| 00:00.06 | [TK]D-Fender | baliktad: No, stop looking at your dialplna, look where your PEER is sending them |
| 00:00.26 | [TK]D-Fender | baliktad: Looking for 14258828080 in phones (domain 192.168.0.116) SIP/2.0 404 Not Found <---------------- YOU set something in sip.conf to "context=phones" |
| 00:01.09 | baliktad | hmmm |
| 00:01.54 | baliktad | so the call came in and matched the [1000] context |
| 00:02.31 | [TK]D-Fender | baliktad: No, it came in and matched an ACCOUNT entry in sip.conf |
| 00:02.51 | [TK]D-Fender | baliktad: a "context" is a term reserverd for the sections of extensions.conf |
| 00:03.01 | baliktad | ok, sorry for that confusion |
| 00:03.26 | baliktad | let me see if I understand the flow so far |
| 00:03.48 | baliktad | I dial a number, which gets matched to a context in extensions.conf |
| 00:03.59 | C4colo | what switch do I use to only install ztdummy for the ./configure ? |
| 00:04.20 | [TK]D-Fender | C4colo: You don't. Thats in "make menuconfig" |
| 00:04.34 | baliktad | I intended to match the [outgoing_calls] context because my number was 10 digits |
| 00:04.42 | [TK]D-Fender | baliktad: First it looks up WHO is placing the call. Then it decids where yto look for the match |
| 00:04.53 | [TK]D-Fender | baliktad: No. |
| 00:04.53 | C4colo | yep just figured that out |
| 00:05.49 | [TK]D-Fender | baliktad: your [1000] section is the one you made for the phone that is placing the call. It tells * to send calls that it places to [phones] in your dialplan, because thats where you TOLD it to in line 43 - "context=phones" |
| 00:05.54 | baliktad | ok, the call is coming from the internal extension 1000 (matched in sip.conf) which dumped me into the [phones] context |
| 00:06.39 | [TK]D-Fender | baliktad: Better |
| 00:06.44 | *** join/#asterisk km2 (n=x@c-98-210-137-171.hsd1.ca.comcast.net) |
| 00:06.50 | [TK]D-Fender | baliktad: And never call a "device" an "extension" |
| 00:06.55 | baliktad | I think I am starting to see |
| 00:07.06 | [TK]D-Fender | baliktad: An extension is a number that you dial. |
| 00:07.17 | [TK]D-Fender | baliktad: Devices place calls to extensions. |
| 00:07.39 | baliktad | since I want my phone to be able to call both internal numbers and external numbers, I need to modify my [phones] context to include => outgoing_calls as well |
| 00:08.24 | [TK]D-Fender | baliktad: You need it to include whatever bits you want it to be able to do. |
| 00:09.20 | [TK]D-Fender | baliktad: But for your first day you're doing pretty good |
| 00:09.52 | baliktad | so the call comes in, matches account [1000] in sip.conf, this tells it the call should handled by the [phones] context in extensions.conf |
| 00:10.17 | baliktad | the [phones] context right now only includes the [internal] context, and my number doesn't match any of those |
| 00:10.33 | [TK]D-Fender | baliktad: Hence the 404 |
| 00:10.45 | [TK]D-Fender | (not found) |
| 00:10.55 | baliktad | my hypothesis is that I need to include the [outgoing_calls] context as well |
| 00:11.02 | baliktad | which I'm going to try right now :) |
| 00:11.11 | [TK]D-Fender | baliktad: Well that would include those extens. Go try and see |
| 00:15.49 | baliktad | well |
| 00:16.08 | baliktad | the good news is I'm getting farther, the bad news is... not far enough to make a call |
| 00:17.31 | UnixDog | yes asterisk 1.6 is working |
| 00:17.40 | UnixDog | coolio time to play |
| 00:18.27 | doolph | Anyone know why after I upgraded asterisk I lost dialtone on all Analog line |
| 00:22.00 | [TK]D-Fender | doolph: What "analog line"? Connected how? What version were you on before? And NOw? |
| 00:22.20 | baliktad | ok, now when it tries to place the call I get a few seconds before it fails, here's my sip debug output: http://pastebin.ca/1083898 |
| 00:24.48 | sharp | anybody have any good recomendations for a single port FXS ATA? |
| 00:26.55 | [TK]D-Fender | sharp: hard to find. Linkksys PAP2T-NA is the best budget choice |
| 00:27.06 | baliktad | I think the problem stems from line 125 "SIP/2.0 407 Proxy Authentication Required" which never gets successfully answered, leading to the eventual line 156 "SIP/2.0 603 Declined" |
| 00:27.07 | [TK]D-Fender | baliktad: Go specify codecs for all of your peers. |
| 00:28.00 | sharp | i've got an old iaxy, but it always had horrible echo problems |
| 00:28.28 | sharp | any idea on how to fix that? |
| 00:28.46 | [TK]D-Fender | baliktad: and [1REDACTEDNUM] should have "nat=no" |
| 00:28.53 | [TK]D-Fender | sharp: Replace it. |
| 00:29.50 | sharp | [TK]D-Fender: you mean call up digium and ask that they replace it? or buy another brand of ata? |
| 00:30.21 | [TK]D-Fender | sharp: You gt echo between it and another local SIP device? |
| 00:31.07 | sharp | when theres a phone call between iaxy and PSTN, the people on the other end hear tons of echo |
| 00:32.07 | [TK]D-Fender | sharp: Don't just say "PSTN". Say EXACTLY how you get there |
| 00:32.52 | baliktad | I added nat=no to that account but it doesn't solve the outbound dialing problem, which seems to be my provider requesting authentication and not getting it |
| 00:33.00 | sharp | iaxy -> local asterisk box -> internets -(sip)-> vitelity -> pstn |
| 00:33.30 | [TK]D-Fender | sharp: Do it happen when using another device than the IAXY |
| 00:33.50 | sharp | it happens on purely voip calls too |
| 00:34.02 | [TK]D-Fender | sharp: Then it just plain sucks. |
| 00:34.25 | sharp | yeah, well i haven't used the device in years |
| 00:34.36 | sharp | but i was wondering if all iaxys sucked so bad |
| 00:34.59 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 00:35.12 | [TK]D-Fender | sharp: Given their price its a complete waste to think of buying another. |
| 00:35.21 | sharp | yeah |
| 00:35.57 | sharp | but i'm wondering if theres any tweaks i can do |
| 00:36.04 | sharp | like hardware wise |
| 00:42.10 | baliktad | does "sip show registry" only show the parties that asterisk has registered to, or should it also show devices that have registered as clients to asterisk? |
| 00:47.14 | angryuser | have anyone used tool 'visual dialplan' ? not free just saw on mailing lists |
| 00:51.41 | Nugget | vi is as visual as I need. |
| 00:53.00 | Qwell | heck, it's got 2 of 6 letters |
| 00:54.34 | angryuser | it's a clone , they stole the code source |
| 00:56.54 | [TK]D-Fender | baliktad: "sip show peer [peerwithoutbaces]" will show an individual. "sip show registy" will show what * is registered ot |
| 00:58.17 | *** join/#asterisk L84supper (n=L84suppe@c-24-12-190-159.hsd1.il.comcast.net) |
| 00:58.29 | gramulhaozin | hey guys |
| 00:58.43 | gramulhaozin | anyone ever had a problem with No AUDIO when playing the IVR files |
| 00:58.46 | gramulhaozin | but audio on calls ? |
| 00:59.08 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 01:00.07 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:02.17 | [TK]D-Fender | gramulhaozin: Stop *. do "rmmod ztdummy", then restart * and test |
| 01:03.06 | gramulhaozin | [TK]D-Fender: did that already |
| 01:03.07 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 01:03.25 | [TK]D-Fender | gramulhaozin: describe everything involved in the call |
| 01:03.25 | gramulhaozin | I wasn't using ztdummy |
| 01:03.28 | gramulhaozin | so I tried loading |
| 01:03.36 | gramulhaozin | simple call |
| 01:03.37 | tzafrir_laptop | rmmod ztdummy? |
| 01:03.44 | gramulhaozin | insmod ztdummy |
| 01:03.44 | tzafrir_laptop | maybe rmmod zaptel as well |
| 01:03.59 | *** join/#asterisk Pokerastu (n=pokerast@lns-bzn-49f-81-56-173-76.adsl.proxad.net) |
| 01:04.04 | Pokerastu | bonjour a tous |
| 01:04.19 | tzafrir_laptop | but there's a simpler test, before loading/unloading modules |
| 01:04.20 | gramulhaozin | can't loose zaptel |
| 01:04.21 | gramulhaozin | :P |
| 01:04.27 | gramulhaozin | echo test ? |
| 01:04.36 | tzafrir_laptop | zttest -c 3 |
| 01:04.41 | tzafrir_laptop | What do you get? |
| 01:05.14 | gramulhaozin | measuring |
| 01:05.19 | gramulhaozin | right now I have ztdummy unloaded |
| 01:05.24 | gramulhaozin | so I think I have to load it to test |
| 01:05.44 | gramulhaozin | how long does it usually takes ? |
| 01:06.05 | gramulhaozin | loaded ztdummy and testes |
| 01:06.09 | gramulhaozin | Best: 0.000 -- Worst: 100.000 -- Average: 100.000000, Difference: 100.000000 |
| 01:06.13 | gramulhaozin | it's hunging |
| 01:06.21 | gramulhaozin | It's not even running the test |
| 01:06.34 | [TK]D-Fender | gramulhaozin: add "noload => chan_zap.so" to modules.conf |
| 01:06.39 | gramulhaozin | I had to ctrl+c to stop it |
| 01:06.55 | gramulhaozin | [TK]D-Fender: but I will loose the two zap modems right ? |
| 01:07.14 | [TK]D-Fender | gramulhaozin: What 2 "modems"? |
| 01:07.17 | doolph | hello I have a question about call pickup |
| 01:07.45 | doolph | I can do call pickups only from local extensions |
| 01:08.07 | doolph | but if it is an incoming call from FXO I cannot pickup the call |
| 01:11.03 | gramulhaozin | two ambient modems |
| 01:11.08 | gramulhaozin | [TK]D-Fender: two ambient modems |
| 01:11.28 | gramulhaozin | ttest -c 3 just hangs forever |
| 01:11.31 | [TK]D-Fender | gramulhaozin : go hunt down the ztdmmy module and just delete it. |
| 01:12.15 | gramulhaozin | but where do I get timming ? |
| 01:12.42 | UnixDog | rtc or ztdummy |
| 01:12.45 | [TK]D-Fender | gramulhaozin: Fro the CARDS themselves |
| 01:12.51 | *** join/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 01:12.56 | *** part/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 01:13.01 | gramulhaozin | o nice |
| 01:13.14 | *** join/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 01:13.15 | gramulhaozin | but it strange that even without ztdummy I still get zttest hanging |
| 01:13.21 | gramulhaozin | rmmod ztdummy |
| 01:13.31 | gramulhaozin | lsmod |grep ztdum (empty) |
| 01:13.34 | Qwell | what cards do you have? |
| 01:14.16 | *** join/#asterisk Steve_J-obs (n=Chris123@ip68-104-45-211.lv.lv.cox.net) |
| 01:14.40 | [TK]D-Fender | Qwell: [21:11]<gramulhaozin>[TK]D-Fender: two ambient modems |
| 01:14.45 | [TK]D-Fender | Qwell: CrapTASTIC! |
| 01:14.54 | gramulhaozin | :P |
| 01:15.05 | gramulhaozin | 05:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 01:15.05 | gramulhaozin | 05:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 01:15.07 | gramulhaozin | :P |
| 01:15.09 | gramulhaozin | Tiger3xxx |
| 01:15.10 | gramulhaozin | :P |
| 01:15.13 | gramulhaozin | Tiger Jet |
| 01:15.18 | gramulhaozin | Fear the TIGERRRRRRRR |
| 01:15.31 | Steve_J-obs | Can anybody please help me install asterisk on a godaddy vps? |
| 01:15.41 | Qwell | Steve_J-obs: ./configure && make && make install |
| 01:16.01 | C4colo | godaddy does vps now? |
| 01:16.17 | Steve_J-obs | I am getting configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 01:16.32 | C4colo | well, download gcc instead |
| 01:16.42 | Qwell | Steve_J-obs: distro? |
| 01:16.51 | C4colo | I've always found cpp to be a bit insane myself |
| 01:16.52 | [TK]D-Fender | Steve_J-obs: Go install all of *'s prerequisites. Its all layed out nice & neat in the BOOK. |
| 01:16.54 | [TK]D-Fender | ~book |
| 01:16.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:17.30 | Steve_J-obs | fedora 7, it has gcc 4.1.2 |
| 01:18.00 | Steve_J-obs | ./configure identifys almost all gcc installed |
| 01:18.17 | [TK]D-Fender | <PROTECTED> |
| 01:18.36 | doolph | anyone know why I cannot pickup a incoming call from fxo?? it is dialing to ext 101, I can pickup the ext 101 if it is dialed from local extensions |
| 01:18.36 | [TK]D-Fender | Steve_J-obs: install everything GCC related |
| 01:18.56 | Steve_J-obs | I think so too... anyone can point where I can doewnload it? |
| 01:19.04 | Qwell | Steve_J-obs: yum |
| 01:19.52 | jaytee | we need a new installer named nom |
| 01:19.56 | Steve_J-obs | ok thanks...I just hope I dontmess up the php setup |
| 01:20.02 | Qwell | ...php? |
| 01:20.12 | Steve_J-obs | yeah, the php libraries |
| 01:20.17 | [TK]D-Fender | Qwell: Yeah, its tucked in beneath the plasma relays |
| 01:20.29 | Qwell | since when does Asterisk use php? |
| 01:20.39 | [TK]D-Fender | Qwell: Just befor you reach the warp manifold & flux capacitor |
| 01:21.11 | Steve_J-obs | I will :) .... never mind me.... thanks guys...will let you know in a couple of hours :) |
| 01:22.10 | baliktad | ok I'm officially stuck on this proxy authentication thing |
| 01:22.26 | baliktad | here's my sip debug :( http://pastebin.ca/1083922 |
| 01:22.35 | jaytee | officially? you have the paperwork to prove it? |
| 01:23.30 | baliktad | does a worn out spot on my desk count? |
| 01:24.33 | [TK]D-Fender | baliktad: add "insecure=port,invite" to your peer |
| 01:24.47 | jaytee | nope, you need a notarized Form 1113111-B22-C from the Vogon homeworld |
| 01:25.56 | [TK]D-Fender | rolls a backhoe over jaytee |
| 01:26.13 | jaytee | ouch!!!!! |
| 01:27.01 | baliktad | [TK]D-Fender: OK, I changed the line insecure=invite to insecure=port,invite under my provider's account, but no such luck, same results |
| 01:27.58 | doolph | hello no body can help me with call pickups? |
| 01:28.01 | [TK]D-Fender | baliktad: Also, change the peer name to [flowroute], and set codecs like I told you to. |
| 01:33.37 | baliktad | [TK]D-Fender: changed the peer name as well as the reference in my dialplan. I've looked over my codec settings but didn't see anything I should change. |
| 01:33.57 | [TK]D-Fender | baliktad: pastebin your setup |
| 01:34.04 | [TK]D-Fender | (msaking only passwords) |
| 01:34.04 | baliktad | sure, 1 sec |
| 01:34.09 | *** join/#asterisk ming_zym (n=ming_zym@220.181.35.203) |
| 01:36.38 | baliktad | http://pastebin.ca/1083928 |
| 01:38.49 | Steve_J-obs | guys, what does error: 'termcap support not found' means?? it ain't letting me do the asterisk install |
| 01:39.17 | jaytee | it means you need to install libtermcap-devel |
| 01:39.20 | Steve_J-obs | I just installed the gcc++ with yum and worked well.... |
| 01:39.30 | jaytee | what distro are you using? |
| 01:39.39 | Steve_J-obs | fedora 7 |
| 01:40.02 | [TK]D-Fender | Steve_J-obs: I means "holy shit, I should go read that list of prerequisites rather than give a play by play while asking to be handfed every little step" |
| 01:40.03 | Steve_J-obs | fedora 7, linux kernel 2.6, gcc 4.1 |
| 01:41.01 | jaytee | ok, go here and follow the instructions for installing all the dependencies at the beginning, it's for CentOS but it should work for Fedora for the deps at least. http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 01:41.10 | [TK]D-Fender | baliktad: try changing "fromuser" to "usernamr" |
| 01:41.28 | [TK]D-Fender | username* |
| 01:41.54 | Steve_J-obs | <PROTECTED> |
| 01:42.35 | *** join/#asterisk friendly12345 (n=friendly@ppp59-167-142-196.lns3.mel6.internode.on.net) |
| 01:43.35 | baliktad | holy F it WORKED! |
| 01:43.46 | jaytee | woohoo!!! |
| 01:44.24 | [TK]D-Fender | baliktad: Merry Christmas |
| 01:45.03 | baliktad | aww thanks TK *hugs* |
| 01:45.35 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 01:48.50 | baliktad | of course, fromuser is what is transmitted as the identity (=caller-ID), username is what asterisk uses to auth. to the provider |
| 01:49.28 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 01:51.03 | jaytee | ok, I'm not sure what purpose this whole thing serves but I thought it was strange enough to share. http://goosh.org |
| 01:53.27 | outtolunc | web s goosh |
| 01:53.48 | outtolunc | one of the hits is '... i'm goosh-ing' <G> |
| 01:54.12 | jaytee | I mean what's the whole point? it's not a real shell if it's hypertext |
| 01:54.27 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 01:56.10 | outtolunc | its is kindof cute, got lang settings and cookies and its mostly on one line for readability <G> |
| 01:56.16 | *** part/#asterisk kimc (n=kimc@c-71-205-244-134.hsd1.mi.comcast.net) |
| 02:01.24 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 02:02.48 | baliktad | *dies* |
| 02:03.02 | baliktad | there's a typo in TFOT that has been killing me for 2 hours |
| 02:04.26 | baliktad | page 101, the example says to include this line in your [incoming_calls] context: exten => _X.,1.NoOp() |
| 02:04.41 | baliktad | of course, if you look closely, it's a period after 1 instead of a comma like it should be |
| 02:04.49 | jaytee | should be a comma, yep |
| 02:05.09 | baliktad | well serves me right for copying and pasting from the damn textbook |
| 02:05.18 | baliktad | I guess I learned some debugging in the process |
| 02:05.36 | *** join/#asterisk MaliutaLap (n=nikolai@124-171-128-169.dyn.iinet.net.au) |
| 02:05.36 | jaytee | and on pages 112, 113 and 114 and 115 where it has examples for setting up IAX trunks the contexts are reversed. |
| 02:05.47 | outtolunc | should have been a error on extensions reload |
| 02:06.24 | baliktad | it didn't cause an error until a call actually came through, then it gives some cryptic error about not finding application '' |
| 02:07.01 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 02:07.27 | outtolunc | i just did a test on a 1.2 box and it gave an error |
| 02:08.07 | jaytee | 1.2? my grandmother ran that on her pbx! |
| 02:09.18 | [TK]D-Fender | baliktad: because without the comma it tok the rest as the priority |
| 02:09.34 | mattwj2002 | hey guys |
| 02:09.52 | mattwj2002 | I have an idea for an Asterisk project....but I am wondering if it is legal to do |
| 02:10.05 | outtolunc | 1.4 also errors.. maybe your grandma .. well we will leave that for another discussion |
| 02:10.23 | jaytee | legal, shmegal, who cares? this is the Bush/Cheney era! Anything goes, even torture. |
| 02:10.41 | mattwj2002 | if I buy a 1800 number and setup so I can make long distance calls can I do that |
| 02:10.54 | mattwj2002 | it would be similar to how a call card works |
| 02:11.18 | [TK]D-Fender | mattwj2002: No, and all the other hundreds of compainies that are doing it are going to be raided. |
| 02:11.27 | jaytee | lol |
| 02:11.31 | mattwj2002 | :O |
| 02:11.48 | mattwj2002 | your kidding right? |
| 02:11.55 | [TK]D-Fender | DUH |
| 02:11.59 | mattwj2002 | lol |
| 02:12.31 | mattwj2002 | okay what if I do that same thing but using a local number |
| 02:12.45 | mattwj2002 | I live in one state and my mom lives in another |
| 02:12.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:13.08 | mattwj2002 | so I have my area code (local DID) -> my mom's number |
| 02:13.09 | jaytee | use a cell phone with a nationwide calling plan |
| 02:13.27 | mattwj2002 | lol |
| 02:13.50 | mattwj2002 | okay and one last question before I do some configuration |
| 02:14.06 | mattwj2002 | what do you guys think of the following two providers if anything.... |
| 02:14.20 | mattwj2002 | http://www.callwithus.com/ |
| 02:15.12 | mattwj2002 | http://www.sellvoip.net/ |
| 02:17.46 | MaliutaLap | anyone know of a linux softphone with upnp support? |
| 02:18.09 | MaliutaLap | I find myself trapped in a network I don't control |
| 02:18.15 | justdave | got a user using XMeeting on OS X trying to connect to asterisk, and it keeps giving him "user unknown" errors when he tried to dial (he's successfully registered). When I turn on SIP debugging for his IP, he's sending INVITE sip:1.2.3.4 instead of sip:extension@ourdomain . |
| 02:18.38 | justdave | I'm not seeing any settings in XMeeting for dial-by-IP or anything... anyone have any ideas? |
| 02:19.13 | justdave | (FWIW, XMeeting is working fine for me, and his settings for the account and location seem to match mine) |
| 02:20.10 | jaytee | MaliutaLap, supposedly Ekiga will do upnp |
| 02:20.24 | justdave | and if anyone knows of discussion areas for XMeeting, feel free to point me there instead. Their website is a bit sparse on the "how to find help" details. |
| 02:21.27 | justdave | (they have an SF tracker and that's about it as far as I can tell) |
| 02:22.23 | [TK]D-Fender | justdave: www.google.com |
| 02:22.39 | justdave | yeah, I googled for a while before I came here |
| 02:24.16 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:26.20 | baliktad | is there a way I can insert a delay/pause in my extension handling? |
| 02:27.20 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 02:29.05 | jaytee | baliktad, have you looked in the book? |
| 02:29.07 | jaytee | ~book |
| 02:29.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:29.23 | baliktad | I searched for pause, delay, wait |
| 02:29.33 | jaytee | because in the book it has all kinds of goodies like the Wait() application |
| 02:29.35 | [TK]D-Fender | baliktad: well "wait" was it |
| 02:30.18 | baliktad | ahh |
| 02:30.25 | baliktad | it's always easier when you know the answer |
| 02:30.31 | jaytee | go forth and do your homework! |
| 02:31.54 | [TK]D-Fender | baliktad: Read up on how to use variables & functions. then go read teh COMPLETE function & application list and read each one |
| 02:32.06 | jaytee | it's always easier to have someone else write your term paper for you too. |
| 02:42.15 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 02:49.39 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 03:03.21 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-34.syrcny.fios.verizon.net) |
| 03:27.51 | *** join/#asterisk davidc (n=david@netman1.us.sargasso.net) |
| 03:38.22 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 04:08.00 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 04:30.11 | *** join/#asterisk codelemur (n=codelemu@24.96.140.159) |
| 04:34.56 | *** join/#asterisk [T]ank (n=ckwall@75-162-174-132.slkc.qwest.net) |
| 04:35.09 | [T]ank | anyone here a grandstream ata user? |
| 04:35.31 | [T]ank | i can get sip phones connected through my firewall... but i cannot get even close to it with my grandstream ata and could use some help |
| 04:36.32 | [TK]D-Fender | ~sipnat |
| 04:36.33 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:38.19 | [T]ank | i dont think its nat... cuz like i said, i can get sip phones to work just fine... its just this damn ata |
| 04:38.32 | [TK]D-Fender | [T]ank: All talk, no show |
| 04:39.01 | [T]ank | [TK]D-Fender: ? |
| 04:39.23 | [TK]D-Fender | [T]ank: And the reason you don't have SIP debug already in a pastebin on asking that is....? |
| 04:41.13 | [T]ank | http://pastebin.ca/1084053 |
| 04:45.43 | [T]ank | any ideas? |
| 04:45.53 | [TK]D-Fender | [T]ank: <--- SIP read from 75.162.174.132:15060 ---> Contact: <sip:102@192.168.15.101> <--- Transmitting (no NAT) to 192.168.15.101:5060 ---> |
| 04:46.02 | [TK]D-Fender | [T]ank: Yes, you didn't set up your peer right |
| 04:47.53 | [T]ank | http://pastebin.ca/1084060 |
| 04:48.26 | [T]ank | is what I have... |
| 04:48.36 | [TK]D-Fender | [T]ank: PB all of [general] as well. |
| 04:49.06 | [T]ank | http://pastebin.ca/1084062 |
| 04:49.12 | [TK]D-Fender | [T]ank: Contact: <sip:102@192.168.1.10> <- and * iis responding back with its PRIVATE IP |
| 04:49.43 | [TK]D-Fender | [T]ank: And you seem not to have done anything about * being set up behind NAT. Go read the guide this time |
| 04:53.13 | [TK]D-Fender | [T]ank: ... AND SIP/2.0 401 Unauthorized <--- the auth isn't even RIGHT. Thats just insult to ingury on the completely absent NAT config |
| 05:01.24 | [T]ank | ok, I have the ata device registered and I can dial it... however I get no sound. I have set the rtp port in the firewall to 10000-10010 as well as my rtp.conf |
| 05:03.12 | [TK]D-Fender | [T]ank: And your new sip.conf? |
| 05:04.49 | [T]ank | http://pastebin.ca/1084075 |
| 05:06.03 | *** join/#asterisk ming_zym (n=ming_zym@203.212.6.223) |
| 05:06.39 | [TK]D-Fender | [T]ank: what exactly have you forwarded? And then follow by a pastebin of a new call |
| 05:07.08 | [T]ank | 5060 udp and 10000-10010 to my asterisks ip address |
| 05:08.20 | [T]ank | http://pastebin.ca/1084076 |
| 05:08.22 | [TK]D-Fender | What 10000-10010? |
| 05:08.29 | [T]ank | udp |
| 05:09.22 | [TK]D-Fender | [T]ank: Retransmitting #4 (no NAT) to 192.168.15.101:5060: <- srtill no good |
| 05:09.31 | [T]ank | what am i missing? |
| 05:09.37 | [T]ank | where do i make that change? |
| 05:10.03 | [TK]D-Fender | [T]ank: You went and said that 102 was "nat=no". |
| 05:10.25 | [T]ank | ive tried both |
| 05:10.42 | [TK]D-Fender | [T]ank: and you go and show me something clearly wrong. |
| 05:12.01 | [T]ank | http://pastebin.ca/1084080 |
| 05:14.59 | [TK]D-Fender | [T]ank: have you forwarded anything on the remote side? |
| 05:15.16 | [T]ank | its plugged directly into the net |
| 05:15.32 | [TK]D-Fender | [T]ank: Not if its reporting a private IP like that it isn't... |
| 05:17.31 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-134-114.dsl.stlsmo.sbcglobal.net) |
| 05:17.44 | *** join/#asterisk Magnus_H (n=magnus@90-228-242-54-no71.tbcn.telia.com) |
| 05:25.21 | *** join/#asterisk [T]ank (n=ckwall@75-162-174-132.slkc.qwest.net) |
| 05:25.48 | *** part/#asterisk [T]ank (n=ckwall@75-162-174-132.slkc.qwest.net) |
| 05:51.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:14.59 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-182-169.dsl.tul2ok.sbcglobal.net) |
| 06:15.17 | *** join/#asterisk bkw___ (n=brian@adsl-70-234-182-169.dsl.tul2ok.sbcglobal.net) |
| 06:34.55 | *** part/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 06:46.56 | *** join/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 06:47.17 | mattwj2002 | hi everyone |
| 06:47.25 | x86 | HAI |
| 06:47.38 | mattwj2002 | well I had a lot of success with my pbx tonight |
| 06:48.00 | mattwj2002 | I have it setup so I can dial an 800 number and dial any long distance number I want :D |
| 06:48.24 | x86 | uh huh |
| 06:48.35 | x86 | that's very elementary stuff ;) |
| 06:48.42 | mattwj2002 | lol |
| 06:48.47 | mattwj2002 | indeed |
| 06:48.55 | mattwj2002 | I am just a hobbyst and do this for fun :) |
| 06:49.21 | x86 | twas how i got started with asterisk 4 years ago ;) |
| 06:49.32 | x86 | before that I knew nothing about telephony :P |
| 06:50.39 | mattwj2002 | I work with Cisco Callmanager for a living |
| 06:50.44 | mattwj2002 | but that is a lot different |
| 06:52.35 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 06:52.47 | x86 | sorry to hear that |
| 06:52.56 | x86 | tis quite unfortunate, ccm |
| 06:53.42 | mattwj2002 | hey it works for me |
| 06:53.52 | mattwj2002 | we all need to do something to get paid :) |
| 06:55.00 | *** join/#asterisk nicox (n=nicox@212-183-45-141.adsl.highway.telekom.at) |
| 06:55.09 | x86 | true that :) |
| 06:56.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:59.21 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 07:03.19 | mattwj2002 | well I am off to bed |
| 07:03.22 | mattwj2002 | good night all :) |
| 07:03.29 | *** part/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 07:03.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:14.41 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 07:32.49 | *** join/#asterisk Steve_J-obs (n=Chris123@ip68-104-45-211.lv.lv.cox.net) |
| 07:40.26 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-6-148.revip2.asianet.co.th) |
| 07:46.25 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 08:18.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:19.04 | *** join/#asterisk cplx (n=cplx@203.206.204.213) |
| 08:19.10 | cplx | hi guys.. anyone here pretty full bottle with a CME or callmanager & Asterisk integration? |
| 08:19.18 | carrar | t |
| 08:19.20 | carrar | err |
| 08:19.37 | carrar | ~centos52bug |
| 08:19.37 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 08:20.36 | carrar | if you are compiling Asterisk on CentOS 5.2, why would you such a old version |
| 08:23.45 | *** part/#asterisk Magnus_H (n=magnus@90-228-242-54-no71.tbcn.telia.com) |
| 08:34.00 | d00gster | guys for a remote office with analog pstn only, can I use a spa9000 or spa400 to bridge inbound calls over to my * server? |
| 08:41.29 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 08:42.13 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 08:45.28 | Steve_J-obs | hello everybody, easy question: I just compiled asterisk and now I can't find the freaking executable |
| 08:51.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:51.51 | *** join/#asterisk ming_zym (n=ming_zym@203.212.6.223) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk davidc (n=david@netman1.us.sargasso.net) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk ixx (n=ixx@70.114.153.233) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 08:51.52 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk [netman] (n=netman@12.Red-88-25-139.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) [NETSPLIT VICTIM] |
| 08:51.52 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM] |
| 08:53.51 | Steve_J-obs | hello everybody, I just compiled asterisk bt I cant find the execubtale |
| 08:57.10 | styelz | did you make install ? |
| 08:57.38 | styelz | type "which asterisk" it should tell you |
| 08:58.33 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) |
| 09:00.26 | Steve_J-obs | yes |
| 09:00.31 | Steve_J-obs | everything ran well |
| 09:01.02 | Steve_J-obs | install successful... no errors...asterisk 1.4.... I just cant find the freaking execuatable |
| 09:01.18 | Steve_J-obs | 'command not found' |
| 09:01.27 | Steve_J-obs | and all the files are in place |
| 09:03.30 | styelz | maybe its not in your path |
| 09:05.09 | Steve_J-obs | but whre is it? it aint at /usr/bin |
| 09:05.43 | styelz | look in /usr/sbin or /usr/local/sbin |
| 09:06.17 | Steve_J-obs | what file should I look for? one specifically called asterisk ? |
| 09:06.24 | styelz | yea |
| 09:08.51 | Steve_J-obs | wow.... strange styelz... on /usr/sbin, I typed ./asterisk, and it worked!... how do I set it to the path? |
| 09:09.25 | styelz | usually you start it from init.d /etc/init.d/asterisk start |
| 09:09.58 | *** join/#asterisk km2 (n=x@c-98-210-137-171.hsd1.ca.comcast.net) |
| 09:10.34 | styelz | check your shell's login script |
| 09:10.36 | Steve_J-obs | you mean, I go cd /etc/init.d? |
| 09:10.55 | styelz | no, just type /etc/init.d/asterisk start|stop|restart |
| 09:11.12 | Steve_J-obs | whre do I check my shells login script? |
| 09:11.19 | styelz | dont know |
| 09:11.54 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 09:12.12 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 09:13.14 | Steve_J-obs | no, there is no file asterisk on folcer /etc/init.d |
| 09:13.20 | tzafrir_laptop | Steve_J-obs, /usr/sbin and /usr/local/sbin should be in the PATH of root |
| 09:13.34 | tzafrir_laptop | OTOH, /etc/init.d shouldn't be |
| 09:13.37 | unpaidbill | how is your cancer mr jobs |
| 09:14.12 | Steve_J-obs | this nickname has given me more trouble than anything else :) |
| 09:15.18 | Steve_J-obs | strangely, I do notice certain antagonism in the community towards steve... dont know why... maybe because of all the money he has... thanks for asking |
| 09:15.35 | MaliutaLap | unpaidbill: that's not funny. As the holder of Highest Circulating Leukaemic count in an adult, I take offence |
| 09:17.21 | styelz | no file in init.d.. i think you need to type make config for that |
| 09:17.25 | styelz | cant remember |
| 09:22.05 | Steve_J-obs | wow stylelz, you are great... make config did put file asterisk on init.d |
| 09:22.38 | Steve_J-obs | had it been in the readme file of digium, I'd have done it without asking |
| 09:25.10 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:27.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:41.04 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:44.41 | *** join/#asterisk nicox_ (n=nicox@213-33-11-28.adsl.highway.telekom.at) |
| 09:49.50 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 10:05.33 | *** join/#asterisk nicox (n=nicox@213-33-15-124.adsl.highway.telekom.at) |
| 10:33.35 | *** join/#asterisk GertZA (n=gert@dsl-240-179-218.telkomadsl.co.za) |
| 10:34.22 | GertZA | Hi all |
| 10:35.14 | GertZA | I need help with making outgoing PTSN calls from asterisk using a spa3102 |
| 10:37.32 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 10:37.47 | GertZA | I need help with making outgoing PTSN calls from asterisk using a spa3102 |
| 10:44.16 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-5652a4eeef405979) |
| 11:03.51 | *** join/#asterisk shtoom (n=shtoom@123.176.41.46) |
| 11:17.47 | *** join/#asterisk nicox (n=nicox@212-183-47-166.adsl.highway.telekom.at) |
| 11:27.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:27.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:29.10 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:46.41 | *** join/#asterisk pjezek (n=pj@193.85.164.154) |
| 11:54.48 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:54.48 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:00.26 | *** join/#asterisk unafilliated (n=nX2k@unaffiliated/unafilliate) |
| 12:21.33 | *** join/#asterisk Dovid (n=Dovid@bzq-79-176-161-20.red.bezeqint.net) |
| 12:22.07 | Dovid | any one here using QUintum ? |
| 13:03.55 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 13:04.22 | Dovid | anyone here have a sugestion for SIP -> H.323 ? |
| 13:11.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:16.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:32.08 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:39.36 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 13:43.27 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 13:48.27 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 13:50.34 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 14:00.29 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 14:02.04 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:09.42 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:14.33 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-76-rrdg-esr-2.dynamic.isadsl.co.za) |
| 14:15.00 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 14:15.57 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 14:29.12 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 14:30.37 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 14:48.20 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 14:56.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:16.51 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 15:18.05 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 15:18.28 | k-man | how can i force my linksys 942 to attempt to reregister more quickly? |
| 15:19.09 | [TK]D-Fender | k-man: and why would you feel the need to do that? |
| 15:19.12 | k-man | err.. i'm trying to get it to register with asterisk and it failed the first time |
| 15:19.19 | k-man | cos i had my settings wrong |
| 15:19.21 | [TK]D-Fender | k-man: reboot it |
| 15:19.26 | k-man | ok |
| 15:19.48 | [TK]D-Fender | k-man: they boot in a few seconds |
| 15:19.48 | k-man | ok, thanks |
| 15:20.13 | *** join/#asterisk xenonex (n=xenonex@89.218.233.159) |
| 15:30.06 | k-man | [TK]D-Fender, should the "user id" in the extension config on the 941 match the [namehere] bit in sip.conf? |
| 15:30.19 | *** join/#asterisk nicox (n=nicox@212-183-47-166.adsl.highway.telekom.at) |
| 15:30.30 | k-man | ie, userid=phone1 and the section in sip,conf should also be [phone1] |
| 15:31.00 | ManxPower | Not "should", but "must" |
| 15:31.12 | k-man | right |
| 15:31.24 | ManxPower | [this_is_the_userid_asterisk_expects_to_receive] |
| 15:31.24 | k-man | thanks ManxPower |
| 15:31.31 | ManxPower | fromuser=username_asterisk_sends |
| 15:31.37 | k-man | yes, thats what I thought, i was just checking |
| 15:36.05 | [TK]D-Fender | k-man: and "userid" is not a valid option. You do not need "ay other naming than [namehere] |
| 15:51.11 | *** join/#asterisk Katty (n=asterisk@hera.copi-rite.com) |
| 15:51.25 | Katty | morning |
| 15:53.26 | *** join/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 15:58.18 | tzafrir_laptop | evening |
| 16:02.26 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 16:10.14 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 16:13.31 | *** join/#asterisk ar3dam (n=ar3dam@189.156.215.56) |
| 16:14.37 | ar3dam | somebody can help, i try make calls to pstn, when i type 7 for make the call display me this msg " Unable to create channel of type 'Zap'", any advice? |
| 16:15.07 | *** join/#asterisk hardwire (n=hardwire@rdbk-5282.mtaonline.net) |
| 16:15.14 | hardwire | y0 y0 y0 y0 |
| 16:16.16 | ar3dam | somebody can help, i try make calls to pstn, when i type 7 for make the call display me this msg " Unable to create channel of type 'Zap'", any advice? |
| 16:18.55 | tzafrir_laptop | ar3dam, please provide a trace from the CLI of the call attempt |
| 16:23.22 | *** join/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 16:23.50 | hardwire | anybody know the story on Newman Telecom? |
| 16:23.52 | hardwire | they still around? |
| 16:24.01 | hardwire | anybody taken over their faxdetect modules? |
| 16:25.11 | ManxPower | hardwire: It could be in the most recent asterisk-addons |
| 16:25.31 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-126-8-150.bflony.east.verizon.net) |
| 16:28.04 | hardwire | hmm |
| 16:28.07 | hardwire | wonders what app_fax is |
| 16:28.09 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 16:29.11 | ManxPower | hardwire: sad that someone that made one of the most useful add-on for Asterisk has mostly disappeared. |
| 16:29.21 | hardwire | it seems to be that way. |
| 16:29.39 | hardwire | tears the internet up looking for app_fax documentation |
| 16:30.00 | ManxPower | we run it against almost every single call that comes into the system from the PSTN |
| 16:30.12 | hardwire | I use zap fax detect |
| 16:30.15 | hardwire | it bothers me |
| 16:30.51 | ManxPower | zap faxdetect is far, far, far too limited. |
| 16:30.57 | hardwire | it needs a timeout |
| 16:31.06 | ManxPower | it needs to be turned on and off in the dialplan |
| 16:31.16 | hardwire | or there needs to be a Set(ZapFaxDetect=1) |
| 16:31.18 | hardwire | yeh |
| 16:31.36 | hardwire | sounds like a simple thing to do really. |
| 16:31.46 | *** join/#asterisk moy (n=moy@189.169.66.1) |
| 16:32.44 | hardwire | sigh |
| 16:32.53 | ManxPower | then code it up! 8-) |
| 16:32.58 | hardwire | yeh.. nomore answering machine detection either. |
| 16:33.11 | hardwire | and app_fax has no detection dialplan functions |
| 16:33.32 | hardwire | just SendFax and ReceiveFax |
| 16:34.11 | hardwire | http://svn.digium.com/view/asterisk/trunk/apps/app_fax.c?view=markup |
| 16:35.07 | hardwire | giggles at app_minivm |
| 16:35.12 | ManxPower | We do things like test for channel type first and only run nvfaxdetect against Zap channels |
| 16:35.52 | hardwire | right now I just detect all faxes inbound on all DID's |
| 16:36.11 | hardwire | it goes to a macro, which boots faxes back to the calling context |
| 16:36.35 | hardwire | it sets ${macro_exten} and the fax,1,.... dialplan entry does the routing |
| 16:36.56 | hardwire | between a fax server and my local hylafax server (iaxmodem over gigabit) |
| 16:37.15 | hardwire | I'm pretty impressed with iaxmodem, but I'd love to not need it. |
| 16:37.29 | *** join/#asterisk PepOSX (n=angeldav@190.72.148.183) |
| 16:42.29 | *** join/#asterisk john_fbac (n=john_fba@241.sub-75-200-32.myvzw.com) |
| 16:42.48 | hardwire | hey neat.. ${faxexten} |
| 16:44.17 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 16:44.38 | hardwire | ManxPower: apparently if I wanted to continue using fax detection via pstn channels, I will have to limit which contexts have a "fax" extension. |
| 16:45.14 | hardwire | and there doesn't appear to be a call timeout for accepting a fax, apparently somebody out there can spend several minutes on the phone then suddenly transmit a fax out of nowhere. |
| 16:45.26 | *** join/#asterisk alami (n=alami@81.192.164.151) |
| 16:53.32 | tzafrir_laptop | ManxPower, nvfaxdetect is channel independent. And chan_zap has a built-in fax detection |
| 16:56.19 | hardwire | ? |
| 16:59.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:03.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:05.51 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 17:09.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:09.52 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 17:20.16 | outtolunc | sneezes |
| 17:20.41 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 17:24.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:28.41 | ManxPower | tzafrir_laptop: Yes, but fax over voip is evil, so we don't support it in our dialplans |
| 17:30.26 | tzafrir_laptop | The problem with turning on fax detection in the dialplan is that by the time it gets there it is often too late to detect, isn't it? |
| 17:30.56 | ManxPower | tzanger: not at all. |
| 17:31.14 | hardwire | tzafrir_laptop: not really, no |
| 17:31.34 | ManxPower | tzafrir_laptop: Unless you are talking about T.38 fax, in that case it would be. |
| 17:31.49 | hardwire | that requires direct extensions |
| 17:32.10 | ar3dam | tzafrir_laptop, hi there... |
| 17:32.20 | ar3dam | tzafrir_laptop, Executing [71604367@default:1] Dial("SIP/102-b6f026b0", "Zap/G2/1604367") in new stack |
| 17:32.20 | ar3dam | [Jul 27 08:25:03] WARNING[3845]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 17:33.01 | hardwire | do you have a group 2? |
| 17:33.02 | ManxPower | ar3dam: Any of these 1) you did not install Zaptel BEFORE installing Asterisk, 2) your /etc/asterisk/modules.conf is screwed up. |
| 17:33.27 | ManxPower | oh, or 3) the zaptel kernel modules are not loaded into memory. |
| 17:33.43 | tzafrir_laptop | ar3dam, pastebin the contents of your zapata.conf and the output of: asterisk -rx 'zap show channels' |
| 17:36.32 | d00gster | guys for a remote office with analog pstn only, can I use a spa9000 or spa400 to bridge inbound calls over to my * server or do I have to make another * server and iax2 between the two ? |
| 17:36.50 | ar3dam | ok, tzafrir_laptop letme pastebin |
| 17:37.41 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-141-153.neo.res.rr.com) |
| 17:37.55 | tzafrir_laptop | d00gster, how many phones in that office? |
| 17:38.10 | d00gster | 2-4 not sure ... |
| 17:38.13 | tzafrir_laptop | err.. sorry, misread |
| 17:39.25 | d00gster | the idea is to build a virtual office for a lawyer in the thailand. want to get 204 analog phone line and when a client calls, I want my local (north america) * to answer |
| 17:40.07 | tzafrir_laptop | What should happen if the internet connection goes down? |
| 17:40.45 | d00gster | then customers don't get anywhere I guess. |
| 17:40.49 | tzafrir_laptop | can they talk to each other? (and does it matter?) |
| 17:40.57 | d00gster | ? |
| 17:41.22 | d00gster | they call the office and get a busy number for all what we care |
| 17:41.41 | d00gster | I think the spa9000 has a pbx in it. |
| 17:41.42 | ar3dam | ok, tzafrir_laptop letme pastebin |
| 17:41.52 | ar3dam | ok, tzafrir_laptop, http://pastebin.com/d4e125d82 |
| 17:43.13 | tzafrir_laptop | ar3dam, hmm... that looks like the sample config file |
| 17:43.29 | tzafrir_laptop | Have you configured zaptel channels in any way? If so: how? |
| 17:45.15 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 17:45.38 | ar3dam | tzafrir_laptop, yes it from make sample files..... |
| 17:46.09 | ar3dam | i configure the channel with genzaptelconf... |
| 17:47.32 | outtolunc | d00gster: does the spa9000 even have the ability to connect to the pstn? |
| 17:47.50 | d00gster | has fxo |
| 17:48.04 | outtolunc | right for connecting an fxs phone |
| 17:48.14 | outtolunc | not to connect to another fxo (pstn) |
| 17:48.44 | outtolunc | at least the last time i looked |
| 17:49.17 | outtolunc | from the pdf |
| 17:49.34 | outtolunc | important: do not conenct the phone port to a telephone wall jack |
| 17:50.02 | d00gster | ah |
| 17:50.16 | d00gster | so what's the best option? |
| 17:50.25 | d00gster | get an * applicance? |
| 17:51.36 | tzafrir_laptop | ar3dam, echo '#include zapata-channels.conf' >>/etc/asterisk/zapata.conf |
| 17:51.50 | outtolunc | that would be a good guess, as they are lawyers, not techies |
| 17:51.56 | tzafrir_laptop | asterisk -rx 'module unload chan_zap.so' |
| 17:52.02 | tzafrir_laptop | asterisk -rx 'module load chan_zap.so' |
| 17:53.09 | d00gster | spa400 is the one then |
| 17:53.51 | ar3dam | when i include zapata-channels.conf show a msg after load chan_zap.so, let me show you... wait pls. |
| 17:55.48 | *** join/#asterisk mshades (n=mirrored@host-69-146-198-70.chy-wy.client.bresnan.net) |
| 17:56.22 | outtolunc | well i think you need both the spa400 and spa9000 |
| 17:57.03 | ar3dam | tzafrir_laptop, this is a msg "[Jul 27 08:52:48] WARNING[3875]: config.c:806 process_text_line: No '=' (equal sign) in line 42 of /etc/asterisk/zapata.conf" |
| 17:57.13 | *** join/#asterisk VE3ABZ (i=Nighthaw@CPE001c103411e5-CM000f9f7b1f0c.cpe.net.cable.rogers.com) |
| 17:57.30 | ar3dam | and in the line 42 i have the include zapata-channels.conf |
| 17:59.27 | tzafrir_laptop | sed -n 42p /etc/asterisk/zapata-channels.conf # to print that line no. 42 |
| 18:02.13 | ar3dam | maybe i can explaint good. u tellme i need to include zapata.conf in zapata-channels.conf or inverse in zapata.conf add include zapata-channels.conf????? |
| 18:03.14 | outtolunc | he wants you to pastebin the 'line 42' of /etc/asterisk/zapata.conf |
| 18:03.43 | outtolunc | as asterisk things there should be an equals sign, but there isn't one |
| 18:03.46 | outtolunc | er thinks |
| 18:04.01 | tzafrir_laptop | it's one line. it can be pasted here |
| 18:04.34 | ar3dam | this is the line 42? .. include 'zapata-channels.conf' |
| 18:04.44 | ar3dam | this is the line 42? .. include 'zapata-channels.conf' |
| 18:04.51 | ar3dam | this is the line 42 .. include 'zapata-channels.conf' |
| 18:04.52 | tzafrir_laptop | can you please paste here that output of that command? |
| 18:05.03 | ar3dam | from sed? |
| 18:05.10 | tzafrir_laptop | right |
| 18:05.56 | ar3dam | sed -n 42p /etc/asterisk/zapata.conf |
| 18:05.57 | ar3dam | include 'zapata-channels.conf' |
| 18:06.21 | tzafrir_laptop | Right. So the '#' sign is missing there at the beginning of the line |
| 18:06.42 | tzafrir_laptop | It should be: #include zapata-channels.conf |
| 18:07.38 | outtolunc | or include => zapata-channels.conf if he reallys wants too |
| 18:07.51 | tzafrir_laptop | outtolunc, no . that 'include' is different |
| 18:08.15 | ar3dam | ok.... look with the # |
| 18:08.23 | tzafrir_laptop | it's something that is only used in extensions.conf, and not generic pre-processing of config files |
| 18:09.46 | mshades | asterisk is my sexual therapy |
| 18:10.42 | hardwire | that's a bit freaky |
| 18:10.44 | hardwire | but ok |
| 18:10.46 | outtolunc | yep, i got = stuck in my head <G> |
| 18:11.02 | ar3dam | tzafrir_latop, i have some warning. |
| 18:11.13 | ar3dam | no, 4 error. |
| 18:12.47 | mshades | i will be presenting my findings at the next AstriCon |
| 18:12.55 | outtolunc | of? |
| 18:13.40 | mshades | the therapeutic value of * |
| 18:15.31 | outtolunc | that reminds me tech i seen this past week that was explaining to a building manager where he was punching down new phone lines... when a call came in.. it was very hard not to laugh my ass off |
| 18:18.51 | mshades | eating paste...sniffing glue...its all the same to me |
| 18:19.00 | tzafrir_laptop | ar3dam, what warnings? |
| 18:19.24 | tzafrir_laptop | What is the output of: asterisk -rx 'zap show channels' |
| 18:27.51 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 18:33.25 | ar3dam | tzafrir_laptop this is the output ... Chan Extension Context Language MOH Interpret |
| 18:33.26 | ar3dam | <PROTECTED> |
| 18:34.50 | tzafrir_laptop | ar3dam, just that? |
| 18:35.18 | tzafrir_laptop | If so, can you pastebin the output of: cat /proc/zaptel/* |
| 18:37.15 | ar3dam | yes, http://pastebin.com/d461d3dc2 |
| 18:39.57 | *** join/#asterisk safa12 (n=safa@92-238-52-92.cable.ubr16.croy.blueyonder.co.uk) |
| 18:45.37 | *** join/#asterisk eliel (n=eliel@151-202-114-200.fibertel.com.ar) |
| 18:49.40 | UnixDog | anyone out there have a full list of makes amd models of sip/iax phones |
| 18:52.21 | tzafrir_laptop | softphones? |
| 18:52.26 | tzafrir_laptop | hardware phones? |
| 18:52.44 | tzafrir_laptop | ar3dam, hmm... maybe restart asterisk |
| 18:52.55 | Qwell | UnixDog: google, I bet |
| 18:54.12 | [TK]D-Fender | ~softphone |
| 18:54.13 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 18:54.43 | *** join/#asterisk zoid_99 (n=zoid_99@24.96.150.105) |
| 18:54.55 | UnixDog | ~hardphone |
| 18:55.12 | UnixDog | ~sipphone |
| 18:55.36 | Qwell | I would be willing to bet that there are no less than 1,500 different models of SIP phones. |
| 18:55.54 | linuxstb | UnixDog: There are too many to list, but this page tries - http://www.voip-info.org/wiki/view/VOIP+Phones&utm_source=voip-info&utm_medium=pbox&utm_campaign=pbox1&utm_content=VOIPP |
| 18:56.10 | UnixDog | ok |
| 18:57.39 | *** join/#asterisk Assimilate (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 18:58.43 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:58.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:00.59 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 19:02.18 | Qwell | "A dialup hard phone is a hard phone with a built-in modem instead of the Ethernet port. It will connect through the modem via a dialup internet service to a remote VoIP server and is therefore self contained." |
| 19:02.19 | Qwell | wow |
| 19:03.06 | Qwell | what genius thought up that garbage? |
| 19:04.33 | jaytee | so instead of 'ring, ring' "Hello?" it's 'shhhhhhhhhhhhh-screeeeeeee-deee-deee-deeee' "Hello?" |
| 19:05.07 | jaytee | yeah, that's progress :-) |
| 19:05.26 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:10.33 | *** join/#asterisk BBHoss (n=hoss@user-24-214-218-77.knology.net) |
| 19:11.49 | [TK]D-Fender | Qwell: I uses patented ADADADADADADADADADADAD converters for maximal call quality ;) |
| 19:12.50 | [TK]D-Fender | It* |
| 19:18.57 | *** join/#asterisk nicox_ (n=nicox@213-33-17-109.adsl.highway.telekom.at) |
| 19:27.39 | *** join/#asterisk Steve_J-obs (n=Chris123@ip68-104-45-211.lv.lv.cox.net) |
| 19:30.32 | *** join/#asterisk JackEStorm (n=no@ip70-171-83-146.no.no.cox.net) |
| 19:56.08 | *** join/#asterisk linuxmaniac (n=maniac@193.153.218.77) |
| 20:01.11 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 20:01.16 | PakiPenguin | hello :) |
| 20:02.13 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 20:09.53 | *** join/#asterisk linuxmaniac (n=maniac@193.153.218.77) |
| 20:14.23 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 20:17.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:21.46 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 20:23.21 | jaytee | lol, http://cdn.davesdaily.com/pictures/302-fukitol.jpg |
| 20:25.25 | *** join/#asterisk rob0 (n=rob0@tuxaloosa.org) |
| 20:26.06 | rob0 | Any stanaphone users having registration troubles, or is it just me? |
| 20:29.09 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 20:30.44 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 20:31.50 | rob0 | my SIP with another origination provider (ipkall) seems to work, as does SIP termination |
| 20:31.55 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 20:36.33 | [TK]D-Fender | rob0: Do you really need us to state the obvious? |
| 20:37.22 | BBHoss | jaytee: thats great |
| 20:37.58 | jaytee | BBHoss, yeah I got a good chuckle from that :-) |
| 20:41.13 | rob0 | nono I just am suddenly getting registration failures, seems like they're down, but the web site works |
| 20:41.20 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 20:41.35 | [T]ank | anyone here a smoothwall express firewall user? |
| 20:42.42 | [TK]D-Fender | [T]ank: You left last night not having fixed all of your SIP settings |
| 20:43.58 | [T]ank | [TK]D-Fender: lol... sorry about that, my battery died and screwed up my irc login when it dumped. I have it all working.. for whatever reason 10 open rtp ports was not enough. i opened it to 10000 open udp ports. now it works. |
| 20:44.12 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:44.30 | [T]ank | what i am wondering now is if anyone is using the smoothwall sip proxy instead of the 10000 open udp ports |
| 20:44.51 | [T]ank | thing that is difficult is testing it all out when i am at home trying to work on it. |
| 20:44.53 | [TK]D-Fender | [T]ank: Tightwad + inability to follow instructions = failure |
| 20:45.26 | [T]ank | why a tightwad? |
| 20:45.31 | Dovid | jaytee: PRICELESS !!!!!! |
| 20:45.45 | [T]ank | inability to follow instructions i will give you... ;-) |
| 20:46.07 | jaytee | Dovid, :-) |
| 20:46.28 | [T]ank | but are you saying i am a tightwad because I am using smoothwall? |
| 20:46.59 | rob0 | There's a town in Missouri called "Tightwad", they used to have a "Tightwad State Bank", got bought out by United Missouri Bank. |
| 20:47.40 | [TK]D-Fender | [T]ank: Sure, lets lock down to 10 ports and hope everything plays nice... |
| 20:47.59 | [TK]D-Fender | ~rtpparanoid |
| 20:48.04 | [TK]D-Fender | ~rtpparanoia |
| 20:48.16 | [TK]D-Fender | Darn, what was that one drmessano made last night... |
| 20:48.30 | [TK]D-Fender | All about 1 ot 100, doesn't matter, its all the same daemon at risk |
| 20:49.25 | [T]ank | so because I am not clear on how things work i am stingy? I thought that one call used only a couple of ports. figured 10 would cover it. |
| 20:49.33 | [T]ank | sorry to have offended you with my inexperience. |
| 20:52.52 | [TK]D-Fender | [T]ank: No, you just need to charnge your approach |
| 20:53.16 | [TK]D-Fender | [T]ank: When things aren't working you shuold be trying to out-think yourself. REMOVE restrictions and look at EVERYTHING. |
| 20:53.29 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 20:53.31 | [TK]D-Fender | [T]ank: It took you far too long to actually look at the SIP debug even. |
| 20:53.55 | [T]ank | never used sip... didnt think to look. this is my first sip peer |
| 20:53.59 | [TK]D-Fender | [T]ank: and step by step I had to point out every blatant place it was listing a private IP where you should have been using a public IP. |
| 20:54.07 | [T]ank | thus=inexperience. |
| 20:54.25 | [T]ank | if i could find all the answers, I wouldnt be here |
| 20:54.26 | [TK]D-Fender | [T]ank: Well hopefully you can absorb all of this, because it was an all-inclusive experience |
| 20:55.57 | jaytee | ~RTPparanoia |
| 20:56.34 | jaytee | ~RTParanoia |
| 20:56.34 | jbot | rumour has it, rtparanoia is [RTParanoia]: 10000 ports, 1 port, who cares.. it's the same daemon listening! |
| 20:56.51 | jaytee | knew I had it in my logs somewhere :-) |
| 21:00.43 | *** join/#asterisk devhen_ (n=devhen@107.235.sfcn.org) |
| 21:01.12 | [T]ank | so i take it then that no one here is using the sip proxy in smoothwall? |
| 21:02.45 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 21:05.37 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 21:07.04 | Dovid | [T]ank: In the past I have found it helpfull to look at the FW logs to see what it may have been blocking |
| 21:12.49 | [TK]D-Fender | Dovid: Even more helpful to provide advise to people who are HERE ;) |
| 21:13.22 | Dovid | TK :) didnt see the quit |
| 21:13.45 | rob0 | He still has spies here. |
| 21:17.45 | jaytee | spies? |
| 21:17.57 | rob0 | everywhere! |
| 21:22.59 | LemensTS | shhh |
| 21:25.51 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 21:35.27 | Katty | afternoon. |
| 21:36.04 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.124.56) |
| 21:41.40 | *** join/#asterisk bsaxon (n=bryantsa@83.sub-75-249-181.myvzw.com) |
| 21:43.19 | outtolunc | afternoon |
| 21:43.52 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 21:44.54 | *** join/#asterisk BBHoss_ (n=hoss@user-24-214-218-77.knology.net) |
| 21:49.03 | errr | hi Katty |
| 21:51.57 | *** join/#asterisk Assimilate (n=Assimila@24-116-182-58.cpe.cableone.net) |
| 21:55.01 | *** join/#asterisk AMUG (n=junky@modemcable156.137-20-96.mc.videotron.ca) |
| 21:55.35 | *** join/#asterisk Freman (i=twitsrus@121.91.0.2) |
| 21:55.51 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 21:55.59 | Freman | Greetings folks. Quick question: What phones are we recommending today? |
| 21:56.19 | Freman | Situation: Small motel, want one largeish phone for the main office, but small simple phones for the rooms |
| 21:58.17 | [TK]D-Fender | Freman: Polycom IP 650 + Attendant modules for reception. SIP Gateway + analog phones for the rooms |
| 21:59.23 | unafilliated | Freman : You can even use wookie toolkies :P |
| 21:59.54 | Freman | Hmmm, was aiming for having a spare ethernet port per room :) |
| 21:59.55 | Freman | they've got an existing proprietry system that's let them down dismally in flexability and ability. |
| 22:00.45 | russellb | check out some aastra phones, too, then ... |
| 22:00.48 | russellb | they have nice simple SIP phones |
| 22:01.07 | russellb | it will be more expensive than the analog phone route, though |
| 22:01.28 | Freman | it'll probably be worth the expense |
| 22:01.45 | Freman | the thing I've likd about the polycoms we installed here is the centralized deployment |
| 22:02.17 | russellb | or the polycom IP 320 for the rooms |
| 22:02.43 | Freman | I also have to pick up 2 analog lines (one for incomming, one for fax, both will be used as failsafe out) |
| 22:03.06 | russellb | polycom phones <3 |
| 22:05.05 | Freman | [TK]D-Fender: So something like the Quadro FXO huh? |
| 22:05.57 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 22:06.08 | [TK]D-Fender | Freman: Wire them for Cat5E, but use it for analog |
| 22:06.31 | [TK]D-Fender | Freman: there is NO point in putting a SIP phon in a room like that. |
| 22:06.56 | [TK]D-Fender | Freman: It will only confuse simple customers and introduces a ton of problems for no good reason |
| 22:07.52 | [TK]D-Fender | Freman: Next thing you know they ask for a wake-up call and accidentally hit DND on the phone and it gets rejected. Or they forward their phone around and crazy shit starts happening. |
| 22:08.56 | Freman | you use astrisk to restrict that :) |
| 22:08.56 | Freman | but yes, I undrstand what you're saying |
| 22:10.51 | [TK]D-Fender | Freman: nice idea, but too much that can go wrong and only leads to confusion. All pain, no gain. |
| 22:11.04 | Freman | Spoze the next question is, is it worth going for something like the Asterisk Appliance - say the S844i or just building one... |
| 22:11.22 | [TK]D-Fender | Freman: Build a real system with real redundency |
| 22:11.44 | [TK]D-Fender | Freman: Do not bottlenexk yourself with a toaster |
| 22:12.53 | Freman | actually, that's kinda what I was planning to do anyway :) - there's no money in this for me, at least until they call and ask for help with something... |
| 22:12.58 | *** join/#asterisk doolph (n=doolph@201.218.241.66) |
| 22:13.00 | doolph | hello |
| 22:13.51 | Freman | an alix board with a tdn400b or 410p... or something... |
| 22:14.17 | Freman | build an image to run on a flash disk. |
| 22:14.19 | doolph | I got a problem with pickups, I can pickup an extension if it is an local call, but if the call comes from a FXO port, I cannot pickup it, -- Executing [**101@user:1] NoOp("SIP/104-b7416898", "Attempt to Pickup 101 by 104") in new stack, but it is not working |
| 22:14.45 | *** join/#asterisk blq (i=Bl@dslb-088-066-226-133.pools.arcor-ip.net) |
| 22:16.48 | *** join/#asterisk riddlebox (n=james@75.132.225.75) |
| 22:18.54 | raz | hum.. is there a way to setup asterisk so that the two participants of a call can type in numbers via dtmf to trigger a sound to be played to both parties? (kind of audio emoticons) |
| 22:20.11 | Freman | [TK]D-Fender: What about something like the Aastra 57I for reception? :) |
| 22:21.22 | errr | Freman: thats a great phone, we have 5 or 6 of them |
| 22:21.57 | errr | and around 120 of the 55i's |
| 22:22.51 | Freman | Heh, not putting $323 phones in the gurst rooms :) |
| 22:23.35 | Freman | I'm tossing up between the sip gateway + analog, or linksys spa942's for guests (they actually work out cheaper then the sipgateway + analogs) |
| 22:25.14 | errr | Freman: I didnt read the history of the conversation you were having with [TK]D-Fender, I was just saying they are good phones |
| 22:25.35 | Freman | I know :) |
| 22:25.42 | Freman | I was just saying it for giggles |
| 22:27.00 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 22:29.41 | beek | ~voip |
| 22:29.42 | jbot | rumour has it, voip is Voice over IP |
| 22:29.48 | beek | ~provider |
| 22:30.04 | _ShrikE | ~itsp |
| 22:30.05 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:30.40 | beek | _ShrikE: Thanks -- I couldn't remember that... Teliax seems to be having problems so I'm off to find a backup. |
| 22:32.49 | beek | ~itsplist-us |
| 22:32.50 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 22:34.51 | Freman | Ok, if I order a switch that outputs 48v and supports 802.3af PoE, and the phones I order normally have 5v bricks, but also support 802.3af PoE - do they work with that switch? or smoke with that switch? |
| 22:35.56 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 22:37.12 | LemensTS | freman - it will know to give the proper power to the phone |
| 22:38.13 | Freman | cos looking at the gateway/analog option, it's awfully expensive compared to even full poe/voip option |
| 22:39.05 | LemensTS | i used a linksys rack poe switch and it has worked nicely so far. It has vlans too, didnt use it. But it was the cheapest i could find. |
| 22:39.48 | Freman | Thinking of using linksys SPA942's - offering the second port on the phone as ethernet to the room |
| 22:41.44 | Freman | scratches head - I always get the two mixed up, fxo vs fxs - which one provides power? it's fxo isn't it? |
| 22:42.36 | Strom_C | fxs |
| 22:43.10 | Freman | hehe, then when you go and configure them in asterisk - it's backwards :) |
| 22:43.24 | Strom_C | no, it isn't |
| 22:43.31 | Strom_C | the signaling type is different from the port type |
| 22:50.01 | *** join/#asterisk AMUG (n=junky@modemcable156.137-20-96.mc.videotron.ca) |
| 22:50.32 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 22:54.29 | *** join/#asterisk marc7 (n=marc@S01060018f877b2e0.gv.shawcable.net) |
| 22:55.46 | *** join/#asterisk bsaxon (n=bryantsa@122.sub-75-200-86.myvzw.com) |
| 23:00.44 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 23:02.21 | Freman | LemensTS: So you guys are running a SRW224P? |
| 23:03.37 | Freman | Or perhaps SRW224F4P |
| 23:03.37 | Freman | fails to see the difference... |
| 23:03.50 | Freman | except the price |
| 23:06.16 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 23:09.26 | *** join/#asterisk BBHoss (n=hoss@user-24-214-218-77.knology.net) |
| 23:12.22 | *** join/#asterisk Bananaskin (n=mike@78-105-249-152.zone3.bethere.co.uk) |
| 23:16.58 | *** join/#asterisk ManxPower (n=manxpowe@191.sub-75-249-181.myvzw.com) |
| 23:20.43 | *** join/#asterisk cplx (n=cplx@ettamo.lnk.telstra.net) |
| 23:21.57 | cplx | Registration from '<sip:EXTENSION@SIP HOST>' failed for 'MY EXT IP' - No matching peer found [Jul 28 07:03:55] NOTICE[12266]: chan_sip.c:15049 handle_request_register: --- it looks like my ccme is trying to register directly with them when the trunk should be? |
| 23:23.42 | *** join/#asterisk seanmh (n=johndoe@c-76-113-38-173.hsd1.nm.comcast.net) |
| 23:26.52 | *** join/#asterisk nvez (n=me@unaffiliated/nvez) |
| 23:27.00 | nvez | good evening guys :) |
| 23:28.06 | nvez | i'm a network/server admin of my own "small business" that's quite growing up, i have a few questions before i start messing around. |
| 23:28.56 | nvez | our datacenter and office are on two different servers, office has no protection and just normal bandwidth, however datacenter has full redundancy, etc. would installing asterisk at the datacenter then connecting the office to it a good idea (for redundancy and no failure)? |
| 23:29.06 | nvez | or it would be just a big waste of bandwidth? |
| 23:29.25 | nvez | also, plenty more bandwidth at DC rather than office.. (sorry if these are newb questions!) |
| 23:36.41 | Dovid | nvez: what kind of lines ? |
| 23:37.01 | nvez | Dovid: lines as in where? |
| 23:37.05 | Dovid | if you are going to go with a Voice PRI then have it in your office. Also why not have an ATM with QoS from your DC to your office |
| 23:37.20 | Dovid | well is it for the office or for clients ? |
| 23:37.29 | nvez | no its for me only |
| 23:37.45 | nvez | (aka office, not clients) |
| 23:38.05 | Dovid | then just put in the office. what do u gain by having it in the DC (unless power backup, etc). |
| 23:38.18 | nvez | well, im guessing that if power goes off here (who knows), i can still be contacted |
| 23:39.27 | nvez | and i can take care of QoS at the office (2950T cisco switch for office) so thats no problem |
| 23:39.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:39.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:43.29 | Freman | So I'm left with the descision - do I build a tiny low powered system based around an Alix board (Think via EPIA but not) or do I just build a cheap system (will probably cost the same in the long run) |
| 23:43.33 | *** join/#asterisk BBHoss_ (n=hoss@user-24-214-218-77.knology.net) |
| 23:44.43 | Katty | herro. |
| 23:48.04 | *** join/#asterisk EtherNet (n=ethernet@host243.190-138-34.telecom.net.ar) |
| 23:50.19 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 23:52.54 | lmadsen | Freman: use the low power system -- save the environment! :) |
| 23:54.43 | Freman | blah |
| 23:54.56 | Freman | the +'s of the low power system are outweighed by the -'s |
| 23:55.05 | Freman | biggest one includes upgrade and flexability |
| 23:56.13 | lmadsen | I don't upgrade systems... I build them to do what they need to do, then buy a new system when I outgrow that one |
| 23:56.22 | lmadsen | upgrading systems rarely makes sense to me |
| 23:56.53 | Freman | I concur, but the alix boards are awfully new, I've not used one before. |
| 23:57.27 | Freman | I shy away from the via epia boards because I find them painful to deal with at times |
| 23:59.58 | *** join/#asterisk Victor_Yure (n=aaa@unaffiliated/victoryure/x-837844) |
