00:00.02 | ManxPower | And it will not solve your problem. If it does not work before the dial it will not work |
00:00.41 | ManxPower | BZBW, If you fight Asterisk you will be very unhapy. If you accept Asterisk's oddities and try to work with them. rather than against them, you will be happy. |
00:01.11 | *** join/#asterisk thermf (i=fadaasfa@adsl-68-73-6-126.dsl.sfldmi.ameritech.net) |
00:01.26 | ManxPower | docelm0, The internet is not reliable. Why would I want to have my business or my customer's businesses rely on the internet for something so basic as phone service. |
00:01.37 | *** join/#asterisk litage (n=nick@203.220.55.70) |
00:01.43 | BZBW | ha, I will never want to fight *, just that it is so powerful that I thought it can be done:) |
00:01.49 | ManxPower | I'll keep my VoIP calls on my managed LAN or QoS WAN, thankyouverymuch |
00:02.05 | ManxPower | BZBW, parkandannounce was not designed to do what you want it to do. |
00:02.06 | thermf | hi, does anyone know where rxfax/txfax is available with t38 support? |
00:02.23 | ManxPower | You are trying to do several steps to avoid learning how to do a supervised transfer |
00:02.28 | ManxPower | thermf, nowhere |
00:02.43 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
00:02.48 | thermf | ManxPower: there was a version, but it doesn't seem to be on steve underwood's site anymore |
00:02.51 | ManxPower | you're going to have to learn how to do a supervised transfer eventually |
00:05.57 | [TK]D-Fender | BZBW : I believe you can do what you want by substituting your SIP/${DIALEDPEERNUMBER} with a LOCAL channel that will add the header and continue to dial. |
00:07.59 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
00:08.19 | BZBW | sorry guys, got to take a call, thanks for the sugguestion, will give it a try. |
00:08.53 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
00:09.26 | BZBW | ManPower: the why that GS does Supervised Transfer does work for this:( |
00:11.16 | ManxPower | BZBW, Call someone. Then transfer them to someone else, but TALK TO THE DESTINATION PERSON before completing the transfer. That is a "supervised transfer" |
00:11.23 | ManxPower | If you can't do that then you can't park a call. |
00:11.50 | ManxPower | OR you can use the DTMF based supervised transfer hack. See features.conf and "w" and "W" options to dial. |
00:12.22 | [TK]D-Fender | ManxPower : No... he should by quality phones and use * as it ws intended :) |
00:12.42 | [TK]D-Fender | Polycom for the win! |
00:12.56 | ManxPower | [TK]D-Fender, you'd think people would learn that Grandstream makes terrible products |
00:13.01 | [TK]D-Fender | Just waiting for stock to come in on the IP 430..... last piece to add to my collection... |
00:13.29 | Zodiacal | things = thinks |
00:13.34 | [TK]D-Fender | ManxPower : Cheap people are dumb people. Some just don't appreciate the truth in YGWYPF |
00:14.18 | ManxPower | You will always get screwed by IP phones, the difference is that Grandstream doesn't use lube. |
00:14.45 | denon | ManxPower: you dont like those new linksys phones? cheap and very functional |
00:14.54 | denon | not ideal for power users, maybe .. but for avg users |
00:15.02 | [TK]D-Fender | ManxPower : I recently bought a katana. I could have cheaped out, but I refuse to waste my money on crap. So I researched like nuts to learn about the the products out there, sourced it right and then made my purchase of something that costs at least twice that of some entry level product. I am naturally thrilled with the end product... |
00:15.15 | [TK]D-Fender | ManxPower : How... colourful.... |
00:15.35 | ManxPower | denon, I've not tried them yet. we standardized on Polycom |
00:15.43 | [TK]D-Fender | denon : Linksys is *ok*, but in North America is simply not with the price difference with Polycom. |
00:15.47 | ManxPower | Users have enough trouble figuring out how to dial |
00:16.30 | [TK]D-Fender | Polycom quality and usability is considerably better than Linksys. |
00:17.15 | [TK]D-Fender | I say this of course having owned an SPA-941 and every Polycom IP desk phone they produce.... |
00:17.42 | copland | Anyone using Avaya 46xx series running the sip 2.2 image |
00:17.49 | ManxPower | [TK]D-Fender, "colorful" gets remembered. |
00:18.03 | denon | [TK]D-Fender: we havent done a ton with polycom .. I guess Im not real thrilled with how they treat partners |
00:18.16 | [TK]D-Fender | ManxPower : Indeed. |
00:18.56 | ManxPower | denon, and you are happier with SIPura/Linksys/Cisco "we won't give you the provisioning manual unless you are a partner and sign an NDA" |
00:19.00 | [TK]D-Fender | denon : Yeah, not the best perhaps, but its the product I care about and I have found nothing disappointing about them really. |
00:19.05 | BZBW | damn, talk about phones, my GS phone is working great other than this feature, I'm cheap:). |
00:19.23 | denon | ManxPower: we've already got good relationships with linksys and cisco, so yeah |
00:19.23 | ManxPower | BZBW, GS claims to support it |
00:19.44 | BZBW | Support what? |
00:19.54 | ManxPower | BZBW, supervised transfer |
00:20.14 | denon | I find the linksys is easier to train people on than a 7960 |
00:20.30 | denon | people whine a lot about the 7960s, but it's still my personal phone of choice |
00:20.57 | MACscr | is there like a voip directory of some sort that lets you input your zip code and prefix to find what voip options are available. Every place i have tried wont let me port my number |
00:21.05 | MACscr | i have my number with vonage right now |
00:21.17 | ManxPower | MACscr, Tried Teliax? |
00:21.37 | dlynes_office | [TK]D-Fender: Is there only certain versions of the kernel sangoma will work with? |
00:22.17 | [TK]D-Fender | dlynes_home : Well I've working mostly on CentOS/FC3 for which the stock always worked.... no idea really... |
00:22.35 | dlynes_office | [TK]D-Fender: but you said it worked on stock slackware 10.2 also? |
00:22.38 | [TK]D-Fender | dlynes_home : I wouldn't suspect it should be a problem. is that what theier techs are now suggesting? |
00:22.49 | MACscr | lol, thanks ManxPower |
00:22.54 | MACscr | they are now an option |
00:22.56 | sevard | [TK]D-Fender: give me free local DIDs and outbound. |
00:22.57 | sevard | go. |
00:22.58 | [TK]D-Fender | dlynes_office : : No, my S518 works fine in Slack.... |
00:22.58 | MACscr | do you use them? |
00:23.00 | dlynes_office | [TK]D-Fender: nope...haven't talked to them yet, but I remember you said it work on stock slack, too |
00:23.15 | [TK]D-Fender | sevard : LOL. |
00:23.20 | dlynes_office | [TK]D-Fender: do you know what module 'cdev_...' is defined int, then? |
00:23.20 | sevard | i'm drinking. |
00:23.31 | [TK]D-Fender | dlynes_home : PM |
00:23.37 | ManxPower | MACscr, I did before I moved and the only IP connection is 900ms - 1500ms with up to 1000ms of jitter. |
00:23.37 | dlynes_office | wtf? |
00:23.43 | dlynes_office | you mean power management? |
00:23.45 | ManxPower | VoIP doesn't work well in that enviroment |
00:23.54 | sevard | i'd go for a free outbound service with a 20 second ad prepending each call |
00:23.58 | [TK]D-Fender | dlynes_home : Private Message |
00:24.01 | dlynes_office | ah |
00:24.05 | *** join/#asterisk litage (n=nick@203.220.55.70) |
00:24.20 | sevard | dlynes_office: [TK]D-Fender is from AOL. He means /msg |
00:24.21 | BZBW | ManxPower: u can not use ParkAndAnnounce() with Supervised Transfer, do you? |
00:24.30 | [TK]D-Fender | sevard : You're cheap.... go get a GXP-2000 to place those calls on! |
00:24.40 | sevard | eww |
00:24.49 | docelm0 | hay I have a GXP2000 sitting right next to me.. |
00:24.53 | docelm0 | it works perfectly fine |
00:25.16 | docelm0 | granted I flashed the hell out of it. |
00:25.18 | sevard | I think I'm going to sign up for a ShellShark account, an unlimited personal, 2 channels 24/mo |
00:25.28 | [TK]D-Fender | docelm0 : 2.2 Gigawatts? ;) |
00:25.33 | ManxPower | BZBW, There is NO NEED FOR PARK AND ANNOUNCE IF YOU USE SUPERVISED TRANSFER |
00:25.40 | sevard | docelm0: you can't flash it to make it feel less like a fisher price phone. |
00:25.51 | BZBW | hey, I got paging/intercom, one way, two way, BLF, all working with my GXP2000:) |
00:25.52 | sevard | BZBW: JESUS CHRIST HOW MANY TIMES DOES HE HAVE TO FUCKING SAY IT |
00:25.59 | docelm0 | dunno.. it works fine for me.. |
00:26.08 | docelm0 | sevard bitch slap em and get over it. |
00:26.19 | sevard | I use that hand for touching myself. |
00:26.27 | ManxPower | People that don't listen get on /ignore. |
00:26.27 | BZBW | Sorry guys, just don't get it in my head:( |
00:26.37 | ManxPower | BZBW, read The Book |
00:26.41 | ManxPower | ~docs |
00:26.48 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
00:26.48 | sevard | ~thebook |
00:26.50 | docelm0 | ~mybutt |
00:26.52 | jbot | mybutt is probably HUGE and stands for some funky stuff... |
00:26.52 | sevard | ~bothammer |
00:27.22 | ManxPower | ~afot |
00:27.26 | ManxPower | ~atfot |
00:27.56 | ManxPower | I admin like 8 asterisk systems, two of the do parking, none of them use parkandannounce |
00:28.16 | sevard | The only transfer I've ever gotten to work is flash dial hangup |
00:28.56 | ManxPower | sevard, try flash, dial, talk, hangup |
00:28.59 | sevard | I've never gotten features.conf to work for that fancy *3 or whatever blind transfer/supervised transfer |
00:29.05 | sevard | ManxPower: that's a supervised transfer |
00:29.12 | sevard | it'd be cool to know how to do it on * though |
00:29.13 | ManxPower | sevard, exactly |
00:29.28 | ManxPower | sevard, on Zap you need threewaycalling and transfer enabled |
00:29.37 | BZBW | ManxPower: ha, I remember why I kind like parkandannounce, I just need to blind transfer to *3, whereas for supervised transfer, I have to press more keys. |
00:29.49 | ManxPower | BZBW, live with it. |
00:30.39 | sevard | ManxPower: hmm, where? |
00:30.56 | ManxPower | sevard, in /etc/asterisk/zapata.conf |
00:31.03 | ManxPower | assuming you are using zap |
00:31.26 | sevard | I'm using zap, but what's stopping a person calling in parking or transfering his own call? |
00:31.50 | ManxPower | sevard, they won't be able to flash the telco line from a remote location |
00:32.11 | sevard | but if they're on a channelbank they will be able to, i'm assuming |
00:32.38 | ManxPower | If they are "calling in" then they are coming from the PSTN and you can't send a flash across the PSTN |
00:32.49 | sevard | ah |
00:32.55 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
00:32.57 | ManxPower | of course you could disable transfer for the PSTN channels |
00:33.07 | ManxPower | enable it for the local analog channels |
00:33.14 | sevard | How I have it working now is that there are only Sipura ATAs on my asterisk box and they dial out or in via my PRI |
00:33.25 | ManxPower | you can do all this stuff on a channel-by-channel basis |
00:33.37 | *** join/#asterisk linlin (i=linlin@c-67-184-152-231.hsd1.il.comcast.net) |
00:33.44 | ManxPower | sevard, then you are not using ANY Zap for extensions |
00:33.48 | sevard | and you can flash on those babys, supervised transfer, blind transfer, etc... Just thinking it'd be sweet to do it like the Pros do it |
00:33.50 | ManxPower | so everything I said is not valid |
00:34.07 | sevard | ManxPower: well, I will be phucking with a tdm2400P pretty soon |
00:34.18 | ManxPower | sevard, I'm sorry. |
00:34.22 | justinu | heh |
00:34.33 | sevard | hooking up 90 some odd analog rooms at a client's location |
00:34.35 | sevard | never done it before |
00:34.37 | sevard | I'm sorry? :) |
00:34.42 | MACscr | sevard, what voip provider do you use |
00:34.51 | justinu | he's got a pri |
00:34.52 | MACscr | sry, just doing a little polling |
00:34.55 | ManxPower | sevard, why not a channelbank and a T-1 port? |
00:34.58 | sevard | MACscr: At the moment I'm using Teliax for my personal shit |
00:35.07 | sevard | ManxPower: That solution had come up too |
00:35.20 | sevard | they're already on 66blocks |
00:35.45 | ManxPower | What does that have to do with anything? |
00:35.47 | MACscr | 66, come on, get with the times, 110's =P |
00:35.51 | sevard | nothing at all |
00:36.05 | sevard | i'm just a little... |
00:36.06 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
00:36.09 | sevard | intoxicated? |
00:36.16 | MACscr | lol |
00:36.29 | sevard | i've been drinking white russians for the past 4 hours |
00:36.33 | ManxPower | I suppose I should hit the drive thru daqueri(sp!) shop on the way hime. |
00:36.46 | sevard | hime? looks like you don't need it. |
00:36.46 | ManxPower | home too. |
00:36.57 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
00:37.00 | ManxPower | sevard, that's just from working since 5am |
00:37.03 | sevard | Man, that's one of my big pet peeves |
00:37.06 | sevard | "thru" |
00:37.07 | sevard | I hate that. |
00:37.20 | justinu | it's a valid spelling |
00:37.24 | justinu | it's even in the dictionary |
00:37.25 | sevard | I don't care. |
00:37.27 | ManxPower | sevard, whyz dat d00d? |
00:37.28 | sevard | It shouldn't be. |
00:37.41 | sevard | If I was in charge of m-w I'd burn that god damn "word" |
00:37.57 | *** join/#asterisk JASON99 (n=jason@jason.unitz.ca) |
00:38.08 | ManxPower | Anyway, I'm outta here. |
00:38.12 | sevard | later man |
00:38.46 | JASON99 | hello, i'm setting up a pre-paid system for sip users. Is there open source software that you recommend using? |
00:39.39 | sevard | http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications |
00:39.44 | sevard | use |
00:39.45 | sevard | the |
00:39.45 | sevard | effing |
00:39.46 | sevard | wiki |
00:39.57 | JASON99 | sevard: FYI i already looked there |
00:40.06 | sevard | cool, then you already know. |
00:40.12 | justinu | <PROTECTED> |
00:40.12 | JASON99 | no.. I asked for recommendations |
00:40.18 | justinu | who the fuck leases a car for 84 months? |
00:40.22 | sevard | AstBill |
00:40.40 | sevard | justinu: Nobody I know |
00:40.59 | justinu | <PROTECTED> |
00:41.04 | Neptune__ | with what command line command (windows or linux) can i check a enum entry? |
00:41.06 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
00:41.21 | JASON99 | Thanks sevard |
00:41.25 | *** join/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net) |
00:41.43 | sevard | justinu:................. |
00:41.54 | omarc55 | Hi all, I am trying to get sip with a video call to work behind nat, does anybody know what ports I have to forward? |
00:41.56 | justinu | full artcle here: http://www.edmunds.com/advice/specialreports/articles/115584/article.html |
00:42.00 | sevard | wow. |
00:42.26 | sevard | that gives me an idea for a new profession |
00:42.38 | sevard | anyone have an insured SUV? |
00:42.49 | docelm0 | yes |
00:43.01 | docelm0 | Well my better half.. 2004 Jeep |
00:43.01 | Neptune__ | omarc55 - 5060 and the rtp range in your rtp.conf |
00:43.07 | sevard | docelm0: how'd you like to trade it in for a charred ashtray and a pile of bling |
00:43.11 | sevard | or coke |
00:43.26 | justinu | heh |
00:43.29 | sevard | or a delorian |
00:43.35 | sevard | dc12 |
00:43.56 | omarc55 | alright, let me give that a shot |
00:44.04 | Neptune__ | the dmc12 doesnt guzzle that mutch of gas... |
00:44.07 | *** join/#asterisk b00mer (n=b00mer@ip24-255-125-65.dc.dc.cox.net) |
00:44.10 | sevard | fish tank? |
00:44.18 | sevard | i'd kill a hobo for a DC12. |
00:44.29 | Neptune__ | yeah me too... |
00:44.38 | Neptune__ | you can get them as kits of spare parts |
00:44.43 | sevard | i'd do it with a pen knife |
00:44.43 | Neptune__ | quite expensive though |
00:44.59 | sevard | in the throat |
00:45.21 | sevard | damn, i always thought it was DC, what's DMC now |
00:45.42 | Neptune__ | the car is called "De Lorean DMC-12" |
00:45.56 | sevard | excuse me, i'm getting low |
00:46.10 | sevard | Does anyone's PRI freaking restart like every hour or some crap |
00:46.46 | Strom_C | sevard: the b-channels are supposed to be restarted every hour |
00:46.47 | Neptune__ | DC12 - could be an airplane - but then it would be MD12 - that was once planned but it never when't into production after MD got bought up by boeing |
00:46.53 | docelm0 | yes |
00:47.16 | omarc55 | Neptune__: do I have to forward them to the pbx or to local sip client in the network? |
00:47.17 | sevard | wazzzzzzzzzzzzup |
00:47.30 | Strom_C | I bought sandals |
00:47.35 | Strom_C | totally awesome sandals |
00:47.38 | Neptune__ | omarc55 to the PBX |
00:47.39 | Strom_C | they're like sex for my feet |
00:47.51 | sevard | i always wanted to learn how to make roap sandals. |
00:47.55 | omarc55 | it didn't work. not sure what I am doing wrong, they can't see me, I can't see them. |
00:48.48 | sevard | omarc55: do you have a line... nat=yes in your sip.conf |
00:48.59 | omarc55 | yes, I do. |
00:49.03 | sevard | Strom_C: where have you been in the last month |
00:49.16 | omarc55 | and I also set the externip |
00:49.17 | Neptune__ | omarc55 - the reinvite won't work - so you need to turn that off |
00:49.32 | Neptune__ | (afaik) |
00:49.39 | *** join/#asterisk litage (n=nick@203.220.55.70) |
00:49.43 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
00:51.10 | Strom_C | sevard: busy as all fuck |
00:51.15 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
00:51.21 | sevard | stop fucking and get on #asterisk |
00:51.33 | Strom_C | I am o #asterisk |
00:51.34 | Strom_C | silly |
00:51.40 | sevard | more often, idiot. |
00:51.59 | sevard | contractor work? |
00:52.15 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
00:52.35 | Strom_C | contract work, consulting, school |
00:53.06 | sevard | school, eh? |
00:53.12 | Strom_C | yes |
00:53.17 | Strom_C | fortunately that finished yesterday |
00:54.46 | sevard | what were you taking? |
00:54.53 | Strom_C | general courses |
00:54.59 | Neptune__ | with what command line command (windows or linux) can i check a enum entry? |
00:55.01 | Strom_C | it all bored me to tears |
00:56.27 | *** join/#asterisk Lino` (n=Lino@i577BD710.versanet.de) |
00:56.53 | sevard | Strom_C: I was just about to type "gay" |
00:57.10 | Strom_C | no, i dont think you can take gay at college |
00:57.20 | drray | lesbian studies |
00:57.22 | sevard | Not according to a friend of mine |
00:57.26 | sevard | he's having the time of his life |
00:57.33 | sevard | -c |
00:58.09 | Strom_C | Cruising 101? |
00:58.59 | sevard | more like dorm-room saussage tag |
01:00.07 | Strom_C | sweet |
01:00.22 | sevard | heh |
01:01.38 | omarc55 | Neptune__: still doesn't work, I am trying to get 2 sip clients to register to asterisk and then video conference between the 2, that is possible right? |
01:02.21 | sevard | very possible |
01:02.34 | Neptune__ | never tried video, but i figure it should work |
01:02.37 | sevard | you ought to put them all on the same lan |
01:02.42 | Neptune__ | are the 2 clients on the same net? |
01:02.47 | sevard | I used eyebeam and h232p |
01:05.25 | omarc55 | no, they are not. |
01:05.30 | omarc55 | and I just got a message on the console |
01:05.53 | omarc55 | Maximum retries exceeded on transmission <number here> for seqno 3 (Non-critical Response) |
01:07.05 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
01:09.27 | sevard | dude, it's totally non-critical. don't even worry. |
01:09.51 | *** join/#asterisk litage (n=nick@203.220.55.70) |
01:09.56 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:11.50 | copland | any avaya 46xx series users in here |
01:12.32 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
01:12.33 | surfdue | hello there! |
01:12.46 | surfdue | I was wondering I have my voicemails sent to my email, how can they be automatcially deleted? |
01:13.11 | Strom_C | surfdue: that setting is right there in the sample voicemail.conf |
01:13.16 | surfdue | k |
01:17.20 | surfdue | ty |
01:17.45 | surfdue | Now to fix sip, for some reason my sip phone keeps saying Can't connect to login server using hostname or ip |
01:17.45 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:17.48 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
01:17.54 | *** join/#asterisk litage (n=nick@203.220.55.70) |
01:18.20 | surfdue | Is there maybe an obvoiusly problem with why this happens, asterisk -r consol shows when people call so it isnt being blocked |
01:18.43 | SkramX | <-- Back from dinner and buying an overpriced iced coffee drink at Starbucks |
01:19.30 | surfdue | aparently that isnt the case |
01:19.44 | surfdue | which is odd asterisk -r with sip debug enables shows no info at all when calling the number |
01:19.59 | JASON99 | i'm trying to compile addons but i keep getting errors |
01:20.00 | JASON99 | cdr_addon_mysql.c:38:19: mysql.h: No such file or directory |
01:20.00 | JASON99 | cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory |
01:20.13 | Strom_C | surfdue: what do you have for "login server" on the phone? |
01:20.20 | Strom_C | JASON99: do you have mysql installed? |
01:20.25 | surfdue | host41.com |
01:20.32 | JASON99 | Strom_C: yes |
01:20.48 | Strom_C | *shrug* |
01:21.02 | JASON99 | Strom_C: Ver 5.0 |
01:21.10 | surfdue | Strom_C, its host41.com |
01:21.26 | Strom_C | surfdue: is host41.com the URL of your asterisk box? |
01:21.33 | surfdue | thats right |
01:21.51 | Strom_C | maybe your phone is on crack |
01:22.11 | surfdue | i was kinda thinking the same |
01:22.12 | surfdue | lol |
01:22.30 | surfdue | I have tried ip and hostnmae |
01:22.37 | JackEStorm | JASON99: dude, use ODBC and forget cdr_addon crap |
01:22.43 | surfdue | Strom_C, can you see maybe test yourself if you can reach the server host41.com on port 5060? |
01:22.56 | JASON99 | JackEStorm: I can write to mysql with ODBC? |
01:23.07 | Strom_C | no, I think I'm going to take a nap instead |
01:23.15 | surfdue | Registration State:Can't connect to login server |
01:23.26 | JackEStorm | JASON99: do you know what ODBC is? |
01:23.30 | JASON99 | yes, I got it :P |
01:24.42 | JASON99 | JackEStorm: I've just never setup a dsn in linux, but I'll figure it out.. Thanks |
01:25.34 | surfdue | Strom_C, Next Registration In: is simply blank? |
01:25.35 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:25.44 | *** join/#asterisk litage (n=nick@203.220.55.70) |
01:25.45 | surfdue | mnow it snot |
01:26.05 | JackEStorm | JASON99: it's simple and easy, look at cdr_odbc.conf, and your odbc install default .ini to figure it out |
01:26.16 | dlynes_office | JASON99: You don't have mysql-dev installed |
01:26.28 | dlynes_office | JASON99: that's why you're getting all those compiler errors |
01:29.29 | JackEStorm | dlynes_office: yeah, but he's better off using ODBC, if he plans on using more than bdb and SQLITE, just makes migration across OS simple. |
01:29.54 | dlynes_office | JackEStorm: yeah, I realize that |
01:30.05 | JASON99 | Thanks guys, I'll use ODBC |
01:30.08 | dlynes_office | JackEStorm: but i figured i'd at least get him up and running first |
01:30.29 | JASON99 | Will I still need mysql-dev if I use the ODBC? |
01:30.36 | dlynes_office | JackEStorm: besides...if he ends up having to compile the unixodbc driver for mysql, he might still need mysql-dev |
01:31.02 | JackEStorm | JASON99: no, if you are properly packagetized |
01:31.18 | JackEStorm | dlynes_office: true...but: |
01:31.24 | JackEStorm | JASON99: what distro? |
01:31.38 | dlynes_office | well, true |
01:31.51 | dlynes_office | his name's all in caps lock...he's probably using LINSPIRE |
01:31.53 | JASON99 | RedHat |
01:32.00 | dlynes_office | heh...close enough :p |
01:32.19 | JackEStorm | dlynes_home: wait |
01:32.30 | JackEStorm | Jason99: RH what? |
01:32.31 | dlynes_office | heh |
01:32.35 | Jason99 | hehe |
01:32.39 | Jason99 | 9 |
01:32.45 | sevard | linspire |
01:32.46 | sevard | redhad |
01:32.48 | sevard | close enough |
01:32.49 | sevard | ahahahahaha |
01:32.55 | Jason99 | :P |
01:33.28 | JackEStorm | dude, UPGRADE!!!!!! |
01:33.28 | dlynes_office | dood |
01:33.28 | dlynes_office | redhat post version 9 isn't free |
01:33.43 | JackEStorm | thats why FC exists |
01:33.43 | sevard | it shouldn't. |
01:33.49 | dlynes_office | and slackware and debian and ubuntu and ... |
01:33.54 | JackEStorm | move to Debian, or something else |
01:33.55 | sevard | slax :) |
01:34.24 | dlynes_office | incidentally |
01:34.29 | sevard | jjjjjjjjjjjjjjjjjjjjjjjj |
01:34.30 | JackEStorm | dlynes_home: never offer slak to someone who might not know what an Admin really Is |
01:34.31 | Jason99 | ok I'll upgrade some day lol |
01:34.33 | dlynes_office | WHEN THE HELL IS PATRICK GOING TO SHIP 11.0? |
01:35.04 | sevard | dlynes_office: I know, as much as I enjoy 10.2 linux has inproved greatly since its release |
01:35.19 | sevard | improved |
01:35.20 | dlynes_office | sevard: well, not just that |
01:35.26 | dlynes_office | i want 2.6 to the default kernel |
01:35.28 | dlynes_office | 2.4 sucks |
01:35.34 | sevard | hey now |
01:35.41 | sevard | i just switched to 2.6 about two months ago |
01:35.41 | dlynes_office | lemme rephrase |
01:35.45 | dlynes_office | 2.4 sucks for voiop |
01:35.47 | sevard | until then 2.4 did wonders for me |
01:35.50 | *** join/#asterisk litage (n=nick@203.220.55.70) |
01:35.52 | sevard | I also used asterisk on 2.4 |
01:36.03 | dlynes_office | sevard: if you want to use sangoma ec, you need 2.6 |
01:36.07 | dlynes_office | it won't work on 2.4 |
01:36.12 | sevard | sangoma shanagomoa |
01:36.33 | justinu|laptop | sangoma rocks |
01:36.37 | sevard | but yeah, i'm willing to bet slackware 10.3 or 11 or whatever it'll be will be 2.6 |
01:36.52 | sevard | i don't even know what sangoma is :) |
01:37.21 | JackEStorm | dlynes_home: I have only used 3Linux Distros, sans roll your own...first was SLS |
01:37.38 | dlynes_office | sls is what slackware was based on |
01:37.44 | dlynes_office | it's what slackware replaced :p |
01:37.57 | *** join/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com) |
01:37.59 | sevard | bob gives us slack. |
01:38.01 | JackEStorm | nod, Slak was #2 |
01:38.04 | justinu|laptop | they're a digium telephony card competitor |
01:38.22 | sevard | thsi kid I know asked me to join the church of the subgenious a couple of weeks ago |
01:38.32 | sevard | I was like duuuddee, i've been using slackware since you were in diapers |
01:38.35 | dlynes_office | sevard: sangoma's a cool Canadian company that makes kick ass telephony cards |
01:38.48 | sevard | dlynes_office: I'll check them out, native * support? |
01:38.56 | justinu|laptop | yeah |
01:39.02 | sevard | what kind of cards? |
01:39.06 | dlynes_office | sevard: yeah...they run on top of zaptel.o/zaptel.ko |
01:39.07 | justinu|laptop | t1, analog |
01:39.20 | sevard | what's the difference in price/ support |
01:39.25 | dlynes_office | sevard: fxs, fxo, single port pri, dual port pri, quad pri, octo pri |
01:39.27 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
01:39.28 | dlynes_office | sevard: hardware ec |
01:39.29 | justinu|laptop | one thing nice about sangoma, is you can terminate a T1 running both PRI and data |
01:39.29 | surfdue | hi |
01:39.33 | justinu|laptop | on one card |
01:39.40 | surfdue | i need some help guys I cant communite with my servers asteirks |
01:39.44 | sevard | nice! |
01:39.56 | surfdue | is tehre any way to make sure its not running behind a nat |
01:40.01 | dlynes_office | justinu|laptop: you mean because the sangoma card is also a network card? |
01:40.09 | sevard | surfdue: yes. |
01:40.11 | justinu|laptop | dlynes_office: sorta like that, yeah |
01:40.15 | dlynes_office | ah |
01:40.24 | justinu|laptop | it'll give you a ppp0 interface |
01:40.26 | sevard | what's the difference in price/ support |
01:40.32 | dlynes_office | so dchannel goes into netowkr |
01:40.33 | justinu|laptop | they're the same price basically |
01:40.36 | dlynes_office | and bchannel is voice? |
01:40.42 | surfdue | sevard, how |
01:40.43 | sevard | what about post-buy support? |
01:40.49 | sevard | digium will give you free setup support |
01:40.50 | justinu|laptop | seemed good when I had a few questions |
01:40.52 | [TK]D-Fender | 2.4 is FINE. I've run * on Slack 2.4 stock kernels for over 2 years now... |
01:40.54 | justinu|laptop | same with sangoma |
01:41.07 | dlynes_office | [TK]D-Fender: hehe |
01:41.25 | sevard | [TK]D-Fender: they are right, 2.6 is a little easier to work with. |
01:41.25 | dlynes_office | [TK]D-Fender: i just like 2.6 cause it's got more features that make it more amenable to telephony |
01:41.36 | justinu|laptop | dlynes_office: you would do something like: run 12 timeslots of PPP (768k) and 11 b-channels, and 1 dchannel |
01:41.43 | sevard | but I'm sad to see my good old 2.4 go |
01:41.46 | [TK]D-Fender | dlynes_office : And yes all of Sangoma's tech is founded in the networking world. Voice = Data. |
01:41.57 | justinu|laptop | sangoma wanpip will give you a ppp0 interface which is that 768kbps serial link delivered via t1 |
01:42.09 | justinu|laptop | and then expose the other 12 channels to asterisk |
01:42.21 | dlynes_office | justinu|laptop: cool |
01:42.25 | [TK]D-Fender | justinu : I have an underutilized PRI I should maybe fractionalize.... |
01:42.25 | justinu|laptop | using the native zaptel driver |
01:42.37 | justinu|laptop | probably a good idea to leverage that T1 pipe |
01:42.37 | sevard | fractionalize? |
01:42.39 | sevard | hahaha |
01:42.56 | [TK]D-Fender | sevard: Split to partial PRI and use the other channels for PPP |
01:43.17 | sevard | right. i'm just laughing at your use of that 'word' |
01:43.31 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:43.38 | JackEStorm | [TK]D-Fender: work out VtoD and I'm in |
01:43.53 | surfdue | anyone? |
01:43.58 | surfdue | sevard, plz |
01:44.04 | sevard | please what niggua |
01:44.12 | dlynes_office | surfdue: type sip show peer peername |
01:44.20 | dlynes_office | surfdue: does it show a private ip in that info? |
01:44.25 | justinu|laptop | lmao |
01:44.31 | justinu|laptop | JackEStorm: VtoD? |
01:44.41 | dlynes_office | prolly voice to data |
01:44.46 | sevard | holy crap i have to take a shower |
01:44.46 | JackEStorm | justinu: voice traffic on demand |
01:44.47 | sevard | bbl |
01:44.49 | justinu|laptop | ah |
01:44.50 | dlynes_office | oh |
01:44.58 | dlynes_office | crap isn't holy |
01:45.03 | dlynes_office | unless it's blessed |
01:45.03 | sevard | mine are |
01:45.05 | [TK]D-Fender | dlynes_office : Mass-recompile still in progress I take it? |
01:45.05 | justinu|laptop | well... atm can do that |
01:45.12 | dlynes_office | [TK]D-Fender: yeah |
01:45.13 | justinu|laptop | you could deliver an ATM link via T1 |
01:45.17 | dlynes_office | [TK]D-Fender: removed all the sangoma shit |
01:45.17 | justinu|laptop | terminate that with a sangoma card |
01:45.20 | sevard | you have to smell my crap to understand its holyness. |
01:45.23 | dlynes_office | [TK]D-Fender: and now i'm rebuilding the kernel |
01:45.26 | sevard | kbbl |
01:45.28 | sevard | kthxBAI |
01:45.30 | surfdue | type "sip show peer peername" into asterisk -r? |
01:45.41 | dlynes_office | [TK]D-Fender: after having reinstalled the sangoma patches to the kernel |
01:45.58 | dlynes_office | [TK]D-Fender: and alex the sangoma tech just signed on for the night on msn |
01:46.10 | [TK]D-Fender | dlynes_office : Yeah, you needed to clean up your SRC folder a bit, put theng back where they belong... but odds are looking up... libpri, zaptel,wanpipe,zaptel,asterisk, adnt hen all should be good. |
01:46.25 | justinu|laptop | dlynes_office: what is your issue? |
01:46.26 | surfdue | dlynes_office, that command isnt reconized in asterisk -r |
01:46.32 | justinu|laptop | lmaro |
01:46.46 | dlynes_office | surfdue: type load chan_sip.so |
01:47.11 | surfdue | Unable to load module chan_sip.so |
01:47.11 | surfdue | Jun 9 21:47:05 WARNING[18806]: loader.c:305 __load_resource: Module 'chan_sip.so' already exists |
01:47.20 | [TK]D-Fender | dlynes_office : Don't... I bet he's trying to type that mass of crap as a poorly formatted Linux CLI command... |
01:47.21 | dlynes_office | surfdue: ummm |
01:47.30 | justinu|laptop | lol |
01:47.39 | dlynes_office | surfdue: wtf, dood? I didn't mean 'peername', literally |
01:47.41 | surfdue | Peer peername not found |
01:47.48 | surfdue | :| |
01:47.50 | surfdue | oh |
01:47.50 | surfdue | lol |
01:47.53 | surfdue | duh. |
01:47.53 | justinu|laptop | "nohup rm -rf / &;logout' |
01:47.56 | [TK]D-Fender | surfdue : YOURE SUPPOSED TO PUT YOUR PEER NAME THERE! |
01:48.04 | surfdue | what do you mean by peer name? |
01:48.07 | surfdue | ip? |
01:48.20 | [TK]D-Fender | surfdue : The named [] entry in sip.conf! |
01:48.21 | dlynes_office | surfdue: whatever the name of the sip peer is that's trying to connect to your box |
01:48.21 | surfdue | extension? |
01:48.24 | surfdue | ok |
01:48.34 | dlynes_office | surfdue: the name you declared in your sip.con |
01:48.34 | [TK]D-Fender | surfdue :... |
01:48.35 | dlynes_office | surfdue: the name you declared in your sip.conf |
01:48.35 | [TK]D-Fender | ~book |
01:48.41 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
01:48.41 | surfdue | ok i see now |
01:48.42 | dlynes_office | surfdue: the one you see when type sip show peers |
01:48.45 | surfdue | now what am i looking for |
01:49.03 | dlynes_office | surfdue: down at the bottom, you'll see an external ip and an internal ip |
01:49.06 | surfdue | <PROTECTED> |
01:49.14 | dlynes_office | ok...and down at the bottom? |
01:49.19 | dlynes_office | the very buttom? |
01:49.24 | dlynes_office | bottom |
01:49.39 | justinu|laptop | <PROTECTED> |
01:49.44 | [TK]D-Fender | surfdue : Pastebin your entire damn sip.conf and lets ahve a look at it. |
01:49.45 | [TK]D-Fender | ~pb |
01:49.46 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
01:49.47 | dlynes_office | justinu|laptop: don't have an issue at the moment |
01:49.49 | surfdue | k |
01:49.50 | justinu|laptop | oh |
01:50.10 | dlynes_office | justinu|laptop: but the issue earlier was that the sangoma card couldn't allocate memory off the pci bus |
01:50.14 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:50.14 | justinu|laptop | oh |
01:50.21 | dlynes_office | justinu|laptop: that was in 2.6 |
01:50.31 | dlynes_office | justinu|laptop: tk got the driver loading in 2.4 though |
01:50.47 | dlynes_office | justinu|laptop: so i'm just rebuildijng everything from scratch to make sure I understand how it works |
01:50.48 | justinu|laptop | all that stuff "just worked" for me |
01:50.51 | [TK]D-Fender | justinu : It compiled wrong somehow but I got it up, and jsut want a clean recompile so that chan_zap doesn't whine... |
01:51.05 | [TK]D-Fender | justinu : For me as well. |
01:51.06 | justinu|laptop | i just followed their install instructions |
01:51.15 | justinu|laptop | but it was an a101 card |
01:51.22 | surfdue | http://host41.pastebin.com/771195 |
01:51.57 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
01:52.02 | kumamoto | I wondering where to buy a good and inexpensive IP Phone with dual nics |
01:52.13 | surfdue | sorry http://host41.pastebin.com/771195 is my sip.conf |
01:52.19 | *** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
01:52.29 | Guest^DJ | hi, does anyone know a SIP based GSM channel bank |
01:52.58 | docelm0 | Check the wiki I just bought a 32 channel one for my australia office |
01:53.28 | Guest^DJ | 32 channel, wow i am thinking of only 4-8 |
01:53.59 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
01:54.15 | docelm0 | Its for the calling card company I used to work for? |
01:54.20 | docelm0 | for... |
01:54.26 | Qwell | for@ |
01:54.53 | Guest^DJ | i did check the wiki, found a Voiceblue, wrote them 2 says ago, and nothing happen |
01:54.54 | kumamoto | I didn't know that one can use calling card with voip |
01:55.19 | Qwell | kumamoto: it's just a call |
01:55.25 | docelm0 | yes.. The company I just resigned from is 100% voip except 8 T1's |
01:55.39 | docelm0 | We have an 80T1 switch and run tons of calls |
01:56.56 | justinu|laptop | excel? |
01:57.21 | *** join/#asterisk AsteriskGURU (i=Asterisk@24-117-117-211.cpe.cableone.net) |
01:57.25 | docelm0 | yep |
01:57.27 | kumamoto | Qwell, I thought there was a special expensive thingie that has to used to convert a pbx to use calling cards |
01:57.33 | justinu|laptop | heh... i worked on excels for 7 years |
01:57.35 | [TK]D-Fender | surfdue : So... pastebin "sip show peers" from the * CLI |
01:57.35 | docelm0 | I have 2 EXS2000's with a fiber ring |
01:57.39 | *** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com) |
01:57.39 | justinu|laptop | wrote a lot of call control code |
01:57.42 | AsteriskGURU | does anyone here know anything about two b channel transfer? |
01:57.53 | Qwell | kumamoto: sure, with a traditional pbx |
01:57.56 | justinu|laptop | glad to be done with those things! |
01:58.31 | docelm0 | I just left the company to take a job @ a company in NYC. Got more money and possibly alot more.. I will know in a couple months.. Depends on a merger in process right now |
01:58.34 | justinu|laptop | i used to be able to decode the EXS HEX API by sight |
01:58.35 | ManxPower | AsteriskGURU, nobody does. Well, except maybe for the Masons and Illumati, but neither of them are talking. |
01:58.40 | kumamoto | Qwell:so nothing special with an asterisk pbx? |
01:58.49 | Qwell | kumamoto: just a free app |
01:58.56 | sevard | ahh, squeaky clean. |
01:58.56 | Qwell | or write one yourself |
01:59.14 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
01:59.14 | AsteriskGURU | lol @ ManxPower U sure? theres code in asterisk for it on 5ESS but NI2 is the big mystery |
01:59.32 | ManxPower | AsteriskGURU, I doubt 2BCT is supported on anything except 5ESS |
01:59.34 | kumamoto | Qwell: oh really? So which app is that? |
01:59.36 | mitcheloc | AsteriskGURU: are you affiliated with asteriskguru.com? |
01:59.41 | mitcheloc | docelmo: are they hiring ;) |
01:59.42 | Qwell | there are several |
01:59.48 | sevard | pretty sure NI2 is supported. |
01:59.51 | AsteriskGURU | um, asteriskguru.com never heard of it, better change my name huh? |
01:59.51 | justinu|laptop | NI2 spec includes 2BCT capability |
02:00.02 | mitcheloc | AsteriskGURU: probably =P |
02:00.05 | sevard | Liar! |
02:00.15 | *** join/#asterisk litage (n=nick@203.220.55.70) |
02:00.15 | ManxPower | yeah, but I doubt Asterisk supports it on NI2 |
02:00.16 | AsteriskGURU | ok, i thought NI2 had it.... but has anybody made it work with asterisk? |
02:00.48 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
02:01.01 | AsteriskGURU | since i doubt i'll be able to collect that 5000 bounty thats been out there for a while what if i said i have it working, got it working 20 minutes ago with 10 extra lines of code and had to gloat |
02:01.27 | mitcheloc | which bounty? |
02:01.36 | justinu|laptop | AsteriskGURU: nice work |
02:01.43 | surfdue | sip show peers and show peer 200, http://host41.pastebin.com/771200 |
02:01.57 | Assid | just curious.. does SIP provide better stability as compared to iax? since it has more streams? |
02:01.58 | AsteriskGURU | somebody had a 5k bounty for 2BCT but it says they retain the rights to the source code..... uh, well mines a patch, not code, so i doubt i can claim it |
02:02.16 | sevard | just fork the source and go :) |
02:02.19 | justinu|laptop | heh |
02:02.35 | sevard | if you want 10 and a half Gs though get T.38 support |
02:02.37 | mitcheloc | do what sevard said and you are 5k richer |
02:02.43 | ManxPower | LOL! Many of the firehouses in New Orleans were damaged, so the city leased trailers for the fire fighters to live in, but the city has not made any payments on the leases in 5 months |
02:03.05 | AsteriskGURU | asterisk does T38 doesnt it? |
02:03.10 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
02:03.11 | justinu|laptop | no |
02:03.13 | sevard | I think only T30 or something |
02:03.14 | surfdue | anyone have any suggestions from the past? |
02:03.16 | surfdue | paste* |
02:03.17 | AsteriskGURU | um, my voip adapter does it |
02:03.29 | docelm0 | AsteriskGURU not yet.. 1.4 will |
02:03.29 | sevard | sure your ata does |
02:03.35 | mitcheloc | what is the reason asterisk hasn't done t.38 yet? |
02:03.47 | AsteriskGURU | i'm really a newbie programmer, i dont know what fork the code means |
02:03.48 | justinu|laptop | apparently the ATAs don't support t38 very well |
02:03.49 | Qwell | mitcheloc: it's being implemented |
02:03.49 | AsteriskGURU | lol |
02:03.54 | AsteriskGURU | and i dont even know how to make a patch really |
02:03.54 | sevard | AsteriskGURU: dude, get your self five grand. |
02:04.01 | [TK]D-Fender | surfdue : Where is SIP/200 relative to your * server? |
02:04.02 | sevard | send me the code and i'll give you 2,500 |
02:04.17 | surfdue | [TK]D-Fender, what do you mean |
02:04.29 | AsteriskGURU | lol @ sevard, heres the other problem, if i get the 5000 bucks from this guy it doesnt help the community any cause they are going to use it for private use and then i cant tell you how i did it |
02:04.42 | sevard | you can't but I can |
02:04.45 | [TK]D-Fender | sevard : If you want to get rich quick, I've got some ocean-side property in nevada I'm willing to sell you cheap ;) |
02:04.57 | sevard | [TK]D-Fender: eat me. |
02:05.07 | mitcheloc | ocean + nevada? |
02:05.16 | sevard | mitcheloc: that's the joke. |
02:05.24 | [TK]D-Fender | sevard : I don't need another case of food poisoning |
02:05.30 | sevard | i am teh ubar poison |
02:05.33 | sevard | all i drink is vodka |
02:05.39 | sevard | water is for gaywads |
02:05.54 | docelm0 | sevard why dont you drink it? |
02:06.01 | sevard | lololololololololololofag |
02:06.05 | [TK]D-Fender | surfdue : The question is pretty self-explanitor. What kind of device is SIP/200 and where is it located relative to you * server |
02:06.09 | hads | Water is good for hangovers |
02:06.14 | surfdue | oh [TK]D-Fender its a linksys pap2 |
02:06.16 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
02:06.22 | kumamoto | I came across something called DID and was wondering what that is |
02:06.24 | sevard | hads: not me, does me worse. solids do me good. |
02:06.24 | docelm0 | PAP2 does NOT support t38 |
02:06.30 | docelm0 | NSE or reinvite only |
02:06.38 | docelm0 | I just went thru this a couple hours ago for faxing |
02:06.56 | docelm0 | Set to NONE and use ULAW and you *SHOULD* be golden |
02:07.14 | mitcheloc | oooh slashdot got a face lift! |
02:07.20 | kumamoto | is there any voip fax thing I haven't heard of any |
02:07.26 | [TK]D-Fender | mitcheloc : nEW css IS really NICE. |
02:07.47 | sevard | I'm using a Sipura 2002 codec: ulaw only to my * box and a PRI trunk, asterisk complains about not being able to understand the codec but faxing works. |
02:08.09 | surfdue | so what do i do? |
02:08.22 | sevard | mitcheloc: OOOOOOoooooooo |
02:08.31 | sevard | I like it! |
02:08.35 | docelm0 | I just told you goober.. ULAW no fax detection for NSE or Reinvite and your good |
02:08.47 | docelm0 | I just set one of these up with asterisk like 2 hours ago |
02:09.47 | sevard | ka pow bang |
02:10.14 | [TK]D-Fender | surfdue : I asked you a question... TWICE. |
02:10.26 | surfdue | i ansered it |
02:10.27 | [TK]D-Fender | surfdue : Where is that PAP2 relative to *? |
02:10.32 | surfdue | i dont get it? |
02:10.35 | surfdue | relative? |
02:11.01 | [TK]D-Fender | suf : On a direct network switch witha n IP local to *? 12000miles away behind a nat router? |
02:11.08 | [TK]D-Fender | WHERE!? |
02:11.38 | surfdue | what |
02:11.43 | surfdue | im a noob explain plz :(O |
02:11.51 | *** join/#asterisk litage (n=nick@203.220.55.70) |
02:11.54 | sevard | google: wtf is a network |
02:11.54 | surfdue | oh! |
02:12.00 | [TK]D-Fender | surfdue : OHow is that stupid box networked? |
02:12.02 | surfdue | [TK]D-Fender, you mean the servers ip in the network? |
02:12.09 | surfdue | Its directly connected to the net |
02:12.12 | surfdue | no local ip |
02:12.18 | [TK]D-Fender | asd;asfd;asfdasfdkjfasdq7wn89-17n438904325b |
02:12.30 | surfdue | :| |
02:12.49 | surfdue | [TK]D-Fender, http://host41.pastebin.com/771207 |
02:13.11 | [TK]D-Fender | Qwell : I asked for sharks with frigen lasers on their heads!!! |
02:13.23 | *** part/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com) |
02:14.20 | sevard | FRICKEN LASERS |
02:14.56 | [TK]D-Fender | surfdue : Well your ATA is clearly not registered with *. Make sure that [200] has "qualify=yes", "canreinvite=no" |
02:15.00 | *** join/#asterisk Jameno123 (n=james@ddsl-216-68-219-38.fuse.net) |
02:15.03 | Jameno123 | join #mandriva |
02:15.06 | Jameno123 | err |
02:15.09 | surfdue | k |
02:15.18 | Jameno123 | forgot my / :( sorry |
02:15.25 | [TK]D-Fender | surfdue : Make those changes, power down your ATA, tun on "sip debug" in * CLI and look for errors when you power it back up. |
02:15.37 | surfdue | k |
02:15.47 | surfdue | k |
02:15.47 | surfdue | ty |
02:15.58 | sevard | so that raises a question |
02:15.58 | [TK]D-Fender | And pastebin it when in doubt |
02:16.11 | sevard | when you have 400 clients connecting to your * box how do you debug -one- |
02:16.26 | [TK]D-Fender | sevard : "sip debug peer [ip or peer name]" |
02:16.33 | *** join/#asterisk litage (n=nick@203.220.55.70) |
02:16.35 | sevard | \i didn't notice that, sweet |
02:16.46 | [TK]D-Fender | Complimentary trout :) |
02:16.48 | sevard | could you 'supress everything except debug' ? |
02:17.03 | sevard | enjoy some pussy. |
02:17.22 | [TK]D-Fender | sevard : No.... few people trust the small stuff when debugging... |
02:17.28 | surfdue | [TK]D-Fender, this is what I get http://host41.pastebin.com/771215 |
02:17.49 | sevard | [TK]D-Fender: sure, but if you have 200 sim calls going on at once.. you want to look at the client in question |
02:17.55 | [TK]D-Fender | surfdue : And you jsut powered on your ATA for being completely powerless? |
02:18.04 | surfdue | correct |
02:18.05 | surfdue | :| |
02:18.17 | [TK]D-Fender | sevard : Hence your ability to specify an IP or peer... |
02:18.31 | [TK]D-Fender | surfdue : then its not even talking to *. Go set it up right! |
02:18.41 | surfdue | huh? |
02:18.42 | sevard | [TK]D-Fender: when you debug one peer there are still 300 calls going on at once |
02:18.44 | surfdue | i dont think it can |
02:18.48 | sevard | lots of messages to sort through |
02:18.52 | surfdue | I had a feeling there was something blocking * |
02:18.52 | surfdue | ? |
02:19.19 | [TK]D-Fender | surfdue : Goddamnit... Your ATA is not even CONTACTING your Asterisk server! Forget about even having the right user& pass. |
02:19.32 | surfdue | i undertstand that |
02:19.37 | surfdue | it could be aport block |
02:19.38 | surfdue | right? |
02:19.39 | [TK]D-Fender | surfdue : Go look at it and your networking setup. |
02:19.41 | sevard | surfdue: get in paint and diagram exactly how your network is laid out |
02:19.50 | [TK]D-Fender | surfdue : You should know.. its your server... |
02:19.54 | sevard | imageshack it. |
02:20.08 | surfdue | i dont use image shack |
02:20.15 | surfdue | [TK]D-Fender, thanks for your help |
02:20.43 | surfdue | [TK]D-Fender, do you wanna look at my configs? They are clean! |
02:21.04 | [TK]D-Fender | surfdue : Whats the point? * isn't even getting called! Go fix your ATA/networking... |
02:22.54 | *** join/#asterisk adker (n=adker@74-33-195-209.br1.glv.ny.frontiernet.net) |
02:26.46 | ManxPower | I want this: http://www.wired.com/news/technology/0,71087-0.html |
02:27.29 | drray | implants? |
02:27.33 | drray | are you nuts? |
02:29.19 | ManxPower | Having a new sense would be cool, especially one that can detect voltage in wires. |
02:29.49 | [TK]D-Fender | ManxPower : I wanna see the first guy to test it out with 220v :D |
02:30.29 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
02:31.46 | drray | or a MRI |
02:32.26 | *** join/#asterisk synthetiq (n=family@c-71-234-203-69.hsd1.ct.comcast.net) |
02:32.44 | *** part/#asterisk MACscr (i=user@adsl-70-235-7-81.dsl.peoril.sbcglobal.net) |
02:32.45 | synthetiq | how do i generate a 183 ringing message vs the typical 183 ringing? |
02:33.21 | *** join/#asterisk litage (n=nick@203.220.55.70) |
02:35.15 | [TK]D-Fender | dlynes_office : My watch is faster! And its analog. |
02:35.27 | dlynes_office | yeah, no kidding :) |
02:35.47 | dlynes_office | I don't know why slackware enables all this crap by default |
02:35.48 | synthetiq | no one?? |
02:35.52 | drray | shouldn't you compile on another box and copy it over? |
02:36.11 | dlynes_office | drray: i suppose |
02:36.12 | drray | sorry |
02:36.20 | dlynes_office | drray: but i'm just doing proof of concept at this point |
02:36.25 | Dr-Linux | still not understand wtf is 183 ringing? |
02:36.30 | drray | :) |
02:36.30 | [TK]D-Fender | dlynes_office : enables what? |
02:36.31 | dlynes_office | drray: that would require too much effort at this point |
02:36.34 | file | 183 Session Progress |
02:36.47 | dlynes_office | [TK]D-Fender: all the drivers and modules and all that other funking crap |
02:36.52 | file | ringing or other progress sent as an audio stream versus as out of band |
02:36.59 | synthetiq | yes file...how do i go about generating one of those |
02:36.59 | [TK]D-Fender | dlynes_office : Helps things "just work".... |
02:37.23 | drray | it did not help my linux powered time machine "just work" |
02:37.24 | file | there's a Progress dialplan application |
02:37.30 | Dr-Linux | what's good link to read about 183 ringing and inband :S |
02:38.23 | synthetiq | i always thought it was done with putting an "r" in the dial string |
02:38.43 | file | that forces ringing to be sent back regardless of what you get from the dialed party |
02:39.52 | russellb | file: go to bed |
02:40.04 | file | russellb: ...no! you go to bed. |
02:40.05 | Qwell | file! russellb! |
02:40.17 | file | omg hi Qwell |
02:40.22 | file | or should I say UberNub |
02:40.32 | Qwell | oh em gee |
02:40.41 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
02:40.50 | russellb | file: ok, I WILL |
02:40.55 | file | russellb: FINE THEN |
02:41.01 | russellb | Qwell: g'night! |
02:41.13 | dlynes_office | drray: it didn't help it just grind away at the hard drive? |
02:41.23 | Qwell | be nice |
02:41.30 | drray | the Nub is the best part of the |
02:41.40 | h0 | can some one help me with a question please |
02:41.47 | Qwell | ~ask |
02:41.51 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a quesiton first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
02:41.51 | h0 | I am looking into to set up asterisk but before I purchase any hardware I want to verify something that I cant seem to find the answer to anywhere else. I have optimum voice (cablevison) who has provided a Motorola voip cable modem. I am just wondering if it maters if I put a SPA-3000 behind voip or a regular phone line. |
02:42.40 | Qwell | h0: analog is analog is analog |
02:43.00 | drray | h0: treat it like running water, if you get a dialtone you are done |
02:43.17 | synthetiq | file what should i search for on voip info |
02:43.22 | *** join/#asterisk litage (n=nick@203.220.55.70) |
02:43.26 | file | synthetiq: about what? |
02:43.32 | h0 | ok thanx alot thats what i thought i just did not want to buy anything and then find out later there is a problem |
02:43.43 | drray | I'd buy stuff late |
02:44.13 | synthetiq | 183 progress |
02:44.20 | synthetiq | the app |
02:44.33 | file | it's a really simple app... |
02:44.47 | file | you run Progress, and then all further indications will be inband... sent as 183 Session Progress |
02:47.52 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
02:48.59 | *** part/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
02:50.19 | synthetiq | al you do is exten => exten,priority,Progress() ? |
02:50.28 | synthetiq | sounds to simple |
02:59.13 | synthetiq | yup it is |
02:59.15 | synthetiq | thanks file |
02:59.26 | file | :D |
03:09.13 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
03:14.54 | *** join/#asterisk thermf (i=fadaasfa@adsl-68-73-6-126.dsl.sfldmi.ameritech.net) |
03:15.28 | thermf | anyone happen to have spandsp-0.0.3pre4 (w. t38bits)? |
03:16.36 | dlynes_office | [TK]D-Fender: so far so good |
03:16.48 | dlynes_office | [TK]D-Fender: doing the final recompile on asterisk now |
03:23.07 | docelm0 | Hay how does one restart the logger in 1.2.9? |
03:25.20 | [TK]D-Fender | dlynes_office : Only now? :) |
03:29.57 | dlynes_office | [TK]D-Fender: everything's up and running the same way now as you had it |
03:30.06 | dlynes_office | [TK]D-Fender: does sangoma not use ztcfg? |
03:30.31 | [TK]D-Fender | it does |
03:30.47 | dlynes_office | ah...i always get invalid argument |
03:31.00 | dlynes_office | does it only count a dual fxo module as 1 channel then? |
03:36.29 | file | and the crowd goes quiet |
03:39.11 | dlynes_office | prolly cause it's so late at night |
03:39.14 | dlynes_office | and it's a friday |
03:39.18 | dlynes_office | everyone's out drinking |
03:39.20 | dlynes_office | like you should be |
03:39.27 | dlynes_office | you're not a true maritimer, you imposter! |
03:39.32 | file | too tired... |
03:39.33 | znoG | have they estimated when Asterisk 1.4 will be coming out? |
03:39.47 | file | oh it'll be out when it's out |
03:40.13 | file | I'd say sometime in July though |
03:45.12 | *** join/#asterisk Eggplant (i=No@dsl-216-155-213-242.cascadeaccess.com) |
03:46.43 | znoG | file: yeah, i realize it'll be out when it's out (glad you clarified that!) :-) |
03:47.00 | znoG | but i was wondering if they planned on releasing it next week, next month, next year, and so on .. |
03:47.09 | file | we're having betas first |
03:47.17 | file | so it won't just be released :) |
03:47.27 | znoG | yeah, i figured |
03:47.50 | znoG | 1.4 will be, more or less, what's in SVN right? |
03:48.01 | file | it'll be trunk |
03:48.04 | Qwell | znoG: at some point in time |
03:48.07 | dlynes_office | [TK]D-Fender: IT'S WORKING!!!!!!!!!!!!! |
03:48.18 | znoG | file: SVN == trunk? |
03:48.28 | file | SVN is a revision control system. |
03:48.30 | Qwell | trunk vs a branch |
03:48.33 | file | :D |
03:48.38 | Qwell | ~svn |
03:48.45 | Qwell | stupid bot |
03:48.47 | bkw_ | docelm0, you do logger reload |
03:48.53 | Qwell | bkw_: nub |
03:49.04 | bkw_ | what? |
03:49.09 | Qwell | y0 |
03:49.12 | bkw_ | sabi |
03:49.23 | znoG | file: yes i know what SVN is, it's like saying "upload to the FTP".. FTP is a protocol, but it's generally used when referring to its contents |
03:49.24 | Qwell | the wa variety is good on sushi |
03:49.29 | Qwell | (or sabi) |
03:49.31 | Qwell | of* |
03:49.37 | Qwell | nm |
03:49.44 | file | znoG: well SVN can refer to a few things, as everything is in SVN - and you can pull 1.2 from SVN |
03:49.52 | file | so trunk is more proper |
03:50.03 | znoG | file: valid point, trunk would be where all the development is going on ... right? |
03:50.05 | file | where all the bleeding edge... development... stuff |
03:50.10 | znoG | ok cool |
03:50.16 | znoG | that answers it |
03:52.33 | *** join/#asterisk bmg505 (n=leon@196.209.47.183) |
03:56.21 | synthetiq | what would cause a 503 error |
03:56.36 | bkw_ | isn't that service unavali? |
03:56.39 | synthetiq | yes |
03:58.06 | docelm0 | 5 minutes until new pricing hits.. :) |
03:58.19 | thermf | of what? |
04:01.10 | docelm0 | US48/CANADA/Toll Free |
04:02.11 | thermf | what carrier? |
04:05.10 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
04:05.15 | mdiehl | Hi all. |
04:06.22 | mdiehl | Do I expect to be able to change the callerid number of an incoming call without hosing the callerid name? |
04:07.12 | mdiehl | I'm trying to strip off a leading "+" if it is present. When I do, the name doesn't display on my cid unit..... |
04:07.13 | grabowski | mdiehl: You could copy the original callerid to another variable before you change it |
04:07.22 | grabowski | mdiehl: oh |
04:07.34 | mdiehl | I've been fighting this for two days now. |
04:08.04 | mdiehl | My provider adds a + to the cid number. Some of my other voip clients do not..... |
04:08.14 | mdiehl | So I can't simply strip off the first char. |
04:08.45 | grabowski | mdiehl: You should be able to, how I am not sure |
04:09.07 | mdiehl | exten => s, 3, gotoif(${REGEX("^\+",${CALLERID(number)})},4,5) |
04:09.07 | mdiehl | exten => s, 4, set(CALLERID(number)=${CALLERID(number):1}) |
04:09.33 | mdiehl | When this code runs, I get the phone number displayed twice on my phone, no name. |
04:12.15 | grabowski | does CALLERID(number) work? I thought it was CALLERID(num) |
04:13.00 | mdiehl | It seems to. |
04:13.46 | grabowski | mdiehl: http://www.voip-info.org/wiki/view/Asterisk+variables |
04:14.02 | grabowski | mdiehl: read the part about ${CALLERIDNUM} |
04:15.08 | mdiehl | I thought that was deprecated. |
04:15.35 | grabowski | meshuga: yes but read the note about it |
04:17.12 | grabowski | mdiehl: sorry that was ment for you ^ |
04:17.14 | mdiehl | Says, I should be using CALLERID(num)..... |
04:17.17 | mdiehl | I got it. |
04:17.20 | mdiehl | Thanx. |
04:17.33 | mdiehl | Strange. Wonder why (number) is working.......? <grin> |
04:18.13 | grabowski | mdiehl: I am also wondering about the "Note that this is not necessarily numeric as the name would indicate" |
04:18.34 | grabowski | mdiehl: Are you sure that CALLERID(num) only contains the number? |
04:18.35 | mdiehl | That's in the case where the caller isn't from the pstn. |
04:18.39 | mdiehl | Yes. |
04:18.44 | grabowski | mdiehl: with the + that you would like to remove of course |
04:18.51 | mdiehl | Right. |
04:19.12 | mdiehl | For example "john doe <sip@home>" |
04:22.37 | *** join/#asterisk sengland (n=info@dsl081-072-084.sfo1.dsl.speakeasy.net) |
04:23.43 | docelm0 | Plainvoip |
04:23.52 | docelm0 | I just sent an email out on the -biz list |
04:24.38 | grabowski | docelm0: Yea I noticed its "till the end of the month" |
04:25.01 | grabowski | mdiehl: sorry, im out of ideas and can't find anything on google |
04:25.21 | sengland | Anyone know of a way to connect active channel to a confrence room using channelspy with the AMI? |
04:25.32 | grabowski | mdiehl: is that regex gotoif syntax correct? |
04:26.08 | mdiehl | Not sure. Seems to be working, but I've not tested it very hard. |
04:26.51 | grabowski | mdiehl: show function regex shows that its a seperate cmd from the dial plan.. |
04:27.51 | grabowski | mdiehl: I have not needed to use regex in any of my dialplans yet so I'm sorry I have no idea |
04:28.45 | mdiehl | What do you mean by a separate cmd from the dialplan? |
04:28.52 | grabowski | mdiehl: show function REGEX |
04:29.28 | grabowski | mdiehl: I mean can you use it within a gotoif or does it need to be a seperate line / priority before the gotoif. |
04:29.57 | mdiehl | It's a function, so I'm assuming I can use it in an expression...... |
04:30.10 | grabowski | mdiehl: I guess |
04:31.22 | compu73rg33k | what's the coolest thing you guys have done with VOIP ? |
04:31.33 | compu73rg33k | in * |
04:32.10 | grabowski | compu73rg33k: Not me but someone http://uc.org/read/Zasterisk |
04:33.16 | mdiehl | I've started my own phone company..... just a few friends, but hey.... |
04:33.25 | compu73rg33k | yeah that's cool |
04:33.33 | compu73rg33k | nice grabowski. I know someone who made http://asterwake.com heh |
04:33.44 | compu73rg33k | bah bad link hold up |
04:34.05 | mdiehl | I went from giving Quest $40/month to spending $15/month. That's WAY cool! |
04:34.58 | compu73rg33k | oh damn he gave up the site :( basically it was a wake up call program in asterisk it was cool |
04:35.08 | compu73rg33k | yeah nice mdiehl |
04:35.40 | mdiehl | Thinking about doing that myself. Using call files created by a perl script run via cron..... |
04:35.51 | compu73rg33k | haha I think tha'ts exactly waht he did |
04:35.56 | docelm0 | I started a real phone company pushing millions of minutes |
04:35.57 | docelm0 | :) |
04:36.06 | compu73rg33k | http://biimsoft.com/pages/projects/code/asterwake/ for the code mdiehl if you want |
04:36.16 | mdiehl | Thanx. |
04:38.45 | *** join/#asterisk scrubadub (i=seeess@dwai.rit.edu) |
04:43.28 | asterisk-dud | any good canadian voip carriers anyone knows of? |
04:44.17 | dlynes_office | asterisk-dud: you looking for retail or wholesale? |
04:44.29 | asterisk-dud | retail |
04:44.55 | dlynes_office | asterisk-dud: you could try looking at www.calltermination.com; they have a huge list of terminators both wholesale and retail |
04:45.01 | asterisk-dud | actually, what's the diff, i work for a fairly larg company |
04:45.11 | dlynes_office | asterisk-dud: define large? |
04:45.24 | Qwell | in minutes per month |
04:45.25 | dlynes_office | asterisk-dud: like how many minutes of long distance per month? |
04:45.26 | asterisk-dud | 75 extentions |
04:45.50 | asterisk-dud | mts lines are costing well over a grand |
04:46.07 | dlynes_office | how many minutes per month though? |
04:46.33 | asterisk-dud | 4000 |
04:46.36 | dlynes_office | and also where in Canada are you located? |
04:46.37 | asterisk-dud | ballpark |
04:46.41 | asterisk-dud | manitoba |
04:47.06 | dlynes_office | asterisk-dud: well, you're probably out of luck for "good" voip carriers in Canada, then simply because of where you are |
04:47.26 | dlynes_office | you're not going to get super great ping times, but you'll get ok ping times |
04:47.34 | [TK]D-Fender | yup, all the good ones are mostly eastern Canada |
04:47.42 | dlynes_office | [TK]D-Fender: and western canada :) |
04:47.44 | [TK]D-Fender | TO/MTL |
04:47.49 | dlynes_office | Vancouver |
04:47.49 | asterisk-dud | well i have vonage and it sucks balls |
04:47.57 | asterisk-dud | is there anyway to improve it/ |
04:47.58 | dlynes_office | vonage sucks a lot of stuff |
04:48.05 | dlynes_office | including investors' money :p |
04:48.06 | [TK]D-Fender | dlynes_office : I've had trouble finding ones to incoming DID.... |
04:48.23 | scrubadub | i have an icable smta icsg101c, couldnt find much using it with asterisk. I'm wondering if it is possible to use it like a sipura device, allowing PSTN inbound calls routed to an asterisk server, and outbound calls from an asterisk server? http://www.icablesystem.com/english/product/sub_ics_g101.html |
04:48.31 | dlynes_office | [TK]D-Fender: well, after I finish this godforsaken billing system, we can sell you vancouver dids |
04:48.43 | Jason99 | I'm just wondering if there is a way to call up a context or something after the call hangs up? |
04:48.44 | scrubadub | before i've only used it as a terminating PSTN/voip box, so i'm unsure if i can use it with asterisk |
04:49.28 | asterisk-dud | what is a pri line? |
04:49.56 | [TK]D-Fender | scrubadub : No, thats for plugging PHONES into, not LINES. |
04:50.17 | [TK]D-Fender | Jason99 : look at the "h" exten. |
04:50.29 | dlynes_office | ~wiki pri |
04:50.37 | dlynes_office | erm |
04:50.39 | scrubadub | right ok what's the major difference between my smta and the sipura box, like what should i be googling for |
04:50.41 | Jason99 | [TK]D-Fender: Thanks |
04:50.46 | dlynes_office | ~wiki primary rate interface |
04:51.12 | [TK]D-Fender | scrubadub : Sipura sells various products, some for plugging PHONES into, and ones fro pluuging LINES into. |
04:51.17 | dlynes_office | [TK]D-Fender: btw |
04:51.22 | sengland | [TK]D-Fender you tried voxbone for incoming in Canada? |
04:51.24 | dlynes_office | [TK]D-Fender: everything's up and running just ducky now |
04:51.52 | dlynes_office | [TK]D-Fender: that sangoma tech that i got promoted to is much better at troubleshooting than the other one |
04:51.58 | [TK]D-Fender | scrubadub : You you looking to use that to plug a PHONE into to talk to *? If so, fine. If you want to take it and try to take an analog phone LINE into *, you're out of luck. |
04:52.10 | [TK]D-Fender | dlynes_office : 100% now? |
04:52.18 | dlynes_office | [TK]D-Fender: yeah, on 2.4, anyways |
04:52.19 | scrubadub | yeah i'm looking for something sitting between my asterisk server and PSTN/Voip i guess "plugging lines into". are there keywords i should google for that would bring up other vendors / products |
04:52.20 | Jason99 | [TK]D-Fender: Will that take effect after the call was connected for a while and then hung up? |
04:52.36 | dlynes_office | [TK]D-Fender: now i need to throw a monkey wrench into it and make it work on 2.6 :) |
04:53.04 | [TK]D-Fender | dlynes_office : I've run A200's on CentOS 2.6 without issue... I don;'t think its a kernel issue so much as a lack of headers to match your custome kernels... |
04:53.21 | Jason99 | nm.. I answered myself |
04:53.31 | [TK]D-Fender | Jason99 : Yes, get reading... |
04:53.38 | dlynes_office | [TK]D-Fender: iirc, make linux, make zaptel, make libpri, Config install, make zaptel, make libpri, make asterisk, right? |
04:53.52 | scrubadub | although [TK]D-Fender it does have both a fxo and fxs port which was why i was wondering |
04:53.55 | Jason99 | [TK]D-Fender: Thanks thats exactly what I was looking for |
04:54.20 | dlynes_office | [TK]D-Fender: couldn't be a lack of headers...I installed the kernel from source, not from kernel-dev/kernel-bin |
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04:55.36 | [TK]D-Fender | scrubadub : Appears I read it a bit wrong... but it may only be a failover outbound port... unsure if it will take calls IN... |
04:56.02 | [TK]D-Fender | dlynes_home : Maybe its a question of locations, or maybe a kernel dependency of some kind... |
04:56.59 | scrubadub | ok, basically i'm looking for hardware for my first asterisk setup, one PSTN line, probably just use skypeout for voip connection, do you have any models you recommend. Cant find a whole lot besides http://www.voip-info.org/wiki/view/Asterisk+setup+home |
04:57.29 | scrubadub | trying to decide if i should go pci card or external box |
04:57.53 | [TK]D-Fender | scrubadub : Linksys SPA-3102 <- |
04:57.56 | scrubadub | budget is around 150 |
04:58.00 | scrubadub | ok thanks i'll check that out |
04:58.04 | [TK]D-Fender | Thats $90 |
04:58.22 | [TK]D-Fender | offers 1 FXO, 1 FXS, each operates independantly |
04:58.50 | dlynes_office | [TK]D-Fender: i'm guessing maybe a botched install of sangoma or something |
04:59.06 | dlynes_office | [TK]D-Fender: when i retry it, i'll just recreate my kernel source directory again |
04:59.20 | dlynes_office | [TK]D-Fender: and then reinstall sangoma according to how I did it on 2.4 |
04:59.45 | [TK]D-Fender | dlynes_office : IIAB..... |
05:00.09 | dlynes_office | ~wiki iiab |
05:00.24 | dlynes_office | iiab? |
05:00.27 | [TK]D-Fender | If It Ain't Broke.... |
05:00.31 | dlynes_office | ah |
05:00.34 | [TK]D-Fender | GLUTTON... |
05:00.34 | dlynes_office | yeah...pretty much |
05:01.21 | dlynes_office | anyways...on that note, i'm heading home for the night |
05:01.40 | [TK]D-Fender | Ok, take care. |
05:01.45 | [TK]D-Fender | I'm off as well... |
05:01.47 | [TK]D-Fender | checkout time... |
05:01.58 | dlynes_office | thanks again for the help earlier |
05:02.21 | dlynes_office | got another a101 on the way, too |
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05:03.36 | [TK]D-Fender | cool, look forward to it bombing out on you too ;) |
05:04.02 | [TK]D-Fender | (j/k) |
05:04.17 | dlynes_office | lol |
05:04.34 | dlynes_office | you're just full of sugar and spice and everything nice, aren't you? |
05:04.54 | [TK]D-Fender | Ok, and you should try it out ont hat C#, and while you're at it, lets just bypass G711 and go right to transcoding to G729, k? |
05:05.09 | dlynes_office | ? |
05:05.13 | [TK]D-Fender | C3 |
05:05.21 | dlynes_office | heh |
05:05.43 | dlynes_office | I do transcoding on those machines |
05:05.47 | [TK]D-Fender | Choke that poor bastard till it smurfs out on you... |
05:05.49 | dlynes_office | but not in g729 |
05:06.00 | dlynes_office | I've tried in ilbc though :P |
05:06.10 | dlynes_office | and gsm |
05:06.29 | [TK]D-Fender | gsm isn't so bad. ILBC is. |
05:06.34 | dlynes_office | anyways..i'm outta here now, though |
05:08.19 | [TK]D-Fender | *yawn* ditto |
05:10.41 | Jason99 | Ok, using the contexts I know when the call is dialed and when the call is hung up, but is there a way to know when the call is answered after it's dialed? |
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05:39.49 | rushowr | hey quick question, should be fairly easy...Anyone know for sure what version the Filter() function is available in? |
05:41.20 | Qwell | rushowr: trunk? |
05:41.29 | Qwell | bed |
05:41.31 | rushowr | so it's not available in 1.2.8 |
05:41.35 | Qwell | dunno |
05:41.36 | rushowr | bed? |
05:41.48 | Qwell | bed, as in...the place I sleep |
05:41.51 | rushowr | ah gotcha |
05:41.52 | rushowr | sorry |
05:41.53 | rushowr | thanks mate |
05:42.11 | Qwell | and, umm.. |
05:42.18 | Qwell | Why are you still using 1.2.8? |
05:42.25 | Qwell | see channel topic |
05:42.27 | rushowr | client's machine |
05:42.32 | rushowr | no IAX |
05:42.45 | rushowr | sip and zap only, I've recommended to them though |
05:42.50 | rushowr | thanks again, will hit them again |
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05:58.15 | stephane_ | jour |
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06:16.58 | dlynes_home | stephane_: Est-ce qu'un triggere? |
06:17.50 | stephane_ | triggere ? |
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06:36.21 | MACscr | since I have a dynamic ip address at my home office, should i not even think about running an asterisk server here? |
06:38.00 | Gabriel25 | MACscr you can fix that |
06:38.12 | MACscr | just run an updater? |
06:38.16 | Gabriel25 | dyndns |
06:38.34 | MACscr | ok, cool |
06:38.48 | MACscr | wasnt sure if it would work ok without using ip addresses directly |
06:39.12 | Gabriel25 | i have like this and is working with no problem |
06:39.23 | MACscr | well, i just used TrixBox and installed everything fine |
06:39.42 | MACscr | havent decided on a voip provider yet |
06:39.48 | MACscr | i wish i could use vonage |
06:40.46 | Gabriel25 | don`t use the word vonage noone like that here |
06:40.47 | Gabriel25 | :)) |
06:41.00 | Gabriel25 | I was almost banet for this word |
06:41.22 | MACscr | thats retarded. They have decent service, the coverage is one of the largest, and they are cheap |
06:41.28 | MACscr | i have 3 lines with them |
06:41.51 | Gabriel25 | https://www.teliax.com/newaccount/ |
06:41.54 | Gabriel25 | some prices |
06:42.03 | MACscr | yep, im looking at them |
06:42.18 | MACscr | i hate going with any company though that doesnt have 24/7 phone support |
06:42.45 | Gabriel25 | you are right about this one :) |
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06:43.07 | MACscr | ive heard good things, but a lot of voip providers run their sites like they are a 1 person company |
06:43.19 | MACscr | and built it with frontpage |
06:43.41 | Gabriel25 | :)) |
06:43.47 | MACscr | i dont know. Im a web host and designer, so im pretty critical |
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06:43.58 | VoIPMasta | Hi there |
06:43.58 | Gabriel25 | I didn`t try |
06:44.07 | Gabriel25 | wow nice |
06:44.18 | MACscr | i worked in telecom for 2 years as well, so im pretty familiar with how things work |
06:44.27 | MACscr | not to much voip experience though |
06:44.33 | Gabriel25 | where you are from ? |
06:44.43 | MACscr | Peoria, IL, USA |
06:44.50 | VoIPMasta | I have a question for those experienced with dialplans... let's say that I have 2 devices, one is IAX and the other one is SIP. Now I want to receive calls to my extension regardless of wether device I'm using to connect... any ideas? |
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06:45.39 | Gabriel25 | I know some people in verizon bussines ex MCI in New York |
06:45.40 | Gabriel25 | :)) |
06:46.30 | MACscr | i worked with SBC (AT&T) mostly |
06:47.03 | MACscr | whats harder about finding a good provider is that i want to find one that i can port my numbers too |
06:47.18 | MACscr | teliax is one that can, but there arent many that i have found that can |
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07:13.53 | x86 | MACscr: no way, I'm from peoria as well ;) |
07:14.04 | MACscr | lol, really? |
07:14.05 | x86 | MACscr: east peoria anyway |
07:14.09 | x86 | yeah man :) |
07:14.10 | MACscr | oh, im sorry |
07:14.11 | MACscr | jk |
07:14.13 | x86 | hehe |
07:14.14 | MACscr | =P |
07:14.17 | x86 | the good side of the river ;) |
07:14.20 | MACscr | im originally from washington |
07:14.23 | x86 | ah ok |
07:14.31 | x86 | right down the road from me ;) |
07:14.38 | MACscr | so you want to configure my asterisk system for me so i can go to bed? |
07:14.40 | MACscr | =P |
07:14.40 | x86 | you work for cat? |
07:14.44 | MACscr | lol, no |
07:14.50 | x86 | hehe |
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07:15.09 | MACscr | im an IT director for a brokerage firm |
07:15.46 | nextime | 'morning |
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09:01.02 | Tmob | hey guys... i have a sip capable phone.. i'm not too familiar with phone tech.. so curios if anoyne here knows where i can buy some accont so i can make calls using sip |
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09:01.24 | Tmob | saves me a lot of $$ onphone bills if i cna do that .. esp when i'm travelling out of country :) |
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09:08.01 | robin_sz | there are thousands of SIP providers |
09:09.31 | Tmob | robin_sz, i am tryig to search for ones which let me use my phone to connect to them |
09:09.40 | Tmob | robin_sz, i found a few but they have their own applications |
09:09.46 | Tmob | robin_sz, can you recommend any? |
09:09.57 | {zombie} | look on voip-info.org |
09:10.07 | {zombie} | there are several lists of VoIP providers |
09:10.15 | robin_sz | what he said, there a list on voip-info.org |
09:10.25 | Tmob | thanks guys :) |
09:10.28 | Tmob | i'll check it out |
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09:12.23 | robin_sz | probably wont save you a lot though .. as you travel around the world, finding an network conneciton is ususally more expensive than just using a mobile to make calls |
09:12.41 | robin_sz | but, in theory, you should save |
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09:17.32 | Tmob | robin_sz, hehe yea, but you will be surprised how many calling cards i use up when in europe |
09:17.53 | Tmob | robin_sz, i have a theory their clocks run faster than ours ;) |
09:19.08 | Tmob | oops.. yes i did fall for that one :( |
09:19.09 | Tmob | sorry |
09:19.12 | robin_sz | hehe |
09:19.21 | Tmob | robin_sz, you work for emc? |
09:19.23 | robin_sz | im in Geneva :) |
09:19.30 | Tmob | ah cool.. |
09:19.44 | robin_sz | emc? the storage people? |
09:20.02 | robin_sz | emc the linux based machien tool controller? |
09:20.11 | Tmob | robin_sz, heh thought it was storage.. nm |
09:20.17 | robin_sz | either way, no |
09:20.32 | Tmob | robin_sz, i'll be in zurich in 3 weeks :) |
09:20.37 | robin_sz | nice, |
09:20.45 | Tmob | but far away from geneva |
09:21.02 | robin_sz | not so far, its a very small country |
09:21.06 | Tmob | usually nothing to do in geneva i think.. except diplomats doing useless meetings.. hehe |
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09:21.11 | robin_sz | in 3 weeks I will be back in the UK |
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09:53.52 | Eric-xx | how to upgrade asterisk? |
10:00.24 | Tili | Eric-xx: download new tarball u want. delete everything in /usr/lib/asterisk/modules. Extract new tarball somewhere. make and then make install |
10:30.56 | x86 | Tili: what about /var/lib/asterisk ? |
10:31.04 | x86 | or /usr/bin/asterisk? heh |
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10:31.22 | Tili | x86: dont need to do anything in there |
10:31.41 | Tili | x86: /var/lib/asterisk holds the agi and sounds etc. these are not such integrated parts of * |
10:31.54 | Tili | x86: /usr/bin/asterisk will get overridden when you do make install |
10:32.28 | Tili | the only thing about modules directory is that modules are loaded from that directory. |
10:32.31 | x86 | so will /usr/lib/asterisk/modules ;) |
10:32.38 | Tili | x86: no |
10:32.42 | Tili | modules dir wont |
10:32.53 | x86 | no? read the make file and try again :) |
10:32.57 | Tili | only those modules that were in old * and are in new * will get overriden |
10:33.04 | Tili | x86: I know what I am doing |
10:33.15 | x86 | read the make file, because you're wrong |
10:33.21 | x86 | ah right |
10:33.34 | Tili | but if some module is not supported anymore, it will still be loaded in * |
10:33.36 | x86 | only same module names will be overridden |
10:33.41 | Tili | yeah |
10:33.57 | Tili | this issue arises mostly when u downgrade. but can happen in upgrade also |
10:34.01 | x86 | well, you didnt tell him to remove /etc/asterisk/*, so the old module will be in modules.conf still ;) |
10:34.05 | Tili | unless make file removes all modules in modules dir |
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10:34.38 | Tili | x86: modules.conf is no problem usually. it tells not to load a module |
10:34.47 | Tili | and removing /etc/asterisk/* is dangerous |
10:34.52 | x86 | very |
10:34.52 | Tili | as you will lose all ur earlier settings |
10:35.00 | x86 | unless you use realtime ;) |
10:35.07 | Tili | x86: true |
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11:24.19 | jhiver | hi all |
11:24.39 | jhiver | is there a way to detect if a gateway has timed out from the dialplan? |
11:25.27 | jhiver | like you do Dial(SIP/number@gateway) <------- if a call couldn't get through because the gateway has timed out, i want to do something about it |
11:26.16 | jhiver | the idea would be to suspend gateway which time out for 60 minutes, and then suspend them permanently and send an email alert on the second time |
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11:35.35 | jhiver | I guess there isn't heh :) |
11:35.42 | jhiver | or maybe this channel is dead :) |
11:35.53 | jhiver | vi rocks, emacs sucks! should sort it out |
11:36.09 | robin_sz | shh, sleeping! |
11:37.55 | drray | three chers for sed! |
11:38.04 | jhiver | lol |
11:38.09 | robin_sz | bah, hex ed |
11:38.12 | robin_sz | hexedit |
11:38.39 | jhiver | see? vi vs emacs always works to wake up a channel :) |
11:39.22 | jhiver | let's do something more outrageous |
11:39.28 | jhiver | OpenPBX rocks! |
11:40.01 | jhiver | hopefully that'll wake up some hardcore digium supporters and then somebody will know about my poor issues :) |
11:40.44 | jhiver | oh no! they go away now :) |
11:41.06 | robin_sz | jhiver, are you familiar with IRC commands? |
11:41.21 | jhiver | not very much no |
11:41.30 | robin_sz | wait, its ok ... I found it .. /ignore |
11:41.35 | jhiver | lol |
11:41.43 | jhiver | allright I'll shut up |
11:41.54 | jhiver | just faffing about really :) |
11:42.00 | robin_sz | no kidding |
11:42.29 | jhiver | well since everybody's asleep a bit of sillyness doesn't hurt does it |
11:43.38 | drray | maybe if you spent your time reading the wiki |
11:43.50 | jhiver | I do |
11:44.03 | jhiver | I've been on Dial and RetryDial pages but haven't seen anything so far |
11:44.36 | drray | you know about the t extension? |
11:45.03 | jhiver | but that's not timing out because the gateway doesn't answer is it |
11:46.17 | drray | ok, so why does the timeout variable in the dial command not work for you? |
11:47.11 | jhiver | isn't it timing out if a call hasn't been answered under 't' seconds? |
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11:47.29 | drray | yes |
11:47.40 | jhiver | that's not what I want to do |
11:48.09 | jhiver | I want to detect if the call hasn't gone through because the GATEWAY has timed out (i.e. it's down for instance) |
11:48.30 | jhiver | so I can suspend it temporarily |
11:49.01 | jhiver | now if somebody lets a phone ring for 60 or so seconds, it's fine by me |
11:49.01 | drray | I don't suppose least cost routing would help |
11:49.29 | jhiver | but if the gateway doesn't answer to INVITE requests under 5 seconds, I want it out |
11:49.34 | jhiver | does it seem reasonable? |
11:50.55 | jhiver | drray, how would it help? |
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12:33.12 | synthetiq | anyone know what would cause a 503 (service unavail) error? |
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12:54.45 | copland | Hello, if a call is coming in on a trunkA that says gets fowarded by follow me to a pstn number that it always uses TrunkB and not any of my other trunks |
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13:04.27 | queuetue | How would I build a tool that allows me to access CID info on my desktop machine? |
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13:13.14 | x86 | queuetue: most likely one such tool already exists |
13:13.20 | x86 | queuetue: google for asterisk screen pop |
13:13.29 | x86 | queuetue: i got it setup on my Mac very easy |
13:13.53 | queuetue | x86, Good, mac integration is what I'm looking for - will it pause iTunes? :) |
13:14.38 | x86 | lol not that i know of |
13:14.47 | x86 | it has Growl integration which is quite nice |
13:15.10 | queuetue | x86, Do you know the exact name? asterisk screen pop doesn't seem to be doing it... |
13:16.59 | x86 | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Notify |
13:17.07 | x86 | you'll need the app_notify.so module |
13:17.24 | queuetue | x86, Ok, that's what I was already investigating. :) Thanks. |
13:17.28 | x86 | http://www.voip-info.org/wiki/view/Asterisk+call+notification |
13:17.47 | x86 | this one shows desktop apps that tie into the asterisk management API and give screen pops |
13:18.34 | x86 | wait no |
13:18.36 | x86 | go here: |
13:18.42 | x86 | http://www.mezzo.net/asterisk/app_notify.html |
13:18.48 | x86 | there is a section on MacOS X client |
13:20.02 | x86 | Growl + Address Book integration |
13:20.06 | x86 | cant be beat :) |
13:20.15 | x86 | http://www.mezzo.net/asterisk/AsteriskNotifyClient-1.0rc5.dmg |
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13:24.59 | x86 | the dialing from Address Book seems to be broken though |
13:25.14 | x86 | but not a big deal for me, as I have custom software I use for that anyway |
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13:34.01 | asterisk-dud | for some reason i can't get fop to work |
13:35.16 | asterisk-dud | index.php doesn't kick in |
13:35.30 | asterisk-dud | i can only see the files in the directory |
13:35.32 | x86 | sounds like your apache configuration skills are weak ;) |
13:36.03 | x86 | ask about that in #apache |
13:36.05 | x86 | not here :P |
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14:02.54 | jhiver | hi again all |
14:03.04 | jhiver | is there a way to detect if a gateway has timed out from the dialplan? |
14:03.10 | jhiver | like you do Dial(SIP/number@gateway) <------- if a call couldn't get through because the gateway has timed out, i want to do something about it |
14:03.23 | jhiver | the idea would be to suspend gateway which time out for 60 minutes, and then suspend them permanently and send an email alert on the second time |
14:03.28 | jhiver | any ideas? |
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14:28.25 | littleball | hello, my domain registrar doesn't support DNS SRV record. who can recommend one domain registra? |
14:29.12 | lunk | godaddy |
14:29.20 | lunk | big tit commercials for the win |
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14:36.22 | eKo1 | Is it possible to configure two spans on the TE410 to share a D-channel? |
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14:38.21 | wunderkin | eKo1, that is what nfas is for |
14:39.48 | eKo1 | Hmm...I'm going to have to play around with this stuff. |
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14:40.26 | robin_sz | littleball, there are no registrars on the planet that "support SRV record", nor will there ever be. |
14:41.05 | robin_sz | littleball, SRV records come out of your DNS, registrars simply add NS records to the root servers |
14:41.44 | robin_sz | two entirely different things |
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14:51.11 | littleball | robin_sz, http://www.voip-info.org/wiki-DNS+SRV |
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14:58.12 | eKo1 | Is there a way to take out a span from operation within the * CLI? |
15:00.22 | tzafrir_laptop | eKo1, what spam? |
15:00.37 | wunderkin | no |
15:01.10 | WiredX | hey everyone.. |
15:01.55 | tzafrir_laptop | eKo1, is that span actually doing something useful? |
15:02.44 | wunderkin | yes.. some butt fucker picked up my email from an asterisk mailing list and is now spamming other users on my domain now.. ugh |
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15:05.12 | eKo1 | tzafrir_laptop: nope, it is complaing about no d-channel |
15:05.15 | eKo1 | I want to take it down. |
15:05.33 | tzafrir_laptop | ztcfg -s NUM ? |
15:05.50 | tzafrir_laptop | Or just remove that span from your zapata.conf |
15:06.40 | tzafrir_laptop | WiredX, you wanted to ask something? |
15:06.47 | eKo1 | I did remove it from zapata.conf. That didn't work. Doesn't ztcfg -s shutdown all the spans? |
15:07.04 | tzafrir_laptop | eKo1, did you restart asterisk? |
15:07.45 | tzafrir_laptop | Without restarting asterisk: you can "destroy" zap channels, but I'm not sure about spans |
15:08.49 | eKo1 | tzafrir_laptop: I don't want to restart asterisk as I have calls going through the other spans. |
15:09.09 | eKo1 | I guess it can't be done. |
15:09.22 | tzafrir_laptop | You can destro the channells of the irrelevant span |
15:09.41 | tzafrir_laptop | Works on analog. Never tried it with ISDN |
15:09.53 | tzafrir_laptop | zap destroy channel NNN |
15:10.06 | WiredX | tzafrir_laptop: no :) just saying hi and making myself feel welcome |
15:10.50 | jhiver | hey, how come when a destination is circuit-busy it executes congestion instead of doing the next Dial() command as specified in my extensions.conf ? |
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15:11.02 | tzafrir_laptop | BTW: areyou related to JCraft's WierdX ? |
15:11.08 | *** part/#asterisk acidchild (i=ash@unaffiliated/acidchild) |
15:12.51 | jhiver | Can anybody take a look at my macro and tell me what's wrong? |
15:13.01 | jhiver | http://pastebin.ca/63732 |
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15:13.39 | Qwell | jhiver: What is it (not) doing? |
15:13.50 | jhiver | when it hits exten => s,100,Dial(SIP/0${ARG1:3}@83.206.114.85) and if it's circuit busy, then it executes congestion instead of jumping to 200 |
15:14.18 | jhiver | ideally I'd like that if the response code is anything but answered, it goes to the next step |
15:14.23 | jhiver | I use 1.2.7.1 |
15:14.45 | jhiver | and I have autofallthrough=yes |
15:15.13 | jhiver | because I have this: |
15:15.15 | jhiver | # |
15:15.16 | jhiver | exten => s,100,Dial(SIP/0${ARG1:3}@83.206.114.85) |
15:15.16 | jhiver | # |
15:15.16 | jhiver | exten => s,101,Goto(200) |
15:15.16 | jhiver | # |
15:15.17 | jhiver | exten => s,200,Macro(loadbalance-orange,${ARG1}) |
15:15.18 | jhiver | # |
15:15.20 | jhiver | exten => s,201,Congestion() |
15:15.22 | jhiver | oops |
15:15.24 | jhiver | sorry |
15:15.30 | jhiver | that was supposed to be only 4 lines |
15:16.15 | robin_sz | so the s,101 line is redundant right? |
15:16.23 | robin_sz | goto the next line? |
15:16.45 | blitzrage | priority numbers need to be sequential |
15:16.47 | jhiver | Well, I don't think it would jump straight from 100 to 200 would it? |
15:16.50 | blitzrage | so its not redundant |
15:16.56 | robin_sz | k |
15:16.56 | Qwell | blitzrage: notice the goto(200)? :p |
15:17.00 | blitzrage | but you should be using 'n', not actual numbers |
15:17.06 | blitzrage | Qwell: yes, I do |
15:17.14 | Qwell | 100>101>200>201 |
15:17.20 | jhiver | yeah well old habits from asterisk 1.0.9 |
15:17.23 | blitzrage | right... thats why the Goto() |
15:17.24 | Qwell | jhiver: check the DIALSTATUS var |
15:18.35 | jhiver | I know it's a circuit busy |
15:18.49 | jhiver | that's what it says on the CLI |
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15:19.18 | SexyKen | Hey there fellas! |
15:19.21 | jhiver | trouble is I can't faff about too much with it because it's in production, I guess I'll have to wait during night time to see what's going wrong |
15:19.36 | Qwell | jhiver: You assume it'll do priority jumping |
15:19.51 | jhiver | but anyway, isn't asterisk supposed to go to the next step when a call isn't answered? |
15:19.58 | SexyKen | Guys -- we're all a pretty interactive community here! We help each other with issues and whatnot.... |
15:20.05 | blitzrage | jhiver: yes |
15:20.14 | jhiver | is there some option I haven't set or something? |
15:20.29 | robin_sz | jhiver, just type extensions reload a few times and you'll soon solve the "in production" annoyance ;) |
15:20.33 | SexyKen | ....maybe you guys would be interested in donating some change for the Relay for Life that I'm doing? It's to raise money for cancer research. |
15:20.39 | jhiver | robin_sz, ? |
15:21.17 | robin_sz | SexyKen, go bother the govenrment about it |
15:21.17 | jhiver | I _do_ do reloads rather than a restart... can't really afford to restart the box except maybe at 4:00 am or something |
15:21.53 | robin_sz | jhiver, well, eventually, reloads seem to make things unstable |
15:22.07 | jhiver | does it? |
15:22.09 | Qwell | robin_sz: eh? |
15:22.10 | jhiver | crap |
15:22.13 | Qwell | no.. |
15:22.19 | jhiver | might be why it's not working then :) |
15:22.24 | robin_sz | shrug .. thats my experience |
15:22.28 | blitzrage | just reload the module you need |
15:22.34 | blitzrage | reload chan_sip.so for example |
15:22.34 | Qwell | robin_sz: got a bug number? |
15:22.41 | Qwell | blitzrage: sip reload! |
15:22.47 | Qwell | (is there actually a difference?) |
15:22.51 | blitzrage | Qwell: I perfer just to reload the module :) |
15:22.55 | blitzrage | Qwell: nah... no difference |
15:22.56 | robin_sz | and thats guaranteed memeroy leak and after effect free? |
15:23.06 | jhiver | ok, well, I'll do some tests tonight at 4am GMT + 4 |
15:23.19 | *** part/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
15:23.27 | jhiver | the channel should be still pretty active so I guess I'll ask when I have some time to fuck about with my Asterisk box |
15:23.51 | jhiver | because right now there is too much traffic for testing |
15:23.55 | robin_sz | personally, after a bit of reloading, I do a stop when conveninet and a resatrt |
15:24.22 | ghenry | Where do you set things to put a zap channel on hold, call another user, then put the original caller through to them by hanging up? i.e. call transfer and call forwarding? zapata.conf? Also, is this the same as Parking a call? I don't think so, as that's to move on to another phone and retrieve the call, correct? Thanks. brb |
15:24.26 | robin_sz | "stop when convenient" is pretty kewl, unless it keeps you sitting there waiting for it to die for an hour |
15:24.35 | jhiver | it's a pretty strange issue though.... |
15:24.57 | jhiver | robin_sz, it would only work at night time for me anyway |
15:24.58 | jhiver | so... |
15:25.15 | robin_sz | people talk too much huh? |
15:25.44 | jhiver | anyway, has this anything to do with autofallthrough=yes? |
15:26.27 | jhiver | ; If autofallthrough is set, then if an extension runs out of |
15:26.28 | jhiver | ; things to do, it will terminate the call with BUSY, CONGESTION |
15:26.28 | jhiver | ; or HANGUP depending on Asterisk's best guess (strongly recommended). |
15:26.31 | Qwell | jhiver: priorityjumping |
15:26.53 | robin_sz | Qwell, http://www.voip-info.org/wiki/view/Asterisk+administration |
15:27.05 | robin_sz | A repetitive reload will not be sufficient, and can actually cause more harm (instability, memory not being released, see bug tracker) than it does good |
15:27.10 | robin_sz | ok? |
15:27.18 | jhiver | I don't have that set |
15:27.18 | Qwell | wtf |
15:27.22 | jhiver | might be it |
15:27.26 | Qwell | there is a ton of BS on this page |
15:27.50 | robin_sz | there it is .. in the wiki .. whenever I ask I always get told "read the wiki" I did . |
15:27.55 | Qwell | WTF |
15:28.29 | jhiver | Dial sets DIALSTATUS to indicate its success. However, under some circumstances, execution will jump to priority n+101 in the current context |
15:28.31 | Qwell | Note that it is recommended to do a clean CVS checkout instead of a CVS update! |
15:28.32 | Qwell | BULLSHIT |
15:28.52 | ghenry | Is it config option as well, that when a zap/sip user puts a call on hold, they hear music? |
15:28.53 | robin_sz | well, feel free to edit it :) |
15:29.00 | jhiver | so |
15:29.18 | jhiver | I guess I have to add priorityjumping=yes to the [general] section |
15:29.47 | jhiver | and then change the extensions.conf to have something smart at extension 201 (since I attempt the dial at 100) ? |
15:29.47 | Qwell | jhiver: Do you want it to jump to 201? |
15:30.02 | Qwell | You said it already is |
15:30.06 | Qwell | which means you have that set |
15:30.25 | jhiver | it isn't! |
15:30.25 | Qwell | so then check DIALSTATUS |
15:30.25 | jhiver | it executes Congestion when everything is fast-busy |
15:30.30 | Qwell | So then it is |
15:30.40 | Qwell | So remove that option, and check DIALSTATUS |
15:30.40 | jhiver | I thought it would go to +1 but apparenty not |
15:31.47 | *** join/#asterisk parag7732 (n=root@de1-b1453.alshamil.net.ae) |
15:33.09 | Damin | That's complete BULLSHIT! You have no idea what the hell you are talking about! |
15:33.27 | Qwell | Damin: troll |
15:33.28 | Qwell | :P |
15:33.30 | parag7732 | Hi...I m wondering that i m able to configure the "external voip provider account" with Linksys SPA942 and Pap2.....but its not working with asterisk....!!!!!!!!! why |
15:33.39 | Qwell | There we go. Line 3. http://www.voip-info.org/wiki/view/Asterisk+administration |
15:33.49 | parag7732 | i created the right trunks |
15:33.56 | parag7732 | and peers |
15:34.45 | robin_sz | Qwell: thats a start, now repeat the process for the rest of the wiki |
15:41.23 | stephane_ | re |
15:45.49 | Dr-Linux | re |
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15:48.47 | *** join/#asterisk Borgon (n=l3orgon@host-69-59-103-160.nctv.com) |
15:49.11 | Borgon | if am using a voip software that uses sip, can i make it connect to my asterisk by sip then use asterisk to dial using iax2? |
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15:52.41 | grabowski | Borgon: of course |
15:52.54 | Borgon | grabowski: hrm ok |
15:53.24 | jaike | anyone running asterisk on AMD dual cores? |
15:54.34 | *** join/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com) |
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15:58.22 | Dr-Linux | why WIKI suggest to restart the asterisk regularly????? |
15:58.26 | Dr-Linux | here: |
15:58.28 | Dr-Linux | Regularly restart (better: stop and start) your PBX during off-hours. A repetitive reload will not be sufficient, and can actually cause more harm (instability, memory not being released, see bug tracker) than it does good. If you run Asterisk provisioned for automatic reloading this could be as simple as placing a cronjob to execute asterisk -rx 'stop gracefully'. |
15:59.43 | Qwell | Dr-Linux: It's wrong |
16:00.06 | kumamoto | Damn Qwell you are still here? |
16:00.19 | Dr-Linux | Qwell: i'm reading the link, that you pasted above |
16:00.22 | Qwell | kumamoto: sure |
16:00.27 | Dr-Linux | <Qwell> There we go. Line 3. http://www.voip-info.org/wiki/view/Asterisk+administration |
16:00.29 | Qwell | Dr-Linux: somebody else pasted it |
16:00.34 | Qwell | Dr-Linux: yes, read line 3 ;) |
16:00.40 | kumamoto | I appreciate your help in this channel |
16:00.54 | Qwell | kumamoto: donations accepted :p |
16:00.58 | Dr-Linux | ok :P |
16:00.59 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
16:01.07 | Qwell | Dr-Linux: me, saying all the stuff there is wrong, heh |
16:01.33 | Dr-Linux | :P |
16:01.36 | jaike | from experience, better restart rather than just reload |
16:01.43 | Dr-Linux | Qwell: what's the line 3 ? |
16:01.49 | kumamoto | Qwell, once I get into the business of voip full time I will be here fulltime and maybe we can talk on donations. |
16:01.58 | Qwell | Dr-Linux: "edit: Half of the stuff in here is completely false. Take it all with a grain of salt." |
16:01.59 | kumamoto | At this time I am still learning |
16:02.15 | Qwell | Dr-Linux: That's line 3 |
16:02.46 | Dr-Linux | ok |
16:03.06 | kumamoto | I was wondering those IP phones with 2 nics is one intended for POE or for bridging |
16:03.18 | Qwell | kumamoto: It's a builtin switch |
16:04.24 | kumamoto | aha so not a POE thing? That is good to know I always thought it was a POE port |
16:04.26 | kumamoto | thanks |
16:04.38 | Qwell | kumamoto: poe uses the same port |
16:04.44 | Qwell | if the phone supports poe, that is |
16:05.03 | kumamoto | so it is interchangeable if it supports POE |
16:10.07 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
16:12.56 | [TK]D-Fender | kumamoto : Some phones have switches, some have PoE, others both or neither. |
16:13.32 | [TK]D-Fender | PoE is only ever enabled on 1 of the 2 ports on the phone if present at all. |
16:13.33 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
16:16.29 | kumamoto | aha nice |
16:18.04 | [TK]D-Fender | kumamoto : What models are you looking at? |
16:20.06 | kumamoto | Budgetstream GS-102 |
16:20.25 | kumamoto | or the polycom 103 |
16:21.09 | wunderkin | heh.. you have that all backwords |
16:21.28 | kumamoto | i thought no one will catch it hehe |
16:22.20 | kumamoto | Grandstream Budgetone 102 |
16:22.36 | Qwell | yuck |
16:22.42 | Qwell | barbietone |
16:22.52 | wunderkin | :D |
16:22.54 | kumamoto | Polycom SoundPoint IP 301 |
16:23.12 | kumamoto | aaah what is now wrong with the grandstream |
16:23.43 | Qwell | My First VoIP Phone (TM) |
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16:24.49 | kumamoto | was it the 101 or 102? |
16:24.56 | kumamoto | or GXP2000 |
16:27.11 | jaike | polycoms very sturdy |
16:27.46 | Qwell | cisco very purdy |
16:27.58 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
16:28.07 | *** join/#asterisk litage (n=nick@203.220.55.70) |
16:28.14 | kumamoto | So what y'all saying is rather spend extra on the polycom than the grandstream |
16:28.20 | Qwell | yes |
16:28.41 | kumamoto | ok how does the faxing work |
16:29.54 | [TK]D-Fender | kumamoto : Forget Grandstream altogther. Cheap crap. |
16:30.37 | kumamoto | true it is cheap |
16:30.55 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
16:31.21 | *** join/#asterisk litage (n=nick@203.220.55.70) |
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16:31.47 | anonymouz666 | grandstream firmware is not good. |
16:31.50 | [TK]D-Fender | kumamoto : And believe me the "crap" is just as true. |
16:32.06 | [TK]D-Fender | kumamoto : Polycom is great stuff.... |
16:32.07 | kumamoto | how about the sipura phone |
16:32.27 | [TK]D-Fender | kumamoto : Sipura/Linksys SPA-94X can be ok, but ti depends. |
16:32.33 | [TK]D-Fender | kumamoto : Where are you located? |
16:32.46 | kumamoto | in dallas |
16:33.13 | [TK]D-Fender | kumamoto : Then Sipura's price point devalidates it. Polycom is a superior product at a price on par with Sipura |
16:33.20 | [TK]D-Fender | kumamoto : www.atacomm.com |
16:34.24 | kumamoto | damn thanks for that links I always seem to stick to voipsupply.com |
16:35.46 | kumamoto | Any reviews on the Snom phone? |
16:36.12 | [TK]D-Fender | Snom is better than the cheap crap, but still unstable firmware, and crappy LCD usability |
16:36.22 | [TK]D-Fender | kumamoto : Stick with Polycom, trust me... |
16:37.35 | [TK]D-Fender | So far the only other phones in Polycom's class are Cisco, and they are PRICY. |
16:37.35 | kumamoto | isn't snom the one running with a linux based firmware? |
16:37.56 | kumamoto | I guess for now polycom wins |
16:37.57 | [TK]D-Fender | kumamoto : Yup... great web interface and all that, but when it comes to using the phone... cik... |
16:38.08 | [TK]D-Fender | (in reference to SNOM) |
16:38.22 | *** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca) |
16:38.30 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
16:38.55 | kumamoto | polycom will work with asterisk@home (tribox)? |
16:38.58 | [TK]D-Fender | Polycom is a bit more complex on setup, but it pays, and is stable and a sturdy product. |
16:39.23 | kumamoto | What is the software that does faxing using voip? |
16:39.32 | [TK]D-Fender | kumamoto : Like any other SIP phone yes, but Trixbox is a craptastic cookie-cutter system that will box you in. |
16:39.36 | kumamoto | I guess polycom it is |
16:39.40 | JakBeatZ | Folks.. having a weird issue with a 7960. have a bunch of SIP extenstions configured on the phone and call waiting is turned on, on the phone, but when I try to dial one extension from the other (on the same phone) I get a 486. I'm running 8.2 SIP on the phone.. anyone seen anything like that before? |
16:39.48 | [TK]D-Fender | You're better off learning how to run * for yourself. there really isn't that much to it. |
16:40.27 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
16:41.04 | kumamoto | It is those configuration files that seem to be intimidating but I will get over it |
16:41.54 | [TK]D-Fender | ok, ewll good luck to you, I'm off... |
16:41.57 | [TK]D-Fender | back later.... |
16:42.17 | kumamoto | thanks to all your help and good information |
16:42.22 | kumamoto | I will be back later |
16:42.25 | *** part/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com) |
16:45.44 | *** part/#asterisk parag7732 (n=root@de1-b1453.alshamil.net.ae) |
16:47.04 | *** join/#asterisk doolph (n=doolph@200.75.204.169) |
16:47.12 | doolph | how can I compile without zaptel |
16:47.26 | Qwell | doolph: same way |
16:47.45 | doolph | in asterisk dir I get chan_zap.c: In function `pri_dchannel': |
16:47.56 | Qwell | uninstall zaptel |
16:48.02 | Qwell | or upgrade |
16:48.25 | doolph | I tried to upgrade zaptel and i get another error |
16:48.28 | doolph | how can I uninstall it |
16:49.15 | doolph | I dont have zap hardware though |
16:50.00 | doolph | erm nvm the problem was zaptel head |
16:51.41 | *** join/#asterisk litage (n=nick@203.220.55.70) |
16:56.17 | ghenry | with faxdetect, is it easy to redirect this call to a fax machine? |
16:56.47 | *** join/#asterisk litage (n=nick@203.220.55.70) |
16:58.18 | *** join/#asterisk dstr_2 (i=[U2FsdGV@kolsyra.mer.nu) |
16:58.56 | *** part/#asterisk doolph (n=doolph@200.75.204.169) |
17:00.45 | dstr_2 | How do I pass the pundkey through asterisk? When calling another pbx they ask me to press 1 and the pound key, which drops me into my own asterisk |
17:05.09 | *** join/#asterisk litage (n=nick@203.220.55.70) |
17:06.42 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
17:07.40 | *** join/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com) |
17:09.45 | jaike | change the transfer key in features.conf? not sure though |
17:10.11 | grabowski | dstr_2: just dont add the T option to your outbound Dial cmd |
17:10.12 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
17:10.59 | grabowski | dstr_2: This disables the feature that allows you to transfer on your own PBX with the pound key tho (for that outbound dial cmd) |
17:11.32 | dstr_2 | grabowski : is there a way to work around it? |
17:12.04 | grabowski | dstr_2: besides what jaike said and change the transfer key no |
17:12.26 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
17:12.35 | grabowski | dstr_2: You can still transfer on a phone that has a transfer button.. |
17:13.09 | dstr_2 | all I find is blind transfer |
17:13.36 | grabowski | I think pound is a blind transfer |
17:13.54 | dstr_2 | ;blindxfer => #1 ; Blind transfer |
17:14.04 | grabowski | dstr_2: http://www.voip-info.org/wiki-Asterisk+config+features.conf |
17:14.34 | grabowski | dstr_2: ;blindxfer => #1 ; Blind transfer, default is # |
17:15.14 | grabowski | so you could uncomment that line and that should make the new blind transfer #1 |
17:16.28 | JakBeatZ | Hey.. anyone have any ideas why a 7960 won't let me dial from one extension to the other on the same phone even though call waiting is enabled? |
17:16.40 | Qwell | JakBeatZ: different contexts? |
17:17.03 | *** part/#asterisk jaike (i=jaike@210.213.168.167) |
17:17.04 | dstr_2 | grabowski : ha! it works now! |
17:17.09 | dstr_2 | i changed it to *9 |
17:17.55 | JakBeatZ | Qwell: No, same context. I get a 486 busy here. It's like the cisco doesn't know call waiting is enabled |
17:18.43 | dstr_2 | Perhapps i should just add a dialplan for the banks phonenumber, without the T? |
17:18.48 | *** join/#asterisk litage (n=nick@203.220.55.70) |
17:19.24 | grabowski | JakBeatZ: I have a 7960 beside me.. when I call myself its rings and I have done nothing fancy in the phone config.. |
17:19.55 | dstr_2 | but *9 wont work. :/ |
17:20.36 | grabowski | dstr_2: I think you need to press it quickly |
17:20.38 | JakBeatZ | grabowski: I know, that's the funny part.. I don't understand what's going on. I can dial from a soft phone to the 7960 no problem and vice-verse. I can call soft phone to soft phone with no issues, but the 7960 won't do it.. What code are you running on your 7960? |
17:21.21 | *** join/#asterisk ast_freak (n=jesse@72.25.129.66) |
17:21.23 | grabowski | JakBeatZ: what do you mean by code? Its the SIP firmware P0S3-04-4-00 |
17:22.09 | grabowski | JakBeatZ: want to take a look at my SIP-mac-addy.cnf file? |
17:22.14 | JakBeatZ | grabowski: code/firmware.. sorry.. that term is interchangable for me.. Wow, 4-4-00. I'm up to 8-2 and it's worked fine so far |
17:22.39 | JakBeatZ | grabowski: except for the call waiting, that is.. hmm... |
17:22.39 | grabowski | JakBeatZ: any reason I should upgrade? |
17:23.21 | JakBeatZ | grabowski: if it works, all the power to ya :) I'm just a child of 'latest and greatest' so that's why I'm up to 8-2. Unless there are new features in newer code you want, but I dunno. |
17:23.23 | dstr_2 | ah! 500ms was a bit to narrow |
17:23.45 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
17:23.49 | JakBeatZ | s/code/firmware :) |
17:24.25 | grabowski | JakBeatZ: you need a cisco login to get the latest sip firmware do you not? |
17:25.11 | JakBeatZ | grabowski: I believe so, yes. And you may need a support contract number tied to the login to be able to download firmware. |
17:25.37 | JakBeatZ | grabowski: but I hear support contracts on phones are dirt cheap so it may be worth while. |
17:26.19 | grabowski | dstr_2: just tryed #1 worked fine for me |
17:26.23 | *** join/#asterisk thermf (i=fadaasfa@d14-69-149-97.try.wideopenwest.com) |
17:26.47 | grabowski | JakBeatZ: so yea it may be something with the new firmware I guess |
17:27.09 | JakBeatZ | sheeit. |
17:27.37 | grabowski | JakBeatZ: again, if you want Ill post my SIP.cnf file |
17:29.47 | JakBeatZ | grabowski: Sure, I'll check it out, if you don't mind. |
17:29.58 | asterisk-dud | i'm trying to recompile asterisk with spandsp and i'm getting compile errors |
17:30.19 | asterisk-dud | any advice? |
17:32.33 | thermf | asterisk-dud: what are the compile errors? |
17:33.46 | grabowski | JakBeatZ: http://pastebin.ca/63787 |
17:33.51 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
17:33.53 | asterisk-dud | there are a whole bunch of no member named errors |
17:34.07 | asterisk-dud | incomplete pointer type |
17:34.25 | grabowski | JakBeatZ: Im going to look though my SIPDefault.cnf as well.. 1 sec |
17:34.27 | robin_sz | dang those pointers |
17:34.31 | asterisk-dud | 't30_flsuh' defined but not used |
17:34.33 | robin_sz | they really should learn to get along |
17:34.49 | asterisk-dud | app_rxfax.c is creating the errors |
17:34.54 | thermf | ah |
17:34.55 | JakBeatZ | grabowski: Ya, nothing out of the ordinary there. Mine is very similar. |
17:35.08 | thermf | are you compiling with the correct spandsp version? |
17:35.48 | grabowski | JakBeatZ: in my SIPDefault.cnf I have call_waiting: "1" |
17:35.49 | asterisk-dud | i have 0.0.3 pre20 |
17:35.55 | grabowski | JakBeatZ: # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) |
17:36.15 | *** join/#asterisk Splat (n=Splat@220-253-105-26.TAS.netspace.net.au) |
17:36.45 | JakBeatZ | grabowski: ditto. except that's in my SIPmac.conf |
17:36.49 | thermf | asterisk-dud: where did you get the app_rxfax from? it is probably made for 0.0.2 |
17:36.57 | asterisk-dud | themf: thats the prob, thanks |
17:37.09 | grabowski | JakBeatZ: maybe it needs to be in the SIPDefault.cnf as well? |
17:37.37 | JakBeatZ | grabowski: Not sure.. I'll check. |
17:37.56 | thermf | asterisk-dud: did you just download 0.0.3pre20 today? i actually have never seen that version |
17:38.27 | asterisk-dud | yes i did, but i'll download the pre26 |
17:39.20 | *** part/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com) |
17:39.53 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
17:42.15 | thermf | does anyone have the version of app_rxfax/app_txfax that makes use of t.38? |
17:44.04 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
17:46.07 | grabowski | JakBeatZ: any luck? |
17:48.33 | JakBeatZ | grabowski: No. I just posted to the Cisco NetPro forums.. |
17:49.06 | JakBeatZ | grabowski: I never used a SIPDefault.cnf file and call waiting worked previously many moons ago when I tried so I don't think it's that. |
17:49.40 | grabowski | JakBeatZ: Well, good luck with that. |
17:50.26 | JakBeatZ | grabowski: thx for your help. |
17:53.08 | dstr_2 | Which IP-phone is the cheapest/best for use with asterisk? could this be a good pick? http://www.ntsweden.se/public/voipsortiment/IP300.pdf |
17:53.56 | dstr_2 | i want something simple at home |
17:54.11 | *** join/#asterisk shmur (n=blern@157.130.10.166) |
17:54.13 | grabowski | dstr_2: cheapest and best do not coincide |
17:54.17 | dstr_2 | hehe |
17:54.25 | grabowski | dstr_2: the best of the cheapest maybe.. |
17:54.45 | dstr_2 | just cheapest then... since it's for home. |
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17:56.05 | grabowski | dstr_2: The Grandstream budgettone (I think) is about the same quality of your average residential phone. For something a little better without much more maybe get a polycom. |
17:56.51 | grabowski | dstr_2: Take a look at http://voip-info.org/wiki/view/Cheapest+ATAs+and+Service in the Phones section. |
17:59.27 | grabowski | dstr_2: I guess get a Grandstream BT-102 so you have the dual RJ45 |
18:01.35 | dstr_2 | looks nice |
18:01.36 | grabowski | dstr_2: The polycom you linked there is a good quality phone. |
18:01.55 | dstr_2 | i think the polycom is about $130 here in sweden |
18:05.19 | JakBeatZ | grabowski: looks like it's firmware. put P0S3-07-5-00 on it and it works fine. |
18:05.27 | asterisk-dud | spandsp and asterisk recompiled without errors but application rxfax is still not found |
18:05.38 | asterisk-dud | how can i make sure it's loaded |
18:05.43 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
18:05.52 | grabowski | JakBeatZ: Yet another reason not to upgrade. |
18:08.09 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
18:10.05 | thermf | asterisk-dud: asterisk -r -x 'show application rxfax' |
18:11.52 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
18:12.01 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.192) |
18:12.52 | shido6 | rxfax and spandsp, eh ? :) you make the changed to your Makefile in the apps directory? |
18:13.08 | yxa | what does this mean: Got SIP response 485 "Ambiguous" back from x.x.x.x |
18:13.59 | asterisk-dud | does asterisk overwrite my extension.conf file when i recompile to a newer version? |
18:14.10 | shido6 | as long as you dont do make samples |
18:14.15 | asterisk-dud | ok, thanks |
18:14.43 | mitcheloc | hey shido6 |
18:14.47 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
18:14.53 | shido6 | hey |
18:15.01 | mitcheloc | how have you been? |
18:16.04 | shido6 | ok||ko |
18:17.18 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
18:17.21 | TripleFFFF | hey akk |
18:18.08 | TripleFFFF | ok where can i find old specs on the TE110P |
18:19.54 | yxa | anyone knows what the SIP response 485 means? |
18:20.07 | TripleFFFF | 4xx is authorization |
18:20.26 | *** join/#asterisk NDT (n=noone@cpe-72-228-10-145.nycap.res.rr.com) |
18:21.37 | NDT | Anyone running 2 or more TE410P cards in a dell 2850? |
18:22.23 | TripleFFFF | bad idea |
18:22.35 | TripleFFFF | so where can in find the te 110 ancestor |
18:22.54 | justinu|laptop | 485 ambiguous means there wasn't enough digits in the address to route the call |
18:23.08 | NDT | I have 1 card that runs fine added another and it doesn't see it...even change the dial on them to 0 and 1 respectively |
18:24.25 | yxa | justinu|laptop means i'm missing something? |
18:25.34 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
18:28.04 | justinu|laptop | 21.4.23 485 Ambiguous The Request-URI was ambiguous. The response MAY contain a listing of possible unambiguous addresses in Contact header fields. Revealing alternatives can infringe on privacy of the user or the organization. It MUST be possible to configure a server to respond with status 404 (Not Found) or to suppress the listing of possible choices for ambiguous Request-URIs. |
18:28.39 | justinu|laptop | Example response to a request with the Request-URI sip:lee@example.com: SIP/2.0 485 Ambiguous Contact: Carol Lee <sip:carol.lee@example.com> Contact: Ping Lee <sip:p.lee@example.com> Contact: Lee M. Foote <sips:lee.foote@example.com> |
18:29.21 | thermf | does anyone have the version of app_rxfax/app_txfax that makes use of t.38? |
18:29.55 | TripleFFFF | hey is t38 avail / ?? boutny is sitll up lol |
18:30.14 | *** join/#asterisk ikey (i=ikey@220.226.47.200) |
18:33.44 | *** join/#asterisk salah (n=salah@216-30-75.0505.adsl.tele2.no) |
18:36.07 | shmur | hi everyone, im looking at setting up an asterisk box at work, and Im wondering about support for a specific dialogic card? i see the support for 240 on the website, but its not the specific model. is the support for dialogic cards fairly universal? |
18:36.36 | *** part/#asterisk salah (n=salah@216-30-75.0505.adsl.tele2.no) |
18:38.07 | asterisk-dud | tiff will not biuld: [tif_stream.lo] Error 1 \ |
18:38.18 | shmur | the specific model is D/240sc-T1 Rev 3 |
18:38.26 | asterisk-dud | line 837: g++: command not found |
18:38.33 | asterisk-dud | i have gcc installed |
18:38.43 | justinu|laptop | shmur: a little advice... forget about the dialogic card, and get a Digium or Sangoma T1 card |
18:38.57 | justinu|laptop | single T1 card is only about $459 |
18:39.24 | shmur | I would like to buy a fully support card, but for right now the boss kind of wants to use what we have in house which is why im wondering |
18:39.44 | websae | who can fuck me? |
18:40.25 | websae | i want to fuck everyone |
18:41.41 | Qwell | I want to do something..that matters.. |
18:42.19 | salviadud | nin song? |
18:42.25 | Qwell | :D |
18:42.37 | Qwell | salviadud: was wondering if anybody would catch that |
18:42.57 | salviadud | Qwell, I like NIN, especially the quake soundtrack |
18:43.01 | Qwell | mmhmm |
18:43.31 | Qwell | For some reason, I don't actually have that |
18:44.04 | *** part/#asterisk littlejohn (n=little@host217-58.pool8717.interbusiness.it) |
18:44.10 | salviadud | well, you can down it from galbadia hotel, or just get a copy of quake 1 |
18:44.20 | Qwell | yeah.. |
18:44.35 | shmur | since the advice is to forget about the dialogic card, can i assume that the support for it in asterisk isn't the greatest? |
18:44.38 | ghenry | how do you transfer a call on a SIP phone? |
18:44.46 | Qwell | ghenry: transfer button? |
18:44.51 | salviadud | Qwell, it's worth your while dude, it's the best MOH if you want them to hang up quick |
18:44.55 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
18:45.00 | Qwell | heh |
18:45.04 | ghenry | Yeah ;-) I see a XFer button |
18:45.06 | SRCR | I'm unable to get inbound SIP calls see http://pastebin.ca/63826 for details. |
18:45.54 | ghenry | Qwell: what about on an anagloue phone? |
18:45.55 | justinu|laptop | SRCR: check line 50 |
18:46.29 | SRCR | justinu|laptop: i have a entry provider-in in extensions |
18:46.41 | justinu|laptop | an entry called "provider"? |
18:48.03 | TripleFFFF | the ancestor of digium there was the code on google once |
18:48.14 | TripleFFFF | anyone remmber ? think it was on a spanish site |
18:48.48 | thermf | does Mithraen visit here? |
18:49.20 | SRCR | justinu|laptop: I have added a [provider] in my extensions.conf.. this should do the trick ? |
18:49.34 | justinu|laptop | srcr: no |
18:49.45 | justinu|laptop | you need an entry in your [provider-in] context |
18:49.49 | justinu|laptop | something that looks like: |
18:50.10 | justinu|laptop | exten => provider,1,Playback(file) |
18:50.28 | *** join/#asterisk anthm (n=anthm@000-445-169.area4.spcsdns.net) |
18:50.28 | *** mode/#asterisk [+o anthm] by ChanServ |
18:50.57 | SRCR | justinu|laptop: ok did that i'll test |
18:51.36 | justinu|laptop | SRCR: file isn't really a file you can play, replace it with something in /var/lib/asterisk/sounds |
18:51.54 | SRCR | justinu|laptop: i know.. :) thanks |
18:52.07 | justinu|laptop | never can tell with people in this room :P |
18:52.54 | SRCR | justinu|laptop: it almost works.. i forgot to fireup the softphone :| |
18:55.21 | SRCR | justinu|laptop: it works.. i stil had entries like 'exten = > s,1,Answer' |
18:55.59 | justinu|laptop | the "s" exten is only really used for macros and channel tech that doesn't provide DNIS info |
18:56.26 | SRCR | justinu|laptop: you lost me.. but i'll figure it out someday |
18:57.18 | justinu|laptop | DNIS is a way that you can tell what number the calling party dialed |
18:57.28 | justinu|laptop | POTS lines for example won't tell you that |
18:58.03 | *** join/#asterisk darby_t (i=darby_t@aaph227.neoplus.adsl.tpnet.pl) |
18:58.06 | justinu|laptop | try the bot for acronyms you don't know |
18:58.09 | justinu|laptop | ~dnis |
18:58.11 | jbot | from memory, dnis is A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination device, which ... |
18:58.14 | justinu|laptop | ~pots |
18:58.16 | jbot | it has been said that pots is Plain Old Telephone Service as in "Old Analogue Crap" |
18:58.29 | TripleFFFF | foud it |
18:59.12 | mitcheloc | there should be a help doc generated off of jbots vocabulary =) |
18:59.23 | justinu|laptop | ~status |
18:59.24 | jbot | Since Tue Jun 6 21:15:44 2006, there have been 76 modifications, 803 questions, 0 dunnos, 0 morons and 637 commands. I have been awake for 3d 21h 43m 39s this session, and currently reference 110978 factoids. I'm using about 18976 kB of memory. With 0 active forks. Process time user/system 11268.56/926.45 child 0.04/0.01 |
18:59.29 | justinu|laptop | jbot, laskjdakdj? |
18:59.53 | NDT | anyone have 2 TE410P with 1st gen and 2nd gen firmware woring in the same box? I am starting to think that is my issue |
19:00.57 | justinu|laptop | pretty slim chance of you finding someone with that specific config, NDT |
19:01.22 | justinu|laptop | does it not work right with both cards? |
19:01.37 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
19:02.20 | NDT | Well...the older one was in the dell 2850...added the newer one and it won't detect the old one |
19:02.39 | justinu|laptop | i assume when the new card comes out, the old one starts working again? |
19:02.44 | NDT | lspci only shows the new one...take the new one out the old one shows |
19:02.53 | NDT | hehe yep |
19:02.57 | justinu|laptop | that's too bad |
19:03.10 | justinu|laptop | you might wanna talk to digium about it |
19:03.16 | justinu|laptop | seems broke |
19:04.09 | NDT | yeah...what a pain in the ass...friggin datacenter is 4 hours away...There is a 2nd gen in anotehr machine...maybe I can bring it down and swp em |
19:04.14 | NDT | err swap em |
19:04.30 | justinu|laptop | sounds like your weekend is toast |
19:05.05 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
19:05.11 | NDT | I didn't think they would make them incompatible ughhh |
19:06.02 | NDT | You can flash sangoma cards right? |
19:06.28 | justinu|laptop | not sure... i've only worked on one system with a sangoma |
19:06.32 | justinu|laptop | A101 card |
19:07.26 | NDT | I saw this on a page |
19:07.28 | NDT | > The 2nd gen firmware has field-upgradeability. The 1st gen firmware does |
19:07.28 | NDT | > not, unfortunately. There is not currently any 3rd gen firmware, but |
19:07.28 | NDT | > when there is, you'll be able to do it yourself |
19:07.37 | NDT | wonder if I can downgrade it then LOL |
19:10.10 | NDT | somewhere else says it isn't upgradeable...blah...don't think digium is even there on the weekend are they? |
19:10.26 | justinu|laptop | prolly not |
19:11.04 | asterisk-dud | well, i can't get asterisk to compile with rx_fax app, it gets all kind of errors, i have checked to see if I have the same versions |
19:11.06 | NDT | nah says they aren't on the site... |
19:11.10 | asterisk-dud | anyone have this trouble? |
19:11.45 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
19:12.45 | asterisk-dud | anyone with spandsp experience |
19:17.07 | reza_ | anyone buy from discountvoipoutlet.com before? reputable? |
19:20.22 | NDT | never...I know I haven't found anyone cheaper on TE410P cards though then netxusa |
19:20.43 | docelm0 | Where can I find cheap TDM400P's? |
19:21.22 | asterisk-dud | make[1]: *** [app_rxfax.o] Error 1 |
19:22.22 | NDT | I buy my quad TE410P cards from http://www.netxusa.com/ guy named rick...for $1262 cheapest I have found so far |
19:22.35 | NDT | maybe they have TDM400s too dunno |
19:22.54 | NDT | yeah says they do on their site |
19:24.24 | asterisk-dud | make[1]: *** [app_rxfax.o] Error 1 |
19:25.27 | docelm0 | does netxusa sell to end users or just businesses? |
19:28.37 | docelm0 | asterisk-dud quit being a dick.. What do you want? |
19:30.22 | thermf | asterisk-dud: rxfax/txfax usually compile without much trouble if you have the right spandsp version installed on your system |
19:30.40 | thermf | asterisk-dud: make sure that you remove all versions of spandsp and then "make install" spandsp 0.0.2 |
19:30.43 | *** join/#asterisk litage (n=nick@203.220.55.70) |
19:31.48 | NDT | docelm0: think both |
19:39.12 | *** join/#asterisk litage (n=nick@203.220.55.70) |
19:40.26 | sengland | Anyone know why I cant get the context to work under Realtime configs and IAX ? |
19:40.41 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
19:41.28 | *** join/#asterisk Whoopie_ (n=Whoopie@p54A7AFF5.dip0.t-ipconnect.de) |
19:43.02 | Whoopie_ | Hi, I'd like to uninstall asterisk 1.0.10. Compiled it myself, but want to switch to debian packages (asterisk 1.2.x.x). Is there somewhere an uninstaller? there's no possibility to do "make uninstall"? |
19:45.35 | *** join/#asterisk Ciber311 (i=Ciber@user-1087e94.cable.mindspring.com) |
19:45.55 | Strom_C | Whoopie_: just compile 1.2.9.1 from source |
19:46.02 | Strom_C | the packages are likely to be out of date |
19:47.17 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
19:48.09 | asterisk-dud | Hunk #2 failed at 100 while trying to patch asterisk for spandsp |
19:48.18 | asterisk-dud | what coudl be wrong |
19:48.44 | Whoopie_ | Strom_C: But are all files replaced or are there then some files from 1.0.10? |
19:52.46 | Ciber311 | so what voip providers are you guys using for your asterisk boxes? |
19:55.36 | *** join/#asterisk Jedirl (n=asdf@154.Red-217-127-168.staticIP.rima-tde.net) |
19:55.38 | Jedirl | Hi! |
19:55.52 | Jedirl | I'm getting this: "Rejecting call on channel 0/8, span 1" |
19:55.53 | Ciber311 | OMG hi2u2 |
19:55.56 | Ciber311 | :P |
19:56.11 | Jedirl | span 1 is always the first E1 in my TE405 card? |
19:56.32 | Jedirl | that message would make sense if coming from the second E1, not the first one |
19:57.36 | Jedirl | span 1 is *always* the first E1? |
19:58.57 | Dr-Linux | ~soccer |
19:58.59 | jbot | gooooooaaaaaogoalgoalgoggogogogogogogogogoaooaaaaaaaaal... oops sorry, you said soccer, didn't you. |
19:59.47 | sengland | Ok what am I missing here. Under realtime I am setting up iax users. I can authenticate to the database and when I do a realtiem load from the cli the correct data is returned. However when I try and make a call the user is not placed in the correct default context. Any ideas? |
20:04.49 | *** join/#asterisk darby_d (i=darby_t@aapf23.neoplus.adsl.tpnet.pl) |
20:05.46 | asterisk-dud | Hunk #2 failed at 100 while trying to patch asterisk for spandsp |
20:08.21 | Jedirl | huh |
20:08.28 | Jedirl | any new release of spandsp? |
20:09.03 | Jedirl | 0.0.3 still not production, right? |
20:09.13 | Ciber311 | so can anyone in here recommend a reliable voip provider for asterisk? |
20:09.37 | *** join/#asterisk litage (n=nick@203.220.55.70) |
20:09.43 | *** join/#asterisk darby__t (i=darby_t@aapj121.neoplus.adsl.tpnet.pl) |
20:10.56 | justinu|laptop | reliable compared to what? ILEC? |
20:10.56 | grabowski | Ciber311: VoicePulse (http://connect.voicepulse.com) seem reliable |
20:11.10 | grabowski | Ciber311: cheap for outgoing US48/Canada too! |
20:13.13 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@24-107-22-85.dhcp.stls.mo.charter.com) |
20:14.37 | Ciber311 | seem? you actually use them right? |
20:14.40 | Ciber311 | Download Speed: 4769 kbps (596.1 KB/sec transfer rate) |
20:14.40 | Ciber311 | Upload Speed: 356 kbps (44.5 KB/sec transfer rate)Download Speed: 4769 kbps (596.1 KB/sec transfer rate) |
20:14.43 | Ciber311 | Upload Speed: 356 kbps (44.5 KB/sec transfer rate)woops |
20:14.44 | Ciber311 | wtf |
20:14.47 | Ciber311 | Upload Speed: 356 kbps (44.5 KB/sec transfer rate)Download Speed: 4769 kbps (596.1 KB/sec transfer rate) |
20:15.01 | file | what's the matter? |
20:15.05 | NDT | justinu|laptop: I have that 2850 here with me cause I brought it back from the datacenter and was supposed to drive it back down asap...heh well...the old card that was working is actually dead...doesn't get any power...so the shmuck I had put the new card in musta fucked something up.. |
20:15.08 | grabowski | Ciber311: only been using them for a little while |
20:15.11 | Ciber311 | accidently pasted that |
20:15.23 | Dr-Linux | Ciber311? |
20:15.29 | NDT | err fudged something up heh |
20:18.41 | sengland | Ciber311 voicepulse is really not that reliable. |
20:18.54 | justinu|laptop | NDT, :/ |
20:19.13 | Ciber311 | so what do you recommend sengland? |
20:19.33 | sengland | For DID try voxbone |
20:19.35 | grabowski | sengland: have you been having problems? I have not had any of yet |
20:19.40 | sengland | Then have several outbounds |
20:19.42 | *** join/#asterisk darby (i=darby_t@aapi69.neoplus.adsl.tpnet.pl) |
20:20.15 | sengland | grabowski yes. They tend to have alot of circuits busy issues |
20:21.42 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
20:21.45 | charles___ | Hey Guys |
20:21.56 | charles___ | Does anyone uses CISCO 7960 ? |
20:22.25 | sengland | gah fixed the realtime issue, Turn off rtcachefriends. |
20:22.36 | Ciber311 | sengland: where are the prices? :P |
20:23.10 | *** join/#asterisk litage (n=nick@203.220.55.70) |
20:23.54 | ghenry | Is this the way everyone else sets up Call Forwarding? http://www.voip-info.org/wiki-Asterisk+call+forwarding |
20:24.00 | ghenry | thanks, bbl |
20:26.23 | sengland | They only handle DID not termination, I use several carriers for outbound Broadvoice, Junction Networks. |
20:26.44 | sengland | To get DID prices you have to sign up |
20:27.01 | sengland | What area are you looking for? |
20:27.07 | Ciber311 | nyc |
20:27.23 | sengland | area code? |
20:27.27 | Ciber311 | 212 |
20:29.05 | sengland | inbound DID is $9 setup 7.50 monthly unlimited inbound |
20:29.20 | justinu|laptop | the markup on DIDs is insane |
20:29.37 | justinu|laptop | our wholesale voip provider charges us $0.50/mo for DIDs |
20:30.21 | sengland | is there an inbound per minute charge? |
20:30.25 | Ciber311 | someone has to pay for the traffic :P |
20:30.42 | sengland | And how many do you have to buy to get that rate. |
20:31.09 | *** join/#asterisk Assid (n=assid@210.18.143.29) |
20:31.29 | sengland | The nice thing about voxbone is they have did's all over the world and they have a port pooling system |
20:31.45 | justinu|laptop | well, the wholesale arrangement required a big monthly commit |
20:31.47 | grabowski | justinu|laptop: yep its worse then getting IP addresses from a colo/dedicated company. |
20:31.53 | justinu|laptop | something like 20grand a month/ |
20:32.50 | sengland | if you need more ports you can buy them for a region (a region is the US, Canada, South America and Europe) and they dynamically allocate them based upon realtime load. |
20:32.50 | justinu|laptop | and yeah, there's an inbound charge |
20:33.03 | sengland | justinu|laptop thats the kicker |
20:33.04 | justinu|laptop | it varies from something sub penny/min to maybe 2.5/min |
20:33.15 | charles___ | Does anyone uses CISCO 7960 ? |
20:33.15 | justinu|laptop | depending on the ILEC that serves that DID |
20:33.23 | justinu|laptop | there's a whole tier chart |
20:33.28 | grabowski | charles___: yes |
20:33.48 | sengland | See with voxbone its a flat rate. For most small installs they just want a flat rate. |
20:33.58 | justinu|laptop | the most expensive places to get DIDs are the smallest indepedent LECs out there |
20:34.21 | justinu|laptop | then comes former GTE/Contel territories |
20:34.24 | grabowski | sengland: is voxbone just reselling DIDs from everywere or do they have actual connectivity everwhere they offer DIDs? |
20:34.31 | justinu|laptop | the RBOCs are the cheapest |
20:34.51 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
20:34.59 | sengland | Russia wants 250 a month for a single did |
20:35.09 | sengland | I dont offer service in Russia as a result |
20:35.22 | justinu|laptop | 250USD? |
20:35.24 | sengland | Both I think |
20:35.28 | sengland | Yes :) |
20:35.37 | justinu|laptop | that's a joke |
20:36.17 | sengland | If your a Russia company you pay much less........ |
20:37.28 | sengland | I have only had one issue with voxbone and they cleared it up right away. They maintain a 24/7 support staff, which most providers dont do. |
20:38.10 | sengland | And you can add and remove dids in realtime from the website. |
20:38.48 | *** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net) |
20:38.59 | knarfly | hello out there????? |
20:39.07 | justinu|laptop | i had quality issues with voxbone |
20:39.12 | justinu|laptop | they wanted me to change to g729 |
20:39.37 | knarfly | Anyone setup with FreeBSD and IPFW? |
20:40.17 | knarfly | My setup only works halfway and I think I have a natd problem.! |
20:40.32 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
20:41.17 | knarfly | Can anyone offer advice on simple sip setup to FWDNET? |
20:42.40 | grabowski | knarfly: It's on the voip-info.org wiki |
20:43.11 | grabowski | knarfly: http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+FWD |
20:43.27 | docelm0 | is trixter in here? |
20:43.31 | knarfly | Thanks, but I have read through that numerous times and think I'm following it ok - but I can't hear anything. |
20:43.33 | docelm0 | ~seen trixter |
20:43.45 | jbot | trixter <n=trixter@65-165-167-217.du.volcano.net> was last seen on IRC in channel #asterisk, 1d 17h 36m 37s ago, saying: 'chino_: depends on how you connect'. |
20:43.58 | grabowski | knarfly: when you dial a FWD number you can't hear anything? |
20:44.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
20:44.38 | grabowski | knarfly: is everything going though just fine on the asterisk CLI? |
20:44.50 | knarfly | grabowski: I hear a dial tone and can dial a # and hear it ring. Then when it appears someone answers I can't hear anything. |
20:45.29 | grabowski | knarfly: Did you try the 613 echo test? |
20:45.33 | knarfly | the CLI looks good but honestly I wouldn't really know if something bad flew by...but there are no warning messages as I attempt the calls. |
20:45.40 | grabowski | knarfly: could it be a one way audio problem? |
20:46.03 | grabowski | knarfly: is your asterisk system behind a NAT? |
20:46.16 | knarfly | I did try the 613 and got it to ring but could not hear anything. |
20:46.46 | knarfly | Yes, I am behind another FreeBSD server running IPFWE and NATD. |
20:47.10 | knarfly | I've tweaked the rules and the redirects to some degree and gotten some of this working. |
20:47.12 | justinu|laptop | ~nat |
20:47.15 | jbot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
20:47.36 | justinu|laptop | pay attention to that... externip, localnet are important to making it work |
20:47.49 | grabowski | knarfly: I suggest you setup your FWD using IAX (which they offer) |
20:48.08 | grabowski | knarfly: You just need to enable it in your FWD account on one of the options pages on their website. |
20:48.38 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
20:49.02 | knarfly | I'll have to double check if I enabled it. I believe I did. But I went through the setup there too and it's still a no go. |
20:51.25 | L|NUX | can some one tell me how can i ring extensions b/w 7am to 7pm |
20:52.00 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
20:52.21 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
20:56.07 | grabowski | L|NUX: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
20:56.55 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
20:57.42 | grabowski | L|NUX: theres some other pages on the wiki about this type of setup, just search around |
20:58.53 | grabowski | L|NUX: actually this page http://www.voip-info.org/wiki/view/Asterisk+day+night+mode+example is better |
20:59.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
20:59.38 | thermf | hey... does Mithraen hang out around here? |
21:02.00 | L|NUX | ok |
21:02.09 | L|NUX | grabowski : thanks bro |
21:02.13 | L|NUX | i will look into this now |
21:03.23 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
21:04.09 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
21:04.34 | asterisk-dud | has anybody installed fax support for asterisk? |
21:05.33 | grabowski | asterisk-dud: yes but not T.38 |
21:05.37 | *** join/#asterisk arekm (n=arekm@pld-linux/arekm) |
21:06.31 | asterisk-dud | what is t.38, forgive my ignorance? |
21:06.48 | Jedirl | store&forward faxing over IP |
21:07.02 | asterisk-dud | ok, i don't need that |
21:07.09 | grabowski | asterisk-dud: http://www.voip-info.org/wiki/view/T.38 |
21:07.11 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
21:07.11 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
21:07.12 | asterisk-dud | i have asterisk 1.2.9.1 |
21:07.29 | arekm | hi, http://pastebin.ca/63901 , when I call 100 it's going trough Zap/1 while IMO it should go via SIP - any ideas why? 1.2.7.1 here |
21:08.21 | asterisk-dud | i have spandsp 0.2 ref26, with all rx_fax and tx_fax and apps_makefile from same version |
21:08.25 | anthm | never put any extens with _ in them in with any other stuff put it in it's own context and include it |
21:08.30 | [TK]D-Fender | arekm : Change the order of _X. and your include and reload |
21:08.42 | asterisk-dud | i have trouble patching the makefile |
21:09.01 | asterisk-dud | i manually edited it and then asterisk compiled and install successfully |
21:09.10 | asterisk-dud | but the applications are not installed |
21:09.13 | anthm | if you made a [zap] context with that _. then you would be fine cos the includes have sequential pref |
21:09.20 | grabowski | arekm: You should read http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns so you understand why that is happening. |
21:09.20 | asterisk-dud | y is there a problem patching the file? |
21:09.33 | anthm | and _ ext in the same context always beats anything included |
21:10.13 | arekm | grabowski: there is exactly such example there and according to it asterisk should go via sip - it doesn't here |
21:10.14 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
21:10.28 | arekm | [TK]D-Fender: no change after order change |
21:10.31 | asterisk-dud | grabowski: do you have any idea what could be wring? |
21:10.33 | asterisk-dud | wrong? |
21:11.14 | anthm | hmm i guess i'm wasting my virtual breath |
21:11.28 | grabowski | asterisk-dud: I guess you need to load the modules.. |
21:11.46 | anthm | arekm quick go read the detailed explanation i just wrote write in front of you..... |
21:12.21 | asterisk-dud | i have an asterisk@home box, and in the apps folder there are three different files for each app, but with the asterisk box i'm working now this is not the case |
21:12.32 | asterisk-dud | should the three files be present? |
21:12.47 | asterisk-dud | .o, .c, |
21:12.47 | grabowski | arekm: read http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
21:13.34 | grabowski | asterisk-dud: let me check what I did |
21:13.36 | asterisk-dud | i cannot figure out why it's giving me an error when i patch the makefile |
21:13.41 | asterisk-dud | ok |
21:13.42 | asterisk-dud | thanks |
21:13.54 | arekm | ah, now it's clear, thanks |
21:14.07 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
21:16.13 | grabowski | asterisk-dud: just the so file the compiled version |
21:17.00 | asterisk-dud | what version did u use, u remember |
21:18.15 | asterisk-dud | grabowski |
21:19.09 | grabowski | asterisk-dud: I just followed the entry on the wiki. I can't remember exactly what version as it was 5-6 months ago. |
21:19.54 | asterisk-dud | well, i'm trying to follow the wiki but it won't patch |
21:20.35 | grabowski | asterisk-dud: the diff wont patch? |
21:20.50 | asterisk-dud | makefile |
21:21.05 | asterisk-dud | how do i load modules or applications into asterisk |
21:21.12 | charles___ | <PROTECTED> |
21:21.19 | charles___ | Sorry about but I miss a SBN file |
21:21.39 | grabowski | asterisk-dud: Just put them in the /usr/lib/asterisk/modules |
21:21.57 | drray | won |
21:22.16 | asterisk-dud | so if i put the rx_fax.so file into that dir and reload it should work? |
21:22.39 | grabowski | asterisk-dud: you will need to do a shutdown of asterisk |
21:23.00 | grabowski | asterisk-dud: yes, if it does not you could add the module to the modules.conf file but it should take it automaticly |
21:23.34 | charles___ | Hey Guys, anyone uses CISCO 7960 phones ? |
21:25.50 | asterisk-dud | loading modules failed |
21:26.05 | grabowski | asterisk-dud: what was the problem? |
21:26.42 | asterisk-dud | grabowski: undefined symbol: fax_set_phase_d_handler |
21:27.09 | *** join/#asterisk areq (n=areq@pld-linux/areq) |
21:27.28 | charles___ | Hey Guys, anyone uses CISCO 7960 phones ? |
21:27.29 | grabowski | asterisk-dud: try google, it brings up a bunch of results |
21:27.47 | *** join/#asterisk Samoied (n=Samoied@201-25-253-150.fnsce703.dsl.brasiltelecom.net.br) |
21:27.49 | asterisk-dud | if i just recompile asterisk to a newer version it will ovewrite the old one, correct? |
21:28.06 | asterisk-dud | i know, i've tried most of them |
21:28.49 | mishehu | I make packages |
21:28.51 | grabowski | charles___: It looks like no one else that is here has a cisco 7960 why don't you try again later? You don't need to keep repeating yourself. |
21:29.03 | Ciber311 | i make smoothies |
21:29.31 | mishehu | charles___: yes, anyway, you might want to ask the real question and not "who uses XYZ?" |
21:29.38 | grabowski | asterisk-dud: yep, just dont do make samples |
21:29.46 | mishehu | dataja - don't ask to ask, just ask |
21:30.42 | asterisk-dud | grabowski: do u remember which wiki u followed? |
21:31.35 | grabowski | asterisk-dud: http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax and http://www.voip-info.org/wiki/view/Asterisk+fax |
21:32.33 | charles___ | grabowski: I know dude |
21:32.38 | charles___ | mishehu: shhhh |
21:34.16 | knarfly | I want to update Asterisk on my FreeBSD server. If I just compile the new source it won't go in the right directories. Is there an easier way? |
21:34.36 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
21:35.43 | grabowski | asterisk-dud: I gtg, good luck getting it to work. |
21:35.45 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
21:36.56 | veto | Is it possible to number portability a number out of a wireless carrier (Sprint) to voip? Is anyone setup to do this? |
21:38.17 | justinu|laptop | my wholesale voip carrier won't let me do LNPs from wireless phones |
21:39.57 | veto | justinu|laptop, I assume it's a PITA for the destination carrier? |
21:46.24 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
21:51.44 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
21:52.51 | justinu|laptop | veto: i'm not sure what the issue is |
21:53.02 | justinu|laptop | we use a major carrier tho |
21:54.32 | veto | I find very little info on this via google, i guess not many people are trying to do it. |
21:55.09 | justinu|laptop | i would check into the laws that mandated LNP |
21:55.24 | justinu|laptop | maybe there's a loophole that says wireless carriers don't have to participate with wireline carriers? |
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22:06.36 | grabowski | justinu|laptop: Wireless carriers were given a LNP extension (at least here in Canada under the CRTC). I believe its somewhat the same situation under the FCC. |
22:06.45 | *** join/#asterisk qdk (n=qdk@x1-6-00-0f-66-90-6b-48.k441.webspeed.dk) |
22:06.59 | justinu|laptop | the thing is, i know you can do LNPs from wireless to wireless carrier |
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22:07.25 | ghenry | Is this the way everyone else sets up Call Forwarding? http://www.voip-info.org/wiki-Asterisk+call+forwarding |
22:07.26 | ghenry | thanks, bbl |
22:07.31 | Dr-Linux | LNP? |
22:07.38 | Dr-Linux | ~LNP |
22:08.13 | grabowski | justinu|laptop: I think they don't have to fully comply till 2008 or something? |
22:08.25 | [TK]D-Fender | Local Number Portability |
22:08.27 | grabowski | Local number portability |
22:08.28 | grabowski | yea |
22:08.52 | justinu|laptop | grabowski: ic |
22:09.13 | ghenry | anyone do it that way? |
22:10.14 | thermf | does anyone have the version of app_rxfax/app_txfax that makes use of t.38? |
22:11.15 | grabowski | ghenry: What exactly do you need? |
22:11.16 | [TK]D-Fender | ghenry : Similarly, yes |
22:11.53 | ghenry | I just need to set it up for a client, and wanted to check if I was way off by following that guide |
22:12.19 | [TK]D-Fender | ghenry : Think of it as "inspiration" and you should tweak it to fit your needs. |
22:12.27 | file | can you feel the love tonight? |
22:12.33 | ghenry | Ay [TK]D-Fender |
22:12.57 | ghenry | It's just nice to have a verbal on the stuff you read on all the * sites and books ;-) |
22:13.11 | ghenry | To make sure it |
22:13.18 | grabowski | justinu|laptop: Actualy the 2008 deadline might be LNP from Wireless carrier to Wireless carrier may be even longer before we see LNP from Wireless to a CLEC |
22:13.19 | ghenry | 's not out of date etc. |
22:13.44 | grabowski | justinu|laptop: or LEC. |
22:13.57 | asterisk-dud | this wiki is telling me to find a symlink, any suggestions on how to go about doing that? |
22:15.03 | ghenry | asterisk-dud: How so? |
22:16.02 | ghenry | [TK]D-Fender: How many * rollouts does it take to know when to "tweak" or when other guides etc. read wrongly? |
22:16.38 | asterisk-dud | ghenry: http://www.asteriskguru.com/tutorials/spandsp.html troubleshooting step 2 |
22:17.47 | [TK]D-Fender | ghenry : I never used guides for much of anything... I find the best way if to just learn the scope of commands at your disposal and implementations will simply come to you. |
22:18.00 | ghenry | asterisk-dud: symlink --help in that directory, if on GNU/Linux |
22:18.14 | ghenry | [TK]D-Fender: COol. |
22:19.07 | ghenry | asterisk-dud: symlink will fix brokn symlinks |
22:19.44 | asterisk-dud | thnks |
22:19.52 | ghenry | np |
22:20.00 | ghenry | .me off for a cupa |
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22:28.08 | ghenry | how do you hook flash on an analogue phone? |
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22:33.03 | teniar | put in on hook then off hook quickly |
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22:37.40 | knarfly | freevsd_fan: Do you have your Asterisk in full working order? |
22:38.38 | knarfly | I could use some insights into the natd settings |
22:38.39 | Ciber311 | sengland: you there? |
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22:52.04 | ghenry | thanks teniar |
22:52.33 | ghenry | is that the normal way to transfer a call on a modern analogue phone? |
22:52.43 | ghenry | if that makes sense teniar ;-) |
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22:53.11 | [TK]D-Fender | ghenry : Analog has always been the same... |
22:53.34 | teniar | ghenry, is it using an ATA? i know sipuras have some special code to transfer |
22:53.54 | [TK]D-Fender | ghenry : Sipuras work multiple ways. |
22:54.17 | ghenry | this is a bog std analogue phone on a zap chan |
23:02.54 | teniar | ghenry, yeah then just hook flash it |
23:03.05 | teniar | ghenry, as if you were picking up call waiting |
23:03.10 | ghenry | thank. |
23:03.54 | *** join/#asterisk litage (n=nick@203.220.55.70) |
23:04.44 | Dr-Linux | litage: :) |
23:07.19 | Dr-Linux | anybody active and free? :) |
23:09.50 | RoyK | <PROTECTED> |
23:10.27 | Dr-Linux | RoyK: hey there :) |
23:11.19 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
23:11.22 | Dr-Linux | RoyK: i wanna discuss about my new asterisk setup. if you got some time |
23:11.25 | Dr-Linux | :) |
23:11.54 | RoyK | pork? |
23:12.19 | *** join/#asterisk litage (n=nick@203.220.55.70) |
23:13.20 | Dr-Linux | pork? :S |
23:13.32 | Dr-Linux | you mean pig's meat? :S |
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23:36.26 | Jick | Does anyone know if a call queue can be configured such that, when an agent is called by the queue to receive a caller, a set of dialplan instructions can be execution before the call is transferred? |
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23:40.14 | asterisk-dud | when recieving fax i get error" TIFFOpen: :Cannot open |
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23:48.57 | Jick | Does anyone know if call queues can be configured to execute a set of dialplan instructions for the agent before connecting the agent to the next caller? |
23:49.23 | [TK]D-Fender | Jick : Use Local/ agents and put it in there |
23:50.26 | Jick | Hmmm |
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23:58.26 | Jick | D-Fender, okay. Let me see if I can understand how Asterisk would handle that... |
23:59.21 | Jick | So an agent would be configured at extension, say, 105 in the [agents] dialplan context... |