| 00:05.54 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 00:06.12 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 00:06.30 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 00:12.27 | Waverly360 | Guys, what would cause an audio file to play all choppy in asterisk? I can copy the file to another pbx, and it plays fine there, but on this one, it sounds like crap. The odd thing is that the file sounds exactly the same every time I play it. |
| 00:15.52 | Waverly360 | No one's had this problem? |
| 00:17.04 | CrashSys | What's your idle system load? |
| 00:17.21 | Waverly360 | CrashSys: 0.16 |
| 00:17.38 | CrashSys | what kind of system? |
| 00:18.05 | Waverly360 | CentOS 2.6.9-34.EL |
| 00:18.16 | Waverly360 | I have around 40 just like it at my disposal |
| 00:18.49 | CrashSys | what kind of interface card? |
| 00:19.10 | Waverly360 | a Sangoma a102D and an a200 |
| 00:19.33 | CrashSys | Are the other systems configured the say way? |
| 00:19.40 | Waverly360 | Yes. |
| 00:20.41 | CrashSys | Hmmmmm |
| 00:20.59 | Waverly360 | I'm going to copy the file off somewhere again, and make sure it's not corrupted |
| 00:21.47 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 00:23.56 | Waverly360 | CrashSys: The file plays perfectly on my pc. I guess I can try copying it to another one. |
| 00:25.16 | CrashSys | How are you connecting to it? |
| 00:25.31 | Waverly360 | to hear the audio file playing on the pbx? |
| 00:25.35 | CrashSys | yes |
| 00:25.43 | Waverly360 | I'm dialing a did attached to an IVR |
| 00:26.05 | CrashSys | And that DId is on this box? |
| 00:26.31 | CrashSys | Is it the A200 or A102? |
| 00:26.47 | Waverly360 | A102 |
| 00:26.53 | Waverly360 | the a200 isn't being used |
| 00:27.02 | CrashSys | Did you look to see if your having T1 circuit issues? |
| 00:27.14 | Waverly360 | I don't have any red lights or anything like that. |
| 00:27.46 | Waverly360 | The only difference between this pbx and the others, is this is the only one with dual pris. |
| 00:27.52 | Waverly360 | all of the others are single. |
| 00:28.03 | CrashSys | PRi's from different providers? |
| 00:28.08 | Waverly360 | I thought maybe it could have been a timing issue |
| 00:28.11 | Waverly360 | no |
| 00:28.13 | Waverly360 | from the same |
| 00:28.16 | Waverly360 | one full pri |
| 00:28.24 | *** join/#asterisk Mattwj2005 (n=Matt@c-76-17-133-96.hsd1.mn.comcast.net) |
| 00:28.24 | Waverly360 | and another that's shy a few channels |
| 00:28.36 | *** join/#asterisk ldsjohn (n=darksage@exchange.tekworks.com) |
| 00:28.49 | Mattwj2005 | vad finns det for sevardheter? |
| 00:29.00 | CrashSys | does ifconfig on the box show any error's for the if? |
| 00:29.20 | Waverly360 | none |
| 00:29.25 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 00:29.54 | Mattwj2005 | anyone speak Swedish? |
| 00:30.04 | ldsjohn | anyone have any luck getting fax to email working on trixbox? |
| 00:30.05 | Waverly360 | Thing I don't get is why the file breaks the exact same way everytime I listen to it. The audio goes in and out at the same times as it always does |
| 00:30.05 | Mattwj2005 | I know I sure don't :D |
| 00:32.05 | CrashSys | does the machine have a soundcard? |
| 00:32.14 | CrashSys | Trying playing the file from the console |
| 00:32.38 | Waverly360 | It doesn't have a soundcard in it. |
| 00:34.59 | Waverly360 | CrashSys: Hmm..it sounds the same on another pbx too. I think this is a bad conversion. |
| 00:35.04 | CrashSys | did it work and now it's not? |
| 00:35.39 | Waverly360 | CrashSys: It's never worked well..I was just looking into the problem...seems that I was mis-informed though. I was told it sounded great on any other system..just bad on this one..that's not the case. |
| 00:35.54 | Waverly360 | CrashSys: I'll try converting it again. Thanks for the help. |
| 00:35.59 | CrashSys | Yup |
| 00:36.07 | Mattwj2005 | I also thought it would be cool to make my asterisk server ring :) |
| 00:36.11 | Mattwj2005 | *always |
| 00:36.27 | CrashSys | It's a safe practice to assume that what they tell you is bogus :) |
| 00:37.25 | *** join/#asterisk xjagox (n=xjagox@190.8.158.12) |
| 00:37.27 | CrashSys | Specially if it involves pots line issues... |
| 00:37.40 | xjagox | wenas |
| 00:37.44 | ldsjohn | anyone have any luck getting fax to email working on trixbox? |
| 00:38.31 | xjagox | hi |
| 00:40.35 | Waverly360 | CrashSys: Well, it'd just be nice to be able to depend on all of my co-workers. |
| 00:40.44 | *** part/#asterisk xjagox (n=xjagox@190.8.158.12) |
| 00:40.51 | Waverly360 | anyways..night :) |
| 00:41.53 | CrashSys | night |
| 00:47.55 | *** join/#asterisk rlx (n=rlx@c-24-22-183-194.hsd1.mn.comcast.net) |
| 00:48.12 | CrashSys | DAmn... stuck cd-rom drive, and not a paper-clip in sight... |
| 00:49.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:52.07 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 00:55.16 | *** join/#asterisk ifnotwhynot (i=dummy-wi@196.211.34.3) |
| 00:59.36 | *** join/#asterisk perf3ktion (n=perf3kt@adsl-68-77-93-206.dsl.ipltin.ameritech.net) |
| 00:59.48 | perf3ktion | where are the conf files at? |
| 01:00.04 | ifnotwhynot | WHITCH CONF FILES? |
| 01:00.08 | perf3ktion | sip |
| 01:00.20 | ifnotwhynot | sorry about the caps |
| 01:01.03 | ifnotwhynot | is this folder/etc/asterisk/sip.conf |
| 01:01.13 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 01:01.56 | perf3ktion | gotcha thanks |
| 01:02.58 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:03.01 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 01:04.18 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 01:04.36 | ifnotwhynot | hi florz |
| 01:04.49 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.192) |
| 01:05.08 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:05.36 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 01:08.09 | florz | ifnotwhynot: hmm? hi :-) |
| 01:10.25 | ifnotwhynot | whats up? |
| 01:14.37 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:19.41 | *** join/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 01:19.44 | BSD_Tech[laptop] | hey |
| 01:20.06 | BSD_Tech[laptop] | when is 1.4.5 asterisk going to be released |
| 01:24.24 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:29.47 | Qwell | when it's ready |
| 01:30.39 | _VoiceMeUp_COM | probably before 1.4.6 |
| 01:30.46 | Qwell | never know |
| 01:30.52 | Qwell | we might change it, just to shake things up a bit |
| 01:31.04 | rob0 | ooooooh sneaky |
| 01:31.22 | _VoiceMeUp_COM | yeah Asterisk (Rand(1,9),Rand(2,99),Rand(1,10) ) |
| 01:31.30 | _VoiceMeUp_COM | could be fun to debug versions lol |
| 01:32.06 | _VoiceMeUp_COM | how bout we start form 99.0 and go down to 1.0 then back up again |
| 01:32.32 | _VoiceMeUp_COM | or reverse the channels.. like sip is now iax , iax is now local , etc |
| 01:32.48 | Qwell | how about not? |
| 01:35.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:43.39 | *** join/#asterisk comando24 (n=comando2@host207-187-dynamic.56-82-r.retail.telecomitalia.it) |
| 01:44.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:45.17 | *** join/#asterisk PMantis (n=sswitzer@cpe-74-65-53-178.rochester.res.rr.com) |
| 01:45.56 | PMantis | Does anyony have a good gsm or wav saying, "Sorry, the party is not accepting private callers..." ? |
| 01:46.21 | rob0 | tt-monkeys |
| 01:46.25 | PMantis | lol |
| 01:46.38 | PMantis | Wanted something a little more professional :) |
| 01:46.44 | rob0 | an all-purpose solution :) |
| 01:50.27 | *** join/#asterisk flashnet (i=fnet@gateway/tor/x-bb947d5943770df5) |
| 01:52.02 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
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| 01:55.55 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:55.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:06.11 | *** join/#asterisk bonderponder (n=test@201.199.68.150) |
| 02:06.50 | bonderponder | helllo |
| 02:07.12 | bonderponder | ANybody can help to build an IVR very simple ? |
| 02:07.36 | BSD_Tech[laptop] | 50 bucks |
| 02:08.00 | bonderponder | BSD_Tech[laptop]: do you have any other contact , besides here ? |
| 02:09.08 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 02:12.45 | *** join/#asterisk JSabines (n=alancast@189.158.186.76) |
| 02:13.10 | wotcha | bonderponder: have a read through this, it'll tell you how to make an ivr plus much much more, and it's free! ;) --> http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:20.54 | Qwell | umm, what is a "PCIe connector", and why would a power supply have one? |
| 02:21.18 | *** join/#asterisk Taadow (n=john@d154-5-91-133.bchsia.telus.net) |
| 02:21.43 | Taadow | Anyone had any success using a ShoreTel 530 (apparently mgcp based) w/ asterisk? |
| 02:21.45 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 02:21.59 | Aces1Up | does the bot have a function to see when someone was last on? |
| 02:22.19 | Qwell | ~last kram |
| 02:22.53 | Aces1Up | ~last putzz |
| 02:24.23 | *** join/#asterisk daveburr (n=Miranda@h46062829.area1.spcsdns.net) |
| 02:24.54 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 02:24.58 | asterisknerds | <PROTECTED> |
| 02:25.16 | Taadow | Such a nice phone... I suspect I may never get to use it w/ *. :( |
| 02:25.46 | Aces1Up | ~last Putzz |
| 02:26.21 | Taadow | Although, I believe it retrieves it's mgcp config as a file download via ftp on the ShoreGear server sitting on the same network, and asterisk does have mgcp support iirc so could very well be possible. |
| 02:29.16 | daveburr | shoretel sucks :) |
| 02:30.29 | Taadow | Yeah, prolly. I never had a chance to play w/ the system. But this phone is off the hook. |
| 02:30.31 | tzanger | what's the sipura ATA that's 1FXS+1FXO... it's the 941? |
| 02:31.01 | Taadow | No pun intended. |
| 02:31.02 | daveburr | just playin.. shoretel is cool.. expensive cool |
| 02:31.39 | wotcha | 941 is a handset |
| 02:31.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:31.54 | tzanger | bah |
| 02:32.00 | tzanger | it's the 3000 or 3102 |
| 02:32.02 | wotcha | 1FXS+1FXO could be a 3000 or 3102 |
| 02:32.08 | wotcha | yeah |
| 02:32.16 | tzanger | yeah what's the main difference between those two? I've never used them before |
| 02:32.53 | wotcha | i _think_ the difference is that the 3102 is a router as well as everything else, while the 3000 isn't |
| 02:33.04 | wotcha | i've only used the 3102 |
| 02:33.08 | rob0 | IIUC the x1xx's have more CPU power or something ... like they can really handle 2 simultaneous calls. Oh, maybe router? Hmmm. |
| 02:33.36 | rob0 | I have a 2000, and it says it can't handle compression on both lines at once. |
| 02:35.06 | wotcha | the 3102s are pretty nifty for what they are - you can even use them as a voip -> pstn gateway if you're really keen |
| 02:38.30 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 02:41.00 | tzanger | hmm |
| 02:41.02 | tzanger | ok |
| 02:41.04 | tzanger | 3102 it is then |
| 02:41.05 | tzanger | :-) |
| 02:41.06 | tzanger | thanks guys |
| 02:43.44 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 02:44.36 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 02:44.59 | giesen | I'm getting an error when doing make in zaptel 1.4.3 on linux 2.4.26 |
| 02:45.25 | giesen | wcte12xp.c: In function `t1xxp_rbsbits': |
| 02:45.25 | giesen | wcte12xp.c:1115: parse error before `)' |
| 02:45.27 | giesen | ... |
| 02:45.37 | giesen | make[1]: *** [wcte12xp.o] Error 1 |
| 02:45.42 | giesen | anyone have any thoughts? |
| 02:46.24 | *** part/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 02:50.06 | *** part/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 02:55.23 | *** join/#asterisk kimosabe (n=kimosabe@189.175.37.61) |
| 02:55.52 | kimosabe | can some one tellme how the majority of the voice over ip providers set up to get those good rates ? |
| 02:56.39 | Qwell | bulk minutes are incredibly cheap |
| 02:57.47 | kimosabe | qwell and where can i view that type info |
| 02:58.07 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:58.08 | kimosabe | when you say bulk minutes do i recive all the service via coper pair or t-1 |
| 02:58.25 | Qwell | well, it wouldn't be voip if it were over copper... |
| 03:01.00 | kimosabe | qwell yes true but if i wanted to become a provider i would recive via ip or via coper then convert to ip ¿ |
| 03:01.40 | Qwell | you won't be able to push a whole heck of a lot of volume over copper |
| 03:01.48 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 03:02.56 | kimosabe | qwell and where can i purchase some thing of this sort for say leasing 20 lines out |
| 03:05.06 | flenders | kimosabe: what exactly do you want to do? |
| 03:05.43 | kimosabe | i want to become a voice over ip provider |
| 03:05.50 | kimosabe | i have some know how |
| 03:05.54 | kimosabe | i live in mexico |
| 03:07.31 | giesen | kimosabe: typically you'd buy DS3s |
| 03:07.35 | giesen | or DS0s |
| 03:07.57 | flenders | kimosabe: you should terminate the calls somehow |
| 03:08.06 | giesen | Qwell: it's still voip if he terminates the calls on a T1 |
| 03:08.07 | flenders | even getting a bunch of PRIs |
| 03:08.16 | giesen | it's gotta get on the PSTN somehow |
| 03:08.34 | *** join/#asterisk CuriosCat (i=stian@mack.bigrig.org) |
| 03:08.36 | CuriosCat | Hi all |
| 03:10.41 | kimosabe | ok where can i purchase something like tis in the usa ? |
| 03:10.52 | giesen | Qwell: you can push over a million minutes a month on a full PRI |
| 03:10.56 | flenders | any telco |
| 03:11.02 | giesen | kimosabe: any telco will sell you a PRI |
| 03:11.04 | kimosabe | i live on a border town im 3 miles from border and i have 16 megabits international crosing via wifi |
| 03:11.08 | giesen | they're VERY common |
| 03:11.22 | giesen | uh |
| 03:11.25 | Qwell | no, no, no, no, no, do not carry voip over wifi |
| 03:11.25 | kimosabe | so i ask for a pri ? |
| 03:11.26 | giesen | voip over wifi = bad. |
| 03:11.28 | Qwell | just no |
| 03:11.45 | giesen | kimosabe: you cant have a pri delivered via wifi |
| 03:11.49 | kimosabe | the links are rather stable |
| 03:11.55 | giesen | it doesnt matter |
| 03:11.56 | Qwell | and full of jitter |
| 03:11.59 | kimosabe | no ican put the box in the us |
| 03:12.06 | kimosabe | then carry the devices via wifi |
| 03:12.13 | giesen | wifi is about the worst thing you can do to voip |
| 03:12.20 | kimosabe | ok |
| 03:12.25 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:13.01 | kimosabe | giesen but i have 32 voice chanels over wifi here and theyve been up for 3 yrs since asterisk came out |
| 03:15.55 | giesen | that's pretty suprising |
| 03:16.03 | giesen | how are they delivered |
| 03:16.04 | giesen | sip? |
| 03:16.07 | giesen | iax? |
| 03:16.30 | kimosabe | sip |
| 03:16.43 | kimosabe | the link is 8 megabits and 4 megabits in bad weather |
| 03:16.53 | kimosabe | giesen can i pm u |
| 03:17.14 | giesen | sorry |
| 03:17.20 | giesen | too busy dealing with other stuff right now :/ |
| 03:17.27 | giesen | like tryig to get zaptel to compile |
| 03:18.39 | kimosabe | oki thanks man do you know the more less cost for a ds1 ? |
| 03:19.18 | Qwell | Who's ready for the nub question of the day? |
| 03:19.26 | Qwell | What ever happened to T2? |
| 03:29.16 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 03:30.21 | `Sean | Guys anyone around here use a PAP2NA, i was wondeirng is there a way to set outgoing callerid? |
| 03:30.41 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
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| 04:18.19 | *** part/#asterisk slingr (n=san@66.29.73.5) |
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| 04:32.13 | perf3ktion | anyone around |
| 04:32.21 | perf3ktion | can't get incoming calls to my asterisk box |
| 04:32.30 | Trevor_b | what do you see on the console? |
| 04:38.00 | perf3ktion | I see alot, you want everything? |
| 04:38.09 | perf3ktion | or you looking for something specific? |
| 04:38.14 | perf3ktion | I see the outside number |
| 04:39.11 | Trevor_b | back in a few, what is the number being dialed, then show me the dialplan entry where you have it defined. Ill be back in 5 or so, if noone else chimes in ill give it a once over. |
| 04:42.08 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 04:42.35 | *** join/#asterisk suma (n=suma@63.83.99.163) |
| 04:43.07 | suma | i want to have unlimited outgoing calls in the US, which provider is cheap and best with asterisk with less hassles |
| 04:43.54 | Trevor_b | back |
| 04:44.06 | Trevor_b | damn rpc |
| 04:44.17 | grey | If I want to get an FXO card just to tinker with Asterisk and a few softphones for now, Where is a good place to get them in Canada? I saw an auction on e-bay to preorder one... but preorder stuff on ebay is sometimes a little shakey |
| 04:46.51 | grey | here's the auction on e-bay, anyone know if it's legit, or if this particular card is supported under OpenBSD preferably: http://cgi.ebay.ca/Authentic-X100P-SE-FXO-PCI-for-Digium-Asterisk-VoIP-PBX_W0QQitemZ130122058261QQihZ003QQcategoryZ61839QQrdZ1QQcmdZViewItem |
| 04:47.19 | suma | authentic is crap |
| 04:47.48 | Trevor_b | x100p.com |
| 04:47.51 | suma | i have experienced professional x100p someone was selling, has zero problems |
| 04:48.34 | Trevor_b | very nice cards, works with no modification to zaptel. At least for me they have been flawless. |
| 04:49.10 | Trevor_b | haha, that ebay sale is for one of that companies cards. |
| 04:49.25 | grey | heh yeah |
| 04:49.37 | grey | but I wasn't sure if it was legit or not, it's a preorder for a manufacturing run |
| 04:49.42 | Trevor_b | they do really good discounts on multiple orders. |
| 04:49.43 | Hmmhesays | the new practice area is good |
| 04:49.53 | grey | I might send them a message and ask them to mail me from a company e-mail address, |
| 04:50.08 | Trevor_b | i ordered direct from them on their x100p.com site |
| 04:50.35 | grey | oh |
| 04:50.36 | perf3ktion | so you'll have to be gentle trevor |
| 04:50.42 | grey | hey their site is selling them for the same amount, durr :P |
| 04:50.43 | perf3ktion | I've gotten this far |
| 04:50.48 | Trevor_b | hehe yeah ;) |
| 04:50.54 | Trevor_b | 8 dollars ship, but discount on 3 or more. |
| 04:50.57 | grey | alright thats excellent, |
| 04:51.02 | perf3ktion | but I've edited everything through the gui |
| 04:51.09 | grey | Do you know if they are supported under openbsd as well as linux? |
| 04:51.12 | [TK]D-Fender | Screw the X100p, get an SPA-3102. |
| 04:51.23 | [TK]D-Fender | Under $100 CAD and you get 1 FXO & 1 FXS |
| 04:51.35 | perf3ktion | I have the ssh up though and can edit the config if need be |
| 04:51.47 | Trevor_b | as far as i know openbsd doesnt do ANY zaptel |
| 04:52.21 | Trevor_b | asterisk works, but just voip. Or thats my understanding of the current affairs, unless zaptel hardware and drivers are built into bsd and I just been missing that part. |
| 04:52.42 | Trevor_b | which is usually the case for their hardware, just not sure. |
| 04:52.43 | perf3ktion | so what do you need trevor? |
| 04:53.07 | perf3ktion | the number is what is supplied from broadvoice |
| 04:53.14 | Trevor_b | i use linux + asterisk + zaptel. if BSD works, then just plug it in and see if its accesible |
| 04:53.19 | Trevor_b | oh |
| 04:53.25 | Trevor_b | the inbound |
| 04:53.33 | perf3ktion | yeah |
| 04:53.39 | Trevor_b | the number and the dialplan that shows the incoming call rule. |
| 04:54.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:54.14 | Trevor_b | you have a incoming call rule, is it for the specific number, or for "all unmatched"? |
| 04:54.21 | perf3ktion | I haven't edited the dialplan explicitly |
| 04:54.25 | perf3ktion | is that the problem? |
| 04:54.26 | Trevor_b | in the gui |
| 04:54.34 | perf3ktion | its for all unmatched |
| 04:54.34 | Trevor_b | under incoming calls |
| 04:54.47 | Trevor_b | pastebin your console for the incoming call. |
| 04:55.34 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 04:55.55 | CrashSys | Anyone know approximately how much TCP traffic a ulaw sip call makes? |
| 04:55.57 | CrashSys | 90kb? |
| 04:56.46 | [TK]D-Fender | 85 |
| 04:56.57 | Trevor_b | i think 64kb was the stock number to work from and then add any overhead? I think voip-info.org has a good workup of all of the codecs |
| 04:57.00 | [TK]D-Fender | CrashSys, and that'd be *UDP* |
| 04:57.08 | CrashSys | err UDP |
| 04:57.46 | Trevor_b | DUH, not thinking overhead would be TCP, guess its getting late. |
| 04:58.24 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 04:59.06 | CrashSys | Hmmm... man isn't working... wonder what I forgot to install :D |
| 05:00.03 | perf3ktion | http://pastebin.ca/560666 |
| 05:01.37 | Trevor_b | whats the verbosity set to? |
| 05:01.37 | CrashSys | Wanpipe cant find a file/directory in it's own tree... |
| 05:02.34 | grey | that SPA3102, does that interface with asterisk?, |
| 05:02.43 | perf3ktion | 3 |
| 05:06.43 | [TK]D-Fender | grey, Yes, its an ATA + SIP gateway |
| 05:07.01 | [TK]D-Fender | grey, So you can use your analog line AND an analog phone (seperate of each other) |
| 05:08.02 | CrashSys | Anyone using the wanpip 3.1.0 drivers? Any stability issues? |
| 05:10.10 | Trevor_b | ill let you know tomorrow or so ;) |
| 05:10.12 | grey | is there an example of setting it up with asterisk somewhere? or of how it behaves? |
| 05:10.20 | Trevor_b | got their 104d model to setup. |
| 05:10.30 | CrashSys | heh |
| 05:10.47 | CrashSys | 2.4.3-9 is missing files in the distro... guess i'll try down a version |
| 05:11.08 | CrashSys | or I'm on crack |
| 05:11.11 | CrashSys | prolly on crack |
| 05:11.23 | Trevor_b | hehe |
| 05:12.54 | *** join/#asterisk Micc (n=dotirc@c-71-231-254-232.hsd1.or.comcast.net) |
| 05:13.08 | [TK]D-Fender | grey, www.voxilla.com , check the forums. Setup is much like an ITSP. |
| 05:13.28 | [TK]D-Fender | grey, SIP is SIP. |
| 05:13.39 | [TK]D-Fender | grey, I've owned an SPA-3000 & a 3102. |
| 05:13.40 | Micc | is there a way I can run an asterisk command in the background from my extensions.conf say like exten => 100,1,Ices(blah.xml) |
| 05:14.11 | Micc | currently when I try it, the next line in extensions.conf does not execute. |
| 05:14.11 | [TK]D-Fender | Micc, no. |
| 05:14.45 | Micc | [TK]D-Fender, so my only option is to use a .call file? |
| 05:14.50 | grey | yeaahhh, I don't even know what ITSP means, I'm just intrigued and wanted to mess around more with it for as little cost as possible, |
| 05:15.33 | [TK]D-Fender | grey, Ok, to play around with * you only need a computer to install it on, and it'd be NICE to have a soft-phone running on another computer or two. |
| 05:15.47 | [TK]D-Fender | grey : anything else is commiting to hardware. |
| 05:16.02 | [TK]D-Fender | grey, * works, untold thousands of users can't be wrong. |
| 05:16.08 | [TK]D-Fender | ~itsp |
| 05:16.20 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 05:16.20 | grey | yeah, I don't mind dropping $30 on one card, I'm not thrilled to drop $80 a peice of hardware, but I would if it's going to be substantially better for me, |
| 05:16.59 | [TK]D-Fender | grey, What do you really want * to do for you? |
| 05:17.04 | Trevor_b | grey: junker pc and that x100p card is all you need, just like TK said. a seperate PC for softphone is very nice as well. |
| 05:18.00 | [TK]D-Fender | Don't need ANY special hardware just to play with *. |
| 05:18.21 | [TK]D-Fender | just use softphones tog et a feel for what its like to call in/out. |
| 05:18.27 | Trevor_b | true true, he had asked about a zaptel card on the cheap earlier, assumign he wants to answer house calls or something. |
| 05:18.44 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-77-93-206.dsl.ipltin.ameritech.net) |
| 05:18.52 | CrashSys | that sucks... spent all that time compiling 2.6 kernel and this stupid nic is giving me framing error's... :( |
| 05:18.54 | [TK]D-Fender | grey, so again, in the big picture, what are you envisioning using * for? |
| 05:19.58 | Trevor_b | CrashSys: Trying something special, or you really need to compile OS instead of using CentOS or something? |
| 05:21.00 | CrashSys | I just like to strip all the stuff I dont use out of the kernels and make 'em monolithic except for the zap/sangoma stuff... |
| 05:22.00 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 05:23.17 | grey | sorry, got pulled away for a sec |
| 05:23.41 | grey | I've got the junker PC, everyone in my house has a computer that can run a softphone, I'm interested in getting a menu system etc. setup because my dad runs a business out of our house, and I'm sick of answering his calls :P |
| 05:24.12 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 05:24.28 | grey | If possible, I'd like to split between personal calls and business calls based on an voice menu, as well as provide all the other features available through *, (Hold, internet calls etc.) |
| 05:24.29 | nowork | hi, how can I check my voicemail remotely dial in?? |
| 05:24.52 | [TK]D-Fender | grey, Ok, well the short answer is, it will work. Taek your pick from an X100P or SPA-3102. |
| 05:25.04 | [TK]D-Fender | nowork, Using whatever YOU setup in your dialplan to allow it. |
| 05:25.21 | grey | I'd like to experiment with it a lot more first, because it's a fair chunk of change to get everyone in the house switched over to ip phones, |
| 05:25.31 | grey | and no one likes using a computer for their phone, |
| 05:25.52 | [TK]D-Fender | grey, Get ATA's then. |
| 05:26.07 | [TK]D-Fender | grey, then you can reuse all your existing analog phones. |
| 05:26.34 | grey | don't I have to get one for each phone then? |
| 05:26.44 | [TK]D-Fender | grey, $70 for an SPA-2102 (2 FXS) letting you use 2 phones independently of each other |
| 05:26.55 | [TK]D-Fender | for $70 USD |
| 05:27.01 | grey | It's not like our existing phones hold a lot of sentimental value, they are cheap phones for the most part, but IP phones are not cheap, |
| 05:27.06 | [TK]D-Fender | so a conversion cost of $35 / ea |
| 05:27.16 | grey | ok, but then how does the wiring for that work? |
| 05:27.24 | [TK]D-Fender | grey, Depends on your idea of "cheap" of course. |
| 05:27.46 | [TK]D-Fender | grey, ATA is just a little box with ethernet on one side, and phone jacks on the other. |
| 05:27.54 | grey | hmm |
| 05:27.58 | [TK]D-Fender | grey, thats what Vonage, etc give you. |
| 05:28.05 | grey | it means I have to have each phone plugged directly into an FXS though right? |
| 05:28.15 | grey | which is going to be a pain given how spread out our phones are, |
| 05:28.34 | [TK]D-Fender | grey, there IS no perfect pretty picture for your scenario I'm sure. |
| 05:28.44 | grey | yeah I figured |
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| 05:28.53 | grey | hmm |
| 05:28.58 | [TK]D-Fender | grey, Esp if you want them to all act INDEPENDENT of each other |
| 05:29.04 | Trevor_b | perf3kt: kicked for spammyness |
| 05:29.07 | grey | that would be nice, but I was just going to ask, |
| 05:29.14 | grey | is there a way to have calls pass through asterisk in some situations? |
| 05:29.29 | [TK]D-Fender | grey, "pass through" how/why/meaning what? |
| 05:29.37 | grey | ie have a voice menu, and for business calls ring only the office phone (Which can easily be an IP phone), and for everything else ring the main house line |
| 05:29.49 | grey | or have the option to go to personal voice mail, |
| 05:30.31 | [TK]D-Fender | well it won't be "ring the house line" anymore. * will be answering pretty much EVERYTHING... it'd be more a question of what would happen afterwards. |
| 05:30.48 | [TK]D-Fender | you can do whatever you FELL LIKE with any call that enteres your system. |
| 05:30.56 | grey | can a single FXS ring the house? |
| 05:31.08 | [TK]D-Fender | It's tuesday night and raining, and its your Aunt calling? DIRECT TO VOICEMAIL! |
| 05:31.12 | grey | that might involve some fancy wiring into the telephone panel I'd imagine, |
| 05:31.35 | [TK]D-Fender | grey, It can... but then that means that all those phones are SHARING that port making it one big "party line". |
| 05:31.45 | [TK]D-Fender | grey, They are clearly NOT independent. |
| 05:31.46 | grey | so, exactly like everything already is? |
| 05:31.56 | [TK]D-Fender | grey, Correct. AKA no change. |
| 05:32.01 | grey | thats fine |
| 05:32.13 | grey | I mainly want to filter out office calls directly to the office, |
| 05:32.21 | [TK]D-Fender | what would it matter if * answers the call only... TO MAKE THEM ALL RING ANYWAYS. |
| 05:32.33 | grey | because I want it to have the option |
| 05:32.37 | grey | at a voice menu, |
| 05:32.46 | grey | either ring them all, or ring only the office phone |
| 05:32.50 | [TK]D-Fender | grey, that means there IS no difference. |
| 05:33.01 | [TK]D-Fender | Thent he office phone would need to be seperate from the rest. |
| 05:33.20 | grey | "Press One to talk to the family, Press Two to talk to the business", one rings the house phones, two rings the office phone (which can be an SIP phone) |
| 05:33.32 | [TK]D-Fender | grey, Yeah, that could work. |
| 05:33.36 | grey | ok cool... |
| 05:34.01 | [TK]D-Fender | grey, So you'd need to do some nifty rewiring, but doable. |
| 05:34.20 | grey | so for that setup, I'd need an FXO, an FXS and an ip phone, and to wire the FXS into the incoming line in the house, |
| 05:34.43 | [TK]D-Fender | grey, Not necessarying an "ip phone", but a seperate device in *'s eyes. |
| 05:34.48 | grey | yeah |
| 05:34.53 | grey | so a phone connected to another FXS device, |
| 05:35.04 | grey | or a software phone, or an SIP phone, etc. |
| 05:35.07 | [TK]D-Fender | grey, You could use an X100P + SPA-2102 (+/- $100 total) |
| 05:35.13 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
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| 05:36.17 | [TK]D-Fender | grey, or an SPA-3102 + SPA-2102 (+/- $150) and have an extra port free as well. thing is that in the case of a power failure, the SPA-3102 wil bridge the FXS & FXO ports so you don't "go dark" |
| 05:36.35 | [TK]D-Fender | grey, Because when your server goes with an X100 all calls are DOA |
| 05:36.53 | grey | oh that would be super nice, |
| 05:37.47 | grey | hmm, ebay says about $160 for those two, not especially bad |
| 05:38.22 | [TK]D-Fender | grey, http://www.telephonydepot.com/product_p/105-054-212.htm |
| 05:38.36 | [TK]D-Fender | grey, http://www.telephonydepot.com/product_p/105-054-312.htm |
| 05:38.48 | [TK]D-Fender | 142$ new |
| 05:39.13 | grey | will they ship to Canada? |
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| 05:40.16 | [TK]D-Fender | grey, Yup. |
| 05:40.30 | nowork | TK: thank you |
| 05:40.34 | *** part/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 05:42.14 | [TK]D-Fender | http://www.canadianvoipstore.com/index.php?manufacturers_id=36 |
| 05:42.35 | [TK]D-Fender | grey, There's VoipSupply's canadian division. Decent pricing surprisingly. |
| 05:42.45 | grey | sweet :) |
| 05:43.30 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 05:43.35 | [TK]D-Fender | THIS is going to rock... http://www.canadianvoipstore.com/product_info.php?manufacturers_id=36&products_id=2912 |
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| 05:44.21 | grey | o_O |
| 05:44.51 | grey | but it requires rewiring every phone line in the house to it, |
| 05:45.00 | [TK]D-Fender | grey, who said for YOU? |
| 05:45.03 | grey | lol |
| 05:45.08 | grey | no I didn't actually mean for me, |
| 05:45.24 | grey | and yeah, I guess you can put it straight into the patch panel or whatever it's called |
| 05:45.39 | [TK]D-Fender | grey, think about older companies wanting more core functionality (menus, etc) vs their old PBX using old wiring, etc. |
| 05:45.59 | *** join/#asterisk pipwerk (i=pip@ringbreak.dnd.utwente.nl) |
| 05:45.59 | [TK]D-Fender | grey, 8 FXS at that price is VERY nice. |
| 05:46.04 | grey | thats true |
| 05:46.12 | [TK]D-Fender | grey, in1 unit with RJ11 + Amphenol |
| 05:46.54 | *** part/#asterisk Mattwj2005 (n=Matt@c-76-17-133-96.hsd1.mn.comcast.net) |
| 05:46.55 | grey | Whats the performance of * like in general? from what I've seen they don't recommend more than 4 active calls on a 400mhz+/- PC, (Which is what my router currently is, I could move it to a tbird 1ghz if I needed to) |
| 05:47.05 | grey | that doesn't seem like a good scale, |
| 05:47.13 | [TK]D-Fender | grey, 1ghz and you're set |
| 05:47.33 | [TK]D-Fender | grey, jsut a little breathing room is recommended. |
| 05:47.53 | grey | how many lines/channels/whatever (I'm not very clear on some of that terminology) should that be able to handle? |
| 05:57.25 | [TK]D-Fender | grey, technically plenty |
| 05:57.26 | [TK]D-Fender | grey, if it doesn't have to translate too many compressed codec calls. |
| 05:57.26 | [TK]D-Fender | grey, depends on your outside VoIP usage. |
| 05:57.26 | [TK]D-Fender | grey, inside is irrelevent (esp as you will technically have 2 phones). |
| 05:57.26 | grey | well in my place yes |
| 05:57.26 | grey | as well, this means in theory I can probably recieve phone calls to my house anywhere I have my laptop logged into an SIP phone eh? |
| 05:57.26 | [TK]D-Fender | grey, anywhere you have IP to your home. |
| 05:57.26 | [TK]D-Fender | grey, local, over the internet, whatever |
| 05:57.26 | grey | cool |
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| 05:57.27 | grey | that would probably be pretty annoying to those in my house who want to use the phone eh? :P |
| 05:57.27 | grey | Sorry guy's, I'm tying up the phone line... FROM ANOTHER House/City/Country :P |
| 05:57.27 | [TK]D-Fender | grey, think of the LD saving that could give you... |
| 05:57.27 | perf3kt | anyone help with incoming calls not coming in |
| 05:57.27 | perf3kt | just set the port forwarding |
| 05:57.27 | [TK]D-Fender | ~sipnat |
| 05:57.29 | jbot | methinks sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:57.29 | [TK]D-Fender | perf3kt, you need a PILE of settings in sip.conf for NAT to work. Read up. |
| 05:57.29 | [TK]D-Fender | ok, checkout time for me here.... back tomorrow. |
| 05:57.30 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 05:57.30 | [TK]D-Fender | well.... anyways.. jbot'll wake up eventually... |
| 05:57.30 | [TK]D-Fender | later.... |
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| 06:02.21 | perf3kt | you have to use cleartext passwords? |
| 06:04.24 | DrAk0 | is 1.4 rdy for production? |
| 06:05.46 | tzafrir_laptop | perf3kt, in iax: no |
| 06:16.35 | perf3kt | unfourtunately using sip |
| 06:16.42 | perf3kt | set the register |
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| 06:48.26 | snuffy22 | got a question.. |
| 06:48.37 | snuffy22 | how would i set a variable of a parent channel |
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| 07:03.34 | Aces1Up | ~last Putzz |
| 07:05.41 | k31th | guys, any one recommend a card I need to allow 6 pstn connections for inbound outbound routing on a Asterisk box. I also need a place to buy it im in the UK. |
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| 07:09.10 | k31th | is the OpenVox A400P any good? |
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| 07:31.28 | tzafrir_laptop | k31th, the A400P can give you up to 4 channels. It is basically a clone of the Digium TDM400P card (and should use the same driver) |
| 07:32.54 | tzafrir_laptop | You can use e.g. the Digium TDM800P for up to 8 channels. |
| 07:35.44 | k31th | I need to make 6 incoming or 6 outgoing calls they have 6 ISDN lines basically. |
| 07:38.55 | EvilGreen | k31th: 6 ISDN lines will give you 6 in AND 6 out, as soon as each ISDN line is 2B+D |
| 07:39.50 | k31th | really, well they can only make or rec 6 calls total. |
| 07:40.35 | EvilGreen | k31th: probably they have 6 analog lines coming out of NT1 box |
| 07:40.50 | EvilGreen | ... and have only 3 ISDN lines |
| 07:40.51 | k31th | this is in the UK not sure if it differs, our infrastructure is normally a grade below the rest of the world here :p |
| 07:41.17 | k31th | thats probably the jist of it tbh. |
| 07:41.31 | tzafrir_laptop | if you have BRI, what would you want to use it as analog? |
| 07:41.36 | tzafrir_laptop | digital i more fun |
| 07:42.35 | tzafrir_laptop | s/i/is/ |
| 07:42.42 | tzafrir_laptop | doh |
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| 07:55.24 | creativx | omfg ¤#"_URQWur |
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| 08:03.04 | walhala | hi |
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| 08:17.55 | festr__ | hello, is it possible to active jitter buffer between sip and misdn? |
| 08:18.48 | creativx | now this is fun, the damn ip10s ignores the ringing tone volume |
| 08:18.49 | creativx | LOUD |
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| 08:47.29 | angryuser | god day |
| 08:49.14 | angryuser | i have some dtmf tons problems from misdn calls, my ivr from some callers does not react on choices, any ideas (i have mISDN latest, all packages latest |
| 09:00.56 | angryuser | can somebody help me on isdn and dtmf working ? point me into direction, where is dtmf setting are set for misdn ? |
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| 10:15.51 | *** join/#asterisk TimothyP (n=timothyp@116.252-243-81.adsl-static.isp.belgacom.be) |
| 10:16.36 | TimothyP | Hello, I have a macro where I playback some message and then I want to do different things based on wether the user presses 1 or 2, I added 2 extentions to that macro 1,1,.... and 2,1,.... but that does not work |
| 10:19.58 | *** join/#asterisk Op3r (n=op3r@125.212.50.108) |
| 10:20.49 | Op3r | can anyone tell me how to point an external number to an extension? ie exten => 1234,Dial(sip/18005551212) ? |
| 10:21.38 | TimothyP | 0p3r what do you mean exactly |
| 10:21.42 | creativx | Op3r: you mean you want to dial local ext 1234 and forward it to number sip/18005551212 |
| 10:21.50 | Op3r | creativx: thats correct |
| 10:22.04 | Op3r | I mean I want to point a pstn number to an extension |
| 10:22.05 | TimothyP | but the external numer probably isn't a sip number right? |
| 10:22.14 | TimothyP | oh that's the other way around |
| 10:22.14 | Op3r | TimothyP: thats correct :( |
| 10:22.21 | creativx | get the direction correct here mate |
| 10:22.23 | TimothyP | so from PSTN to SIP |
| 10:22.28 | creativx | are we talking inbound or outbound :-) |
| 10:22.33 | Op3r | creativx: outbound |
| 10:22.37 | Op3r | I mean |
| 10:22.41 | TimothyP | so from SIP TO PSTN |
| 10:22.44 | Op3r | yes |
| 10:22.47 | TimothyP | ok |
| 10:22.48 | Op3r | thats correct |
| 10:23.19 | Op3r | any way to do it? |
| 10:23.33 | TimothyP | then it depends on what you're using, in my case it's Dial(mISDN/g:myoutsidelines/0003343434,60,r) |
| 10:23.37 | TimothyP | where 00033... is the number |
| 10:24.04 | TimothyP | in my case I'm using mISDN , but might be different for you |
| 10:24.05 | penguinFunk | Op3r: are you using an FXO gateway to analogue lines? |
| 10:24.10 | Op3r | just sip |
| 10:24.16 | TimothyP | .... |
| 10:24.33 | TimothyP | so how do you get from SIP to PSTN? |
| 10:24.34 | penguinFunk | what do you mean by external number then? |
| 10:24.44 | Op3r | so Dial(SIP/provider/18005551212,60,tTo) ? |
| 10:24.54 | Op3r | penguinFunk: a pstn number |
| 10:25.28 | penguinFunk | so you can't be using just sip then |
| 10:25.36 | TimothyP | sip connects pphones on the internet/network, not over classlic landlines |
| 10:25.38 | penguinFunk | what hardware are you using to connect to pstn? |
| 10:25.45 | Op3r | Im using voip |
| 10:25.50 | penguinFunk | duh |
| 10:25.57 | penguinFunk | voip is not hardware |
| 10:26.01 | penguinFunk | it is a protocol |
| 10:26.03 | Op3r | nada zip |
| 10:26.09 | Op3r | :( |
| 10:26.18 | matsk | VoIP ia a acronym |
| 10:26.22 | penguinFunk | then how are you analogue lines physically connected to you network? |
| 10:26.28 | matsk | SIP is a protocol |
| 10:26.29 | TimothyP | then you can't make calls to a normal pstn/isdn number |
| 10:26.38 | Op3r | errr? |
| 10:26.41 | Op3r | ok |
| 10:27.15 | TimothyP | you need something which connects your asterisk server to the normal phone lines, so the first thing to check is probably if you have a cable running from your server to the phone socket in the wall |
| 10:27.20 | TimothyP | if so, check what card it's connected to |
| 10:27.24 | penguinFunk | are your digital calls magically drifting through the air to the pstn network |
| 10:27.59 | penguinFunk | and getting modularised/demodularised by god |
| 10:28.06 | Op3r | heres what I wanted to do. My cellphone number is for example a us toll free number 8005551212. I am using teliax as my provider. I am registering to teliax using sip. now I want to point my extension 1234 to my 8005551212. |
| 10:29.11 | TimothyP | aah :) |
| 10:29.14 | TimothyP | now we're talking |
| 10:29.26 | TimothyP | you're using a SIP gateway to connect to your mobile phone |
| 10:29.50 | Op3r | I was thinking to put in my extensions.conf like exten => 1234,1,Dial(SIP/provider/18005551212) |
| 10:29.56 | Op3r | is that how its done? |
| 10:30.02 | TimothyP | so I'm guessing , wild guess here Dial(800555121@whateveryourprovider.com) |
| 10:30.04 | Op3r | TimothyP: yes thats correcnt |
| 10:30.44 | Op3r | cos I put a Macro on top of default like TRUNKSIP=SIP/teliax |
| 10:31.12 | Op3r | or not a macro but |
| 10:31.15 | Op3r | err |
| 10:31.18 | TimothyP | variable |
| 10:31.21 | Op3r | yeah |
| 10:31.23 | TimothyP | :) |
| 10:31.24 | Op3r | variable |
| 10:31.25 | Op3r | hahaha |
| 10:31.29 | Op3r | so basically |
| 10:31.33 | Op3r | is that how its done? |
| 10:32.01 | TimothyP | Dial($TRUNKSIP/thenumber) sorry for bad syntax, but my macbook pro isn't really fit to type special chars, so for exact syntax you'll have to look online |
| 10:32.26 | Op3r | oh |
| 10:32.47 | TimothyP | Anyway, I'll be back... got to get some food :d |
| 10:32.48 | creativx | Dial(${TRUNKSIP}/number) |
| 10:32.54 | creativx | brackets!! |
| 10:33.05 | creativx | hmm food |
| 10:33.06 | TimothyP | can't type brackets with my macbook |
| 10:33.07 | creativx | sounds like an idea. |
| 10:33.16 | creativx | macs.... |
| 10:33.17 | TimothyP | it's an azerty keyboard and they left out loads of keys |
| 10:33.19 | TimothyP | yeah |
| 10:33.20 | TimothyP | it's a test :d |
| 10:33.24 | TimothyP | my desktop is Ubuntu :d |
| 10:33.26 | creativx | not only does it lack a mouse key |
| 10:33.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:33.32 | TimothyP | oh mine doesn't :d |
| 10:33.32 | creativx | but also brackets! |
| 10:33.33 | creativx | :> |
| 10:33.37 | creativx | oic |
| 10:33.44 | TimothyP | it checks where my fingers are on the one mouse buttons and acts accordingly :d |
| 10:33.50 | TimothyP | even use that mouse on windows |
| 10:33.51 | TimothyP | :d |
| 10:33.53 | TimothyP | it rocks |
| 10:34.06 | TimothyP | and has a 360 degree mouse ball/wheel/thingy |
| 10:34.08 | TimothyP | :p |
| 10:34.12 | creativx | keep trying |
| 10:34.13 | creativx | its still a mac |
| 10:34.14 | creativx | ;) |
| 10:34.15 | TimothyP | :p |
| 10:34.17 | TimothyP | true :) |
| 10:34.27 | TimothyP | I prefer ubuntu but oh well :d |
| 10:34.30 | TimothyP | part of the job :d |
| 10:34.39 | TimothyP | anyway g2g now, I need some help later, but I'll be back :d |
| 10:35.04 | Op3r | so is this correct |
| 10:35.04 | Op3r | exten => 1234,1,Dial(${TRUNKSIP}/18005551212) |
| 10:35.04 | Op3r | exten => 1234,2,Hangup() |
| 10:35.09 | Op3r | ? |
| 10:35.19 | TimothyP | should be |
| 10:35.23 | TimothyP | you'll have to try |
| 10:35.35 | TimothyP | and while you do make sure you like asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 10:35.36 | TimothyP | :d |
| 10:35.39 | TimothyP | that should give you a clue |
| 10:36.10 | TimothyP | if it's any help, I haven't been able to get outgoing calls to FWD working either, only incoming calls |
| 10:37.03 | Op3r | TimothyP: its working |
| 10:37.04 | Op3r | :) |
| 10:37.06 | Op3r | :D |
| 10:37.20 | TimothyP | :d |
| 10:37.21 | TimothyP | sweet :d |
| 10:37.38 | Op3r | Executing Dial("SIP/2111-007b8db0", "SIP/provider/18005551212") in new stack |
| 10:37.38 | Op3r | <PROTECTED> |
| 10:38.06 | Op3r | sweet indeed |
| 10:38.07 | Op3r | thanks |
| 10:38.14 | Op3r | now on mac |
| 10:38.18 | Op3r | I wanted macbookpro |
| 10:38.19 | Op3r | :( |
| 10:38.19 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:38.29 | TimothyP | that's what I'm using :) |
| 10:38.32 | TimothyP | but to be honest |
| 10:38.38 | TimothyP | you're better off buying an iMac |
| 10:38.49 | TimothyP | macos really isn't a notebook os |
| 10:39.02 | TimothyP | I ordered a Sony Vayo 11" notebook instead :d |
| 10:39.09 | Op3r | but you cant take imac on the road and like me I have to bring my own machine when going to work |
| 10:39.26 | penguinFunk | imac's are for women |
| 10:40.03 | TimothyP | :d |
| 10:40.08 | TimothyP | I'll tell my collegue that, hang on :d |
| 10:40.26 | TimothyP | lol anyway, brb now :) |
| 10:41.10 | penguinFunk | lol |
| 10:41.48 | penguinFunk | http://www.anekdotov.net/pic/montazh2/imac_for_women.jpg |
| 10:41.52 | Hymie | hmm, it seems that when I use a call group "Dial(Zap/g1/number%SIP/place)" it stops ringing the SIP line when it finds a valid zap line? ;( |
| 10:42.46 | penguinFunk | do you mean when you get an incoming call you want to dial a call group? |
| 10:43.29 | Hymie | no, just dialing out |
| 10:44.09 | Hymie | a zap call group will find the first free zap channel in that group, and use it to call out... |
| 10:44.54 | Hymie | I can use dial(zap/1/number&SIP/place) and thawt rings both lines and only stops when one answers |
| 10:45.12 | Hymie | but, dial(zap/g1/number&sip/place) stops when it finds the first free zap line |
| 10:45.17 | Hymie | and only rings that |
| 10:47.07 | penguinFunk | why would you want to use 2 analogue lines to ring the same place? |
| 10:47.48 | penguinFunk | incoming i can understand |
| 10:47.56 | penguinFunk | but not outgoing, that doesn't make sense |
| 10:48.02 | Hymie | eh? |
| 10:48.11 | Hymie | I'm using one analog line |
| 10:48.14 | Hymie | not two |
| 10:48.26 | penguinFunk | so it is a sip call group, not a zap call group |
| 10:48.27 | penguinFunk | ? |
| 10:48.34 | Hymie | no |
| 10:48.44 | creativx | sip zap bham |
| 10:48.47 | Hymie | it's a zap group, not a call group |
| 10:48.56 | Hymie | it finds the first free zap channel |
| 10:48.58 | Hymie | and uses it |
| 10:49.01 | Hymie | the sip is another call |
| 10:49.03 | Hymie | to a local sip |
| 10:49.28 | penguinFunk | we have 1 zap device per analogue line |
| 10:49.36 | Hymie | so do I |
| 10:49.36 | penguinFunk | what are your other zap devices assigned to then? |
| 10:49.42 | penguinFunk | oh |
| 10:49.43 | penguinFunk | wtf |
| 10:49.54 | Hymie | zap/g1 not zap/1 |
| 10:49.56 | penguinFunk | <Hymie> I'm using one analog line |
| 10:50.06 | Hymie | yes, to call out |
| 10:50.20 | Hymie | I have many of them, but you assumed thawt I was calling the same number multiple times |
| 10:50.25 | Hymie | with multiple zaps |
| 10:50.28 | Hymie | all at once |
| 10:50.30 | Hymie | I'm not |
| 10:50.38 | Hymie | I'm using one analog line at once |
| 10:51.11 | Hymie | that's what groups do |
| 10:51.17 | Hymie | they find the first free zap line |
| 10:51.18 | Hymie | and use it |
| 10:51.25 | Hymie | zap/g1, not zap/1 or zap/4 |
| 10:51.38 | Hymie | zap/g1 = zap/1 + zap/4 + zap/whatever in a gropup |
| 10:51.44 | penguinFunk | yes |
| 10:52.00 | penguinFunk | i still dont think i understand your problem |
| 10:52.13 | Hymie | when I use a call group "Dial(Zap/g1/number%SIP/place)" it stops ringing the SIP line when it finds a valid zap line? ;( |
| 10:52.16 | penguinFunk | judging from the silence in the channel, neither does anyone else |
| 10:52.33 | Hymie | if I dial zap/1&SIP/place |
| 10:52.43 | Hymie | it won't stop calling the SIP line unless the xap line is answered |
| 10:52.44 | penguinFunk | why is that a problem? |
| 10:52.57 | penguinFunk | once you have found one that is available that you can use, surely you dont need to worry about other ones? |
| 10:53.20 | penguinFunk | call gets bridged to the zap, off you go |
| 10:53.25 | Hymie | if I use zap/g1&sip/place, it stops when it finds the first free zap line, NOT when it answers, as & denotes |
| 10:53.27 | TimothyP | so while I wait for the oven to warm up.... I Background(somefile) and if the user presses 1 a specific macros should be called, if he pressed 2 a different macro etc.;... this is already inside a macro, how can I solve this? |
| 10:53.43 | Hymie | & means that a line has been answered, not bridged |
| 10:54.31 | Hymie | anyhow, it's very non-intuitive, and it makes groups fairly useless in a dialplan, if you want to do multiple destination dialouts with & |
| 10:54.43 | penguinFunk | i see |
| 10:55.03 | Hymie | there's no other easy method to find a free zap channel, while dialing with a & |
| 10:55.14 | penguinFunk | why not just have multiple outgoing dial commands, trying each channel in order? |
| 10:55.16 | Hymie | sure, I could write a routine, but a bit of a pITA |
| 10:55.33 | Hymie | because, I need SIP at the same time |
| 10:55.36 | penguinFunk | if 1, fails goto2, if 2 fails goto 3 etc |
| 10:55.40 | penguinFunk | ah |
| 10:55.58 | Hymie | it will always succeed, I assume, since the SIP will always succeed |
| 10:56.05 | Hymie | so I'll rarely get a zap |
| 10:59.00 | TimothyP | Hymie doesn't zap have groups like mISDN? |
| 10:59.10 | Hymie | er |
| 10:59.12 | Hymie | I'm using groups |
| 10:59.17 | TimothyP | I createad a group for my mISDN channels and then I dial mISDN/g:myoutsidelines/ |
| 10:59.22 | TimothyP | so it finds a free channel automatically |
| 10:59.52 | Hymie | TimothyP: yes, I know... I think you just read the last two lines ;) That's not the problem... |
| 11:00.13 | TimothyP | np :d |
| 11:00.28 | Hymie | when I use a call group "Dial(Zap/g1/number%SIP/place)" it stops ringing the SIP line when it finds a valid zap line |
| 11:03.24 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:03.35 | puzzled | hi |
| 11:04.47 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 11:06.06 | *** join/#asterisk gardo (n=gardo@124.107.38.214) |
| 11:06.45 | penguinFunk | hi puzzled, who do you have to butter up to get an xs4all.nl sub domain then? |
| 11:08.52 | puzzled | penguinFunk: just get their adsl service. they first give you your-ip-address.xs4all.nl but you can change it in the service section |
| 11:09.46 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 11:12.42 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:13.11 | Zeeek | why would audio not pass for echo() unless I playback() a file right before the echo? |
| 11:13.48 | puzzled | playback answers than chan first. echo probably does not. so first answer the chan before you do echo |
| 11:14.18 | Zeeek | puzzled yes I already answer() |
| 11:15.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:16.27 | penguinFunk | puzzled: nice, they give you static ip by default then too? :] |
| 11:16.58 | puzzled | penguinFunk: yup |
| 11:17.16 | penguinFunk | how much is dsl in .nl? |
| 11:17.21 | penguinFunk | we get ripped off over here |
| 11:17.52 | penguinFunk | £38/month for 8Mbps Max...most people won't get higher than 6Mbps here |
| 11:18.31 | penguinFunk | really bad contention ratio's too |
| 11:18.31 | Zeeek | anyone hear of any NAT issues particular to 1.4 ? |
| 11:18.41 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 11:18.53 | puzzled | penguinFunk: I pay about €27 for 6Mbps/1Mbps (not sure about the 1Mbps) |
| 11:19.06 | puzzled | Zeeek: there are always NAT issues :) |
| 11:19.17 | thevoke | :> |
| 11:19.18 | Zeeek | the identical phones worked fine on 1.2 |
| 11:19.34 | KpoH | hey people |
| 11:19.58 | penguinFunk | 27 euro? omg thats like £18 |
| 11:20.10 | penguinFunk | nice |
| 11:20.43 | puzzled | penguinFunk: I have had it since they launched the service years ago so I'm still on an old cheap plan while over the years they kept increasing the speed gratis |
| 11:21.24 | penguinFunk | do you live near the city? |
| 11:21.27 | KpoH | I recently build asterisk with non standart --prefix and installed him. Now I'm trying to build addons, but "make" complain |
| 11:21.30 | KpoH | app_addon_sql_mysql.c:15:22: asterisk.h: No such file or directory |
| 11:21.31 | KpoH | In file included from /usr/include/asterisk/ |
| 11:21.35 | KpoH | and so on |
| 11:21.48 | puzzled | penguinFunk: I live in one of the bigger cities in .nl |
| 11:22.03 | penguinFunk | rotterdam or utrecht? |
| 11:22.06 | penguinFunk | i been to both |
| 11:22.11 | *** join/#asterisk kakarot (n=kakarot@16.Red-81-33-10.staticIP.rima-tde.net) |
| 11:22.11 | puzzled | The Hague |
| 11:22.13 | penguinFunk | ahh |
| 11:22.36 | Zeeek | we pay about €25 for 2Meg/256k down |
| 11:22.37 | KpoH | i've try ./configure --includedir=/my/dir/with/aster-header-files |
| 11:22.39 | penguinFunk | i went to tiesto in concert in arnhem week before last, was amazing |
| 11:22.45 | penguinFunk | :] |
| 11:22.47 | puzzled | KpoH: it obviously can't find asterisk.h. Did you check in the addons Makefile where it is looking for it? |
| 11:22.57 | penguinFunk | Zeeek: where you from? |
| 11:23.00 | Zeeek | Paris |
| 11:23.15 | Zeeek | The 100M fibre is coming soon |
| 11:23.16 | KpoH | puzzled: i forgot to say, addons 1.4.1 |
| 11:23.31 | penguinFunk | Zeeek: waw, that be v nice |
| 11:23.42 | Zeeek | not expensive either |
| 11:23.43 | puzzled | Zeeek: beautiful city, and with 100M fibre even better |
| 11:23.44 | KpoH | whereis no path to include in Makefile |
| 11:24.05 | puzzled | KpoH: I only work with 1.2 so can't help you there |
| 11:25.28 | Zeeek | hmmmmm it may not even be a NAT issue, FWD is UNREACHABLE too |
| 11:25.29 | penguinFunk | aren't you worried about people bending? |
| 11:25.31 | penguinFunk | :P |
| 11:25.40 | *** join/#asterisk Corydon76-home (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 11:25.40 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:25.59 | puzzled | Zeeek: define "not expensive" |
| 11:26.11 | penguinFunk | you can put a micro-bend in the cable, leaking a small amount of light through, enough to sniff all your data |
| 11:26.22 | Zeeek | under €80 for 100Mbits symettric is what I call inexpensive |
| 11:26.47 | puzzled | indeed |
| 11:27.01 | puzzled | I would get that service if they offered it here |
| 11:27.05 | stoffell | hm, would it be possible, upon entering a queue, to give the caller a few rings before giving the caller a message and then music for the rest of the duration? |
| 11:27.26 | creativx | ringing(); queue() |
| 11:27.52 | puzzled | Zeeek: if it's reliable €80 is cheap cause you can save on hosted websites and host them on your own box |
| 11:27.56 | stoffell | creativx, meaning the caller enters the queue only after X rings? |
| 11:28.29 | Zeeek | puzzled I would never host websites on my own box |
| 11:28.35 | Zeeek | I would use asterisk though |
| 11:29.11 | Zeeek | ok, wait, EVERYONE SIP is unreachable on 1.4. This looks like a different problem |
| 11:29.18 | puzzled | Zeeek: if they are important business critical ones I off course agree. |
| 11:29.24 | creativx | stoffell: yeah.. inside queue() you dont have much options |
| 11:29.31 | creativx | as to playing tones or whatelse |
| 11:29.51 | stoffell | creativx, okay, thanks, i'll have a go with this, seems like the best way indeed.. |
| 11:32.12 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 11:34.22 | Zeeek | Ok, let's think about this: all SIP peers are unreachable on this box. WHy? |
| 11:34.26 | alin` | can somebody tell me if asterisk supports timer? (I mean if it supports the RFC 4028, Session Timer) |
| 11:34.32 | Zeeek | Let's look at port forwarding |
| 11:35.45 | Zeeek | quilify is just an OPTIONS message snet via UDP or TCP, correct? |
| 11:36.15 | puzzled | yes think it's an OPTION via udp (in case of asterisk) |
| 11:37.07 | Zeeek | so I have 5060 forwarded to the asterisk local ip |
| 11:39.11 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:46.28 | alin` | nobody can explain me? |
| 11:47.07 | puzzled | alin`: not sure but I don't recall it ever being mentioned so probably not |
| 11:48.23 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 11:48.40 | alin` | I could find however in its sources channels/chan_sip.c:/* RFC4028: SIP Session Timers */ |
| 11:48.50 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:49.10 | puzzled | alin`: ah ok, better ask in #asterisk-dev when they are awake or ask on the asterisk-dev mailing list |
| 11:49.40 | alin` | puzzled: thanks |
| 11:49.54 | alin` | /* RFC4028: SIP Session Timers */ |
| 11:49.54 | alin` | { SIP_OPT_TIMER,NOT_SUPPORTED,"timer" }, |
| 11:50.10 | alin` | in fact from the sources, I can understand that it is not supported |
| 11:55.53 | *** part/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 11:58.37 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:59.14 | DrukenLPY | exactly how many enum registries are there out there? and which one is the best? |
| 11:59.55 | puzzled | DrukenLPY: afaik there are 2 big ones. check voip-info.org for more info |
| 12:03.56 | Zeeek | If I remove qualify, state change to Unmonitored and the phone is reachable |
| 12:05.40 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 12:05.56 | *** part/#asterisk alin` (n=user@193.226.173.50) |
| 12:06.08 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 12:06.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:16.31 | penguinFunk | what is the latency between sip clients and asterisk? |
| 12:16.46 | Zeeek | on what network? |
| 12:16.56 | tzanger | penguinFunk: what's the velocity of an unladen swallow? |
| 12:16.58 | penguinFunk | if it is over a certain threshold it will count the sip clients as unreachable |
| 12:17.10 | penguinFunk | lol tzanger |
| 12:17.12 | tzanger | your question is meaningless |
| 12:17.16 | penguinFunk | i meant for Zeeek's problem |
| 12:17.20 | tzanger | heh |
| 12:17.29 | penguinFunk | thats the whole point of qualify |
| 12:17.31 | tzanger | ah that makes a lot more sense |
| 12:17.35 | tzanger | my apologies |
| 12:17.45 | Zeeek | I can't tell since qualify isn't working |
| 12:17.54 | penguinFunk | if latency > threshold then [client = unreachable] |
| 12:18.06 | penguinFunk | because if the call was allowed to proceed with a very large latency |
| 12:18.13 | penguinFunk | it would be very poor quality |
| 12:18.23 | Zeeek | true but somehow I don't think that's the problem |
| 12:18.32 | penguinFunk | have you tested latency ? |
| 12:18.33 | Zeeek | I can try a high qualify though |
| 12:19.49 | Zeeek | the latency bewteen the two routers is 60ms |
| 12:19.51 | tzanger | you can qualify high? |
| 12:20.25 | [TK]D-Fender | tzanger: American or European? :) |
| 12:20.39 | tzanger | [TK]D-Fender: I don't know... wait AUUUUUGHHHHH |
| 12:21.07 | [TK]D-Fender | *thud* |
| 12:21.42 | penguinFunk | zap! |
| 12:21.49 | tzanger | SIP |
| 12:21.52 | Zeeek | The various latencies are all under 200ms |
| 12:22.45 | Zeeek | what is odd is that if qualify is off, the phone seem to be reachable |
| 12:23.09 | Zeeek | I've seen this with specific peers in th past but this seems to be the case with all peers |
| 12:23.27 | Zeeek | the same ones are fine on the other asterisk box with the same accounts/peer settings |
| 12:24.03 | Zeeek | SO THE QUESTION IS: why is qualify suddenly not working on this new 1.4.4 box ? |
| 12:24.36 | Zeeek | exit |
| 12:24.39 | Zeeek | not |
| 12:29.20 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:37.18 | *** join/#asterisk kova (n=kova@tech.quentris.com) |
| 12:37.35 | kova | Hi everyone |
| 12:38.34 | kova | anyone here knows how to get AMR support into * |
| 12:39.36 | EvilGreen | kova: you have Nokia phone ? |
| 12:39.48 | kova | yes |
| 12:39.56 | EvilGreen | me too |
| 12:40.23 | kova | and I would like to connect to another system that talks h263 and amr-nb |
| 12:40.45 | kova | it's a content system for 3G video |
| 12:41.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:41.09 | EvilGreen | I have no idea about AMR formal status |
| 12:41.41 | kova | I understood that it's patented ... so not free |
| 12:41.58 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 12:42.02 | kova | but you can download C code at 3gpp.org |
| 12:42.41 | coppice | its a darned good codec, though |
| 12:43.09 | kova | so I was hoping someone already added it to asterisk |
| 12:43.33 | kova | not in the SVN though, because of licensing issues |
| 12:44.39 | kova | anyone? |
| 12:45.08 | EvilGreen | kova: I suppose you should ask in asterisk-dev |
| 12:49.14 | *** join/#asterisk tanacsdavid (n=david@office.axpnet.com) |
| 12:49.22 | tanacsdavid | Hellello! |
| 12:51.56 | penguinFunk | Zeeek: if you turn qualify off then all works fine? and when you turn it back on nothing is reachable? |
| 12:52.41 | tanacsdavid | Can You help me in my problem? I'm using trixbox, I reached the semi-good state. The extensions can call eachother, outcalling works great, but when I try to call the number from an outer station, I always get the 'busy' signal. I don't use any PSTN/ISDN cards, just pure VoIP. My voip server is sip.e.fone.hu |
| 12:53.36 | mocker | tanacsdavid: You might try #trixbox, it's harder to debug because of all the macro's it uses. |
| 12:54.00 | tanacsdavid | mocker: I tried, but there are only 15 virtual people... |
| 12:54.03 | s0ck | tanacsdavid: sip show registry ? |
| 12:54.29 | Zeeek | who is running 1.4.4 ? |
| 12:54.48 | Zeeek | penguinFunk that seems to be what I'm seeing, yes |
| 12:55.18 | Zeeek | there are only two possible causes: local router and 1.4.4 settings I don't know about |
| 12:55.20 | tanacsdavid | s0ck: is that a command? I do not have the sip script/program/anything. |
| 12:55.52 | s0ck | that's a command you can issue from the cli to show whether you've actually registered/authed with your sip provider |
| 12:56.12 | s0ck | you would normally ssh to the box and issue an 'asterisk -r' |
| 12:56.40 | s0ck | i do believe you can issue cli commands from the http interface too, though |
| 12:57.33 | *** join/#asterisk DrukenHME (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:58.49 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 12:58.56 | *** join/#asterisk perf3kt (i=perf3kt@149.166.34.169) |
| 12:59.22 | tanacsdavid | s0ck: asterisk1*CLI> sip |
| 12:59.22 | tanacsdavid | No such command 'sip' (type 'help' for help) |
| 12:59.36 | [TK]D-Fender | ~trixbox |
| 12:59.38 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 12:59.50 | perf3kt | ~pbx |
| 12:59.51 | jbot | i guess pbx is a Private Branch eXchange |
| 12:59.59 | perf3kt | ~asterisk |
| 13:00.01 | jbot | hmm... asterisk is the best free PBX in the world |
| 13:00.08 | s0ck | tanacsdavid: you need the full command :) |
| 13:00.12 | tanacsdavid | ~me should find an other job... |
| 13:00.14 | jbot | moi? |
| 13:00.20 | perf3kt | lol |
| 13:00.25 | perf3kt | ~gui |
| 13:00.27 | jbot | hmm... gui is (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 13:00.41 | perf3kt | ~cli |
| 13:00.43 | jbot | [cli] a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
| 13:00.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:00.54 | perf3kt | cool |
| 13:01.14 | tzanger | ~seen a one eyed, one horned, flying purple people eater |
| 13:01.33 | jbot | i haven't seen 'a one eyed, one horned, flying purple people eater', tzanger |
| 13:01.52 | tanacsdavid | s0ck: good. :) Made a typo, too. The correct command sais this: |
| 13:01.54 | tanacsdavid | sip.e.fone.hu:5060 16004357 105 Registered |
| 13:03.04 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:03.07 | s0ck | any context specified in the trunk/peer details? |
| 13:03.44 | EvilGreen_ | tanacsdavid: now type "sip show peers" and finf the name for that peer |
| 13:04.05 | s0ck | you might want to try 'set verbose 99' and make a call, see where it's going wrong |
| 13:04.09 | tanacsdavid | 57test/16004357 213.253.219.151 5060 Unmonitored |
| 13:04.10 | s0ck | failing that, sip debug |
| 13:04.14 | tanacsdavid | 20/20 88.151.100.144 D N 5060 OK (22 ms) |
| 13:04.22 | tanacsdavid | 20 is my phone |
| 13:04.33 | EvilGreen_ | ok, sip show peer 57test |
| 13:04.56 | EvilGreen_ | and check the context name |
| 13:05.00 | puzzled | coppice: can you please tell me which version of spandsp/rxfax/txfax is best to use with asterisk 1.2? |
| 13:05.08 | tanacsdavid | it's a little too long to paste is here |
| 13:05.15 | tanacsdavid | so.. |
| 13:05.21 | tanacsdavid | Context : from-sip-external |
| 13:05.22 | EvilGreen_ | just check the context name |
| 13:06.16 | EvilGreen_ | tanacsdavid: no open extensions.conf and check [from-sip-external] |
| 13:06.22 | [TK]D-Fender | EvilGreen : He's on Trixbox, hence FreePBX..... |
| 13:06.27 | EvilGreen_ | now ;) |
| 13:06.32 | [TK]D-Fender | ~wglwat |
| 13:06.34 | jbot | rumour has it, wglwat is well, good luck with all that |
| 13:06.36 | EvilGreen_ | oops |
| 13:07.05 | s0ck | tanacsdavid: see if you have any [contexts] specified in the peer details |
| 13:07.11 | s0ck | they prolly do not need to be there |
| 13:07.12 | EvilGreen_ | ok, then log& debug |
| 13:07.42 | tanacsdavid | I only have there exten lines |
| 13:07.44 | coppice | puzzled: spandsp-0.0.2 is antique, but I haven't adapted the app_txfax and app_rxfax for spandsp-0.0.4. 0.0.4 is wwwaaaayyyy ahead of 0.0.2 |
| 13:07.53 | waKKu | talking about fax.. which do u prefers rxfax or hylafax ? |
| 13:08.12 | EvilGreen_ | s0ck: he answered that - [from-sip-external] |
| 13:08.58 | puzzled | coppice: thanks. so I understand: from a spandsp p.o.v it's best to use 0.0.4 with 1.2. it's just that app_{r,t}xfax need to be updated to work with spandsp 0.0.4 (and 1.2)? |
| 13:09.30 | tanacsdavid | These are my lines: |
| 13:09.32 | tanacsdavid | exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN}) |
| 13:09.32 | tanacsdavid | exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})}) |
| 13:09.32 | tanacsdavid | exten => _.,n,Goto(s,1) |
| 13:09.42 | tanacsdavid | exten => s,1,Ringing |
| 13:09.42 | tanacsdavid | exten => s,n,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1) |
| 13:09.42 | tanacsdavid | exten => s,n,Set(TIMEOUT(absolute)=15) |
| 13:09.46 | [TK]D-Fender | tanacsdavid: STOP |
| 13:09.48 | coppice | yep. there is not much to do, but I just don't run * any more |
| 13:09.49 | tanacsdavid | ok |
| 13:09.54 | [TK]D-Fender | tanacsdavid: to NO spam that junk here |
| 13:10.00 | puzzled | tanacsdavid: use a pastebin |
| 13:10.00 | [TK]D-Fender | do NOT* |
| 13:10.14 | [TK]D-Fender | tanacsdavid: And even then don't bother... this is FREEPBX |
| 13:10.34 | [TK]D-Fender | tanacsdavid: there is NO point touching extensions.conf at all for this. |
| 13:10.48 | tanacsdavid | puzzled: what is pastebin? |
| 13:11.07 | [TK]D-Fender | ~pb |
| 13:11.09 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 13:11.29 | [TK]D-Fender | tanacsdavid: And again, your extensions.conf file will not do much for you. |
| 13:12.51 | [TK]D-Fender | tanacsdavid: "but when I try to call the number from an outer station, I always get the 'busy' signal." <- call "the number" (which?) from an "outer station" (meaning what & where exactly? And called how? |
| 13:13.02 | tanacsdavid | I just wanted a quick solution to the asterisk system... I see, I made a big mistake... |
| 13:13.07 | Zeeek | [TK]D-Fender asterisk doesn't work, please help :) |
| 13:13.16 | Zeeek | I have phones |
| 13:13.21 | [TK]D-Fender | Zeeek: www.drphil.com :D |
| 13:13.41 | Zeeek | have you messed with 1.4 ? |
| 13:13.45 | [TK]D-Fender | tanacsdavid: Well FreePBX isn't supported here and Better info yeilds better answers. |
| 13:13.52 | [TK]D-Fender | Zeeek: I use it at home. |
| 13:14.09 | [TK]D-Fender | Zeeek: But hardly enough to truely differentiate it from 1.2 |
| 13:14.12 | *** join/#asterisk jmacz (n=jmacz@190.24.96.186) |
| 13:14.21 | EvilGreen_ | guys, have that message - what does it mean? |
| 13:14.21 | EvilGreen_ | DEBUG[12691]: chan_iax2.c:4852 raw_hangup: Raw Hangup x.x.x.x:4569, src=2, dst=16 |
| 13:14.31 | Zeeek | [TK]D-Fender I'm having an unusual problem - nothing allows qualify. Both phones and peers with be UNREACHABLE if qualify is used |
| 13:14.38 | tanacsdavid | [TK]D-Fender: the number is the registrated number with the sip provider. When I call out from the asterisk, this number is sent as CID. The "out" is my mobile phone |
| 13:14.44 | Zeeek | the same phones and peers as used with 1.2 box |
| 13:15.06 | Zeeek | I'm comparing every setting but hav"nt found differences yet |
| 13:15.17 | [TK]D-Fender | tanacsdavid: So when you use a cell phone to call the DID provided to you by your ITSP * doesw not seem to see the call coming in? |
| 13:15.33 | [TK]D-Fender | doesn't* |
| 13:15.34 | Zeeek | if I remove qualify, the peers work,n but are unmonitored |
| 13:15.38 | tanacsdavid | Now I see, that these frontends are not so good when trying to mak an asterisk pbx. Could You recommend me a good howto? |
| 13:15.50 | [TK]D-Fender | tanacsdavid: |
| 13:15.52 | [TK]D-Fender | ~book |
| 13:15.54 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:15.54 | [TK]D-Fender | ~wikis |
| 13:15.56 | jbot | i guess wikis is http://www.voip-info.org |
| 13:16.12 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:16.24 | [TK]D-Fender | tanacsdavid: Here's a decent qucik start as well that I contributed to : http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 13:16.25 | tanacsdavid | [TK]D-Fender: I cannot see anything coming in... |
| 13:16.37 | Zeeek | also this new one: http://www.the-asterisk-book.com/ |
| 13:16.43 | tanacsdavid | Thank You, I check those resources |
| 13:17.17 | Zeeek | ~dasbook http://www.das-asterisk-buch.de/ |
| 13:17.25 | Zeeek | oops |
| 13:18.17 | tanacsdavid | Zeeek: My german is maybe even poorer than my english, but I'll try that one, too. |
| 13:18.29 | Zeeek | there is an english version |
| 13:18.32 | tanacsdavid | O, I see, that's the same |
| 13:18.46 | tanacsdavid | maybe a good hungarian translation? ::) |
| 13:18.57 | Zeeek | the one you'll do in a few months, yes |
| 13:18.57 | tanacsdavid | Or I should do that? :) |
| 13:19.00 | Zeeek | yes |
| 13:19.13 | Zeeek | then you can tell me why qualify isn't working out |
| 13:20.04 | [TK]D-Fender | Zeeek: By then.... it WILL ;) |
| 13:20.11 | tanacsdavid | :) |
| 13:20.14 | Zeeek | also unusual: Answer;Echo; no sound. Answer; playback(duh); echo works normally |
| 13:20.29 | [TK]D-Fender | tanacsdavid: Quick guess here : Your * is behind NAT, correct? |
| 13:20.31 | puzzled | Zeeek: wasn't there something with qualify requiring dynamic hosts? are you perhaps using static hosts and qualify? |
| 13:20.40 | Zeeek | why does echo not work without playback =even after answer? |
| 13:21.07 | Zeeek | puzzled all of the config is the same with the 1.2 box. qualify works with 99% of all peers. |
| 13:21.14 | puzzled | ok |
| 13:21.20 | [TK]D-Fender | ~wifisip |
| 13:21.22 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 13:21.30 | Zeeek | in fact I don't know why it doesn't work with one peer only |
| 13:21.56 | [TK]D-Fender | Zeeek: Not SUPPORTED by their switch perhaps? |
| 13:22.06 | Zeeek | deltathree |
| 13:22.09 | tanacsdavid | Only a quick question more, and I'll shut my mouth... How can I watch the asterisk log in real time? Only tail -f, or is there another way? |
| 13:22.37 | [TK]D-Fender | tanacsdavid: "asterisk -r" |
| 13:22.44 | tanacsdavid | aha |
| 13:22.47 | tanacsdavid | Ok, thank You. |
| 13:23.04 | mocker | [TK]D-Fender: Took your advice from yesterday on the analog adapter. |
| 13:23.08 | Zeeek | FWIW qualify works fine with IAX2 |
| 13:23.09 | mocker | Wish me luck. :) |
| 13:23.21 | tanacsdavid | So my call doesn't get to the asterisk. It sais nothing, just blinks that dumb cursor... |
| 13:23.38 | Zeeek | tanacsdavid use debug to see what's happening |
| 13:23.47 | creativx | asterisk -rvvvvvvvvvvvvvvvvvvT |
| 13:23.50 | creativx | :) |
| 13:23.53 | creativx | veeeeeeeeee |
| 13:25.36 | tanacsdavid | thank You for the support! |
| 13:25.56 | tanacsdavid | I cannot promise, I won't be back... :) |
| 13:26.16 | creativx | I guarantee you, that you will. |
| 13:26.20 | tanacsdavid | :) |
| 13:26.24 | *** join/#asterisk kakarot (n=kakarot@16.Red-81-33-10.staticIP.rima-tde.net) |
| 13:26.33 | Zeeek | free for a limited time only |
| 13:26.39 | tanacsdavid | that should be true |
| 13:26.47 | tanacsdavid | Zeeek: You're not funny. :)) |
| 13:26.49 | Zeeek | then $30 for the forst 90 days, $3000 thereafter |
| 13:26.54 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 13:27.36 | tanacsdavid | Have a nice day/night, whereever You are! |
| 13:27.39 | tanacsdavid | Bye! |
| 13:27.45 | Zeeek | ty |
| 13:27.54 | *** part/#asterisk tanacsdavid (n=david@office.axpnet.com) |
| 13:29.05 | Zeeek | let's see now: what does qualify do? It sends an options msg to the peer? THe peer needs to respond. THe response maybe is going to the wrong place? |
| 13:29.27 | Zeeek | why would calls work and not qualify? |
| 13:29.30 | puzzled | perhaps it's a routing issue |
| 13:29.45 | Zeeek | yes??? |
| 13:29.56 | Zeeek | please elaborate with examples |
| 13:30.10 | mutilator | anyone wanna but a lightly used te110p? $200 |
| 13:30.20 | Zeeek | this happens with phones nearby and peers on other continents |
| 13:30.21 | puzzled | Zeeek: if the remote peer does not have a proper route back than you will never receive an answer to the qualify |
| 13:30.39 | Zeeek | puzzled but then how does my call proceed? |
| 13:30.40 | [TK]D-Fender | mutilator: ebay it |
| 13:30.56 | puzzled | Zeeek: ah good question. guess it's not a routing issue then |
| 13:31.00 | Zeeek | heh |
| 13:31.01 | EvilGreen | Zeeek: peer may not support OPTIONS ? |
| 13:31.04 | mutilator | i would, asking is cheaper tho |
| 13:31.06 | mutilator | :P |
| 13:31.11 | Zeeek | this is some really wacky thing, but what? |
| 13:31.13 | mutilator | and less work |
| 13:31.53 | Zeeek | EvilGreen they do on the 1.2 box, same peers and phones |
| 13:32.18 | EvilGreen | Zeeek: ok, I see |
| 13:32.25 | Zeeek | yes, but I don't :) |
| 13:35.16 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:35.42 | DrukenHME | mutilator: te110p is which ? |
| 13:36.19 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:39.32 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:40.15 | Qwell | http://www.digium.com/en/mediacenter/news/viewpress.php?id=Digium-launches-full-line-of-PCI-Express-cards :D |
| 13:40.35 | mutilator | single t1/e1 card |
| 13:42.37 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 13:43.10 | ramindia | any one assist me. any tool available convert sip.conf to mysql ? |
| 13:43.37 | [TK]D-Fender | ramindia: Try using.. YOUR HANDS |
| 13:43.43 | Qwell | [TK]D-Fender: see above |
| 13:43.51 | [TK]D-Fender | Qwell : I read.... |
| 13:44.03 | [TK]D-Fender | Qwell : Keep up with the jones' |
| 13:44.14 | ramindia | i know that, just asking is there any tool which convert to real time |
| 13:45.28 | tzanger | *sigh* I'm an idiot |
| 13:45.34 | [TK]D-Fender | ramindia: I seriously doubt anyone cared enough to write one. |
| 13:46.19 | tzanger | I *EXPLICITLY* tell the kernel "HAI LIKE THIS PARTITION IS RO, RESPECT" and then on the other side I'm screaming WTF YOU BETTA LET ME WRITE TO FLASH BISH |
| 13:46.25 | ramindia | ok. what is the best method to use recordings. and make more calls to accomidate ? any suggestions |
| 13:46.38 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 13:47.19 | ramindia | iam not able to achive more than 50calls with my server, most of the time due to recordings, iam getting DEAD LOCK Message |
| 13:47.42 | [TK]D-Fender | tzanger: and kernel goes like "I IS FREEDOM OF SPEAKING! KTHXBIBI" |
| 13:48.05 | *** join/#asterisk rogerz (n=highvolt@nucleabio.com) |
| 13:48.25 | tzanger | [TK]D-Fender: indeed. I think in cases like this the kernel should utterly scramble the contents of the entire device, permanently lock it and say "FUCK YOU LEARN TO CODE YOU WANNABE" |
| 13:49.33 | ramindia | how to over come DEAD Lock messages |
| 13:49.43 | tzanger | seance? |
| 13:50.24 | tzanger | grab the ouijatag |
| 13:54.02 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:54.49 | [TK]D-Fender | "HAI I CHANNELZ UR HBO!" |
| 13:55.47 | tzanger | IM IN UR SMB SHARE CORRUPTING UR PR0NZZ |
| 13:57.15 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 13:57.58 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:58.13 | *** join/#asterisk kissand (n=kissand@asterix.ucnet.uoc.gr) |
| 13:58.14 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 13:58.17 | kissand | hello again |
| 13:58.24 | kissand | anyone beronet + fax ? |
| 13:58.37 | _VoiceMeUp_COM | pri's answer when like you get a call on zap1 and dial box2 ? |
| 13:59.24 | _VoiceMeUp_COM | cause my cell gets charged.. PSTN -> zap1-> box2 -> deadagi (fputs outgoing) ->hangup |
| 13:59.45 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:59.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:59.51 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 13:59.54 | [TK]D-Fender | _VoiceMeUp_COM: Funny talks Yoda think you? hmmmMMMM!??!? |
| 14:00.01 | _VoiceMeUp_COM | lol |
| 14:00.16 | _VoiceMeUp_COM | ok what i mean ( i did a callback agi.. php fputs to a outgoing file) |
| 14:00.36 | [TK]D-Fender | _VoiceMeUp_COM: you also canjust just say "box2" in there. WTF do we know about how you're treating that call. |
| 14:00.54 | _VoiceMeUp_COM | call from cell to ZAP to my did callflow is ZAP (sip)-> Box2(sip) then on this boz its deadagi |
| 14:00.55 | [TK]D-Fender | can't* |
| 14:01.25 | [TK]D-Fender | _VoiceMeUp_COM: Sorry.. completely untrustworthy description. We don't know if your other side answers that call. |
| 14:01.33 | _VoiceMeUp_COM | didnt |
| 14:01.33 | [TK]D-Fender | _VoiceMeUp_COM: Pastbin code |
| 14:01.40 | _VoiceMeUp_COM | k |
| 14:02.49 | *** join/#asterisk anYc (i=mario@hadince17.hadiko.uni-karlsruhe.de) |
| 14:05.51 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 14:06.58 | *** join/#asterisk galeras (n=root@201.244.240.115) |
| 14:08.10 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 14:08.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:09.44 | _VoiceMeUp_COM | http://pastebin.ca/561493 |
| 14:10.13 | _VoiceMeUp_COM | makes sense ? |
| 14:11.13 | _VoiceMeUp_COM | the callback agi simply fputs to a file the channel LOCAL/callerid@outboundcallback |
| 14:11.29 | _VoiceMeUp_COM | and send the other leg to cocntext callback |
| 14:11.49 | _VoiceMeUp_COM | let me know if you want that as well |
| 14:12.02 | _VoiceMeUp_COM | but theres no asnwe in there.. |
| 14:12.06 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-02e031be77c6333f) |
| 14:12.07 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:12.18 | [TK]D-Fender | _VoiceMeUp_COM: you're only supposed to call deadAGI on a dead channel... |
| 14:14.09 | _VoiceMeUp_COM | hmm |
| 14:14.11 | _VoiceMeUp_COM | ah |
| 14:14.22 | _VoiceMeUp_COM | think thats the prob ? |
| 14:14.29 | _VoiceMeUp_COM | the cell co thinks i asnwered somehow |
| 14:15.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:15.27 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:17.28 | [TK]D-Fender | _VoiceMeUp_COM: Well... since I also don't see what that AGI is gdoing.... |
| 14:17.51 | [TK]D-Fender | _VoiceMeUp_COM: and you provided no CLI output.... |
| 14:18.30 | [TK]D-Fender | _VoiceMeUp_COM: Is it just me, or does everybody else feel your odds shrinking by the second? :)_ |
| 14:19.39 | _VoiceMeUp_COM | lol |
| 14:19.41 | _VoiceMeUp_COM | hold on |
| 14:20.29 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 14:20.42 | _VoiceMeUp_COM | http://pastebin.ca/561514 |
| 14:20.43 | _VoiceMeUp_COM | is the agi |
| 14:21.35 | _VoiceMeUp_COM | cli coming |
| 14:22.04 | *** join/#asterisk CunningPike (n=CunningP@204.239.12.183) |
| 14:22.38 | [TK]D-Fender | _VoiceMeUp_COM: Hrm.... |
| 14:22.52 | [TK]D-Fender | _VoiceMeUp_COM: Basic callfile return script.... should be fine. |
| 14:23.01 | _VoiceMeUp_COM | yeah |
| 14:23.11 | [TK]D-Fender | _VoiceMeUp_COM: that should jsut be AGI. |
| 14:23.16 | _VoiceMeUp_COM | i modded |
| 14:23.20 | _VoiceMeUp_COM | ill test right now |
| 14:23.36 | _VoiceMeUp_COM | does the cell co need a congestion anywhere to think its busy or unavail ? |
| 14:23.39 | _VoiceMeUp_COM | i cant just hangup |
| 14:23.42 | [TK]D-Fender | _VoiceMeUp_COM: and you shouldn't create the call file in there direct, you should create it elsewhere and mv it in. |
| 14:23.44 | _VoiceMeUp_COM | its not RFc lol |
| 14:23.51 | TimothyP | so while I wait for the oven to warm up.... I Background(somefile) and if the user presses 1 a specific macros should be called, if he pressed 2 a different macro etc.;... this is already inside a macro, how can I solve this? |
| 14:23.57 | _VoiceMeUp_COM | yeah i know |
| 14:24.02 | _VoiceMeUp_COM | working on that |
| 14:24.11 | [TK]D-Fender | _VoiceMeUp_COM: You can definately congestion it. |
| 14:24.19 | _VoiceMeUp_COM | ok after the agi |
| 14:24.23 | _VoiceMeUp_COM | befor e hangup |
| 14:24.25 | [TK]D-Fender | _VoiceMeUp_COM: That translates back from SIP/PRI |
| 14:24.39 | _VoiceMeUp_COM | yeah ill pri debug the zap to see return code |
| 14:24.42 | _VoiceMeUp_COM | response cod ei mean |
| 14:27.41 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:27.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:30.06 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 14:30.09 | _VoiceMeUp_COM | ok the congestion send to the pri |
| 14:30.11 | _VoiceMeUp_COM | Got SIP response 503 "Service Unavailable" |
| 14:30.41 | _VoiceMeUp_COM | every time i dial a number on the disa prompt i get fast busy |
| 14:30.48 | _VoiceMeUp_COM | so a patern not matching |
| 14:30.58 | _VoiceMeUp_COM | i need a dot after _NXXNXXX,1, ?? |
| 14:31.10 | _VoiceMeUp_COM | tough dots where like * for everyting aftr |
| 14:31.30 | [TK]D-Fender | _VoiceMeUp_COM:pastebin it at verbose 10 |
| 14:31.35 | _VoiceMeUp_COM | k |
| 14:32.40 | _VoiceMeUp_COM | well it orked now |
| 14:32.42 | _VoiceMeUp_COM | no idea why |
| 14:32.51 | _VoiceMeUp_COM | gotta dial fast or its dtmf issues |
| 14:33.13 | _VoiceMeUp_COM | TIMEOUT(digit)=7 and ResponseTimeout=10 |
| 14:33.20 | _VoiceMeUp_COM | so its prolly DTMF |
| 14:33.35 | *** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 14:33.47 | _VoiceMeUp_COM | ok no charge this time all good thanks |
| 14:33.52 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:33.55 | _VoiceMeUp_COM | the ocngestion and no dead helped me alot thanks tdk |
| 14:36.31 | [TK]D-Fender | _VoiceMeUp_COM: np |
| 14:36.46 | [TK]D-Fender | _VoiceMeUp_COM: Now change it so it mv's the file in and you'll be set ;) |
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| 14:38.51 | Zeeek | by jove |
| 14:41.09 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 14:42.13 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 14:44.16 | shay|work | the cable needed for connecting an ISDN box to an ISDN card is a normal RJ-45 CAT5 Ethernet cable, right? |
| 14:45.28 | *** join/#asterisk PoWeRKiLL (n=powerkil@LSt-Amand-152-31-40-167.w82-127.abo.wanadoo.fr) |
| 14:45.40 | PoWeRKiLL | Hi |
| 14:46.22 | PoWeRKiLL | Someone know why even using canreinvite=no I see the rtp on my asterisk only for 30 seconds then it's disappear |
| 14:47.47 | perf3kt | I understand there are alot of settings for sip and nat |
| 14:48.02 | EvilGreen_ | shay|work yes, you may use straight cable |
| 14:48.17 | shay|work | EvilGreen, thanks |
| 14:48.22 | perf3kt | can someone point me in the general direction, I got everythign coming in and internal working, but recieivn gfrom extermal gives me just a busy signal |
| 14:48.46 | *** join/#asterisk kova (n=kova@tech.quentris.be) |
| 14:50.05 | [TK]D-Fender | perf3kt: pastebin the [general] section of your sip.conf |
| 14:50.18 | *** join/#asterisk bbryant (i=brett@nat/digium/x-9b4fe2eb9517517f) |
| 14:51.38 | EvilGreen_ | shay|work make sure it's full cable, i.e. with all the wires; in fact you need pairs 1 & 2 which corresponds to pins 1,2,4,6 |
| 14:52.16 | shay|work | EvilGreen, sure, not like the 10 Mbps cables. |
| 14:52.29 | EvilGreen_ | just in case ;) |
| 14:52.32 | shay|work | EvilGreen, I brought the cables today, so it's a full cable for sure ;) |
| 14:53.22 | shay|work | it even looks nice, has gold plated contacts |
| 14:53.45 | *** part/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com) |
| 14:54.43 | EvilGreen_ | shay|work if you need the perfect cable then you need to buy/order/make one with RJ-48 connecors and 2x2 shielded cable |
| 14:55.16 | shay|work | EvilGreen, I don't think that there's a need for that on a testing system :) |
| 14:55.44 | EvilGreen_ | agree |
| 14:56.23 | shay|work | now It's time to test the ISDN line |
| 15:00.01 | *** part/#asterisk anYc (i=mario@hadince17.hadiko.uni-karlsruhe.de) |
| 15:00.34 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 15:01.03 | alin` | can somebody tell me how a SIP message of keep-alive looks like? |
| 15:02.30 | [TK]D-Fender | alin`: Go enable it and start sniffing. |
| 15:02.48 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 15:02.53 | [TK]D-Fender | alin`: * uses OPTIONS packets for qualify for this purpose |
| 15:04.08 | uwe | hello, i know this should be asked in #debian, but no one there seems to know, what is the diffrence between package asterisk and asterisk-classic, asterisk-classic is the original digium version, but what would that mean ? what is the diffrence ? |
| 15:04.50 | perf3kt | quick question is there a quick way to output a file to the clipboard to paste? |
| 15:04.55 | kombi_ | uwe: the best is to look that up in packages.debian.org |
| 15:05.10 | [TK]D-Fender | perf3kt: Depends how you're seeing it. |
| 15:05.30 | kombi_ | perf3kt: what os are you under? |
| 15:05.37 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 15:05.38 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:05.45 | *** part/#asterisk alin` (n=user@193.226.173.50) |
| 15:06.30 | perf3kt | i'm running ssh and want to get the contents of a config fiel to notepad to view easier |
| 15:06.34 | perf3kt | winxp |
| 15:07.12 | kombi_ | putty I assume? just mark the text |
| 15:07.26 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 15:07.56 | kombi_ | is the audio of a phone conversation available as a stream somewhere? i.e. to feed it into ice/shout/whatever-cast? |
| 15:08.40 | perf3kt | you mean highlight? |
| 15:08.47 | kombi_ | yip |
| 15:08.49 | perf3kt | but can't get the whole conf file at one |
| 15:08.51 | perf3kt | *once |
| 15:08.58 | perf3kt | you knwo how long those thigns are |
| 15:09.03 | kombi_ | copy it to your local machine then |
| 15:09.23 | kombi_ | use winscp |
| 15:09.43 | [TK]D-Fender | perf3kt: What are you using for SSH? |
| 15:10.16 | EvilGreen_ | perf3kt: just enable logging in your terminal program, scroll the file, all the content will be saved locally in the log |
| 15:10.45 | perf3kt | putty |
| 15:11.01 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 15:11.07 | perf3kt | the logging has the output looking like a mess |
| 15:11.08 | blitzrage | or use screen |
| 15:11.20 | blitzrage | then you can detach from the screen even when you're not SSH'd in |
| 15:11.28 | blitzrage | and reconnect and look at the scroll back later |
| 15:12.16 | anonymouz666 | how can I see what's consuming 99% of CPU in asterisk process? |
| 15:12.21 | anonymouz666 | or I can't? |
| 15:12.24 | [TK]D-Fender | perf3kt: yes, you CAN get it all. click on the bottom and drag UP. it will SCROLL. |
| 15:12.45 | anonymouz666 | no strange log msgs on CLI |
| 15:14.09 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 15:14.22 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 15:14.31 | perf3kt | tk: with less, or nano? |
| 15:15.09 | kombi_ | less and then G |
| 15:15.24 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 15:15.40 | [TK]D-Fender | perf3kt: neither |
| 15:16.10 | [TK]D-Fender | perf3kt: just "cat sip.conf" scroll to the bottom of what you want. Click Drag up & release. Then paste in pastebin.ca |
| 15:16.27 | perf3kt | oh yeah cat, duh |
| 15:17.36 | perf3kt | but it only scrolls so far |
| 15:17.50 | Qwell[] | uuoc.com, get uuocpipe |
| 15:18.03 | Qwell[] | cat sip.conf | uuocpipe --someoptions |
| 15:18.27 | Qwell[] | of course, that won't filter passwords or anyhint |
| 15:18.28 | Qwell[] | anything.. |
| 15:19.58 | purplet | Hello, is there someone who can give me a push in the right direction to solve a problem with dropped calls on an IAX connection? Calls in one direction are getting dropped after about 11 minutes. I'am not sure where to look anymore. I'am suspecting NAT, but can't find any evedince for that... |
| 15:19.58 | [TK]D-Fender | perf3kt: Go into your host setup and go into options anc change your scrollback to at least 2000 lines |
| 15:20.22 | perf3kt | yeah just noticed that geesh |
| 15:20.43 | [TK]D-Fender | perf3kt: While you're at it, make sue you add a keep-slive to it... |
| 15:21.47 | perf3kt | add a keep alive? |
| 15:22.48 | perf3kt | how long? |
| 15:22.49 | [TK]D-Fender | perf3kt: under "connection" |
| 15:22.54 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 15:22.56 | [TK]D-Fender | perf3kt: I use 10s personally. |
| 15:23.01 | *** join/#asterisk EvilGreen (n=Miranda@ppp85-140-136-108.pppoe.mtu-net.ru) |
| 15:24.13 | kombi_ | purplet: any way to try the same conf without nat? |
| 15:24.14 | *** join/#asterisk mivck (i=1000@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 15:24.42 | perf3kt | can I use winscp to get the fiels directly off the server? |
| 15:25.01 | kombi_ | yes you can |
| 15:27.17 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:27.23 | [TK]D-Fender | perf3kt: just copy & paste right from cat in PuTTY |
| 15:27.43 | [TK]D-Fender | perf3kt: Holy crap this should have taken 3 SECONDS! |
| 15:27.47 | perf3kt | lol |
| 15:28.08 | perf3kt | I already did, but connecting via my ftp client is just as easy |
| 15:28.23 | purplet | kombi_: well, that's going to be difficult, cause one server (it's an IAX connection between two * servers BTW) is behind an ADSL modem with builtin router ... |
| 15:28.41 | *** join/#asterisk Runlvl (n=juan@200.69.219.113) |
| 15:28.43 | Runlvl | Helo |
| 15:29.00 | Runlvl | any one know a sip-phone for command mode? |
| 15:29.33 | Runlvl | i wanna use a voip phone in a terminal |
| 15:29.34 | Runlvl | :D |
| 15:29.37 | Runlvl | any one? |
| 15:29.48 | kombi_ | puplet: indeed, that would be the easiest to rule out NAT though.. have you checked all logs? |
| 15:32.37 | perf3kt | so after all that I think I have to update the nat setting in the sip.conf |
| 15:32.59 | [TK]D-Fender | ...... |
| 15:33.13 | kombi_ | Runlvl: no such thing to my knowledge, there might one the accepts console commands though |
| 15:33.50 | perf3kt | was that for me tk? |
| 15:33.54 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:34.16 | kombi_ | perf3kt: are you aware that there are editors for the command line? |
| 15:34.43 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:34.46 | purplet | kombi_: yes, that's true.. Maybe I can replace it with a different model for testing, but that would be one of my last options. |
| 15:35.15 | kombi_ | purplet: what do the logs say after those 11 minutes? |
| 15:35.35 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:35.46 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
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| 15:36.24 | [TK]D-Fender | perf3kt: DUH :) |
| 15:36.55 | perf3kt | tk: like nano |
| 15:36.56 | purplet | kombi_: the logs aren't very conclusive... In the debug log chan_iax2 on one end (behind the adsl) it says We're hanging up, other side says it received a hangup... |
| 15:37.00 | [TK]D-Fender | perf3kt: All this time we were working just so you could friggin pastebin the [general] section of your sip.conf so we could tell you what you skipped that is prventing it from working. |
| 15:37.30 | [TK]D-Fender | perf3kt: just cat the stupid file, click, drag, release, PASTE to pastebin.ca and be done with it! |
| 15:37.44 | purplet | kombi_: Verbose log says just a hangup |
| 15:37.51 | [TK]D-Fender | perf3kt: It took longer for me to write that than it should take you to DO it! |
| 15:37.59 | perf3kt | tk: like nano |
| 15:38.02 | kombi_ | purplet: have you increased verbosity to the insane level? |
| 15:38.07 | perf3kt | http://pastebin.ca/561715 |
| 15:38.36 | perf3kt | but like I said, you're gonna yell cause I hadn't set the nat settings |
| 15:40.24 | *** join/#asterisk minerale (i=achille@about/cooking/alfredo/Minerale) |
| 15:40.49 | [TK]D-Fender | perf3kt: No... first I'm going to yell at you for leaving all that sample CRAP in there :) |
| 15:41.12 | minerale | can I make multiple simultaneous outboud calls on a single asterisk server connected to a single voip provider? |
| 15:41.17 | [TK]D-Fender | perf3kt: PERMANENTLY remove every comment that you did not explicitly make yourself from your sip.conf and repastebin it |
| 15:41.28 | purplet | kombi_: no i haven't. Can i configure that in logger.conf ? |
| 15:41.55 | minerale | I would like to re-implement the features found on grandcentral.net -> call one number -> It calls all your phones and conects to the first one, not sure if I can do it with only a remote server running asterisk and a voip line |
| 15:41.57 | [TK]D-Fender | perf3kt: don't worry, ditch it... its useless crap you don't need. |
| 15:42.10 | kombi_ | purplet: or start cli with -rvvvvvvvvvvvvvvvvv... |
| 15:42.28 | [TK]D-Fender | minerale: "show application dial" |
| 15:43.01 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:43.35 | minerale | [tk]d-fender: I don't have asterisk, I'm considering signing up for a server + sip, just want to make sure It can be done |
| 15:43.58 | [TK]D-Fender | minerale: then "Yes". |
| 15:44.07 | perf3kt | tk: I was abotu to say that would leave me with like one line |
| 15:44.22 | purplet | kombi_: ok, thx.. i'am going to do another testcall with that |
| 15:44.39 | [TK]D-Fender | perf3kt: if thats the case you've set NOTHING up and you can go back to try and actually DO something :) |
| 15:45.06 | perf3kt | tk: and I am totally fine with doing that |
| 15:45.19 | minerale | [tk]d-fender: two more questions: is asterisk its own distro .. or is it an app that I can simply run on debian, and 2) where can I get a list of voip (sip?) providers ? |
| 15:45.29 | perf3kt | I just honestly need jsut a little direction, I'm not asking anyone to DO it, just tell me what needs to be done |
| 15:45.37 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 15:45.40 | [TK]D-Fender | perf3kt: I hope you weren't serious when you said 1 line.... |
| 15:45.46 | Trevor_b | /koin #asterisk-gui |
| 15:45.49 | Trevor_b | doh |
| 15:46.08 | [TK]D-Fender | perf3kt: Otherwise what the hell did you set up? Without any config for your provider whats your worry with NAT? You have nothing to receive AGAINST |
| 15:46.30 | perf3kt | the gui |
| 15:46.33 | perf3kt | :( |
| 15:46.41 | Trevor_b | perf3kt: you get your sip through your firewall? Sorry about lst night, wife was still up and had to hang out ;) |
| 15:46.54 | minerale | what exactly is asterisk? a distro or an application? |
| 15:47.04 | Trevor_b | telephony application |
| 15:47.07 | [TK]D-Fender | perf3kt: sounds like you have officially done NOTHING... not much I can do for you there. Ditch that flaming pile ofcrap and come back when you're ready. |
| 15:47.57 | s0ck | a B410P should work fine with uk isdn, yeh? |
| 15:48.06 | [TK]D-Fender | "Oh no [TK]D-Fender , tell us how you REALLY feel. Don't hold back now!" |
| 15:48.16 | Trevor_b | hahahahah |
| 15:49.05 | *** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net) |
| 15:50.29 | *** join/#asterisk elg (n=fugalh@azerial.fastwave.biz) |
| 15:50.49 | elg | what is a "power alarm" on zaptel? tdm400p |
| 15:51.15 | Qwell[] | elg: don't quote me on this, but I think it means you need to plug in a molex power connector |
| 15:51.47 | *** join/#asterisk absa (n=absa@193.219.45.243) |
| 15:52.06 | sci_05 | ok got a question for you guys. I have an asterisk box (1.4.4) with a pri, its gets it calls and sends them to an offsite server. For some reason the offsite server they will get oneway audio in the middle of a call. I have the iax trunk in a vpn tunnel also. Could this be a QOS problem (as in their upload is maxed out)? |
| 15:54.44 | perf3kt | tk: okay |
| 15:55.32 | *** join/#asterisk EvilGreen_ (n=Miranda@ppp85-140-136-108.pppoe.mtu-net.ru) |
| 15:56.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:58.27 | [TK]D-Fender | perf3kt: Oh, and in ditching the GUI, trash that users.conf BS while you're at it. |
| 15:59.01 | perf3kt | hey, I just use what they put out |
| 15:59.27 | [TK]D-Fender | perf3kt: Microsoft made this program called "Bob"... you should go use it! :) |
| 15:59.38 | file | [TK]D-Fender: omg |
| 15:59.45 | [TK]D-Fender | file: pwned |
| 15:59.47 | perf3kt | tk: I thought if it was under the asterisk name, it was sanctioned and everyone was working to make it better |
| 16:00.06 | [TK]D-Fender | perf3kt: Yeah... and democracy works ;) |
| 16:00.24 | [TK]D-Fender | perf3kt: Any more illusions I can shatter for you while you are here? |
| 16:00.33 | [TK]D-Fender | perf3kt: How 'bout some movie spoilers? ;) |
| 16:01.59 | [TK]D-Fender | Ok, off to lunch for a few minutes... |
| 16:03.03 | Zeeek | ... |
| 16:03.10 | Zeeek | how much is a few? |
| 16:03.24 | Zeeek | tha'ts already three |
| 16:03.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 16:08.54 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 16:11.14 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:12.55 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 16:13.28 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 16:13.55 | perf3kt | tk: come baaack |
| 16:14.00 | perf3kt | I want movie spoilers |
| 16:15.28 | perf3kt | lol |
| 16:15.28 | KpoH | folks, how to enable asterisk update regserver column in peer table |
| 16:15.28 | KpoH | ? |
| 16:15.28 | KpoH | i've enable asterisk.conf |
| 16:15.28 | KpoH | systemname => server1 |
| 16:15.37 | KpoH | and sip.conf |
| 16:15.38 | KpoH | displaysystemname=yes |
| 16:15.45 | KpoH | rtsavesysname=yes |
| 16:16.25 | KpoH | a look to full log give me sql stataments that was executed, but where is no regserver |
| 16:18.43 | [TK]D-Fender | Zeeek: Back fast enough for you? |
| 16:19.29 | [TK]D-Fender | perf3kt: http://www.threadless.com/product/844/Spoilt |
| 16:20.01 | KpoH | MySQL RealTime: Update SQL: UPDATE peer SET ipaddr = '86.106.208.182', port = '5060', regseconds = '1181668687', username = '7325928432' WHERE name = '7325928432' |
| 16:20.20 | KpoH | that is what was executed |
| 16:20.29 | KpoH | wtf with aster? |
| 16:22.21 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:23.17 | KpoH | anybody? |
| 16:23.27 | KpoH | :( |
| 16:26.55 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 16:28.21 | *** join/#asterisk elg (n=fugalh@azerial.fastwave.biz) |
| 16:31.34 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 16:32.51 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 16:34.13 | Trionnis | is there a dialplan command similar to sleep or pause? something where I can force it to wait for x seconds before continuing the steps? |
| 16:34.26 | Trionnis | I'm not seeing anything in the voip-info wiki |
| 16:34.39 | KpoH | Wait() |
| 16:34.58 | Trionnis | ah hah |
| 16:35.00 | Trionnis | thank you! |
| 16:35.01 | Trionnis | :) |
| 16:35.03 | KpoH | Wait(10) will wait 10 seconds |
| 16:35.48 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 16:36.12 | blitzrage | show applications |
| 16:40.11 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:40.35 | irule | how can I react to certain return codes? |
| 16:41.15 | blitzrage | like the -1? |
| 16:41.17 | blitzrage | you can't |
| 16:41.28 | [TK]D-Fender | irule: Depends what kind of codes and where you;re returning from :) |
| 16:41.36 | [TK]D-Fender | Trionnis: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
| 16:41.41 | [TK]D-Fender | Trionnis: Its there.... |
| 16:41.45 | irule | thanks |
| 16:41.56 | blitzrage | usually something like a ${DIALSTATUS} is what you're going to match on |
| 16:42.02 | Trionnis | hm... I don't think Wait is going to work for my needs, since I'm using a dropped call file into outgoing. I need to retry the call if the first one fails |
| 16:42.17 | blitzrage | Trionnis: so loop |
| 16:42.24 | Trionnis | basically, call, test, if failed, wait 1 minute, try again, etc |
| 16:42.29 | blitzrage | so do that |
| 16:42.31 | blitzrage | Dial() |
| 16:42.52 | blitzrage | Exec(${IF($[${DIALSTATUS} != ANSWER]?Wait(60):NoOp())}) |
| 16:42.56 | *** join/#asterisk MrTelephone (n=na@h64184192-5.picriverisp.net) |
| 16:43.16 | MrTelephone | if you Dial out how do you execute the next in sequence after a user picks up? |
| 16:43.18 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 16:43.19 | blitzrage | or you can just do a GotoIf($[${DIALSTATUS} != ANSWER]?retry_dialer) |
| 16:43.20 | MrTelephone | such as txfax |
| 16:43.31 | blitzrage | if you Dial(), and the call answers, you don't continue on |
| 16:43.38 | blitzrage | although I think the 'g' option will let you do that |
| 16:43.42 | blitzrage | show application dial |
| 16:43.50 | Trionnis | well, it's not only answer, I'm testing for a dtmf tone |
| 16:43.50 | blitzrage | amazing the things you can find in the documentation... :) |
| 16:44.04 | blitzrage | Trionnis: ok, so make the matching more/less specific |
| 16:44.15 | Trionnis | if it fails, I need to retry the call, but the part that's throwing me is that it's not a "normal" call |
| 16:44.25 | blitzrage | what does "normal call" mean |
| 16:44.34 | perf3kt | tk: where are my spoilers? |
| 16:44.36 | blitzrage | nothing is "normal" in asterisk |
| 16:44.39 | Trionnis | heh |
| 16:45.38 | `pariah | I need help with my dialplan, if anyone is willing to help here is it http://rafb.net/p/c3te7E62.html what i am trying to do is have a SIP call forwarded to zap and ring an extension on our PBX. it pretty much works just some small things like sometime zap wont quit ringing or time out after a specified time. |
| 16:45.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:45.44 | [TK]D-Fender | perf3kt: I linked you AGES ago. Wake up! |
| 16:46.02 | perf3kt | sorry had my asterisk-gui cahnnel open, LOL |
| 16:47.18 | MrTelephone | g: When the called party hangs up, exit to execute more commands in the current context. |
| 16:47.27 | MrTelephone | exten => 700,1,Dial(ZAP/g1/2293367,g) |
| 16:47.27 | MrTelephone | exten => 700,2,txfax(/var/www/apache2-default/faxes/1175190265.270.tif) |
| 16:47.36 | perf3kt | tk: that shirt is the greatest |
| 16:48.16 | *** join/#asterisk AeGu2 (n=Jeff@70.230.169.91) |
| 16:48.25 | [TK]D-Fender | perf3kt: I went bang-for-buck with that link... |
| 16:48.59 | [TK]D-Fender | MrTelephone: Um.... that sure doesn't LOOK like it'll work.... |
| 16:49.36 | MrTelephone | yeah I know it doesn't, wish it would :P |
| 16:49.40 | [TK]D-Fender | MrTelephone: Call a guy, when they hang up try sending a fax to DEAD AIR. |
| 16:49.44 | MrTelephone | haha |
| 16:49.48 | [TK]D-Fender | MrTelephone: Your approach is BROKEN ;) |
| 16:50.02 | MrTelephone | the only other dial command is running a macro when answered? |
| 16:50.17 | AeGu2 | Does anyone know if you can get asterisk to work if I just have 3 old dial up fax modems? or what card should I buy im trying to build this for a small company of 3 employees |
| 16:50.43 | [TK]D-Fender | MrTelephone: You'd need to better describe the actions you would take and how you would like * to react so we can advise you. |
| 16:50.53 | `pariah | AeGu2: you can find X100P clones for next to nothing, but they don't work as well as digium products |
| 16:50.57 | [TK]D-Fender | AeGu2: Go check up the hardware compatability list on the WIKI |
| 16:51.02 | [TK]D-Fender | ~wikis |
| 16:51.18 | jbot | [wikis] http://www.voip-info.org |
| 16:51.21 | [TK]D-Fender | AeGu2: Anfd no, your pile of craptastic winmodems are almost guaranteed worthless to * |
| 16:51.36 | MrTelephone | Just dial a number and run txfax when the called party picks up.. heres what I'll try.. |
| 16:51.45 | MrTelephone | [prdc] |
| 16:51.45 | MrTelephone | exten => 700,1,Dial(ZAP/g1/2293367,M(send_fax)) |
| 16:51.51 | MrTelephone | [macro-send_fax] |
| 16:51.51 | MrTelephone | exten => s,1,txfax(/var/www/apache2-default/faxes/1175190265.270.tif) |
| 16:51.53 | perf3kt | well tk, i'm off to my oriely book and fresh install of asterisk |
| 16:51.57 | [TK]D-Fender | AeGu2: What harddware we'd suggest depends on the actual number of lines and kind that you require/desire |
| 16:52.10 | [TK]D-Fender | MrTelephone: WHO dials that? |
| 16:52.20 | [TK]D-Fender | perf3kt: yay |
| 16:52.23 | MrTelephone | its a testfax option from the console |
| 16:52.33 | perf3kt | tk: just know, that I'll be back |
| 16:52.35 | [TK]D-Fender | MrTelephone: Far to say "system generated? |
| 16:52.52 | AeGu2 | well basically all we would like do is have voicemail boxes, fax, and then 3 possibly 4 phones |
| 16:53.05 | [TK]D-Fender | perf3kt: look at that quicky guide I linked you earlier. Thats a super fast start. |
| 16:53.11 | MrTelephone | i just want to send a fax to a specific number by dialing 700 from the console.. |
| 16:53.13 | AeGu2 | then the ability to conference call |
| 16:53.17 | MrTelephone | let me try it |
| 16:53.22 | [TK]D-Fender | AeGu2: * can likely handle everything you want and more. |
| 16:53.36 | [TK]D-Fender | AeGu2: But we're just talking hardware here. What kind of lines, and how many? |
| 16:54.05 | AeGu2 | Oh we just have one plan old phone line |
| 16:54.17 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 16:54.18 | AeGu2 | with caller-id call waiting... |
| 16:54.20 | [TK]D-Fender | AeGu2: And thats what you're hoping to keep? |
| 16:54.28 | [TK]D-Fender | AeGu2: Forget call waiting with *. |
| 16:54.36 | [TK]D-Fender | AeGu2: its not made with taht concept in mind. |
| 16:54.37 | AeGu2 | for the time being yes |
| 16:54.44 | perf3kt | tk: crap can't find it |
| 16:54.49 | MrTelephone | I just thought maybe there was an alternative command to execute next step on answer when using the dial command.. that wasn't doccumented ;) |
| 16:55.05 | [TK]D-Fender | AeGu2: Crappy analog concetps don't works so well with a PBX in mind. |
| 16:55.25 | [TK]D-Fender | perf3kt: http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 16:55.31 | perf3kt | thanks |
| 16:55.43 | perf3kt | does that cover installing the distro and packages? |
| 16:55.47 | [TK]D-Fender | MrTelephone: What you are describing should be done via a .call file or AMI originate. |
| 16:56.03 | ^majik^ | is it possible to specify more than one voicemail box to a single channel in zapata.conf? I see some examples on voip-info.org (ie, mailbox=123@office,124@office ) -- I just didn't know if this would work for the message waiting light on our phones |
| 16:56.04 | [TK]D-Fender | perf3kt: No, you are assumed to be be capable of installing your own distro |
| 16:56.09 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-79.ph.ph.cox.net) |
| 16:56.25 | [TK]D-Fender | perf3kt: But it does cover DLing and installing * from source. |
| 16:56.34 | perf3kt | tk:I can do that distro, just get confused when it comes to packages |
| 16:56.41 | *** join/#asterisk mutilator (n=WebChat@65.111.201.122) |
| 16:56.44 | ManxPower | ^majik^: yes, it works just fine |
| 16:56.46 | *** join/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 16:56.54 | [TK]D-Fender | ^majik^: multiple individual mailbox lines. |
| 16:56.55 | *** join/#asterisk logyati (n=suporte@201.29.73.49) |
| 16:56.59 | logyati | hi ppl |
| 16:57.17 | [TK]D-Fender | perf3kt: No packages, jsut DL the source and compile. instructions are on www.asterisk.org |
| 16:57.28 | MrTelephone | it worked but it wanted a timeoute before the M() operator.. Dial(ZAP/g1/82123,20|M(send_fax)) |
| 16:57.30 | logyati | yesterday i was here and someone told me to buy a book, i did it and im reading, but i have some doubts about what can i do with asterisk |
| 16:57.41 | ManxPower | [TK]D-Fender: somehow I'm a bit wary of recommending a newbie document written by JerJer 8-) |
| 16:57.50 | logyati | the book is "building telephony systems with asterisk" |
| 16:57.57 | [TK]D-Fender | ManxPower: *I* edited the whole thing for him :) |
| 16:58.08 | ManxPower | [TK]D-Fender: that makes me feel better |
| 16:58.14 | BSD_Tech[laptop] | but is it for 1.2 or 1.4 |
| 16:58.16 | [TK]D-Fender | <- Standards |
| 16:58.21 | BSD_Tech[laptop] | alot of things change in 1.4 |
| 16:58.23 | [TK]D-Fender | BSD_Tech[laptop]: 1.4 |
| 16:58.27 | BSD_Tech[laptop] | ok |
| 16:58.38 | ManxPower | BSD_Tech[laptop]: and nobody reads UPGRADE.txt |
| 16:58.58 | BSD_Tech[laptop] | I know its a big issue |
| 16:58.59 | [TK]D-Fender | "I would like to thank Andrew Oulton for his technical editing of this article." |
| 16:59.17 | BSD_Tech[laptop] | I feel like I spend half my days dealing wth users who dont read thigns |
| 16:59.24 | MrTelephone | I just finished my kickass billing software in perl |
| 16:59.34 | JerJer | ManxPower: why is that? |
| 16:59.39 | BSD_Tech[laptop] | MR Tell you going to share |
| 16:59.50 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:59.59 | [TK]D-Fender | JerJer: You have benn... deprecated! Oh noes! :O |
| 17:00.38 | BSD_Tech[laptop] | I just wish Digium/Kram/Other head staff would get more html/Javascript people to work on the gui so it would be usefull |
| 17:00.41 | JerJer | that single post gets very serious search engine love |
| 17:00.42 | MrTelephone | its kind of basic though.. what are you planning to do with your phoen system? |
| 17:00.50 | BSD_Tech[laptop] | only 2 people working on it is not enough |
| 17:01.08 | logyati | i have some questions... first is, if i wanna build a voip server, to make voip-to-voip and voip-to-pstn and pstn-to-voip, wich softwares should i use? |
| 17:01.12 | tzanger | word jerjer |
| 17:01.26 | [TK]D-Fender | logyati: Asterisk. (Duh) |
| 17:01.39 | logyati | i dont need openser??? |
| 17:01.43 | MrTelephone | I would like to do more java but I find there is a lot of conflict in versions, don't you find BSD? |
| 17:01.49 | logyati | freeradius |
| 17:01.55 | logyati | mediaproxy |
| 17:01.59 | ManxPower | logyati: only if you have a large instalation |
| 17:02.00 | tzanger | U SHULD USE TEH AWESOEMEST SOFTWAREZ ONLY PERSONALLY I RCCMND ASTERIX IT IS AWESOEM |
| 17:02.06 | [TK]D-Fender | logyati: Need, no. Want... depends on the size of your deployment, etc. Planning on becoming an ITSP? |
| 17:02.56 | logyati | well, what if i want a large instalation, yes i wanna be a ITSP |
| 17:02.56 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:02.57 | BSD_Tech[laptop] | I just wanted to se it |
| 17:02.57 | MrTelephone | BSD i reccommend you start with some mysql tables for your customers.. and write a couple perl scripts to read mysql and print the output in html |
| 17:02.57 | [TK]D-Fender | logyati: If you have to ask.... you're already unqualified :) |
| 17:03.01 | logyati | hehehe |
| 17:03.02 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:03.02 | ManxPower | logyati: well, get a small installation working first without all that extra stuff |
| 17:03.03 | *** join/#asterisk canberk (n=canberk@212.156.213.131) |
| 17:03.06 | Ryushin | Is there any way to increase the gain on IAX clients from the server? |
| 17:03.06 | canberk | hello |
| 17:03.07 | logyati | im qualifying :D |
| 17:03.10 | JerJer | [TK]D-Fender: tru dat |
| 17:03.30 | JerJer | then again we can teach monkeys to communicate |
| 17:03.34 | ManxPower | logyati: You should expect to spend at least 6 months just learning enough to be an ITSP, maybe as much as a year if you don't have a background in networking, linux, or telecom |
| 17:03.37 | [TK]D-Fender | logyati: And yes, you'd be wanting a serious SIP proxy, soft-switch, redundant PRI setup, etc. |
| 17:03.38 | logyati | yes but i wanna know the softwares u should use, to get a way on my studies |
| 17:04.01 | ManxPower | Ryushin: no, that is the job of the device that converts analog to digital like a SIP phone, ATA, or Zap line |
| 17:04.04 | logyati | hmmm |
| 17:04.04 | canberk | i have a question, is it possible to create a pseudo sip channel to call a local sip user with SIP/LocalHostProfile/int.number format? |
| 17:04.18 | logyati | i made a project here |
| 17:04.19 | JerJer | logyati: this is the Asterisk channel - obviously its going to be biased toward asterisk |
| 17:04.21 | Ryushin | ManxPower: I was hopping that wasn't the answer. |
| 17:04.25 | Ryushin | Oh well. |
| 17:04.29 | ManxPower | canberk: your sentence makes no sense. |
| 17:04.32 | logyati | using LDAP to handle openser accounts |
| 17:04.34 | canberk | i see |
| 17:04.35 | canberk | look |
| 17:04.39 | ManxPower | Ryushin: you need to fix it before it gets to IAX |
| 17:04.46 | logyati | freeradius do comunicate openser with ldap |
| 17:04.47 | canberk | i want to create a SIP channel where i can call local users |
| 17:04.52 | Ryushin | Okay, thanks much. |
| 17:05.01 | logyati | mediaproxy to override nat |
| 17:05.04 | [TK]D-Fender | canberk: Create from WHERE? |
| 17:05.05 | canberk | instead of just SIP/extension |
| 17:05.05 | ManxPower | canberk: pretty much every single asterisk install that uses SIP does that |
| 17:05.08 | JerJer | logyati: try joining #OpenSER |
| 17:05.10 | canberk | locally |
| 17:05.11 | logyati | and asterisk |
| 17:05.14 | canberk | look |
| 17:05.19 | ManxPower | canberk: WRONG! WRONG! WRONG! |
| 17:05.20 | [TK]D-Fender | canberk: "locally" is NOT an answer. |
| 17:05.29 | ManxPower | canberk: SIP/sipconfentry |
| 17:05.38 | ManxPower | you don't dial extensions, you dial sip accounts |
| 17:05.40 | canberk | yeah ManxPower |
| 17:05.41 | logyati | no no, im here cos i wanna know if asterisk can do alone everything this programs i said do |
| 17:05.58 | ManxPower | logyati: it can't do what ANY of those programs do. |
| 17:05.59 | canberk | i want to dial a sip account, but that sip account will be the the asterisk server itself |
| 17:06.08 | canberk | and then i will call the sip clients |
| 17:06.12 | logyati | hmm |
| 17:06.12 | canberk | did you get it? |
| 17:06.12 | [TK]D-Fender | logyati: No * is NOT enough to run an ITSP on alone. |
| 17:06.24 | *** join/#asterisk brussel_ (n=brussel@adsl-71-154-207-12.dsl.sndg02.sbcglobal.net) |
| 17:06.28 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 17:06.29 | [TK]D-Fender | logyati: You WILL need a pile of other stuff, something like SER most assuredly. |
| 17:06.31 | logyati | thats what i wanted to know :D |
| 17:06.36 | ManxPower | canberk: yes, I do it all the time. It's just a phone call and an IP device, just like all other phone calls and sip devices. |
| 17:06.52 | ManxPower | canberk: read The Book, set up your own asterisk server, see how it works |
| 17:06.55 | canberk | ManxPower, can you explain a litte bit |
| 17:07.02 | canberk | i have my own asterisk server |
| 17:07.04 | [TK]D-Fender | canberk: Yes, you can. Now go read the book, install * and start playing around. |
| 17:07.06 | canberk | working perfectly |
| 17:07.06 | [TK]D-Fender | ~book |
| 17:07.09 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:07.22 | logyati | just more one question, should i need a gatekeeper like gnugk? or asterisk does its features? |
| 17:07.26 | ManxPower | canberk: You must already be doing what you want to do. |
| 17:07.40 | ManxPower | logyati: gatekeeper is of H323 |
| 17:07.52 | [TK]D-Fender | logyati: No, * does not include much by way of H.323 support. It isn't stable with what little it DOES now.. |
| 17:08.02 | canberk | oh god |
| 17:08.04 | canberk | http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 17:08.09 | canberk | this is what i am talking about |
| 17:08.16 | logyati | ty :D |
| 17:08.31 | ManxPower | canberk: If you had said so in the first place I would never have wasted 20 mins of my life I'll never get back. |
| 17:08.41 | [TK]D-Fender | canberk: You need to start over and be very descriptive of the tool used to start this "call" you are talking about, and what you want * to do for you. |
| 17:08.47 | canberk | ManxPower, i'd give you my 20 mins if it was possible... |
| 17:08.57 | ManxPower | canberk: no, I do not use Dial(Local/extension) much at all, as it is only needed in a few situations |
| 17:09.11 | canberk | and i am in one of those situations |
| 17:09.19 | [TK]D-Fender | ManxPower: then again... its usually little better than a GOTO. |
| 17:09.28 | ManxPower | canberk: I have not seen any reason from you that would require Local/ |
| 17:09.39 | [TK]D-Fender | canberk: Your description does not tell us anything about needing Chan_local. |
| 17:09.39 | canberk | so |
| 17:09.47 | canberk | i am using asterisk2billing |
| 17:09.51 | [TK]D-Fender | canberk: There, heard it from both of us |
| 17:09.54 | canberk | and i need to bill the local calls too |
| 17:09.57 | ManxPower | canberk: as [TK]D-Fender mentioned Local/ is pretty much a fancy goto |
| 17:09.58 | [TK]D-Fender | canberk: now START OVER. nice and slow. |
| 17:10.02 | canberk | okay |
| 17:10.15 | ManxPower | canberk: I cannot help you with billing, as I do not bill for calls that go thru my system. |
| 17:10.15 | canberk | i am using a2billing for billing, and a2billing has its own sip/iax friends billing |
| 17:10.39 | [TK]D-Fender | canberk: Oh... a GUI... |
| 17:10.43 | canberk | however, this system cannot monitor the calls and bill them properly |
| 17:10.44 | logyati | omg im at chapter 5, creating a dialplan... its so hard to understand extensions.conf :( |
| 17:10.59 | ManxPower | canberk: I suspect you'll have to go to the a2billing support for help with that |
| 17:11.01 | canberk | and i need to make a2billing seem like it is dialing a trunk for pstn and connecting to a remote server |
| 17:11.19 | [TK]D-Fender | canberk: Time to start hitting the books and reading up on a2billing's support mailing lists, etc./ |
| 17:11.20 | canberk | i will create a new trunk in a2billing for Local extensions |
| 17:11.29 | [TK]D-Fender | canberk: Because GUI's are NOT supported here. |
| 17:11.35 | ManxPower | canberk: why can't it ACTUALLY dial a trunk for pstn and connect to remote server. |
| 17:11.56 | [TK]D-Fender | canberk: And don't be too liberal with the use of "Local". This has NOTHING to to do with BILLING. |
| 17:12.12 | [TK]D-Fender | canberk: Or dialing, or ANYTHING. |
| 17:12.27 | [TK]D-Fender | canberk: chan_local is an internal mechanism for other puroposes. |
| 17:12.44 | ManxPower | canberk: Dial(Local/1234@fred) is almost exactly the same as Goto(fred,1234,1) |
| 17:12.46 | [TK]D-Fender | logyati: Keep reading |
| 17:12.57 | logyati | hehe im doing it |
| 17:13.07 | ManxPower | logyati: you don't even have the concept of how much work you have ahead of you. |
| 17:13.20 | ^majik^ | one other question, in zapata.conf, group is seperate from callgroup is seperate from pickupgroup - all seperate from one another, right? |
| 17:13.39 | canberk | ok guys |
| 17:13.43 | ManxPower | ^majik^: callgroup and pickupgroup are related |
| 17:13.49 | canberk | you wanted me to explain what i am going to do with the local channel |
| 17:13.51 | canberk | and i explained |
| 17:13.51 | ManxPower | canberk: you really need to talk to the a2billing people |
| 17:13.53 | logyati | lol, that was to encourage me.... |
| 17:13.56 | logyati | hehehehe |
| 17:14.12 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 17:14.15 | canberk | a2billing is a nice software with NO support |
| 17:14.18 | perf3kt | logyati I'm with you |
| 17:14.28 | ManxPower | canberk: no real need to explain. We will tell you it is wrong to use chan_local. If you want to be told something else then go to the a2billing people, as they seem to require it. |
| 17:14.30 | perf3kt | don't try to swim without reading that whole book |
| 17:14.36 | perf3kt | I just did and got reamed |
| 17:14.45 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 17:15.05 | ^majik^ | ManxPower: ok, but if I'm using group=1 for our outgoing channels, I could still use callgroup=1,pickupgroup=1 for a group of phones in the office, for *8, right? |
| 17:15.23 | ManxPower | We worked our asses off, put in massive amounts of time, work, money, reading, reading, reading, learning, experimenting. We don't tollerate people well that don't want to do that. |
| 17:15.32 | [TK]D-Fender | ^majik^: Pretty much. |
| 17:15.33 | canberk | thanks |
| 17:15.38 | ManxPower | ^majik^: correct. group= is totally unrelated to the other two |
| 17:15.44 | logyati | perf3kt together we are invicible hahaha |
| 17:15.59 | *** join/#asterisk Delta_Offset (n=abrahamc@201.226.130.55) |
| 17:16.09 | [TK]D-Fender | ^majik^: group= is for pooling zap channels for an outbound dial. |
| 17:16.28 | ManxPower | It is pretty common too. We even joke about people coming into the channel and wanting to start an ITSP in a week. |
| 17:16.34 | ManxPower | Mostly because it is SO funny. |
| 17:16.43 | ^majik^ | ManxPower, [TK]D-Fender: ok. just wondering.. cuz the guy before me used, ie, callgroup & pickupgroup=2, as if he couldn't use 1 because of the earlier group=1 |
| 17:16.48 | Delta_Offset | im having this problem on the cli |
| 17:16.49 | ^majik^ | right |
| 17:16.53 | Delta_Offset | Jun 12 13:16:31 WARNING[11023]: pbx.c:1720 pbx_extension_helper: No application 'Meetme' for extension (default, 8600, 1) |
| 17:16.53 | Delta_Offset | <PROTECTED> |
| 17:17.08 | Delta_Offset | sorry.. |
| 17:17.08 | ManxPower | ^majik^: he was either an idiot or had some OTHER reason. |
| 17:17.25 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:17.29 | ManxPower | Delta_Offset: looks like you didn't install zaptel before compiling Asterisk |
| 17:17.42 | logyati | manxpower i dont wanna do this what u saying, i just wanna know what i should learn, to have a direction to study |
| 17:17.51 | [TK]D-Fender | Delta_Offset: Indeed. |
| 17:18.06 | Delta_Offset | mmm..let me compile again |
| 17:18.37 | [TK]D-Fender | Delta_Offset: And even if you DID compile & install zaptel before * and recompiled * after you still need to have a timing source configured (ztdummy is not build by default if you aren't using a hardware timing source) |
| 17:18.53 | [TK]D-Fender | Delta_Offset: Pay attention to its build instructions. |
| 17:18.54 | ManxPower | logyati: I understand. The thing is that some of these concepts are so hard to understand we simply do not have the time to do a 1 day seminar just so you know enough to ask intellegent questions. That is why the book, mailing list archives, and to a lesser extent the Wiki are places to read |
| 17:19.07 | Delta_Offset | yes i will fender |
| 17:19.17 | Delta_Offset | the thing is that i was having problems before |
| 17:19.41 | logyati | hehe im reading |
| 17:19.54 | logyati | to make those questions |
| 17:19.58 | logyati | ^ |
| 17:20.00 | logyati | ^^ |
| 17:20.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:20.11 | Delta_Offset | and i installed libpri, zaptel and then asterisk |
| 17:20.17 | Delta_Offset | i had problems with zaptel |
| 17:20.23 | Delta_Offset | so i did zaptel again |
| 17:20.31 | Delta_Offset | but i forgot to do asterisk |
| 17:20.47 | [TK]D-Fender | Delta_Offset: Ok, get to it and when/if it fails, let us know. |
| 17:21.27 | Delta_Offset | sure thing. |
| 17:21.56 | ber111 | in the AGI interface, if i invoke a script with DeadAGI to dial a number, does that invoked script block at the dial |
| 17:22.11 | ber111 | so when i query for the ANSWEREDTIME later in the script it is accurate |
| 17:22.17 | ManxPower | ber111: ALL Dial invocations clock |
| 17:22.24 | ManxPower | block that is |
| 17:22.27 | ber111 | ah cool |
| 17:22.41 | ManxPower | I don't even know IF you can Dial from exten => h |
| 17:22.41 | ber111 | so if i query for answered time afer the DIAL command i will be safe |
| 17:23.02 | ManxPower | ber111: assuming you are not using FXO ports to connect to the PSTN, yes. |
| 17:23.26 | ber111 | ok cool |
| 17:23.32 | ber111 | i am using sip |
| 17:24.00 | ManxPower | Any decent ITSP would be using PRI to connect to the PSTN, so you should be safe. |
| 17:25.19 | *** join/#asterisk bonderponder (n=test@201.199.68.150) |
| 17:25.24 | *** join/#asterisk rafael-ec (n=rafael@200.93.218.202) |
| 17:25.51 | bonderponder | hello, anybody can help me with an IVR for phonebanking demostration ? |
| 17:25.52 | Delta_Offset | same |
| 17:25.54 | Delta_Offset | Jun 12 13:25:30 WARNING[22154]: pbx.c:1720 pbx_extension_helper: No application 'Meetme' for extension (default, 8600052, 1) |
| 17:25.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:26.18 | *** join/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 17:26.43 | JoeDeveloper | Hey, I am using originate to start a call. How do I get asterisk to run a context if there is no answer or a timeout? |
| 17:26.48 | bonderponder | hello, anybody can help me with an IVR for phonebanking demostration ? I need to have 12-16 digits Credit card number followed by the pound key , then please enter your pin num followed by the pund key, with no AGI just simple |
| 17:27.14 | Qwell[] | bonderponder: a simple read() would work |
| 17:27.32 | [TK]D-Fender | Delta_Offset: do you have a Zaptel card? |
| 17:28.13 | bonderponder | Qwell: but I need to use the pund key to continue to the next option , the pin # how can I do that? |
| 17:28.13 | [TK]D-Fender | JoeDeveloper: use a local channel for your originate and put the logic in there. |
| 17:28.32 | [TK]D-Fender | bonderponder: "show application read" |
| 17:28.49 | bonderponder | where can I find about that? |
| 17:28.58 | [TK]D-Fender | bonderponder: enter that at * CLI |
| 17:29.09 | [TK]D-Fender | bonderponder: and READ THE INSTRUCTIONS. |
| 17:29.09 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 17:30.08 | bonderponder | but I need the # key to work. maxdigits have been entered (without requiring the user to |
| 17:30.08 | bonderponder | <PROTECTED> |
| 17:30.35 | [TK]D-Fender | bonderponder: don't USE max digits |
| 17:30.49 | [TK]D-Fender | bonderponder: ***READ*** |
| 17:30.52 | *** join/#asterisk raptorra1 (n=rathomps@lnxwc1.shsu.edu) |
| 17:31.08 | [TK]D-Fender | bonderponder: the instructions can't be any clearer. |
| 17:31.37 | raptorra1 | anyone in here have a working cisco 2800/2600 config with asterisk? |
| 17:31.56 | bonderponder | ok let me find some examples to play arround |
| 17:32.28 | raptorra1 | trying to setup a 2600 series router with a vwic/pri to send land line calls to the asterisk box |
| 17:33.24 | raptorra1 | the best I've been able to do is only one way audio and the noreinvites don't seem to be helping. I think it is my cisco config b/c I have no idea what I'm going in cisco |
| 17:33.56 | [TK]D-Fender | raptorra : I guess we'll just have to take your word for it. |
| 17:36.07 | Delta_Offset | [TK]D-Fender: its still doing the same thing |
| 17:36.37 | Delta_Offset | dahh |
| 17:36.44 | Delta_Offset | i guess i have to reload asterisk |
| 17:36.50 | [TK]D-Fender | Delta_Offset: before starting * : modprobe zaptel ; modprobe ztdummy ; ztcfg-vvvv |
| 17:37.03 | Delta_Offset | yep |
| 17:37.06 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.192) |
| 17:37.17 | [TK]D-Fender | Delta_Offset: Oh.. you though you could recompile *, reinstall it while its still RUNNING and it would jsut start working?! :) |
| 17:37.24 | Delta_Offset | :D |
| 17:37.37 | Delta_Offset | i just... well.. i too tired now |
| 17:37.41 | ManxPower | [TK]D-Fender: Ah, the innocence of youth. |
| 17:37.52 | raptorra1 | [TK]D-Fender: I take it you haven't messed with anything like this? |
| 17:38.02 | Delta_Offset | im falling asleep here in front of the laptop |
| 17:38.20 | Delta_Offset | xDD |
| 17:38.28 | ManxPower | Delta_Offset: then go do something else for a while. Sleepy Asterisk Admin = Broken PBX |
| 17:38.31 | [TK]D-Fender | raptorra : You haven't shown us your configs so we have NO idea how many things you've done wrong so we cannot place blame or assist you. |
| 17:38.42 | Delta_Offset | i know.. |
| 17:38.44 | Delta_Offset | :D |
| 17:38.58 | Delta_Offset | lol |
| 17:39.00 | [TK]D-Fender | ManxPower: 2 words : Natural-&^$@ing-selection <- |
| 17:39.18 | raptorra1 | [TK]D-Fender: I dont' have much in the way of configs it I was asking for |
| 17:39.26 | raptorra1 | let me hook something on pastebin |
| 17:40.23 | [TK]D-Fender | raptorra1 : better. |
| 17:40.23 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 17:40.24 | Delta_Offset | even my typing is getting screwed |
| 17:40.24 | bonderponder | [TK]D-Fender: any example for the Read command that can be usefull ? |
| 17:41.01 | [TK]D-Fender | bonderponder: It takes 2 stupid parameters and the instructions tell you to your FACE that if you don't give it a maxdigits that it will wait until #. |
| 17:41.02 | Delta_Offset | chachin... |
| 17:41.03 | Delta_Offset | working |
| 17:41.12 | Delta_Offset | :D thanks |
| 17:41.13 | [TK]D-Fender | bonderponder: Get off your ass and TRY IT. Its 1 stupid line. |
| 17:41.23 | Delta_Offset | hoo... /me |
| 17:41.36 | bonderponder | take it easy brother |
| 17:42.01 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 17:42.55 | Zeeek | [TK]D-Fender topology? |
| 17:42.59 | [TK]D-Fender | bonderponder: Silly mirc user! |
| 17:43.42 | Zeeek | did you solve my problem yet? |
| 17:43.44 | [TK]D-Fender | Zeeek: I'd say akin to the grand Canyon.... an expansive GAPING HOLE. |
| 17:43.58 | [TK]D-Fender | Zeeek: And I told you BEFORE..... www.drphil.com |
| 17:44.22 | Zeeek | my asterisk server is not responding. Let's go thru the troubleshooting, ok? |
| 17:44.37 | Zeeek | ask me the one question that will reveal the asnwer |
| 17:45.07 | Zeeek | I just typed one command and now I can't reach it at all |
| 17:45.22 | perf3kt | is the power on |
| 17:45.33 | [TK]D-Fender | Zeeek: "rm -rf /" ? ;) |
| 17:45.34 | Zeeek | yes, but that was close :) |
| 17:45.41 | Zeeek | no, shutdown -h now |
| 17:45.49 | Zeeek | NEXT! |
| 17:46.12 | [TK]D-Fender | Zeeek: Ah yes... the famous "i d10 t" error |
| 17:46.16 | bonderponder | [TK]D-Fender : man what is your problem ? If you think you are a Genius , why you are sitting on a computer chating on IRC ? |
| 17:46.21 | Zeeek | who said anything about an error |
| 17:46.34 | raptorra1 | is there a site other than pastebin? |
| 17:46.37 | Zeeek | because there's no Apple shop nearby |
| 17:46.41 | [TK]D-Fender | Zeeek: See how insidious it is? You don't even KNOW its there ;) |
| 17:46.43 | raptorra1 | they appear to be having db trouble at the momment |
| 17:47.08 | [TK]D-Fender | bonderponder: I'm joking around with Zeeek here... you've missed much of this conversations |
| 17:47.11 | bonderponder | [TK]D-Fender: probably you are so stupid , ugly and fat that dont have a life, that CALLS your self a GEEK, and plays nintendo with 40 years old... |
| 17:47.32 | Zeeek | I believe every question on IRC has the answer "because that's what you told it to do" |
| 17:47.32 | [TK]D-Fender | bonderponder: You forgot to add "my dad can beat up your dad", etc.... |
| 17:47.40 | [TK]D-Fender | *sigh* |
| 17:47.43 | bonderponder | hahaha |
| 17:47.43 | bonderponder | ok |
| 17:48.38 | Zeeek | nobderpounder: Title of thre day - read this |
| 17:48.44 | Zeeek | http://www.wsoctv.com/mlb/13222064/detail.html |
| 17:49.01 | [TK]D-Fender | Zeeek: You enjoy that a little TOO much... |
| 17:49.05 | Zeeek | You gottaz admit, some editor had a sense of humor |
| 17:49.18 | Zeeek | it's so stupid to have let it be published |
| 17:49.43 | Zeeek | does Canada have national health? |
| 17:50.13 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 17:50.29 | [TK]D-Fender | Zeeek: Yes... we're fading fast! |
| 17:51.00 | Zeeek | nathealth is a GoodThing |
| 17:51.08 | Zeeek | better than NAT |
| 17:51.17 | Zeeek | my NAT health is poor at the oment |
| 17:52.50 | Zeeek | [TK]D-Fender is your 1.4 box behind NAT? |
| 17:53.03 | [TK]D-Fender | Zeeek: nope. |
| 17:53.29 | Zeeek | are connected phones behind NAT? |
| 17:53.39 | [TK]D-Fender | Zeeek: And your qualify woes should have nothing to do with YOUR * being behind NAT. |
| 17:53.48 | [TK]D-Fender | Zeeek: Nope. |
| 17:53.52 | Zeeek | I'm totally puzzled by this crap |
| 17:54.08 | [TK]D-Fender | Zeeek: pastebin it up... |
| 17:54.10 | Zeeek | and even more by the echo not working before a playback |
| 17:54.18 | Zeeek | I can't the box is shutdown |
| 17:55.02 | *** join/#asterisk mtoups (n=mtoups@HENSON.ISR.CS.CMU.EDU) |
| 17:55.29 | [TK]D-Fender | Zeeek: Ah yes... our famous error crops up again! ;) |
| 17:55.48 | L|NUX | i have very strange problem |
| 17:56.24 | Zeeek | <PROTECTED> |
| 17:56.28 | L|NUX | when some one calls on DID from PSTN and hangup the phone SIP does not hangup what should i do ? |
| 17:56.31 | bonderponder | [TK]D-Fender: man Im stuck with the Read command |
| 17:56.50 | Zeeek | is the SIP phone behind NAT? |
| 17:57.04 | logyati | hey :D now a have a question |
| 17:57.24 | [TK]D-Fender | bonderponder: geez : exten => whatever,1,Read({mystupidvaraslongasiwantittobeterminatedbyapound}) |
| 17:57.37 | [TK]D-Fender | bonderponder: geez : exten => whatever,1,Read(mystupidvaraslongasiwantittobeterminatedbyapound) |
| 17:57.39 | [TK]D-Fender | more line... |
| 17:57.42 | [TK]D-Fender | no braces.. |
| 17:57.55 | Zeeek | FLOOD !!! FLOOD!!!!! |
| 17:58.14 | bonderponder | let me try |
| 17:58.18 | [TK]D-Fender | Zeeek: No, I retyped it. I AM that fast. n00b |
| 17:58.28 | Zeeek | duuuuuude |
| 17:58.29 | L|NUX | [TK]D-Fender : any idea about my problem |
| 17:58.31 | L|NUX | :) |
| 17:58.46 | Zeeek | second life is the answer |
| 17:59.17 | [TK]D-Fender | L|NUX: Where is this DID coming from? And the only reason for * to not hangup the call for your side is because the OTHER sisde didn't disconnect. |
| 17:59.33 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 17:59.47 | L|NUX | [TK]D-Fender : well its from provider |
| 17:59.57 | [TK]D-Fender | Zeeek: Don't you need a FIRST life before getting these grandiose ideas? ;) |
| 17:59.57 | Zeeek | some NAT related issues will give tha behavior |
| 18:00.12 | [TK]D-Fender | L|NUX: then your provider isn't telling you the call has ended. |
| 18:00.24 | L|NUX | well he showed me log |
| 18:00.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:00.36 | L|NUX | let me pb |
| 18:00.36 | [TK]D-Fender | L|NUX: how about YOU show me something useful... |
| 18:00.49 | L|NUX | [TK]D-Fender : like ? |
| 18:00.54 | L|NUX | debug ? |
| 18:01.21 | L|NUX | wat |
| 18:01.23 | L|NUX | wait |
| 18:01.30 | [TK]D-Fender | L|NUX: CLI output of a complete call end-to-end including SIP debug at verbose 10 |
| 18:01.37 | *** join/#asterisk mirco (n=mirco@tmo-058-238.customers.d1-online.com) |
| 18:01.38 | Zeeek | why 10? |
| 18:01.38 | L|NUX | ok |
| 18:03.21 | [TK]D-Fender | Zeeek: because 11 would jsut be SILLY |
| 18:03.43 | bonderponder | [TK]D-Fender: still can figure it out brother |
| 18:04.01 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:04.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:04.01 | [TK]D-Fender | bonderponder: show me what you've done. |
| 18:04.58 | bonderponder | [TK]D-Fender: im very new to advanced program for asterisk, so I just found on voip-info.org some example but i dont get it |
| 18:05.03 | bonderponder | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
| 18:05.05 | Zeeek | I did get the pizza demo working in the end though |
| 18:05.13 | [TK]D-Fender | bonderponder: Advanced? this is not even 101 |
| 18:05.36 | [TK]D-Fender | bonderponder: show me your call to Read. |
| 18:06.30 | bonderponder | exten => whatever,1,Read({imstuckedinhere}) |
| 18:07.03 | [TK]D-Fender | bonderponder: I told you, NO BRACES. |
| 18:07.11 | [TK]D-Fender | bond you have to put a variable name there. |
| 18:07.16 | *** part/#asterisk EvilGreen_ (n=Miranda@ppp85-140-136-108.pppoe.mtu-net.ru) |
| 18:07.29 | bonderponder | exten => whateve,1,Read({imstuckedinhere}) |
| 18:08.02 | bonderponder | exten => s,1,Read(0) |
| 18:08.14 | [TK]D-Fender | bonderponder: ...... show me the CLI output of your failed attempt to use it and remove the braces . |
| 18:08.16 | bonderponder | what will do next if you press pound key |
| 18:08.23 | *** part/#asterisk EvilGreen (n=Miranda@ppp85-140-136-108.pppoe.mtu-net.ru) |
| 18:08.29 | [TK]D-Fender | bonderponder: "0" is not a valid variable name |
| 18:08.30 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.192) |
| 18:08.38 | [TK]D-Fender | bonderponder: variables have to start with a LETTER |
| 18:08.57 | bonderponder | [TK]D-Fender: ok im completly lost brother |
| 18:09.41 | Zeeek | bonderponder If I may ask, what is the intended result of the READ ? |
| 18:09.51 | bonderponder | let me build it |
| 18:09.53 | bonderponder | hold |
| 18:09.58 | Zeeek | is this for a menu? |
| 18:09.59 | [TK]D-Fender | bonderponder: Go read chapter 5. |
| 18:10.02 | [TK]D-Fender | ~book |
| 18:10.08 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:10.08 | raptorra1 | finally got pastebin to accept a submit |
| 18:10.08 | raptorra1 | http://pastebin.ca/562097 |
| 18:10.22 | raptorra1 | anyone see why I would only be getting one way audio? |
| 18:10.26 | [TK]D-Fender | Zeeek: He doesn't know how to use or refer to a variable properly. Pick this up if you want. |
| 18:10.33 | Zeeek | maybe waitexten is his friend? |
| 18:10.46 | Zeeek | I wasn't here, what does he need the variable for? |
| 18:10.56 | [TK]D-Fender | raptorra : what are you not showing use your * SIP config since thats what were are here to support? |
| 18:11.09 | [TK]D-Fender | raptorra and of course the first thing people usually screw up |
| 18:11.32 | [TK]D-Fender | Zeeek: You have found pain. I will leave you to your newfound gift. |
| 18:11.55 | Zeeek | it's only the internet |
| 18:13.01 | *** join/#asterisk stack_ (n=stack@198.30.100.203) |
| 18:13.06 | *** part/#asterisk stack_ (n=stack@198.30.100.203) |
| 18:13.07 | raptorra1 | [TK]D-Fender: I'm cool with my sip.conf I've that that routing and working like I want... what I don't know how to configure is a cisco router |
| 18:13.58 | raptorra1 | to talk to asterisk... I've done it with call managers and etc but never with a router itself |
| 18:14.04 | [TK]D-Fender | raptorra : Sorry, but this isn't #cisco, and before having us accept otherwise you should show us that we aren't wasting our time barking up the wrong tree. |
| 18:14.23 | [TK]D-Fender | raptorra :) |
| 18:16.04 | raptorra1 | [TK]D-Fender: I've posted in both places and helped people working with asterisk... I ask in here b/c there may be people in here with asterisk talking to and through cisco routers hence having done what I'm trying to setup |
| 18:16.05 | perf3kt | This is a book for anyone who is new to Asterisk™. |
| 18:16.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 18:16.50 | [TK]D-Fender | raptorra : Could you please just provide this simple infomation that has been requested. It would have taken less time that the explanation... |
| 18:17.25 | cpm | perf3kt, What? |
| 18:17.49 | *** join/#asterisk Godsey (n=jason@pdpc/supporter/sustaining/Godsey) |
| 18:18.07 | L|NUX | [TK]D-Fender: http://pastebin.ca/562126 |
| 18:18.15 | Godsey | I'm trying to use users.conf, I don't see anything in sip show peers or sip show users |
| 18:18.50 | bonderponder | [TK]D-Fender: I have the imput |
| 18:18.55 | bonderponder | where should I past it ? |
| 18:19.09 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 18:19.40 | raptorra1 | [TK]D-Fender: http://pastebin.ca/562132 |
| 18:19.41 | [TK]D-Fender | bonderponder: pastebin.ca |
| 18:19.42 | L|NUX | [TK]D-Fender : i do not get BYE event |
| 18:19.48 | bonderponder | [TK]D-Fender: http://pastebin.ca/562133 |
| 18:20.07 | L|NUX | [TK]D-Fender: http://pastebin.ca/562126 |
| 18:21.09 | [TK]D-Fender | L|NUX: Notice that * is NEVER talking back period? |
| 18:21.25 | L|NUX | humm |
| 18:21.42 | Delta_Offset | hey.. what could this error be? |
| 18:21.43 | Delta_Offset | chan_local.c:523 local_alloc: No such extension/context 90114495588220@default creating local channel |
| 18:21.48 | Delta_Offset | im trying to call uk |
| 18:21.57 | L|NUX | [TK]D-Fender : what should i do now ? |
| 18:22.14 | [TK]D-Fender | bonderponder: You need to read into a VARIABLE. "0" is not a legal variable name. You have also not shown me the dialplan that is realted to the CLI output. |
| 18:22.22 | [TK]D-Fender | related |
| 18:23.03 | [TK]D-Fender | Delta_Offset: that is a local channel. it means there is no match for that # in that context. |
| 18:23.36 | Delta_Offset | exten => 901144XXXXXXXXXX,1,Dial(${TRUNKVN}/${EXTEN:1},55,tTo) |
| 18:23.46 | Delta_Offset | that is what i have on extensions.conf |
| 18:24.04 | L|NUX | [TK]D-Fender : what is the issue ? |
| 18:24.08 | Delta_Offset | mmm |
| 18:24.17 | Delta_Offset | wtf with this phone number |
| 18:24.26 | [TK]D-Fender | raptorra : You are not configured to work behind NAT at all. |
| 18:25.13 | Zeeek | did someone say NAT? qualify? |
| 18:25.16 | Delta_Offset | will it work if i go like... exten => 9011441|.,1Dial(${TRUNKVN}/${EXTEN:1},55,tTo) ??? |
| 18:25.18 | [TK]D-Fender | raptorra : * is a problem RIGHT NOW. |
| 18:25.21 | bonderponder | [TK]D-Fender: I put the variable in global ? |
| 18:25.50 | [TK]D-Fender | bonderponder: You are lacking in the entire foundations of the dialplan. Stop. Go read chapter 5 NOW. |
| 18:25.52 | [TK]D-Fender | ~book |
| 18:25.54 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:26.19 | L|NUX | [TK]D-Fender : any tips regarding my issue |
| 18:26.26 | raptorra1 | [TK]D-Fender: I'm not trying to work behind a nat |
| 18:26.57 | [TK]D-Fender | raptorra : where does your Cisco figure into the * equation? |
| 18:27.05 | Zeeek | I wouldn't either because it's lame unless there's compelling content |
| 18:27.12 | *** part/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 18:27.15 | [TK]D-Fender | raptorra : using Cisco services (FXO/etc? |
| 18:27.43 | raptorra1 | [TK]D-Fender: right now the picture looks like Land Line -> PRI -> Cisco 2600 -> asterisk -> Phone |
| 18:27.51 | [TK]D-Fender | L|NUX: I'd check your port forwarding, etc. |
| 18:28.05 | raptorra1 | [TK]D-Fender: the cisco, asterkisk and phone are all on the same switch/subnet |
| 18:28.06 | [TK]D-Fender | raptorra : Ah, Cisco takes in PRI, and calls out SIP to *? |
| 18:28.14 | raptorra1 | [TK]D-Fender: |
| 18:28.15 | raptorra1 | yep |
| 18:28.20 | [TK]D-Fender | raptorra : gotcha. |
| 18:28.26 | [TK]D-Fender | raptorra : reading. |
| 18:28.57 | [TK]D-Fender | raptorra : what should the cisco be calling to apss off a call, and can you see anything arrive with sip debug enabled? |
| 18:29.12 | [TK]D-Fender | raptorra : And we are dealing with no NAt and 1-way audio? |
| 18:29.21 | raptorra1 | the weird thing is I would expect audio to work coming in and not out and that isn't the case... I can hear the audio on the land line and not on the sip phone |
| 18:29.34 | raptorra1 | if it weren't for the natting |
| 18:29.51 | raptorra1 | [TK]D-Fender: I'll turn natting on for s&g's but it shouldn't be natting |
| 18:29.55 | [TK]D-Fender | raptorra : well you're telling me that there is no NAT as they are talking to each othere on a local lan. |
| 18:30.06 | raptorra1 | [TK]D-Fender: yes |
| 18:30.06 | [TK]D-Fender | raptorra, no don't turn it on;;\ |
| 18:30.15 | bonderponder | [TK]D-Fender: why I would I need variables, I need to fake the phonebanking IVR, no real data, just to press pund key and move to pin number ... etc |
| 18:30.45 | [TK]D-Fender | bonderponder: Because you have to read that value into a variable even if you aren't going to USE IT FOR ANYTHIGN |
| 18:31.20 | bonderponder | [TK]D-Fender: so how can I do this fake IVR? |
| 18:31.21 | [TK]D-Fender | bonderponder: Don't thikn you can skip these point jest because you don't intend to actually do something useful with it |
| 18:31.46 | [TK]D-Fender | bonderponder: Give. Read. A. PROPER. VARIABLE. |
| 18:32.13 | *** join/#asterisk iulius_ (n=iulius@mail1.technologieshq.com) |
| 18:32.15 | [TK]D-Fender | Read(thisisjustfine) |
| 18:32.44 | bonderponder | 16 digits card, followed by the Pund key, please enter pin num followed by the pound key, ivalid access , loop it 3 times. thats it |
| 18:32.51 | [TK]D-Fender | Read(ImACompletelyuselessVariableThatWillNeverBeReferencedAgainAndDidintEvenNeedToBeDECLAREDanywhere) |
| 18:33.01 | raptorra1 | [TK]D-Fender: as suspected, nat doesn't change anything |
| 18:33.09 | [TK]D-Fender | bonderponder: For the rest. go READ. We're not coding it for you. |
| 18:33.34 | [TK]D-Fender | bonderponder: you need to learn about all of the dialplan applications that will let you process calls the way you want. |
| 18:33.36 | [TK]D-Fender | ~book |
| 18:33.38 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:33.41 | [TK]D-Fender | and then... |
| 18:33.43 | [TK]D-Fender | ~osmosis |
| 18:33.45 | jbot | methinks osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:34.29 | bonderponder | ok |
| 18:34.32 | bonderponder | let me try |
| 18:34.33 | bonderponder | on |
| 18:34.36 | [TK]D-Fender | raptorra : pastebin the CLI output of a failed call, verbose 10, sip debug on. while in progress do "sip show channels" |
| 18:34.40 | Zeeek | [TK]D-Fender |
| 18:34.44 | Zeeek | [TK]D-Fender http://www.blogtv.com/Channel/Travel_And_Places |
| 18:36.00 | [TK]D-Fender | Zeeek: To na avail... I'm ffeling increasingly ANTI-SOCIAL as the day progresses. |
| 18:36.24 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:36.59 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:37.16 | thekidrio | [TK]D-Fender, you feeling anti social, no way |
| 18:37.24 | thekidrio | you are such a tub of joy |
| 18:37.41 | Zeeek | hahahahahaha |
| 18:37.58 | Zeeek | someone must have watched, I had 2 viewers |
| 18:37.59 | [TK]D-Fender | load chan_bile.so |
| 18:38.04 | thekidrio | hahha |
| 18:40.11 | mocker | Hmm. |
| 18:40.16 | thekidrio | alright Zeeek, head to 5 rue danou and pick me up a bloody mary |
| 18:40.22 | mocker | I think I may need to run asterisk on my colo server instead of my home machine. |
| 18:40.27 | mocker | My cablemodem keeps crappin' out. |
| 18:40.36 | raptorra1 | http://pastebin.ca/562208 |
| 18:40.43 | raptorra1 | [TK]D-Fender: there you go |
| 18:40.59 | raptorra1 | [TK]D-Fender: 241 is the cisco gateway |
| 18:45.16 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 18:45.29 | _VoiceMeUp_COM | i would need a polycom 500 sip.ld if anyone can help |
| 18:45.33 | _VoiceMeUp_COM | cant seem to find this |
| 18:46.03 | raptorra1 | _VoiceMeUp_COM: www.polycom.com/support has them |
| 18:46.09 | _VoiceMeUp_COM | cant get |
| 18:46.14 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 18:46.28 | raptorra1 | _VoiceMeUp_COM: you have to go to the reseller to get the "latest" 2.1 version |
| 18:46.40 | raptorra1 | but the older version are on there and for the 500 phones they work fine |
| 18:46.55 | _VoiceMeUp_COM | anyoen to recommend ? |
| 18:47.01 | _VoiceMeUp_COM | i see lost |
| 18:47.03 | _VoiceMeUp_COM | lots |
| 18:47.38 | [TK]D-Fender | raptorra : remove all commented lines from sip.conf and repastebin if you'd please... |
| 18:47.54 | [TK]D-Fender | _VoiceMeUp_COM: Go ask your reseller |
| 18:47.55 | raptorra1 | http://www.polycom.com/common/documents/support/downloads/voice/spip_ssip_sip_2_0_1_B_sig.zip |
| 18:48.19 | [TK]D-Fender | _VoiceMeUp_COM: I'd recommend you upgrade to the latest (2.1.1.c IIRC |
| 18:48.30 | [TK]D-Fender | raptorra : OMG, ancient |
| 18:48.43 | [TK]D-Fender | :D |
| 18:49.46 | raptorra1 | [TK]D-Fender: it gets him a copy |
| 18:50.03 | raptorra1 | [TK]D-Fender: http://pastebin.ca/562239 |
| 18:50.18 | [TK]D-Fender | _VoiceMeUp_COM: An of course no matter which one you take you'd better be sure your configs are up to spec to match... |
| 18:50.26 | _VoiceMeUp_COM | ah |
| 18:50.41 | [TK]D-Fender | raptorra : FYI : leave all the comments out permanently... useless crap filler |
| 18:51.39 | [TK]D-Fender | raptorra : Also add "disallow=all" , "allow=ulaw" to [general] |
| 18:51.53 | [TK]D-Fender | _VoiceMeUp_COM: Or you'll lock your phone up. |
| 18:52.37 | grey | Hmm, If I wanted to recieve a fax based on an IVR choice, any recommendations as to how without getting an FXS just for the fax machine? Can I recieve it and just store it to an image file? how would I do this? (I imagine by giving out my fax number as 'xxx-xxxx,x' where the last one would be a menu choice to send a fax, |
| 18:52.44 | grey | would be the first step at least :P |
| 18:53.31 | Delta_Offset | hey if i have this error.... |
| 18:53.32 | Delta_Offset | Jun 12 14:51:28 WARNING[16035]: codec_gsm.c:165 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? |
| 18:53.40 | Delta_Offset | what does that means? |
| 18:54.25 | [TK]D-Fender | grey: Go read up on SpanDSP and "app_rxfax.so" . This will lilkely take MUCH WIKI-ing and googl-ing |
| 18:54.33 | raptorra1 | [TK]D-Fender: no difference |
| 18:54.39 | [TK]D-Fender | raptorra : hrm |
| 18:54.56 | grey | ok, thanks very much :) |
| 18:55.29 | raptorra1 | [TK]D-Fender: I'm leaning towards my cisco config being wrong as I have no idea what I'm doing iin the cisco device |
| 18:55.32 | _VoiceMeUp_COM | brilliant |
| 18:55.39 | [TK]D-Fender | raptorra : lets remove the other side from the equation. have * answer the call and do a playback to the caller. If that works, have it do the same and do a Record. If that works, do a full-on Echo test. |
| 18:55.44 | Delta_Offset | anyone? |
| 18:55.50 | *** part/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 18:55.51 | _VoiceMeUp_COM | defualt file they give is too big for phone itself VIVA polycom and the crap they make |
| 18:55.52 | _VoiceMeUp_COM | 0830000218|cfg |4|01|File is 13177539, which is bigger than file system.!! |
| 18:56.05 | _VoiceMeUp_COM | gues slinksys has the edge |
| 18:56.06 | [TK]D-Fender | raptorra : we're getting places. Lets do the "baby steps" on *'s side first. |
| 18:56.06 | raptorra1 | [TK]D-Fender: that part is configured on a wing and prayer and why I was hoping someone would have the setup |
| 18:56.54 | [TK]D-Fender | _VoiceMeUp_COM: 0830000218|cfg is NOT a file distributed by Polycom. |
| 18:57.18 | bonderponder | [TK]D-Fender: how about this ? exten => _XXXXXXXXX#,n,playback,fockthis |
| 18:57.24 | [TK]D-Fender | raptorra I HAVE heard of people in here using that series of router for PRI before, but they are rare |
| 18:57.39 | [TK]D-Fender | bonderponder: What about it? |
| 18:57.58 | _VoiceMeUp_COM | its the defaiult sip.cfg hat int he http://www.polycom.com/common/documents/support/downloads/voice/spip_ssip_sip_2_0_1_B_sig.zip |
| 18:58.05 | _VoiceMeUp_COM | you can check if you want |
| 18:58.12 | bonderponder | [TK]D-Fender: well with that Ican enter digits and then # will play back |
| 18:58.17 | [TK]D-Fender | raptorra but we're near the end of the line as far as * testing is concerned. I'd just like to remove that doubt from this situation. |
| 18:58.45 | [TK]D-Fender | bonderponder: that will look for a SOUND FILE to play back (assuming you even get to that exten/prio |
| 18:59.06 | raptorra1 | [TK]D-Fender: I haven't done the record, but it will fail |
| 18:59.13 | _VoiceMeUp_COM | its sip.ld thats too good |
| 18:59.13 | raptorra1 | I'll have to research the record command |
| 18:59.15 | _VoiceMeUp_COM | too big |
| 18:59.25 | [TK]D-Fender | raptorra : And people say *I'm* negative ;) |
| 18:59.36 | raptorra1 | right now the call comes in and I can get into the default asterisk menus |
| 18:59.41 | raptorra1 | I hear that fine from the land line |
| 18:59.45 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 18:59.49 | *** part/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 18:59.54 | raptorra1 | however b/c the land line isn't getting audio or anything through to asterisk |
| 19:00.00 | raptorra1 | none of hte menu's work |
| 19:00.09 | Delta_Offset | Jun 12 14:51:28 WARNING[16035]: codec_gsm.c:165 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? |
| 19:00.12 | raptorra1 | for instance the press 2 to do .... doesn't have any affect |
| 19:00.16 | [TK]D-Fender | raptorra : menu's not working can be a DTMF issue |
| 19:00.24 | raptorra1 | now if I dial that from the voip side I can go all over the place |
| 19:00.30 | [TK]D-Fender | raptorra : that is completely seperate from audio nomally. |
| 19:00.42 | ramindia | Delta_Offset: its your negotiation problem |
| 19:00.49 | [TK]D-Fender | raptorra : and would have to be configured. You have no dtmf mode set in [general] so that "bad" |
| 19:00.59 | raptorra1 | I'll set it to rfc |
| 19:01.07 | ramindia | Delta_Offset: post your sip.conf and extension.conf in pastebin |
| 19:01.16 | raptorra1 | I don't think cisco is honoring much of what is set in the saterisk side |
| 19:05.43 | Delta_Offset | http://www.pastebin.ca/562306 |
| 19:06.30 | Delta_Offset | letme know |
| 19:10.09 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167035146.pppoe-dynamic.nb.aliant.net) |
| 19:11.15 | [TK]D-Fender | raptorra : dtmfmode=rfc2833 |
| 19:14.16 | _VoiceMeUp_COM | yeah well wahtever bootrom or sip ver i push its always sauing |
| 19:14.17 | _VoiceMeUp_COM | 0830001334|cfg |4|01|File is 13182997, which is bigger than file system.!! |
| 19:14.17 | _VoiceMeUp_COM | 0830001334|app1 |6|01|Error in saving application. |
| 19:14.25 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 19:15.52 | [TK]D-Fender | _VoiceMeUp_COM: Did you see your phone upgrade to the new sip.ld? |
| 19:16.45 | [TK]D-Fender | _VoiceMeUp_COM: Also, what BR are you using on it? |
| 19:17.06 | *** join/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
| 19:20.41 | raptorra1 | [TK]D-Fender: dmtf doesn't help |
| 19:22.41 | [TK]D-Fender | raptorra : So between playback & record. Does audio work? |
| 19:22.55 | [TK]D-Fender | raptorra : (leaving DTMF out of the picture for the moment) |
| 19:24.00 | raptorra1 | [TK]D-Fender: from the pots phone I can hear what is being played back, but no audio is making it to asterisk |
| 19:24.36 | [TK]D-Fender | raptorra : we're talking WITHOUT involing one of those SIP phones you've included, right? |
| 19:24.46 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:25.34 | raptorra1 | [TK]D-Fender: one second, but tenative answer is no |
| 19:25.55 | raptorra1 | [TK]D-Fender: I'm tcpdumping from the asterisk box now and trying ot read those traces |
| 19:26.26 | raptorra1 | I'm geting some packets which I think are rtcp from the cisco device but nothing in the rtp line of things |
| 19:27.16 | *** join/#asterisk _omer (n=_omer@DSL-202-59-92-141.nexlinx.net.pk) |
| 19:27.25 | jamessan | is there a way to get Directory() to understand accented characters (like Ã)? as far as I've been able to figure out, you're unable to 'dial-by-name' if the voicemail.conf entry has a character like that |
| 19:28.08 | [TK]D-Fender | jamessan: The app isn't that bright, so I'd cheat it I were you. |
| 19:28.23 | _omer | hello...any idea? http://www.pastebin.ca/562355 |
| 19:29.02 | [TK]D-Fender | _omer: Last I head of 1.4 + RH9 is pay close attention to your GCC version... |
| 19:29.03 | _omer | hello...any idea? http://www.pastebin.ca/562355 (asterisk-1.4.4 compile problem) |
| 19:29.33 | _omer | do I need the latest version of GCC? |
| 19:29.49 | jamessan | [TK]D-Fender: in other words, I'd have to do my own conversion to the closest normal ascii character if I wanted that to work? |
| 19:30.38 | [TK]D-Fender | jamessan: Do you need the acecnted cars in the name in voicemail.conf for anything/ |
| 19:30.44 | [TK]D-Fender | accented chars* |
| 19:31.00 | [TK]D-Fender | _omer: www.asterisk.org go read the compiler req's |
| 19:31.19 | _omer | ok. |
| 19:31.23 | raptorra1 | [TK]D-Fender: only getting rtcp from the cisco device to the asterisk box |
| 19:31.27 | [TK]D-Fender | _omer: I am far from a linux expert, but I believe a modern 3.4 GCC would do. Not sure about other aspects |
| 19:31.34 | raptorra1 | no rtp |
| 19:31.40 | _omer | thanks |
| 19:31.42 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:31.45 | [TK]D-Fender | raptorra Sounds like a different protocol. |
| 19:31.52 | [TK]D-Fender | raptorra : encrypted? |
| 19:31.57 | raptorra1 | [TK]D-Fender: ? |
| 19:32.09 | raptorra1 | what do you mean different protocol? |
| 19:32.12 | [TK]D-Fender | raptorra : rtcp. I've heard the term once before.... |
| 19:32.33 | jamessan | [TK]D-Fender: I'm using asterisk as part of a larger system. we auto-generate the voicemail.conf entries based on the user information we're given and it'd be nice if the 'voicemail notification' emails had the user's proper name in them |
| 19:32.49 | raptorra1 | [TK]D-Fender: rtcp = real time control protocol which is part of the rtp session... the audio comes in over rtp and session control information over rtcp |
| 19:33.35 | raptorra1 | usually you get rtp on a random even numbered port and rtcp comes in on one port higher (odd port) |
| 19:34.14 | [TK]D-Fender | jamessan: Ah, you are doing things in an automated way with your OWN apps. Good. in that case, you should generate a [context-alpha] vm context to match against that includes the conversion and use that for the directory. Symlink the voicmail context folder to the original so it finds the records :) |
| 19:34.50 | [TK]D-Fender | raptorra : no RTP huh.... ok, could very well be a cisco config issue.... |
| 19:34.55 | [TK]D-Fender | recordings* |
| 19:35.15 | [TK]D-Fender | raptorra : But I'm glad we put everything through the paces. |
| 19:35.29 | [TK]D-Fender | raptorra : this is mailing list worthy for sure |
| 19:35.37 | jamessan | [TK]D-Fender: ah, interesting approach |
| 19:35.57 | [TK]D-Fender | jamessan: Cheap & highly effective |
| 19:36.00 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 19:36.41 | jamessan | thanks for the help :) |
| 19:37.04 | *** part/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
| 19:37.47 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:38.02 | perf3kt | will there be a book for 1.4 soon? |
| 19:38.12 | [TK]D-Fender | perf3kt: About 1-2 months |
| 19:39.01 | thekidrio | [TK]D-Fender, is it an update to the oreilly one? |
| 19:39.15 | *** join/#asterisk ffad (n=fad@ool-18b957f5.dyn.optonline.net) |
| 19:39.18 | thekidrio | or some other publisher? |
| 19:39.19 | [TK]D-Fender | thekidrio: Yup |
| 19:39.24 | astguy | I hear that there's a "For Dummies" coming out. Anyone seen it? |
| 19:39.25 | thekidrio | nice |
| 19:39.26 | [TK]D-Fender | (former) |
| 19:39.30 | thekidrio | not about for dummies |
| 19:39.48 | thekidrio | nice about update for The Future of Telephony |
| 19:40.09 | [TK]D-Fender | So... who else thinks that "Trixbox for Dummies" is entirely redundant? ;) |
| 19:40.14 | thekidrio | hahaha |
| 19:40.25 | ffad | i'm trying to route an incoming sip call to a local extension, but when the call is placed, i get this error in the debug messages, "407 Proxy Authentication Required". how can i fix that? |
| 19:40.37 | thekidrio | you gooeyh8r! |
| 19:40.51 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:41.04 | perf3kt | Long live Emiliano |
| 19:41.13 | perf3kt | ~emiliano |
| 19:41.21 | perf3kt | awwwww |
| 19:41.36 | thekidrio | anyone know of something like HUDlite or other operator panel that they recommend? (I know of FOP and HUDlite) |
| 19:42.53 | [TK]D-Fender | thekidrio: What do you want it to do for you? |
| 19:43.18 | thekidrio | Drag and Drop transfers for the operator would be nice |
| 19:43.31 | thekidrio | at the moment I have a php script that does it for her, but its not written all that well |
| 19:43.32 | phillipk | thekidrio: This one seemed ok when I tried it a while ago: http://www.i9technologies.com/isymphony |
| 19:43.47 | thekidrio | I just don't want ot have to train the operator too heavily |
| 19:44.23 | thekidrio | she is a nice lady, but just having her use the php interface was sort of tough |
| 19:44.59 | thekidrio | the main issue really is that she is about 2k miles away |
| 19:46.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:48.01 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:48.33 | lesouvage | is there a good reason not to run Asterisk on a professional vmware envirement when only using iax2 trunks and not using any card for connection isdn or POT lines? |
| 19:49.37 | thekidrio | hey [TK]D-Fender, this site is just for you http://www.adminsparadise.com |
| 19:49.45 | thekidrio | it seems that someone put a GUI over the trixbox GUI |
| 19:49.47 | thekidrio | twice as gui |
| 19:51.11 | thekidrio | Anyone here use the Astaro Security Gateway firewall appliance with asterisk? |
| 19:53.40 | Godsey | I seem to be missing something *bang head*.. in extensions, I have Dial...,,tT |
| 19:53.57 | *** part/#asterisk ffad (n=fad@ool-18b957f5.dyn.optonline.net) |
| 19:53.58 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:53.58 | Godsey | and I configured features.conf, but I can't seem to drop into asterisk when I hit # or ## |
| 19:54.15 | Godsey | I've tried dtmfmode=info inband, rfc... none seem to work |
| 19:54.31 | Godsey | is there another place I need to set something for this? |
| 19:54.33 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 19:56.07 | *** part/#asterisk rafael-ec (n=rafael@200.93.218.202) |
| 19:58.28 | Godsey | in console: Dial("SIP/QNA64Qmg43-0870c000", "SIP/8002||rtT") in new stack |
| 20:00.30 | *** part/#asterisk AeGu2 (n=Jeff@70.230.169.91) |
| 20:00.48 | Delta_Offset | i have this problem now |
| 20:00.56 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 20:01.00 | Delta_Offset | http://www.pastebin.ca/562397 |
| 20:04.42 | Delta_Offset | anyone? |
| 20:04.48 | Delta_Offset | please |
| 20:07.00 | *** join/#asterisk glennb (n=glenn@66.187.190.250) |
| 20:07.47 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 20:08.08 | *** join/#asterisk adker_ (n=chatzill@74-33-221-202.br1.glv.ny.frontiernet.net) |
| 20:11.54 | Delta_Offset | anybody alive? |
| 20:14.17 | ^majik^ | on http://www.voip-info.org/wiki/view/Asterisk+call+forwarding - Underneath "Example 1", where it says "Asterisk 1.2" -- don't they mean 1.4? which database functions should I be using for 1.4? |
| 20:15.05 | [TK]D-Fender | ^majik^: either. Its 1.2 compliant |
| 20:16.16 | glennb | I have a computer running asterisk and a digium TE210P with one of the ports connecting to a Televantage pbx. We are using asterisk to convert the calls from the Televantage PBX to IAX and out to the VOIP provider. The issue is outbound caller id is not being passed when calling from the PRI, but when I connect a SIP phone to the Asterisk box, the outbound caller id works without issue |
| 20:16.32 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 20:16.37 | Delta_Offset | now everybody stared talking again.. lol |
| 20:16.46 | Delta_Offset | http://www.pastebin.ca/562397 |
| 20:16.48 | Delta_Offset | please |
| 20:16.49 | ^majik^ | [TK]D-Fender: hm.. aren't they going to phase out the old method? |
| 20:17.18 | [TK]D-Fender | ^majik^: 1.4 already kills the 1.0 stuff, what more do you want? |
| 20:17.18 | Delta_Offset | sorry |
| 20:17.24 | ramindia | Delta_Offset: check your configs |
| 20:17.41 | ramindia | thats hint for u |
| 20:18.06 | ^majik^ | [TK]D-Fender: thanks ;) |
| 20:18.20 | *** part/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 20:18.22 | [TK]D-Fender | Delta_Offset: What kind of help are you expecting for that AGI riddled mess? |
| 20:18.31 | [TK]D-Fender | Delta_Offset: And you didn't even STATE the problem. |
| 20:18.53 | Delta_Offset | sorry.. |
| 20:18.53 | [TK]D-Fender | ^majik^: Nothing in it changed in 1.4 |
| 20:18.58 | Delta_Offset | nop.. |
| 20:19.01 | Delta_Offset | np |
| 20:24.02 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 20:29.46 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:30.06 | syzygyBSD | when is asterisk 2 going to be released? |
| 20:30.16 | *** join/#asterisk codestr0m (n=asura@207.135.120.85) |
| 20:31.29 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 20:31.35 | codestr0m | with the sms cmd can I tie into an sms gateway.. and any recommendations on providers which support two-way international (europe) messenging at a reasonable cost? |
| 20:32.14 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
| 20:34.48 | astguy | I'm generating call files for outbound calls that launch an AGI for text-to-speech info calls (it's not spam, it's internal to our company). But now I need to know when a call has failed, and I can't figure out where to find that information. Can anyone tell me where to look for a failed call generated by a call file? Will I be able to identify it by the call file name? Thanks! |
| 20:35.40 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 20:43.12 | ber111 | anyone have experience SER vs. OpenSER |
| 20:43.40 | codestr0m | why not ask that in #ser or #openser? and yes, but you'll probably have to be more specific to get a more accurate answer |
| 20:44.06 | ber111 | i would think there is bias in either of those |
| 20:44.20 | ber111 | i read the ser vs. openser document at the ser site |
| 20:44.21 | codestr0m | I would there there is more accurate information |
| 20:44.38 | codestr0m | and the bias is always going to be on the side of the product a person has choosen |
| 20:44.45 | codestr0m | iow. ymmv |
| 20:45.04 | codestr0m | bottom line. ask something more specific |
| 20:45.30 | ber111 | just looking for code stability and ease of configuration |
| 20:45.55 | ber111 | i have SER running as a proxy right now seems to be pretty stable and its easy to configure |
| 20:46.23 | codestr0m | ber111: that's still not a very specific question... |
| 20:48.12 | *** join/#asterisk ReDNeQ (n=iBuMp@rrcs-71-42-227-6.sw.biz.rr.com) |
| 20:48.27 | ReDNeQ | sup sup |
| 20:48.34 | *** part/#asterisk codestr0m (n=asura@207.135.120.85) |
| 20:49.42 | *** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com) |
| 20:49.54 | ReDNeQ | if you have a VPN setup between locations do you still need to include udp statements in the firewalls of the VPN's? |
| 20:50.16 | ReDNeQ | the reason I ask is because I am getting random phones connecting or able to connect across the wans |
| 20:51.28 | mocker | ReDNeQ: Is your VPN stable? |
| 20:51.51 | ReDNeQ | well thats another question all together |
| 20:52.00 | ReDNeQ | we are using Timewarner on both ends |
| 20:52.10 | mocker | Hah. |
| 20:52.12 | ReDNeQ | and the connections are 1.5 min |
| 20:52.17 | ReDNeQ | but you know how that goes |
| 20:53.36 | mocker | ReDNeQ: If you have a VPN you should have to worry about firewalls. |
| 20:54.16 | mocker | Unless you have some local firewalls behind the VPN |
| 20:55.22 | eatmypiano | Hi. I'm about to move to Canada (from the UK) and I want to set up Asterisk in my new house. In the new house I have a phone line supplied by Rogers (cable company) and some structured wiring which I'll be expanding. I want to have 4 phones (office, kitchen, living room, bedroom). Where can I find what kit i need to buy (including the server spec)? |
| 20:56.02 | ReDNeQ | well what I have are 2 dlink 804HV's |
| 20:56.07 | ramindia | eatmypiano: simple PC do that job |
| 20:56.15 | ReDNeQ | and nothign for firewall installed on the voip box |
| 20:56.21 | ReDNeQ | not even selinux |
| 20:56.35 | ramindia | eatmypiano: u can just download * and install, if not download Trixbox, does the whole job |
| 20:56.52 | ReDNeQ | but some phones across the wan can connect and make calls but hte phones have not stayed stable at all |
| 20:57.01 | ReDNeQ | dropped calls, unable to check voicemail |
| 20:57.16 | ReDNeQ | the like.. We are using digium td808p and g729a codec |
| 20:57.54 | mocker | ReDNeQ: Any chance you can test w/o the VPN? |
| 20:58.11 | mocker | If it's the # of ports you're worried about, IAX only needs you to have one opened up. |
| 20:58.13 | eatmypiano | How simple? I have a PIII with 640Mb and a PII laying around. |
| 20:58.34 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:58.35 | ramindia | eatmypiano: if u talking only 4 extension should do that hardwre |
| 20:59.11 | eatmypiano | OK - Thanks! |
| 20:59.57 | grey | If I (or actually a friend I got interested in Asterisk while talking about my plans) Wanted to pass data traffic through asterisk, for the purposes of a dial up connection specifically, is that going to be possible? are there any codecs recommended for best results when passing data? |
| 21:00.15 | ReDNeQ | i have to have the vpn to keep the locations connected |
| 21:00.25 | ReDNeQ | i dont have 2 systems (voip boxes) |
| 21:01.01 | mutilator | anyone wanna but a lightly used te110p? $200 |
| 21:02.02 | Qwell[] | I keep hearing "lightly used"... as opposed to what, somebody who slams the card on their desk daily? :) |
| 21:02.12 | mutilator | heh |
| 21:02.18 | mutilator | as opposed to being run in a system for a year |
| 21:02.27 | Qwell[] | ahh |
| 21:02.58 | mutilator | i think i'll have to just throw it on ebay heh |
| 21:03.02 | Qwell[] | yeah |
| 21:03.25 | mutilator | sux 2 that |
| 21:06.00 | irule | how can I see info for an application in the CLI? |
| 21:06.25 | jkiff | "show application (application name)" |
| 21:11.23 | [TK]D-Fender | mutilator, e-bay it. |
| 21:12.04 | [TK]D-Fender | grey, Don't even think about it.. |
| 21:12.11 | grey | >_> |
| 21:14.07 | krdian_ | <PROTECTED> |
| 21:14.28 | irule | thanks |
| 21:14.46 | grey | hmm, so no work around to it? It still needs a second phone line? |
| 21:15.24 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 21:17.04 | [TK]D-Fender | grey, highly inadvisable. |
| 21:20.44 | *** join/#asterisk lwh (n=lwh192@rdsl-0469.tor.pathcom.com) |
| 21:23.54 | grey | ok, I'll take your word for it |
| 21:23.57 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 21:24.35 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:25.14 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.105) |
| 21:28.33 | syzygyBSD | what is the option to pass to gcc so "isdef VARIABLE" is true? |
| 21:29.44 | grey | is it the latency or the codec that kills it? We're looking at an incoming dial up line used by a single user most likely, |
| 21:32.54 | *** join/#asterisk adker (n=chatzill@74-33-221-202.br1.glv.ny.frontiernet.net) |
| 21:34.00 | [TK]D-Fender | grey, Any data over voip is highly unstable. Faxes don't make it reliably. |
| 21:34.12 | grey | >_> |
| 21:34.22 | grey | what causes it to be unstable? |
| 21:34.29 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-79.ph.ph.cox.net) |
| 21:34.37 | Maxxed | man is there any thing i need 2 change on the pbx when i have my DNIS changed? |
| 21:34.52 | grey | I get that converting back and forth between data/audio a few times might cause problems, especially depending on the codec |
| 21:35.23 | Maxxed | the telco was sending the last 4 digits, now they are sending 10, and i cant dial in |
| 21:35.40 | Maxxed | i get an error msg |
| 21:35.43 | grey | but over 10/100 LAN, why not just use a PCM/WAV codec with little loss even if you're just stuffing it out an FXS and into a modem? (Which would suck but would allow a lot of flexibility) |
| 21:35.48 | Maxxed | from the telco |
| 21:35.53 | *** join/#asterisk ectospasm (i=Spasm@nat/digium/x-4dabd08e9e27fab7) |
| 21:36.35 | Maxxed | anybody know anything about DNIS ? |
| 21:38.10 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 21:38.28 | Aces1Up | hey tkd, were you on the asterisk users podcast? |
| 21:41.01 | [TK]D-Fender | Aces1Up, The first, yes |
| 21:41.21 | [TK]D-Fender | Aces1Up, Actually... not sure if we;re talkijng about the same thing |
| 21:41.38 | [TK]D-Fender | Aces1Up, I was on Zeeek's Talkshoe conference. |
| 21:41.53 | [TK]D-Fender | Aces1Up, the first, and a few minutes of the last one. |
| 21:42.34 | [TK]D-Fender | grey, Doesn't work that way. telephony lines don't encode that way. Trust me. This is a dead end. |
| 21:42.49 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 21:43.02 | [TK]D-Fender | Maxxed, make sure your dialplan matches what theya re sending. |
| 21:43.09 | grey | :-\ |
| 21:43.17 | [TK]D-Fender | Maxxed, pastebin the incoming call that gets rejected at verbose 10 |
| 21:43.58 | Trevor_b | grey: sampling makes it unstable. Codecs take small samples of the true sounds, and then use those small samples to reconstruct the audio. To human hearing its not noticed, to a fax working of every little warble and squak it makes a HUGE difference. |
| 21:44.18 | grey | ah |
| 21:44.19 | Trevor_b | squak(?) hmm |
| 21:44.25 | grey | Squawk |
| 21:44.28 | Trevor_b | thanks |
| 21:44.29 | grey | :P |
| 21:45.04 | Trevor_b | yeah so just think that were less sensitive, so loosing 50% of the conversation (in miniscule bits) we dont even notice, very rarely does a fax like that AT ALL. |
| 21:45.20 | grey | thank God you explained that, It makes sense now, It's one of those things where I know TKD knows better than I do, so I Want to believe him and keep from wasting my time, but it just seems like something that 'should work' to first look at :P |
| 21:46.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:46.12 | [TK]D-Fender | grey, You'll hear all sorts of crap about faxing in here, and remember that usually 14.4k or less.... |
| 21:46.12 | Trevor_b | well there is work to TRY and get it to work, it SUCKS, and fails badly ;). TK is dead on target when it comes to voip from everything i have heard him say |
| 21:46.24 | Trevor_b | I do TONS of faxing on asterisk ;) |
| 21:46.30 | Trevor_b | its all zap hardware lines mind you :) |
| 21:46.57 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 21:47.11 | [TK]D-Fender | Trevor_b, still not that relaible... echo often tends to get thrown in, lines get staticy through many cards, etc... |
| 21:47.45 | [TK]D-Fender | Trevor_b, bridged zaptel is definately the best way if you need * involved at all, but typically not suggested |
| 21:48.00 | Trevor_b | Yeah, its not wonderful, i never run more then a single TDM2400 in a system. But as long as the lines are good or you fxotune it seems to be pretty damn stable (compared to the cost of a 12 port card from ANY fax hardware place).... |
| 21:48.14 | Trevor_b | bridged zaptel? like between 2 servers? |
| 21:48.33 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 21:49.21 | [TK]D-Fender | Trevor_b, No, 2 ports on a card (FXS -> FXO |
| 21:49.54 | [TK]D-Fender | Trevor_b, You talking about SpanDSP / Hylafax here? |
| 21:50.11 | Trevor_b | SpanDSP inbound fax server, not outbound, you meaning outbound? |
| 21:50.31 | [TK]D-Fender | Trevor_b, either = same risk :) |
| 21:50.48 | [TK]D-Fender | Trevor_b, are you running a "from scratch" installation? |
| 21:52.15 | Trevor_b | [TK]D-Fender: hehehe, yeah i have had SpanDSP running on an OLD OLD asterisk build for about 16 months. Not really from scratch, its a borged out Asterisk@Home, rebuild asterisk, rebuilt zaptel, just really using the freepbx from the initial installation. All our current work is our own RPM's for CentOS (got really sick of TrixCrap after helping him get the RPM's rolling, and it was half assed) |
| 21:52.17 | Aces1Up | tkd yeh i was listening to the one on talkshoe about double-natting. |
| 21:52.31 | Aces1Up | and he mentioned tk when referring to someone, thought it might be you. |
| 21:52.41 | [TK]D-Fender | Aces1Up, Yup |
| 21:52.46 | [TK]D-Fender | Aces1Up, That was me |
| 21:52.46 | Aces1Up | cool. |
| 21:52.47 | grey | If I'm happy with this stuff running at 14.4 is it going to be a big deal? the dial up is just a sort of emergency administration sort of thing afaik, |
| 21:53.06 | Aces1Up | good job on it. |
| 21:53.27 | [TK]D-Fender | grey, If you're thinking about it working through that SPA solution we talked about yesterday.... the odds are UGLY |
| 21:53.29 | grey | and what sort of reliability can I expect out of those app_rxfax stuff? |
| 21:53.52 | *** join/#asterisk _omer (n=_omer@DSL-202-59-92-141.nexlinx.net.pk) |
| 21:53.53 | [TK]D-Fender | grey, Faxing definately requires a better card than an X100. TDM400P minimum. |
| 21:54.07 | _omer | hello.... |
| 21:54.13 | _omer | http://www.pastebin.ca/562355 ....anyhelp? |
| 21:54.13 | Aces1Up | anyone here use adhearsion? |
| 21:54.22 | [TK]D-Fender | Trevor_b, I STILL can't get SpanDSP (rx_fax) not to crash on 1.2.17+ |
| 21:54.23 | grey | well, I need to recieve faxes at home here, but the place thats looking at dial up AND fax is a little better, has battery backups and is an office/server room type place, |
| 21:54.26 | Aces1Up | wondering what your thoughts on it are. |
| 21:54.49 | grey | and does that mean the linksys 3102/2102 stuff is right out the window? I really like that failover to POTS |
| 21:54.53 | [TK]D-Fender | grey, And all this..... off 1 line! |
| 21:55.15 | [TK]D-Fender | grey, You are building up a house of cards around a single line. |
| 21:55.26 | grey | thats why I like the failover >_> |
| 21:55.55 | [TK]D-Fender | grey, and how is * to know how to process a call when it can be Data/fax/voice/morse-friggen-code ? :) |
| 21:56.15 | grey | I'd think an IVR choice? have the fax number include a pause then another number |
| 21:56.15 | [TK]D-Fender | _omer, we heard you the first dozen times. |
| 21:56.21 | grey | can usually be done easily with a comma |
| 21:56.38 | [TK]D-Fender | grey, nobody SENDING you a fax will go through an ivr. |
| 21:56.46 | _omer | Updated GCC ...still didn't work...same error msg |
| 21:56.58 | [TK]D-Fender | grey, face it, you're going to have to actually SPEND money. |
| 21:57.02 | grey | heh dang |
| 21:57.05 | Trevor_b | hmm yeah i think this is like a 1.2.0x system i have in play. I dont like screwing with spanDSP since i got it working flawlessly (with a little code to stop spamming empty PDF's to the inbox) |
| 21:57.56 | grey | It's not THAT huge of a deal, my first objective is just to split incoming office calls out from incoming home calls, |
| 21:58.09 | Trevor_b | grey: TK's right, how many times have you sat at your fax with phone in hand and dialed up hitting digits until you could hit the send button? |
| 21:58.22 | Trevor_b | ;) |
| 21:58.37 | grey | I know that, but I also know most devices can insert a pause into a phone number |
| 21:58.51 | Trevor_b | sure, i more meant the IVR into fax |
| 21:58.57 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:58.59 | [TK]D-Fender | grey, not AFTER dialing in await of a MENU. |
| 21:59.06 | [TK]D-Fender | grey, that'd be psychotic. |
| 21:59.28 | grey | buh? |
| 21:59.46 | *** join/#asterisk JSabines (n=alancast@189.158.186.76) |
| 22:00.02 | [TK]D-Fender | grey, so 1 line for home voice calls, business call, fax calls, inbound dial-up-isp calls, and THEN morse-code calls? |
| 22:00.13 | grey | no, not inbound dial up on my home phone |
| 22:00.14 | [TK]D-Fender | grey, "Would you like fries with that, sir?" |
| 22:00.16 | Trevor_b | [TK]D-Fender: yeah its 1.2.4 with a 289 day uptime. I have thought up upgrading but dealing with spanDSP in anything new i just didnt want to think about ;) |
| 22:00.38 | grey | sorry, lets drop that for now, it's for a different setup that will be months down the road, just my friend mentioned it when I started talking about tinkering with * |
| 22:00.45 | [TK]D-Fender | Trevor_b, I had it working under 1.2.7.1 last and siince then it just shits all over me... |
| 22:01.26 | grey | basically press 1 for my family, press 2 for the business, press 3 to send a fax (Or heck it doesn't even have to say it if we give the fax number out as xxx-xxxx,x) |
| 22:01.52 | grey | the fax machine might not like the menu being read off to it, but should probably ignore it while it's still dialing right? |
| 22:02.03 | [TK]D-Fender | grey, no fax machine I've ever heard of can do it AFTER dialing. |
| 22:02.15 | [TK]D-Fender | grey, and you'd be asking the world at large to try. |
| 22:02.16 | Trevor_b | [TK]D-Fender: great. Haven't even looked into if its compatible with 1.4.x yet. Sounds crappy considering it worked so well in old revisions. Although I think Trixbox is still touting fax with DSP, may have to look at their spec files and see if i can figure out how they are doing it. |
| 22:02.33 | [TK]D-Fender | Trevor_b, TrixBox seems to get ti right... |
| 22:02.41 | grey | nah, faxes are quite rare, this is a pretty small business and it's a yard care business, not something with a lot of incoming paperwork |
| 22:02.48 | [TK]D-Fender | Trevor_b, I just need to figure out what they use I guess. |
| 22:03.16 | [TK]D-Fender | grey, You are quite welcome to try, but you are definately looking at a pricier card. |
| 22:03.19 | Trevor_b | grey: depends, some devices error if not getting the correct tone to start the call and get a prompt, but im not paying enough attention to keep up with where your at |
| 22:03.20 | grey | I'll have to peek at a few fax machines and see if they allow a delay during a dial, |
| 22:03.48 | grey | I know my cell phone does, I know windows does if you are printing to a virtual fax machine under windows XP/2k |
| 22:04.33 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:04.38 | Trevor_b | [TK]D-Fender: my only local Trix is the 2.0 which htey kept at 1.2.9, ill be rolling a xen box of the new 2.2 to see what they did (refuse to use the product in production anymore), since I probably will have to upgrade this fax server fairly soon to get it into the Rack. |
| 22:05.21 | grey | worse comes to worse I can drop whole fax thing, afaik right now the only one using it my idiot brother who uses it because he's too stunned to use e-mail, and happens to have a fax machine (It's a lot easier for him, but a huge pain in the ass on this end, have to go put windows into 'recieve fax mode' every time, then click answer etc.) |
| 22:07.03 | Qwell[] | hi |
| 22:07.05 | Qwell[] | erm |
| 22:07.08 | Qwell[] | wrong channel ;p |
| 22:07.10 | Trevor_b | grey: fxo=>fxs like TK mentioned sounds like the way to deal with that, takes 1 new line and probably a TDM400. Or at least thats my guess from now ;) |
| 22:07.21 | Maxxed | [TK]D-Fender>: i dont get anything |
| 22:07.37 | Maxxed | i think its a telco thing |
| 22:07.53 | Maxxed | 210-408-1205 |
| 22:07.56 | grey | TDM400 is about $160 right? |
| 22:08.13 | Maxxed | listen 2 the error, thats telco up n down |
| 22:08.13 | Maxxed | im on the phone with them |
| 22:08.30 | Maxxed | there chekin it out |
| 22:08.39 | [TK]D-Fender | I'm off for a while... |
| 22:08.43 | grey | ok, |
| 22:08.45 | grey | thanks for the help TKD |
| 22:09.41 | Trevor_b | looks like about $300 with 2 fxo on it and 1 fxs (for the fax). |
| 22:09.52 | Trevor_b | hell |
| 22:09.55 | Trevor_b | WTF am i thinking |
| 22:10.18 | Trevor_b | no reason to do that at all, just directly conect that second inbound line to the fax, DUH. if you get a second line anyway. |
| 22:10.28 | grey | heh yeah I'm thinking thats the likely solution |
| 22:10.30 | Trevor_b | Sorry i never deal with faxing unless its a dedicated line for fax. |
| 22:10.43 | grey | no I know, if at all possible I'd prefer to avoid getting a second line |
| 22:10.43 | Trevor_b | which means its just inbound to PDF conversion. |
| 22:10.59 | grey | but if it's get a $6/month second line, or a $300+ card... |
| 22:11.06 | Trevor_b | you could do inbound fax to lpr to pring on a local printer. |
| 22:11.18 | Trevor_b | s/pring/print |
| 22:11.20 | grey | I'm even happy with the fax->pdf thing, thats all I need |
| 22:11.47 | Trevor_b | there is pause code on inbound calls to wait for fax initiation on the same line, i rarely use it, but freepbx has it. |
| 22:12.07 | Trevor_b | just delays the caller being dropped to an extension to see if they initiate a tone for fax first. |
| 22:12.21 | grey | ok, how long of a delay is that usually? |
| 22:12.26 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 22:13.10 | grey | also apparently most fax machines do support the dial-pause-dial approach, so I could have the faxes just navigate the first level of the IVR as long as I can direct them straight to whatever fax recieving program I have right after that |
| 22:13.16 | grey | but I'm getting the impression thats inadvisable |
| 22:13.55 | sunsmasher | whats the big difference between 1.4.4 and 1.2.18? Why release two different versions? |
| 22:13.56 | Mahmoud | hey guys, I need IAX2 protocol specification |
| 22:14.20 | Qwell[] | Mahmoud: You can google the draft RFC |
| 22:14.32 | Mahmoud | Qwell, the number? |
| 22:14.57 | Qwell[] | there isn't one, it's a draft |
| 22:15.36 | Trevor_b | grey: configurable delay |
| 22:15.48 | Trevor_b | more the better obviously, as more time to get a slow fax starting |
| 22:16.17 | Trevor_b | lots or work, lots of headache when it breaks if theres lots of steps. |
| 22:16.22 | grey | but for voice callers I don't want it to be too long |
| 22:16.33 | grey | are we talking 1-2 seconds? or 5-10? |
| 22:16.35 | Trevor_b | 2-3 seconds is the average i think, |
| 22:16.40 | grey | ok thats not too bad |
| 22:16.49 | *** join/#asterisk techie (n=gus@antibala.net) |
| 22:17.04 | grey | any leads as to how to suppotr that on *? |
| 22:17.38 | Trevor_b | they probably are doing a Wait with an NVFaxDetect command |
| 22:17.47 | Trevor_b | 'help application nvfaxdetect" |
| 22:17.52 | *** join/#asterisk lee_is_me (n=chatzill@66.16.60.61) |
| 22:18.09 | grey | excellent :) |
| 22:18.29 | lee_is_me | would anyone mind helping me with an echo issue? I'm 2 1/2 hours away at this customer site again and I just can't seem to track down the issue |
| 22:18.48 | Mahmoud | Qwell[], any plans to change it? how stable is it? |
| 22:19.17 | Mahmoud | Qwell[], i'm thinking to write an IAX2 softphone for symbian.. not sure if that draft is enough or not |
| 22:19.19 | Qwell[] | well, it can't change much now that it's a draft RFC... |
| 22:19.37 | Qwell[] | and we can't change the protocol enough that it breaks reverse compat with asterisk 1.2 |
| 22:19.45 | Qwell[] | and hell, 1.0 for that matter |
| 22:19.51 | JSabines | lee_is_me where do you have the echo |
| 22:19.56 | JSabines | sip or tdm cards |
| 22:20.09 | lee_is_me | JSabines: sangom a200 |
| 22:20.21 | JSabines | without echo card? |
| 22:20.21 | lee_is_me | JSabines: 2 channels |
| 22:20.33 | lee_is_me | It was supposed to come with software echo can. |
| 22:20.37 | lee_is_me | Octware? |
| 22:21.00 | lee_is_me | can't get it installed because the register utility won't recognize that I have sangoma card installed |
| 22:21.19 | lee_is_me | so I'm trying to do my best, but getting frustrated with this card...supposed to be better |
| 22:21.42 | lee_is_me | doesn't happen all the time. randomly |
| 22:22.00 | lee_is_me | you can hear an echo of your own voice so load it makes talking difficult |
| 22:22.07 | JSabines | the card was installed without the echo option? |
| 22:22.13 | lee_is_me | Yes |
| 22:22.27 | lee_is_me | I thought the Octware software echo cancel would work |
| 22:22.45 | *** join/#asterisk blueneon (i=hfklows@dsl-146-31-219.telkomadsl.co.za) |
| 22:23.10 | lee_is_me | I'm about to RMA the sangoma and get a TDM400, maybe. Tough to keep coming back out here 2 1/2 hours away ;) |
| 22:23.31 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.105) |
| 22:23.32 | JSabines | when do you check the wanpipe conf over /etc/sangoma i think |
| 22:23.41 | lee_is_me | when? |
| 22:23.56 | lee_is_me | I used whatever the defaults were during the installation |
| 22:24.10 | JSabines | what do you see |
| 22:24.12 | flenders | I have a couple of TDM400Ps |
| 22:24.17 | flenders | 4 FXO modules each |
| 22:24.34 | lee_is_me | i do not have /etc/sangoma directory |
| 22:24.35 | flenders | get rid of echo was a pain in the ass, but fxotune helped a lot |
| 22:25.02 | blueneon | hi... i have FC3 and a digium TDM400p, i just compiled the zaptel drivers, then compiled asterisk etc, i've entered the 3 channels i have in /etc/zaptel.conf and in /etc/asterisk/zapata.conf, i start up asterisk and get no errrors at all. But its not seeing the zap channels at all, show channels returns 0... but ztcfg -vv shows 3 chans as configured |
| 22:25.11 | blueneon | any ideas what i might be doing wrong? |
| 22:25.17 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:25.53 | JSabines | i do not remember where sangoma cards write its config files |
| 22:26.00 | JSabines | but is under /etc |
| 22:26.32 | JSabines | there you should have the echo option on |
| 22:26.43 | lee_is_me | it's "/etc/wanpipe" |
| 22:26.59 | flenders | blueneon: pastebin your zaptel.conf and zapata.conf |
| 22:27.09 | lee_is_me | ok hold 1 |
| 22:28.44 | JSabines | TDMV_HWEC=yes on wanpipex.conf |
| 22:29.05 | lee_is_me | http://www.pastebin.ca/562812 |
| 22:29.12 | lee_is_me | JSabines: I'll try that now |
| 22:29.43 | lee_is_me | is it normal that I should have a wanpipe1.conf ? |
| 22:29.55 | JSabines | yes |
| 22:30.20 | JSabines | wanpipe1.conf is your conf file for that card |
| 22:30.34 | flenders | I had a pretty weird problem with a sangoma a101 the other day |
| 22:30.39 | lee_is_me | should I rmmod wanpipe? |
| 22:30.51 | lee_is_me | it was set to NO, btw |
| 22:31.25 | flenders | first it wouldnt bring up the PRI, then, when it did (for a brief moment), I had a 'clicking' noise on the outgoing audio |
| 22:31.28 | JSabines | no |
| 22:31.46 | flenders | I ended up moving the card to a different slot on the box, and it fixed the problems |
| 22:31.51 | JSabines | wanpipe is used for the card along the other drivers |
| 22:32.11 | blueneon | http://www.pastebin.ca/562824 |
| 22:32.16 | blueneon | thats my zapata.conf |
| 22:32.18 | lee_is_me | JSabines: oh, ok. I thought maybe I should unload and restart it |
| 22:32.37 | flenders | blueneon: I'll have a look |
| 22:32.45 | blueneon | http://www.pastebin.ca/562828 |
| 22:32.50 | blueneon | thats my zaptel.conf |
| 22:33.02 | lee_is_me | JSabines: Should I restart the system then? |
| 22:33.25 | Mahmoud | Qwell[], is it this? http://www.ietf.org/internet-drafts/draft-ietf-enum-iax-02.txt |
| 22:33.28 | JSabines | lee_is_me do you use setup_sangoma |
| 22:33.50 | JSabines | when you installed the second time? |
| 22:33.57 | Qwell[] | Mahmoud: believe so |
| 22:33.58 | blueneon | if i do a ztmonitor and pick up the handset and talk i can see movement, so i know its working, and as i said ztcfg -vv shows the channels, it seems that asterisk is just ignoring them completely tho |
| 22:34.13 | lee_is_me | I belive I did |
| 22:34.19 | Mahmoud | Qwell[], there is another one named 01 instead of 02.. wondering what does it mean |
| 22:34.27 | lee_is_me | console base installation utility, right? |
| 22:34.33 | JSabines | yes |
| 22:34.38 | lee_is_me | then yes |
| 22:34.53 | flenders | blueneon: did you install zaptel before you installed asterisk? |
| 22:34.57 | blueneon | yes |
| 22:35.12 | JSabines | well restart it and see what happens |
| 22:35.22 | lee_is_me | sure thing. Wait a couple... |
| 22:35.57 | blueneon | one wierd thing is that when i do, show channeltypes i dont see ZAPTEL in that list, am I meant to? |
| 22:36.23 | blueneon | http://www.pastebin.ca/562837 |
| 22:36.26 | blueneon | this is what i get |
| 22:36.35 | flenders | blueneon: you should see 'Zap' |
| 22:37.10 | blueneon | ok, so whats wrong then, cause its not there :(? |
| 22:38.23 | lee_is_me | JSabines: I restart asterisk |
| 22:38.35 | lee_is_me | the thing is that it only happens randomly |
| 22:38.49 | lee_is_me | so I will not know if it did any good for a while |
| 22:39.11 | lee_is_me | what does the setting do when there is no echo card on the sangoma? |
| 22:39.20 | ManxPower | blueneon: did you install zaptel BEFORE Asterisk |
| 22:39.49 | blueneon | yes |
| 22:39.52 | blueneon | i did |
| 22:39.52 | JSabines | nothing the echo is managed by asterisk |
| 22:40.07 | lee_is_me | oh, what was the setting change for then? |
| 22:40.11 | lee_is_me | curious... |
| 22:40.21 | flenders | blueneon: you did a 'make install' on zaptel? |
| 22:40.30 | blueneon | yes |
| 22:40.36 | flenders | blueneon: try recompiling asterisk |
| 22:40.37 | JSabines | to be managed for the hardware instead |
| 22:40.43 | blueneon | already tried that |
| 22:41.08 | flenders | blueneon: so, youre installing both from source? |
| 22:41.24 | flenders | blueneon: what versions of zaptel and asterisk? |
| 22:41.27 | blueneon | yes |
| 22:41.30 | blueneon | sec |
| 22:41.40 | blueneon | asterisk-1.2.18 |
| 22:41.44 | JSabines | what kind of machine did you put the sangoma a200 |
| 22:41.45 | blueneon | zaptel-1.4.3 |
| 22:41.51 | flenders | there you go |
| 22:41.53 | blueneon | ? |
| 22:41.57 | flenders | you should use asterisk 1.4.x |
| 22:42.06 | blueneon | hmm |
| 22:42.24 | blueneon | shit, ok, so how would i remove this now that its installed by src? |
| 22:42.33 | lee_is_me | p4 3.0 Ghz on CentOS 4.4 |
| 22:42.41 | JSabines | you problem could be missed interrupts |
| 22:42.44 | lee_is_me | or were you asking about the mobo? |
| 22:42.49 | JSabines | yes |
| 22:43.03 | flenders | not sure if 'make uninstall' works on asterisk |
| 22:43.05 | lee_is_me | Asus |
| 22:43.11 | lee_is_me | not sure which model though |
| 22:43.30 | JSabines | has hyperthreading? |
| 22:43.37 | *** join/#asterisk bonderponder (n=test@201.199.68.150) |
| 22:43.39 | lee_is_me | I think so |
| 22:43.41 | flenders | blueneon: but I just recompiled/installed 1.4.x on top of 1.2.x and it worked |
| 22:43.47 | lee_is_me | it's an smp |
| 22:43.48 | JSabines | oks |
| 22:44.00 | blueneon | nope |
| 22:44.04 | JSabines | but are you using irqbalance? |
| 22:44.04 | blueneon | it doesnt :( |
| 22:44.08 | blueneon | oh |
| 22:44.12 | blueneon | ok i'll do that |
| 22:44.13 | blueneon | sec |
| 22:44.20 | lee_is_me | lol, since I don't know what it is, I'll say no |
| 22:44.24 | lee_is_me | sorry |
| 22:44.29 | JSabines | if so cancel it and assign one processor to your card |
| 22:44.43 | flenders | blueneon: make sure you backup your zapata.conf, as I believe it's ready to go |
| 22:44.53 | lee_is_me | JSabines: not sure how to do that |
| 22:44.55 | flenders | and don't forget the make samples |
| 22:44.58 | bonderponder | Hello, I need to develop and finish my currrent IVR. But I cant figure it out how to finish , anyone can help me ? |
| 22:45.17 | flenders | bonderponder: what do you need? |
| 22:45.28 | blueneon | kk |
| 22:46.30 | blueneon | damn the digium server is slow |
| 22:46.42 | Qwell[] | blueneon: "the digium server"? |
| 22:46.50 | Qwell[] | I'm pretty sure we have more than one... |
| 22:46.56 | blueneon | ftp.digium.com |
| 22:46.57 | lee_is_me | JSabines: Linux localhost.localdomain 2.6.9-55.ELsmp #1 SMP Wed May 2 14:28:44 EDT 2007 i686 i686 i386 GNU/Linux |
| 22:47.02 | Qwell[] | blueneon: shouldn't be |
| 22:47.08 | blueneon | *shrug* |
| 22:47.23 | blueneon | flenders: im getting asterisk-1.4.4.tar.gz is that the one i should be getting? |
| 22:47.25 | Qwell[] | try ftp1 or ftp2 |
| 22:48.33 | Qwell[] | ftp1 will probably be faster right this second... |
| 22:48.46 | blueneon | too late already using ftp. |
| 22:48.47 | blueneon | :/ |
| 22:50.08 | *** join/#asterisk plut0 (i=plut0@cpe-74-70-152-114.nycap.res.rr.com) |
| 22:50.28 | plut0 | greetings |
| 22:50.33 | bonderponder | flenders: can you help me ? |
| 22:51.19 | russellb | Qwell[]: updating? ;) |
| 22:51.22 | Qwell[] | indeed! |
| 22:51.29 | Qwell[] | about 90 packages total, without -u |
| 22:51.36 | Qwell[] | we'll see how it goes after this is done |
| 22:51.41 | russellb | heh, cool |
| 22:51.45 | Qwell[] | I masked glibc and gcc :D |
| 22:51.50 | russellb | good call |
| 22:51.52 | Qwell[] | yeah |
| 22:51.56 | russellb | and i wouldn't recommend upgrading the kernel, either .. |
| 22:51.57 | Qwell[] | 2.3.6 > 2.5 |
| 22:52.01 | Qwell[] | would've been very bad |
| 22:52.04 | russellb | because if it breaks ........ well .... we're screwed |
| 22:52.06 | Qwell[] | yeah |
| 22:52.22 | bonderponder | Hello, I need to develop and finish my currrent IVR. But I cant figure it out how to finish , anyone can help me ? |
| 22:53.59 | Qwell[] | emerge --unmerge emacs vi |
| 22:54.00 | Qwell[] | erm, wrong window |
| 22:54.04 | Qwell[] | :P |
| 22:54.30 | plut0 | gentoo fan? |
| 22:54.39 | Qwell[] | eh, I use it at home |
| 22:54.51 | Qwell[] | it's a PITA when it isn't maintained though |
| 22:54.56 | plut0 | i'm using it on our asterisk pbx |
| 22:54.58 | russellb | (like this server) |
| 22:55.03 | Qwell[] | exactly |
| 22:55.11 | Qwell[] | which is why one of the ftp servers may be slow right now :p |
| 22:55.15 | Qwell[] | the load is at like 3.5, heh |
| 22:55.20 | plut0 | i only upgrade for security advisories |
| 22:57.22 | blueneon | flenders: thanks, the newer ver work :) |
| 22:59.36 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:59.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:59.54 | flenders | blueneon: glad it worked |
| 23:01.52 | blueneon | for audio playbacks... where is the default folder? |
| 23:02.01 | blueneon | if i do PlayBack(filename) |
| 23:02.06 | blueneon | where is Asterisk looking? |
| 23:02.15 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 23:06.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:06.53 | *** join/#asterisk canberk (n=canberk@212.156.213.131) |
| 23:06.55 | canberk | hello |
| 23:07.40 | canberk | i have welltech wellgate 4804 fxo gateway, i set up the gateway to send out the caller id to asterisk and i can see that the device can read the caller id from pstn network from device's debug, however asterisk is not showing the caller id |
| 23:07.43 | canberk | why do you think is this |
| 23:07.46 | ManxPower | blueneon: it is set in /etc/asterisk/asterisk.conf |
| 23:15.40 | *** join/#asterisk nephfl (n=no@wsip-70-184-144-158.ga.at.cox.net) |
| 23:15.56 | nephfl | does anyone know if meridian phones will work with asterisk |
| 23:19.27 | nephfl | anyone here? |
| 23:20.29 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 23:20.51 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 23:21.08 | nephfl | hello |
| 23:22.06 | *** join/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 23:22.10 | BSD_Tech[laptop] | ok need info |
| 23:22.21 | BSD_Tech[laptop] | on 1.4 I knwo they use users.conf now |
| 23:22.31 | BSD_Tech[laptop] | for user extensions |
| 23:22.41 | blueneon | which file in the sounds dir is the one with the american woman saying "please leave a message after the tone..." |
| 23:22.41 | blueneon | ? |
| 23:22.53 | BSD_Tech[laptop] | but do they plan to move trunkx to a trunks.conf |
| 23:23.03 | Qwell[] | blueneon: You can look in the sounds.txt file, and it'll tell you what each sound says |
| 23:23.07 | Qwell[] | BSD_Tech[laptop]: no, no plans |
| 23:23.19 | BSD_Tech[laptop] | hmm they should |
| 23:23.51 | BSD_Tech[laptop] | they group the trunks in the users.conf and it makes it ugly |
| 23:24.25 | nephfl | anyone ever have to deal with nortel/meridian phones? |
| 23:24.27 | blueneon | there is not "please leave a message..." in sounds.txt |
| 23:24.35 | BSD_Tech[laptop] | but I like the fact it moves the users extensions out of sip.con and iax.conf and puts them in 1 managed area |
| 23:25.13 | blueneon | nvm |
| 23:25.17 | blueneon | found it |
| 23:25.19 | blueneon | doh |
| 23:25.40 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:25.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:26.53 | snuffy22 | how would i set a variable of a parent channel |
| 23:28.44 | BSD_Tech[laptop] | ok other question |
| 23:28.45 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:28.57 | BSD_Tech[laptop] | are zap users also ut in the users.conf ? |
| 23:29.23 | BSD_Tech[laptop] | ut/put |
| 23:30.17 | mrdigital-work | hwy BSD_Tech[laptop]: i got a FTP Question |
| 23:30.28 | mrdigital-work | *hey, |
| 23:30.45 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 23:30.48 | mrdigital-work | im trying to set up ftp but any user i give it it says invaild |
| 23:31.14 | iBuMp | good evening |
| 23:32.48 | BSD_Tech[laptop] | what ftp client |
| 23:32.53 | BSD_Tech[laptop] | server |
| 23:35.12 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-77-93-206.dsl.ipltin.ameritech.net) |
| 23:35.18 | mrdigital-work | VSFTPD |
| 23:35.25 | mrdigital-work | i want 1 user to access /backups |
| 23:35.52 | mrdigital-work | client is FlashFXP |
| 23:36.47 | tzafrir_laptop | why not use sftp? |
| 23:36.52 | mrdigital-work | sftp? |
| 23:37.02 | mrdigital-work | holdon |
| 23:37.19 | tzafrir_laptop | anyway, ask this in #$DISTRO |
| 23:37.37 | tzafrir_laptop | sftp is a file transfer protocol on top of ssh |
| 23:38.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:38.58 | Know1 | any thoughts on why I'd be able to register to a sip provider but get SIP/2.0 401 Unauthorized when trying to make a call? |
| 23:39.18 | tzafrir_laptop | BSD_Tech[laptop], yes, zap users can be defines in users.conf. And in a saner syntax, actually |
| 23:39.27 | nephfl | I cant find any information about it, has anyone tried to connect nortel meridian phones to asterisk? |
| 23:39.35 | *** join/#asterisk DMark (n=root@vps-71-6-209-250.lylix.net) |
| 23:39.53 | tzafrir_laptop | if you add a zapconf=<channelspec> line to a users.conf entry, it is processed by chan_zap |
| 23:40.37 | tzafrir_laptop | and basically read as a small zapata.conf snippet which ends with 'channel => <channelspec>' |
| 23:41.14 | tzafrir_laptop | But the nice thing is that those definitions are not carried onwards to the next channel, so the order and such is no loger significant |
| 23:42.24 | CrashSys | http://www.pastebin.ca/562963 I get these error's when I try to install wanpipe... it's right as it tries to compile kernel modules... but the files aren't in the distribution... am I missing something? |
| 23:42.25 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:42.59 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 23:43.02 | dotSlashW | hi, I'm having some trouble debugging a SIP trunk |
| 23:44.05 | Mahmoud | damn.. |
| 23:44.05 | tzafrir_laptop | CrashSys, wtf? |
| 23:44.12 | dotSlashW | any help over here ? |
| 23:44.14 | tzafrir_laptop | those are all generated files |
| 23:44.34 | tzafrir_laptop | someone generated sloppy patches? |
| 23:45.04 | CrashSys | Hmmm... maybe this is it... sangoma's wiki says to basically install asterisk before installing wanpipe |
| 23:45.09 | CrashSys | Maybe that's the ticket :) |
| 23:49.05 | grey | any tips on getting decent performance out of MusicOnHold with app_mp3? It seems to stutter every few seconds, pretty annoying, the server load is extremely low, and asterisk isn't using very much cpu or memory at all, |
| 23:49.32 | CrashSys | Use files mode? |
| 23:50.22 | grey | it is, |
| 23:51.51 | BSD_Tech[laptop] | ok |
| 23:51.51 | dotSlashW | where should I read to learn how to debug a SIP trunk ? |
| 23:52.29 | blueneon | thanks for the help all.. im off for now |
| 23:52.31 | blueneon | \o |
| 23:53.03 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 23:54.42 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 23:54.55 | linagee | ack. indian telemarketers trying to sell me packet 8. hah |
| 23:55.01 | linagee | i was playing with them. :) |
| 23:55.15 | linagee | "will you be able to provide a SIP or IAX2 connection for my existing asterisk system?" |
| 23:55.16 | linagee | lol |
| 23:55.33 | CrashSys | "Yes sir... credit card please" |
| 23:55.57 | linagee | CrashSys: LOL |
| 23:56.08 | linagee | CrashSys: i was telling her of all the features that i have that she doesn't have. :) |
| 23:56.17 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 23:56.20 | CrashSys | One of those telemarketing scams where the idea is to charge the card in a foreign country... |
| 23:56.20 | linagee | CrashSys: "i can record phone calls outgoing and incoming. like this call right now even." :) |
| 23:56.24 | CrashSys | think nigeria here :) |
| 23:56.34 | linagee | CrashSys: "i can record phone calls outgoing and incoming. like this call right now even." :) |
| 23:56.55 | linagee | i think that made her a bit nervous. LOL. :) |
| 23:56.55 | CrashSys | Yup |
| 23:57.24 | linagee | CrashSys: i was like, "i pay half a cent per minute and about $50/mo for about 8 phone lines. can you do that?" |
| 23:57.35 | CrashSys | Although in florida (where I live) it's illegal to record the conversation without consent... |
| 23:57.40 | CrashSys | 2-party consent here |
| 23:57.53 | linagee | CrashSys: she was like, "$50 per phone line, right?" me: "no. $50 altogether". her: "sir, i think you are misunderstanding your phone bill. it couldn't be $50 for 8 lines" LOL. :) |
| 23:58.27 | linagee | CrashSys: then i had to explain, "I have 8 DIDs." her: "sir, what's a DID?" heh |
| 23:58.29 | CrashSys | Are they hard-lines? or VoIP? |
| 23:58.35 | linagee | CrashSys: voip of course. :) |
| 23:58.36 | grey | in most cases, a telemarketer just has to say 'I do not consent to this call being recorded' then can continue on, that makes the recording inadmissable as evidence (Thats afaik, it's what we were told to do when I was working tech support and someone stated they were recording the call) |
| 23:59.00 | linagee | grey: true true. she never said that though. and a remark like that would likely get an instant hangup from anyone. :) |
| 23:59.10 | *** part/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 23:59.12 | CrashSys | grey: Yup... and only like 13 states are 1-party consent... |
| 23:59.20 | linagee | grey: "thanks for playing. goodbye" |
| 23:59.32 | grey | eh, I've told people that before and they were just like 'uh.. ok whatever, lets continue' |
| 23:59.43 | linagee | CrashSys: i was like, "i think you did a great job reading from your script, but i really just don't think you can beat my current wholesale provider." hehehehe. :-> |
| 23:59.56 | linagee | CrashSys: she was like, "sir, i'm not reading from a script!" lol |
