:41.03 | murdock_ut | I know that the answer app changed between 1.4 and 1.6 |
| 00:41.26 | murdock_ut | This dialplan works fine in 1.2. |
| 00:41.30 | jaytee | murdock_ut, add a comma after the r, save your extensions.conf and do a dialplan reload |
| 00:42.39 | murdock_ut | jaytee: That didn't change anything. |
| 00:43.10 | NovceGuru | murdock_ut: does an extension on the same network as the asterisk box get ringing? |
| 00:43.18 | murdock_ut | Now that is weird. |
| 00:43.23 | *** join/#asterisk hakr (n=hakr@pdpc/supporter/active/hakr) |
| 00:43.47 | murdock_ut | I tested the ivr ringing in through the pstn and it rings. |
| 00:43.56 | murdock_ut | Why would that be? |
| 00:44.03 | jaytee | what are you calling from? |
| 00:44.09 | murdock_ut | cell phone. |
| 00:44.26 | jaytee | the cell phone does indicate ringing when calling through the PSTN? |
| 00:44.34 | murdock_ut | Yes. |
| 00:44.46 | jaytee | when you call and don't get ringing what are you calling from? |
| 00:45.32 | murdock_ut | sip phone register to * box but on different network connected through ipsec vpn. |
| 00:46.27 | jaytee | so if you call extension 1 does voicemail pickup and do you hear the prompts? |
| 00:46.55 | murdock_ut | Yes. |
| 00:47.33 | murdock_ut | It is when I dial 0 I don't hear ringing but it is ringing the 1399 |
| 00:49.04 | murdock_ut | kinda weird. |
| 00:51.05 | *** join/#asterisk steerpike (n=Unknown@unaffiliated/steerpike) |
| 00:51.20 | Zippoman | <Zippoman> Does any one know of any way I could accomplish text to speech on asterisk without playing it through the microphone just directly on the box? Also does anyone know of any way to accomplish speech to text where it turns the persons words they are speaking into text? Is this even Possible? |
| 00:51.28 | steerpike | hi, where can i find mobile phones that support sip over wifi? |
| 00:52.25 | murdock_ut | steerpike: I don't work for these guys, but here is a pretty good list: http://www.8774e4voip.com/category_s/36.htm |
| 00:52.39 | steerpike | thanks :) |
| 00:54.00 | NovceGuru | murdock_ut: can you try with 1.4? |
| 00:54.02 | steerpike | pricey :\ |
| 00:54.45 | murdock_ut | I will later. The good this is that it works for outside callers which is who it is for. |
| 00:55.00 | murdock_ut | I'm also going to upgrade to 1.6.0.3 which came out today. |
| 00:55.39 | NovceGuru | good idear |
| 00:55.59 | murdock_ut | Thanks for you help. Just wanted to make sure it wasn't something obvious. |
| 00:56.38 | murdock_ut | Now if only I can get the one step parking to allow reparks I would be happy. |
| 00:56.39 | *** join/#asterisk freakazoid0223 (n=matt@pool-68-238-180-205.phil.east.verizon.net) |
| 00:59.00 | FinboySlick | I'm trying to migrate to dahdi... do I still Dial(Zap/....)? |
| 00:59.46 | murdock_ut | dial(DAHDI/ |
| 01:00.17 | mchou | ipguy: which specfic router are you using? |
| 01:00.59 | jaytee | murdock_ut, there is a parameter you can add that allows you to use the Zap command with DAHDI |
| 01:01.12 | jaytee | excuse me, Zap channel |
| 01:01.25 | murdock_ut | for backward compatibility, I know. |
| 01:01.58 | jaytee | yes, so if you wan't you can leave the Dial(Zap/channel) the same |
| 01:02.51 | NovceGuru | damn I couldn't get imap voicemail storage to work on the first try |
| 01:02.55 | NovceGuru | gives up |
| 01:03.12 | FinboySlick | app_dial.c:1242 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) :( |
| 01:03.31 | FinboySlick | Bad day to start experimenting with new stuff I guess ;) |
| 01:04.49 | ipguy | so if i'm away and use a softphone to plug into my * box at home, my softphone will not ring if i gat a call on the home sip account ? |
| 01:05.21 | NovceGuru | deciphers question |
| 01:06.10 | NovceGuru | you would need to configure the 'home sip account' to ring your softphone extension and whatever else is connected to that home sip account |
| 01:08.25 | ipguy | NovceGuru: my sip hanset is registered to my * box, ext 101, if i call the extension, the phone rings, no issues there, the problem is that when i'm away and i use a softphone to connect to ext 101, the softphone will not ring if i recieve a call, the sip phone does though |
| 01:08.55 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-154-243.dsl.stlsmo.sbcglobal.net) |
| 01:08.59 | ipguy | NovceGuru: is there any way round that ? |
| 01:10.01 | NovceGuru | you can't connect multiple devices to one sip account |
| 01:10.33 | ipguy | NovceGuru, yes but apparently only the first device that registeres will ring |
| 01:10.36 | NovceGuru | you'll need to create another extension for your softphone to connect to and ring both, as one option |
| 01:10.48 | ipguy | i c |
| 01:11.04 | ipguy | good idea ! |
| 01:11.22 | ipguy | will give that a go ! |
| 01:11.25 | NovceGuru | gluck |
| 01:11.32 | ipguy | thanks |
| 01:11.47 | Zippoman | what do you guys think about festival ? |
| 01:16.48 | LemensTS | Im having a heck of a time getting phpagi working. Here is the .php file i wrote that im calling from the dial plan, and on the bottom i put what the cli shows http://pastebin.com/m660e0db8 |
| 01:16.53 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:18.03 | LemensTS | I never see it receive answer or hangup in the cli |
| 01:20.28 | *** part/#asterisk steerpike (n=Unknown@unaffiliated/steerpike) |
| 01:27.16 | *** join/#asterisk smps (n=smps@193.170.53.51) |
| 01:30.13 | FinboySlick | Interesting. I have no HWEC at all with dahdi... (I just loaded no sofware echo canceller) Do I need to load extra modules for HWEC? |
| 01:31.29 | *** join/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 01:34.02 | jaytee | what kind of card? |
| 01:34.12 | FinboySlick | jaytee: TDM800P |
| 01:34.24 | LemensTS | ahh got it, i had to have phpagi-asmanager.php in the directory too |
| 01:34.28 | jaytee | which has the HWEC module on it |
| 01:34.54 | FinboySlick | The bill says so. How would I physically know? |
| 01:35.10 | jaytee | there would be a purple colored module attached to the card |
| 01:35.18 | carrar | The Answer: http://www.tysknews.com/LiteStuff/financial_theorem.htm |
| 01:35.56 | FinboySlick | jaytee: You mean besides the FXO? |
| 01:36.01 | jaytee | yes |
| 01:37.45 | FinboySlick | jaytee: Now I'm dumbfounded... Where on the card should it be (it's obviously not there, just wondering) |
| 01:38.21 | FinboySlick | Under the FXO or somesuch? |
| 01:38.53 | jaytee | http://www.telephonydepot.com/Catalog/Digium-Accessories/Digium-VPMADT032-Echo-Cancellation-Module |
| 01:39.13 | jaytee | that's what it looks like and it clips onto the card at the bottom under the FXO or FXS modules |
| 01:39.53 | FinboySlick | Great... |
| 01:40.05 | FinboySlick | And here I wondered why I have no HWEC. |
| 01:40.36 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:41.37 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:42.17 | FinboySlick | I feel so dumb... I spent hours and hours trying to get the echo problem fixed by hardware :P |
| 01:43.17 | FinboySlick | In the meantime, which would be the best software echo canceller I can use? |
| 01:43.29 | NovceGuru | broken image link! |
| 01:43.57 | jaytee | whenever you order a card either analog or digital made by Digium or Sangoma you have to specify the HWEC module. Some vendors break down the cards into different model numbers. A Digium TE212P card comes with HWEC but the TE210P doesn't. Same card, one with, one without. |
| 01:44.10 | jaytee | FinboySlick, try MG2 |
| 01:47.57 | jaytee | man, this move Wanted is great! |
| 01:48.15 | Zippoman | ya |
| 01:48.41 | *** join/#asterisk ryoohki (n=ryoohki@208.96.15.252) |
| 01:50.01 | FinboySlick | jaytee: That's what I was using without knowing prior to switching to dahdi. |
| 01:50.53 | ryoohki | is asterisk able to forward voice mail to a voicemail box not associated with a mac address/voip phone? in other words, a general mailbox that is not associated with anyphone? |
| 01:52.08 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 01:52.26 | ryoohki | alternatively, can one voip phone have two extension associated with one mac address? last time i tired this the voip system wouldn't start( but that may have been a mysql problem) |
| 01:53.22 | carrar | if the phone has two line buttons why not have each register with a different SIP registery |
| 01:54.15 | carrar | that way you can pick how your call goes out |
| 01:54.48 | *** part/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:54.51 | NovceGuru | ryoohki: a vmbox doesn't have to be accociated with an extension |
| 01:55.00 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:56.29 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:56.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:57.48 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 01:58.26 | jaytee | FinboySlick, unlike zaptel that needed to have the echo canceller defined at compile time, DAHDI uses loadable modules so you need to specify it in /etc/dahdi/system.conf |
| 01:59.08 | FinboySlick | I modprobed it as a test. |
| 01:59.26 | FinboySlick | I have to specify on each channel too? |
| 02:00.00 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 02:02.19 | jaytee | FinboySlick, nope, just at the bottom beneath all your channel signalling just add one line for echocanceller=mg2 |
| 02:02.58 | jaytee | need to restart dahdi service after that |
| 02:03.03 | *** join/#asterisk hakr (n=hakr@pdpc/supporter/active/hakr) |
| 02:03.03 | jaytee | and asterisk |
| 02:06.10 | FinboySlick | jaytee: Wonderful... I might actually sleep tonight. |
| 02:06.24 | jaytee | sleep is a good thing |
| 02:06.51 | FinboySlick | jaytee: What's the param to have dahdi handle dial(Zap/ ... ? |
| 02:07.37 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5ccb0230c242f18f) |
| 02:07.46 | jaytee | dahdichanname = no |
| 02:07.53 | jaytee | that goes in asterisk.conf |
| 02:08.14 | jaytee | in the [options] section |
| 02:09.28 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 02:09.37 | *** join/#asterisk andresmujica (n=andresmu@190.24.94.102) |
| 02:10.34 | FinboySlick | jaytee: That one didn't seem to work after restarting asterisk. Do I have to restart dahdi too? |
| 02:11.33 | jaytee | is asterisk running as a service? |
| 02:12.23 | FinboySlick | jaytee: Yeah. |
| 02:12.30 | jaytee | what distro? |
| 02:12.34 | FinboySlick | gentoo |
| 02:13.30 | jaytee | stop asterisk and then stop dahdi and then restart dahdi then asterisk and then run asterisk -vvvr |
| 02:13.39 | jaytee | and then type help |
| 02:14.11 | FinboySlick | Gah, nevermind. |
| 02:14.19 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 02:14.21 | FinboySlick | The [options] part was commented out. |
| 02:15.48 | FinboySlick | Works now. |
| 02:16.00 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:16.09 | FinboySlick | jaytee: You're a lifesaver. |
| 02:16.26 | jaytee | no, but I'm sucking on one at the moment. Wintergreen :-) |
| 02:17.07 | jaytee | from the economy size 6.88 oz bag no less |
| 02:17.51 | jaytee | what an awesome ending to the movie. kind sad though |
| 02:17.56 | [TK]D-Fender | Tweet tweeet, triddle twiddle... only one candy with the hole in the middle... |
| 02:18.08 | [TK]D-Fender | just aged another 10 years for that jingle.... |
| 02:18.20 | jaytee | Angelina Jolie has an ass that would make a bishop kick out a stained glass window for a better look. |
| 02:18.23 | FinboySlick | Interesting coincidence. Since I have your attention though. /etc/dahdi/modules Doesn't seem to load anything by itself... Is it supposed to just call modprobe on all the modules in the list when it starts? |
| 02:18.57 | jaytee | not sure, never had to mess with it. |
| 02:19.26 | FinboySlick | I'll take care of that tomorrow anyway. |
| 02:20.11 | jaytee | i think it's a list that gets read by dahdi when it loads as a service. if you don't want to load any of the modules comment them out with a # |
| 02:20.12 | FinboySlick | Oddly enough, chanel1 has okay volume, 2 is extremely low... but I don't have any gain settings. |
| 02:20.35 | jaytee | 1.4 or 1.6? |
| 02:20.36 | FinboySlick | jaytee: That's what I did, my problem is that it doesn't load those I didn't comment out. |
| 02:20.44 | FinboySlick | this is 1.4 |
| 02:20.47 | jaytee | ah |
| 02:21.53 | FinboySlick | I'll try to force gain to 0.0 |
| 02:22.36 | FinboySlick | These are brand new FXOs, I hope they're not giving out already. |
| 02:24.36 | drmessano | X100Ps? |
| 02:24.54 | FinboySlick | Um, 4 ports. |
| 02:25.20 | jaytee | FinboySlick, gain should default to 0.0 |
| 02:25.24 | drmessano | Oh god, they have a 4 port X100P now |
| 02:25.29 | drmessano | THE WORLD IS ENDING |
| 02:25.44 | drmessano | switches to microsoft products ASAP |
| 02:25.46 | FinboySlick | Well, I don't know if they're x100p, just that they're 4 port. |
| 02:26.03 | drmessano | Heh |
| 02:26.05 | NovceGuru | MICROSOFT COMMUNICATOR 2007 TO THE RESCUE |
| 02:26.12 | drmessano | FAIl |
| 02:26.13 | jaytee | try setting txgain and rxgain both to 1.0 and then increase in increments of 1 after each dahdi/asterisk restart until you're satisfied. |
| 02:26.40 | NovceGuru | I'm kinda bummed polycom makes a phone for that :( |
| 02:26.42 | drmessano | MICROSOFT OFFICE LIVE COMMUNICATOR 2007 OMG LOTS OF ADJECTIVES EDITION ENTERPRISE ADD SUFFIXES TOO |
| 02:26.50 | [TK]D-Fender | FinboySlick: Swap the lines to confirm if its the line and not the module |
| 02:27.06 | jaytee | it's probably the lines |
| 02:27.20 | [TK]D-Fender | Either way, PROVE IT |
| 02:27.31 | FinboySlick | That requires a step ladder, so I'll file that in my 'tomorrow' folder too ;) |
| 02:27.34 | jaytee | lighten up, lenny!!! |
| 02:27.35 | FinboySlick | I'm going to bed now. |
| 02:27.58 | [TK]D-Fender | FinboySlick: Ladder? Go to the back of your server and swap the damn connectors! |
| 02:28.12 | FinboySlick | [TK]D-Fender: I need a ladder to get to the server. |
| 02:28.17 | jaytee | wtf? |
| 02:28.34 | jaytee | who builds a computer room with racks that high? |
| 02:28.38 | [TK]D-Fender | jaytee: Up there so the crocs don't get to it obviously! |
| 02:28.49 | jaytee | they have crocs in B.C.? |
| 02:29.14 | FinboySlick | jaytee: I'm at the extreme other end of Canada actually. |
| 02:29.22 | FinboySlick | Far-eastern Quebec. |
| 02:29.27 | jaytee | your IP says different |
| 02:29.32 | FinboySlick | Indeed it does. |
| 02:29.39 | jaytee | hackerboy! |
| 02:29.44 | drmessano | Oh god |
| 02:29.52 | FinboySlick | Actually, no, lazy provider. |
| 02:29.58 | [TK]D-Fender | FinboySlick: Whereabouts? |
| 02:29.59 | drmessano | Reading Mitnicks book doesnt make you a hacker |
| 02:30.19 | FinboySlick | [TK]D-Fender: Um, end of the Gaspé penninsula. |
| 02:30.36 | [TK]D-Fender | FinboySlick: Yup, that'd to it... |
| 02:30.36 | jaytee | reading Hackers (the book by Ken Kesey, not the book based on the Angelina Jolie movie) doesn't make you a hacker either. |
| 02:30.58 | jaytee | having a 10 years running subscription to 2600 magazine might though |
| 02:31.13 | [TK]D-Fender | jaytee: NOW you're talking :0 |
| 02:31.24 | *** join/#asterisk mosty (n=mosty@eth1426.vic.adsl.internode.on.net) |
| 02:31.28 | [TK]D-Fender | jaytee: Used to love their phreaking guides.... |
| 02:31.32 | jaytee | some of us still have our Cap'n Crunch whistles :-) |
| 02:32.01 | [TK]D-Fender | jaytee: pay-phone DTMF trrickery, shorting off the other wirse to turn them into veritable slot-machines, etc |
| 02:32.04 | *** join/#asterisk Trionnis (n=rboggs@s233-51-251.nap.wideopenwest.com) |
| 02:32.13 | jaytee | although they only have sentimental value now |
| 02:32.17 | FinboySlick | almost scoffs at 10 years of 2600, then realises that he has been over 10 years on the internet, feels like he wasted his life... And gets depressed. |
| 02:32.19 | drmessano | I love 2600 |
| 02:32.40 | drmessano | 10 years on the internet? Thats not long |
| 02:32.55 | FinboySlick | It is when you still look young! |
| 02:33.15 | drmessano | How old? |
| 02:33.44 | FinboySlick | Hah, a bit shy of midlife crisis actually. |
| 02:34.04 | FinboySlick | And it's been more than 10 years, he's just amazed by how fast it went by. |
| 02:34.29 | FinboySlick | is so troubled that he messes up his pronouns. |
| 02:34.31 | drmessano | Are you speaking in the 4th person? |
| 02:34.42 | FinboySlick | See? |
| 02:34.45 | jaytee | in my collection of computer oddities I still have a BASF 8" 16K single sided floppy disk. I actually used it at one time. Reagan was president. |
| 02:35.20 | FinboySlick | jaytee: Yeah, but that makes you venerable and cool. I started on Tandy. |
| 02:35.32 | jaytee | TRS-80? |
| 02:35.44 | FinboySlick | second gen, yes. |
| 02:35.53 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 02:36.18 | Trionnis | Atari 800XL |
| 02:36.33 | jaytee | first tech job was for Analog Devices back when they made portable CP/M "laptops". Didn't have a monitor, just a line printer that echoed what you typed. |
| 02:36.34 | Trionnis | *with* the tape drive! |
| 02:36.46 | jaytee | oooooh!!! |
| 02:36.48 | FinboySlick | Trionnis: So you were one of the cool kids with games... I had to re-type mine all the way in basic every time! |
| 02:36.55 | jaytee | I'm gettin wood! |
| 02:37.05 | Trionnis | lol |
| 02:37.06 | drmessano | has a C64 + VIC 1525 printer + 1541 Floppy Drive |
| 02:37.27 | Trionnis | yeah |
| 02:37.33 | Trionnis | Temple of Apshai |
| 02:37.36 | FinboySlick | And sometimes the blue square didn't move the way I wanted. |
| 02:37.37 | Trionnis | 3 tapes |
| 02:37.39 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 02:37.48 | Trionnis | took almost 25 minutes to load |
| 02:37.58 | jaytee | ok, I bet I can top all of you. Perforated paper tape anyone? been there, done that! |
| 02:38.16 | Trionnis | hm, not paper tape |
| 02:38.21 | Trionnis | plenty of punchcards |
| 02:38.25 | jaytee | it was like the friggin player piano of computer systems |
| 02:38.29 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 02:38.33 | Trionnis | yep |
| 02:38.35 | Trionnis | heh :) |
| 02:39.01 | jaytee | used to use punch cards for encryption on old AUTOSEVOCOM lines back in the 70's. |
| 02:39.31 | drmessano | I have you all beat... I used to work in radio... 1920's FTW |
| 02:39.55 | Trionnis | ok, so riddle me this |
| 02:40.04 | drmessano | Unless you've worked in newspaper.. then you pwn me |
| 02:40.18 | jaytee | I did HF, VHF and troposcatter microwave back in the 70's. and occassionally S-band satcom |
| 02:40.24 | FinboySlick | I drew stuff on cave walls? ;) |
| 02:40.29 | jaytee | lol |
| 02:40.40 | [TK]D-Fender | I STILL draw stuff on cave walls. |
| 02:41.10 | [TK]D-Fender | loves playing head-games with paleontologists = :p |
| 02:41.20 | Trionnis | 1.6.0.3, fresh clean install, talking to a Voxeo Prophecy server... get the initial invite everything's peachy... Prophecy sends a second invite with a different RTP port. * throws back a 100, then a 200, but insists on sending RTP to the original port, not the new one |
| 02:41.54 | Trionnis | completely reproduceable, 1.4.21.1 works fine, and honors the second invite |
| 02:41.55 | [TK]D-Fender | Trionnis: make sure your peer is "nat=no" |
| 02:41.59 | Trionnis | yep, it is |
| 02:42.03 | jaytee | yeah, we should all find caves nearby and draw on tech stuff on the walls. 10000 years from now future paleontologists will find drawings of iPhones, laptops, etc and go WTF? |
| 02:42.11 | Trionnis | lol @ jaytee |
| 02:42.56 | FinboySlick | will draw Paris Hilton. She totally defines our time. |
| 02:43.40 | Trionnis | http://pastebin.ca/1304007 |
| 02:45.14 | Trionnis | I guess what I'm wondering... is there a new sip.conf setting in 1.6 that is somehow related to this and I'm just overlooking it? |
| 02:46.00 | Trionnis | and for the record, that exact same entry for the peer that's in the pastebin works 100% in 1.4.x |
| 02:48.29 | joobie | guys anyone got a decent walk through of the setup of a sangoma isdn card for asterisk? |
| 02:48.58 | joobie | i found one that has mention of zaptel and dahcdi or something.. two different methods, not sure which i usd |
| 02:49.00 | joobie | -d+e |
| 02:51.12 | [TK]D-Fender | joobie: http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi |
| 02:53.00 | *** join/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net) |
| 02:53.04 | joobie | [TK]D-Fender, why dahdi though ?I found another article on that wiki for a zaptel version... why not zaptel? |
| 02:53.14 | Trionnis | zaptel is dead |
| 02:53.16 | joobie | when i bootup my box with asterisk installed i see zaptel come up btw |
| 02:53.21 | Trionnis | it was replaced with dahdi |
| 02:53.23 | joobie | like reference to the zaptel wording in services |
| 02:53.37 | Trionnis | no longer updated either |
| 02:53.41 | joobie | eek |
| 02:53.53 | joobie | because the zaptel wording comes up when i bootup, does this mean im running the older asterisk? |
| 02:53.58 | joobie | like services names etc are zaptel.. |
| 02:54.01 | joobie | is there a way i can check? |
| 02:54.02 | [TK]D-Fender | joobie: Zaptel got RENAMED and is NO MORE |
| 02:54.04 | Trionnis | it means you have zaptel installed |
| 02:54.13 | joobie | .. is there an upgrade path |
| 02:54.28 | joobie | bugger - just put the box in the data center yesterday |
| 02:54.28 | joobie | ergh |
| 02:54.30 | Trionnis | if only there were a place one could perform some kind of locating function |
| 02:54.48 | Trionnis | perhaps an "engine" that would let you "search" for something |
| 02:54.51 | Trionnis | ;) |
| 02:55.10 | joobie | #asterisk @ irc ? |
| 02:55.15 | joobie | ;P |
| 02:55.22 | [TK]D-Fender | joobie: its all in their WIKI, go read |
| 02:55.34 | joobie | ok thanks fender, trio |
| 02:55.44 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 02:57.05 | [TK]D-Fender | Sangoma B-series cards = awesome value |
| 02:57.13 | *** join/#asterisk loather-work (n=khudson@internal-nat.djnetworks.net) |
| 02:57.22 | Trionnis | eh... I'm a Digium guy all the way |
| 02:57.40 | Trionnis | (and yes, that stuff on the end of my nose *does* happen to be brown... why do you ask?) |
| 02:58.15 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 02:58.16 | loather-work | does dahdi have the same limitation as zaptel regarding the inability to reconfigure PRI without a restart of asterisk? |
| 02:58.39 | Trionnis | afaik yes |
| 02:58.44 | Trionnis | haven't done much with it though |
| 02:58.57 | loather-work | ugh, thats such a whore of a limitation |
| 02:58.59 | jaytee | who was it got asked to leave some restaurant or somethign because they were wearing a Sangoma shirt when Mark Spencer and everyone was going in at Astricon? |
| 02:59.24 | loather-work | what's wrong with sangoma? |
| 02:59.28 | Trionnis | I must have missed that |
| 02:59.29 | jaytee | nuthin |
| 02:59.34 | loather-work | that's what i though |
| 02:59.38 | *** part/#asterisk ipguy (n=ipguy@129.94.190.121) |
| 02:59.45 | [TK]D-Fender | loatherNo, you do not have to take down * for that. |
| 02:59.47 | Trionnis | of course, I was quite busy schmoozing :) |
| 03:00.45 | loather-work | [TK]D-Fender: ok, excellent. i'm using an ancient version of both asterisk and zaptel, and the ability to do things like change my outbound dial plan while asterisk is running would be nice |
| 03:01.16 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 03:01.21 | [TK]D-Fender | loather-work: Dialplan != zaptel |
| 03:01.23 | LemensTS | jaytee: u series about the sangoma shirt |
| 03:01.34 | loather-work | [TK]D-Fender: PRI dialing plan |
| 03:01.41 | jaytee | someone said that awhile back in here |
| 03:01.46 | loather-work | e.g. national, international, unknown, etc. |
| 03:02.06 | LemensTS | huh, lol that is funny i would have worn a grandstream shirt |
| 03:02.36 | loather-work | trying to debug a caller ID issue, and everything the telco's asking me to change requires a restart of asterisk right now |
| 03:02.47 | loather-work | and that's not possible when ~20 people are on the phone :) |
| 03:07.54 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 03:08.02 | *** join/#asterisk joako (n=joako@adsl-144-103-238.mia.bellsouth.net) |
| 03:08.13 | Trionnis | sure it is |
| 03:08.16 | NovceGuru | loather-work: it is in fact- possible |
| 03:08.34 | NovceGuru | but being possible without pissing people off... |
| 03:08.38 | Trionnis | just restart it then when someone comes to find you, look really concerned and tell them that you'll look into it right away |
| 03:09.03 | loather-work | yeah ok |
| 03:09.06 | Trionnis | hehe :) |
| 03:09.08 | loather-work | i need it to be nondisruptive :) |
| 03:09.09 | NovceGuru | OMG WTF happend |
| 03:09.10 | xuser | just disconnect your phone, turn off you mobile, go offline and do the change ;) |
| 03:09.11 | NovceGuru | say that |
| 03:09.16 | NovceGuru | ohh emmm geee |
| 03:09.42 | loather-work | i already stealth restarted it twice today when nobody was using it |
| 03:11.05 | Trionnis | so is there a comprehensive list somewhere that shows the sip.conf and chan_sip changes in 1.6? the changelog isn't of much use in trying to figure this thing out |
| 03:17.34 | *** join/#asterisk simonr (n=simonr@125.38.15.204.static.thewire.ca) |
| 03:19.59 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 03:29.40 | f0urtyfive | hrmmm |
| 03:29.47 | f0urtyfive | my asterisk seems to have just lost all its peers... |
| 03:30.00 | f0urtyfive | I mean, all the entries are still there, but all the Dyn hosts it doesnt know the host |
| 03:30.05 | f0urtyfive | or the qualify status |
| 03:30.10 | f0urtyfive | like everything stopped registering |
| 03:30.39 | f0urtyfive | wierd |
| 03:30.43 | *** join/#asterisk JJ2110 (n=James@222-152-235-244.jetstream.xtra.co.nz) |
| 03:30.49 | f0urtyfive | sip show channels shows a ton of Rx : register |
| 03:30.52 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 03:34.47 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 03:34.52 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:37.55 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 03:39.01 | f0urtyfive | nevermind |
| 03:39.04 | f0urtyfive | figured it out |
| 03:43.30 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 03:44.22 | *** join/#asterisk chendy (n=chatzill@121.35.145.40) |
| 03:46.55 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 03:54.35 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 03:54.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 04:00.29 | *** join/#asterisk freakazoid0223 (n=matt@pool-68-238-180-205.phil.east.verizon.net) |
| 04:05.59 | *** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
| 04:14.35 | *** join/#asterisk DaPrivateer (n=matt7229@crimson.66fruit.com) |
| 04:15.31 | DaPrivateer | Is it possible to include mailboxes from one context in another context? I tried and "include => " in the voicemail.conf but it doesn't seem to be working |
| 04:16.04 | [TK]D-Fender | DaPrivateer: What is the point? |
| 04:16.29 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 04:16.45 | DaPrivateer | Seperate directories based on inside versus outside calls, but i obviously don't want to make people that occur in both have two mailboxes |
| 04:22.03 | drmessano | I guess jeev got tired of trolling #asterisk.. hes trolling #freeswitch now |
| 04:24.25 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-247-113-221.dsl.pltn13.sbcglobal.net) |
| 04:24.25 | jaytee | really? |
| 04:25.21 | drmessano | yes lol |
| 04:29.27 | DaPrivateer | [TK]D-Fender Do you have any idea if what I am asking is possible? I mean, I can get around it with some ln -s's but i'd prefer not to... |
| 04:29.35 | jaytee | i'm gonna go into "stealth" mode and go listen in :-) |
| 04:30.30 | [TK]D-Fender | DaPrivateer: no reason you can't make a DUPLICATE entry in another VM context exclusively for directory purposes |
| 04:30.46 | [TK]D-Fender | drmessano: No, he got banned. |
| 04:31.01 | DaPrivateer | i could, but then they'd have to record their name twice |
| 04:31.39 | DaPrivateer | either that or i'd have to symbolic link the name file for each duplicate entry |
| 04:31.46 | drmessano | [TK]D-Fender: bans arent permanent.. and hes not tried to come back since it expired |
| 04:31.55 | drmessano | So "tired of trolling" is accurate |
| 04:32.08 | [TK]D-Fender | DaPrivateer: thats what it comes down to. |
| 04:33.45 | jaytee | maybe the crowd that hangs in #freeswitch is more impressed with someone who has an American Express Black Card |
| 04:34.18 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 04:35.41 | ricko73 | jaytee: is that why teliax 2.0 works so well? |
| 04:35.55 | jaytee | what? |
| 04:36.15 | ricko73 | teliax moved their new platform to freeswitch |
| 04:36.29 | jaytee | I didn't know |
| 04:36.34 | ricko73 | it's been a rough transition |
| 04:36.54 | ricko73 | I'm still on their legacy product so I haven't been affected |
| 04:37.14 | ricko73 | everthing from complete outages to DTMF not working |
| 04:39.31 | drmessano | Apparently the DTMF issues were due to an idiot implementation in the upstreams switches |
| 04:39.37 | ricko73 | yeah |
| 04:40.02 | ricko73 | and I realize that's really not their fault |
| 04:40.06 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 04:40.08 | jaytee | so should we all just abandon asterisk and go with freeswitch? is that the future? |
| 04:40.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 04:40.39 | ricko73 | two different tools from what I've seen |
| 04:40.55 | drmessano | I've been looking at the SIP TCP implementation as an answer to Exchange UM |
| 04:41.14 | ricko73 | drmessano: doesn't asterisk 1.6 support sip tcp? |
| 04:41.32 | jaytee | drmessano, just use sipX until Asterisk 1.8 is ready :-) |
| 04:41.33 | drmessano | Asterisk 1.6 sorta does |
| 04:42.51 | drmessano | Its also sorta supports XMPP integration.. which it seems in further along in FS |
| 04:44.42 | jaytee | I'd like to run NoSwitch. |
| 04:44.58 | jaytee | No phones, no fuckin users, no complaints :-) |
| 04:45.24 | drmessano | Im just tired of having to shoehorn app servers into app servers into app servers |
| 04:46.28 | jaytee | I'm tired of asshole bosses that want cadillac quality at yugo prices and thing every system implementation is a cut, dried and simple as installing Microsoft Arcade. |
| 04:46.31 | drmessano | Unified Communications is the biggest joke ever.. Everyone claims to integrate into someone else.. but they go so half-assed into it, it's more a hook for someone else to write LOTS OF CODE to bridge to them versus a real olive branch |
| 04:47.35 | drmessano | I think they teach "How to write an API" before they teach basic programming |
| 04:48.00 | drmessano | "Our shit doesnt integrate with anyone else, but -------------------> Check out the PDF of our API" |
| 04:48.10 | drmessano | Thats a nice happy go lucky "go screw yourself" |
| 04:48.30 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 04:50.13 | jaytee | I've been using Cepstral with the Marta mexican-spanish voice to create spanish prompts for my ivr for the past two days. I keep getting this urge to embed "chenga tu madre! Learn english or get the f*&^ outta the country!" |
| 04:50.31 | ricko73 | lol |
| 04:51.25 | drmessano | Its just insane |
| 04:51.28 | drmessano | Its like |
| 04:52.47 | jaytee | ugh, it's late |
| 04:52.54 | jaytee | time for sleep |
| 04:52.58 | jaytee | niters |
| 04:53.00 | drmessano | "We don't support integration with X, but we wrote three lines of code and a config option to add the x-blah header to invites so if you write the other 20000 lines of code on your end to integrate, we're there dude" |
| 04:53.11 | jaytee | hehehee |
| 04:57.31 | *** join/#asterisk chendy (n=chatzill@219.134.30.111) |
| 05:00.10 | loather-work | do i first compile zaptel then wanrouter? |
| 05:02.03 | ricko73 | loather-work: ask Sangoma, but I believe you must do that then recompile zaptel afterwards |
| 05:02.39 | drmessano | That makes no sense |
| 05:02.57 | ricko73 | it's sangoma. It's supposed to make sense? |
| 05:03.03 | drmessano | You wouldnt compile zaptel, compile Wanrouter, then recompile zaptel afterwards |
| 05:03.13 | drmessano | What use is the first zaptel compile? |
| 05:03.13 | loather-work | yeah, sangoma's documentation isn't exactly clear |
| 05:03.23 | loather-work | i know there's an order to this, i just don't know what |
| 05:03.24 | *** join/#asterisk simonr (n=simonr@dsl-207-112-76-110.tor.primus.ca) |
| 05:03.47 | ricko73 | drmessano: the compilation tells you to recompile zaptel afterwards |
| 05:03.51 | mosty | uncompress zaptel, compile wanpipe (and tell it where zaptel is), then build zaptel |
| 05:04.02 | [TK]D-Fender | loather-work: libpri, zap, wanpipe (redoes Zap for you), * |
| 05:04.03 | drmessano | Thats makes actual sense |
| 05:04.11 | loather-work | got it, thanks a bunch. |
| 05:04.21 | loather-work | will i need to recompile asterisk after recompiling zaptel? |
| 05:04.28 | drmessano | .... |
| 05:04.38 | [TK]D-Fender | loather-work: not if it was the same version as prior |
| 05:04.44 | loather-work | this is my first time doing this, i inherited the system. |
| 05:04.53 | loather-work | ok, i'll be upgrading zaptel so yes. |
| 05:05.01 | [TK]D-Fender | loatherthere you have it |
| 05:05.11 | drmessano | If you do then you need to recompile zaptel but not before wanrouter but only after zaptel but not if libpri isn't compiled 3rd, 5th, and 8th |
| 05:05.24 | mosty | is there a maximum length for channel variables? |
| 05:05.26 | [TK]D-Fender | drmessano: Who's on 1st? |
| 05:05.26 | loather-work | libpri is already done :p |
| 05:07.04 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 05:08.28 | murdock_ut | How do I find out if this option "parkedcallsrepark" is in available in asterisk 1.6 features.conf per this bug report: http://bugs.digium.com/view.php?id=13390 |
| 05:10.08 | drmessano | Try the example config |
| 05:11.05 | murdock_ut | It's not there. |
| 05:11.41 | murdock_ut | I'm trying to find out why I can't repark a call that was parked using the new one-touch parking option. |
| 05:11.50 | murdock_ut | in 1.6 |
| 05:12.26 | murdock_ut | If the call times out and is answered the the person who parked the call they can repark it. But if someone picks up the parked call they cannot repark the call. |
| 05:13.26 | [TK]D-Fender | murdock_ut: Sure they can, just not with features.conf |
| 05:13.41 | [TK]D-Fender | murdock_ut: unless you do a little dialplan trickery. |
| 05:14.11 | drmessano | Why would that ever NOT work? |
| 05:14.40 | drmessano | If I park a call, you pick it up, then want to repark, it's no different than my originally parking the call |
| 05:14.52 | murdock_ut | That's how I see it. |
| 05:14.54 | drmessano | Theres no "wasparked" bit |
| 05:15.00 | drmessano | Sounds like operator error |
| 05:15.16 | murdock_ut | I can reproduce it on both my 1.6 systems. |
| 05:15.42 | [TK]D-Fender | murdock_ut: Whats to reproduce? Just look at how you RETRIEVE the parked call. that determines the rules |
| 05:15.46 | drmessano | Yes, and in this case, you would be the operator |
| 05:16.07 | [TK]D-Fender | murdock_ut: And this is only limited to features.conf. What shit phones are you using that you'd depend on it for this? |
| 05:16.42 | murdock_ut | I retrieve the call by dialing 701 which is where the call is placed. |
| 05:17.08 | murdock_ut | how else would I retrieve the call? |
| 05:17.42 | [TK]D-Fender | murdock_ut: You talk about "dialing 701" without paying attention to what it DOES, WHY, and what else YOU could be doing. |
| 05:18.44 | murdock_ut | Sorry let me be more precise. I dial 701 which fires off exten => 701,1,ParkedCall(701) |
| 05:18.53 | murdock_ut | Is that better? |
| 05:19.14 | drmessano | Ok, so now youve got the call |
| 05:19.20 | drmessano | you talk.. blah blah blah |
| 05:19.24 | drmessano | Then what? |
| 05:19.41 | [TK]D-Fender | murdock_ut: Shows your understanding, and yes, ParkedCall() does NOT support features.conf. There are ways areound this, and again only ends up enabling functionality you'd get on any decent phone WITHOUT depending on features.conf in the first place |
| 05:19.51 | murdock_ut | I press the button on my phone that is configured as a dtmf button to dial *4 |
| 05:20.35 | drmessano | and what happens? |
| 05:20.41 | murdock_ut | I'm using snom phones. They do have a park orbit feature, however the last time I used it, it would disconnect the call before * had a chance to tell you where it parked the call. |
| 05:21.01 | murdock_ut | drmessano: Nothing, I just hear dtmf tones and nothing happens. |
| 05:21.04 | [TK]D-Fender | murdock_ut: Transfer ->700 |
| 05:21.17 | [TK]D-Fender | murdock_ut: Who the hell needs features.conf? |
| 05:22.39 | murdock_ut | I'm not a 100% sure, but I don't think snom phone will subscribe to a hint if the button is configured as a transfer. |
| 05:22.58 | murdock_ut | Only as an extension or blf |
| 05:23.17 | murdock_ut | hold on strike that last statement. I don't subscribe to 700. |
| 05:23.37 | murdock_ut | Let me try that and see if it cuts off. |
| 05:23.47 | [TK]D-Fender | murdock_ut: Now is a great time to pay extreme attention to what you're doing... |
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| 05:33.12 | murdock_ut | Well when I use the transfer button on the phone and transfer to 701 it cuts asterisk off. The only way I've got it to work consistently is to configure a button to dial *3700 which is a blind transfer to 700. |
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| 05:35.15 | drmessano | Why are you transferring to 701? |
| 05:35.21 | drmessano | Transfer to 700 |
| 05:35.39 | murdock_ut | That is what I ment sorry. |
| 05:36.10 | drmessano | That sounds like soe dialplan foolishness |
| 05:36.12 | drmessano | some* |
| 05:36.41 | [TK]D-Fender | murdock_ut: If you get cut off then you simply don't know how to do an attended transfer properly |
| 05:36.55 | murdock_ut | I can also do a *4 which is the one touch park in features.conf and it works fine. |
| 05:37.15 | murdock_ut | The first time. |
| 05:37.20 | [TK]D-Fender | murdock_ut: Which is a dead end from the way you are picking up your call. |
| 05:37.31 | murdock_ut | How else would you pick up the call |
| 05:38.13 | [TK]D-Fender | murdock_ut: Think of what you could do in the dialplan to allow you to pick up the call and still be able to park it again. |
| 05:38.20 | [TK]D-Fender | murdock_ut: this is minor trickery. |
| 05:38.49 | murdock_ut | Ok, questions. After I pickup the call does it stay in that context? |
| 05:40.21 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 05:40.35 | [TK]D-Fender | murdock_ut: its all just DIALPLAN |
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| 06:14.54 | [TK]D-Fender | ok, checkout time... later all |
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| 06:50.00 | mchou | it's 'bricked' man :) |
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| 07:00.59 | LemensTS | anyone do perl and phpAGI both? |
| 07:01.35 | carrar | I do all my agi in perl |
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| 07:03.51 | LemensTS | Think it is more powerful language for working with asterisk? |
| 07:04.10 | carrar | depends who you ask and what you are trying to do I suppose |
| 07:04.15 | carrar | could ne subjective |
| 07:04.24 | carrar | also depends what languages you know |
| 07:04.29 | LemensTS | php |
| 07:04.31 | carrar | perl has been around a long time |
| 07:04.52 | carrar | then write it in php |
| 07:04.58 | carrar | use what you know |
| 07:05.04 | carrar | thats the beauty of it |
| 07:05.10 | carrar | use rex! |
| 07:05.11 | carrar | heh |
| 07:07.40 | LemensTS | :) Yea I just started using phpAGI, was curious on what other asterisk people wrote there agi's in. Ive wrote some long programs in AEL in the dialplan, what was I thinking lol. |
| 07:08.07 | carrar | I like perl so thats what I use |
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| 07:13.47 | ScribbleJ | Ok, I know this is a tricky one... any suggestion for a VOIP softphone that will run on a 2.4 linux kernel and support v4l video? Oh yeah, on 64mb ram machine. |
| 07:14.49 | mchou | ScribbleJ: come on man |
| 07:14.54 | ScribbleJ | I know, I know. |
| 07:15.01 | ScribbleJ | <sigh> |
| 07:15.07 | ScribbleJ | Aw well. |
| 07:15.11 | mchou | ScribbleJ: at least drop the vieo requirement |
| 07:15.17 | mchou | video* |
| 07:16.41 | mchou | ScribbleJ: besides, softphones suck anyways |
| 07:16.53 | ScribbleJ | Hah, doubtless. |
| 07:16.55 | mchou | crappy voice quality for the far end |
| 07:17.36 | ScribbleJ | I was just trying to find a way to make this old Xbox more useful... gunna send it to my dad and if I could get a way to /see/ him in the process it'd be a big win. |
| 07:18.21 | mchou | your dad?? |
| 07:18.37 | mchou | you goona configure the softphone for him? |
| 07:18.44 | mchou | gonna* |
| 07:19.05 | ScribbleJ | Yeah, that's the plan. |
| 07:19.09 | mchou | How old is he? |
| 07:19.47 | ScribbleJ | Uhm, 65-ish, his girlfriend has a kid who's about 10 though. He doesn't know/jack/ about technology... but I figure if the kid has anything fun in the house at least he can learn. |
| 07:21.28 | mchou | if you're going to do all the configuring you'd be way better off sending him a linksys PAP2 |
| 07:21.53 | mchou | be far more useful for him probably |
| 07:22.11 | mchou | but also a lil bit boring |
| 07:22.20 | ScribbleJ | Oh heck yeah. I'm sure I'll send him something to that effect down the line. |
| 07:23.02 | mchou | if you really want him to have fun, set up a jabber client |
| 07:23.24 | ScribbleJ | Oh? I don't play with jabber; isn't that just another IM protocol? |
| 07:23.43 | mchou | probably do v4l and the whole sheban on that box |
| 07:24.05 | mchou | pretty much |
| 07:24.17 | mchou | IM with video |
| 07:24.32 | ScribbleJ | Oh... is there some hardware client box for this? |
| 07:24.52 | ScribbleJ | Or you're just htinking the old Xbox could handle a jabber client. |
| 07:24.55 | mchou | maybe, havent looked into HW |
| 07:25.15 | mchou | old Xbox can handle jabber client |
| 07:25.32 | ScribbleJ | I was thinking myself, maybe just drop the VOIP idea and pop vlc on there to do the streaming... might work OK. I'm trying to compile an old version of linphone on there now though just to see. |
| 07:25.55 | mchou | bah |
| 07:26.02 | mchou | way too much work |
| 07:26.34 | mchou | jabber is probably easier than vlc |
| 07:26.38 | ScribbleJ | It's all too much work at this point, I've been screwing with this project in one form or another for two days now. |
| 07:26.45 | mchou | lol |
| 07:27.04 | mchou | 2 days is nothing in the scheme of things for this kind of stuff |
| 07:27.22 | mchou | you want plug and play get a pap2 |
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| 07:27.52 | ScribbleJ | I appreciate the recommendation... I'm going to be looking for some kind of FXS for my home real soon now. |
| 07:27.57 | ScribbleJ | So I might get one of those for myself. |
| 07:28.03 | mchou | be prepared to spend weeks |
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| 07:28.45 | ScribbleJ | Well, by 'some kinf of FXS" I mean anything I can plug a phone into, the PAP2 looks like it'd work just fine. |
| 07:29.13 | mchou | I recommend pap2 (unlocked) highly |
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| 07:30.02 | mchou | it's more full featured than even a lot of business desk phnes |
| 07:30.07 | mchou | phones* |
| 07:30.55 | mchou | ScribbleJ: I mean I dont know what you use for land line now but just getting one to play with is a lot of fun |
| 07:31.38 | mchou | that's not even taking the "useful" aspects into consideration |
| 07:31.42 | ScribbleJ | Yeah, haven't ha da land line in years; now that I pay for SIP from a couple providers, that's what I got. Got a cuople old deskphones gathering dust 'till I get something. PAP2 looks pretty nifty, but I guess I'm going to have to learn how to make sure I get one that's hackable. |
| 07:32.27 | mchou | ScribbleJ: nah. dont bother. just get a unlocked one retail |
| 07:32.58 | mchou | they're the same price as used locked so why not |
| 07:33.11 | mchou | save yourself the grief |
| 07:34.02 | mchou | local target was clearing out vta-vr for $20, and I tried hacking it (successfully) but the process took DAYS |
| 07:34.38 | ScribbleJ | Hrm... can I assume places that are just selling the thing, without saying VONAGE, are unlocked? Like at telephonydepot? |
| 07:34.39 | mchou | I returned it and git myself unlocked new pap2 retail at 2x the price |
| 07:35.24 | mchou | ScribbleJ: buy retail, and make sure it says PAP2T-NA. those are the unlocked versions |
| 07:35.47 | mchou | ScribbleJ: accept no substitutes |
| 07:35.47 | ScribbleJ | Right on, thanks. |
| 07:36.29 | mchou | ScribbleJ: what ip phones you have gathering dust? |
| 07:37.17 | ScribbleJ | Well, I meant regular phones; but I've got an officefull of some old IP phone system that does SIP, I just have to bother to go in and have a look at them. I'm not sure what they are; we use the nice Cisco stuff for our offices but inherited some other comnpany's cruft recently. |
| 07:37.32 | mchou | nice |
| 07:37.45 | ScribbleJ | I'm planning on getting a rack for my house and some mountable cases from the junk too. Whee! |
| 07:37.45 | ScribbleJ | heh |
| 07:37.59 | mchou | send some over her :) |
| 07:38.05 | mchou | here* |
| 07:38.37 | mchou | I need a decent desktop ip phone |
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| 09:11.24 | angryuser | hello |
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| 09:12.18 | angryuser | i need to verify if new messages exist for some mailbox and send some audio if true, is it possible ? |
| 09:12.31 | angryuser | sorry not a mailbox but a voicemail |
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| 09:31.02 | Stese | returns |
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| 09:42.47 | angryuser | i have instelled net-snmp net-snmp-devel net-snmp-utils , but still asterisk do net detect netsnmp , what am i missing ? |
| 09:42.51 | angryuser | installed* |
| 09:43.11 | angryuser | it's for res_snmp |
| 09:43.26 | Stese | Hmm, what is the error message |
| 09:44.06 | angryuser | Stese: there is no error message, i can not select it under "menuselect" |
| 09:44.39 | angryuser | i am under centos 5.2 |
| 09:46.59 | Stese | and heres me hoping it would be something obvious! |
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| 09:47.27 | alrs | angryuser: SNMP is broken |
| 09:48.02 | alrs | alrs: last I checked people had some luck getting it working in CentOS by recompiling the SNMP stuff by hand |
| 09:48.33 | angryuser | alrs: ;( ok i will try to do it |
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| 09:48.52 | alrs | angryuser: I haven't looked at it for almost a year, so things might be different. Check the email lists. |
| 09:49.34 | angryuser | alrs: i have found that bug, but it was in the middle of 2007 ;) |
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| 09:55.48 | phpboy | chan_dahdi.c: !! Got S-frame while link down <--- what is that suppose to mean? |
| 10:00.09 | angryuser | alrs: worked from sources |
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| 10:04.01 | a-s | When I translate from a codec to another using ast_translate, I do not how to insert a frame of silece in the case that the rtp packet does not arrive. |
| 10:04.13 | a-s | can someone help me please? |
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| 10:15.09 | virtualme123 | How do I know if dahdi_dummy is working and configured correctly with Asterisk? |
| 10:16.06 | virtualme123 | I run 'dahdi show status' and dahdi show channels and it looks fine, but how do I know it is doing anything on a live call? |
| 10:31.22 | a-s | does asterisk have support for silence for every codec? |
| 10:33.21 | angryuser | virtualme123: hello type "dahdi show status" |
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| 10:34.26 | phpboy | why in goodness name does this stupid PRI keep resetting itself :/ |
| 10:36.48 | nix8n82 | I had that problem once..I don't work for the company anymore..but it took a long time fighting with the phone company for it to work |
| 10:38.46 | phpboy | Yesh, that's the thing here |
| 10:39.08 | phpboy | the two E1's from the one telco work 100% fine |
| 10:39.19 | phpboy | the two from the other don't work at all |
| 10:39.26 | phpboy | well, notproperly at least |
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| 10:51.51 | virtualme123 | angryuser: I've run that with no errors, but does this just pretty much confirm that it is loaded? |
| 10:52.17 | jgoo | 12 months ago I bought Polycom handsets, 6 months ago the same - but now I am curious - what are the best / cost / feature / build quality sets to buy for office voip solutions? |
| 10:52.34 | alrs | Polycom. |
| 10:52.59 | jgoo | One things I am concerned about is wiring - I'd like a full wifi system this time - any issues there? I've seem the Linksys wifi adapters and SIP adapters for lines - what do you guys use? |
| 10:53.40 | alrs | jgoo: I do not recommend voip-over-wifi. |
| 10:53.43 | jgoo | I am thinking polycom - I have 8 buildings to connect, total 30 extensions - so wifi is certainly the way to go |
| 10:53.53 | jgoo | alrs, the hops cause too much delay? |
| 10:54.07 | alrs | Someone turns on the microwave and calls drop |
| 10:54.16 | jgoo | I can have high signal strength and minimize hops between repeaters etc |
| 10:54.20 | alrs | wifi is half duplex |
| 10:54.38 | alrs | very easy to have packet loss and jitter problems |
| 10:54.44 | jgoo | alrs - really? no way... so microwaves also stop normal wifi internet access? :-/ hrm. |
| 10:54.47 | virtualme123 | angryuser: When I'm on a live call what should I look for or monitor? |
| 10:54.48 | alrs | yes |
| 10:54.54 | alrs | that's why 2.4 is free to use |
| 10:54.57 | jgoo | Well, if we go fully wired... the costs of laying down cat5 is waaaay to high |
| 10:55.10 | alrs | you could go wireless building to building |
| 10:55.15 | angryuser | virtualme123: in older versions you could see ztdummy there, not sure about dahdi_dummy |
| 10:55.28 | alrs | using mikrotik, perhaps |
| 10:55.50 | jgoo | ok, so a p2p wifi connection, that could work, although there is copper between the buildings... any network solutions that can reuse copper? |
| 10:55.51 | alrs | that is with very directional static antennas |
| 10:56.09 | jgoo | (I mean, can I use some router / ATM system to use the copper wiring in my network) |
| 10:56.10 | alrs | phones don't need that much speed. If you have CAT3 you should be able to run 10bt |
| 10:56.24 | virtualme123 | angryuser: Yes thats right, I see dahdi_dummy. The thing I'm worried about is how do I know asterisk is using it, or is that confirmation enough? |
| 10:57.21 | jgoo | alrs, those isdn cards that work with the capi driver - I've used on in one setup, worked fine, But I have yet to try putting 4 cards in one machine - are there interrupt problems? |
| 10:57.46 | alrs | I'm in the US where BRI is nearly non-existent |
| 10:57.55 | jgoo | I have four incoming ISDN lines - what is the best solution for taking them? an OpenVox card or Digium or those cheap and cheerful ISDN cards? |
| 10:58.08 | jgoo | oh, true, that was the conversation I had last time too :p |
| 10:58.08 | angryuser | virtualme123: if dummy is not working well, you will see it during calls, bad/robo audio , audio degradation during day even with internal call's |
| 10:58.26 | jgoo | Shame, two channels on one wire, PRI / E1 - nice setups |
| 10:58.58 | virtualme123 | angryuser: That is what I was hoping to fix with the dummy loaded, thats the trouble ... :( Maybe it just hasn't fixed my problem. |
| 10:59.25 | alrs | virtualme: I'm just coming in. Are you running Asterisk in a vps? |
| 10:59.28 | jgoo | 4 port ISDN card is 300 euros... a 1 port ISDN card is 30 euros... |
| 10:59.37 | jgoo | I figure I can stuff 4 one port cards into a box... |
| 11:00.18 | virtualme123 | alrs: No not a vps |
| 11:00.29 | angryuser | virtualme123: you need a timing source then, if you are sure that it is timing |
| 11:01.13 | virtualme123 | angryuser: But I thought the dahdi_dummy would get a timing source from something like the usb drivers? |
| 11:01.23 | virtualme123 | Is that right? |
| 11:01.59 | virtualme123 | Also I assumed this part of it worked because when I run dadhi_test it sits there ticking away nicely |
| 11:02.00 | alrs | virtualme123: I don't think it uses USB any more. It is still not an optimal solution. |
| 11:03.22 | virtualme123 | alrs: Do you mean dahdi isn't an optimal solution or USB? |
| 11:03.33 | alrs | dahdi-dummy |
| 11:03.43 | alrs | maybe it's better, I haven't used it since ztdummy |
| 11:03.55 | virtualme123 | alrs: Oh, is there any other solution for timing?? |
| 11:04.01 | alrs | buy a card |
| 11:04.22 | alrs | or if you have another machine with a zaptel card you can pass the timing over ethernet |
| 11:05.46 | virtualme123 | alrs: Would passing it over the ethernet produce unwanted lag though? |
| 11:06.08 | alrs | I haven't noted any |
| 11:06.51 | angryuser | virtualme123: i heard there was some cheap timing sourse sold, from sangoma ? google |
| 11:07.13 | virtualme123 | alrs: ok so I do have a zaptel card on another machine maybe I can try using that. So do you believe that this dadhi_dummy is working but maybe just not very well? |
| 11:07.24 | virtualme123 | angryuser: I'll have a look |
| 11:11.04 | virtualme123 | angryuser: However I like alrs idea with the remote timing as might be more convienant for me at this time ... |
| 11:12.10 | alrs | http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE |
| 11:12.39 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
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| 11:14.55 | virtualme123 | alrs: ok reading that now ... :) |
| 11:22.34 | a-s | into a translation how can I insert a frame of silence ? |
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| 11:32.03 | virtualme123 | alrs: Just one point with my dahdi_dummy I feel I should mention is that when I run 'dahdi show status' show a 1 under alarm ... |
| 11:33.21 | virtualme123 | alrs: well I say 1 for alarm but this is it - Description Alarms IRQ bpviol CRC4 |
| 11:33.23 | virtualme123 | DAHDI_DUMMY/1 (source: Linux26) 1 UNCONFIGUR 0 0 0 |
| 11:33.49 | virtualme123 | alrs: but looking at it I'm not sure if that 1 lines up with the Alarm column or not ... |
| 11:35.40 | virtualme123 | angryuser: does that ring any bells with you? |
| 11:39.43 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
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| 11:45.57 | *** join/#asterisk riksta (n=rick@office.encompassmedia.co.uk) |
| 11:46.15 | riksta | does asterisk 1.6 still require a kernel driver for timing, for things like meetme? |
| 11:48.29 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 11:48.52 | *** join/#asterisk tokozedg (n=h3kt0r0@85.118.98.122) |
| 11:48.53 | virtualme123 | riksta: You could use dahdi_dummy, which replaces ztdummy, I'm struggling to get it to work for me however ... |
| 11:49.17 | virtualme123 | riksta: think that works with 1.6 |
| 11:49.20 | tokozedg | hi, how to convert asteris record files (GSM) to mp3 ? |
| 11:49.48 | riksta | virtualme123: yeah - i'm in a xen virtual machine so was just wondering |
| 11:50.53 | tokozedg | is it possible? |
| 11:51.53 | virtualme123 | tokozedg: think there is a command on the client to convert audio files that might help |
| 11:52.20 | tokozedg | what kind of command? |
| 11:53.25 | virtualme123 | tokozedg: Thats all I've got but should be able to find it on voip-info.org |
| 11:53.55 | tokozedg | virtualme123: ok thank you |
| 11:54.42 | tokozedg | and is there gsm online player? |
| 11:56.50 | virtualme123 | tokozedg: Would have thought so, but don't know a name.. |
| 12:07.12 | virtualme123 | alrs: I believe dahdi does away with the use of the old zapata.conf(where it suggests setting up the dynamic driver), so where do I do the changes with dahdi? |
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| 12:12.34 | Rico29 | hello all |
| 12:12.39 | Rico29 | I have a DTMF problem |
| 12:12.50 | Rico29 | i'm making a call from my * to a GSM |
| 12:13.02 | Rico29 | by placing a file in /var/spool/asterisk/outgoing |
| 12:13.32 | Rico29 | everything works fine, except that DTMF are not send from GSM to my * serv |
| 12:13.53 | Rico29 | i'v tried with dtmfmode = auto and dtmfmode=rfc2833 |
| 12:13.58 | Rico29 | in iax.conn |
| 12:13.59 | Rico29 | conf |
| 12:14.12 | Rico29 | (i pass through an IAX trunk for placing my call) |
| 12:16.23 | Rico29 | I have disabled jitterbuffer |
| 12:16.25 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:17.15 | *** join/#asterisk ming_zym (n=ming_zym@125.39.45.55) |
| 12:17.46 | Rico29 | can anyone help me please ? |
| 12:21.54 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 12:23.33 | virtualme123 | Rico29: Can't think but is the dtmfmode set anywhere else? |
| 12:24.00 | virtualme123 | Rico29: I would say auto you see, unless it is being overwritten ...? |
| 12:24.03 | Rico29 | I only use IAX |
| 12:24.20 | Rico29 | and I set dtmfmode=rfc2833 ( which is set by default) |
| 12:24.47 | Rico29 | i'll try with another VoIP line |
| 12:25.41 | Rico29 | same problem with a VoIP line (SIP) |
| 12:27.12 | virtualme123 | Rico29: I found this note in my iax.conf - dtmfmode=inband only works with ulaw or alaw |
| 12:27.24 | virtualme123 | Rico29: not sure if that is the issue? |
| 12:29.40 | Rico29 | i'll try |
| 12:29.48 | Rico29 | but when I'm using GSM codec... ? |
| 12:31.05 | virtualme123 | Well I send everything as ulaw and it all works for me. |
| 12:31.36 | Rico29 | doesn't work better with dtmfmode=inband |
| 12:32.45 | virtualme123 | No I must have misunderstood that line I found, sorry |
| 12:32.46 | *** join/#asterisk shodan (n=shodan@197.58-ppp.3menatwork.com) |
| 12:33.12 | virtualme123 | probably won't help ... |
| 12:33.31 | shodan | ok, I'm tired of paying rogers 125$ per month, time to set me up an asterisk box again :) |
| 12:41.15 | *** join/#asterisk stevie[xxx] (n=stevie@85.183.21.87) |
| 12:42.49 | stevie[xxx] | hello, there was a patch for 1.6.0 version and this url http://bugs.digium.com/view.php?id=13958 tells me that it was added in the source. but if i open the latest 1.6.0.3 src, there is no IGNOREDSPVERSION , did i missed sth? |
| 12:47.55 | stevie[xxx] | there is also no r165180 listed in the changelog |
| 12:52.30 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:52.30 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:52.34 | *** join/#asterisk kotique (n=picachu@78.129.232.75) |
| 12:52.50 | kotique | how do I set ring tone for particular dial command ? |
| 12:59.42 | shodan | is there a way to find out if, for example, 450-755 can call 450-668 locally or if it is long-distance ? (other than going there and trying it) |
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| 13:14.35 | GazD | can anyone tell me what an Index telephone system is please? |
| 13:15.52 | eppigy | TRABAJO |
| 13:18.10 | *** part/#asterisk imchandave (n=chandave@ip155.bb203.pacific.net.hk) |
| 13:18.20 | *** join/#asterisk imchandave (n=chandave@ip155.bb203.pacific.net.hk) |
| 13:20.20 | virtualme123 | What is an exceptable tollerance for dahdi_test? I'm getting the odd 93% with a majority 98-99%, I've recently heard that the kernel version can effect timing. |
| 13:21.25 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:21.33 | eppigy | hello [TK]D-Fender |
| 13:21.38 | [TK]D-Fender | you are dave |
| 13:21.52 | eppigy | virtualme123: this can be cause by many things |
| 13:21.57 | eppigy | IRQ sharing |
| 13:22.03 | eppigy | Hard drive |
| 13:22.05 | eppigy | etc. |
| 13:22.29 | eppigy | try lspci -v |
| 13:22.37 | virtualme123 | eppigy: So do you think a drop to 93% is not expected. |
| 13:22.46 | eppigy | no that is terrible |
| 13:22.58 | eppigy | it should be 99.8 |
| 13:23.02 | eppigy | consistently |
| 13:23.34 | gambler1 | Hi, I am trying to handle CHANUNAVAIL error in my dial plan but it seems that * just hangup the chan and does not play some file for example. Anyone have experience with this in * 1.6.0.1? |
| 13:24.04 | [TK]D-Fender | gambler1: * doesn't just "play a file", you have to tell it to in your dialplan. |
| 13:24.10 | eppigy | virtualme123: if you run hdparm -t <device name> |
| 13:24.13 | [TK]D-Fender | gambler1: pastebin what you're doing now. |
| 13:24.14 | eppigy | while running zttest |
| 13:24.15 | [TK]D-Fender | ~pb |
| 13:24.16 | jbot | [~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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 13:24.18 | [TK]D-Fender | ^^^^^^^^ |
| 13:24.23 | eppigy | and your accuracy rate drops sharply |
| 13:24.27 | eppigy | replace your HD |
| 13:24.35 | virtualme123 | eppigy: You may have just saved my life, being faffing with this dahdi stuff for ages not thinking it was working, ok will try your test .. |
| 13:24.49 | gambler1 | [TK]D-Fender: I told * to play a file.. I will use pastebin and send a link |
| 13:25.02 | eppigy | also make sure your cards have their own irq |
| 13:25.05 | eppigy | if possible |
| 13:26.02 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-76.static.insightbb.com) |
| 13:27.24 | gambler1 | [TK]D-Fender: http://pastebin.com/d7979fdeb |
| 13:30.10 | [TK]D-Fender | gambler1: What is that SIP device youre checking? |
| 13:30.38 | [TK]D-Fender | gambler1: I do not believe its valid to specify an IP like that... |
| 13:30.51 | virtualme123 | eppigy: Well I ran hdparm -t /dev/md0 and the dadhi_test dropped to 85%. So I'm running a Software Raid could that be an issue? |
| 13:30.55 | gambler1 | [TK]D-Fender: our upstream provider |
| 13:31.21 | [TK]D-Fender | gambler1: What are you actually trying to test for? |
| 13:31.45 | eppigy | virtualme123: it is possible |
| 13:31.50 | gambler1 | [TK]D-Fender: the problem I am trying to solve is very hmmm strange |
| 13:32.10 | eppigy | you need to reemove softraid |
| 13:32.16 | eppigy | use lvm |
| 13:32.18 | eppigy | or something |
| 13:32.25 | eppigy | or use a real raid card |
| 13:32.27 | gambler1 | [TK]D-Fender: when I use cisco to call some dest and I get address incomplete error the chan just hangup |
| 13:32.34 | eppigy | I mean it could be a bad drive as well |
| 13:33.03 | gambler1 | [TK]D-Fender: but when I use sip hpone (linksys, softphone) I get a cdr no matter if call was successful or not |
| 13:33.18 | gambler1 | gambler1: if you understand me because of my bad english :) |
| 13:33.29 | virtualme123 | eppigy: Well it is a brand you hosted server, so not sure but could be the hard drive ... ? |
| 13:33.51 | gambler1 | /s/gambler1/[TK]D-Fender/ |
| 13:34.10 | gambler1 | nevermind my search and replace :) |
| 13:34.57 | [TK]D-Fender | gambler1: Show me your whole exten. |
| 13:35.34 | gambler1 | [TK]D-Fender: is it possible to not show on public chan? |
| 13:35.58 | [TK]D-Fender | gambler1: PM |
| 13:38.58 | [TK]D-Fender | gambler1: I don't believe your use of DEVICE_STATE is proper there. use ${DIALSTATUS} to check the result |
| 13:40.13 | [TK]D-Fender | gambler1: I also see nothing in your new PB that would deserve being classed as "private" |
| 13:42.36 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
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| 13:43.55 | kotique | <PROTECTED> |
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| 14:04.49 | virtualme123 | eppigy: Going to put Asterisk on a new machine without Software Raid and test it out. Many thanks! |
| 14:12.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:12.34 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:88c:c7d0:805e:9d44) |
| 14:13.35 | eppigy | virtualme123: yw |
| 14:13.43 | eppigy | TRABAJO |
| 14:17.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 14:20.40 | ScriptFanix | I'm a bit confused : what *is* NBS (Network Broadcast Sound)? |
| 14:23.00 | ScriptFanix | google didn't help me much, as almost every search results are asterisk related, and none give a definition |
| 14:25.01 | *** join/#asterisk Zeeeeeeek (n=Zeeek@bdx.resmo.net) |
| 14:25.13 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:25.27 | Zeeeeeeek | hello to each and every one of you a happy new year |
| 14:25.39 | Zeeeeeeek | was this channel down a few moments ago? |
| 14:26.37 | Zeeeeeeek | is it down now? |
| 14:26.43 | Zeeeeeeek | am I? |
| 14:27.03 | kannan | Zeeeeeeek , no |
| 14:27.30 | Zeeeeeeek | must be local problems |
| 14:28.30 | phpboy | :/ |
| 14:30.21 | phpboy | ERROR[5359] chan_dahdi.c: !! Got I-frame while link state 8 <--- what would this error generally mean? |
| 14:33.32 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-4c5386e56d55316b) |
| 14:33.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:38.33 | Katty | good morning all you beautiful people! |
| 14:39.27 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-d8ed521de8e4f223) |
| 14:39.51 | *** join/#asterisk rethus (n=rethus@xdsl-84-44-158-104.netcologne.de) |
| 14:40.33 | rethus | i use class phpagi-asmanager()... now i want to use phpagi the AGI()-Klaass itself. |
| 14:40.58 | rethus | i have a phpagi.conf in /etc/asterisk, but there is only a section für asmanager.. not for php-agi |
| 14:41.13 | rethus | what have i to do to get access via agi to asterisk? |
| 14:41.19 | jaytee | hugs Katty |
| 14:41.24 | Katty | jayyyteeaaaa! |
| 14:41.26 | Katty | hugs jaytee |
| 14:41.38 | Katty | how are you faring this morning mister tee |
| 14:41.41 | rethus | have i to enter the same data like in asmanager-section? |
| 14:41.50 | jaytee | I'm fair to poor, poor to middlin |
| 14:42.06 | Katty | checks jaytee for a temperature. |
| 14:43.43 | Katty | jaytee: are you feeling okay? |
| 14:43.48 | Katty | jaytee: did you forget it's friday? |
| 14:45.06 | jaytee | friday only matters to people whose week actually comes to an "end". It's all the same week for me at this job, everyday is either monday or wednesday with an occassional tuesday thrown in here and there at random. |
| 14:45.59 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
| 14:46.01 | jasonwoot | werd |
| 14:46.10 | jaytee | werd? |
| 14:46.21 | Katty | someone needs to take a break and feed some penguins. |
| 14:46.27 | Katty | orders fish for jaytee |
| 14:46.32 | jaytee | is that like almost weird? or a cooler way of saying "word" |
| 14:46.51 | Katty | not sure. werd does not parse properly. |
| 14:47.01 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 14:47.13 | Nugget | http://macnugget.org/photos/strange/werd |
| 14:47.44 | Katty | hugs on Nugget |
| 14:47.45 | Stese | Arrrggghhh my eyes! |
| 14:47.53 | Stese | Why!!!! |
| 14:47.55 | jaytee | Nugget, "well, my days of not takin you seriously are certainly comin to a middle!" |
| 14:48.07 | jasonwoot | jaytee: werd= what the kids say these days |
| 14:48.12 | Nugget | messes up jaytee's hair |
| 14:48.12 | Katty | Nugget: <3 |
| 14:48.21 | Katty | Nugget: that's superdeduper. |
| 14:48.27 | Katty | Nugget: may i blog that? |
| 14:48.34 | jasonwoot | I'm so hip, I have a hard time seeing over my own pelvis |
| 14:49.30 | Stese | jaytee > Where is that from? |
| 14:49.50 | Zeeeeeeeeek | anyone tried playing with VoicePHP yet? |
| 14:49.57 | Zeeeeeeeeek | {{{{{Katty}}}}} |
| 14:49.58 | Katty | you, apparently. |
| 14:50.03 | Katty | hugs Zeeeeeeeeek |
| 14:50.05 | Zeeeeeeeeek | my life is an open book |
| 14:50.11 | Katty | my, but you have an unusually large ammount of es in you today. |
| 14:50.23 | Zeeeeeeeeek | has tried a lot of things, too many things |
| 14:50.37 | Zeeeeeeeeek | lots of leeeeeeeead in my pencil |
| 14:50.52 | jasonwoot | what's prozac like? |
| 14:50.53 | Zeeeeeeeeek | also a keeeeeeeeeeeeeey is stuck on the kbd |
| 14:51.08 | Zeeeeeeeeek | jasonwoot: legal, so i wouldn't know! |
| 14:51.18 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 14:51.26 | Zeeeeeeeeek | But I can tell you, THC turns you into a martian |
| 14:51.38 | Katty | Never had any experience with Prozac |
| 14:51.40 | Zeeeeeeeeek | but this is not asterisk talk |
| 14:51.46 | Katty | I can tell you that dorvaset makes me feel drunk. |
| 14:51.49 | Zeeeeeeeeek | back to VoicePHP! |
| 14:51.49 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-94406abf42416def) |
| 14:51.57 | eppigy | this is asterisk talk for sure |
| 14:51.58 | Assid | voicephp? |
| 14:52.04 | Zeeeeeeeeek | beer, wine, whiskey make me feel drunk |
| 14:52.05 | eppigy | lets eat some oxycontin |
| 14:52.06 | Katty | never heard of it, actually zeek |
| 14:52.10 | Zeeeeeeeeek | VoicePHP!! Yes!! |
| 14:52.14 | Zeeeeeeeeek | WHAT? |
| 14:52.15 | jasonwoot | Zeeeeeeek: coping with depression is in the Asterisk manual, or should be |
| 14:52.16 | Assid | whats that |
| 14:52.20 | Zeeeeeeeeek | http://voicephp.com |
| 14:52.30 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 14:52.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:52.50 | Zeeeeeeeeek | Very cool (and h-gee, coincidentally the subject of a certain weekly live teleconference that begins in just over an hour) |
| 14:53.05 | Zeeeeeeeeek | VoicePHP is like VoiceXML without the XML |
| 14:53.42 | Katty | looks |
| 14:53.45 | Zeeeek | I still can't lose the extra e |
| 14:54.04 | Zeeeek | so it's like echo "hi Katty" will say that instead oif printing it |
| 14:54.04 | Katty | ah, meh. |
| 14:54.27 | Katty | I would have to find some useful, practical application of it. |
| 14:54.47 | Zeeeek | you can call my stupid demo here: (567) 244-9762 |
| 14:55.28 | Zeeeek | It's mostly of interest if you already know php or have people that do. |
| 14:55.35 | Katty | yeah. |
| 14:55.40 | Katty | i was forced into ASP, sadly. |
| 14:55.43 | Zeeeek | It's neat tho |
| 14:55.46 | Katty | but i do have an ole dusty php book somewhere |
| 14:55.48 | Zeeeek | Ewwwwwwww. |
| 14:55.57 | Zeeeek | there's only one thing worse than ASP |
| 14:56.02 | Assid | hrmm i gotta look into it |
| 14:56.05 | Zeeeek | aspx .NET |
| 14:56.13 | Zeeeek | hates .net |
| 14:56.14 | jaytee | Stese, that's from Firefly, the "Our Mrs. Reynolds" episode |
| 14:56.16 | Katty | i am forced to agree. |
| 14:56.22 | Zeeeek | sorry, my prejudice comes out |
| 14:56.26 | Katty | mmm, Firefly |
| 14:56.29 | Katty | oh! |
| 14:56.31 | Katty | speaking of firefly |
| 14:56.31 | Zeeeek | I absolutely ABHOR .net |
| 14:56.37 | Katty | they've discovered a new squid! |
| 14:56.38 | Zeeeek | I like firefyl |
| 14:56.43 | Katty | Dana Octopus Squid |
| 14:56.52 | Zeeeek | delecious on toast |
| 14:56.58 | Katty | it is a deep sea bioluminescent squid which glows bright blue when attacking food |
| 14:57.09 | Katty | it's gorgeious. |
| 14:57.12 | Zeeeek | $but sadly, eating squid on toast is deprecated |
| 14:57.13 | Katty | ...in terms of squid. |
| 14:57.25 | jaytee | mmmmm, calamari |
| 14:57.28 | Zeeeek | so it can replace those neat blue LEDs? |
| 14:57.29 | Assid | okay time to hit back to work |
| 14:57.32 | Katty | firefly squid are also pretty |
| 14:57.36 | jaytee | with aoili sauce for dipping |
| 14:57.41 | Zeeeek | work? There's no room for that word here |
| 14:57.58 | *** join/#asterisk moy (n=moy@74.12.127.97) |
| 14:58.06 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:58.06 | Zeeeek | aioli sauce is among the possible candidates for the better ideas the French have had |
| 14:58.08 | Katty | werd. |
| 14:58.19 | Katty | oh! and champaign |
| 14:58.32 | Zeeeek | Champagne is a good one, too |
| 14:58.49 | Zeeeek | someone stole my Zeeek name :( |
| 14:59.02 | Zeeeek | ~seen Zeeek |
| 14:59.04 | jbot | zeeek is currently on #debian (1h 16m 45s) #openmoko (1h 16m 45s), last said: 'poop'. |
| 14:59.04 | Katty | :< |
| 14:59.11 | jjshoe | in asterisk 1.6 are ztcfg and fxotune still applicable? |
| 14:59.51 | jjshoe | I'm guessing not |
| 15:00.07 | *** part/#asterisk elfguy516 (n=elfguy51@96.56.103.35) |
| 15:00.31 | Zeeeek | I am registered as Zeeek so how does that happen? |
| 15:01.05 | *** join/#asterisk jarekk (n=jarek@h-81-15-194-37.dolsat.pl) |
| 15:01.13 | Katty | kick it |
| 15:01.19 | jarekk | hi all |
| 15:01.22 | Zeeeek | I don't have the power |
| 15:01.35 | Zeeeek | I've never been able to master IRC !ç |
| 15:01.48 | Katty | it's /msg nickserv ghost Zeeek whateveryourpasswordwas |
| 15:02.24 | Zeeeek | Yeah did that and it says ok, but the zeeek is still there. Is it case sensitive? |
| 15:02.54 | Katty | probably |
| 15:03.12 | Katty | after you kick you should do a whois |
| 15:04.19 | Katty | horay! you made it! |
| 15:04.25 | Katty | (frodo! don't wear the ring!) |
| 15:04.34 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:04.35 | Zeeek | so it must be an issue of the two cases? |
| 15:04.42 | Sargun_screen | Hi Katty |
| 15:04.46 | Sargun_screen | Hi Zeeek |
| 15:04.49 | Sargun_screen | Morning, how are you? |
| 15:05.01 | Katty | lil sleepy actually |
| 15:05.03 | Katty | hugs anthm |
| 15:05.12 | Zeeek | hello all |
| 15:05.14 | Sargun_screen | Katty: what TZ are you in again? |
| 15:05.23 | anthm | =D |
| 15:05.33 | Sargun_screen | Man, I love cak. |
| 15:05.33 | Katty | REF: http://www.youtube.com/watch?v=SWf3iJjqYCM |
| 15:05.36 | Sargun_screen | Cake even. |
| 15:05.46 | eppigy | you already said you love cak |
| 15:05.50 | eppigy | there is no going back |
| 15:05.57 | Zeeek | "cak" heh, that was the term used for female many years ago |
| 15:06.07 | Katty | orly |
| 15:06.11 | Zeeek | "get me some cak" |
| 15:06.17 | Sargun_screen | Zeeek: well, I love that too. |
| 15:06.18 | Katty | that parses as yak |
| 15:06.29 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 15:06.30 | lowtek | Katty: gay++ |
| 15:06.34 | eppigy | i was pretty sure it was heavily accented cock |
| 15:06.44 | Katty | lowtek: baroo? |
| 15:06.50 | eppigy | i mean lets be honest here |
| 15:06.55 | lowtek | lol, the video |
| 15:07.00 | Sargun_screen | o.O |
| 15:07.03 | Katty | lowtek: oh, right. |
| 15:07.10 | Katty | (the magical blingbling!) |
| 15:07.24 | Zeeek | no way. It was "cack" as in cackle maybe |
| 15:07.28 | Sargun_screen | Man, asterisk is far from "real" telecom people, who are usually ugly old men with no sense of style, hapiness, and fun. |
| 15:07.40 | Zeeek | gimmie a break this is sin city here |
| 15:07.43 | Katty | we're not real telcom people |
| 15:07.45 | Katty | we're data people |
| 15:07.46 | Sargun_screen | happiness. |
| 15:07.48 | Katty | (or at least me) |
| 15:07.49 | eppigy | my style is impeccable |
| 15:07.49 | Zeeek | AND we're all major geeks for VoIP |
| 15:08.00 | Katty | YOU"RE impeccable |
| 15:08.02 | Sargun_screen | Katty: Network admin here. |
| 15:08.02 | Zeeek | data people? Get out |
| 15:08.10 | Katty | Zeeek: stuff it, or you get another hug. |
| 15:08.12 | Zeeek | don't try pecking me to find out |
| 15:08.13 | Sargun_screen | Data is sexy. |
| 15:08.26 | eppigy | Katty: :D |
| 15:08.30 | Zeeek | Datum is good. Data is kinky |
| 15:08.35 | Katty | waits for reply |
| 15:08.38 | Katty | eyes eppigy |
| 15:08.40 | Katty | come on then |
| 15:08.40 | jarekk | I have a question about cdr_pgsql in asterisk1.6.0.3, where can I set path to pgsql schema ? |
| 15:08.42 | Katty | let's get those insults going |
| 15:08.49 | eppigy | GIRL U LOOKIN KINDA GOOD |
| 15:08.54 | Katty | facepalm |
| 15:08.55 | Katty | nonono |
| 15:08.57 | Sargun_screen | Prolonging the Magic [Explicit]/3 - Never There [Explicit] |
| 15:09.06 | eppigy | :[ |
| 15:09.06 | Katty | let's try this agian. |
| 15:09.07 | Sargun_screen | haha |
| 15:09.09 | *** join/#asterisk UQlev (n=yuriy@proton.sallbay.com) |
| 15:09.13 | lowtek | looks for the exit ... |
| 15:09.23 | Katty | 09:08 < eppigy> my style is impeccable |
| 15:09.26 | Sargun_screen | I apparently drove the channel nuts. |
| 15:09.28 | Katty | you're impeccable |
| 15:09.33 | Katty | you're FACE is impeccable |
| 15:09.41 | Katty | waits. |
| 15:09.44 | Sargun_screen | Katty: I belive it's your. |
| 15:09.47 | eppigy | i'll make your mouth impeccable |
| 15:09.47 | Sargun_screen | ewadewaoijdijwad |
| 15:09.49 | Sargun_screen | Nevermind |
| 15:09.52 | Sargun_screen | it's far too early |
| 15:09.54 | Sargun_screen | and I misread that |
| 15:09.56 | Katty | sad. |
| 15:10.01 | Sargun_screen | shotos himself in face. |
| 15:10.01 | lowtek | says load "*",8,1 |
| 15:10.04 | Katty | the appropriate reply is your mom's face is impeccable |
| 15:10.13 | eppigy | i see |
| 15:10.20 | Katty | we need to send eppigy back to insults 101 |
| 15:10.31 | eppigy | YOUR SECOND COUSIN'S FACE IS IMPECCABLE |
| 15:10.43 | *** join/#asterisk maddog01 (n=minotaur@mail.upperjamestoyota.ca) |
| 15:10.49 | Katty | wonders if she has a second cousin ^_- |
| 15:10.52 | eppigy | Definitions of impeccable on the Web: |
| 15:10.53 | eppigy | * faultless: without fault or error; "faultless logic"; "speaks impeccable French"; "timing and technique were immaculate"; "an immaculate record" |
| 15:10.56 | eppigy | for one thing |
| 15:11.09 | eppigy | i dont think that statement is insulting |
| 15:11.15 | Katty | of course it isn't. |
| 15:11.24 | Katty | what's that have to do with anything? |
| 15:11.26 | eppigy | i see |
| 15:11.32 | eppigy | 8[] |
| 15:11.56 | Katty | <bkw> NEXT!!!!! |
| 15:12.21 | Katty | i passed my samsung certifications |
| 15:12.23 | Katty | 94.67% |
| 15:12.25 | Katty | i am pleased. |
| 15:12.41 | Katty | another shiny certification paper to frame. |
| 15:12.42 | Zeeek | Katty: can you get me a deal on a 42" LCD ? |
| 15:12.50 | eppigy | http://pastebin.com/d4534f29 |
| 15:12.56 | eppigy | i am getting this error |
| 15:12.59 | eppigy | during compile |
| 15:13.02 | Katty | Zeeek: no, just stuff in the telcomm division |
| 15:13.03 | eppigy | what is the deal |
| 15:13.04 | eppigy | ? |
| 15:13.10 | Katty | Zeeek: try newegg. |
| 15:13.49 | Sargun_screen | Katty: Dude, I was dealing with someone from Samsung telecom, they were complete idiots. |
| 15:13.57 | Sargun_screen | Who do you guys hire for your peering director? |
| 15:13.58 | Katty | Sargun_screen: I am not a Dude. |
| 15:13.59 | *** join/#asterisk AlienPenguin (n=Miranda@151.13.106.31) |
| 15:14.14 | Sargun_screen | Dudete? |
| 15:14.16 | Katty | Sargun_screen: and i don't work for samsung, we're just a certified partner now |
| 15:14.23 | Katty | <PROTECTED> |
| 15:14.25 | Sargun_screen | Ah, how do you enjoy dealing with them? |
| 15:14.30 | eppigy | hello i am dave |
| 15:14.32 | errr | hi Katty =) |
| 15:14.35 | Katty | getting tech support is a bitch. |
| 15:14.43 | Katty | but otherwise, it's descent |
| 15:14.44 | eppigy | hello what does this mean |
| 15:14.46 | eppigy | <PROTECTED> |
| 15:14.50 | Katty | lib |
| 15:15.33 | Sargun_screen | Katty: What do you guys do with Samsung exactly? Do you use their network for transit, and/or peer with their network? |
| 15:15.39 | Sargun_screen | whther ist be their voice or data network? |
| 15:15.39 | *** join/#asterisk fun330 (n=manning_@2.223.188.72.cfl.res.rr.com) |
| 15:15.54 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:16.19 | Katty | Sargun_screen: we install mostly 7100 phone systems. |
| 15:16.19 | AlienPenguin | hi all, i am experiencing the following problem: when trying to trasnfer a bridged call between different technologies (sip/pstn) i get the following error messages builtin_atxfer: Did not read data. i read over internet that some ppl complained about it and one solution was to set the __TRANSFER_CONTEXT to something sensible, but is this the right solution? is there a more "clean" workaround? |
| 15:16.46 | Katty | eppigy: i dunno, i don't think i've seen that problem before )= |
| 15:17.38 | eppigy | D: |
| 15:17.44 | eppigy | neither has i |
| 15:17.44 | Katty | i sowwie :< |
| 15:17.48 | eppigy | its all good |
| 15:18.18 | eppigy | nothing a little HAIR TRIGGER VIOLENCE cant fix |
| 15:18.23 | [TK]D-Fender | AlienPenguin: What phones are your users using? |
| 15:18.24 | Katty | yep |
| 15:18.25 | Zeeek | BANG |
| 15:18.38 | Katty | does not comment. |
| 15:18.57 | Katty | http://www.youtube.com/watch?v=32_tkje6NjU <- Distraction! |
| 15:19.48 | AlienPenguin | [TK]D-Fender : the ip phone is a grandstream gxp 2000 (but i tried also with a linksys) and the pstn is bridged trhough a patton |
| 15:20.02 | jaytee | brb |
| 15:20.14 | [TK]D-Fender | AlienPenguin: And you're trying to transfer from the GS? |
| 15:20.14 | Katty | :< |
| 15:20.37 | Sargun_screen | Katty: Ah, do you don't deal with any of samsung's "Big iron" |
| 15:21.07 | Katty | If i did, they'd probably call me Sargent Slaughter. |
| 15:21.11 | AlienPenguin | [TK]D-Fender: yes, gs (or LS) calls through the patton then i try to transfer (correct params were provided to the Dial() app) after the first dtmf digit i get that error and the call is resumed |
| 15:21.39 | AlienPenguin | i found these posts showing the same behaviour |
| 15:21.40 | AlienPenguin | http://www.asteriskguru.com/archives/asterisk-dev-builtin-transfer-vt90183.html |
| 15:21.46 | AlienPenguin | http://lists.digium.com/pipermail/asterisk-users/2005-November/129112.html |
| 15:21.58 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:22.12 | fun330 | does anyone know how to do a follow me? |
| 15:22.25 | [TK]D-Fender | AlienPenguin: You are using a real SIP phone, you should not be using DTMF transfers. use your phone's NATIVE trasfter features. |
| 15:23.10 | Sargun_screen | Wow, there are 1028 users part of the "asterisk" group on facebook. |
| 15:23.16 | [TK]D-Fender | fun330: Dial whatever you feel like dialing and on failure or whatever go dial something else afterwards. Maybe prompt for input first to offer it, etc. |
| 15:23.54 | AlienPenguin | [TK]D-Fender : why is that? bridged calls within the same technologies work as expected, and i didnt find any advice on "not-to-use" dtmf transfers |
| 15:24.44 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:24.51 | jaytee | I am back |
| 15:24.55 | jaytee | I am not dave |
| 15:25.13 | Zeeek | Happy New Year [TK]D |
| 15:25.22 | Zeeek | dave's not here |
| 15:25.43 | coppice | is hal here? |
| 15:25.54 | Zeeek | I can't answer that, steve |
| 15:26.02 | coppice | open the firewall port, hal |
| 15:26.10 | Zeeek | I'm sorry I can't do that |
| 15:26.33 | Zeeek | I can't even control IRC today |
| 15:26.40 | fun330 | can someone help me with follow me |
| 15:26.49 | lowtek | fun330: What's your question? |
| 15:27.44 | fun330 | is follow me just an advacnced call forward that will go back to the voicemail on the system? |
| 15:28.10 | lowtek | fun330: essentially |
| 15:28.37 | fun330 | and how can i set it up for all extensions not just selected extetions? |
| 15:28.59 | lowtek | fun330: _8xx,1,followme(args) |
| 15:29.02 | eppigy | JUST DANCE |
| 15:29.55 | lowtek | fun330: docs -> http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe |
| 15:30.22 | fun330 | do i have to configure followme.conf too? |
| 15:30.26 | stupidnic | Is this the best way to handle a rollover situation http://pastebin.com/d27557d4b |
| 15:31.17 | lowtek | fun330: yes, that's where you stack up your numbers for the extensions ... |
| 15:31.32 | lowtek | fun330: It's easy, see followme.conf about half way down on that page I posted. |
| 15:31.44 | fun330 | yeah |
| 15:32.47 | fun330 | my ext numbers are 3XX and voicemail is 6XX will that screw with exten => _4411,5,VoiceMail(u${EXTEN}) |
| 15:32.58 | fun330 | should i change ext # and voicemail to be the same |
| 15:35.20 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 15:35.50 | lowtek | fun330: You just add your VoiceMail() cmd after the FollowMe() command ... FollowMe() by itself doesn't do anything but place calls or fail (allowing fall-through to VoiceMail() or whatever) |
| 15:36.44 | lowtek | fun330: so you can keep whatever extension logic you like |
| 15:37.08 | fun330 | okay, so the follow me will call the designated numbers local and remote, can the user enter in the follow me remote number? |
| 15:37.30 | lowtek | fun330: If I understand your question, "No". |
| 15:37.46 | jgoo | I am researching an office voip system over wifi - I've heard there are disadvantages to wifi, but then again I've read a lot of forums regarding people using it (and used the nifty calculator) - has anyone recently implemented this and has a cost breakdown for the hardware they've used? |
| 15:38.13 | lowtek | fun330: They have to be specified by the * administrator via follome.conf (or some custom app you've written that the user can interact with) |
| 15:38.23 | jgoo | I did a wired solution (20 phones) polycom 12 months ago, and another 8 phones 6 months ago, now I am going wifi only - so I need ethernet/wifi bridges and other thigns... thinking about channels etc |
| 15:38.42 | lowtek | jgoo: Local PBX or remote via wifi? |
| 15:39.34 | jgoo | local PBX |
| 15:39.44 | jgoo | 4 ISDN lines |
| 15:40.17 | fun330 | okay so i need to get the users who want to do follow me a head of time and enter it, do i have to use seperate contexts for each follow me exteions |
| 15:40.18 | jgoo | BRI - I am thinking either a 300 euro 4 channel ISDN card OpenVox or something, or 4x 30 euro ISDN cards (if anyone knows they work, I have one working fine at a test rig, with just one card) |
| 15:40.20 | fun330 | extension |
| 15:40.26 | lowtek | jgoo: There's not that many choices for wireless phones. The best wireless solution I've found is the Aastra 480i CT (wired ethernet phone with radio handset, not wifi) |
| 15:40.43 | lowtek | fun330: no, same context |
| 15:41.01 | jgoo | lowtek, that isn't too bad.... if the handset has all functions... I could call the base a 'charging station' and that would then also give me no need for power sockets there |
| 15:41.46 | jgoo | lowtek, you have an idea for all the ethernet/wifi bridges? I've tries the Linksys SP400... I think that is it... wasn't too impressed with the cd based configuration... damn thing give me ssh to .conf files any day |
| 15:42.07 | jgoo | missing comma - but bonus points for a nice text based or remote config option |
| 15:42.15 | *** join/#asterisk iomari (n=iomari@41.222.209.142) |
| 15:42.47 | jgoo | Why don't companies think of these things? I swear, I am waiting for Apple to make a voip server, whatever the cost I will buy it because I am certain they would have thought of this, and 10 other things I didn't think about |
| 15:42.51 | lowtek | jgoo: Well, my philosophy is to just run fiber for mission critical stuff like phones if the cat5 distance is too great. Leave wireless for laptop users, but ymmv |
| 15:43.01 | jgoo | is afk for 1 hour |
| 15:43.08 | jgoo | ok, but cat is a serious PITA |
| 15:43.19 | lowtek | jgoo: Maybe, but it's reliable |
| 15:43.25 | fun330 | lowtek: can you paste bin an example for me i am not following |
| 15:43.45 | Zeeek | ANyone here on Twitter we have a directory list of Voip telephony people building here: http://tr.im/voipview ADD YOUR ID here: http://tr.im/voipform |
| 15:44.18 | iomari | will 2 ip phones and asterisk software be enough to commuicate beyween 2 hosts on a lan? |
| 15:44.36 | lowtek | fun330: my examples wouldn't help you much, it's really just as easy as build followme.conf, call it with _3XX,1,FollowMe(args), then _3XX,2,VoiceMail(args) |
| 15:44.50 | lowtek | fun330: using your 3xx extensions |
| 15:44.51 | FinboySlick | [TK]D-Fender: I found a ladder, it's not the lines ;) |
| 15:45.25 | jaytee | I found a chair, it's not the shoes :-) |
| 15:45.55 | fun330 | so if there is no follow me for that ext it will only dial the local number |
| 15:46.02 | FinboySlick | jaytee: Heh... Actually, I have a bit more detail on the issue now. Audio fades out, it starts low and fades into nothingness. |
| 15:46.21 | jaytee | on both channels? |
| 15:46.32 | FinboySlick | jaytee: Just two. |
| 15:46.38 | jaytee | out of how many? |
| 15:46.56 | FinboySlick | jaytee: Heh, I was about to specify, just channel two. |
| 15:47.17 | FinboySlick | jaytee: Channel 3 and 4 aren't used, I'll see if they have the same problem. |
| 15:47.42 | jaytee | how many FXO channels do you have on that card? try moving the line to another channel. |
| 15:47.53 | lowtek | fun330: no, followme() will simply fail and proceed to the next line in your dialplan |
| 15:48.22 | jaytee | on earlier Digium cards the FXO modules were single line modules but now they make modules that are 4 FXO or 4 FXS. Not sure what your card takes. |
| 15:48.29 | FinboySlick | jaytee: I have 4 channels, and yeah, that's what I said. If this is a bad fxo channel though, I'm begining to think that my box eats them, this FXO was just RMAed. |
| 15:48.50 | FinboySlick | Mine has the 4 FXO module. |
| 15:48.57 | jaytee | the quad mod? |
| 15:49.06 | FinboySlick | Yes, quad module. |
| 15:50.00 | jaytee | I'd still try moving the line to another port on that module |
| 15:50.36 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 15:51.04 | FinboySlick | jaytee: I will in a bit, yeah. But that makes me wonder if the line kills my FXO modules. As I said, this one is about two weeks old. |
| 15:51.22 | *** join/#asterisk Trionnis (n=rboggs@s233-51-251.nap.wideopenwest.com) |
| 15:52.55 | *** join/#asterisk mog (n=mog@nat/digium/x-eedc0101edb4ec75) |
| 15:52.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:56.00 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f33beab5eb424557) |
| 15:56.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:03.43 | *** join/#asterisk Sargun (n=Sargun@66.151.148.225) |
| 16:09.37 | *** join/#asterisk seaq (n=seaq@correo.seaq.com.co) |
| 16:10.50 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 16:12.32 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:12.59 | FinboySlick | jaytee: Switched from port 2 to port 3, everything is back to normal. |
| 16:13.15 | FinboySlick | jaytee: Now I neet to figure out what kills my FXO modules. |
| 16:14.57 | jaytee | could be surges on the line, do the come into protected entry terminal blocks at the demarc? |
| 16:15.58 | FinboySlick | jaytee: Well, this is not a very big shop... We put them through your standard type of powerbar surge protectors. I guess it's not enough. |
| 16:20.00 | jaytee | FinboySlick, not the AC power source, I'm talking about the phone lines themselves |
| 16:21.06 | FinboySlick | jaytee: Yes, that's what I meant too. Power bars with RJ-11 surge protected outlets. |
| 16:22.00 | jaytee | ah, ok. |
| 16:22.05 | *** join/#asterisk mog (n=mog@nat/digium/x-4e15fe8d01b2f36e) |
| 16:22.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:22.29 | jaytee | FinboySlick, you said you'd already RMA'd one module prior to this? |
| 16:22.57 | FinboySlick | jaytee: Just called our carrier, they suggested that we might be way too close to the PSTN (I don't think we have more than a few hundread feet of wire to the PSTN) and they've had similar problems at other PSTN. |
| 16:23.25 | FinboySlick | They'll come put some resistors on the lines. |
| 16:24.14 | jaytee | cool, I was going to ask if it was the same port on the last module you had issues with that you had to RMA. |
| 16:24.23 | FinboySlick | We're in nowhereland... They tend to crank the output to compensate for very long wire runs. |
| 16:24.39 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:24.44 | FinboySlick | jaytee: They eventually all died on the previous module. |
| 16:25.10 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:25.22 | jaytee | yup, definitely sounds like the case. hope adding LBO resistance helps |
| 16:25.37 | FinboySlick | Heh, we'll send them the bill if it doesn't. |
| 16:26.14 | *** join/#asterisk ttyS1 (n=julian@adsl-074-246-089-066.sip.bct.bellsouth.net) |
| 16:27.11 | ttyS1 | how can I allow caller ID passthrough ? |
| 16:28.22 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:28.29 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 16:28.29 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 16:28.42 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 16:28.44 | [TK]D-Fender | ttyS1: From what to what? |
| 16:28.44 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:29.52 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:30.11 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 16:32.49 | *** join/#asterisk ziro_axis (n=ziro_axi@41.208.79.249) |
| 16:32.56 | ziro_axis | hello |
| 16:33.02 | Zeeek | This is where I say goodbye and solicit you to /JOIN #voip-users-conference and call in to participate on 463#22622#1@proxy.ideasip.com if you'd like to talk |
| 16:33.28 | ziro_axis | here i'm again with the same damn problem of attaching a USB to *now |
| 16:33.37 | Zeeek | Sorry the SIP URI is: 7463#22622#1@proxy.ideasip.com if you'd like to talk |
| 16:33.40 | ziro_axis | so some body can help |
| 16:35.16 | jgoo | lowtek, really, can we rule out any reliability in wifi? surely it must be *sorta* reliable... it can't be a complete miss |
| 16:35.22 | carrar | ziro_axis, try #asterisknow |
| 16:35.31 | jgoo | I will wire each building perhaps, but, darn the time / costs.... |
| 16:38.04 | alrs | jgoo: still going on about wifi? Don't do it. |
| 16:38.20 | *** join/#asterisk metfan2007 (n=jc@201.103.19.81) |
| 16:38.39 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:38.50 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:39.24 | metfan2007 | Hi all!!! I'm trying to insert 2 values in the same mysql statement using func_odbc, but I'm having problems while trying to pass two arguments in the same fun_odbc call |
| 16:39.30 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:39.44 | *** part/#asterisk ziro_axis (n=ziro_axi@41.208.79.249) |
| 16:40.14 | metfan2007 | I just tried something like Set(ODBC_INSERTAEVENTO()=1\,1) but it takes "1,1" as one value |
| 16:41.59 | carrar | How about Set(ODBC_INSERTAEVENTO()="1,1") |
| 16:42.46 | jgoo | alrs, You know any solution for networking over copper? |
| 16:43.04 | metfan2007 | carrar: same, as one value :( |
| 16:43.06 | [TK]D-Fender | jgoo: Yes, we call it "ethernet" |
| 16:43.09 | alrs | jgoo: If it is CAT3 you can run 10BT over it. |
| 16:43.19 | jgoo | I just don't want to wire up 8 whole bloody houses... seriously... and alrs, I've read a lot about people doing voip over Wifi... this isn't enterprise stuff... |
| 16:43.29 | carrar | 10base2 |
| 16:43.46 | alrs | 10bt is twister pair, right? |
| 16:43.49 | ttyS1 | [TK]D-Fender: from a carrier sending in to the asterisk box and out to another carrier |
| 16:43.49 | alrs | twisted |
| 16:43.50 | jgoo | [TK]D-Fender, I mean phone line, existing phone lines in a building (is phone line cat3?) |
| 16:43.58 | eppigy | 210:43 < [TK]D-Fender> jgoo: Yes, we call it "ethernet" |
| 16:44.00 | eppigy | lollin |
| 16:44.00 | carrar | no |
| 16:44.05 | [TK]D-Fender | ttyS1: Depends ont he carrier if they let you set it or not |
| 16:44.10 | jgoo | eppigy, ignorant lollin' |
| 16:44.40 | eppigy | calm down guy |
| 16:44.40 | mort_gib | jgoo: Can't you run CAT5 betwen houses?? |
| 16:44.56 | jgoo | because he wasn't exactly stating something obvious... unless ethernet over phone cable is something obvious (And that was what I was asking, so 'yes we call it ethernet' isn't a good answer...) |
| 16:45.06 | eppigy | it is if you know standards |
| 16:45.16 | eppigy | 10:44 < jgoo> eppigy, ignorant lollin' |
| 16:45.17 | eppigy | lollin |
| 16:45.35 | jgoo | mort_gib, I might do that... others suggest wifi between buildings... I mean configuring PBX's is all well and good, but give me a jackhammer, a road and some cable, and I will go wardriving to find open wifi |
| 16:45.46 | jgoo | eppigy, trollfail lollin' |
| 16:45.47 | [TK]D-Fender | jgoo: Obvious? You just said "copper". An ELEMENT on the periodic table |
| 16:45.58 | eppigy | i mean seriously |
| 16:46.06 | eppigy | talking about ignorance |
| 16:46.09 | rob0 | Copper, you'll never take me alive!! |
| 16:46.11 | carrar | metfan2007, you can pass more then 1 value |
| 16:46.11 | eppigy | IRONY |
| 16:46.18 | eppigy | IRON-Y |
| 16:46.21 | eppigy | humor |
| 16:46.25 | [TK]D-Fender | rob0: How to speak "hick" |
| 16:46.26 | mort_gib | jgoo: Have a look at the PowerStation 2-5 units |
| 16:46.31 | rob0 | oh damn, more metallurgy |
| 16:46.31 | carrar | metfan2007, ${VAL1},${VAL2} |
| 16:46.40 | jgoo | [TK]D-Fender, yes, sadly the english language allows for synonyms to real nouns, such as referring to phone lines as copper. It is a tragedy and one I hope we can resolve with, if not violence, some form of political action. |
| 16:46.44 | jgoo | thanks mort_gib |
| 16:47.00 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:47.04 | mort_gib | np |
| 16:47.10 | jgoo | I am voting for violence though, it is cheaper in the long run |
| 16:47.17 | rue_mohr | ok, day, which? on building this phone system |
| 16:47.18 | ttyS1 | [TK]D-Fender: yes the carrier is sendin a valid caller id and the outgoing carrier accepts any valid number. however asterisk is removing the original incoming caller id. I just want to forward the original incoming caller id out to the other carrier |
| 16:47.45 | rue_mohr | so who is really familiar with the polycom 601 sets? |
| 16:47.48 | jgoo | alrs, I use a linksys wifi handset, admittedly it is a piece of crap, but it has never had any connection problems... |
| 16:47.50 | rob0 | I would recommend using pigeons. |
| 16:47.59 | rob0 | RFC 1149 |
| 16:48.13 | jgoo | rfc1149 yes, any compliant routers though? |
| 16:48.16 | rue_mohr | the packet loss aparently wasn't that bad |
| 16:48.24 | ttyS1 | [TK]D-Fender: btw I'm using sip |
| 16:48.41 | eppigy | TRABAJO |
| 16:48.46 | rue_mohr | terrible latency though |
| 16:48.47 | jgoo | I heard they were expensive and the built in tracking was messy |
| 16:49.10 | rue_mohr | but hey, you cant knock that collision avoidance! |
| 16:49.10 | jgoo | Plus don't forget PDOS attacks |
| 16:49.39 | jgoo | Pellican Denial of Service attacks.... www.youtube.com/watch?v=PO5ifLzLYiU |
| 16:50.15 | jgoo | rue_mohr, but, based on the hollywood films, I can see problems during electrical storms. |
| 16:50.38 | rue_mohr | that can be an issue with 802.1 anyhow |
| 16:50.46 | rob0 | Not safe to talk over copper in a storm anyway. |
| 16:51.00 | rob0 | STFU and hide under the bed! |
| 16:51.02 | carrar | use fiber |
| 16:51.22 | rue_mohr | 1149 is mechanically isolated though |
| 16:51.27 | jgoo | so, it is 2009, and we have no real way of installing a phone system without bending over and hammering pieces of metal into a wall to secure cables to a building? |
| 16:51.46 | rue_mohr | eherm I do that for a living? |
| 16:52.02 | rob0 | Oh I bet it could be done over wireless. Just not many have tried it. |
| 16:52.06 | mort_gib | jgoo: sure you do, just don't use WiFi phones |
| 16:52.07 | rue_mohr | besides, copper is secure and reliable |
| 16:52.24 | rob0 | directional antennae from router to router |
| 16:52.30 | rue_mohr | sure sure |
| 16:52.32 | jgoo | mort_gib, eh? explain what you mean... I don't intend using wifi phones (but ethernet/wifi bridges) |
| 16:52.42 | mort_gib | jgoo: if you must go wireless use dect phones |
| 16:53.01 | rue_mohr | ****is it possable to have a polycom 601 connect to asterisk when you hit the line key*** |
| 16:53.11 | jgoo | rue_mohr, the fail with that is, I've read lots since 2005 talking about it, but since the shelf life of electronics is 4-6 months, it is useless... I can't find the same routers I bought 3 months ago |
| 16:53.12 | rue_mohr | I'm gonna start messing with dialplans now |
| 16:53.39 | jgoo | mort, I would like to use non-wifi phones, and network with wireless.... if that makes sense... seperate the wifi aspect out of the phone system |
| 16:53.51 | jgoo | wifi phones == newer == less reliable |
| 16:54.07 | jgoo | I'd like a 4th / 5th gen successful + cheap SIP phone model |
| 16:54.25 | rue_mohr | cmon, polycom 601... anyone? |
| 16:54.26 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 16:54.36 | mort_gib | jgoo: ok, no problem, have a look at the powerStation2-5 tehy are great outdoor reliable WiFi devices |
| 16:54.42 | mort_gib | Use that to link houses |
| 16:54.46 | *** join/#asterisk willianmazzardo (n=willianm@187.4.15.116) |
| 16:54.47 | rob0 | yup |
| 16:54.51 | willianmazzardo | hi |
| 16:54.59 | mort_gib | Then use whatever phones you like, but NOT WiFi phones! |
| 16:55.01 | alrs | mort_gib: things like powerstation aren't so great for VOIP, as they are half-duplex |
| 16:55.03 | willianmazzardo | hpec echo cancelator, work in E1 system? |
| 16:55.29 | alrs | mort_gib: Perhaps Microtik n-streme running SR9 |
| 16:55.36 | mort_gib | alrs: Ok, I have some 10 users in one end of a link and the asterisk server in the other.... |
| 16:55.49 | alrs | mort_gib: over 2.4? |
| 16:56.06 | [TK]D-Fender | rob0: "Omlette" : You dun' me wrong... but omlette you off easy this time! |
| 16:56.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:56.10 | rue_mohr | [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT < thats std north american, right? if I delte it all togethor will it do an immediate connect to the asterisk machine? |
| 16:56.16 | *** part/#asterisk psy0nid3 (n=b0red@bookit-dev.com) |
| 16:56.17 | mort_gib | alrs: over 5Ghz |
| 16:56.52 | jgoo | alrs, thanks for the roundabout recommendation.... |
| 16:56.59 | [TK]D-Fender | rue_mohr: You can have it DIAL immediately. Never just use the word "connect" like it has some magical meaning |
| 16:57.11 | [TK]D-Fender | rue_mohr: And FFS go read the admin guide |
| 16:57.11 | rue_mohr | ok :) |
| 16:57.12 | alrs | jgoo: I recommended you look at Microtik 10 hours ago |
| 16:57.19 | rue_mohr | I am... |
| 16:57.22 | rue_mohr | its thick |
| 16:57.32 | [TK]D-Fender | rue_mohr: Strangely appropriate |
| 16:57.35 | carrar | heh |
| 16:57.47 | rue_mohr | not as thick as the aastra |
| 16:57.56 | jgoo | alrs, yeah? well, I was... *thinks* I dunno where I was 10 hours ago, but the coffee I just taken has increased my cognitive powers considerably, so now I am looking :-) |
| 16:57.58 | rue_mohr | 340 pages vs 1200 |
| 16:58.20 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:59.17 | rue_mohr | I cant find anything like, immediate send |
| 16:59.35 | carrar | when do you want it to dial? |
| 16:59.43 | carrar | as soon as you pick up the receiver? |
| 16:59.46 | rue_mohr | when I hit ht line key |
| 16:59.51 | carrar | or after 1 pressed digit |
| 16:59.55 | alrs | 02:52 < alrs> you could go wireless building to building |
| 16:59.55 | alrs | 02:53 < alrs> using mikrotik, perhaps |
| 16:59.59 | rue_mohr | before 1 digit |
| 17:00.09 | willianmazzardo | please, i have this problem. I`ve been buyed 2 licences of HPEC to use in 2 clients with Asterisk and E1 Telephony link. The Echo canceller will work in that system? |
| 17:00.10 | alrs | sorry, six hours ago |
| 17:00.20 | willianmazzardo | I need to buy 30 licences? one per channel? |
| 17:00.54 | rue_mohr | willianmazzardo, I dont know |
| 17:01.09 | willianmazzardo | rue_mohr, thanks for you atention :) |
| 17:01.17 | rue_mohr | :) |
| 17:03.34 | rue_mohr | I'm not aware of an hp echo can that works with asterisk, where did you get that idea from? |
| 17:03.43 | jgoo | alrs, what timezone is that 02:52? |
| 17:03.51 | jgoo | aaah ok, that was this morning, I was still asleep |
| 17:04.12 | willianmazzardo | rue_mohr, i have downloaded from digium, and installed ... |
| 17:04.28 | willianmazzardo | my only one doubt, is about the number of licences do buy |
| 17:04.31 | jgoo | You even spelled it correctly then :-) you must be tired now, thanks all the same. So, One of those... and what about ethernet / wifi bridges? |
| 17:04.46 | rob0 | [TK]D-Fender: "Mayonnaise" : Mayonnaise gotta fix dat pot-hole in de road, I durn near lost m' pikup truck in it! |
| 17:04.51 | willianmazzardo | because the echo canceller worked out with E1, but i need to confirm with you if that is OK |
| 17:04.52 | willianmazzardo | :) |
| 17:05.24 | rue_mohr | I presume you just need one liscence per stream (up to your 25 or whatever channels) |
| 17:05.28 | [TK]D-Fender | willianmazzardo: Yes, you pay PER CHANNEL |
| 17:05.44 | willianmazzardo | [TK]D-Fender, damn!! |
| 17:05.48 | rue_mohr | I'm wrong of couse |
| 17:06.02 | rue_mohr | willianmazzardo, why didn't you go with a hardware echo can? |
| 17:06.12 | willianmazzardo | so more expensive |
| 17:06.20 | rue_mohr | ebay? |
| 17:06.40 | willianmazzardo | i cant buy from US !! im from Brazil :) |
| 17:06.51 | rue_mohr | course I'm a hardware guru so thats easy for me to say |
| 17:06.58 | willianmazzardo | yeah ... |
| 17:07.16 | rue_mohr | do the asterisk software echo cans not work for E1? |
| 17:07.23 | [TK]D-Fender | willianmazzardo: then buy from CANADA. Everybody loves us! |
| 17:07.28 | willianmazzardo | eaiuhaeiuh |
| 17:07.35 | jgoo | alrs, are those things over 30k in price?!?!? |
| 17:07.57 | rue_mohr | I bet you could get a rack of echo cans for like $250us |
| 17:08.03 | rue_mohr | (from canada and all) |
| 17:08.06 | willianmazzardo | rue_mohr, some channels still get the eco in call, some others not !! |
| 17:08.07 | alrs | jgoo: no, they are a couple of hundred dollars |
| 17:08.14 | rue_mohr | oh |
| 17:08.16 | willianmazzardo | i have tested OSLEC, MG2 and not so good |
| 17:08.17 | jgoo | alrs, uuh... ok... I |
| 17:08.37 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:10.15 | rue_mohr | arg, the polycom wont send till I dial something, so I can dial 1, delete it, and hit send |
| 17:10.20 | *** join/#asterisk nexu (n=nexu@s5590800f.adsl.wanadoo.nl) |
| 17:10.38 | rue_mohr | what if I make a dialplan [0-9] |
| 17:10.41 | carrar | rue_mohr, make it a speed dial |
| 17:11.02 | rue_mohr | i CANT make it a speed dial cause its a line key. its a polycom 601 |
| 17:11.18 | rue_mohr | and I need it to be a line key to that it can take INCOMMING calls |
| 17:11.24 | carrar | so re-assign it as a speed dial |
| 17:11.41 | rue_mohr | then how do I make that key take incomming calls? |
| 17:11.52 | carrar | use a different line key |
| 17:12.02 | carrar | You have 6 |
| 17:12.02 | rue_mohr | I only have 6, I need 5 lines |
| 17:12.05 | willianmazzardo | rue_mohr, [TK]D-Fender, thanks for all ... good bye |
| 17:12.25 | rue_mohr | and at this rate I'm gonna need an 'intercom' |
| 17:12.36 | rue_mohr | which I dont want to talk about... |
| 17:12.42 | *** part/#asterisk willianmazzardo (n=willianm@187.4.15.116) |
| 17:12.46 | carrar | assign a extension number as a intercom, not a button |
| 17:13.00 | carrar | get a sidecar |
| 17:13.14 | rue_mohr | getting sidecars for these is NOT an option |
| 17:14.01 | carrar | Is this a "door phone"? |
| 17:14.05 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:14.11 | stupidnic | Is this the best way to handle a rollover situation http://pastebin.com/d27557d4b |
| 17:14.29 | rue_mohr | right now, all I need is for the polycom to send when I hit the line key, as an analog phone would with immediate=1 |
| 17:14.46 | *** join/#asterisk km2 (n=x@32.178.17.6) |
| 17:14.55 | rue_mohr | no, its an office system with a really really messed up arrangement of phone lines, people, and businesses |
| 17:15.47 | rue_mohr | ok, different approach, how do I get the web interface on the 601 woring |
| 17:16.15 | carrar | it's on by default |
| 17:16.18 | rue_mohr | it is working now, good |
| 17:16.26 | carrar | rtfm |
| 17:16.32 | rue_mohr | I am |
| 17:16.55 | *** part/#asterisk virtualme123 (n=chatzill@fentech.gotadsl.co.uk) |
| 17:17.03 | carrar | best way to configure those phones is via the ftp config |
| 17:18.01 | carrar | look at autoOffHook call.autoOffHook |
| 17:18.03 | rue_mohr | which is GREAT _IF_ you have a proper list of paramiters for the phone, and there isn't one, there are a bunch of half descent blog configs you can download from polycom |
| 17:18.13 | rue_mohr | :) thankyou |
| 17:20.43 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:25.03 | fun330 | has anyone had the following problem? |
| 17:25.47 | rob0 | I did. |
| 17:25.48 | fun330 | Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 17:26.20 | rob0 | My problem was that I was being followed. |
| 17:26.24 | fun330 | when trying to dial an extesion |
| 17:26.46 | carrar | fun330, sounds like that extension is a SIP phone that is not registered |
| 17:27.41 | fun330 | it is not |
| 17:27.47 | fun330 | so that is normal then |
| 17:27.53 | carrar | no |
| 17:28.36 | fun330 | so if the phone was registered would it work? |
| 17:28.53 | carrar | in theory |
| 17:29.01 | fun330 | this is my first asterisk setup just going thoguhg so road bumps |
| 17:29.14 | *** join/#asterisk andresmujica (n=andresmu@190.24.94.102) |
| 17:29.27 | carrar | Could be you are daling a none existant sip device also |
| 17:29.33 | carrar | typo |
| 17:29.46 | fun330 | it shows up in sip show peers |
| 17:29.48 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:30.04 | fun330 | just not registered i didn't program phoens yet just testing the dial plan |
| 17:30.10 | carrar | doesn't mean you typed it right in the dialplan |
| 17:31.07 | *** join/#asterisk h[a]kr (n=hakr@pdpc/supporter/active/hakr) |
| 17:32.40 | fun330 | okayu i am going to reg phone |
| 17:34.03 | *** join/#asterisk h[a]kr (n=hakr@pdpc/supporter/active/hakr) |
| 17:35.10 | *** join/#asterisk DarkRift (n=dark@65.92.170.122) |
| 17:36.55 | *** join/#asterisk `paul (n=admin@122.55.36.3) |
| 17:37.14 | `paul | how would you know if the cdr_mysql module is already installed? |
| 17:37.59 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 17:38.00 | angryuser | `paul: search it's module in asterisk modules |
| 17:39.12 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 17:39.43 | rue_mohr | module shoe like cdr ? |
| 17:39.47 | rue_mohr | show even |
| 17:40.41 | *** join/#asterisk af_ (n=getsmart@88-149-240-27.dynamic.ngi.it) |
| 17:42.12 | *** join/#asterisk andresmujica (n=andresmu@190.24.94.102) |
| 17:44.25 | rue_mohr | its webpage isn't working again |
| 17:44.59 | *** join/#asterisk hakr (n=hakr@pdpc/supporter/active/hakr) |
| 17:45.04 | rue_mohr | and now it is, grrr |
| 17:45.30 | carrar | download the ftp config file from polycom |
| 17:45.33 | carrar | and use that |
| 17:46.09 | rue_mohr | ok, should I worry itdosn't have any <ip601> in it, just 600 and 500? |
| 17:46.46 | rue_mohr | it has entries for all sorts of phones kinda a mess really |
| 17:47.28 | carrar | again you have not read the manual |
| 17:47.31 | rue_mohr | I dont remember where I got this one, but its missing all sorts of stuff, I cant get the phone to accept this autooffhook paramiter, tried feeding it to it 4 different ways |
| 17:47.51 | rue_mohr | I cant read every manual front to back, I'd be sitting here for a year |
| 17:48.04 | carrar | just read the 1 admin guide |
| 17:48.28 | carrar | 600 covers the 600 series phones |
| 17:49.18 | [TK]D-Fender | and what you want isn't MODEL SPECIFIC |
| 17:50.25 | *** join/#asterisk ocnarf (n=chatzill@122.2.246.148) |
| 17:51.20 | ocnarf | Hi, anyone here experienced when asterisk stops ringing phones when in queue? |
| 17:51.48 | ocnarf | then when i hit reload, it will start ringing the phones again (agents) |
| 17:52.57 | [TK]D-Fender | ocnarf: Next time SHOW us the status of the queue & its callers, agents, etc |
| 17:53.44 | rue_mohr | hmm the web login wont accept admin:456 Admin:456 administrator:456 or Administrator:456 |
| 17:54.07 | rue_mohr | and I cant find anything in the manual about a different username to log in with |
| 17:54.40 | rue_mohr | scrolls back to the top of the manual again |
| 17:55.02 | [TK]D-Fender | rue_mohr: So much for admin guide... |
| 17:56.13 | rue_mohr | ok ok, I'm gonna delete all the configs I'v spent a week doing and start with the ones from polycom |
| 17:57.01 | rue_mohr | and resetting the phone to defaults dosn't work, it dosnt reset everything |
| 17:57.45 | rue_mohr | I'm gonna be lucky if this phonesystem dosn't end my job |
| 17:57.54 | Katty | nomnomnomnomnoms on subway |
| 17:58.37 | jaytee | nomnomnomnoms on a Rallys Big Buford and some chili cheese fries |
| 17:58.42 | Katty | oooh |
| 17:58.46 | Katty | rallies fries are the BEST |
| 17:59.05 | jaytee | no, White Castle fries are the best but their burgers suck |
| 17:59.11 | Katty | i remember, back in the day, when they used to be bigger tho |
| 17:59.12 | Katty | and more nomable |
| 17:59.17 | beek | made due with a Kashi bar. |
| 17:59.29 | Katty | not really a fan of the white castle stuff--except for their hot chocolate. |
| 17:59.35 | Katty | they have very nice hot chocolate. |
| 17:59.36 | jaytee | I miss the original McDonalds fries fried in real beef tallow |
| 17:59.40 | jaytee | aka lard |
| 17:59.48 | Katty | they were awful for your health |
| 18:00.09 | Katty | not that we care. |
| 18:00.12 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 18:00.19 | jaytee | everything that actually tastes good is awful for your health, you just need to drink alot of red wine, that's what the french do. |
| 18:00.39 | rue_mohr | to get the config files I download the firmware package? |
| 18:00.41 | beek | Now *that* is a theory I subscribe to. |
| 18:01.06 | beek | really enjoys Merlot & Sheraz |
| 18:01.59 | rue_mohr | I cnt find it, do I want sample applications!?!?! where did they hide the config files!!!!!!!!!!! |
| 18:02.22 | rue_mohr | http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip601.html#download |
| 18:02.31 | rue_mohr | tries not to panic |
| 18:03.16 | jaytee | rue_mohr, the config files are bundled with the sip firmware package |
| 18:03.29 | rue_mohr | there isn't a firmware package there!? |
| 18:03.34 | jaytee | and you should read the whitepaper on configuration as well as the admin guide |
| 18:04.14 | rue_mohr | they dont have a link to a whitepaper on that page?! |
| 18:04.19 | rue_mohr | is this the wrong page!? |
| 18:04.32 | jaytee | rue_mohr, what model of Polycom phone? |
| 18:04.38 | rue_mohr | ip601 |
| 18:04.55 | [TK]D-Fender | rue_mohr: Downloads section on that very same page. Direct F-ING link. WAKE UP |
| 18:04.55 | ttyS1 | asterisk is sending the extension number instead of the origianl caller id. how can I change that si that it forwards the originating party caller id instead ? |
| 18:05.04 | jaytee | rue_mohr, http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip601.html |
| 18:05.13 | jaytee | that's for the download page for the firmware |
| 18:05.28 | [TK]D-Fender | rue_mohr: http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_3_1_1RevB_relnotes.pdf < |
| 18:05.35 | jaytee | http://www.polycom.com/common/documents/support/technical/products/voice/white_paper_configuration_file_management_on_soundpoint_ip_phones.pdf |
| 18:05.39 | [TK]D-Fender | rue_mohr: http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_1_release_sig.zip |
| 18:05.46 | jaytee | and that's the configuration guide whitepaper |
| 18:06.24 | [TK]D-Fender | ttyS1: So far you haven't proved taht your outbound provider ALLOWS you to set your callerid <- |
| 18:09.58 | Katty | distributes oreos (i almost typed auras) |
| 18:12.19 | Nugget | yum |
| 18:12.46 | eppigy | :D |
| 18:13.06 | *** join/#asterisk delphus (n=delphus@unaffiliated/delphus) |
| 18:14.04 | delphus | I have nat=yes in my peer and after sip reload, sip show peers tells me peer has nat N, how come ? |
| 18:15.08 | Katty | man, i've been on hold with this pharmacy for 20min |
| 18:15.16 | Katty | this is kinda stupid |
| 18:15.44 | eppigy | yeah |
| 18:15.45 | delphus | all my peers can't call anymore, there is no global nat=no set in sip.conf... might be some bug |
| 18:15.52 | eppigy | i can give you all the pharma advice you need |
| 18:16.01 | Katty | can you refill my prescription too? |
| 18:16.56 | eppigy | WHY YES |
| 18:17.02 | eppigy | what u need gurl |
| 18:17.08 | rue_mohr | they say to write all the configs from scratch!? |
| 18:17.13 | Katty | Yaz |
| 18:17.23 | Katty | i don't want children |
| 18:17.32 | eppigy | =( |
| 18:18.23 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:19.22 | rob0 | Katty, take one aspirin tablet, held between your knees. ;) |
| 18:19.54 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:20.00 | Katty | maybe when i turn 50 |
| 18:20.22 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 18:20.56 | Katty | OH NOES |
| 18:20.57 | eppigy | past 30 your chances of birthing an autistic or otherwise flawed child increase |
| 18:21.00 | Katty | my prescription as EXPIRED |
| 18:21.05 | Katty | sobs |
| 18:21.14 | eppigy | lets preserve our small minority of an intelligent populous |
| 18:21.19 | eppigy | att all costs please |
| 18:21.25 | Katty | calls doctor |
| 18:21.27 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
| 18:21.50 | Katty | but i don't want kids. |
| 18:22.43 | Corydon76-dig | Katty: would you birth kids for your gay friends? |
| 18:22.49 | Katty | no |
| 18:23.07 | Corydon76-dig | crosses Katty off his list |
| 18:23.20 | Katty | ha |
| 18:23.43 | Katty | adopt. there are plenty of children who need homes. |
| 18:23.56 | Corydon76-dig | is looking for that special female... who will birth one kid each for him and his partner |
| 18:24.10 | Corydon76-dig | May, eventually |
| 18:24.20 | eppigy | is looking for a special female that would not be afraid to birth his spawn |
| 18:24.21 | Katty | somehow i don't think ryan would be okay with that ^_- |
| 18:24.53 | Corydon76-dig | Ryan is your bf? |
| 18:25.01 | Katty | fiance |
| 18:25.04 | Corydon76-dig | Ah |
| 18:25.06 | Katty | but yes |
| 18:25.25 | Corydon76-dig | thought Katty was male-phobic |
| 18:25.34 | Katty | just creepy tall people |
| 18:25.43 | Corydon76-dig | hunches over |
| 18:25.47 | Katty | kthx |
| 18:26.13 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 18:26.27 | rob0 | has kids ... they make nice pets |
| 18:26.39 | eppigy | TRABAJO |
| 18:27.04 | Katty | http://flickr.com/photos/izaah/3021644675/in/set-72157608822234737/ <- Individual on Right |
| 18:27.54 | ttyS1 | [TK]D-Fender: I can modify the caller ID and set it to any 10 digit number. The outgoing carrier accepts it I've already test this. However the original caller ID is not passedthrough |
| 18:29.29 | Katty | ttyS1: noop what's going before you send the call |
| 18:30.15 | Katty | rob0: i have pets... they make nice kids. |
| 18:30.29 | Katty | rob0: in which case, i have 5 kids |
| 18:33.22 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:33.40 | ttyS1 | Katty: it sends the extension number instead |
| 18:35.19 | Katty | ttyS1: well that's why it's not showing up |
| 18:35.28 | Katty | ttyS1: a carrier cannot parse a 3 digit phone number |
| 18:36.40 | Katty | someone take these oreos away from me. |
| 18:40.42 | *** join/#asterisk zapa (n=hzavala@201.116.9.58) |
| 18:42.49 | ttyS1 | Katty: yes, so how can stop asterisk from sending the extension number and send the original caller id instead ? |
| 18:43.30 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 18:45.05 | verywiseman | [TK]D-Fender, which is better audiocodes or rhino? |
| 18:46.02 | [TK]D-Fender | verywiseman: Different products. |
| 18:46.18 | [TK]D-Fender | verywiseman: they don't make the exact same gear and I gave you my recommendation already |
| 18:47.46 | zapa | hi all, i have a trouble with echo, with a tdm800p 8 fxo ports, PSTN <- FX0 <- ASTERISK -> SIP EYEBEAM , I only recive echo in the EYEBEAM, i have and asterisk 1.4.22 with dadhdi fxsks=1 |
| 18:47.46 | zapa | echocanceller=mg2,1 |
| 18:48.13 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:48.15 | zapa | but not in the pstn is perfect |
| 18:48.18 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 18:48.20 | zapa | any clue? thanks |
| 18:49.34 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:50.56 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 18:54.41 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:58.01 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:59.50 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:00.08 | `paul | can i blocklist a list of numbers so that every time someone dials them asterisk wont allow it |
| 19:01.11 | stupidnic | just put them as extension patterns |
| 19:01.25 | stupidnic | that point at Congestion |
| 19:02.55 | stupidnic | exten => 8675309,1,Congestion |
| 19:03.06 | stupidnic | throws out an obscure reference |
| 19:04.01 | *** join/#asterisk rdk5 (n=jeff@75-27-14-205.lightspeed.iplsin.sbcglobal.net) |
| 19:04.16 | BBHoss | ha |
| 19:04.28 | beek | Is anyone using 1.6, AMI, and Originate successfully? |
| 19:04.28 | *** join/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 19:04.37 | *** join/#asterisk SparFux (n=raoul@e182031014.adsl.alicedsl.de) |
| 19:04.52 | `paul | is it possible to put em in a DB and use something like in_array.... if(in_array(EXTEN,array_of_blacklisted_nos)) then { hangup }? |
| 19:05.19 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 19:05.38 | stupidnic | `paul: with an AGI script sure |
| 19:05.52 | stupidnic | but that is beyond the scope of what I can assit you with |
| 19:06.29 | `paul | ok thanks |
| 19:06.32 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 19:06.41 | [TK]D-Fender | `paul: "core show function DB" , "core show application gotoif" |
| 19:06.52 | [TK]D-Fender | `paul: Go read the book. |
| 19:07.03 | FinboySlick | After peeking at my init.d script... should dahdi_cfg actually load the modules listed in /etc/dahdi/modules ? |
| 19:07.24 | FinboySlick | Or are they just listed there to look pretty? |
| 19:08.38 | *** join/#asterisk pdfhacker (n=dd@38.104.98.118) |
| 19:08.58 | rdk5 | <PROTECTED> |
| 19:10.00 | pdfhacker | I can't get Say (SayDigits/SayNumber/etc) to use the language-specific directory (yes, I have a digits/ subdirectory with all of the appopriate files), but Playback/Background use it without a problem. Are there any known issues? |
| 19:10.46 | pdfhacker | rdk5: are you trying to connect by name or just IP addresses? I recommend always using IP addresses directly when first testing |
| 19:10.57 | rdk5 | pdfhacker: i am using the IP address |
| 19:11.12 | rdk5 | and I am able to ping/nmap the asterisk machine from the machine the SIP softphone is on. |
| 19:11.58 | [TK]D-Fender | pdfhacker: pastebin yoru backup |
| 19:12.12 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 19:13.01 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 19:13.47 | *** join/#asterisk joako (n=joako@adsl-144-103-238.mia.bellsouth.net) |
| 19:15.13 | *** join/#asterisk pittstains (n=frank@mx1.distributivenetworks.com) |
| 19:15.27 | pdfhacker | rdk5: does your asterisk console display anything? Do you have sip debug on? |
| 19:18.19 | pdfhacker | [TK]D-Fender: What specifically? You mean the directory listing of [mylanguage]/digits/* ? |
| 19:18.47 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:20.49 | [TK]D-Fender | pdfhacker: You call attempt to use them normally, the one where you call them directly, folder dumps, etc |
| 19:21.17 | *** part/#asterisk SparFux (n=raoul@e182031014.adsl.alicedsl.de) |
| 19:23.24 | *** join/#asterisk h[a]kr (n=hakr@pdpc/supporter/active/hakr) |
| 19:23.42 | jaytee | file you here? |
| 19:23.48 | file | yes |
| 19:24.53 | jaytee | file, if I have one option in my IVR tree use the Dial statement will that be treated the same as a Hangup() and free up the lumenvox port or do I need to precede it with SpeechDestroy()? |
| 19:25.16 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 19:25.18 | file | if you don't call SpeechDestroy the port is kept open for the duration that the channel is alive |
| 19:25.30 | file | so call it after you are done |
| 19:26.23 | jaytee | ok, I get it, the port is kept open until the calling party hangs up or SpeechDestroy() is invoked. |
| 19:26.32 | file | right |
| 19:26.47 | jaytee | file, thanks! |
| 19:26.49 | *** join/#asterisk qdk (n=qdk@79.138.241.68.bredband.3.dk) |
| 19:27.04 | pittstains | i have a question about bridging calls... or maybe it's three way calls.... using AMI, I am generating a call to a user, and dumping that call into a context in my dialplan. when the user and answers the call, i want to be able to play them a message, then then initiate a three-way phone call |
| 19:27.57 | pittstains | i've heard mixed reviews about MeetMe, and i'm not sure if it's the right tool for the job. can anyone shed a little light on it? |
| 19:28.55 | pdfhacker | This is what I'm seeing with Say: http://pastebin.com/d1a7089f9 |
| 19:29.48 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 19:30.08 | rdk5 | pdfhacker: my console doesn't show anything -- it's like it's not even listening for a SIP connection or something. nmap doesn't show port 5060 as open, which i think it shold. |
| 19:30.51 | pdfhacker | rdk5: core set debug 9999 |
| 19:30.58 | pdfhacker | rdk5: sip set debug 9999 |
| 19:31.22 | pdfhacker | (assuming you used asterisk -r to connect to the instance) |
| 19:31.47 | rdk5 | pdfhacker: I did the core set debug, worked, sip set debug 9999 say no such command |
| 19:32.27 | pdfhacker | rdk5 sorry / sip set debug |
| 19:33.05 | pdfhacker | no number |
| 19:33.19 | rdk5 | pdfhacker: hmm, it still says no suck command 'sip set debug' |
| 19:33.23 | rdk5 | it's like my sip is missing |
| 19:33.38 | pdfhacker | rdk5: try core show channeltypes |
| 19:33.44 | pdfhacker | SIP should be listed |
| 19:34.02 | rdk5 | nope |
| 19:34.09 | rdk5 | pdfhacker: not listed |
| 19:35.11 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:35.16 | rdk5 | pdfhacker: and that is weird, I am using the out of the box asterisknow, all I did was add the extension. |
| 19:35.18 | pdfhacker | Did you do much to modify this from a default install? The SIP module isn't getting loaded. I'd make sure it's in /usr/lib/asterisk/modules |
| 19:35.47 | pdfhacker | make make sure it's set to get loaded in /etc/asterisk/modules.conf |
| 19:36.03 | pdfhacker | (or try a new default install, if you haven't gotten far anyway :) ) |
| 19:36.16 | rdk5 | This is my second default install |
| 19:36.18 | rdk5 | :( |
| 19:36.43 | nny_1 | I am trying to write a simple GotoIf statement that checks to see if the areacode on an inbound call is 912. I have a gotoif, is CALLERID the proper variable to use in this case? |
| 19:36.53 | rdk5 | i don't see a sip line in modules.conf though |
| 19:37.30 | nny_1 | actually CALLERID(num) |
| 19:42.01 | lowtek | nny_1: ${CALLERID(num):3} |
| 19:43.14 | rdk5 | pdfhacker: is there a way i can make sure that the sip module is loaded, force it somehow from the asterisk cli? |
| 19:44.08 | *** join/#asterisk FABN1977 (n=fneto@189-19-75-184.dsl.telesp.net.br) |
| 19:44.23 | FABN1977 | Hi all |
| 19:44.46 | rdk5 | pfhacker: I did module load chan_sip, and now it works. Why this is not enabled in the default install, I have no idea. |
| 19:44.50 | lowtek | Greetings, mighty baud warrior! |
| 19:45.38 | kannan | does asterisk support video through H323? |
| 19:46.27 | lowtek | kannan: http://www.voip-info.org/wiki/view/Asterisk+video |
| 19:46.45 | kannan | lowtek , thanks |
| 19:46.59 | FABN1977 | I'm having echo problem in asterisk, I have an dual E1 with echo cancel in hardware that is working fine, and a 4 port fxo analog that I use and that are having a lot of problems with echo |
| 19:47.12 | FABN1977 | does someone have this problem before? |
| 19:47.24 | lowtek | FABN1977: What echo canceller do you have for the 4-port fxo? |
| 19:47.52 | FABN1977 | it seems that 2 e1 turn on the echo canceller in hardware and disable the software echo cancel of the fxo card |
| 19:48.36 | lowtek | FABN1977: Someone else may jump in here with more experience with that hardware but I'm pretty sure you have to have an echo canceller for each path/tech/channel |
| 19:51.25 | FABN1977 | lowtek: where should I look to see the path/tech/channel?? Ins asterisk or in /proc? |
| 19:51.55 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:52.36 | lowtek | FABN1977: What I meant was that calls to your E1 card should use the E1 card's echo canceller and calls to your fxo card should use a software based echo canceller ... But this is a semi-educated guess, I don't use any of that hardware |
| 19:53.45 | FABN1977 | lowtek: I agree with you, but it seems that asterisk are ignoring ehco configuration in the analog card. That's why I think I have a problem! |
| 19:55.16 | lowtek | FABN1977: Again, just a guess, but try swapping the card slots. If the ec module is detecting hardware ec and exiting before it detects your fxo card. |
| 20:03.38 | *** join/#asterisk dieguito84 (n=diego@host248-192-dynamic.10-87-r.retail.telecomitalia.it) |
| 20:06.44 | jaytee | [TK]D-Fender, you busy? |
| 20:07.46 | beek | Hi jaytee -- waking the lion? |
| 20:07.51 | jaytee | yeah |
| 20:07.53 | jaytee | or trying |
| 20:08.10 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 20:08.34 | eppigy | TRABAJO |
| 20:08.51 | beek | If you want to get his attention just say how much you love grandstream products. |
| 20:09.31 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:09.41 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 20:09.56 | fun330 | how do you do dial by extesion in the auto attendent |
| 20:10.19 | tzafrir_laptop | FABN1977, what card is it? What version of Zaptel / DAHDI? |
| 20:10.41 | jaytee | beek, I keep getting this NOTICE message on one of the 3 piece of crap Grandstream phones I've got, all the other phones are Polycoms. :-) |
| 20:10.46 | jaytee | http://pastebin.ca/1304547 |
| 20:11.35 | [TK]D-Fender | jaytee: Yes? |
| 20:12.37 | jaytee | [TK]D-Fender, I keep getting the message in the above pastebin from a grandsuck phone. It's one of 3 and they all work and show registered but I get that coming across the console at least once an hour or so. |
| 20:12.50 | jaytee | I think it's a flaky phone but I can't be certain. |
| 20:13.07 | jaytee | everything is fine with it's sip entry in sip.conf |
| 20:13.19 | pittstains | i have a question about bridging calls... or maybe it's three way calls.... using AMI, I am generating a call to a user, and dumping that call into a context in my dialplan. when the user and answers the call, i want to be able to play them a message, then then initiate a three-way phone call. playing the message is the easy part :-). i've heard mixed reviews about MeetMe, and i'm not sure if it's the right tool for the a three-way cal |
| 20:13.41 | [TK]D-Fender | jaytee: Full SIP debug please... |
| 20:14.20 | [TK]D-Fender | pittstains: Who is the THIRD party? |
| 20:15.43 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 20:17.10 | pittstains | D-Fender: Maybe I'm wrong, but aren't there three parties? The user, Asterisk (or the dialplan or whatever), and the party I want to connect the user to? |
| 20:17.18 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 20:17.37 | jaytee | [TK]D-Fender, I've set sip debug ip to that phone, is that good enough? I've got alot of calls going on at the moment so a global would be like finding a needle in a haystack. It ain't an urgent thing anyways, just a curiosity factor. |
| 20:18.21 | pittstains | Are you saying I could just use Dial after I play back the message? |
| 20:20.32 | flewid | [TK]D-Fender. yo, rmember my issue yesterday with directory - turns out it wasn't a freepbx issue, it was the guys last name :) |
| 20:20.44 | flewid | asterisk, and freepbx don't take into account "la rocque" |
| 20:20.51 | flewid | i had to change the last name to not have a space, then she works |
| 20:24.02 | *** join/#asterisk nexu (n=nexu@86.85.169.45) |
| 20:26.47 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 20:28.21 | murdock_ut | Me wonders if Russell is still alive based on the fact he hasn't updated is blog since October 29th. :( |
| 20:28.39 | russellb | :-/ |
| 20:28.41 | russellb | I'm alive ... |
| 20:28.49 | russellb | I have been completely consumed by some internal projects |
| 20:28.58 | Juggie | and marriage |
| 20:29.09 | murdock_ut | He did say internal projects. :) |
| 20:29.16 | russellb | heh, not marriage .. |
| 20:29.40 | murdock_ut | I figured you were busy. |
| 20:30.34 | murdock_ut | russellb: Is one touch parking supposed to allow you to repark a call after a person picks it up? |
| 20:30.50 | Katty | hugs russellb |
| 20:31.02 | russellb | murdock_ut: good question ... in theory, yes |
| 20:31.38 | murdock_ut | russellb: I thought so, but can only get it to work is the call times out and rings back. |
| 20:32.14 | murdock_ut | russellb: 1.6.0.1 and 1.6.0.3 both have this problem. |
| 20:32.24 | russellb | bugs.digium.com i guess |
| 20:32.36 | murdock_ut | russellb: Will do. |
| 20:33.05 | murdock_ut | russellb: Thanks, wasn't sure if there was a features.conf setting that needed to be set to make it work. |
| 20:33.07 | [TK]D-Fender | flewid: Its still a stupid AGI.... I blame "poor design" |
| 20:33.22 | murdock_ut | Oh crap grumpy [TK]D-Fender is here. |
| 20:33.26 | Katty | i blame fender. |
| 20:33.40 | Katty | it's always his fault. |
| 20:33.44 | Katty | [TK]D-Fender: are you grumpy today dear? |
| 20:34.10 | [TK]D-Fender | murdock_ut: Oh crap I see you still haven't thought about the 3 lines of dialplan trickery it would take to make this function <- |
| 20:34.13 | murdock_ut | [TK]D-Fender: I tried to figure out what you were trying to tell me last night, but I could you another hint. I played with Park() it kinda worked. |
| 20:34.36 | murdock_ut | murdock_ut: Man my grammer sucks today. |
| 20:34.56 | Katty | mhmmm |
| 20:35.03 | Katty | k'then |
| 20:35.15 | Katty | testosterone spill in aisle 4 |
| 20:35.19 | Katty | wanders off to find mop |
| 20:35.23 | *** part/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 20:35.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:35.58 | FinboySlick | Somebody said testosterone? |
| 20:37.57 | *** join/#asterisk sack (n=sack@244.Red-79-148-188.staticIP.rima-tde.net) |
| 20:38.54 | *** join/#asterisk saftsack (n=oliver@g226196248.adsl.alicedsl.de) |
| 20:39.03 | murdock_ut | [TK]D-Fender: Am I correct in assuming I need to use the park() application? I'm all for doing thing the right way, but discovering what that is has proven difficult at times. |
| 20:39.27 | [TK]D-Fender | murdock_ut: *1* command lets you talk to someone and care about features.conf. That is what you need to do to pick up the call and be able to re-park accordingly |
| 20:39.57 | [TK]D-Fender | murdock_ut: And your problem isn't "park", its how to enable the ability once you RETRIEVE the call. |
| 20:40.06 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 20:42.12 | murdock_ut | [TK]D-Fender: The only option I know of enable that is the kK options in the dial application. |
| 20:42.19 | *** join/#asterisk kisu_ (n=kisu@2001:5c0:1100:9900:1d4:cb1a:114b:4dd0) |
| 20:44.36 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 20:44.57 | [TK]D-Fender | murdock_ut: And that is what you must use in picking up your parked call in order to be able to repark it. |
| 20:45.06 | flewid | [TK]D-Fender. that's what i'm saying though it has nothing to do with their agi, even using the internal app directory instead of app phonebook provided by freepbx |
| 20:45.09 | flewid | causes the same issue |
| 20:45.13 | flewid | it was just the space in the name |
| 20:46.06 | murdock_ut | [TK]D-Fender: Ok, so I don't use the ParkedCall app to pick up the call then? |
| 20:50.53 | *** join/#asterisk maddog01 (n=minotaur@mail.upperjamestoyota.ca) |
| 20:52.37 | FABN1977 | tzafrir_laptop: Sorry I have to leave my room for a moment... |
| 20:52.59 | FABN1977 | tzafrir_laptop: I'm using latest dahdi version drivers and asterisk 1.4.22 |
| 20:53.30 | tzafrir_laptop | what EC do you use? do you actually use one? |
| 20:53.41 | tzafrir_laptop | do you have echocanceller lines in system.conf? |
| 20:54.26 | tzafrir_laptop | (for the FXOs) |
| 20:55.01 | [TK]D-Fender | murdock_ut: Yes... + dial |
| 20:55.12 | FABN1977 | tzafrir_laptop: No configuration in system.conf |
| 20:55.18 | [TK]D-Fender | flewid: A smart AGI wouldn't care about the space. |
| 20:55.22 | FABN1977 | I'm compiled dahdi with oslec |
| 20:55.28 | [TK]D-Fender | flewid: like I said.... dumb AGI |
| 20:55.35 | FABN1977 | tzafrir_laptop: I'have compiled dahdi with oslec |
| 20:56.19 | tzafrir_laptop | FABN1977, you need to have explicit echocanceller line for each channel you want to use a software echo canceller |
| 20:56.50 | FABN1977 | tzafrir_laptop: I've read it now in the system.conf |
| 20:57.07 | tzafrir_laptop | http://docs.tzafrir.org.il/dahdi-tools/#_echo_cancellers |
| 20:57.17 | tzafrir_laptop | for OSLEC is it "oslec" |
| 20:57.24 | FABN1977 | tzafrir_laptop: I make this mistake because I rename the default file and create another empty file |
| 20:57.53 | FABN1977 | tzafrir_laptop: I will test it now! Thanks by now! |
| 20:58.50 | maddog01 | i have a question about Cepstral Text-to-Speech. they have voices for download on there website. can i use that download directly with asterisk or do i need to purchase the product before i can use it. |
| 20:59.20 | *** join/#asterisk sack (n=sack@101.Red-79-148-188.staticIP.rima-tde.net) |
| 20:59.35 | gambler1 | [TK]D-Fender: ping |
| 21:00.05 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-b72d9c42547f4c9b) |
| 21:00.13 | WHYS | is trying * intergration with CCM |
| 21:00.33 | WHYS | I can get calls from the call manager side, but call out through * yeald ChanIsAvail |
| 21:01.18 | *** join/#asterisk gviewbaron (n=gviewbar@67.96.159.72) |
| 21:02.54 | WHYS | - that is, I can recieve calls in *, but not call out. : ChanIsAvail("SIP/2200-b7600868", "SIP/callman02&SIP/callman01") |
| 21:04.17 | flewid | [TK]D-Fender. then blame asterisk as much as you blame freepbx :) |
| 21:04.32 | [TK]D-Fender | WHYS: ChanisAvail does not call out. |
| 21:04.48 | [TK]D-Fender | WHYS: Show actual SIP debug for a real call attempt |
| 21:05.09 | [TK]D-Fender | flewid: For Directory(), sure. For the other, that's 100% FreePBX |
| 21:05.38 | [TK]D-Fender | gambler1: Yes? |
| 21:05.46 | WHYS | Yeah, I was reading that wrong. Let me look again. |
| 21:05.57 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 21:06.14 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-868a18f72883537c) |
| 21:06.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:06.18 | jameswf | chrome in linux sweeeeeet |
| 21:07.32 | *** part/#asterisk pittstains (n=frank@mx1.distributivenetworks.com) |
| 21:07.37 | FABN1977 | tzafrir_laptop: I have the file compiled dahdi_echocan_oslec in my kernel tree but If I choose oslec dahdi says that it isn't available |
| 21:07.55 | FABN1977 | tzafrir_laptop: the oslec module is loaded |
| 21:08.00 | flewid | [TK]D-Fender. yes, and that's what i am saying, directory() does the same shit :) |
| 21:08.02 | gambler1 | [TK]D-Fender: remember the problem about devicestate variable today? It seems that problem was in 1.6.0.1, I have just upgraded to 1.6.0.3 and it works perfect... |
| 21:08.04 | murdock_ut | [TK]D-Fender: I thought the only way to pickup a call was with the parkedcall app how would I do that with the dial app? |
| 21:08.22 | gviewbaron | anyone have a cisco 7970 /c SIP? I have it working but when I make outbound calls with "RNA" instead of getting that persons voicemail, I get a fast busy on the phone and "Reorder" displayed |
| 21:08.32 | *** join/#asterisk fusss (n=chatzill@ip70-187-234-43.dc.dc.cox.net) |
| 21:08.35 | flewid | gviewbaron. i'm using a 7970, no issues like that though |
| 21:08.36 | [TK]D-Fender | murdock_ut: use BOTH. Think on it... |
| 21:08.37 | gambler1 | [TK]D-Fender: anyhow... thank you once again for your help and patience... |
| 21:09.18 | tzafrir_laptop | FABN1977, what is the exact error? |
| 21:10.30 | murdock_ut | [TK]D-Fender: Dial a context that contains the parkedcall app?? |
| 21:10.45 | [TK]D-Fender | murdock_ut: Almost there... go run with it a bit. |
| 21:11.34 | FABN1977 | look at dmesg outputdahdi_echocan_oslec: Unknown symbol oslec_create |
| 21:11.35 | FABN1977 | dahdi_echocan_oslec: Unknown symbol oslec_update |
| 21:11.35 | FABN1977 | dahdi_echocan_oslec: Unknown symbol oslec_free |
| 21:11.49 | FABN1977 | It's a module problem |
| 21:12.43 | loather-work | crap, after my spurious asterisk upgrade last night, queueing seems to be bj0rken. the queue seems to ignore the fact that an agent is already on the phone and sends them additional calls. |
| 21:12.45 | FABN1977 | tzafrir_laptop: I've download and compiled oslec from the source code at page, but I haven't changed the kernel version |
| 21:14.01 | tzafrir_laptop | what is the exact error you get that "oslec isn't available"? |
| 21:14.39 | FABN1977 | the oslec canceller isn't available because the module can't be loaded |
| 21:15.13 | *** join/#asterisk tacvbo (n=tacvbo@189.146.192.147) |
| 21:15.22 | gviewbaron | is "REORDER" a sip message sent from Asterisk during call progress? |
| 21:15.45 | *** join/#asterisk fusss (n=chatzill@ip70-187-234-43.dc.dc.cox.net) |
| 21:16.33 | fusss | (sorry if this is repeated, but i wasn't connected when i asked earlier) do I need any special hardware other than a network card if I'm using a Vitelity trunk? |
| 21:17.03 | gviewbaron | fusss: No, I use vitelity and works great |
| 21:17.22 | fusss | do you use it with a soft-phone? |
| 21:17.31 | gviewbaron | yes and hard phones |
| 21:17.54 | fusss | so basic asterisk installation and that's all i have to worry about? :-) |
| 21:18.04 | fusss | i know my way around unix, just not telephony |
| 21:18.21 | gviewbaron | don't forget a good internet connection...dsl may not be adequate. same with cable..just depends |
| 21:18.33 | fusss | we have a T1 |
| 21:19.11 | FABN1977 | tzafrir_laptop: I'm using CentOS 5.2 with kernel 2.6.18 do you know if exist some oslec patch for it? |
| 21:19.15 | gviewbaron | should be fine then. Just setup your bandwidth allocation so that some one browsing the internet won't kill it |
| 21:19.41 | fusss | ok, thanks :-) |
| 21:19.47 | tzafrir_laptop | FABN1977, could you please start by reporting the exact error? |
| 21:20.09 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 21:20.11 | tzafrir_laptop | hmm... what error do you see then in /var/log/messages ? |
| 21:21.54 | *** join/#asterisk SparFux (n=raoul@e182031014.adsl.alicedsl.de) |
| 21:22.19 | SparFux | Is there a software phone which can directly speak to ISDN cards? So that I can use it as an ISDN phone? |
| 21:23.20 | FABN1977 | DAHDI_ATTACH_ECHOCAN failed on channel 63: Invalid argument (22) |
| 21:23.46 | FABN1977 | FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 21:23.54 | loather-work | SparFux: most ISDN hardware is customer-premise side and doesn't support network-side signalling |
| 21:24.18 | SparFux | loather: Ok, I mean ISDN hardware with HFC-S chip. |
| 21:24.27 | FABN1977 | did you get the errors? |
| 21:24.57 | loather-work | SparFux: normally to connect ISDN BRI phones without a real switch you need something like aN ISDN BRI simulator. Normally they take a PRI as input and demux it to a bunch of BRIs. |
| 21:25.16 | SparFux | aha. |
| 21:25.40 | loather-work | hardware that can do network-side PRI is relatively cheap and not difficult to find |
| 21:25.59 | loather-work | any of your sangoma/digium/etc. equipment can do it. |
| 21:28.41 | loather-work | SparFux: i completely misread your question. you want something to use a softphone with your already-established BRI line. |
| 21:29.46 | loather-work | SparFux: if that's the case, then as long as your BRI board is supported by e.g. zaptel/dahdi/etc. then you should be able to use it with asterisk, then connect the softphone up to asterisk. just use it for translation between sip and isdn. |
| 21:29.57 | SparFux | loather: Yes. |
| 21:30.26 | SparFux | But then as long as I am on the same machine, the sip part is very much overhead. |
| 21:30.27 | loather-work | anyhow, bbl. |
| 21:32.14 | troy- | does anyone have a script whereby you dial a number from console and it drops that user a dialtone? |
| 21:33.25 | lowtek | troy: Disa() |
| 21:34.08 | SparFux | And sip <-> isdn is a loss of quality, too. |
| 21:36.52 | SparFux | Yes, Disa(), I use that. |
| 21:37.37 | stupidnic | is there anyway with Followme() to force it to not prompt the caller for their name? |
| 21:39.12 | ricko73 | stupidnic: it's not supposed to if you leave the 'a' option out (see 'show application followme') |
| 21:39.13 | *** join/#asterisk propellerhead (n=yogurt2u@host9.190-138-94.telecom.net.ar) |
| 21:39.33 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 21:39.49 | stupidnic | the a option function was clear in that |
| 21:39.58 | troy- | lowtek, but can i use that to push a dialtone to someone versus calling in and getting one? |
| 21:42.19 | lowtek | troy: Yea, just call their extension with either AGI or a call file and then just put them in a context where s,1,Disa(args) exist. |
| 21:42.49 | lowtek | troy: you trying to hand calls to sales people? |
| 21:43.24 | stupidnic | ricko73: just a note... if you leave off the | it does the defaults, which appears to include the 'a' option |
| 21:44.38 | stupidnic | http://www.engadget.com/2009/01/09/openpeak-intros-atom-powered-proframe-voip-phone/ <- Okay... I need one of these, screw the Cisco 7960 :) |
| 21:45.08 | troy- | ah, thanks |
| 21:46.00 | NovceGuru | that's hot |
| 21:46.06 | stupidnic | ain't it? |
| 21:46.20 | stupidnic | that is one sexy phone |
| 21:47.36 | WHYS | the APPLICATION cut is depreciated right? exten => s,2,Cut(AVAILCHAN=AVAILCHAN,,1) |
| 21:47.40 | NovceGuru | seems silly to a point though, "calendar access etc." Isn't that what your computer is for? |
| 21:47.44 | jjshoe | stupidnic i just cmae a little |
| 21:47.45 | thehar | anyone know any ports besides 80 (http provisioning) and 5060 that need to be opened for a linksys pap2t? |
| 21:47.54 | FABN1977 | tzafrir_laptop: thank you by now, I'm using the mg2 and it's solving my problem by the time! |
| 21:47.57 | lowtek | WHYS: Yea, but replaced with a function. |
| 21:47.58 | FABN1977 | Thank's all! |
| 21:48.19 | jjshoe | NovceGuru not when it's in my living room |
| 21:48.21 | WHYS | How do I write this now? I am following an age old wiki |
| 21:48.27 | NovceGuru | jjshoe: yes |
| 21:48.33 | lowtek | thehar: describe your configuration. |
| 21:49.29 | thehar | the pap2t has a static ip assigned to it, i need to acl in/out to it. using http for config and it's registering to a softswitch for line 1 |
| 21:49.32 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) |
| 21:49.33 | thehar | SIP |
| 21:49.34 | jjshoe | NovceGuru look at their pics for ideas... http://www.openpeak.com/OpenFrame.html |
| 21:49.42 | lowtek | thehar: network configuration |
| 21:49.48 | Lyma | hi! |
| 21:50.20 | NovceGuru | iPhone! |
| 21:50.35 | NovceGuru | tablet PC with a stand! |
| 21:51.02 | thehar | iad > switch > router on this vlan > internets cloud > wholesaler with sonus switch |
| 21:51.27 | NovceGuru | looks awesome though |
| 21:51.38 | lowtek | thehar: is your firewall "anything" allowed out? |
| 21:51.49 | thehar | no.. neither inbound |
| 21:51.55 | Lyma | I have this errors in my asterisk log... anyone knows what is it? |
| 21:51.55 | thehar | must open ports |
| 21:51.55 | Lyma | [Jan 9 18:35:31] WARNING[18192]: channel.c:2755 set_format: Unable to find a codec translation path from g729 to slin |
| 21:51.55 | Lyma | [Jan 9 18:35:31] WARNING[18192]: res_agi.c:2101 eagi_exec: Unable to set channel 'IAX2/glx-163654' to linear mode |
| 21:52.10 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:52.20 | lowtek | thehar: The source ports are going to vary due to NAT. Maybe let anything on that vlan out? |
| 21:52.29 | Lyma | (sorry for bad english) |
| 21:52.43 | lowtek | Lyma: Do you have g.729 licenses installed? |
| 21:52.57 | thehar | lowtek: i didn't want to any.. but i figured as much. |
| 21:53.36 | murdock_ut | [TK]D-Fender: Well this seems to work. Do you approve? http://pastebin.ca/1304629 |
| 21:53.53 | lowtek | thehar: You should be able to get one device out with SIP 5060-5061 and RTP 10000-20000. |
| 21:55.12 | lowtek | thehar: SIP just sets up and manages the session (Session Initiation Protocol). RTP is where the audio is. |
| 21:55.18 | thehar | yah |
| 21:55.54 | Lyma | lowtek: disabling the codec to test my billing. thanks for your help! |
| 21:56.58 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:58.45 | [TK]D-Fender | murdock_ut: Not bad, you got it :) |
| 21:59.02 | thehar | mmm cisco acls are FUN |
| 21:59.45 | lowtek | Yikes! You'll have to enable sip fixup |
| 22:00.11 | thehar | oh? |
| 22:00.20 | lowtek | Which model cisco? |
| 22:00.42 | thehar | 6509 |
| 22:01.17 | lowtek | So your controlling layer 3 at the switch? |
| 22:01.46 | lowtek | sip fixup is/was for PIX, the ASA's don't need it. Dunno about the switches. |
| 22:01.57 | thehar | i'll try this out and go from there |
| 22:02.14 | eppigy | CONTROLLING PACKETS AT THE SWITCH |
| 22:02.25 | thehar | lolz |
| 22:02.57 | *** join/#asterisk tacubo (n=tacvbo@189.146.185.216) |
| 22:03.01 | lowtek | I don't see that as a problem really, just don't know the implications. |
| 22:03.07 | murdock_ut | [TK]D-Fender: I just needed a little nudge, thanks. Now the question I have is, is that the way one touch parking is supposed to be handled, or is there a bug and this is the work around? |
| 22:06.01 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:51cd:9052:ec05:fd39) |
| 22:07.11 | gviewbaron | does anyone have a cisco 79xx phone working /c Asterisk & SIP? |
| 22:07.27 | lowtek | gviewbaron: Yes. Use SIP firmware 8.2 for best results where NAT is used. |
| 22:08.22 | lowtek | gviewbaron: If it's the java based phones, they won't do NAT with Asterisk. These generally end in "1" such as the 7941, 7961, etc. |
| 22:09.04 | gviewbaron | mine is a 7970, and I have sip working, but whenever I call a cell phone, or 800 number, etc that xfers to voicemail on farend, the phone does a reorder |
| 22:09.25 | gviewbaron | can't figure out what setting would change the reorder tone detection |
| 22:10.13 | lowtek | gviewbaron: Define reorder? |
| 22:10.24 | lowtek | gviewbaron: You mean just congestion? |
| 22:10.25 | gviewbaron | Fast busy. Phone actually displays "reorder" |
| 22:10.39 | lowtek | gviewbaron: Which firmware? |
| 22:10.46 | gviewbaron | what I hear is 5-6 rings, then when it would normally go to the cell phone voicemail, it says "reorder" |
| 22:10.48 | gviewbaron | checking... |
| 22:11.16 | lowtek | I did not know a fast busy was the same as a reorder, learn somethign new every day, neat |
| 22:11.28 | gviewbaron | SIP70.8-3-1S |
| 22:13.16 | lowtek | gviewbaron: one sec, looking up the firmware ... |
| 22:13.19 | gviewbaron | OS Load & App Load are 8.3.0 |
| 22:13.38 | lowtek | gviewbaron: Is NAT employed in your setup? |
| 22:13.43 | gviewbaron | actually 8.3.0.50. No nat |
| 22:13.57 | lowtek | gviewbaron: Ok, downgrade to SIP 8.2 and see if the condition still occurs. |
| 22:14.09 | lowtek | gviewbaron: or all the way up to latest 8.8(2) I think. |
| 22:14.24 | lowtek | gviewbaron: 8.2 is what most of us use. |
| 22:14.28 | gviewbaron | ok..guess I need to purchase the cisco maint |
| 22:14.34 | gviewbaron | will try that |
| 22:14.36 | gviewbaron | thanks |
| 22:14.37 | lowtek | gviewbaron: No, 8.2. is a free download. |
| 22:14.45 | gviewbaron | is it? Do you have a url? |
| 22:14.47 | lowtek | gviewbaron: The only one that's free afaik |
| 22:14.54 | lowtek | gviewbaron: one sec |
| 22:15.53 | lowtek | gviewbaron: looks like the link I have is broke, looking on cisco.com now |
| 22:16.08 | lowtek | Heck, I can email it to you, PM me your email address. |
| 22:17.41 | gviewbaron | man..wish I knew how to pm in mirc |
| 22:17.53 | stupidnic | /msg |
| 22:18.17 | gviewbaron | thx |
| 22:18.24 | stupidnic | do you just need the .bin? |
| 22:18.34 | gviewbaron | I think so |
| 22:19.11 | gviewbaron | I believe the bin expands out to give the other files |
| 22:21.18 | lowtek | gviewbaron: If you've never done a cisco firmware upgrade, you may not want to thank me just yet, LOL, at least you already have SIP on the phone. |
| 22:21.51 | stupidnic | yeah its a huge pain especially if you aren't well versed in tftp,xml, config editing |
| 22:22.19 | gviewbaron | ya, that was fun. I got it working with SKinny, but thought sip would be more compatible with the rest of my setup |
| 22:22.24 | stupidnic | I wish I knew how call manager handled it so well, but I have to think that software really is a mess behind the scenes |
| 22:23.08 | lowtek | gviewbaron: What I emailed you is the solaris tftp server already setup to do the upgrade, just replace one of the SIP000000000.cnf files with SIP<your mac address>.cnf, disable your location DHCP server, start up the tftpd.exe, and reboot your phone. |
| 22:23.40 | lowtek | Well, I got an NDR back saying the attachment was illegal. Sending again, just firmware this time. |
| 22:23.50 | gviewbaron | ok, thx |
| 22:24.03 | stupidnic | hmm |
| 22:24.08 | stupidnic | I am running 8.6 |
| 22:24.19 | stupidnic | I could have sworn I downgraded |
| 22:28.01 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:34.13 | *** join/#asterisk rainerj (n=rainer@wurzel.jochem.name) |
| 22:34.24 | *** part/#asterisk rainerj (n=rainer@wurzel.jochem.name) |
| 22:34.53 | loather-work | SparFux: if you use G.711 (ulaw in the us, alaw in europe/everywhere else) as your sip codec then it'll bridge it natively. g.711 is the protocol almost all isdn voice lines use |
| 22:41.47 | gviewbaron | Am I correct that the SEP<MAC>.xml is where you change the load file for the firmware? |
| 22:41.54 | gviewbaron | is there another place in adddition? |
| 22:43.33 | *** join/#asterisk iceyp (n=icepick@60.234.68.250) |
| 22:44.03 | iceyp | hey guys, anyone know what would cause voice not to work between extensions? Outbound calls are fine, it's just calls to other extensions there is no audio |
| 22:45.02 | stupidnic | can one side hear the other, or is there nothing at all? |
| 22:45.08 | stupidnic | in either direction |
| 22:45.17 | iceyp | nothing at all |
| 22:45.28 | iceyp | I've tested from 2 extensions behind the same DSL line |
| 22:45.36 | iceyp | then also from one dsl to another |
| 22:45.43 | iceyp | the PABX is colocated in a DC |
| 22:46.02 | iceyp | the ports I've opened are 5060+16384-53999 |
| 22:46.28 | iceyp | if I call via the PABX to a sip trunk it's fine |
| 22:46.36 | iceyp | so it's only internal calling doesnt work |
| 22:47.11 | iceyp | if either of the extensions dial an extension (10) which dials the IVR on the PBX that is fine |
| 22:47.14 | stupidnic | what is your canreinvite set to? |
| 22:47.20 | iceyp | 1 sec |
| 22:47.39 | iceyp | no |
| 22:47.51 | stupidnic | hmmm shoots my theory down |
| 22:48.49 | stupidnic | are you watching the console when the extension to extension calls are setup? |
| 22:49.04 | iceyp | yeh |
| 22:49.07 | carrar | gviewbaron: SEP<MAC>.cnf.xml |
| 22:50.12 | carrar | <loadInformation>SIP70.8-4-2S</loadInformation> |
| 22:50.34 | iceyp | The call sets up correctly from what I can tell |
| 22:50.39 | gviewbaron | carrar: thx |
| 22:51.35 | *** join/#asterisk ta^3 (n=tacvbo@189.146.193.18) |
| 22:51.48 | *** join/#asterisk nirz (i=c075ec1d@gateway/web/ajax/mibbit.com/x-5bb500b6128204f6) |
| 22:52.15 | nirz | hello, can i run a cli command whithin agi EXEC command ? |
| 22:52.32 | stupidnic | iceyp: its been a while since I have dealt with sip -> pbx -> sip |
| 22:52.44 | stupidnic | I mainly handle iax -> pbx -> iax now |
| 22:53.03 | iceyp | ok |
| 22:53.07 | stupidnic | iceyp: perhaps its a code issue? |
| 22:53.10 | stupidnic | codec |
| 22:53.28 | stupidnic | the last time I had an issue like this... it was related to canreinvite |
| 22:53.43 | stupidnic | where the two SIP clients were trying to hand the call off to each other directly |
| 22:53.48 | stupidnic | and that wouldn't work |
| 22:53.49 | iceyp | both ends are canreinvite=no |
| 22:54.08 | stupidnic | yeah but the call would never get setup if that were the case |
| 22:54.15 | stupidnic | so that isn't your problem I don't think |
| 22:56.51 | nirz | hello, can i run a cli command whithin agi EXEC command ? |
| 22:57.05 | stupidnic | iceyp: can you disable the firewall on the asterisk box at the DC and see if that might be causing problems? |
| 23:00.52 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:02.08 | iceyp | mmm, i just opened up my firewall and it works |
| 23:02.12 | iceyp | so I must be missing something |
| 23:02.52 | iceyp | I'm only allowing 5060+16384-53999 to the PBX |
| 23:03.03 | gviewbaron | the cisco phone will go to a screen that says 'upgrading' then flash error and go to bulls eye screen. I am running tcpdump, and it never attmepts to do a tftp xfer, and gets ip after upgrade attempt |
| 23:03.10 | gviewbaron | I did a factory reset |
| 23:03.15 | gviewbaron | screwed? |
| 23:05.08 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:05.55 | *** part/#asterisk nirz (i=c075ec1d@gateway/web/ajax/mibbit.com/x-5bb500b6128204f6) |
| 23:06.30 | *** join/#asterisk morglum (n=morglum@ip-62.81.126.206.dsl-cust.ca.inter.net) |
| 23:07.32 | morglum | Hi everyone. I just got a new (formerly locked) SPA2102 connected to my pbx. When I dial an extension starting with *, it only passes on the first 2 digits. For example, asterisk only receives *66 when I dial *66664 . Any idea? Thanks!! |
| 23:17.55 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:51cd:9052:ec05:fd39) |
| 23:27.41 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:33.40 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:44.02 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-868a18f72883537c) |
| 23:54.35 | beek | morglum: Look at the dialplan on that device (not Asterisk, the device.) |
| 23:59.50 | SlicerDicer | ok does anybody have any insight into why extensions following 6000, 6004, 6005, are able to dial extension@ip however... 6001 6002 6003 cannot dial.. I get a proxy authentication required? any ideas where something could be wrong? my sip.conf is identical for the extensions... |
| 23:59.52 | SlicerDicer | I am baffled here |