| 00:00.13 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-32-84.vic.bigpond.net.au) |
| 00:00.46 | JT | so has the popularity of religious lunacy in politics in the US reduced a little recently, or is it as strong as always? |
| 00:00.51 | De_Mon | [TK]D-Fender what about the various world religious islam vs christianity vs hindu etc, can they all be "right?" |
| 00:01.19 | De_Mon | it isnt making the news as often, does that mean anything? ;p |
| 00:01.27 | Corydon-w | They're all wrong. There is no god. Never was. No heaven nor hell, either. |
| 00:01.51 | mercestes | JT: religious lunacy? |
| 00:01.57 | [TK]D-Fender | De_Mon : actually, if you look at it Judaism, Christianity, and Islam are all of the same branch. Islam viewed Jesus as a great prophet, just not the Son of God. |
| 00:02.04 | JT | mercestes: policy based on religion |
| 00:02.19 | russellb | the religion discussion needs to end right now ... |
| 00:02.20 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 00:02.25 | JT | seems quite prevlant in the US |
| 00:02.27 | *** join/#asterisk jsmith (n=jsmith@000-150-930.area3.spcsdns.net) |
| 00:02.33 | mercestes | you athiests have nothing to complain about. Your just as obtrusive and offensive as the baptists. |
| 00:02.34 | [TK]D-Fender | mercestes : I'm sure the Pagan's would object to out mis-use of the term "lunacy" as it miscredits their cycle ;) |
| 00:02.35 | J4k3 | wow, the f1000g has a massive flaw in its design :/ |
| 00:02.42 | *** mode/#asterisk [+b %mercestes!*@*] by russellb |
| 00:02.47 | *** mode/#asterisk [+b %[TK]D-Fender!*@*] by russellb |
| 00:02.49 | russellb | I mean it. |
| 00:03.10 | De_Mon | whats an f1000g? |
| 00:03.15 | J4k3 | it works acceptably well and all, but then I took it over to a customer's place to show it off, and I couldn't figure out how to stick his wpa-psk password in :P |
| 00:03.23 | J4k3 | ut starcom's vowifi phone |
| 00:03.27 | J4k3 | the cheap-o one. |
| 00:03.33 | De_Mon | Oooo vowifi |
| 00:03.58 | J4k3 | yeah, its works "ok", but I also have a 3600 sqft house with wooden walls, 2 story |
| 00:04.03 | J4k3 | wooden interior walls. |
| 00:04.07 | *** mode/#asterisk [-b %mercestes!*@*] by russellb |
| 00:04.07 | mercestes | j4k3: Yea, there is a spot to put wpa-psk..but..I never got it to work on a Linksys router. |
| 00:04.14 | *** mode/#asterisk [-b %[TK]D-Fender!*@*] by russellb |
| 00:04.27 | J4k3 | mercestes: yeah... serious design flaw. |
| 00:04.33 | mercestes | well..... |
| 00:04.38 | J4k3 | oh well, it'll work around the office. ;) |
| 00:04.40 | mercestes | I never go tmy laptop to take wpa on a linksys router either |
| 00:04.45 | J4k3 | hehe |
| 00:04.54 | mercestes | so my jury is still out on who is at fault. |
| 00:05.02 | De_Mon | mercestes you need a linksys vowifi phone :) |
| 00:05.12 | mercestes | of course I installed linux on both these routers too so I might not even be able to blame Cisco, it coudl be Openwrt. |
| 00:05.28 | *** part/#asterisk MACscr (n=MACScr@adsl-64-149-152-65.dsl.peoril.sbcglobal.net) |
| 00:09.16 | *** join/#asterisk pat_lehem (i=lehem@bzq-82-81-170-196.red.bezeqint.net) |
| 00:10.13 | *** join/#asterisk Kyler (n=Kyler_La@74-132-227-26.dhcp.insightbb.com) |
| 00:11.00 | mercestes | it's so quiet now |
| 00:11.23 | Kyler | I'm trying to use a ClearOne MAX IP with Asterisk. The thing doesn't send authentication data with REGISTER. |
| 00:11.57 | mercestes | gtg..byes |
| 00:12.22 | jsmith | Kyler: That's crazy! |
| 00:14.10 | Kyler | jsmith: and annoying. It has an option to enable authentication but it doesn't seem to work. The Web interface to it is *horrible*. It's a mess of JavaScript and Java. It breaks constantly. |
| 00:15.09 | jsmith | Kyler: REGISTER is one of the basic SIP primitives. If it can't do that right, I wouldn't expect it to do much at all |
| 00:15.14 | diclophis-work | whats the highest quality codec i can use between two asterisk boxes over a SIP channel? |
| 00:15.16 | diclophis-work | u-law? |
| 00:15.33 | Kyler | jsmith: I think it's more of a matter of the stupid configuration tool not getting it right. |
| 00:15.48 | pat_lehem | g729, but you need to pay for it |
| 00:16.01 | jsmith | diclophis-work: G722 would give you higher audio quality, if you're using 1.4 and your endpoints support it |
| 00:16.05 | *** join/#asterisk HushPe (n=HushPe@mail.kamar.co.nz) |
| 00:16.14 | jsmith | pat_lehem: Actually, g729 is compressed, so the audio quality isn't quite as good as ulaw |
| 00:16.21 | diclophis-work | jsmith: yea, i have 1.4 ... how do i know my endpoints support it ? |
| 00:16.42 | jsmith | diclophis-work: Most don't support G722... what endpoints are you using? |
| 00:16.43 | pat_lehem | oh. isn't g722 compressed as well? |
| 00:16.59 | diclophis-work | jsmith: the connection is between two asterisk 1.4 installations |
| 00:17.03 | jsmith | pat_lehem: No, not really... g722 is a wideband codec |
| 00:17.18 | jsmith | diclophis-work: g722! |
| 00:17.23 | diclophis-work | i was thinking ulaw was like... raw standard pstn audio with no compression? |
| 00:17.30 | pat_lehem | oh, ok. |
| 00:17.59 | jsmith | diclophis-work: It is, with 8-bit sampling. G722 is almost the same, but with 16 bit sampling. (Now, I'm simplifying a bit, but that's basically it) |
| 00:18.13 | diclophis-work | oh.. well thats technically better |
| 00:18.19 | diclophis-work | how do i enable it? |
| 00:18.25 | jsmith | diclophis-work: But your audio has to come from somewhere outside of the Asterisk servers, so ulaw is probably best for you |
| 00:18.26 | diclophis-work | i know how the sip.conf allow/disallow works |
| 00:18.35 | jsmith | diclophis-work: Yup, just the same as any other codec |
| 00:18.37 | diclophis-work | jsmith: yea one box has a pri |
| 00:18.49 | diclophis-work | so... its not gonna 'increase' quality is it? |
| 00:18.51 | jsmith | diclophis-work: PRI would be ulaw (or alaw outside of the US) |
| 00:18.56 | jsmith | diclophis-work: Nope. |
| 00:19.05 | diclophis-work | ok, i will leave it as is |
| 00:19.27 | JT | umm |
| 00:19.30 | diclophis-work | i am not 'losing' quality am it? |
| 00:19.32 | diclophis-work | er am i? |
| 00:19.43 | jsmith | diclophis-work: No, not as long as you have adequate bandwidth |
| 00:20.01 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 00:20.04 | JT | jsmith: g.711 Mu law or A law has companding, which is like compression, it gives closer to 16bits of audio dynamic range, using only 8 bits of bandwidth |
| 00:20.07 | HushPe | hi JT would you be able to have a look at my dial plan (it's a small one)... when i dial in on one of my lines I can't dial extensions :( |
| 00:20.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:20.13 | diclophis-work | yea, internal network... and the machines are on the same rack (1ms delay between them) |
| 00:20.26 | JT | HushPe: i suppose, if you pastebin it :P |
| 00:20.31 | jsmith | JT: I'm well aware of that... that's why I said I was simplifying what I said |
| 00:21.30 | HushPe | JT: cheers, I wasn't planning on flooding ;) http://pastie.caboo.se/36607 |
| 00:22.01 | JT | ah ok, well i'm saying there may likely be not that much difference in experienced audio quality from g.711 to g.722 since most are used to companded g.711 |
| 00:22.10 | HushPe | JT: this might help too: http://pastie.caboo.se/36609 |
| 00:22.38 | jsmith | JT: One cool thing about G722 is that it's uses adaptive differential modulation, so that it uses the same bandwidth as g711, but you get more dynamic sound |
| 00:22.59 | jsmith | JT: (G722 is also companded... so it's more like 16-bit ADPCM/G726) |
| 00:23.05 | diclophis-work | right... so the sound is 'warmer' or something |
| 00:23.21 | JT | right |
| 00:23.26 | J4k3 | screw it, gimmie that 44.1k/16bit/stereo streaming FLAC phonecall action :) |
| 00:23.26 | jsmith | Exactly |
| 00:23.45 | jsmith | J4k3: We're saving that for Asterisk 3.0 |
| 00:24.12 | JT | HushPe: just a little thing, could you use pastebin.ca next time? that pastebin site caboo.se is really annoying, it double frames everything with awful iframes so i need to double scroll, it's quite hard to read |
| 00:24.33 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 00:25.34 | HushPe | JT: will do :) |
| 00:26.52 | JT | HushPe: so the problem is when you dial into asterisk from outside over an FXO line? |
| 00:27.16 | HushPe | yeah, the vodafone incoming, the other line is fine |
| 00:27.26 | HushPe | it doesn't seem to pick up the dial done |
| 00:27.28 | HushPe | tone* |
| 00:29.20 | JT | what is vodafone incoming? |
| 00:29.31 | JT | some sort of mobile phone adapter cable? |
| 00:29.47 | HushPe | JT: yeah, it's a GSM box, on a regular cable |
| 00:30.12 | JT | hrm, i've been interested in doing something like that |
| 00:30.26 | HushPe | it picks up you can talk sweet as, and while talking you can dial numbers, but it just doesn't seem to pick them up when it gets the welcome |
| 00:30.53 | HushPe | JT: we have an erricson box, works quite well, but you'll notice i had to pump up the volume a little, as it's a bit quiet |
| 00:31.16 | JT | what doesn't it pick up? |
| 00:31.18 | HushPe | bossman wants to use it to dial out (hence disa), not one else really uses that line |
| 00:31.36 | *** part/#asterisk jsmith (n=jsmith@000-150-930.area3.spcsdns.net) |
| 00:31.57 | HushPe | it goes like this: ring ring, welcome to blah blah, (dial an exten, nothing), wait... ring through rest of dial plan |
| 00:32.35 | HushPe | so when dialing the numbers it's just not doing it's thing, it like it's deaf to them haha |
| 00:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.26 | JT | but you're saying dtmf sometimes works on the gsm line? |
| 00:40.23 | HushPe | once the call is connected and talking on the phone you can dial tones and can hear them |
| 00:40.53 | JT | you can hear them |
| 00:41.05 | JT | but have you ever had any luck with actually detecting the tones? |
| 00:41.15 | HushPe | yeah, but it doesn't pick the tones |
| 00:41.24 | HushPe | yeah, it works great with the landline :) |
| 00:41.31 | JT | you've never got it to work under any circumstance? |
| 00:41.31 | JT | hrm |
| 00:41.47 | HushPe | not with the vodafone :( |
| 00:42.41 | JT | well, gsm cannot reliably carry dtmf in the traffic channel, it would have to be carried in digital messages in the signalling channel, and your phone should reliably recreate those in audio if received |
| 00:43.03 | JT | but maybe gsm doesn't support OOB DTMF messages from the network to the phone |
| 00:43.10 | JT | or your network has disabled it |
| 00:43.12 | JT | or you have a hardware issue |
| 00:43.30 | coppice | A GSM network reliably carries DTMF, though it greatly alters the timing |
| 00:43.58 | JT | through AF? pretty sure that from phone to network there are OOB DTMf indications |
| 00:44.11 | J4k3 | hmm, is there a way to digitally increase the volume to a specific extension? |
| 00:44.22 | HushPe | i can hear it, and it sounds right, just doesn't seem to do the job |
| 00:44.33 | coppice | it is out of band, but that is transparent to the user, apart from the timing being greatly altered |
| 00:44.50 | *** join/#asterisk predder (n=predder@203.220.55.70) |
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| 00:45.22 | JT | HushPe: if you want to much around, put MixMonitor or just Monitor on the extension and save it as .wav, and look at it with some decent audio software to see if you're getting a clean waveform of 2 tones transposed |
| 00:45.53 | JT | coppice: right, so do you know for certain if OOB indications always work from network to phone? |
| 00:46.10 | kuku5 | dtmf issues? |
| 00:46.24 | coppice | carrying DTMF OOB is part of the GSM specs |
| 00:46.27 | JT | seems like something that either isn't supported properly, or is bared by carrier (probably not wanting people running IVRs on gsm phones) |
| 00:46.48 | JT | coppice: definately from phone > network |
| 00:47.04 | kuku5 | hushpe: on a voip link ? |
| 00:47.12 | JT | kuku5: no, gsm |
| 00:48.14 | HushPe | JT: i transferred to a telecom voice menu, it seems to work, but not reliably... sometimes it does it, others it doesn't |
| 00:48.49 | JT | what do you mean a telecom voice menu? |
| 00:49.25 | HushPe | like a press 1 for the world clock, sorry i'm a bit slow today :P |
| 00:49.50 | JT | is this part of asterisk? |
| 00:50.33 | HushPe | no, completely another system using zap to zap |
| 00:50.51 | HushPe | gsm > zap > asterisk > zap > another IVR |
| 00:50.52 | JT | so not really relevant |
| 00:50.54 | coppice | JT: I've never seen anyone block it. It would be stupid to stop IVR access from a phone. the implementations vary somewhat in their timing, though |
| 00:51.17 | JT | the point is that it's looking more like an issue either with you gsm hardware or provider, than the dialplan |
| 00:51.37 | JT | coppice: no no, the gsm phone is RECEIVING the call |
| 00:51.53 | HushPe | JT: i think so, whether is the gsm box, or the cable or something else... |
| 00:51.57 | JT | he has a gsm gateway hooked up to an FXS port |
| 00:52.29 | HushPe | JT: are there any tweaks i could/should try with zapata? |
| 00:52.41 | coppice | JT: oh, that usually works, but I guess some phones or networks might screw that up |
| 00:53.11 | JT | not that i can think ok, but i've never had to tweak dtmf |
| 00:53.30 | JT | HushPe: if you feel like investigating, you could do what i suggested of looking at the audio waveform |
| 00:54.34 | JT | coppice: networks could conceivably not want people running inbound IVRs on cheap gsm plans, shrug |
| 00:54.48 | JT | it's also an uncommon use case scenario |
| 00:55.11 | JT | most people's phones do nothing on receipt of far end originating dtmf |
| 00:55.52 | coppice | looking at the waveform gets you nowhere. use an audio tool that lets you look at the spectrum. you expect 2, and only 2, nice clean spectral lines of simiar amplitude |
| 00:56.07 | HushPe | JT: can i fire that just after answering the phone? |
| 00:56.15 | JT | yes, spectrum analysis would be more straightforward :P |
| 00:56.20 | JT | HushPe: yes |
| 00:56.26 | HushPe | JT: sweet i'll try that |
| 00:57.10 | HushPe | you'll notice the wait on the vodafone incoming, that's because i found it didn't quite realize the phone had been answered (that was the gsm box i think) |
| 00:57.22 | *** join/#asterisk [1]naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 00:57.44 | JT | ah |
| 00:59.15 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 00:59.45 | JT | HushPe: also, with full debug on, like in the full log, it shows when asterisk detects a dtmf digit |
| 00:59.47 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 01:00.05 | JT | HushPe: might be worth having a look at the full log, or enabling it if it isn't already, in logger.conf |
| 01:03.24 | HushPe | dodgy really, it would cut off the welcome message for a brief moment |
| 01:04.36 | JT | what's dodgy? |
| 01:05.22 | *** join/#asterisk tb0301s (n=sowa@brln-4db13075.pool.einsundeins.de) |
| 01:05.33 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-67-79-2-146.sw.biz.rr.com) |
| 01:06.57 | CoffeeIV | I just checked out asterisk 1.2 branch from svn, and when I compiled it I got the error "codec_zap.c:147: error: dereferencing pointer to incomplete type" -- anyone recognize that and know how to fix it ? |
| 01:08.29 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 01:11.31 | *** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net) |
| 01:11.36 | SimoAmi | hi there |
| 01:12.26 | SimoAmi | I have an issue with picking calls on hold. Sometimes, I don't hear the caller, but they can hear me |
| 01:12.41 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 01:13.00 | *** join/#asterisk tb0301s (n=sowa@brln-4db13075.pool.einsundeins.de) |
| 01:13.12 | MRH2 | qwell can you tell me what the time is? ;) |
| 01:13.20 | Qwell | funny you should ask |
| 01:13.31 | Qwell | my gnome-clock still hates me :P |
| 01:13.41 | MRH2 | lol is it almost later today? |
| 01:13.49 | Qwell | ahh |
| 01:14.03 | Qwell | he said "today or tomorrow" a little while ago |
| 01:14.35 | *** part/#asterisk pat_lehem (i=lehem@bzq-82-81-170-196.red.bezeqint.net) |
| 01:14.57 | MRH2 | he who? |
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| 01:15.17 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 01:15.28 | MRH2 | or do you mean him? |
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| 01:16.51 | *** join/#asterisk DavoFrom818 (n=Vito@cpe-76-173-56-41.socal.res.rr.com) |
| 01:16.54 | DavoFrom818 | hi |
| 01:17.08 | Qwell | MRH2: I assume you mean the zaptel stuff? |
| 01:17.13 | MRH2 | yep |
| 01:17.18 | MRH2 | lol |
| 01:17.20 | Qwell | he == kpfleming |
| 01:17.21 | DavoFrom818 | i just downloaded asterisk 1.4 where are the instructions to install it?? |
| 01:17.27 | MRH2 | ah |
| 01:17.33 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-32-84.vic.bigpond.net.au) |
| 01:18.06 | MRH2 | cool |
| 01:18.28 | MRH2 | toodles |
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| 01:20.41 | DavoFrom818 | do i just run the install-sh? |
| 01:21.15 | Carp1 | Can someone please help me get NuFone working? I've been trying for like 3 hours and I'm no where. |
| 01:21.56 | DavoFrom818 | can someone help me install asterisk 1.4???? |
| 01:22.01 | *** join/#asterisk tb0301s (n=sowa@brln-4db13075.pool.einsundeins.de) |
| 01:22.08 | calvinn | vi INSTALL |
| 01:22.22 | DavoFrom818 | what do you mean vi install? |
| 01:22.39 | HushPe | JT: i'm off the phone now, really busy today... do you want a copy of the wav too? they sound really crisp |
| 01:23.25 | JT | i don't have audio software, or even a soundcard on this pc |
| 01:23.57 | DavoFrom818 | calvinn?? |
| 01:24.10 | *** join/#asterisk Flauto (n=zhao@adsl-69-213-131-119.dsl.chcgil.ameritech.net) |
| 01:24.30 | Flauto | i just got 1.4 but musiconhold is not working |
| 01:24.36 | JT | DavoFrom818: vi is a common text editor |
| 01:25.06 | HushPe | JT: looks pretty clean, i.e. 2 definiate tones. |
| 01:25.31 | JT | hmm |
| 01:25.54 | JT | i guess you could check with dtmf tables to see if they're actually the right tones |
| 01:26.35 | HushPe | hmmm |
| 01:26.54 | JT | and check the full log to see if asterisk detects any dtmf |
| 01:27.05 | Carp1 | Can someone please help me get NuFone working? I've been trying for like 3 hours and I'm no where. |
| 01:28.42 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 01:28.42 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 01:40.41 | HushPe | JT: it's seems they're either too quiet, or not quite right |
| 01:40.57 | JT | hmm |
| 01:41.17 | HushPe | i'll try increasing the gain a little more |
| 01:42.24 | Carp1 | I cant get anything working on asterisk, can someone please help me. |
| 01:42.39 | The_DoC^ | anyone have any 66 bridging clips the want part with |
| 01:42.40 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 01:42.41 | HushPe | Carp1: you have to be a little more specific than that |
| 01:42.50 | HushPe | Carp1: does the demo work? |
| 01:43.28 | Carp1 | No |
| 01:43.34 | Carp1 | I am trying to get NuFone registered |
| 01:43.37 | Carp1 | But I'm no where. |
| 01:43.47 | Carp1 | Ive been working on it for about 3 hours |
| 01:45.00 | Carp1 | I finally got a SIP phone connected |
| 01:45.06 | Carp1 | maybe Im not fowarding the right ports. |
| 01:46.06 | HushPe | are you running it behind a firewall>\? |
| 01:46.14 | J4k3 | wow... this f1000g sounds about 100x better with my office AP set to 802.11g-only. |
| 01:46.24 | Carp1 | no firewall |
| 01:46.27 | *** join/#asterisk tb0301s (n=sowa@brln-4db13075.pool.einsundeins.de) |
| 01:46.37 | Carp1 | Its behind a linksys router flashed with DD-WRT |
| 01:46.47 | HushPe | Carp1: are you using NAT |
| 01:47.09 | Carp1 | Im not really familiar with it (ive heard alot about it). But I dont think so. |
| 01:47.43 | JT | if the machine doesn't have a public IP address, it's behind NAT |
| 01:47.44 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 01:47.48 | HushPe | if you share your internet using it you'll probably be using NAT, in which case SIP doesn't do NAT very well at all |
| 01:48.18 | Carp1 | Yeah, it has an internal IP assigned by DHCP |
| 01:48.21 | HushPe | in short SIP is either directly on the internet, or only inside your network, unless you're running a VPN |
| 01:48.27 | Carp1 | Only the router has a public. |
| 01:48.30 | JT | it would have NAT |
| 01:48.36 | Carp1 | This is IAX protocol |
| 01:48.45 | Carp1 | through NuFone.net |
| 01:48.49 | JT | i thought you said SIP phone |
| 01:48.53 | HushPe | hmmm |
| 01:49.04 | JT | well, which bit has the trouble? |
| 01:49.24 | Carp1 | Yeah, I just finally got a SIP softphone connected. |
| 01:49.34 | Carp1 | Now Im trying to get NuFone working. |
| 01:49.36 | Carp1 | Sorry, |
| 01:50.41 | Carp1 | I have port 4569 (IAX port) fowarded to my asterisk box |
| 01:51.50 | JT | udp? |
| 01:51.57 | *** join/#asterisk predder (n=predder@203.220.55.70) |
| 01:51.58 | Carp1 | TCP |
| 01:52.05 | JT | err it's the UDP port |
| 01:52.14 | JT | iax and sip don't use TCP |
| 01:52.26 | JT | it's inappropriate for streaming |
| 01:52.34 | Carp1 | Yeah, I have UDP and TCP fowarded, sorry |
| 01:52.38 | JT | ok |
| 01:52.46 | *** join/#asterisk litage_ (n=nick@203.220.55.70) |
| 01:52.58 | Kyler | JT: Uh...SIP *can* use TCP, right? |
| 01:53.05 | HushPe | JT: i'm suspecting it's not 'crisp' enough (GSM), therefore it just doesn't do the job... :( |
| 01:53.25 | Carp1 | I might have my iax.conf set up wrong too |
| 01:53.36 | JT | Kyler: dunno if anyone uses it, and the RTP stream will always be UDP anyway |
| 01:53.39 | Carp1 | NuFone has shitty suppots unless you have a question about your account |
| 01:53.51 | Deep6 | guys do phones need to show up in sip registry before they can be called.... .they're showing up under sip channels but not registry |
| 01:53.56 | Kyler | JT: I just noticed that my ClearOne MAX IP has an option for TCP SIP transport. |
| 01:54.15 | Kyler | Carp1: If you're dealing with NuFone...well, good luck. |
| 01:54.29 | Carp1 | :( |
| 01:54.32 | JT | HushPe: umm, did you miss the whole conversation about your problem before? you cannot send dtmf reliably over a compressed codec like dtmf, it should be sent out of band |
| 01:54.40 | JT | HushPe: anyway, scroll up a bit |
| 01:54.55 | Deep6 | I'm getting an E630 Reg retries exceeded |
| 01:55.13 | JT | code like GSM i meant |
| 01:55.17 | JT | codec |
| 01:55.20 | JT | gar |
| 01:56.12 | HushPe | JT: bugger, so that means that the GSM box doesn't send it on through probably |
| 01:56.51 | JT | or your network blocks it |
| 01:56.59 | JT | or it's not supported in the gsm spec |
| 01:57.15 | JT | (it's not common for mobiles to receive dtmf and act upon it) |
| 01:57.52 | Carp1 | No ideas for my probelm? |
| 01:58.28 | JT | Carp1: you really need to provide more details, like configuration or error messages received |
| 01:58.55 | HushPe | pb please :) |
| 01:59.17 | Carp1 | no messages at all. |
| 01:59.22 | HushPe | JT: valid point :) |
| 01:59.23 | Carp1 | I dont think its even trying to connect. |
| 01:59.42 | Carp1 | I dont think there is a command to run for iax either |
| 02:00.10 | JT | you should pastebin the CLI output of your call attempts |
| 02:00.12 | HushPe | can you telnet the port? |
| 02:00.14 | *** join/#asterisk jesselang|laptop (n=jesse@64.241.37.140) |
| 02:01.20 | HushPe | (seeing if it's connecting/open) |
| 02:01.46 | Carp1 | i can in a second, I am pastebining my iax.conf |
| 02:02.39 | *** join/#asterisk distatica (n=distatic@unaffiliated/distatica) |
| 02:02.49 | Carp1 | http://pastebin.ca/332546 |
| 02:02.53 | JT | i'm quite interested in what the call attempt looks like in the cli, with the verbosity turned up |
| 02:02.54 | Carp1 | iax.conf |
| 02:03.20 | distatica | I am looking for python modules for asterisk API and there's a number of different options, has anyone tried them and would be able to recommend the best option? |
| 02:03.41 | JT | oh don't tell me you're using freepbx |
| 02:04.11 | Carp1 | on the CLI right now, it says: registered SIP '100' at 192.168.1.106 on port 44567 expries 3600 |
| 02:04.22 | Carp1 | I am using freepbx |
| 02:04.42 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:04.52 | HushPe | JT: i know this is probably a bad idea, but is there a 'tolerance' for the tones? even if i could tweak it just one line |
| 02:05.08 | Deep6 | JT is freepbx no good? |
| 02:05.11 | JT | Carp1: use ; to comment out, not # |
| 02:05.19 | JT | ~freepbx |
| 02:05.21 | jbot | hmm... freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:05.25 | blitzrage | # is only for zaptel.conf and agents.conf |
| 02:05.45 | Carp1 | I would be using Asterisk without the trixbox package if they didnt change from CVS |
| 02:06.04 | blitzrage | CVS is ooooold |
| 02:06.07 | blitzrage | SVN is good |
| 02:06.07 | Carp1 | and when i downloaded a linux version, it didnt have CVS and I really dont know much linux |
| 02:06.08 | JT | and if you can't see why using # to comment is bad, look at your telephone :P |
| 02:06.15 | blitzrage | yum install subversion |
| 02:06.21 | file | </3 yum |
| 02:06.27 | blitzrage | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 |
| 02:06.32 | blitzrage | <3 yum! |
| 02:06.35 | JT | you don't need cvs or svn, just install a proper release, Carp1 |
| 02:06.43 | JT | Carp1: latest 1.2.x is a safe bet |
| 02:06.43 | Carp1 | What should I run for a distro? Fedora? |
| 02:06.50 | blitzrage | Carp1: it doesn't matter |
| 02:06.52 | blitzrage | whatever you know |
| 02:06.56 | Carp1 | I will gladly switch from FreePBX |
| 02:06.57 | JT | it doesn't matter, whatever you're familiar with and have kernel sources to |
| 02:07.12 | Carp1 | I'm not familia with any linux reeally lol |
| 02:07.15 | blitzrage | I like CentOS, other like Debian, tzanger likes Slackware |
| 02:07.26 | JT | i like debian |
| 02:07.35 | JT | so really, it's all up to you |
| 02:07.41 | blitzrage | I can't stand Debian :) so there you go |
| 02:08.03 | JT | well that's just weird |
| 02:08.05 | JT | :P |
| 02:08.14 | file | to each their own |
| 02:08.56 | Carp1 | Does CentOS have yum? |
| 02:09.15 | JT | yes, it's just mainly a repackage of RHEL |
| 02:09.40 | Carp1 | Well, I will get a copy of CentOS and go from there I guessl |
| 02:09.42 | Carp1 | Thanks. |
| 02:10.00 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 02:11.15 | Carp1 | I can run off just the first ISO right? CentOS-4.4-i386-bin1of4.iso |
| 02:12.29 | blitzrage | yes, if you do a min. install |
| 02:12.30 | JT | i'm still curious what happens in the cli when you try to make a call atm |
| 02:12.42 | blitzrage | or get the ServerCD |
| 02:12.57 | JT | can you get netinst cds for centos? |
| 02:13.54 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 02:13.56 | Carp1 | sorry |
| 02:14.00 | Carp1 | nothing happens |
| 02:14.06 | JT | ? |
| 02:14.07 | Carp1 | I am watching the CLI and no movements |
| 02:14.13 | JT | set verbose 10 |
| 02:14.17 | JT | then try again |
| 02:14.46 | Carp1 | is says verbose was 11 now set at 10 |
| 02:14.58 | JT | oh well, what are you calling from? |
| 02:15.06 | Carp1 | X-Lite |
| 02:15.21 | Carp1 | when I open X-Lite, it says registered on the CLI |
| 02:15.26 | JT | clearly it's not even making it to asterisk |
| 02:15.42 | Carp1 | But it registered with asterisk |
| 02:15.49 | JT | try enabling sip debug and seeing if anything comes up when you make a call |
| 02:15.55 | *** join/#asterisk demigod2k (n=joey@cpe-65-29-113-212.twmi.res.rr.com) |
| 02:16.00 | *** join/#asterisk mak0dk (n=fuck@RH-CypressS-CS1-138-118.njit.edu) |
| 02:16.08 | JT | registration doesn't mean it can make successful calls :) |
| 02:17.18 | Carp1 | there is a ton of debug info when I try. |
| 02:17.21 | mak0dk | anyone have SIP firmware 7.4 or 7.5 for Cisco 7940 IP phone? |
| 02:17.33 | mak0dk | im in desperate need |
| 02:17.37 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 02:17.39 | JT | can you pastbin it? |
| 02:17.58 | Carp1 | let me SSH in becuase I dont know any way else of getting the information |
| 02:17.58 | JT | i think we need sip.conf too |
| 02:18.09 | JT | sounds like a sip issue at this stage |
| 02:19.19 | mak0dk | anyone know how to get broadvoice on a cisco 7940 running firmware SIP 8.2? |
| 02:19.28 | mak0dk | or if its even possible |
| 02:20.43 | *** join/#asterisk RB2 (n=none@pool-71-127-218-246.nwrknj.east.verizon.net) |
| 02:21.29 | Carp1 | http://pastebin.ca/332558 |
| 02:22.45 | JT | something wrong in sip.conf i bet |
| 02:23.03 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:23.39 | Carp1 | let me look |
| 02:23.53 | JT | pastebin it too, removing secrets |
| 02:24.04 | Carp1 | ok |
| 02:24.59 | Carp1 | I used FreePBX to add that extension |
| 02:25.06 | Carp1 | it doesnt show up in sip.conf :-\ |
| 02:25.29 | JT | i see |
| 02:25.59 | JT | btw, extensions are added in extensions.conf, sip peers and users go to sip.conf |
| 02:27.28 | Carp1 | yeah |
| 02:27.30 | Carp1 | I meant that |
| 02:27.41 | Carp1 | I have extensions.conf set to ring the softphone |
| 02:27.56 | Carp1 | let me remove info from FreePBX and hard write it in sip.conf |
| 02:27.56 | JT | so there's no entry in sip.conf at all? |
| 02:27.59 | Carp1 | no |
| 02:30.38 | JT | i guess that could be an issue :) |
| 02:31.08 | JT | in the sip debug, the main message that rang alarm bells was the "407 proxy authentication required" one |
| 02:33.10 | Carp1 | hmm |
| 02:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:33.39 | Carp1 | let me pastebin my extensions.conf, iax.conf you already have, and my sip.conf |
| 02:33.44 | Flauto | <PROTECTED> |
| 02:33.44 | Flauto | [Jan 29 20:33:07] WARNING[27619]: file.c:553 ast_openstream_full: File /var/lib/asterisk/moh-native/10 does not exist in any format |
| 02:33.44 | Flauto | [Jan 29 20:33:07] WARNING[27619]: res_musiconhold.c:243 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh-native/10': No such file or directory |
| 02:33.53 | Flauto | i have a problem with 1.4 |
| 02:34.00 | Flauto | musiconhold does not work |
| 02:35.46 | Carp1 | http://pastebin.com/870861 |
| 02:35.50 | Carp1 | sip.conf |
| 02:36.47 | Carp1 | http://pastebin.com/870863 |
| 02:36.50 | Carp1 | extentions |
| 02:37.58 | *** join/#asterisk bkw__ (n=brian@ip68-0-120-100.tu.ok.cox.net) |
| 02:42.43 | aptura | Flauto, what is the issue |
| 02:44.03 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-130.mtco.com) |
| 02:44.22 | distatica | I'm currently reading the O'Reilly book on Asterisk and am just a little confused over something... can I use a normal modem to connect to PSTN? Or do I need some special hardware? |
| 02:44.32 | distatica | Just for a hobby system |
| 02:44.50 | distatica | 1 phone line. |
| 02:45.25 | demigod2k | I personally use the special equipment but that's for work |
| 02:45.33 | demigod2k | I've heard its possible with the right drivers |
| 02:45.52 | DrkShdw | distatica: you need hardware that is compatible with the zaptel drivers |
| 02:45.58 | distatica | crap |
| 02:46.39 | DrkShdw | for a single phone line, test system. a $20 x100p clone is sufficient. if you plan on actually *using* it, I'd consider a TDM4xx with 1 FXO module. |
| 02:46.57 | DrkShdw | the x100p is a piece of crap. it is useful for nothing more than learning. ;) |
| 02:47.00 | aptura | Flauto, check that res_musiconhold.so is loading in modules.conf |
| 02:47.21 | distatica | DrkShdw, I just want to be able to call my box and execute commands, heh |
| 02:47.24 | JT | distatica: no, you can't use a normal modem |
| 02:47.35 | distatica | I don't even know where you would get a x100p clone, heh |
| 02:47.36 | aptura | yea dump the card |
| 02:47.45 | JT | ebay |
| 02:47.45 | aptura | a20 by Sangoma |
| 02:48.05 | CoffeeIV | I just checked out asterisk 1.2 branch from svn, and when I compiled it I got the error "codec_zap.c:147: error: dereferencing pointer to incomplete type" -- anyone recognize that and know how to fix it ? |
| 02:48.15 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 02:48.18 | JT | all the x100p clones available now are not made by digium, and the chipset is no longer made by manufacturer, so they're often factory seconds etc |
| 02:48.40 | DrkShdw | the x100p's never were made by digium. |
| 02:48.52 | JT | i know, but at least they were badged by them |
| 02:49.00 | JT | now they use factory second silicon |
| 02:49.25 | aptura | DrkShdw what if any digium cards dont have the echo or tx/rx issues? |
| 02:49.39 | aptura | when you want one fxo/fxs :) |
| 02:50.04 | DrkShdw | distatica: any winmodem with the ambient md3200 chipset would work. but again, only use it for testing :) |
| 02:50.13 | Carp1 | no idea's JT? |
| 02:50.53 | JT | sorry just got lunch |
| 02:50.56 | JT | will check in a sec |
| 02:51.06 | Carp1 | Thanks. |
| 02:51.15 | Carp1 | Lunch? its almost 10PM here lol. |
| 02:51.40 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 02:51.53 | JT | it's almost 2pm here |
| 02:51.54 | Bobthehunter | if you start 67 a pri pstn and dial i get this |
| 02:52.01 | Bobthehunter | Jan 29 21:46:12 WARNING[3060505520]: chan_sip.c:6609 build_rpid: Unsupported callingpres (-1) |
| 02:52.04 | JT | on tuesday |
| 02:52.05 | Bobthehunter | anyway to fix this ? |
| 02:52.11 | aptura | man sangoma is such a nice card. So flexible so scalable. |
| 02:52.11 | Bobthehunter | i gues astrisk has no fixup();) |
| 02:52.18 | Bobthehunter | sangoma rules yes |
| 02:52.24 | Bobthehunter | playing with a A102D |
| 02:52.31 | Bobthehunter | superb qual and echo can |
| 02:52.32 | Flauto | aptura, how can i check that? |
| 02:54.35 | JT | Carp1: wow, pastebin.com is so incredibly slow |
| 02:55.17 | aptura | Bob, need to ask you a question but also need to leave whats your email? |
| 02:55.27 | Carp1 | i know |
| 02:55.28 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 02:55.44 | Carp1 | shit |
| 02:55.49 | Carp1 | I meant to use pastebin.ca |
| 02:55.54 | Carp1 | because .com was going so slow |
| 02:56.20 | JT | i only recommend using pastebin.ca :) |
| 02:56.25 | CrashHD | hey fellas |
| 02:56.28 | Carp1 | lol |
| 02:56.29 | CrashHD | you guys seeing a lot of "asterisk RTCP SR transmission error" |
| 02:56.31 | CrashHD | on 1.4? |
| 02:56.43 | CrashHD | I actually had a dropped call or two |
| 02:58.50 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 03:01.18 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 03:02.06 | *** join/#asterisk X-Rob_ (n=Rob@ppp214-210.static.internode.on.net) |
| 03:05.43 | distatica | Is there something that's a 'next model up' from the X100P's? I found some really cheap one's on ebay, but a lot of google results show them as being beyond crap. Someone on a forum recommended a Linksys SPA3102 but they're like $100+ which is a ridiculous for my purposes. |
| 03:06.00 | Flauto | aptura, it does show the musiconhold module should be loaded |
| 03:08.06 | JT | distatica: an SPA3102 is an excellent candidate |
| 03:08.19 | JT | it's a very good price |
| 03:08.42 | JT | distatica: prices only go up from there really |
| 03:08.52 | distatica | ouch |
| 03:08.54 | JT | there's a few other SPA models if you don't need all the features of the 3102 |
| 03:09.07 | JT | proper telephony hardware is not cheap |
| 03:09.18 | distatica | and the only reason a normal PCI modem won't work is because it can't transmit voice and recieve voice at the same time? |
| 03:09.26 | JT | the SPA3102 has 1 FXS port and one FXO port |
| 03:09.32 | JT | the fact there's no driver |
| 03:09.43 | JT | and they're not really designed for it |
| 03:09.47 | distatica | oh yeah, that makes things difficult too, lol |
| 03:09.48 | distatica | ok] |
| 03:10.19 | JT | what are your requirements? |
| 03:10.49 | Carp1 | yeah |
| 03:10.53 | Carp1 | whoops, wrong window |
| 03:11.28 | distatica | I want to be able to call my landline from another phone and be able to either a) leave a voicemail, or b) enter the system and using tones just select some stuff, like check the status of my server, get weather info, whatever. Nothing serious |
| 03:11.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 03:11.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:11.41 | distatica | just playing around, which is why $100 when i'm broke is a little much... |
| 03:12.02 | JT | i guess being broke is the issue |
| 03:12.06 | distatica | also I want people to be able to find out where I am, so I can just set my status as 'off to store' or whatever. |
| 03:12.09 | distatica | most certainly |
| 03:12.24 | distatica | it's always the issue |
| 03:12.33 | JT | setting status like that makes it easier for theives |
| 03:13.12 | J4k3 | damn right |
| 03:13.16 | J4k3 | who wants to get robbed while they're at home? |
| 03:13.20 | J4k3 | thats dangerous! |
| 03:13.34 | distatica | providing they know how to access the menu |
| 03:13.52 | JT | well i assume you want just random people to know where you are |
| 03:14.11 | distatica | or hack my computer, but if they're that desperate to steal my cheap bastard computer equipment they must need it bad |
| 03:14.21 | distatica | I wonder what drug dealers give for a 300MHz machine these days.... |
| 03:14.37 | distatica | JT, either that or people I know ;) |
| 03:14.39 | J4k3 | one rock two rock three rock four |
| 03:14.43 | J4k3 | five rock six rock seven rock more. |
| 03:14.56 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 03:15.45 | JT | you can buy an X100P, just don't expect it to work |
| 03:16.02 | distatica | hrm |
| 03:16.41 | JunK-Y | jt: the new x100p sounds pretty cool. |
| 03:17.02 | JT | what new x100p? |
| 03:17.41 | JunK-Y | http://www.x100p.com/products/FXO.php |
| 03:19.59 | JT | i still call bullshit on x100p.com |
| 03:20.09 | JT | they keep calling themselves genuine and authentic |
| 03:20.14 | JT | yet they aren't digium |
| 03:20.34 | JunK-Y | dont ya think the hardware looks good quality? |
| 03:20.44 | distatica | Just a thought this most likely won't work with asterisk but you folks understand telephony so maybe you'd know. What if I set something up with my sound card and parallel port to trigger pickup/hangup of a normal phone? Could a sound card understand the touch tone's coming from a phone? Or is it out of their frequency abilities? |
| 03:20.53 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:20.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:20.54 | Carp1 | JT, should I just intall CentOS and download asterisk packages? |
| 03:26.03 | dendrite | distatica: It's certainly not beyond the frequency range of a sound card. |
| 03:26.27 | distatica | hmmm |
| 03:26.39 | JT | Carp1: if you want, i'e never used centos |
| 03:26.42 | distatica | maybe that would be a more plausible solution for what i'm trying to accomplish |
| 03:26.46 | dendrite | distatica: You mean drive a POTS handset with a sound card? |
| 03:27.20 | dendrite | distatica: You'll need to power the handset, and if you want to ring the phone, that would be... tricky. |
| 03:27.20 | JunK-Y | JT: we ordered one of that new card to take a look at the hardware, we'll see how good it is. |
| 03:27.29 | JT | distatica: there is an asterisk channel driver for alsa, but it ties you to using the asterisk console in most situations |
| 03:27.34 | distatica | If by POTS you meant to say "P.O.S" phone fromthe dollar store |
| 03:27.52 | distatica | hmmm |
| 03:27.56 | JT | Carp1: that entry in sip.conf, was it already there or did you just add it? |
| 03:28.11 | matt_ | i had some issues earlier today where calls that came from a IAX connection and went out to a sip connection didn't hangup when the calling party hung the phone up, does anybody know why ? |
| 03:28.13 | dendrite | distatica: So you want to use a sound card as a FXS interface??? |
| 03:28.39 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:28.43 | Bobthehunter | chan_sip.c:6609 build_rpid: Unsupported callingpres (-1) |
| 03:28.52 | Bobthehunter | any idea on this ? its from a *67 zap channel |
| 03:28.57 | distatica | hehe, one sec, looking that up |
| 03:29.15 | JT | ~fxofxs |
| 03:29.16 | jbot | fxofxs is, like, An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 03:29.17 | aptura | Bob, what model of Sangoma do you use and have any sound volume issues? |
| 03:29.18 | dendrite | distatica: You'll need a few acronyms to survive in telephony. |
| 03:29.31 | distatica | i'm working on it, heh |
| 03:29.33 | JT | foreign exchange office, foreign exchange station |
| 03:30.28 | distatica | sorry folks, i have a young son (1.5 years) who is climbing my leg in a mad attempt to destroy my data, so i'm sort of distracted at the same time, trying to look this all up |
| 03:30.53 | matt_ | lol :) |
| 03:31.14 | dendrite | distatica: Take your time, the kid is more important than the channel! |
| 03:31.25 | distatica | hence why i'm broker than a "Family Circus" joke. |
| 03:31.37 | distatica | more broke rather, heh |
| 03:31.43 | distatica | me likes english |
| 03:31.46 | JT | JunK-Y: i guess it'll be interesting to see, but i don't see how it can be that good considering the specific intel winmodem chipset has been discontinued |
| 03:32.15 | dendrite | distatica: How much is your time worth? There are some not-too-expensive ways to attach phones to computers... |
| 03:33.03 | distatica | dendrite, i'm a stay at home dad, and have very little cash, but love mucking around with things I don't fully understand, my time is worth about $0.20 / hour. |
| 03:33.19 | dendrite | distatica: Ah, then hack away! :-) |
| 03:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:33.30 | JT | distatica: you may wish to check out the book |
| 03:33.32 | JT | ~thebook |
| 03:33.34 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:33.38 | Carp1 | and is there a go to work wife involved? |
| 03:33.50 | distatica | Carp1, yeah |
| 03:34.07 | Carp1 | Lonestar - Mr Mom |
| 03:34.15 | Carp1 | That song now makes me think of you lol. |
| 03:34.21 | distatica | JT: That's the book i'm reading |
| 03:34.31 | JT | alright |
| 03:34.31 | distatica | And it's what sparked the "can I use a normal modem" question |
| 03:34.40 | JT | distatica: you know where the wiki is too? |
| 03:34.42 | distatica | because he was talking about the X100P and a couple other interfaces |
| 03:34.47 | distatica | the voip-info one? |
| 03:34.47 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 03:35.29 | JT | yes |
| 03:36.21 | matt_ | i have a X100P card lying around |
| 03:36.22 | distatica | I have a friend who works for a voip company, and his wife pointed me to that site. |
| 03:36.28 | Corydon76-home | distatica: the problem with most modems is that they are half duplex devices. A telephone needs to be full duplex |
| 03:36.34 | matt_ | tried to use it with an old pbx but it didn't seem to work :( |
| 03:36.49 | distatica | Corydon-w, so they can transmit and recieve voice at the same time right? |
| 03:36.52 | matt_ | so i just ended up using a supra box |
| 03:37.04 | Corydon76-home | distatica: unless you don't mind only being able to send sound in one direction at a time. ;-) |
| 03:37.17 | distatica | Corydon-w, to be honest that fits my needs perfectly |
| 03:37.32 | distatica | voice mail is only one direction.. and everything else is touch tones |
| 03:37.40 | distatica | with the computer responding with voice |
| 03:37.40 | Corydon76-home | Well, the interfaces in 1.0 did that. They were discontinued in 1.2 |
| 03:38.24 | JT | distatica: how will tones work if the hardware can't monitor it while making audio? |
| 03:38.35 | Corydon76-home | The guy who came up with the original X100P drivers now sells them on ebay for $17 |
| 03:38.57 | distatica | I figured it would just spit out the menu options and then switch to recieve mode |
| 03:39.08 | distatica | of course I couldn't interupt the menu options, whcih would suck |
| 03:39.34 | distatica | but when you're poor, you do everything the annoying way |
| 03:39.53 | distatica | of course the lack of drivers will make life much more difficult |
| 03:39.56 | Corydon76-home | Right, which is why I'm suggesting the $17 way |
| 03:40.15 | JT | is he x100p.com as well? |
| 03:40.30 | Corydon76-home | Probably |
| 03:40.37 | Corydon76-home | Martin P |
| 03:40.55 | distatica | Corydon-w, you mean hack the drivers to work for a PCI modem? |
| 03:40.56 | JT | where would he be sourcing the cards from now? surely they're not custom made |
| 03:41.44 | Corydon76-home | distatica: No, I mean you should get a working board, just so you know what it can do |
| 03:42.32 | Corydon76-home | It's all well and good to spend 60 hours on something, then get something that might partially work, but unitl you have a board that works, you won't know what all to expect |
| 03:42.35 | distatica | oh yeah, but I can't afford something more than $20 (so I can basically afford the X100P) but everywhere that i've read about that card says it sucks |
| 03:42.45 | distatica | very true |
| 03:42.56 | Corydon76-home | Well, it's entry level |
| 03:43.07 | Corydon76-home | It's not terrible, but it's not great either |
| 03:43.18 | Corydon76-home | Half duplex would be terrible |
| 03:43.31 | distatica | so crap for any serious telephony hobbyist, but should work for my purposes? |
| 03:43.51 | Corydon76-home | Crap for production use, but fine for a hobbyist |
| 03:43.52 | *** join/#asterisk chiang_sg (i=kodok@bb121-6-190-224.singnet.com.sg) |
| 03:44.04 | Corydon76-home | I happen to use an X100P in my home system |
| 03:44.27 | Qwell | crap for hobbyist too, but it does the job...sorta |
| 03:45.08 | mog | its good deal for 10 bucks |
| 03:45.24 | Qwell | not anymore... they're selling for $20+ on ebay these days |
| 03:45.35 | mog | yeesh |
| 03:45.35 | distatica | you can get one for <$10 |
| 03:45.41 | *** join/#asterisk Mavvie (n=edwin@ppp0-250.lns1.syd7.internode.on.net) |
| 03:45.55 | Corydon76-home | I don't get callerid at all, but that's not a huge concern |
| 03:46.36 | Qwell | distatica: plus shipping |
| 03:46.58 | Qwell | they used to be like $8 with like $5 shipping |
| 03:47.07 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 03:47.25 | distatica | oh yeah, forgot about that |
| 03:47.37 | Qwell | some of the ones now need a custom driver too |
| 03:47.54 | mog | meh |
| 03:48.05 | distatica | Corydon76-home, why don't you get Caller ID? Is your card supposed to support it but just doesn't work? |
| 03:48.10 | distatica | http://cgi.ebay.com/Real-X100P-FXO-card-oem-for-Digium-asterisk-voip-pbx_W0QQitemZ170073432055QQihZ007QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 03:48.22 | distatica | that one says it has CID but I don't know, heh |
| 03:48.24 | Corydon76-home | distatica: it works if you have nothing else listening on the line |
| 03:48.29 | distatica | ahh |
| 03:49.02 | Corydon76-home | Once you drop the REN by putting a bunch of phones on the line, the X100P is unable to retrieve callerid reliably |
| 03:49.24 | distatica | oh |
| 03:49.29 | Corydon76-home | If I really wanted to fix that, I could do some rewiring, but it's just not worth it to me |
| 03:49.31 | distatica | that should be ok for me though.. |
| 03:49.35 | distatica | I see |
| 03:53.48 | distatica | Well this is some good news at least, I got all excited at the idea of setting this system up and felt sort of crushed when I found out the X100P was crap, renewed hope now, heh. Thanks a lot folks :) |
| 03:54.05 | J4k3 | my "X101P" works pretty good |
| 03:54.13 | J4k3 | its just a resource pig |
| 03:57.46 | Carp1 | why do I not see SVN info on the asterisk download page? |
| 04:00.01 | Bobthehunter | expire -1 = means its expired ? |
| 04:00.10 | Bobthehunter | how can i make them last for 1 month |
| 04:00.15 | Bobthehunter | i hate this crap |
| 04:00.28 | Bobthehunter | 2 minutes is just tooo low |
| 04:01.02 | JT | 1 month? why bother registering at all then? |
| 04:01.12 | *** join/#asterisk shodan (n=shodan@ip119.96-113-216.pppoe1.joliette.intermonde.net) |
| 04:01.16 | Bobthehunter | coz i dont know where they from |
| 04:01.21 | Bobthehunter | and they can still change meanwhile |
| 04:01.29 | Bobthehunter | but u want last known to last 1 month |
| 04:01.30 | Bobthehunter | ;) |
| 04:01.44 | Bobthehunter | can that be done ? |
| 04:02.06 | JT | don't think so |
| 04:02.27 | Bobthehunter | hmm what sets the expire then ? |
| 04:02.58 | *** join/#asterisk orlock (i=[5iKzGtw@202.44.174.4.static.nexnet.net.au) |
| 04:03.03 | Carp1 | why do I not see SVN info on the asterisk download page? |
| 04:03.27 | JT | because the normal way to download, is to download a release |
| 04:03.31 | JT | why do you need svn? |
| 04:04.00 | Qwell | Carp1: that's a good question... |
| 04:04.04 | Bobthehunter | where do asterisk get its expire value ? |
| 04:04.18 | JT | sip.conf |
| 04:04.25 | orlock | Bobthehunter: there is a default unless specified iirc |
| 04:04.32 | Qwell | Carp1: http://www.asterisk.org/developers/get-source |
| 04:05.04 | orlock | I have two SIP phones set up, one via asterisk, the other connects directly to the sip provider. the one going via asterisk is getting echos |
| 04:05.06 | orlock | any suggestions? |
| 04:06.07 | Carp1 | Sorry... |
| 04:06.13 | Carp1 | Becuase I dont know much linux |
| 04:06.25 | Carp1 | and I really am unsure how to install the tar.gz |
| 04:06.41 | distatica | you don't install tar.gz, heh |
| 04:06.45 | distatica | it's a compressed file, like .zip |
| 04:07.00 | distatica | tar -zxf filename.tar.gz will decompress to a directory called 'filename' |
| 04:07.10 | Carp1 | Well, I mean once its uncompressed, I dont know how to install. |
| 04:07.25 | Carp1 | Right, then I enter that directory |
| 04:07.29 | Carp1 | and enter ../install |
| 04:07.30 | Carp1 | ? |
| 04:07.31 | distatica | is it source? If so it's 'make' and then as root 'make install' |
| 04:07.32 | distatica | oh |
| 04:07.42 | distatica | sorry, I don't have the source here so it's difficult for me to check |
| 04:07.56 | JT | you use wget to download the .tar.gz |
| 04:07.59 | Carp1 | I will download the tar.gz if its not hard |
| 04:08.04 | Carp1 | Ok, I will. |
| 04:08.10 | JT | then something like tar -zxvf |
| 04:08.11 | distatica | Carp1, shouldn't be, but sometimes it can be daunting |
| 04:08.20 | Carp1 | Lol. |
| 04:08.23 | JT | Carp1: it's an essential skill to learn |
| 04:08.24 | Carp1 | I need to learn linux. |
| 04:08.33 | distatica | if you have problems though i'm sure someone here will help you, or as a last resort join #linuxhelp they're pretty good |
| 04:08.49 | distatica | JT, he's entering the directory so I assume he has decompressed the file |
| 04:08.55 | distatica | it's the compilation he doesn't know how to do |
| 04:09.11 | Carp1 | Exactally. |
| 04:09.24 | JT | Carp1: you need to read the documentation in each tarball |
| 04:09.27 | distatica | and I don't know if it's straight 'make; su -c 'make install' or if there's a config script to run first |
| 04:09.28 | JT | either INSTALL or README |
| 04:09.34 | distatica | Carp1, yeah, nevermind what JT said |
| 04:10.14 | chiang_sg | i got 2 asterisk server, all phones are connected via SIP. how do i share both dialplan so that phone registered at one server can call others in different server? |
| 04:10.48 | JT | chiang_sg: easy way to do it would be to have different extension prefixes for each server |
| 04:11.20 | chiang_sg | like 8xxxx for server 1, and 7xxxx for server 2 ? |
| 04:11.40 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:11.44 | JT | yes |
| 04:12.04 | chiang_sg | i'll try to propose that to my client. Thanks JT :) |
| 04:12.32 | JT | there are more complicated ways |
| 04:12.38 | JT | like dundi, enum |
| 04:12.45 | JT | realtime (sql database) |
| 04:12.48 | Nukemizer | must digium cards be configured by hand in zaptel.conf ? or has there been a board configuration utility created for assignment of PRI and clear channel T1's ? |
| 04:12.54 | [TK]D-Fender | chiang_sg : http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 04:13.02 | JT | or very complicated dialplans with lots of extensions in each |
| 04:13.08 | erickperez | Hi, I need some assistance with FXS interfaces being connected to an avaya system.... |
| 04:13.15 | chiang_sg | [TK]D-Fender, thanks, i give it a shot :) |
| 04:13.25 | chiang_sg | JT, i'll googling more about dundi :) |
| 04:13.29 | JT | Nukemizer: usually you need to configure it yourself to get it to work |
| 04:13.36 | [TK]D-Fender | chiang_sg : Forget dundi for your scenario |
| 04:13.53 | JT | chiang_sg: 7XXXX & 8XXXX is by far the easiest |
| 04:14.30 | erickperez | Nukemizer: I haven't seen any GUI that helps you configure digital lines... |
| 04:14.55 | chiang_sg | JT, yes it is, but right now they don't want to add more digit to the extension number |
| 04:15.32 | JT | chiang_sg: can't you just split up the existing extension numbers? |
| 04:15.34 | Nukemizer | Thank you both.. I just switch ports in our test lab from PRI to a lot and did not want to keep doing it by hand. I suppose a smart guy maight create mulitple .conf files ;) |
| 04:15.41 | erickperez | is anyone familiar with busytone detection with an avaya system? My TDM400P/ASterisk seems unable to detect busy tone from the avaya. |
| 04:16.14 | JT | busy tone detection over analogue can be dodgy |
| 04:16.16 | *** join/#asterisk ShadowTech (n=jerespet@c-66-176-202-207.hsd1.fl.comcast.net) |
| 04:16.31 | chiang_sg | jt, i have 2203xxx in 1 server and all others number in the other server. for 2203xxx i can put a patern there, but for others i have no idea how should i create the patern |
| 04:16.56 | JT | i'm sure it's possible |
| 04:17.17 | erickperez | JT: The asterisk uses fxo to talk to the avaya and reach PSTN. When pstn goes onhook the avaya goes onhook too but asterisk keeps a busy signal for like 20 seconds and then hang up. |
| 04:17.17 | JT | you'll have to describe the extension numbers further though |
| 04:18.16 | chiang_sg | JT, I will try to do it :) i got 250 extensions in the other server. hope i can get the pattern :) |
| 04:18.42 | JT | chiang_sg: what is the number range? |
| 04:19.55 | chiang_sg | 2203xxxx (server a), for server B there is alot (2201xx, 2202xx, 7xx, 4xx, 8xx) |
| 04:20.12 | chiang_sg | i think, i will just put wildcard |
| 04:20.24 | JT | well that still seems easy |
| 04:20.34 | *** join/#asterisk JerJer[mobile] (n=jj@66.194.234.154) |
| 04:20.47 | chiang_sg | thanks jt, you open my eyes :) i'll try it now |
| 04:20.49 | JT | _220[12]__ |
| 04:20.58 | JT | _[478]XX |
| 04:21.02 | chiang_sg | ow.... |
| 04:21.44 | JT | _220[12]XX i meant |
| 04:22.39 | chiang_sg | jt, one more thing, instead using exten => _7XXX,2,Dial(SIP/myserver:passwordA@SIPserverA/${EXTEN:1},30,r) is that possible to hide username and password ? |
| 04:23.10 | JT | yes, you should add an entry in sip.conf |
| 04:23.15 | JT | then it's SIP/entry |
| 04:23.21 | chiang_sg | mm type = peer ? |
| 04:23.39 | JT | either both a peer and user entry, or friend |
| 04:23.44 | JT | if it needs to go both ways |
| 04:24.15 | chiang_sg | thanks JT |
| 04:26.55 | *** part/#asterisk DavoFrom818 (n=Vito@cpe-76-173-56-41.socal.res.rr.com) |
| 04:28.49 | SomeOne1 | JT!!!! |
| 04:28.53 | SomeOne1 | howya doin buddy |
| 04:28.56 | SomeOne1 | hows the good ol outback |
| 04:29.09 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 04:29.23 | JT | the outback is pretty far away |
| 04:29.27 | JT | hey |
| 04:29.30 | orlock | the dropbears and dingoes are having an all out war |
| 04:29.36 | JT | haha |
| 04:29.49 | SomeOne1 | do you have a really cool accent? |
| 04:30.13 | JT | i can put on whatever accent you want |
| 04:30.47 | JT | my normal accent is not as redneck sounding as say, steve irwin, though |
| 04:32.23 | J4k3 | crikey! |
| 04:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 04:33.24 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 04:33.47 | orlock | hey rob |
| 04:34.42 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:40.56 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.94) |
| 04:46.48 | bkw__ | Hey everyone call 7128723344 |
| 04:48.18 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 04:48.19 | JT | why |
| 04:48.31 | JerJer[mobile] | he gets recip :) |
| 04:51.43 | JT | JerJer[mobile]: ? |
| 04:51.43 | Carp1 | hey |
| 04:51.50 | Carp1 | Whats the test file called? |
| 04:52.03 | Carp1 | I want to be able to call my box and have it run the success test |
| 04:52.21 | Carp1 | JT, I am now getting calls from NuFone. |
| 04:52.31 | Carp1 | I installed CentOS and got asterisk running smooth |
| 04:52.49 | JT | ah cool |
| 04:52.55 | JT | so does it all work? |
| 04:53.49 | Carp1 | It says it rejects the call in CLI becuase it cant reach s@ |
| 04:53.59 | Carp1 | which I ammsume because there is nothing in the extensions.conf file. |
| 04:54.08 | JerJer[mobile] | smells like you are using sip |
| 04:54.48 | JT | Carp1: rofl yeah |
| 04:55.04 | JT | Carp1: installing the sample configs might be helpful |
| 04:55.11 | JT | so you can at least read the comments |
| 04:55.20 | Carp1 | Yeah I have them |
| 04:55.23 | Carp1 | I'm not using SIP |
| 04:55.40 | Carp1 | IAX from NuFone (ps. thanks for the email back nufone) |
| 04:56.02 | JT | you need an s extension in the default context for the sip entry |
| 04:56.08 | Carp1 | I know |
| 04:56.11 | JerJer[mobile] | odd it should call whatever exten you have configured in the number detail section |
| 04:56.21 | Carp1 | I have nothing configured |
| 04:56.25 | Carp1 | only a connection to NuFone |
| 04:56.41 | JerJer[mobile] | that might be your problem :D |
| 04:56.41 | Carp1 | Whats the name of the file that says someth9ing about the monkey |
| 04:56.42 | Carp1 | lol |
| 04:57.06 | JerJer[mobile] | its one of the tt-* files |
| 04:57.27 | InHisName | tt-monkeys.[wav,WAV,gsm] |
| 04:58.01 | Carp1 | I can do exten=>s,1,Playback(tt-monkeys) |
| 04:58.10 | Carp1 | exten=>s,2,Hangup() |
| 04:58.11 | Carp1 | ? |
| 04:59.11 | JerJer[mobile] | can you ? |
| 04:59.31 | JerJer[mobile] | might wana answer and wait a tid so the first word won't get cliped |
| 05:00.37 | Carp1 | ok |
| 05:00.44 | Carp1 | I am still getting rejected for some reason |
| 05:01.02 | JerJer[mobile] | got a [NuFone] type=user ? |
| 05:01.08 | JerJer[mobile] | with a proper secret and context ? |
| 05:01.10 | JerJer[mobile] | no host |
| 05:01.24 | Carp1 | yws |
| 05:01.26 | Carp1 | yes* |
| 05:02.40 | Carp1 | chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach 's@' |
| 05:05.28 | Carp1 | maybe I need to foward more ports? |
| 05:05.40 | Carp1 | I have 4569 fowarded and thats it. |
| 05:05.52 | J4k3 | wow |
| 05:06.02 | J4k3 | is it common for grandstream budgetone 101's to lock up regularly? :P |
| 05:06.22 | Carp1 | yes |
| 05:06.26 | Carp1 | I had one like 3 years ago |
| 05:06.31 | Carp1 | and it froze like twice a day. |
| 05:06.33 | J4k3 | thats... sad |
| 05:06.39 | J4k3 | this ones frozen twice in the last 5 minutes |
| 05:06.44 | Carp1 | lol. |
| 05:06.51 | J4k3 | haha |
| 05:06.53 | Carp1 | Thats why they're only like $50 |
| 05:06.54 | Strom_C | the answer to "Is it common for Grandstream [name of product] to [behave unreliably]" is always "yes" :) |
| 05:06.54 | J4k3 | good thing they're lightweight |
| 05:07.01 | J4k3 | cuz this pos is going back to where it came from |
| 05:07.02 | J4k3 | hehe |
| 05:07.36 | [TK]D-Fender | GrandSuck is to be avoided with extreme prejudice.... |
| 05:07.52 | Carp1 | Any ideas to my problem? |
| 05:09.15 | J4k3 | how/why do they stay in business? |
| 05:09.42 | Carp1 | becuase people buy their product |
| 05:09.43 | Carp1 | lol |
| 05:09.48 | [TK]D-Fender | J4k3 : YOU bought one didnt you? :0 look in the mirror. |
| 05:10.00 | J4k3 | yeah, but I can unbuy it easily. |
| 05:10.00 | J4k3 | ;) |
| 05:10.17 | Carp1 | maybe i'm getting this error becuase I have nothing in [general] in extensions.conf? |
| 05:10.44 | [TK]D-Fender | The world is full of cheap people who care more about how much they spend instead of how much they get. |
| 05:11.08 | J4k3 | well, whats sad is the phone otherwise seems decent |
| 05:11.44 | [TK]D-Fender | J4k3 : Yeah its great.... except for the SUCKING part... |
| 05:11.50 | JT | rofl |
| 05:13.45 | Carp1 | AHH! |
| 05:14.01 | J4k3 | well, it seems no less physically pathetic than the AT&T 4-line analog phones we've been using for about 10 years. |
| 05:14.12 | J4k3 | and no more physically pathetic, either. |
| 05:14.29 | J4k3 | except for the whole sucking part. |
| 05:14.48 | [TK]D-Fender | J4k3 : Shit looks very good when compared to crap.... |
| 05:16.30 | J4k3 | hehe |
| 05:16.58 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 05:17.10 | [TK]D-Fender | J4k3 : Euthanasia. |
| 05:17.14 | J4k3 | things usually have a reason to lock up... the power supply looks like a potential source of shittyness. |
| 05:17.20 | k-man_ | whats THE book called again? |
| 05:17.24 | Strom_C | ~book |
| 05:17.37 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:17.37 | Strom_C | ~thebook |
| 05:17.39 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:17.39 | Strom_C | stupid bot |
| 05:17.51 | k-man_ | thanks stoffell |
| 05:17.54 | k-man_ | oops |
| 05:17.58 | k-man_ | i mean Strom_C |
| 05:18.11 | k-man_ | stupid tab completion |
| 05:18.19 | Strom_C | hehe |
| 05:18.21 | k-man_ | it really annoys me how tab completion can't read my mind |
| 05:18.23 | [TK]D-Fender | k-man_ : perhpas you should use more than 1 character :) |
| 05:18.26 | Strom_C | you're welcome k<tab> |
| 05:18.40 | [TK]D-Fender | k-man_ : it can... its jsut not sure what to do with all the STATIC ;) |
| 05:18.54 | k-man_ | [TK]D-Fender, actually, i did, but i misread Strom_C's nick as storm<something> |
| 05:19.15 | k-man_ | [TK]D-Fender, hey! that was uncalled for.... i prefer to think of it as pink noise |
| 05:19.51 | k-man_ | so should i buy the book? or wait |
| 05:19.59 | Strom_C | download it for FREEEEEEEEEEEEEEE |
| 05:20.03 | k-man_ | is there an upgrade price for the book when it comes out? |
| 05:20.04 | [TK]D-Fender | k-man_ : Yeah, the first hit is on the house... after that you have to pay ;) |
| 05:20.12 | k-man_ | oh.. i can't stand reading that much book in a pdf |
| 05:20.29 | *** join/#asterisk hi365_m (n=hi365_m@ti100710a081-2850.bb.online.no) |
| 05:21.29 | JT | there's a telco here offering PRIs for AUD$20/chan/month |
| 05:21.37 | JT | so AUD$200/mo for 10 channels |
| 05:21.43 | Strom_C | wow |
| 05:21.44 | Qwell | "One channel please" |
| 05:21.44 | [TK]D-Fender | k-man_ : ream of paper = 4$. 400 pages @ 1.2c / p (max reasonable printing cost) < 10$ printed. |
| 05:21.50 | Strom_C | qwell: haha yes |
| 05:21.52 | JT | that's about USD$160 |
| 05:21.56 | JT | minimum 10 :) |
| 05:21.57 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 05:22.01 | JT | 10, 20, 30 |
| 05:22.03 | k-man_ | [TK]D-Fender, you didn't factor in binding |
| 05:22.07 | k-man_ | and trimming |
| 05:22.25 | JT | it's almost tempting to get one, just to do *something* with it |
| 05:22.26 | k-man_ | [TK]D-Fender, and all the effort i have to go to to get that done |
| 05:22.55 | k-man_ | jt, who is offering that btw? |
| 05:22.59 | JT | optus |
| 05:23.04 | k-man_ | oh |
| 05:23.08 | JT | not all areas are covered though |
| 05:23.11 | k-man_ | oh |
| 05:23.13 | k-man_ | i see |
| 05:23.28 | JT | they are also rolling out their own analogue local exchange equipment within telstra exchanges |
| 05:23.36 | JT | $20/mo is the rate for everything, right now |
| 05:23.40 | JT | digital or analogue |
| 05:23.45 | JT | they don't do BRI though |
| 05:23.49 | k-man_ | cool |
| 05:23.52 | [TK]D-Fender | k-man_ 10$ including a binder... |
| 05:24.04 | JT | k-man_: you in .au? |
| 05:24.06 | k-man_ | [TK]D-Fender, and labour cost? |
| 05:24.12 | k-man_ | jt, yeah, i am |
| 05:24.17 | J4k3 | JT: I used to get PRIs for about $250 after tax, but it required paying to be in the colo room, and the smallest rental was a full rack @ $750/mo |
| 05:24.23 | JerJer[mobile] | GRRR you do not need to port forward |
| 05:24.26 | JT | k-man_: cool |
| 05:24.28 | k-man_ | jt, when i move my businss ill be looking to go isdn and voip i think |
| 05:24.28 | J4k3 | T1 PRIs, so 23 channels |
| 05:24.32 | [TK]D-Fender | k-man_ : Ctrl-P in Acrobat reader != difficult. them 5 mins hole punching while watching DivX movies is imperceptable :) |
| 05:24.52 | JerJer[mobile] | Carp1: you obviously have something misconfigured |
| 05:24.58 | k-man_ | [TK]D-Fender, why can't i just buy the book? |
| 05:24.59 | JT | k-man_: depending on number of employees, this pri deal is pretty hard to not look at |
| 05:25.21 | k-man_ | jt, currently we have 4 lines (pots) wich is enough for now |
| 05:25.24 | Carp1 | Everything I see online says this is uaually becuase of a typo |
| 05:25.26 | Carp1 | but I dont see one |
| 05:25.30 | k-man_ | i doubt we will need more for another 6 months to 1 year |
| 05:26.04 | JT | k-man_: yeah, i wonder how much they cost per month though |
| 05:26.21 | JT | it's worth paying a slight premium to go digital, even if you don't need more channels |
| 05:26.22 | k-man_ | jt, how much what costs? |
| 05:26.26 | JT | for business |
| 05:26.29 | JT | your pots lines |
| 05:26.37 | k-man_ | jt, yeah and you save a lot on the instalation cost of the lines |
| 05:27.13 | k-man_ | jt, i did some somes a while ago, and (at the time i hadn;t heard of these new optus rates) the savings for 4 lines were quiet small and would not be worth it because of the disconnect/reconnect fees |
| 05:27.20 | file | Strom_C: Heute Ist Mein Tag! |
| 05:27.34 | Strom_C | arrest my dog? |
| 05:27.38 | k-man_ | but if you are having new lines connected anyway, it was very worth it... so when moving premesis will be a good time for us to convert |
| 05:27.54 | Carp1 | hmm....I enabled debug and it says No Authority Found |
| 05:27.58 | JT | k-man_: right, but from an asterisk perspective, analogue is just a damn pain :) |
| 05:28.06 | JT | so definately worth looking at |
| 05:28.28 | k-man_ | jt, oh... right... well, i have not made the switch yet... |
| 05:28.34 | k-man_ | to asterisk i mean |
| 05:28.35 | J4k3 | and cancel all but the 1st pots line in the rotary |
| 05:28.37 | J4k3 | problem solved. |
| 05:28.41 | k-man_ | we still have an NEC Xen |
| 05:29.10 | orlock | k-man_: NEC PABX? |
| 05:29.18 | orlock | you should be able to get a SIP interface for it |
| 05:29.40 | JT | k-man_: analogue doesn't have the same signalling supervision as digital, so if you want to do anything more complicated then a basic system, asterisk may have some difficulties |
| 05:29.42 | k-man_ | orlock, no, its not a pabx, its what they call a key switched system... and its a slightly older model and cannot do proper sip |
| 05:29.52 | orlock | ahh |
| 05:29.54 | JT | also you can't set outbound callerid on analogue |
| 05:29.59 | file | Carp1: pastebin iax.conf without passwords, and an iax2 debug of an attempt |
| 05:30.04 | orlock | k-man_: my voip provider is NEC basically. |
| 05:30.19 | k-man_ | jt, yes, i plan to go to asterisk only when we move and get isdn and / or voip |
| 05:30.24 | orlock | k-man_: they run an ISP here, as well as their DSL research center and stuff, is cool |
| 05:30.30 | JT | k-man_: right |
| 05:31.00 | k-man_ | i'm still trying to work out how to get started with asterisk |
| 05:31.35 | k-man_ | hence wanting a hard copy of that book |
| 05:31.44 | JT | amazon.com :) |
| 05:31.49 | k-man_ | yeah |
| 05:31.56 | k-man_ | i'd like to get it from aus. |
| 05:32.11 | JT | i have no idea why, but okay |
| 05:32.40 | k-man_ | err... its easier and quicker? |
| 05:32.43 | k-man_ | isn't it? |
| 05:32.51 | k-man_ | and cheaper |
| 05:32.52 | JT | quicker, that's it |
| 05:32.57 | JT | way more expensive |
| 05:33.08 | JT | it's like $70 in bookstores here |
| 05:33.14 | JT | it's something like USD$20 on amazon |
| 05:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:33.23 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:33.26 | k-man_ | your joking? |
| 05:33.28 | JT | no |
| 05:33.31 | k-man_ | oh yeah |
| 05:33.33 | k-man_ | cripes |
| 05:33.41 | k-man_ | why do australian vendors rip us off |
| 05:33.42 | JT | retail bookstores are incredibly expensive at times |
| 05:33.43 | k-man_ | *sigh* |
| 05:33.49 | JT | a lot of stuff is cheaper to self import |
| 05:33.52 | JT | no idea |
| 05:35.06 | [TK]D-Fender | k-man_ : FFS just print the damned thing! Don't make things so damned hard on yourself... |
| 05:35.31 | k-man_ | [TK]D-Fender, ok! bloody hell |
| 05:35.39 | [TK]D-Fender | :D |
| 05:35.52 | Carp1 | now i have a problem |
| 05:35.58 | Carp1 | asterisk isnt seeing my call at all now |
| 05:36.31 | [TK]D-Fender | Remember people : the best way to make a point is but usinge a whet stone, and then piercing with the blade horizontal so as not to get jammed in the ribs! |
| 05:36.33 | JT | [TK]D-Fender: you're crazy |
| 05:36.48 | [TK]D-Fender | JT : Flattery will get you everywhere :) |
| 05:36.55 | JT | heh |
| 05:37.22 | k-man_ | 4 hundred pages double sided |
| 05:37.30 | k-man_ | i wonder how long that will take my poor little printer |
| 05:37.33 | [TK]D-Fender | Its, late, and I'm tired, and my typing is turning to rat shit :) |
| 05:37.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:37.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:37.53 | Juggie | russellb!!! |
| 05:37.55 | JT | k-man_: what about all the cut lines, too? |
| 05:38.00 | *** join/#asterisk litage_ (n=nick@203.220.55.70) |
| 05:38.15 | [TK]D-Fender | JT : screw the cut lines. |
| 05:38.16 | JT | k-man_: you know if you actually pay for your printing costs, it's cheaper to buy the book than make it? |
| 05:38.24 | k-man_ | jt, yeah, i'm going to post it to [TK]D-Fender so he can trim it up for me ;) |
| 05:38.53 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 05:38.59 | k-man_ | ok, jt and [TK]D-Fender, please debate if i should print or buy, and then just tell me which i should do |
| 05:39.02 | k-man_ | ;) |
| 05:39.07 | [TK]D-Fender | k-man_ : who cares about a tiny bit of trim around the page? BIG DEAL. Any little excuse to slow you from getting off your ass is it? ;) |
| 05:39.30 | [TK]D-Fender | k-man_ : How'd ya like them apples? ;) |
| 05:39.40 | JT | k-man_: order it from the us and read online in the meantime, but up to you, hey it's your time and money :P |
| 05:40.10 | k-man_ | hehe |
| 05:42.18 | [TK]D-Fender | k-man_ : Don't let not having the book stop you from installing and learning *. its far from the be-all and end all. You'll find you'll use MAYBE half of it if you're lucky. the rest you pick up as you go. |
| 05:42.52 | JT | or you could just bug #asterisk with your problems without reading it at all |
| 05:42.54 | JT | ;) |
| 05:42.55 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-170-132.hsd1.ca.comcast.net) |
| 05:43.02 | k-man_ | [TK]D-Fender, ok... well... i have tried installing it |
| 05:43.08 | k-man_ | and i did install it |
| 05:43.13 | k-man_ | i just dunno what to do now |
| 05:43.15 | k-man_ | next i mean |
| 05:43.30 | [TK]D-Fender | k-man_ : As in..... |
| 05:43.37 | De_Mon | k-man_ if you dont know, buy |
| 05:43.44 | *** join/#asterisk predder (n=predder@203.220.55.70) |
| 05:43.53 | De_Mon | otherwise, books are for people who cant learn by experience |
| 05:43.59 | [TK]D-Fender | k-man_ : What are you expecting, world domination? Something better than sliced bread? Nirvana? |
| 05:44.04 | k-man_ | how do i hook asterisk up to my sip provider and how do i hoop my sip phone up to asterisk |
| 05:44.30 | Hmmhesays | first take off your pants |
| 05:44.37 | Hmmhesays | then light some scented candles |
| 05:44.41 | De_Mon | next, take of Hmmhesays 's pants |
| 05:44.47 | Hmmhesays | um no |
| 05:44.53 | [TK]D-Fender | k-man_ : For all the dozens of guides & articles out there and the fact you could ahve skimmed the book by now, there really is no reason you shouldn't have gotten some phones placing calls, and doing VM, etc... |
| 05:45.02 | De_Mon | un less ack, Hmmhesays isn't wearing any pants! |
| 05:45.07 | [TK]D-Fender | TMI! |
| 05:45.18 | Hmmhesays | its quarter to midnight of course i'm not wearing pants |
| 05:45.22 | k-man_ | i think your missunderstanding me |
| 05:45.35 | k-man_ | i am working on that in parrallel to seeking the book |
| 05:46.05 | Hmmhesays | tech support was better when computers were more cryptic to your average joe |
| 05:46.15 | Carp1 | I found my problem....not sure how to fix it yet |
| 05:47.32 | [TK]D-Fender | k-man_ : And you can read it on your computer jsut fine.... |
| 05:47.59 | Carp1 | On my IAX provider, I can set a destination, but I dont really understand what it means |
| 05:49.26 | [TK]D-Fender | Carp1 : Your previous error should have tipped you off that it didn't have a context to search for "s" in. that should have beena tip-off to set on in your User config, and another for [general] |
| 05:51.42 | Carp1 | I still don't get it. |
| 05:51.48 | Carp1 | I am really new at this |
| 05:52.21 | Carp1 | I set a context in iax.conf |
| 05:52.35 | Carp1 | and in extensions.conf, it has that context with s in it. |
| 05:54.40 | [TK]D-Fender | Carp1 : pastebin those 2 files, masking ONLY passwords. |
| 05:54.41 | [TK]D-Fender | ~pb |
| 05:54.49 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:55.05 | Carp1 | ok just a second |
| 05:56.58 | Carp1 | http://pastebin.ca/332705 |
| 05:57.58 | Carp1 | http://pastebin.ca/332706 |
| 05:59.39 | *** join/#asterisk Ubeguy (n=Vito@cpe-76-173-56-41.socal.res.rr.com) |
| 05:59.42 | Ubeguy | hi |
| 05:59.55 | Ubeguy | anyone here have voiceon up and running? |
| 06:03.12 | [TK]D-Fender | Carp1 : add "context=inbound" to [general]. |
| 06:03.54 | Carp1 | ok |
| 06:04.11 | Carp1 | in extensions.conf? |
| 06:05.05 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 06:05.52 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 06:08.14 | [TK]D-Fender | Carp1 : no, in iax.conf |
| 06:09.12 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 06:10.05 | Carp1 | ok |
| 06:11.00 | Carp1 | Still errors |
| 06:11.19 | Carp1 | What should I set my destination as on the account oage? |
| 06:11.21 | Carp1 | page* |
| 06:12.01 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 06:17.19 | Carp1 | no ideas? |
| 06:17.36 | [TK]D-Fender | account page? |
| 06:18.08 | [TK]D-Fender | have you done "reload" in* CLI? |
| 06:18.23 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
| 06:18.32 | hack1 | hello friends |
| 06:19.02 | hack1 | i am trying to login using http interface as http://192.168.1.1:8088/asterisk/manager?action=login&username=vivek&secret=asdf |
| 06:19.13 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 06:19.17 | hack1 | and it gives me this Invalid/unknown command |
| 06:19.22 | hack1 | does anyone know this |
| 06:19.24 | Carp1 | yes, i reload alot |
| 06:19.37 | Carp1 | i can set the destination on my NuFone page |
| 06:19.49 | *** join/#asterisk andyshack (n=andyshac@reels.iinet.net.au) |
| 06:19.51 | [TK]D-Fender | hack1 : Please read the channel topic |
| 06:19.52 | andyshack | evening. |
| 06:20.17 | hack1 | [TK]D-Fender: which channel topic? some docs or something |
| 06:21.20 | Carp1 | the message that is displayed when you entered this room |
| 06:21.58 | [TK]D-Fender | Carp1 : Run with this for a bit : http://www.voip-info.org/wiki/view/Asterisk+settings+nufone |
| 06:22.22 | [TK]D-Fender | - Check out the new http://www.asterisk.org !!! -=- Asterisk, Asterisk-addons, Zaptel, and Libpri 1.4.0 released!!! (December 23, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 06:22.32 | [TK]D-Fender | means this isn't the place for * GUI support |
| 06:24.20 | *** join/#asterisk osas (n=nnnnnnnn@CABLE-72-53-75-252.cia.com) |
| 06:24.27 | hack1 | ok, i will do it |
| 06:24.29 | hack1 | thanks |
| 06:25.37 | andyshack | im playing with asterisknow. re : asterisk, all the conf files are in /etc/asterisk/ i assume these are the ones im meant to be adjusting ? |
| 06:26.58 | [TK]D-Fender | andyshack : youa ren't typically going to be doing too much in there as the entire point of running a GUI is as a concession to your inability to do so normally. |
| 06:27.13 | andyshack | well the gui is beta and doesnt work |
| 06:27.47 | andyshack | so ive ssh'd in to see if i can make it work manually |
| 06:28.11 | Carp1 | wow, this is driving me nuts |
| 06:28.51 | [TK]D-Fender | andyshack : ..... |
| 06:28.52 | [TK]D-Fender | ~book |
| 06:28.54 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:28.56 | [TK]D-Fender | ~docs |
| 06:28.58 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 06:28.59 | [TK]D-Fender | ~wikis |
| 06:29.00 | jbot | i guess wikis is http://www.voip-info.org |
| 06:30.44 | andyshack | did you get denial a headie last night or something ? |
| 06:31.02 | andyshack | denied even |
| 06:32.11 | [TK]D-Fender | nope, just pointing you to all the places that should tell you what you need to get started. |
| 06:32.11 | Carp1 | I can now make outgoing calls, but I cant get the incoming calls to work |
| 06:32.33 | [TK]D-Fender | Carp1 : turn up the IAX debug and see whats going on.... |
| 06:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:33.39 | Carp1 | I will pastebin it |
| 06:33.40 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:34.35 | [TK]D-Fender | Carp1 : make sure to do "verbose 10" jut to make sure you get everything else too... |
| 06:34.58 | [TK]D-Fender | <PROTECTED> |
| 06:35.42 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 06:35.54 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:36.02 | Carp1 | http://pastebin.ca/332732 |
| 06:36.53 | Carp1 | localhost*CLI> iax2 show registry |
| 06:36.53 | Carp1 | Host dnsmgr Username Perceived Refresh State |
| 06:36.53 | Carp1 | 66.225.202.80:4569 N carpenter 24.92.37.135:4569 60 Registered |
| 06:38.38 | [TK]D-Fender | Carp1 : here's our tip-off : [Jan 30 01:34:35] NOTICE[30753]: chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach '8002081258@' |
| 06:38.50 | Carp1 | Yeah |
| 06:38.56 | Carp1 | I get that without setting debug |
| 06:38.59 | Carp1 | I get it everytijme |
| 06:39.02 | [TK]D-Fender | Carp1 : firs you need an exten to match that #, and second, there is no context mentioned after the @. |
| 06:39.03 | Carp1 | everytime* |
| 06:39.06 | [TK]D-Fender | This is not a good thing. |
| 06:39.42 | Carp1 | my destination is set to my 800 number |
| 06:39.48 | Carp1 | on my nufone account page |
| 06:39.51 | Carp1 | I can change it to anything I want |
| 06:40.10 | Carp1 | whatever I change it to, I will get the same error, .....trying to reach 'WHATEVER@' |
| 06:40.28 | Carp1 | I don't add the @ sign tho |
| 06:40.37 | [TK]D-Fender | Carp1 : Well you need to do exten => 8002081258,1,DoSomethingUsefulHere...... |
| 06:41.12 | Carp1 | I have that now in inbound |
| 06:41.12 | Carp1 | I replaces s with the 800 number |
| 06:41.12 | Carp1 | replaced* |
| 06:41.13 | [TK]D-Fender | Carp1 : no, that jsut the naming method. exte @ context. the fact a context isn't mentioend is concerning... |
| 06:41.20 | *** join/#asterisk plasko (n=plasko@209-33-199-246-dsl.infowest.net) |
| 06:41.26 | [TK]D-Fender | Carp1 : pastebi your new configs again. |
| 06:41.33 | Carp1 | Ok. |
| 06:42.32 | Carp1 | http://pastebin.ca/332734 |
| 06:42.53 | [TK]D-Fender | Carp1 : Remove the @ |
| 06:42.59 | [TK]D-Fender | Carp1 : that is NOT part of the # |
| 06:43.02 | plasko | do you call it "connecting to the trunk" ? |
| 06:43.16 | Carp1 | http://pastebin.ca/332735 |
| 06:43.28 | plasko | the point where your asterisk server talks to the actual phone system? |
| 06:43.29 | Carp1 | Oh yeah, I was just testing that earlier and forgot to remove it |
| 06:44.47 | Carp1 | um? |
| 06:45.11 | Carp1 | where the phone system connects to the server.....connecting to the trunk |
| 06:45.15 | [TK]D-Fender | plasko : somewhat appropriate, but ultimately useless term. |
| 06:46.05 | plasko | [TK]D-Fender: I'm trying to find more info on this connection. Is there a provider of some kind? |
| 06:46.32 | plasko | The way I understand it, I can have my asterisk server all set up, but I need some higher power to connect me into the phone system proper. is this correct? |
| 06:46.34 | [TK]D-Fender | plasko : what KIND of connection? The are MANY ways to get to the PSTN. You need to be more specific |
| 06:46.53 | [TK]D-Fender | plasko : Or you can jsut a card, plug it into your sever,and plug lines in direct. |
| 06:47.34 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
| 06:47.48 | plasko | Eventually I'd want to be able to make simultaneous calls. I used to work for a company that spend big money to be able to make 1000 simultaneous calls. I wouldn't need that many. |
| 06:48.11 | Carp1 | The destination is remote extension to send calls to. |
| 06:48.21 | Carp1 | thats what it says in my NuFone control panel |
| 06:48.30 | Carp1 | that the feild I have set to the 800 number |
| 06:48.40 | [TK]D-Fender | Carp1 : perhaps you should try setting the context in there as well... |
| 06:48.42 | Carp1 | and thats why you see the 800 number with the @ after it |
| 06:48.53 | Carp1 | I dont add the @ sign tho |
| 06:49.06 | Carp1 | so like 800xxxxxxx@inbound |
| 06:49.07 | Carp1 | ? |
| 06:49.16 | [TK]D-Fender | Carp1 : I know, THEY do. so enter something like 800xxxxx@inbound |
| 06:49.18 | [TK]D-Fender | yes |
| 06:49.22 | [TK]D-Fender | give ti a shot |
| 06:49.35 | Carp1 | ok |
| 06:49.37 | Carp1 | I changed it |
| 06:49.37 | [TK]D-Fender | plasko : You still haven't stated what you want to actually DO. |
| 06:49.47 | Carp1 | takes a minute or 2 to change in the system |
| 06:50.12 | plasko | I want to feed 20 phone numbers into the system and send the same call to 20 people. at the same time, or nearly the same time. |
| 06:50.32 | plasko | is basically what I'm looking to accomplish. |
| 06:50.33 | [TK]D-Fender | plasko : What do you mean the SAME call? |
| 06:51.24 | plasko | a recording essentially, with some numbers (text to speech, probably) inserted specifically for each call. |
| 06:52.12 | J4k3 | North Korea (Landline & Mobile)0.2009 |
| 06:52.13 | Carp1 | same error |
| 06:52.22 | plasko | so its sort of like "Dear *JOE* you have *11* contact requests in the last hour." |
| 06:52.31 | Carp1 | ....who was trying to reach '8002081258@inbound' |
| 06:52.36 | [TK]D-Fender | plasko : well you'll have to think about what kind of tech you intend to use for that. Either a PRI, or using a VoIP provider (which typically use PRI's to terminate calls to the PSTN |
| 06:53.21 | [TK]D-Fender | Carp1 : pastebi your extensions.conf, and I trust you did a "reload" recently. |
| 06:53.38 | Carp1 | yes |
| 06:53.45 | [TK]D-Fender | Carp1 : this is my last swing at this tonight.... already way too late... |
| 06:54.02 | plasko | are there any reasonably inexpensive providers for development and testing? |
| 06:54.18 | Carp1 | Ok....Thanks alot for all the hours you helped me tonight |
| 06:55.16 | [TK]D-Fender | plasko : Depends where you are and what your usage will look like. You'll just hav to start researching that yourself. |
| 06:55.29 | [TK]D-Fender | Carp1 : last pastebin please... |
| 06:56.02 | plasko | yep. I will. So their called "PRIs" and "VOIP Providers" that's what I'm looking for right? |
| 06:56.17 | Carp1 | Ok. Just one second. |
| 06:56.34 | plasko | "VOIP provider", I understand. I'm not familiar with the PRI terminology. Thanks a lot. |
| 06:56.57 | [TK]D-Fender | plasko : PRI is a T1/E1 signalling method. |
| 06:57.05 | Carp1 | http://pastebin.ca/332739 |
| 06:57.10 | [TK]D-Fender | plasko : A digital link to the telco |
| 06:57.23 | plasko | gracias |
| 06:57.51 | Carp1 | http://pastebin.ca/332742 |
| 06:58.18 | [TK]D-Fender | Carp1 : Ok, I don't get it right now. keep at it and you'll figure it out |
| 06:58.22 | [TK]D-Fender | GTG for now |
| 06:58.23 | [TK]D-Fender | later all |
| 06:59.51 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 07:01.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:02.08 | J4k3 | hmm, nice, I think it is the power supply locking up the grandstream phone. |
| 07:02.20 | J4k3 | it happens when my all-electric heat pump system kicks on |
| 07:02.24 | J4k3 | which is a nasty little brownout |
| 07:03.25 | Carp1 | WHY WILL THIS NOT WORK! |
| 07:04.35 | *** join/#asterisk chiang_sg (i=kodok@bb121-6-190-224.singnet.com.sg) |
| 07:05.03 | chiang_sg | hi. is it allow to have callerid="Ko Ben <228888>" |
| 07:05.03 | chiang_sg | inside sip.conf ? |
| 07:07.06 | *** join/#asterisk shinux__ (n=shinux@196.220.28.81) |
| 07:08.26 | chiang_sg | because i got this error " No closing quote found in " |
| 07:09.15 | Carp1 | Yes |
| 07:09.21 | Carp1 | like this: |
| 07:09.41 | Carp1 | callerid="Ko Ben" <228888> |
| 07:09.57 | Carp1 | callerid=("Ko Ben" <228888>) |
| 07:10.01 | Carp1 | no, the first one |
| 07:10.04 | Carp1 | lo |
| 07:10.18 | chiang_sg | so my "ko ben <228888>" is wrong |
| 07:10.30 | Carp1 | move the last quotation after ben |
| 07:10.32 | Carp1 | and it should work |
| 07:10.38 | chiang_sg | oke let me try |
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| 07:18.37 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:24.05 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 07:26.27 | Carp1 | anyone available to help me connect to nuFone? Ive been trying for almost 9 houts |
| 07:26.27 | Carp1 | hours |
| 07:26.47 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 07:26.47 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
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| 07:27.20 | Carp1 | anyone available to help me connect to nuFone? Ive been trying for almost 9 houts |
| 07:27.22 | Carp1 | hours |
| 07:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 07:34.22 | *** part/#asterisk s1gny|wrk (n=s1gny@p54916945.dip.t-dialin.net) |
| 07:36.52 | Chris-NB | Carp1, what do you need? probably I can help u. |
| 07:38.02 | Chris-NB | anyone had the problem: when defining multible ISDN ports in one group and calling vial Dila(Zap/g1/....) it is not possible if the first port is disconnected |
| 07:38.09 | Carp1 | Hey Chris! |
| 07:38.13 | Chris-NB | is there a way to bypass that problem? |
| 07:38.16 | Chris-NB | jup? |
| 07:38.22 | Chris-NB | what are u trying? |
| 07:38.29 | Chris-NB | nuFone, is that a provider? |
| 07:38.43 | Carp1 | Yes |
| 07:38.50 | Carp1 | I will PM you the email I just sent |
| 07:38.55 | Chris-NB | ok |
| 07:39.35 | Carp1 | I've been working all night on this and have no idea. |
| 07:41.17 | chiang_sg | hi, when i tried to call zap, i got "No translator path from g723 to alaw" why zap using g723? |
| 07:42.10 | JT | call zap from what |
| 07:42.45 | chiang_sg | sip |
| 07:42.49 | chiang_sg | all the sip is ulaw and alaw |
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| 07:45.35 | JT | hrm |
| 07:45.46 | Ubeguy | anyone here got voiceone to work? |
| 07:46.01 | Carp1 | JT: this is still not working :( |
| 07:48.22 | JT | hrm |
| 07:48.34 | JT | remind me in about an hour when i get home, i might be able to take a look |
| 07:49.40 | Carp1 | lol |
| 07:49.40 | Carp1 | ok |
| 07:49.46 | Carp1 | That will be 3AM for me |
| 07:49.52 | Carp1 | hopefully I cna stay up |
| 07:50.56 | JT | hah |
| 07:51.04 | JT | what's the url again then? :P |
| 07:51.27 | chiang_sg | 3 AM?here 3 PM :) |
| 07:51.47 | Carp1 | :) |
| 07:51.48 | Carp1 | Here are my config files: |
| 07:51.48 | Carp1 | http://pastebin.ca/332742 |
| 07:51.48 | Carp1 | http://pastebin.ca/332739 |
| 07:52.51 | JT | what the problem again? |
| 07:53.08 | Carp1 | PMing you |
| 07:53.10 | Carp1 | the email I sent |
| 07:53.25 | chiang_sg | i still can't figure out why zap using g723 |
| 07:53.52 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 07:54.47 | yidiyuehan | hello everybody, can anyone tell me how to use Ztmonitor for the asterisk? i run ztmonitor 1 -vvv and it shows Unable to open /dev/dsp: No such file or directory |
| 07:55.25 | JT | yidiyuehan: you need a soundcard setup |
| 07:56.23 | yidiyuehan | i have the zaptel card there |
| 07:56.32 | yidiyuehan | but just don't know how to load ztmonitor. |
| 07:56.42 | chiang_sg | yidiyuehan: tried ztcfg ? |
| 07:56.49 | yidiyuehan | ztcfg is fine |
| 07:57.14 | yidiyuehan | ztcfg -vvvvvvv shows 1 channel configured. |
| 07:57.36 | yidiyuehan | zttool shows the status is ok. |
| 07:57.48 | JT | yidiyuehan: this has got nothing to do with your zaptel card. |
| 07:57.56 | yidiyuehan | i go to /usr/src/zap/ and run ./ztmonitor 1 -vvv, still got the same error |
| 07:58.03 | JT | yidiyuehan: you must have a sound card setup and operating on /dev/dsp |
| 07:58.09 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:58.10 | JT | /dev/dsp is the audio device |
| 07:58.19 | JT | setup of audio is outside the scope of asteriskl |
| 07:58.21 | JT | -l |
| 07:58.27 | yidiyuehan | JT, now i understand your meaning. ;-) |
| 07:58.45 | yidiyuehan | you mean in order to run ztmonitor for zaptel card, i must have the sound card installed? |
| 07:58.57 | JT | well yes, because the whole point is to monitor it |
| 07:58.58 | yidiyuehan | is it necessary or i can work around it |
| 07:59.07 | JT | what do you need ztmonitor for? |
| 07:59.12 | yidiyuehan | ok. thanks man, i will figure out the way to do it. |
| 07:59.36 | yidiyuehan | to adjust my rx and tx gain |
| 07:59.59 | JT | hmm ok |
| 08:00.13 | Carp1 | JT, you got my PM right.......sometimes I think the server rejects big messages |
| 08:00.18 | yidiyuehan | as i want to run fax as well voice for the server, but i just blindly put Rxgain=9, and Txgain =-4. |
| 08:00.28 | chiang_sg | wooow |
| 08:00.37 | chiang_sg | just put 1.0 is enough |
| 08:00.45 | yidiyuehan | 1 for which one? |
| 08:00.51 | chiang_sg | 9 is over i think |
| 08:00.59 | chiang_sg | for both |
| 08:01.15 | chiang_sg | did you read the wiki? then you put 9 and -4 ? |
| 08:01.16 | yidiyuehan | as i read the articles , for FAX, should put higher for Rxgain otherwise it cannot detect the fax tone |
| 08:01.27 | yidiyuehan | yes correct. i just some guys there |
| 08:01.43 | JT | Carp1: i got it |
| 08:01.44 | chiang_sg | yidiyuehan, there is "for example" |
| 08:01.47 | Carp1 | Ok. |
| 08:01.58 | chiang_sg | so no need to follow it :) |
| 08:02.01 | yidiyuehan | yes, chiang_sg ;-) |
| 08:02.04 | *** join/#asterisk antlers (n=antlers@ip70-189-187-211.lv.lv.cox.net) |
| 08:02.14 | yidiyuehan | and by the way chiang_sg , you are in singapore as well right? |
| 08:02.19 | chiang_sg | yea |
| 08:02.28 | JT | Carp1: do calls not work both ways? |
| 08:02.36 | yidiyuehan | great, singapore use Loopstart or ? |
| 08:02.36 | chiang_sg | love this "modern" island |
| 08:02.38 | Carp1 | outgoing work |
| 08:02.40 | Carp1 | incoming error |
| 08:02.49 | chiang_sg | my server is not in singapore :) |
| 08:02.54 | chiang_sg | it's in indonesia |
| 08:03.05 | yidiyuehan | well....then do you know something about it? |
| 08:03.21 | chiang_sg | last time, i run genzaptelconf to generate the settings |
| 08:03.22 | yidiyuehan | as i am not sure with it...some articles said singapore only has loopstart |
| 08:03.24 | chiang_sg | it just work like charm |
| 08:03.41 | yidiyuehan | in singapore or indonesia? |
| 08:03.49 | chiang_sg | indonesia |
| 08:04.01 | JT | Carp1: sorry man, got to go home soon |
| 08:04.06 | yidiyuehan | well, ok. ;-) |
| 08:04.12 | Carp1 | thats ok. |
| 08:04.12 | JT | Carp1: try doing an iax debug then make incoming call |
| 08:04.13 | chiang_sg | but last time i got my tdm 400 P with 4 fxo, also work fine here in singapore |
| 08:04.15 | Carp1 | thanks |
| 08:04.17 | JT | and saving to pastebin |
| 08:04.17 | yidiyuehan | and do you actually use the fax functions for the server? |
| 08:04.24 | Carp1 | Ok |
| 08:04.27 | chiang_sg | naah :) just voice trunk :) |
| 08:04.33 | chiang_sg | sorry can't help much :) |
| 08:04.39 | yidiyuehan | no problem brother. |
| 08:04.46 | yidiyuehan | i am using TDM400p. ;-) |
| 08:05.03 | yidiyuehan | and yes for voice i think pretty much no need to adjust rxgain and txgain |
| 08:05.09 | yidiyuehan | but for fax it's a little bit tricky. |
| 08:05.16 | chiang_sg | hehe :) |
| 08:05.21 | yidiyuehan | i have spent 2 weeks on this,.with iaxmodem and hylafax |
| 08:05.31 | JT | Carp1: it sounds quite easy, it just hasn't hit me what the problem is yet :) |
| 08:05.39 | JT | haven't done iax in a little bit |
| 08:05.40 | Carp1 | http://pastebin.ca/332777 |
| 08:05.40 | yidiyuehan | anyhow i will install sound card to test for it.thanks all guys |
| 08:05.54 | Carp1 | Yeah, this sucks lol. |
| 08:05.54 | *** join/#asterisk getsmart (n=getsmart@ip-164-140.sn2.eutelia.it) |
| 08:06.04 | chiang_sg | good luck bro |
| 08:06.36 | *** join/#asterisk UVSoft (n=UVSoft@80.254.48.58) |
| 08:06.38 | UVSoft | hi there |
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| 08:14.55 | [hC] | any of you guys had an experience where sometimes randomly calling out on PRI returns everyone is busy/congested, even though that number is not busy? |
| 08:17.00 | UVSoft | so, i'm writting a driver for an FXS device, and i've got very strange asterisk's behavior when i hangup (two phones, two FXS devices, call from one to another, pick up receiver, hang up receiver), so the driver gets from the device ONHOOK signal and sends it to zaptel (ZT_RXSIG_ONHOOK), than in several seconds zaptel sends me ZT_TXSIG_KEWL, and than the DIAL application finishes with non-zero status!!! than ZT_TXSIG_ONHOOK, so i can do nothing but hang u |
| 08:17.00 | UVSoft | p (caller phone), but I want to be able to do something according to the dial plan. why can it be so? |
| 08:19.28 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:21.47 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 08:25.34 | tzafrir | UVSoft, do you work with RBS or not? |
| 08:30.11 | UVSoft | tzafrir: ) i dont know what it is |
| 08:32.28 | tzafrir | what version of Asterisk? What do you get in DIALSTATUS ? |
| 08:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:33.35 | UVSoft | so if an application exits with non-zero result, the dialplan isn't executed any more, and i cant look an DIALSTATUS |
| 08:33.45 | tzafrir | zaptel should send you the KEWL signal at the end of the call. this is a hint for you to indicate to the remote FXO device that the call has ended. |
| 08:33.59 | tzafrir | (if you use fxoks) |
| 08:34.25 | tzafrir | change the channel's signalling to fxols and that kewl signalling will probably be gone |
| 08:36.40 | UVSoft | so zaptel sends me that signal, and than DIAL finishes with non-zero, and asterisk doesn't execute any commands of the dialplan |
| 08:37.59 | UVSoft | i'd like to find out why it finishes with non-zero result to be able to go on, to do something according to the dialplan or something |
| 08:39.38 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:39.47 | tzafrir | you wrote above that you pick up the reciever and hang it up again |
| 08:40.07 | tzafrir | A channel was hardly generated |
| 08:40.29 | tzafrir | If you want a channel to be generated immeditely on an FXS channel, set immediate=yes |
| 08:43.36 | [hC] | is it possible to use something like a standard sipura ata to dial a particular extension as soon as the handset is lifted? |
| 08:43.43 | [hC] | ala the bat phone sort of idea? |
| 08:43.50 | [hC] | or a taxi courtesy phone |
| 08:44.03 | [hC] | i know it is with FXS devices, but... SIP? |
| 08:44.53 | UVSoft | tzafrir: i hang it up not so fast |
| 08:45.04 | tzafrir | set verbose 3 |
| 08:45.05 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 08:45.13 | tzafrir | do you see a channel generated? |
| 08:45.32 | tzafrir | 'show channels' will show you where it is in the dialplan |
| 08:46.05 | UVSoft | tzafrir: i even cant talk and hear my voice on the other phone |
| 08:46.29 | tzafrir | fix signalling first. then get the voice working... |
| 08:46.58 | UVSoft | tzafrir: the device is an FXS device, so signelling should be FXO |
| 08:47.07 | UVSoft | tzafrir: and it is |
| 08:47.14 | tzafrir | UVSoft, the FXS is your device, right? |
| 08:47.21 | UVSoft | right |
| 08:47.31 | tzafrir | Has voice ever worked before? |
| 08:47.50 | UVSoft | it works perfect |
| 08:48.07 | UVSoft | but whan i hang up the receiver |
| 08:48.08 | UVSoft | ... |
| 08:51.33 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
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| 09:02.12 | J4k3 | is there any way for a sip provider to know what kind of sip device (ie - asterisk, fxs, voip phone) that you're connecting with? |
| 09:02.25 | J4k3 | other than behaviorial patterns? |
| 09:02.39 | gordonjcp | not really |
| 09:02.45 | gordonjcp | although they can pass back a user agent |
| 09:04.42 | tzafrir | and you can fake a user agent, if you really want |
| 09:12.55 | denon | J4k3: some switches do use more than agent string to ident the client |
| 09:12.56 | denon | and deny |
| 09:13.01 | denon | if applicable |
| 09:13.23 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 09:13.34 | J4k3 | ah |
| 09:14.21 | J4k3 | brb |
| 09:16.01 | x86 | what do you guys think of this: https://voip.shellshark.net/index-whatitis.x86 |
| 09:19.16 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
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| 09:22.04 | nfi|ermes | till now i used Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o with florz's patch, now i d like to update everything at the 1.4 version; is there also zaphfc and florz patch for that ? |
| 09:23.08 | sumasuma | Asterisk Appliance Developer Kit - How many FXO Modules I can have on it ? I don't want to have FXS ! |
| 09:24.34 | sumasuma | Is it possible to use the standard FXO modules used with the TDM400P cards to the Asterisk Appliance Developer Kit ? |
| 09:31.25 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
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| 09:36.00 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
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| 09:46.19 | sbingner | sumasuma, 4 |
| 09:46.29 | *** join/#asterisk chiang_sg (i=kodok@bb121-6-190-224.singnet.com.sg) |
| 09:46.31 | sbingner | and yes |
| 09:46.53 | sumasuma | so can i have 12 fxo ports ? |
| 09:47.23 | sumasuma | i don't need fxs ports |
| 09:48.15 | sbingner | huh |
| 09:48.19 | sbingner | I said 4, not 12 |
| 09:48.25 | sumasuma | ? |
| 09:48.27 | sumasuma | yes |
| 09:48.47 | sumasuma | looks like it can accomodate 12 ports ! (8fxs and 4 fxo ) |
| 09:48.57 | sbingner | no, it can do 4 ports. |
| 09:49.01 | sbingner | fxs or fxo |
| 09:49.24 | sumasuma | One 4- port FXS module & two 4-port FXO modules |
| 09:49.37 | sumasuma | this is from the digum web page |
| 09:49.50 | sbingner | then you need to read it more closely |
| 09:49.56 | sumasuma | http://www.digium.com/en/products/hardware/aadk.php |
| 09:50.02 | sbingner | and pay attention to what type of card those modules are for |
| 09:50.14 | sbingner | oh |
| 09:50.19 | sbingner | appliance developer's kit |
| 09:50.25 | sbingner | missed that part |
| 09:50.35 | sumasuma | oh |
| 09:50.50 | sumasuma | i guess you misunderstood somewhere |
| 09:51.40 | sbingner | I can't say for sure but I don't believe you can modify what's in it when you buy it. If you can open it up and replace a card and you can order said card you may be able to do what you want |
| 09:52.53 | sumasuma | i c |
| 09:54.05 | sbingner | it may very well be the TDM400P modules... somebody else may know more than I |
| 09:54.14 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
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| 10:01.10 | tzafrir | nfi|ermes, there's no bristuff for asterisk 1.4 yet |
| 10:02.16 | *** join/#asterisk asteriskdude (n=test@59.93.56.163) |
| 10:02.47 | asteriskdude | hello - can someone suggest me a reliable UK Toll Free Provider? We are looking for about 250 such numbers. |
| 10:03.07 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 10:03.33 | x86 | origination? |
| 10:04.48 | asteriskdude | yes, origination |
| 10:04.57 | x86 | you can try didx.org |
| 10:05.11 | x86 | i've got a few UK DID's from them before that worked fine |
| 10:05.38 | asteriskdude | any ballpark estimate? |
| 10:05.44 | chiang_sg | anyone ever try using TDM400P with freebsd ? |
| 10:08.22 | *** join/#asterisk Avochelm (n=damien@60.226.239.115) |
| 10:11.30 | x86 | asteriskdude: didx.org |
| 10:17.32 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 10:19.11 | jeremy_g | shalt thou say unto me is asterisk-1.4 's sip stuff stable |
| 10:19.20 | jeremy_g | ? |
| 10:19.38 | jeremy_g | i heard 1.4 's sip stack is not production ready |
| 10:19.58 | x86 | thou shalt not covert they neighbor's RTP |
| 10:20.01 | x86 | thy* |
| 10:20.39 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 10:21.39 | jeremy_g | thou inform me of a possibility so sinful that I shall not adventure to tread into 1.4 realm |
| 10:22.30 | jeremy_g | are you words carrying the freight of truth x86? |
| 10:24.11 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 10:25.24 | jeremy_g | cmon guys, i have to upgrade my prod server to it.shud i or tell the management that its sip stack is not super stable as yet |
| 10:26.04 | bhrobinson | where have you gotten your feeling that the stack is not stable? |
| 10:26.14 | jeremy_g | from the stack author |
| 10:27.00 | *** join/#asterisk Dibbler_XP (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 10:27.43 | bhrobinson | I do not have any info on it, but would assume it is stable. I have it loaded on a standalone for testing right now. |
| 10:28.04 | jeremy_g | isn't there anyone using it in production? |
| 10:28.55 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 10:28.55 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:29.00 | angryuser | good day |
| 10:30.27 | jeremy_g | yo angryuser |
| 10:31.22 | angryuser | i have a question, foe example i have sip 350 call by a client, sip 400 want to pickup that call, i need to program a button like pickup(350@context) for 400? |
| 10:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 10:45.20 | *** part/#asterisk s1gny|wrk (n=s1gny@p54916945.dip.t-dialin.net) |
| 10:53.05 | *** join/#asterisk flimmern (n=sym@85.183.139.244) |
| 10:58.10 | tzafrir | angryuser, you were making some statements. And your question is? |
| 10:59.12 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 11:00.07 | angryuser | tzafrir:nevermind, cmd pickup worked fine |
| 11:01.38 | *** join/#asterisk bkw__ (n=brian@ip68-0-120-100.tu.ok.cox.net) |
| 11:03.52 | flimmern | anyone experienced with voismart vgsm cards an asterisk? |
| 11:04.35 | *** join/#asterisk tld (n=terje@elde.net) |
| 11:04.48 | flimmern | i think i'll need a little wink :/ |
| 11:14.15 | Ahrimanes | any realtime experts here? am experiencing funny stuff with sip peer regseconds field in the database |
| 11:17.50 | Ahrimanes | is there a description of each db field for realtime anywhere? |
| 11:22.16 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 11:22.38 | *** join/#asterisk rootfield (n=rootfiel@200.103.96.98) |
| 11:22.41 | rootfield | hi |
| 11:23.26 | rootfield | how can I send Invite without register? i need to start call from ras to asterisk.. any ideas ? |
| 11:24.33 | *** join/#asterisk EyeCue (n=eyecue@220-253-93-124.QLD.netspace.net.au) [NETSPLIT VICTIM] |
| 11:25.51 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 11:25.53 | *** join/#asterisk ftexcom (n=info@14.Red-80-26-4.staticIP.rima-tde.net) |
| 11:25.56 | ftexcom | hi |
| 11:26.09 | ftexcom | anyone knows a "dual" phone...I mean...rj11 and Ethernet |
| 11:26.34 | tzafrir | is that RJ45 really an ethernet? |
| 11:27.14 | ftexcom | Ok, if you wanna put that way...I'm looking for a dual phone, that admitts rj11 and rj45 |
| 11:27.29 | tld | PSTN and VoIP? |
| 11:27.30 | jeremy_g | ftexcom:what is rj11 for? |
| 11:27.33 | jeremy_g | pstn |
| 11:27.38 | tld | RJ-11 and RJ 45 are just plugs |
| 11:27.41 | tzafrir | what type of voip? |
| 11:27.43 | jeremy_g | yup |
| 11:27.45 | ftexcom | yes, pstn and VoIP |
| 11:27.56 | jeremy_g | but can you put an rj11 in front of ethernet |
| 11:27.57 | *** join/#asterisk Avochelm (n=damien@60.226.239.115) |
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| 11:28.03 | jeremy_g | what are the tradeoffs i wonder |
| 11:28.14 | jeremy_g | of mixing connectors :P |
| 11:28.16 | tld | You could run ADSL over either RJ-11 or RJ-45 |
| 11:28.23 | tld | or them being plugs, I guess through them |
| 11:28.41 | tzafrir | tld, there is an ADSL modem in the middle |
| 11:28.43 | tld | I don't know of any phones supporting ADSL natively |
| 11:29.12 | tld | ftexcom, What is it actually you're trying to do? |
| 11:29.20 | tld | ftexcom, Have a VoIP phone that can call PSTN as well? |
| 11:30.21 | ftexcom | Here's the case...A client dosn't want to re-cable his entire hotel with ethernet cable...so he want's to stay with his pstn cables... |
| 11:30.21 | tld | ftexcom, If you just want to call, without any "fancy" VoIP stuff, you could just get a normal adapter which supports a PSTN line. |
| 11:30.21 | ftexcom | I don't know If I'm explaining correctly |
| 11:30.22 | tld | ftexcom, Now you did. |
| 11:30.22 | rootfield | how can I send Invite without register? i need to start call from ras to asterisk just send invite, no send register.. any ideas ? |
| 11:30.22 | tld | ftexcom, Just put a lot of adapters in where his cables run from? |
| 11:30.22 | Makenshi | our buildings have only rj45 with which we deliver analog, digital and ip phones |
| 11:30.28 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 11:30.31 | PakiPenguin | hi there , |
| 11:30.39 | tld | ftexcom, He could keep both the cables, AND the phones. |
| 11:30.44 | PakiPenguin | i have a question regarding the gtalk channel |
| 11:30.55 | PakiPenguin | whenever i send a call to the gtalk channel , asterisk crashes |
| 11:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:33.48 | *** join/#asterisk santibiotico (n=santi@208.Red-83-58-113.dynamicIP.rima-tde.net) |
| 11:33.49 | santibiotico | hi |
| 11:33.58 | ftexcom | tld yes, that's what I'm thinking. |
| 11:34.04 | santibiotico | i'm using spandsp to receive faxes... |
| 11:34.45 | santibiotico | and i'm trying to receive a fax from a voxbone did |
| 11:34.52 | santibiotico | but i have no success at all |
| 11:35.03 | ftexcom | but, anyone knows of a "dual" phone? |
| 11:35.13 | santibiotico | does anybody know any way to get it functioning??? |
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| 11:39.18 | PakiPenguin | :( |
| 11:40.56 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 11:41.37 | Ahrimanes | if i have a fullcontact on a realtime sippeer, but the peer has expired (current timestamp > regseconds) - can * still call this peer? |
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| 11:47.32 | santibiotico | anyone using voxbone and fax? |
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| 12:00.02 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 12:01.21 | littleball | hello, i have E1/PRI, i got a few DID/DDI numbers from operators. How to specify the caller id when the system make a outgoing call? so that the receipents can see a specific phone number |
| 12:03.34 | uwe | littleball, when you create an extension, set the outgoing cid |
| 12:03.58 | uwe | *outbound cid |
| 12:05.33 | littleball | uwe, what is cid? |
| 12:05.59 | littleball | Callerid: Wakeup Call Service |
| 12:06.01 | littleball | liek this |
| 12:06.02 | littleball | ? |
| 12:09.37 | uwe | i suppose the format is "caller name" <blah> |
| 12:10.00 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 12:11.17 | uwe | littleball, well, im not sure, i use the freepbx interface ... sorry |
| 12:12.56 | *** join/#asterisk oej (n=olle@216.64.24.250) |
| 12:14.33 | PakiPenguin | hi can anyone take a look at this http://pastebin.com/871150 <-- i am trying my luck with 1.4 and gtalk |
| 12:19.25 | littleball | PakiPenguin, what you are trying to do? |
| 12:20.07 | PakiPenguin | littleball: trying to map a DID to a google talk account ( call comes in from did and passes it on to gtalk ) |
| 12:20.16 | PakiPenguin | but as soon as the call comes in and passes to the gtalk channel |
| 12:20.19 | PakiPenguin | asterisk crashes! |
| 12:20.36 | littleball | where do you get the gtalk channel lib? |
| 12:20.48 | PakiPenguin | asterisk |
| 12:20.49 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.105) |
| 12:20.57 | PakiPenguin | libiksemel or something like that |
| 12:21.00 | PakiPenguin | 1.4 comes with that |
| 12:21.15 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:21.21 | littleball | i see. libiksemel should be reliable. it could be your os |
| 12:21.35 | PakiPenguin | CentOS release 4.4 (Final) |
| 12:21.58 | puzzled | hi |
| 12:22.00 | littleball | ok. i am using CentOS also |
| 12:22.08 | PakiPenguin | yeah littleballand it works okay for you? |
| 12:22.15 | dasenjo | Hi! I'm using madplay to play moh without problems, I want to make it reproduce the mp3 file from the beginning every time that the moh starts on a channel, ¿what can I do? |
| 12:22.24 | littleball | i didn't try... but i am fan of gtalk.. |
| 12:22.58 | *** part/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 12:23.19 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 12:24.19 | Caitriona | dasenjo: moh is a continious stream shared by all channels. |
| 12:24.22 | Caitriona | it isn't possible. |
| 12:25.00 | Caitriona | it's designed to be a shared resource. |
| 12:25.02 | dasenjo | Caitriona, :( ¿do I have to compile format_mp3? |
| 12:27.22 | Ahrimanes | format_mp3 only enables you to play mp3 natively instead of using madplay |
| 12:27.27 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 12:27.28 | ThoMe | hiho |
| 12:28.11 | ThoMe | i have a agi script, in php written. this send the name from the caller to my display |
| 12:28.14 | ThoMe | with |
| 12:28.16 | ThoMe | what is here wrongß |
| 12:28.19 | ThoMe | $run .= 'EXEC Set(CALLERID "'.$telefonnummer.'<'.$telefonbuch_vorname.' '.$telefonbuch_nachname.'>")' |
| 12:28.23 | ThoMe | servetux:/usr/share/asterisk/agi-bin# ./test.php 01786323765 |
| 12:28.23 | ThoMe | EXEC Set(CALLERID "01786323765<Thomas Mebes>") |
| 12:28.27 | ThoMe | is it wrong? |
| 12:28.30 | ThoMe | if yes.. why? |
| 12:29.10 | Ahrimanes | dont you need a = somewhere in there? |
| 12:29.37 | ThoMe | hm. i dont know... :/ |
| 12:29.43 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 12:30.32 | Ahrimanes | Set(CALLERID="...") |
| 12:31.00 | Ahrimanes | or Set(CALLERID(name)="Name") and Set(CALLERID(number)=number) |
| 12:32.32 | JT | Ahrimanes: it's ago, but yes, CALLERID(num) etc is now the correct way to do it |
| 12:32.46 | JT | s/ago/agi/ |
| 12:33.05 | Ahrimanes | hehe |
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| 12:35.35 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 12:36.12 | JT | [Jan 30 07:31:07] NOTICE[30754]: chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach '8002081258@' |
| 12:36.54 | JT | the last bit, does that mean the machine was trying to reach extension 8002081258 at context NULL, or something else? |
| 12:37.08 | ThoMe | JT not |
| 12:37.09 | ThoMe | $run .= 'EXEC Set(CALLERID(name)=test)'; |
| 12:37.09 | ThoMe | ? |
| 12:37.26 | JT | i dont know ThoMe |
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| 12:52.04 | ManxPower | OT: Any linux sound gurus here? If so, /msg me, I have a couple of questions. |
| 12:53.31 | *** part/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 13:03.04 | bhrobinson | anyone here have an example of an extensions.conf that uses the h extension I can use as a reference? |
| 13:05.43 | UVSoft | there's one more problem(( there're two devices: FXO and FXS, the FXO one is connected with a phone through TLE, so I call from the FXO to the FXS, after a short talk I hang up the phone, and the TLE plays a dial tone to the FXO, and neither Asterisk nor Zaptel detect this tone, so they continue working, and from the other phone (connected direct to the FXS) I hear that dial tone. but Asterisk(Zaptel?) should hang up!!! |
| 13:07.07 | orlock | *** #slashdot You need to be invited to that channel (Invite only channel) |
| 13:07.17 | orlock | heh |
| 13:07.28 | *** join/#asterisk burnproof (n=jsharryp@58.69.177.28) |
| 13:07.57 | ThoMe | Ahrimanes: no i have... |
| 13:09.22 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 13:10.30 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:13.51 | Ahrimanes | ThoMe, huh? |
| 13:20.14 | *** join/#asterisk AstaWerksDotCom (n=doug@63.161.96.170) |
| 13:20.33 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 13:20.52 | ThoMe | Ahrimanes: i have |
| 13:20.53 | ThoMe | <PROTECTED> |
| 13:20.55 | ThoMe | <PROTECTED> |
| 13:20.59 | ThoMe | echo $run; |
| 13:21.54 | Ahrimanes | ThoMe, afair those are deprecated applications for callerid... what version of asterisk are you using? |
| 13:22.47 | ThoMe | oeh |
| 13:22.48 | ThoMe | servetux*CLI> show version |
| 13:22.48 | ThoMe | Asterisk 1.2.14 built by kk @ nyx on a x86_64 running Linux on 2007-01-15 11:08:58 UTC |
| 13:26.36 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 13:27.27 | anonymouz666 | 1 GHz can handle about how much g711 calls? |
| 13:27.33 | anonymouz666 | many |
| 13:27.41 | JT | delivered how? |
| 13:27.57 | anonymouz666 | e1 |
| 13:28.26 | Corydon76-home | Approximately 28 G.711 calls |
| 13:28.36 | Corydon76-home | Depending upon what protocol you use |
| 13:28.57 | Corydon76-home | IAX2 will allow more calls than SIP, because IAX2 uses less header space |
| 13:29.23 | Corydon76-home | If you use a compressed codec, you can do more |
| 13:29.44 | anonymouz666 | Corydon76-home: did you test with sipp tool? |
| 13:30.17 | Corydon76-home | Nope, just giving you the theory behind it |
| 13:31.03 | Ahrimanes | ThoMe, Set(CALLERID(name)="my name") and Set(CALLERID(num)=mynumber) are the ones to use afaik |
| 13:31.36 | ThoMe | Ahrimanes: if i try it in context with agi i can use noet set( .. |
| 13:31.52 | Ahrimanes | hm |
| 13:33.58 | Ahrimanes | functions aren't available from AGI ? |
| 13:33.58 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:33.58 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:33.58 | ThoMe | hm. i 'dont know. asterisk said if i try your context.... |
| 13:33.58 | ThoMe | mom..i test it |
| 13:34.43 | Ahrimanes | k |
| 13:36.08 | ThoMe | <PROTECTED> |
| 13:36.08 | ThoMe | <PROTECTED> |
| 13:36.11 | ThoMe | Jan 30 14:35:58 WARNING[22602]: res_agi.c:1115 handle_exec: Could not find application (Set(CALLERID(num)=1234567)) |
| 13:36.15 | ThoMe | Ahrimanes: |
| 13:36.15 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 13:39.22 | ThoMe | Ahrimanes: looking? ;) |
| 13:40.26 | Ahrimanes | ThoMe, ah yes... only dialplan applications are available.. not dialplan functions... damn |
| 13:40.44 | ThoMe | hm. ok. |
| 13:40.45 | ThoMe | :-) |
| 13:41.26 | Ahrimanes | that kind of sucks actually |
| 13:41.41 | ThoMe | hihi |
| 13:43.17 | Corydon76-home | Ahrimanes: eh? |
| 13:43.32 | Corydon76-home | SETVAR CALLERID(num) 1234567 |
| 13:44.34 | UVSoft | [phone1 <-> TLE <-> FXO <-> Asterisk + Zaptel <-> FXS <-> phone2], call from phone1 to phone2, hang up phone1, and still hear tial tone from TLE on phone2, so who is in charge of detecting a dial tone and than hanging up? |
| 13:45.02 | UVSoft | :%s/tial/dial |
| 13:45.09 | Corydon76-home | errr, SET VARIABLE CALLERID(num) 1234567 |
| 13:45.39 | ThoMe | Corydon76-home: can i use this also for isdn? |
| 13:45.49 | ThoMe | SET VARIABLE CALLERID(name) Thomas |
| 13:45.49 | ThoMe | ? |
| 13:45.56 | ThoMe | or SETCIDNAME.. ? |
| 13:46.05 | Corydon76-home | You can use this for AGI. What the ISDN channel supports is another thing |
| 13:46.15 | Ahrimanes | ah setvar |
| 13:46.33 | ThoMe | Corydon76-home: hm. ok |
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| 14:13.49 | guillecabeza | hi! |
| 14:13.59 | x86 | HEY! |
| 14:14.20 | guillecabeza | anyone with time to discuss an issue? |
| 14:14.49 | guillecabeza | w're having some issues with deadlock on ChanSpy |
| 14:14.56 | hi365 | lookes like fop is going to be depreciated soon! |
| 14:14.57 | hi365 | http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29 |
| 14:15.09 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:15.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:15.15 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 14:15.30 | guillecabeza | I did open an incident , 8922 |
| 14:15.43 | guillecabeza | and wanna discuss alternatives... |
| 14:15.46 | *** join/#asterisk puthre (n=puthre@80.97.12.10) |
| 14:15.46 | Ahrimanes | hm not really, fop has an open socket via flash to do real time updating of status on the webpage.. with ajam, the browser has to initiate the request afaik |
| 14:15.52 | *** join/#asterisk af_ (n=getsmart@ip-164-140.sn2.eutelia.it) |
| 14:16.27 | x86 | ajam? |
| 14:16.29 | hi365 | im glad its staying. i really like it |
| 14:16.30 | hi365 | does anyone use microsoft access to connect to the asterisk manager? |
| 14:16.56 | b11d | morning all |
| 14:17.52 | Ahrimanes | x86, see the link hi365 posted earlier |
| 14:18.30 | x86 | oh that's hawt! |
| 14:18.52 | Ahrimanes | x86, should make coding webapps for * somewhat easier |
| 14:19.23 | coppice | its like ajax, but they've now solved for x |
| 14:19.48 | Ahrimanes | solved? |
| 14:19.53 | hi365 | a bit of java should replace the flash though, dont you think? |
| 14:20.12 | x86 | java sucks for web stuff |
| 14:20.23 | Ahrimanes | hi365, that's quite possible... still i dont like having java or flash as a requirement for browsers |
| 14:20.38 | x86 | flash > java, imho |
| 14:21.05 | puthre | does anyone know how much will cost an asterisk appliance? |
| 14:21.05 | hi365 | oh well |
| 14:21.13 | hi365 | soes anyone use asterisk + access? |
| 14:21.37 | Ahrimanes | x86, there are projects out there to make small java applets for such use |
| 14:21.45 | JT | they're both poo |
| 14:21.49 | JT | neither would be ideal :D |
| 14:21.55 | Ahrimanes | haha |
| 14:21.58 | Ahrimanes | access mostly so |
| 14:22.12 | JT | java/flash i was refering to |
| 14:22.50 | hi365 | acces with asterisk? as in the manager? |
| 14:24.14 | guillecabeza | anyone knows if there's an asterisk-dev channel? |
| 14:25.14 | dwmw2 | guillecabeza: /list seems to suggest that there is |
| 14:26.15 | Ahrimanes | JT, ah yes |
| 14:26.38 | santibiotico | i'm using nvfaxdetect over a voxbone DID but it doesn't detect faxes |
| 14:27.00 | santibiotico | anyone using fax to email with dids? |
| 14:28.54 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 14:29.29 | JoNate | Holy hell theres a lot of you in here! |
| 14:29.40 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:29.46 | Ahrimanes | yup we're all over |
| 14:30.28 | JoNate | So, I've been given a task here, and I've only been given three days, and I was just wondering if I could get some opinions from you guys |
| 14:31.11 | JoNate | I need to get an Asterisk box up by Friday, and was wondering what distro of Linux everyone uses. This is just a base install, no bells and whistles, and we'll use it for a small office |
| 14:31.19 | *** join/#asterisk profounded (n=pro@ool-44c4e6c0.dyn.optonline.net) |
| 14:31.39 | profounded | i have a quick and easy question, should i be purchasing an fxo or an fxs module so I can setup my fax machine ? |
| 14:32.09 | profounded | I want to fax over voip (unless thats a really bad idea) |
| 14:32.58 | hypnox | it doesnt work |
| 14:33.13 | profounded | ? |
| 14:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:33.32 | hypnox | fax over voip mostly doesn't work, so yeah it's a bad idea |
| 14:34.19 | hypnox | if you send the fax voip over a LAN as far as your asterisk box and then a landline to send it onwards, that sometimes is ok |
| 14:34.56 | profounded | yea.. im considering not using fax or maybe a digital service now |
| 14:35.25 | profounded | ty hypnox |
| 14:35.44 | Gido-E | our fax shizzle works, ok for us. |
| 14:35.55 | Ahrimanes | ymmv |
| 14:36.03 | profounded | over voip? and everytime? |
| 14:36.21 | Gido-E | inbound is an sip account. |
| 14:36.33 | Gido-E | outbound, we dont use. |
| 14:36.35 | profounded | how about outbound? |
| 14:36.38 | profounded | got it |
| 14:36.52 | AstaWerksDotCom | inbound always works fine for me with voicepulse and rxfax |
| 14:36.56 | Gido-E | fax is so OLD :-) |
| 14:37.10 | x86 | i've had great success with inbound VoIP faxes using rtxfax / spandsp |
| 14:37.12 | hypnox | yeah fax sucks but some people insist it MUST be used :( |
| 14:37.22 | profounded | do you use an fxo or an fxs module to setup inbound, |
| 14:37.51 | hypnox | a sipura/linksys ata sends/recieves faxes ok too |
| 14:38.23 | x86 | hypnox: if your provider supports T.38, sure |
| 14:38.23 | x86 | which asterisk can't handle |
| 14:38.32 | x86 | need openpbx or yate or such to do that |
| 14:39.03 | Ahrimanes | 1.4 has T.38 passthrough afair? |
| 14:41.56 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 14:42.31 | *** join/#asterisk jacekowski (n=jacekows@aaot198.neoplus.adsl.tpnet.pl) |
| 14:43.34 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 14:44.48 | angryuser | i have extensions t,1,playback(file), but they dont really work, after 200,1,Background(intro) i got a hang up, any ideas? |
| 14:45.44 | L|NUX | i have very strange problem with My ATA 186, when it registered and after 1 minute when i try to call some one i got no dial tone |
| 14:45.50 | L|NUX | any one fimilar with this problem |
| 14:45.57 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 14:49.35 | *** join/#asterisk saftsack (n=saftsack@pD9E0593B.dip.t-dialin.net) |
| 14:49.36 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:49.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:49.56 | x86 | L|NUX: dont buy Grandstream products... they're crap |
| 14:50.28 | L|NUX | humm |
| 14:50.34 | L|NUX | its Cisco ATA :( |
| 14:51.15 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 14:51.39 | SomeOne1 | cisco PBX > asterisk |
| 14:51.46 | *** join/#asterisk kodz (n=kodz@nowoplex.concrete.pl) |
| 14:51.54 | JT | err what? |
| 14:52.04 | SomeOne1 | haha jk! |
| 14:52.06 | SomeOne1 | jk |
| 14:52.08 | SomeOne1 | hey JT |
| 14:52.18 | JT | hi |
| 14:52.56 | kodz | is there a possibility to hangup (or interrupt actually) the call that wasn't connected yet? i'm originating a call from manager api interface, but i can't cancel it unless its connected already. how to do that? |
| 14:53.14 | jeremy_g | is 1.4 production ready? |
| 14:53.14 | jeremy_g | <jeremy_g> i heard its sip stack is not as stable as that of 1.2 |
| 14:53.14 | jeremy_g | <jeremy_g> is that a rumour? |
| 14:53.44 | kodz | or how to get the channel for unconnected call on zap interface? (before CONNECT message is received) |
| 14:54.09 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:54.44 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 14:54.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:54.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:55.21 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:55.48 | kodz | any better place for that question? |
| 14:55.53 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:57.21 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 14:57.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 14:58.34 | *** join/#asterisk JerJer[mobile] (n=jj@208.65.54.162) |
| 14:59.36 | anonymouz666 | jeremy_g |
| 14:59.39 | anonymouz666 | JerJer[mobile]! |
| 14:59.44 | mut | are the gains in zapa reverse? |
| 14:59.49 | mut | 6 = quiet and -6 = loud |
| 14:59.53 | anonymouz666 | JerJer[mobile]: do you use truphone on nokia e61? |
| 15:00.13 | anonymouz666 | the native SIP software using the last firmware didn't reply the 407 for the register sent by asterisk |
| 15:01.45 | *** part/#asterisk kodz (n=kodz@nowoplex.concrete.pl) |
| 15:01.57 | tzafrir | mut, no |
| 15:02.54 | jeremy_g | anonymouz666:yes |
| 15:04.42 | *** join/#asterisk ez` (n=ez@c66.203.210-59.clta.globetrotter.net) |
| 15:05.06 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 15:11.45 | JerJer[mobile] | anonymouz666: truphone would not install on my E61 |
| 15:12.11 | JerJer[mobile] | but then again my E61 functions with asterisk, behind nat perfectly fine |
| 15:12.24 | JerJer[mobile] | however for the life of me I cannot make it work with SER |
| 15:13.23 | anonymouz666 | JerJer[mobile]: i give up on sip stuff using e61 |
| 15:13.31 | anonymouz666 | i can't put to register on asterisk |
| 15:13.42 | JerJer[mobile] | turn security off |
| 15:13.54 | anonymouz666 | e61 does not answer the register handshake 407 |
| 15:14.02 | JerJer[mobile] | make sure sip: proceeds your username, proxy, registar, etc |
| 15:14.38 | anonymouz666 | yes, i see that 100 million times |
| 15:14.48 | JerJer[mobile] | we got it to work |
| 15:15.11 | JerJer[mobile] | even still running the 2.X firmware |
| 15:15.33 | anonymouz666 | I am using the lastest firmware |
| 15:15.45 | JerJer[mobile] | PL3 is supposed to fix a lot more but for whatever reasons nokia hasn't released that code all the way yet |
| 15:16.16 | JerJer[mobile] | define latest - *#0000# |
| 15:16.31 | anonymouz666 | 3.06.33.09.04 |
| 15:17.47 | JerJer[mobile] | fun - i have 2.0618.06.05 and it works with Asterisk even behind NAT |
| 15:17.53 | JerJer[mobile] | but refuses to work with SER |
| 15:18.07 | JerJer[mobile] | ser thinks its registsered |
| 15:18.19 | anonymouz666 | e61 does not re-send the register request with credentials |
| 15:18.31 | JerJer[mobile] | is it getting the message? |
| 15:18.39 | JerJer[mobile] | try using a different port than 5060 |
| 15:21.00 | anonymouz666 | JerJer[mobile]: how I will know that the packet is reaching my e61? |
| 15:21.12 | anonymouz666 | there some kind of ethereal? ngrep? |
| 15:22.22 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 15:26.13 | tzanger | north american ringback is 2s of tone, 2s silence, right? |
| 15:26.32 | coppice | 2 on 4 off |
| 15:26.36 | tzanger | 2 on 4 off, ahh okay |
| 15:27.23 | coppice | the gap is so long that europeans visiting keep thinking the ringing has stopped :-\ |
| 15:27.56 | tzanger | :-) |
| 15:28.01 | tzanger | how are things, coppice? |
| 15:28.35 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 15:29.19 | coppice | OK |
| 15:29.19 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:30.02 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-e810110e89d2c3e3) |
| 15:30.31 | *** join/#asterisk Lurchtoke (i=Lurchtok@adsl-75-37-77-14.dsl.frs2ca.sbcglobal.net) |
| 15:30.36 | Lurchtoke | morning people |
| 15:31.26 | anonymouz666 | JerJer[mobile]: do you use something as realm? |
| 15:31.29 | anonymouz666 | it's blank here |
| 15:32.02 | Lurchtoke | stupid question from me (the noob).... |
| 15:32.25 | x86 | there are no stupid questions, only stupid people |
| 15:32.30 | x86 | or something like that ;-) |
| 15:32.33 | *** join/#asterisk trixman (n=andy@rrcs-67-53-168-147.west.biz.rr.com) |
| 15:33.00 | Lurchtoke | i need to change a sound file in one of my menus...which configuration file references the audio file I need to have played in a certain option? |
| 15:33.10 | Lurchtoke | so I can change the association |
| 15:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:33.23 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:34.04 | mercestes | Lurchtoke: Extensions.conf. Your not using trixbox are you? |
| 15:34.04 | Lurchtoke | I just began learning this week..and im not technically able to do it myself at this point |
| 15:34.17 | Lurchtoke | ok... |
| 15:34.26 | Lurchtoke | extensions.conf. |
| 15:34.30 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 15:34.31 | mercestes | yea, that one. |
| 15:34.41 | *** part/#asterisk mut (n=ana@65.111.222.120) |
| 15:34.41 | Lurchtoke | ~book |
| 15:34.42 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:35.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:35.24 | errr | what are the packet types sent when you are talking on your phone? Im doing some QoS stuff with pfsync on my freebsd system and I want to give priority to the asterisk packets |
| 15:35.55 | mercestes | errr: Your probably referring to the RTP stream via udp. |
| 15:36.37 | trixman | wouldn't the packets be on the media ports? |
| 15:36.51 | errr | mercestes: so in /etc/protocols I have crtp and irtp is it one of these? |
| 15:38.12 | *** join/#asterisk Hairai (n=Hairai@m108.net81-65-51.noos.fr) |
| 15:40.15 | *** join/#asterisk tb0301s (n=sowa@brln-4db12501.pool.einsundeins.de) |
| 15:40.22 | Lurchtoke | another dumb question....can i just d/l the extensions.conf. file...edit it and then upload it back? |
| 15:40.27 | Lurchtoke | is it that easy? |
| 15:40.55 | De_Mon | Lurchtoke you have to tell asterisk to reload it |
| 15:40.59 | Lurchtoke | ok |
| 15:41.10 | De_Mon | restarting work too, but that would be over kill |
| 15:41.36 | Lurchtoke | I know...I need to do more studying...but im just trying to change our business address recording... |
| 15:41.44 | Lurchtoke | how do you tell asterisk to reload it? |
| 15:41.50 | Lurchtoke | what command at the root? |
| 15:42.47 | Carp1 | Can anyone help me with this: http://forums.digium.com/viewtopic.php?t=13207 |
| 15:42.56 | Carp1 | Ive been working for hours and hours on it and no progress |
| 15:45.09 | mercestes | errr: ? never had to mess with /etc/protocols. I dunno. |
| 15:45.50 | trixman | reload |
| 15:45.55 | trixman | is what you type |
| 15:47.30 | errr | mercestes: it looks to me like QoS via pfsync isnt gonna work because of this |
| 15:47.51 | errr | mercestes: do people gennerally need to worry about QoS with asterisk? |
| 15:48.02 | x86 | yes |
| 15:48.04 | Carp1 | Depends on your setup |
| 15:48.12 | errr | we have ~200 phones |
| 15:48.30 | errr | what can you use for QoS then? |
| 15:48.36 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:48.40 | Carp1 | most definately need to worry about QiS |
| 15:48.47 | Carp1 | QoS* |
| 15:49.14 | *** join/#asterisk andrebarbosa (n=support@62.48.215.205) |
| 15:50.07 | errr | our asterisk system will be running on linux, but our firewalls and routers are freebsd and openbsd, can anyone reccomend something for QoS? |
| 15:50.47 | andrebarbosa | ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 |
| 15:51.03 | andrebarbosa | is this normal? my telecom does not detect my busy sign |
| 15:51.10 | andrebarbosa | analog connection~btw |
| 15:51.10 | andrebarbosa | :s |
| 15:52.10 | andrebarbosa | i have on my dialplan a simple hangup |
| 15:52.33 | andrebarbosa | it hangups up the channel, but my telecom does not detect it and continues to ring |
| 15:52.44 | *** join/#asterisk amdtech (i=amdtech@nat/digium/x-a5f8c860cacbb73c) |
| 15:53.00 | andrebarbosa | few seconds after my tdm400p still detecting the ring tone and think it's a new call, and dials again |
| 15:53.01 | andrebarbosa | :S |
| 15:53.58 | *** join/#asterisk erickperez (n=lolo@200.75.246.173) |
| 15:54.37 | erickperez | Hi people, If I use asterisk SVN, how do I "update" my installation? |
| 15:55.04 | amdtech | in the asterisk source directory, type "svn update" |
| 15:58.24 | erickperez | aamdtech: and after that when i do make all make install all the libraries and stuff will be overwritten, right? |
| 15:58.34 | amdtech | yep |
| 15:58.42 | erickperez | excellent. |
| 15:58.58 | amdtech | if you get any errors, you can do rm -f /usr/lib/asterisk/modules/* |
| 15:59.01 | amdtech | and make install again |
| 16:00.04 | erickperez | amdtech: do you have an idea as to what steps should I take to document a deadlock that occurs when I park a call in a sip-zap conversation? It's a VIA EPIA 1.2ghz 1 gb ram board with only 5 users on it. |
| 16:00.29 | erickperez | amdtech: the deadlock freezes asterisk. and i have to kill/restart. |
| 16:00.54 | amdtech | what version of asterisk are you running, and is it always reproducible? |
| 16:01.35 | mercestes | errr: It's not really too * to control QoS. It's to the routers to flag VoIP traffic with a higher priority |
| 16:01.45 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 16:01.54 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 16:02.02 | erickperez | In asterisk 1.2 branch SVN 51363 |
| 16:02.02 | erickperez | zaptel svn 1980 |
| 16:02.02 | erickperez | libpri svn 393 |
| 16:02.02 | erickperez | addons svn 332 |
| 16:02.02 | erickperez | My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a |
| 16:02.02 | erickperez | tdm400p (4fxo). |
| 16:02.04 | erickperez | A call comes from zap, a SIP ulaw receives the call, talks for a while |
| 16:02.06 | erickperez | and when SIP users tries to park the call, then dozens of... |
| 16:02.08 | erickperez | WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial |
| 16:02.09 | mercestes | errr: You can either do it by interface address or by traffic type. Tell your openBSD box to prioritize RTP from the * box. |
| 16:02.10 | erickperez | deadlock for '0x91bb840', 10 retries! |
| 16:02.12 | erickperez | I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL |
| 16:02.14 | erickperez | I posted the digium script output from autosupport. It is available at: |
| 16:02.16 | erickperez | http://pastebin.com/868590 |
| 16:02.46 | mercestes | erickperez: I can't read what you typed. Could you say that again? This time pressing enter after each word so it spaces out? |
| 16:02.48 | *** join/#asterisk wnfaknd (i=wnfaknd@cpe-24-30-183-141.socal.res.rr.com) |
| 16:03.18 | erickperez | ok sorry..i cut/paste |
| 16:03.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 16:03.45 | mercestes | can you say "pastebin"? |
| 16:03.51 | amdtech | lol |
| 16:03.58 | amdtech | i got the messages fine :) |
| 16:04.00 | mercestes | erickperez: Do you have a conference room setup? |
| 16:04.13 | erickperez | zaptel snv 1980, libpri svn 393, asterisk svn branch 51363 addons svn 332 |
| 16:04.28 | erickperez | via epia motherboard 1.2ghz, 1gbram |
| 16:04.43 | mercestes | erickperez: omg....that's not production is it? |
| 16:05.04 | amdtech | best bet... try upgrading asterisk to the latest version and try again |
| 16:05.10 | erickperez | call comes from zap tdm400p (fxo) received by ulaw SIP. they talk fine. but if SIP parks the call then: |
| 16:05.20 | erickperez | WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial |
| 16:05.20 | erickperez | deadlock for '0x91bb840', 10 retries! |
| 16:05.28 | mercestes | erickperez: ....dude. I was kidding |
| 16:05.39 | erickperez | mercestes: yes is production. |
| 16:05.50 | erickperez | why? |
| 16:05.57 | mercestes | erickperez: oh.....nothing. |
| 16:06.01 | amdtech | how are you parking? |
| 16:06.25 | erickperez | it's only an 4 agent queue with 1:1 ratio over a 4fxo. |
| 16:06.43 | amdtech | right, but what are you using to park users? |
| 16:07.10 | erickperez | so 1 agent, 1 fxo...the customer does not want any of their customers to be put on MOH..they want live experience..and it's only outbound. . |
| 16:07.25 | amdtech | let me see your features.conf |
| 16:07.48 | erickperez | ok, let me get to code and put it into pastebin. |
| 16:10.02 | ThoMe | hello.. i have a script in php for call about web |
| 16:10.02 | ThoMe | http://tm.muc.de/anruf.phps |
| 16:10.21 | ThoMe | but with how command / function i can this call cancel FROM the asterisk console? |
| 16:11.04 | Carp1 | any nufone users? |
| 16:11.17 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 16:11.49 | ThoMe | hm? |
| 16:13.19 | erickperez | machine specs: http://pastebin.com/868590 |
| 16:13.33 | erickperez | asterisk config: http://pastebin.com/871316 |
| 16:13.38 | nosbig | I am having trouble with call parking as well... I can get a call to park in a parking lot using the phone's built-in transfer, but the phone disconnects before * can announce the parking lot extension. |
| 16:14.12 | amdtech | are you doing a blind transfer? |
| 16:14.12 | erickperez | amdtech: my agents use AgentLogin to login to asterisk. |
| 16:14.39 | amdtech | erickperez: i see that, i'm asking about the parking function |
| 16:14.40 | erickperez | yep |
| 16:14.42 | amdtech | what're you using to park? |
| 16:14.58 | nosbig | amdtech: Yes, I am... But I haven't tried adding the T flag to my dial string... |
| 16:15.14 | *** join/#asterisk AvoidingDeadlock (n=brian@m115e36d0.tmodns.net) |
| 16:15.37 | nosbig | amdtech: So I can't use # to initiate a transfer, as the other directions I have read indicate... But I am able to park the call and pick it up... |
| 16:16.22 | amdtech | nosbig: you have to use an attended transfer to get the audio back from asterisk |
| 16:16.29 | amdtech | blind transfer just dials and forwards, no audio |
| 16:16.29 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 16:16.46 | Lurchtoke | another question |
| 16:16.59 | nosbig | oic |
| 16:17.04 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 16:17.10 | Carp1 | I cant get NuFone to work with asterisk, I am connected via IAX but I get an error in CLI when I call in....can someone PLEASE help me? |
| 16:17.32 | erickperez | amdtech: I tranfer then to an extension where i only play MOH. |
| 16:17.38 | *** join/#asterisk robl^ (n=robl@unaffiliated/robl/x-000001) |
| 16:17.44 | Lurchtoke | if I have the current audio file playing information which I want to replace...cant I just delete the current audio file...and replace it with the new one (providing i change its name to the previous name)> |
| 16:17.56 | Lurchtoke | ? |
| 16:18.13 | amdtech | Lurchtoke: yes |
| 16:18.20 | Lurchtoke | ok |
| 16:18.36 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:18.48 | amdtech | erickperez: in your dialplan, what extension are you forwarding them to? |
| 16:19.07 | erickperez | exten => 8302,1,Answer |
| 16:19.11 | erickperez | exten => 8302,2,Playback(conf) |
| 16:19.13 | Lurchtoke | because I can create an audio file in a box (6060) on our machine...I go into the dir with the audio files....find the one thats referenced...and delete it....then copy the new file over into the dir...and the rename it? |
| 16:19.16 | erickperez | exten => 8302,3,Hangup |
| 16:19.17 | Carp1 | chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach '8002081258@' |
| 16:19.29 | amdtech | Lurchtoke: as long as it's the same type :) |
| 16:19.33 | Lurchtoke | ok.. |
| 16:19.33 | erickperez | it's a 30 minute audio file. |
| 16:19.38 | Carp1 | I was working on this error all day yesterday with no luch |
| 16:19.46 | amdtech | erickperez: why not use the actual park functions? |
| 16:19.55 | Lurchtoke | there are diff types (forgive me...im a noob at *...if pretty strong in other stuff)) |
| 16:19.59 | amdtech | and use that audio file as the music on hold |
| 16:20.14 | Lurchtoke | no\ |
| 16:20.19 | erickperez | amdtech: what are those? |
| 16:20.21 | Lurchtoke | not hold music |
| 16:20.44 | Lurchtoke | its just where our * states our store location...we moved :P |
| 16:20.57 | Lurchtoke | so I need to change the phone menu recording |
| 16:21.05 | amdtech | Lurchtoke: helping too many people at once |
| 16:21.05 | amdtech | :) |
| 16:21.20 | Lurchtoke | no prob...you are appreciated! |
| 16:21.20 | *** part/#asterisk smackus (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 16:21.21 | Lurchtoke | :P |
| 16:21.22 | Carp1 | chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach '8002081258@' |
| 16:21.28 | Carp1 | I was working on this error all day yesterday with no luck |
| 16:21.33 | amdtech | erickperez: look here: http://www.voip-info.org/wiki-Asterisk+call+parking |
| 16:22.31 | erickperez | amdtech: will look inmediately...will go back to this channel and repost my findings...thanks a lot!! |
| 16:22.42 | amdtech | erickperez: no problem :) |
| 16:22.57 | amdtech | Lurchtoke: do you already have the audio file recorded? |
| 16:23.23 | mercestes | Lurchtoke: Do you have the name of the old file? |
| 16:24.00 | Lurchtoke | yes |
| 16:24.04 | Lurchtoke | ummm |
| 16:24.08 | Lurchtoke | I need to look it up |
| 16:24.21 | mercestes | Look it up. |
| 16:24.23 | Lurchtoke | ok |
| 16:24.29 | mercestes | then do a grep -i old_file_name extensions.conf |
| 16:24.47 | mercestes | and then do a mv extensions.conf extensions.emergency.backup |
| 16:24.49 | Lurchtoke | grep=replace? |
| 16:25.07 | mercestes | then do a cat extensions.emergency.backup | sed s/old_filename/new_filename/g > extensions.conf |
| 16:25.09 | Corydon-w | grep is just search |
| 16:25.18 | Lurchtoke | ok |
| 16:25.19 | mercestes | then do a chown asterisk:asterisk extensions.conf |
| 16:25.27 | Lurchtoke | ok... |
| 16:25.30 | mercestes | and a nice asterisk -rx 'extensions reload' |
| 16:25.38 | Lurchtoke | ill be back in a few minutes...gotta run over to the store.... |
| 16:25.38 | ThoMe | Corydon-w: hi |
| 16:25.39 | mercestes | That's the *bad* way to do it |
| 16:25.50 | jart | what's up my minions |
| 16:25.54 | mercestes | the *right* way would be to edit extensions.conf...find the old file name, and change it, manually to the new file name. |
| 16:26.04 | mercestes | bu tmy way allows you to do it without having a clue what's really happening. |
| 16:26.04 | ThoMe | Corydon-w: say. i have a script http://tm.muc.de/anruf.phps . with this script i can call a phonenumber about a webpage |
| 16:26.13 | ThoMe | Corydon-w: how i can get the channel |
| 16:26.17 | ThoMe | servetux*CLI> show channels |
| 16:26.17 | ThoMe | Channel Location State Application(Data) |
| 16:26.17 | ThoMe | SIP/26-081eb638 (None) Ringing (None) |
| 16:26.19 | mercestes | and gives you an "oops I screwed up" life line. |
| 16:26.22 | ThoMe | example: SIP/26-081eb638 |
| 16:26.42 | ThoMe | and how i can hang up this channel via my php script http://tm.muc.de/anruf.phps |
| 16:26.54 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:26.57 | PakiPenguin | evening |
| 16:27.38 | jart | top of the pengiun to ya |
| 16:27.40 | Corydon-w | ThoMe: you should read the AGI documentation |
| 16:27.48 | ThoMe | Corydon-w: ok. danke |
| 16:27.51 | ThoMe | aaeh thank you |
| 16:28.12 | nosbig | Thanks, guys.... |
| 16:28.15 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 16:28.17 | nosbig | Attended transfer worked... |
| 16:28.33 | b11d | anyone ever deal with a "config file error 0x0" on a Polycom 501? |
| 16:29.37 | mercestes | b11d: Upgrade your bootrom |
| 16:29.44 | mercestes | b11d: If gives better error messages. |
| 16:29.57 | mercestes | b11d: And let me guess, you *did* upgrade your sip firmware...but you used yoru old configs, right?? |
| 16:30.05 | b11d | no |
| 16:30.09 | mercestes | oh. |
| 16:30.17 | mercestes | it's basically a syntax error in yoru file. |
| 16:30.29 | b11d | new phone.. plugged in.. contacted provisioning server.. upgraded bootrom and sip to standard versions.. |
| 16:30.35 | b11d | reads config file.. dies. |
| 16:30.37 | b11d | reboots. |
| 16:30.39 | b11d | etc.. |
| 16:30.46 | amdtech | FUN |
| 16:30.54 | b11d | it cant be a syntax error.. the config file is the SAME as on another phone of the same model & versions |
| 16:31.06 | mercestes | b11d: permissions? |
| 16:31.09 | amdtech | same firmware too? |
| 16:31.14 | b11d | yes |
| 16:31.31 | amdtech | try forcing the phone to do a factory reset |
| 16:31.35 | PakiPenguin | :) |
| 16:32.29 | b11d | good call |
| 16:32.58 | b11d | hey paki |
| 16:33.01 | b11d | good to see you |
| 16:33.04 | PakiPenguin | hey b11d |
| 16:33.05 | b11d | permissions are all good |
| 16:33.07 | PakiPenguin | sup? |
| 16:33.15 | b11d | this phone issue :) |
| 16:33.18 | b11d | trying to grind it out |
| 16:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.24 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.30 | b11d | i'll try the factory reset.. |
| 16:33.49 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 16:36.02 | *** join/#asterisk Gankhuu (n=luken@72-166-51-162.dia.static.qwest.net) |
| 16:36.53 | b11d | ouhh |
| 16:37.00 | b11d | factory reset, now I get error 0x4020 |
| 16:37.53 | b11d | neato.. on the right track |
| 16:38.04 | robl^ | b11d: try reverting to default config files.. don't use any encrypted ones. |
| 16:38.17 | mercestes | b11d: awesome. Check your permissions and your ftp/tftp/sftp/http/scp/http-get/zodiac/fung-shuy/i-ching/karmanic balance settings. |
| 16:38.41 | mercestes | b11d: And what robl^ said, use the default config files that came with yoru sip firmware release. |
| 16:38.46 | robl^ | b11d: and check your faily fiber intake... |
| 16:38.49 | b11d | yeah already doing that |
| 16:39.00 | robl^ | s/faily/daily |
| 16:39.00 | b11d | faily fiber is the best |
| 16:39.02 | b11d | oh |
| 16:39.03 | b11d | :_) |
| 16:41.02 | PakiPenguin | :) |
| 16:41.03 | PakiPenguin | hahah |
| 16:43.02 | robl^ | b11d: if you were using the encryption tool for teh config files, then reverted to factory default, you wiped out the encryption ey. So you need to use the default config files.. Re-add the key. Then you can use the encrypted files again |
| 16:43.07 | b11d | this is not encrypted |
| 16:43.09 | b11d | this is clear text |
| 16:43.26 | b11d | this is just a new phone, out of the box, trying to get it 'up' using the traditional means.. |
| 16:43.28 | *** join/#asterisk Egonis (n=root@70.54.211.179) |
| 16:43.49 | b11d | back to 0x0 |
| 16:43.50 | robl^ | b11d: polyViagra? |
| 16:43.54 | b11d | Diagara |
| 16:44.04 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 16:44.09 | Egonis | I just completed an * 1.4 install fresh, w/ asterisk-gui from svn -- when running install, it says it cannot find any analogue ports, however when I run 'zap show channels' I see 4 of them as per what I truly have. Is there something basic I am missing? |
| 16:45.21 | mercestes | Egonis: Do they show as up? |
| 16:45.38 | mercestes | b11d: what v. of bootrom/firmware are you using? |
| 16:46.19 | Egonis | <PROTECTED> |
| 16:46.21 | Egonis | that's a sample |
| 16:46.47 | Egonis | mercestes: if I run the 'zapscan' util included, it says 'Skipping Zap Scanning' |
| 16:46.50 | *** join/#asterisk ArkyLady (i=ArkyLady@dsl-205-234-115-182.madisoncounty.net) |
| 16:48.23 | b11d | bootrom 3.2.2 and sip 2.0.1 |
| 16:48.26 | mercestes | Egonis: that dosen't look like a workign analog channel. That looks like a pseudo channel from ztdummy or something. |
| 16:48.43 | mercestes | Egonis: Did you modprobe your card drivers? |
| 16:49.10 | mercestes | b11d: you should be getting better error messages than that. check your networking settings under setup and check yoru net connectivity. I don't think your even getting your config files. |
| 16:49.22 | mercestes | b11d: In fact...if you look, I'll bet your not running the firmware you are thinking youare either. |
| 16:49.22 | b11d | yes. i am. |
| 16:49.25 | b11d | i can see the ftp logs |
| 16:49.32 | mercestes | and you get an ftp success? |
| 16:49.41 | b11d | i can assure you, network connectivity is valid.. and the ftp login & downloads are a success |
| 16:49.50 | b11d | it even uploads the -boot.log file properly |
| 16:50.12 | mercestes | got another phone in that batch? |
| 16:50.25 | b11d | yeah im getting close to breaking it out.. |
| 16:50.34 | b11d | trying one more thing here.. disabled dhcp, trying a static IP.. |
| 16:50.38 | mercestes | yea, bust out the phone, and just rename the mac.cfg to the new phone's mac addy |
| 16:50.50 | mercestes | dhcp won't be the issue ifyou are showing an ftp success |
| 16:51.01 | b11d | i know, but I want to be able to get to the phones status & advanced menus |
| 16:51.01 | Egonis | mercestes: I use a sangoma A200, that could be the issue, I copied my zaptel.conf and zapata.conf from that, could this be the issue? |
| 16:51.07 | amdtech | yeah, but if there's something wrong with the mac.cfg you'll have problems |
| 16:51.08 | mercestes | it could be the flash memory in that phone went RO on you or something. |
| 16:51.10 | b11d | whereas right now I cannot even get to that point |
| 16:51.16 | b11d | the mac.cfg is correct |
| 16:51.25 | mercestes | amdtech: exactly..:) |
| 16:51.34 | amdtech | try taking the mac.cfg out totally |
| 16:51.46 | mercestes | nooo.....if it's the config files I want it to break again on a new phone...:( |
| 16:51.47 | *** join/#asterisk deb_user (n=debian_u@albuquerque.agroinnovations.com) |
| 16:51.48 | amdtech | rename it so the phone doesn't grab it, make ABSOLUTE sure it's not that file :) |
| 16:51.49 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 16:51.52 | b11d | i'll put that in the queue of things to try |
| 16:51.53 | deb_user | hello all |
| 16:52.04 | ThoMe | Corydon-w: how i can get the active channel? |
| 16:52.08 | ThoMe | getparm? |
| 16:52.26 | mercestes | Egonis: I have no clue what you just said but that you said "copy" zaptel.conf and zapata.conf that gives me pause for concern. |
| 16:52.26 | deb_user | is it possible to set up a voicemail box to send to multiple email addresses WITHOUT resorting to aliases? |
| 16:52.43 | Egonis | mercestes: I will review my configs, thank you. :) |
| 16:53.46 | mercestes | =/ Your welcome. |
| 16:54.12 | b11d | deb_user.. what about to an email account with rules already setup to forward incoming email to multiple recipients? |
| 16:54.13 | *** join/#asterisk MonkeyHugs (n=jojo@63.149.122.93) |
| 16:55.59 | deb_user | b11d: I'm thinking of using mailman |
| 16:56.11 | deb_user | I already have it all configured, would just have to add a new mailing list |
| 16:56.24 | Carp1 | chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach '8002081258@' |
| 16:56.29 | Carp1 | I was working on this error all day yesterday with no luck |
| 16:56.30 | amdtech | btw, yes, you can deb_user |
| 16:56.33 | amdtech | without mailing lists |
| 16:56.55 | deb_user | amdtech: could you elaborate on the configuration please? (running the latest 1.4.0 btw) |
| 16:57.13 | Carp1 | any ideas? |
| 16:57.31 | deb_user | amdtech: it would be cool, because setting up a mailing list for two users seems like an overkill |
| 16:57.54 | in-pt | can anyone tell me if there exist a channel for cdrtool |
| 16:58.00 | deb_user | and having to manually insert aliases into my sql database seems like a pain |
| 16:58.02 | mercestes | Carp1: It looks like an allow/deny issue or an authentication issue from a remote sip connection. |
| 16:58.05 | amdtech | there's two ways i think... one would be to set up two totally different mailboxes for each user, and doing Voicemail(b1&b2) |
| 16:58.24 | amdtech | that way you can use the mailboxes separately if necessary |
| 16:58.29 | Carp1 | Its via IAX Connection |
| 16:58.37 | deb_user | amdtech: that's a little more than I need |
| 16:58.44 | Carp1 | I get this eooe when I try to call into the box |
| 16:58.48 | mercestes | Carp1: s/sip/iax2/g then. |
| 16:58.57 | amdtech | easier than setting up a mailing list for two users ;) |
| 16:59.04 | deb_user | i just want the emails to get sent to two users in case one isn't in the office |
| 16:59.10 | Carp1 | I dont understand......Im pretty new at this. |
| 16:59.15 | mercestes | amdtech: I think he wants to use one voiecmail box and send email notification to multiple users. |
| 16:59.19 | Carp1 | Worked all day on the erro yesterday |
| 16:59.24 | deb_user | mercestes: exactly |
| 16:59.25 | mercestes | not use multiple email address boxes. |
| 16:59.33 | mercestes | tho..that would work but it would leave duplicates |
| 16:59.40 | mercestes | unless you did some magic symlinking |
| 16:59.46 | deb_user | mercestes: duplicates? |
| 16:59.48 | mercestes | but that could cause breakage. |
| 16:59.55 | amdtech | you can just have it delete out of the second box automatically |
| 17:00.02 | deb_user | mercestes: if you send the vm to two different boxes |
| 17:00.06 | amdtech | and no, it won't cause breakage unless both users log in at the same time |
| 17:00.13 | mercestes | That's what I was thinking..:) |
| 17:00.19 | mercestes | that would be breakage |
| 17:00.33 | tzanger | wtf |
| 17:00.40 | *** part/#asterisk Gankhuu (n=luken@72-166-51-162.dia.static.qwest.net) |
| 17:00.43 | deb_user | well, on the flip side, the advantage of setting up a mailing list (which is absurdly easy to maintain, btw), is I can add users really easily |
| 17:00.44 | tzanger | http://www.niagarasafetyvillage.on.ca/asterisk_conference_call.html |
| 17:00.47 | amdtech | so force the second box to auto delete after emailing and you've got it :) |
| 17:00.57 | *** join/#asterisk bkw__ (n=brian@m115e36d0.tmodns.net) |
| 17:01.06 | amdtech | it's all in how you want to program it :) |
| 17:01.08 | Carp1 | Here are my config files: |
| 17:01.08 | Carp1 | http://pastebin.ca/332742 |
| 17:01.08 | Carp1 | http://pastebin.ca/332739 |
| 17:01.20 | amdtech | i prefer to let asterisk manage it, not the mailserver |
| 17:01.24 | deb_user | amdtech: I was hoping it was already a native feature in * |
| 17:01.28 | tzanger | has anyone done any largeish (50-150 user) conferences? Any caveats or issues that are known? |
| 17:01.49 | deb_user | you know, just change the syntax a little in voicemail.conf and it does it automagically |
| 17:02.01 | deb_user | i saw on the mailing lists that its in the works, but not quite there yet |
| 17:02.03 | amdtech | if you're using a database you can |
| 17:02.07 | tzanger | I'm lucky in the sense that most callers will be coming in from a pair of PRIs but there will be the handful of SIP participants too |
| 17:02.35 | mercestes | tzanger: well, one, last I checked, PRI will only handle 23 callers. |
| 17:02.43 | mercestes | tzanger: So your limit is 48 right there. |
| 17:03.02 | mercestes | unless you have one of those new fangled 50 channel PRI lines or something. |
| 17:03.03 | tzanger | mercestes: as I said "pair of PRIs" but obviously more clients means either more SIP or more PRI spans |
| 17:04.04 | mercestes | tzanger: yea. I would honestly test first. I've run up to 10-20 successfully, but I'm thinking 100 might present a bit of a load. |
| 17:04.17 | tzanger | yeah |
| 17:04.35 | Carp1 | I think its trying to send to the wrong context.....i dont get it. |
| 17:04.36 | tzanger | anthony created an alternative conference bridge that used a very cool idea to avoid mixing audio streams |
| 17:04.44 | tzanger | he simply played a frame from each of the talkers, in turn |
| 17:05.07 | tzanger | I think he had some logic in there too to detect if a tlaker was actually talking and therefore play more of his frames than others, but anyway |
| 17:05.48 | mercestes | not sure. It can be done and I would eliminate transcoding but I couldn't sleep at night if I told you to go ahead with it and not warned you to test first |
| 17:05.55 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:05.55 | tzanger | heh |
| 17:06.02 | tzanger | I'm not new to asterisk by any means, just conferencing |
| 17:06.07 | Qwell[] | http://www.amcommproservices.com/owned.jpg |
| 17:06.23 | tzanger | Qwell[]: yeah I saw that :-) |
| 17:06.25 | TripleFFFF | a |
| 17:06.34 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:06.34 | amdtech | Qwell[]: that's awesome |
| 17:07.16 | mercestes | lol |
| 17:07.19 | Carp1 | well, thanks for all the help guys :-\ |
| 17:08.56 | tzanger | it's teh shido |
| 17:09.30 | *** join/#asterisk oej (i=olle@nat/digium/x-11127d077a4206f3) |
| 17:14.36 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:16.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 17:16.47 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 17:19.14 | wnfaknd | anyone familiar with asterisk/vicidial, particularly its setup |
| 17:19.21 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 17:19.35 | wnfaknd | and how to get it off the ground |
| 17:19.56 | mercestes | wnfaknd: vicidial? |
| 17:20.41 | mercestes | yea, you'd have to go to the astGUIclient ppl |
| 17:22.10 | mercestes | that may be #asterisk-gui but....I dunno |
| 17:22.17 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:22.37 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 17:23.47 | angryuser | what is the default transfer key for asterisk? *8+extension? |
| 17:24.24 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 17:25.19 | mercestes | I thought it was #. What does features.conf say? |
| 17:26.45 | angryuser | #1 for blind |
| 17:27.00 | tzafrir_laptop | #asterisk-gui is something different |
| 17:33.29 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:33.29 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:34.03 | angryuser | but when i do #1+ extension nothing happens;) i have tw option enable on dial |
| 17:36.03 | mercestes | angryuser: Try wWtT adn then try it |
| 17:36.37 | Qwell[] | tT is good for hdparm, but bad for app_dial |
| 17:36.54 | *** join/#asterisk saftsack (n=w@p54A7FB4C.dip.t-dialin.net) |
| 17:37.24 | saftsack | hi, how is the telephone book number lookup service called? |
| 17:40.11 | CoffeeIV | my compile of asterisk (1.2 branch from svn) goes through the channels sub-directory without compiling chan_zap.c or making chan_zap.so. Why might that be ? |
| 17:41.02 | mercestes | Qwell[]: Just for testing. I don' tknow if he's calling or answering. |
| 17:41.08 | mercestes | ....I guess wW is for recording. =/ |
| 17:41.35 | Qwell[] | CoffeeIV: install zaptel |
| 17:42.00 | ThoMe | Qwell[]: hello |
| 17:42.08 | CoffeeIV | Qwell: I did, and zttool shows a card with no alarm |
| 17:42.10 | ThoMe | Qwell[]: how i can with phpagi get the current channel? |
| 17:42.33 | *** join/#asterisk Egonis (n=root@70.54.211.179) |
| 17:42.59 | Qwell[] | CoffeeIV: You installed zaptel 1.2? |
| 17:43.08 | Egonis | Just upgraded to * 1.4, and am trying to register all of my phones, I get 'Really destroying SIP dialog 'blah' Method: REGISTER, but 'sip show registry' shows no clients |
| 17:43.59 | CoffeeIV | Qwell[]: I installed zaptel from the svn trunk; the 1.2 branch put a zaptel.h in the system that didn't define some struct or something, so asterisk (1.2 branch) would not compile |
| 17:44.07 | Qwell[] | CoffeeIV: you can't do that |
| 17:44.24 | Qwell[] | gotta use the same branch |
| 17:44.52 | CoffeeIV | Qwell[]: OK, let me go back to 1.2 and try again |
| 17:44.59 | ThoMe | Qwell[]: can u help me? |
| 17:45.31 | Qwell[] | ThoMe: no |
| 17:45.40 | ThoMe | Qwell[]: sorry. thx. |
| 17:45.48 | Qwell[] | but I'm sure the other kind folks here would be more than happy to help |
| 17:46.19 | ThoMe | folk: how i can get the current channel? |
| 17:46.30 | ThoMe | with php agi ( asterisk console?) ? |
| 17:46.41 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 17:46.55 | ThoMe | Chris-NB: hi |
| 17:47.02 | Chris-NB | hio |
| 17:47.04 | saftsack | can i resolv an incoming call automatically by doing a lookup in a database? |
| 17:47.31 | ThoMe | Chris-NB: sag, kennst du dich mit asterisk a bissl aus? |
| 17:47.34 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:47.56 | Chris-NB | ThoMe, jep, I think so. but depends on the problem |
| 17:48.04 | ThoMe | Chris-NB: mh ok. |
| 17:48.13 | ThoMe | Chris-NB: wie kann ich denn den momentan channel herrausfinden? |
| 17:48.20 | ThoMe | der bentzt wird.. |
| 17:48.25 | ThoMe | via asterisk console |
| 17:48.31 | Chris-NB | show channels |
| 17:48.33 | ThoMe | also gibts irgend wie ein response vom asterisk? |
| 17:48.42 | Chris-NB | iax/zap/misdn show channels |
| 17:48.44 | ThoMe | Chris-NB: naja. ich hab solch php lib die via php agi zugreift. |
| 17:48.49 | saftsack | oder halt sip |
| 17:48.54 | saftsack | oder? |
| 17:48.59 | ThoMe | hm. weis ned. |
| 17:49.08 | ThoMe | ich brauch eine automatisch rueckmeldung vom asterisk geht das ned? |
| 17:49.15 | ThoMe | das man sagen kann get_channelid? |
| 17:49.27 | ThoMe | ich will via script sagen koennen hangip(channel) |
| 17:49.35 | ThoMe | mit dem hier http://tm.muc.de/anruf.phps waehel ich bsp. |
| 17:49.42 | b11d | wass est los! |
| 17:49.54 | b11d | doh |
| 17:49.56 | b11d | ist |
| 17:50.02 | Chris-NB | ThoMe, show channels shows all active channels |
| 17:50.12 | mercestes | urgen durgen virgin burgen durgen wurgen. |
| 17:50.15 | ThoMe | Chris-NB: hm ja. but ich brauch den currenten! |
| 17:50.19 | b11d | hahaha mercy |
| 17:50.22 | Qwell[] | mercestes: nobody here speaks dutch |
| 17:50.29 | Qwell[] | </trolling> |
| 17:50.45 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:51.25 | Chris-NB | Qwell, that's not dutch |
| 17:51.35 | Chris-NB | Qwell, thats german : D |
| 17:51.55 | mercestes | dutch, german, same diff. Dutch is german spoken with emotion. |
| 17:52.26 | b11d | it might be yogging.. it might have a soft j.. |
| 17:52.32 | Chris-NB | mercestes, deff not. dutch is mix between german and english |
| 17:52.50 | b11d | quiet.. must not think.. of.. amsterdam. |
| 17:52.57 | Chris-NB | *hrhr |
| 17:53.02 | ThoMe | Chris-NB: ich brauch ein response wie diesen: |
| 17:53.02 | ThoMe | servetux*CLI> show channels |
| 17:53.02 | ThoMe | Channel Location State Application(Data) |
| 17:53.02 | ThoMe | SIP/26-081fa110 |
| 17:53.05 | ThoMe | aber schon beim waehlen. |
| 17:53.11 | ThoMe | geht das ned? |
| 17:53.15 | ThoMe | bzw das: -081fa110 |
| 17:55.17 | ThoMe | Chris-NB: irgend eine idee? |
| 17:57.37 | Chris-NB | ThoMe, qry |
| 17:57.49 | ThoMe | Chris-NB: hm? |
| 17:58.03 | ThoMe | ah |
| 17:59.21 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:02.22 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 18:04.22 | *** join/#asterisk jacekowski (n=jacekows@aaot198.neoplus.adsl.tpnet.pl) |
| 18:04.24 | Egonis | exit |
| 18:10.31 | wnfaknd | mercestes, thanks for the info |
| 18:11.02 | wnfaknd | mercestes, btw, you familiar with vicidial? |
| 18:11.38 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 18:12.02 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 18:12.28 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 18:18.55 | *** join/#asterisk candyman50 (n=x@pool69-59-255-15.kewr1.s.vonagenetworks.net) |
| 18:19.47 | candyman50 | anyone know if Asterisk 1.4 can play MP3s natively via the Playback() command in extensions.conf? |
| 18:19.57 | Qwell[] | candyman50: if you install asterisk-addons |
| 18:20.20 | candyman50 | only with the addons? |
| 18:20.39 | candyman50 | I saw something about "native support" |
| 18:20.46 | candyman50 | was that just for MOH? |
| 18:20.54 | Qwell[] | native if you install -addons |
| 18:21.24 | candyman50 | :) thanks |
| 18:23.43 | Lurchtoke | hey fellas....another question |
| 18:24.28 | Lurchtoke | what dir has all my extensions.conf. audio files? |
| 18:25.22 | Lurchtoke | i need to browse them...to find a particular audio file (our company address recording) so I can replace it with another files I have in a mailbox |
| 18:25.50 | Lurchtoke | im a semi-noob at linux and *...so I just need a puch in the right direction |
| 18:25.58 | Lurchtoke | err push |
| 18:26.19 | candyman50 | usually it's in /var/lib/asterisk/sounds, but you can check /etc/asterisk/asterisk.conf |
| 18:26.24 | candyman50 | look for astdatadir => /prod/asterisk/var/lib/asterisk |
| 18:26.38 | candyman50 | that's mine (non-standard place) |
| 18:27.01 | b11d | got that strange polycom 501 issue worked out.. |
| 18:27.14 | b11d | did was was suggested, formatted the fs, and updated it with the default configs.. |
| 18:27.21 | candyman50 | you could always do find / -name "*gsm" -print :) |
| 18:27.23 | b11d | replaced the config's with the ones i wanted to, rebooted, and works fine. |
| 18:28.05 | Lurchtoke | ok |
| 18:28.11 | Lurchtoke | brb |
| 18:29.54 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 18:30.38 | *** join/#asterisk bkw__ (n=brian@m0a5e36d0.tmodns.net) |
| 18:32.20 | *** join/#asterisk topping (n=topping@204.152.96.50) |
| 18:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:35.22 | Bobthehunter | SIP/2.0 503 Unavailable isnt that a congestion ? |
| 18:35.33 | Carp1 | ANyone use NuFone? |
| 18:36.18 | amdtech | where'd you get the phone from b11d |
| 18:37.01 | Lurchtoke | <candyman50> its in another dir i guess |
| 18:37.04 | Lurchtoke | bummer |
| 18:37.44 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-54-119.red.bezeqint.net) |
| 18:41.10 | *** part/#asterisk candyman50 (n=x@pool69-59-255-15.kewr1.s.vonagenetworks.net) |
| 18:41.34 | b11d | Qwest |
| 18:43.34 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 18:48.04 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 18:48.56 | *** join/#asterisk degadar (n=chatzill@217.14.137.77) |
| 18:50.24 | CoffeeIV | I just checked out asterisk 1.2 branch from svn, and when I compiled it I got the error "codec_zap.c:147: error: dereferencing pointer to incomplete type" -- anyone recognize that and know how to fix it ? I am using the zaptel from the 1.2 branch also |
| 18:50.31 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:50.42 | *** join/#asterisk wannabeastguru (n=root@ti100710a080-4763.bb.online.no) |
| 18:51.04 | b11d | well thats a compilation error alright |
| 18:51.09 | b11d | what OS? |
| 18:51.21 | b11d | hrm.. actually.. do you even have zaptel installed yet? |
| 18:52.09 | CoffeeIV | blld: I do have zaptel installed, and I can do ztcfg -vvvv and it looks ok, sees my card |
| 18:52.28 | CoffeeIV | the OS is linux (Fedora Core 4) |
| 18:53.07 | aptura | one of the few linux versions I do like CoffeeIV |
| 18:53.23 | b11d | hrm.. |
| 18:53.25 | b11d | i dunno then mang :) |
| 18:53.27 | b11d | sorry |
| 18:53.55 | degadar | is the zaptel the right version? Obvious questions - sorry if you're one step ahead of me CoffeeIV |
| 18:53.55 | CoffeeIV | I think the problem is that zt_transcode_header should be defined in a .h in /usr/include/linux/zaptel.h or somewhere, and it is not |
| 18:54.08 | b11d | yeah you should start messing with the source.. |
| 18:54.10 | b11d | good start |
| 18:54.10 | b11d | :) |
| 18:54.14 | b11d | haha |
| 18:54.14 | CoffeeIV | I installed the zaptel from the 1.2 svn branch |
| 18:54.16 | SomeOne1 | i <3 cheeseballs |
| 18:54.36 | b11d | have you searched the bug reports for a similiar entry CoffeeIV? |
| 18:55.04 | CoffeeIV | I searched google for that compile error -- should I look in bugs.digium.com or something ? |
| 18:56.27 | b11d | yeah |
| 18:56.39 | b11d | im not saying there IS one in there.. but there might be.. |
| 18:56.59 | *** join/#asterisk shidan (n=chatzill@206.191.69.226) |
| 18:57.16 | b11d | is there any active development on the 1.2 branch anymore? Could you have checked out an unstable version? |
| 18:57.27 | b11d | did you try building from the source tarballs on asterisk.org ? |
| 18:57.32 | CoffeeIV | blld: I don't see anything like that in there |
| 18:57.34 | Carp1 | ANyone use NuFone? |
| 18:57.50 | b11d | i hear a lot of complaints about nufone, so yeah.. people must use their service in here.. |
| 18:57.55 | CoffeeIV | in bugs.digium.com that is . . . I can try the tarballs from asterisk.org, let me see if that works |
| 18:58.13 | Carp1 | yeah, like it wont work! |
| 18:58.23 | Carp1 | chan_iax2.c:6754 socket_process: Rejected connect attempt from 66.225.202.80, who was trying to reach '8002081258@' |
| 18:58.34 | Carp1 | I get that error on CLI when I call into the box |
| 18:59.19 | b11d | do you need a 1 before the 800? |
| 18:59.25 | b11d | are you stripping it off by accident? |
| 18:59.40 | Carp1 | no |
| 18:59.42 | Carp1 | no 1 |
| 18:59.44 | b11d | ok |
| 19:01.57 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 19:01.57 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 19:02.04 | Carp1 | Does anyone know of a reliable IAX provider? |
| 19:02.42 | Lurchtoke | can someone help me for a minute? |
| 19:03.09 | b11d | Lurch, you just have to ask your question.. |
| 19:03.19 | b11d | if you wanna toke.. meet me out back in 15 mins |
| 19:03.20 | b11d | :) |
| 19:03.59 | Lurchtoke | hehe |
| 19:04.12 | Lurchtoke | I actually need a few momnets of time |
| 19:04.37 | Lurchtoke | can someone help me navigate and change an audio file?? |
| 19:04.55 | SomeOne1 | CHEESE BALLS |
| 19:05.20 | Lurchtoke | one sec |
| 19:05.38 | sevard | Lurchtoke: transcoding or cutting, splicing, and other sorts of editing |
| 19:05.40 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 19:05.57 | Lurchtoke | easier than that |
| 19:06.12 | Lurchtoke | I just need help navigating through my * box and finding a file to replace |
| 19:06.26 | Lurchtoke | the audio file that lets our customers know our address |
| 19:06.47 | mercestes | Carp1: teliax |
| 19:08.09 | *** join/#asterisk bkw__ (n=brian@m0e5e36d0.tmodns.net) |
| 19:15.49 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 19:17.08 | perd | ~telephony101 |
| 19:17.20 | perd | damn how does that thing work |
| 19:17.20 | *** join/#asterisk juice (n=juice@mo-76-0-47-34.dhcp.embarqhsd.net) |
| 19:17.21 | perd | ~telephony_101 |
| 19:18.20 | *** join/#asterisk arny1988 (n=Miranda@p5481BA3E.dip0.t-ipconnect.de) |
| 19:18.38 | wunderkin | ~terd |
| 19:18.50 | Bobthehunter | wahts proper way to send to a specific context on a dial ? |
| 19:18.59 | Bobthehunter | dial ( sip/CONTEXT@HOST/DID ?) |
| 19:19.18 | wunderkin | don't think you can with sip |
| 19:19.25 | wunderkin | you have to specify that on the other end |
| 19:19.55 | Bobthehunter | shit |
| 19:20.07 | perd | use callerid |
| 19:20.08 | Bobthehunter | ok well im seperating inbound/outbound oon 2 internal trunks..sip |
| 19:20.22 | Bobthehunter | so BOX A has peer/user and Box B same.. |
| 19:20.35 | Bobthehunter | so i can outbound from A to inbound in B |
| 19:20.39 | Bobthehunter | and vice versa |
| 19:20.48 | arny1988 | hello.... i need a german tutorial for asterisk - anyone? |
| 19:21.03 | wunderkin | volkswagon weinerschnitzel |
| 19:21.13 | Bobthehunter | but its never hitting the right context from B to A.. ex pstn -> B send to B-out.. to A -in |
| 19:21.57 | Bobthehunter | basicaly wich is user and wich is peer.. |
| 19:22.37 | Bobthehunter | from box A point of view........ it has box B user and peer right ? so it call from B going to A........it should match in type=user or type=peer |
| 19:22.45 | Bobthehunter | on Box A |
| 19:23.14 | Bobthehunter | or said differnetly.. is B a user or peer for A .. when call flow is B to -> A |
| 19:23.20 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 19:23.56 | arny1988 | hello.... i need a german tutorial for asterisk - anyone? |
| 19:24.41 | mercestes | arny1988: try #asterisk-de |
| 19:24.55 | mercestes | bobthehunter: you do realize that there is no such thing as a sip trunk, right? |
| 19:25.14 | Bobthehunter | well im not trunking |
| 19:25.18 | Bobthehunter | but you get the point |
| 19:25.32 | mercestes | Ok, in sip.conf, when you specify the incoming user, you set the context. |
| 19:25.35 | Bobthehunter | i wanna isolate calls from A to B and B to A in differnet conexts.. EX inbound outbound for each.. |
| 19:25.45 | Bobthehunter | i did |
| 19:25.46 | arny1988 | mercestes there is no one :( |
| 19:25.46 | mercestes | and all the appropriate pattern matching must be done in that context that you specify |
| 19:26.07 | mercestes | arny1988: Oh...then sorry, there is no one who can help you in german. |
| 19:26.09 | Bobthehunter | but PSTN -> B -> A ( on A i always see it connecting as outbound conext wich is a peer deficnition not user) |
| 19:26.29 | *** join/#asterisk pasqu (n=pasqu@ip-156-18-static.adsl.intratec.it) |
| 19:27.21 | mercestes | Bobthehunter: Do you ever see a user definition? |
| 19:27.38 | Bobthehunter | hmm no wahts it for ? lol |
| 19:27.56 | pasqu | hi all, is there anyone that have some experiences with asterisk cluster? |
| 19:28.19 | *** join/#asterisk oys (n=eoyslebo@84.208.78.84) |
| 19:29.33 | mercestes | BobTheHunter: The answer to your problem may be right there. Try changing the peer context to whatever you want it to be. |
| 19:30.44 | Lurchtoke | mercestes...can you assist me for a second? |
| 19:31.32 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:32.00 | Lurchtoke | anyone? |
| 19:32.24 | pasqu | for what? |
| 19:32.42 | Lurchtoke | i need help navigating and changing an audio file out |
| 19:32.52 | Lurchtoke | with another |
| 19:33.27 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:33.27 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:34.31 | Lurchtoke | just 5 minutes peeps |
| 19:34.33 | Lurchtoke | <PROTECTED> |
| 19:35.03 | oys | I am having trouble compiling asterisk with the new chan_cellphone channel. anyone played with this yet? |
| 19:35.14 | oys | bluetooth works, but when compiling asterisk from svn after applying the patches, make menuconfig tells me this channel depends on <depend>bluetooth</depend> |
| 19:35.19 | pasqu | emh...for change an audio file go in the sounds directory and put your new file |
| 19:35.20 | oys | so a point in the right direction (i.e how to tell ./configure that I have bluetooth in /usr/include/bluetooth/bluetooth.h) would be greatly appreciated :) |
| 19:35.23 | pasqu | where is the problem |
| 19:35.33 | pasqu | if I understand the question |
| 19:35.38 | Lurchtoke | no problem |
| 19:36.02 | Lurchtoke | just trying to change a file |
| 19:36.30 | Lurchtoke | I need basic help identifying an audio file thats played in a part of our phone menu |
| 19:36.37 | Lurchtoke | and replaceing it with a new one |
| 19:36.43 | sevard | Lurchtoke: paste your god damned dialplan |
| 19:37.12 | Lurchtoke | fuck off sevard....go try to charge someone else in a private message |
| 19:37.33 | Lurchtoke | this is a free channel for people neededin g help...not a place for you to make money |
| 19:37.41 | sevard | i'm not talking about that dillhow |
| 19:37.50 | sevard | i said paste your damned dialplan in pastebin |
| 19:38.20 | Lurchtoke | anyways |
| 19:38.43 | b11d | christ almighty |
| 19:38.49 | b11d | Lurch.. go smoke a fatty and relax eh |
| 19:38.53 | b11d | :) |
| 19:39.19 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 19:39.35 | b11d | make sure you throw a crutch in the end |
| 19:39.36 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 19:39.38 | b11d | and no tobacco |
| 19:39.38 | b11d | :) |
| 19:39.53 | Lurchtoke | im relaxed...i just felt insulted when I ask for help and someone msg's me offering to help for a charge...when this is a free channel |
| 19:40.06 | sevard | nobody said this is a 'free channel' |
| 19:40.13 | b11d | yeah where exactly did sevard ask for money? |
| 19:40.32 | Lurchtoke | ill drop it....ask him yourself... |
| 19:40.39 | b11d | hey sevard, i need a hand with this.. |
| 19:40.39 | mercestes | In a PM obviously |
| 19:40.41 | sevard | you can't blame a guy for trying to score some burger monies |
| 19:40.46 | b11d | hahahha |
| 19:40.53 | b11d | ahh you get what you pay for :) |
| 19:41.03 | sevard | but then you get all pissy about me doing it for free |
| 19:41.03 | mercestes | Guys. Chill out, #asterisk is shit enough without everyone scaring off the n00bs. |
| 19:41.08 | sevard | heh |
| 19:41.09 | b11d | yeah |
| 19:41.22 | b11d | can we start a #realasterisk |
| 19:43.00 | tzanger | heh |
| 19:43.13 | Carp1 | i am going to shoot myself if i cant get this working! |
| 19:43.25 | CoffeeIV | b11d and Qwell[]: I downloaded tarballs of 1.2 zaptel and asterisk and got things working. perhaps there is something wrong in current svn branch for 1.2. Thanks for the help. |
| 19:43.43 | pasqu | b11d, is a good idea also for the ml |
| 19:43.50 | tzanger | Carp1: what |
| 19:43.56 | mercestes | Carp1: try Teliax. Otherwise, IAX is a bit.....tricky to get installed. It's syntax is far less than natural. Try google asterisk wiki iax |
| 19:43.59 | mercestes | and get THE BOOK |
| 19:44.03 | mercestes | ~book |
| 19:44.05 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:44.24 | mercestes | Carp1: My advice. Use the examples verbatim, and just change the names as you require just to get it working...then tinker from there. |
| 19:44.52 | AF-Slash | Anyone know of any reason to get dCAP certified? |
| 19:45.01 | oys | no-one try the bluetooth stuff (using a cellphone as a FXS)? |
| 19:45.02 | b11d | anytime |
| 19:45.07 | AF-Slash | or is it pretty much pointless at this time |
| 19:45.15 | b11d | im glad that worked CoffeeIV |
| 19:45.19 | Carp1 | verbatim? |
| 19:45.23 | b11d | 1.2 must still be being developed.. |
| 19:45.40 | b11d | btw, you can totally deploy asterisk and do it well without "THE BOOK" |
| 19:45.53 | b11d | people and their crutches :) |
| 19:45.56 | b11d | oh wait.. irc is a crutch :) |
| 19:46.01 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:46.31 | Carp1 | i copied the setting from my nufone account |
| 19:47.18 | Carp1 | to the config files |
| 19:48.44 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:49.06 | pasqu | OwOw, nobody try the get asterisk working with heartbeat, ultramonkey, dundi network ecc ecc? |
| 19:49.27 | mercestes | Carp1: If you are trying to hook to a provider than the provider should be providing directions and technical support. |
| 19:50.27 | Carp1 | i know |
| 19:50.40 | Carp1 | NuFone wont email me back unless its a simple question |
| 19:51.04 | Carp1 | I will goto another provider that has IAX and allows me to get a toll free number. |
| 19:51.05 | mercestes | then might I suggest a better provider? |
| 19:51.08 | mercestes | or a consultant maybe? |
| 19:51.22 | Carp1 | I dont know of any other providers. |
| 19:51.27 | ChrisN | I'd like to be able to programatically create an outgoing call and transfer it to a given context and extension if the call is connected. Any pointers on what's needed to do this? |
| 19:52.05 | mercestes | Carp1: Teliax |
| 19:52.14 | Carp1 | ChrisN: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 19:54.06 | ChrisN | Carp1: Thanks. That looks like exactly what I need. |
| 19:54.53 | SplasPood | Recently I upgraded from asterisk 1.2.7 to 1.2.14 and now when calls are transfered by my queue agents using SIP transfers on their phones the caller is dropped .. They hit transfer, speak to the person they're transferring to, then hit transfer again to connect.. before they hit transfer the 2nd time the caller hears MOH, then disconnected. Any thoughts? |
| 19:55.31 | SplasPood | Logs don't indicate anything much.. it appears that the call never gets linked to the channel they're being transferred to, so when the agent hangs up the calls is dropped |
| 20:05.29 | *** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
| 20:08.10 | mercestes | SplasPood: random guess but canreinvite=yes maybe? |
| 20:08.35 | *** join/#asterisk shidan (n=chatzill@206.191.69.226) |
| 20:09.01 | *** join/#asterisk hi365_m (n=hi365_m@ti100710a081-2850.bb.online.no) |
| 20:09.23 | SplasPood | mercestes: nope, ... well.. it's always been set = no, unless some default changed somewhere in asterisk that's causing it to be yes now |
| 20:09.38 | SplasPood | mercestes: transfers work fine when the call didn't come in via a Queue/Agent combo |
| 20:09.48 | mercestes | Splaspood: I was suggesting changing it to yes and trying again |
| 20:09.59 | SplasPood | also via *2, cept the queue system maps * to hangup with no way to disabled |
| 20:10.10 | SplasPood | mercestes: oh... well I can try that, cept these phones cannot reinvite.. |
| 20:10.22 | mercestes | oh |
| 20:13.38 | *** join/#asterisk JoseBravo (n=JoseDavi@190.9.74.134) |
| 20:14.16 | JoseBravo | I can't listen music on hold. And in the CLI I see this message: "Stopped music on hold on Zap/1-1" |
| 20:14.42 | anonymouz666 | how much it's a fair value to install/configure a SIP proxy? |
| 20:14.55 | anonymouz666 | 1000 USD? 5000? |
| 20:16.47 | mercestes | anonymouz666: 5000. |
| 20:16.52 | mercestes | are you paying by cash, check, or paypal? |
| 20:16.54 | mercestes | I may be available |
| 20:17.11 | anonymouz666 | there is a guy asking for me to configure his server |
| 20:17.18 | mercestes | oh |
| 20:17.26 | mercestes | I hate you |
| 20:17.33 | Carp1 | Does anyone know Teliax live support hours? |
| 20:17.34 | mercestes | lol |
| 20:17.34 | anonymouz666 | I don't know how much it's a fair value |
| 20:17.47 | mercestes | weren't you in here asking questions two days ago? |
| 20:18.01 | anonymouz666 | ? |
| 20:18.02 | mercestes | Carp1: now until about 3 hours frmo now. |
| 20:18.22 | Carp1 | says its offline |
| 20:18.23 | mercestes | anonymouz666: didn't I teach you? ..for free? |
| 20:19.03 | anonymouz666 | mercestes: huh? |
| 20:19.11 | mercestes | anonymouz666: lol. It's up to you. I can't give yo ua fair value. |
| 20:19.17 | mercestes | anonymouz666: I was teasin gyo ubtw. |
| 20:19.27 | anonymouz666 | approximately |
| 20:19.34 | mercestes | Carp1: They are online. |
| 20:19.48 | mercestes | <PROTECTED> |
| 20:19.51 | mercestes | I just called them. |
| 20:19.52 | mercestes | nice people |
| 20:20.02 | Carp1 | Live chat support is offline, I meant....sorry. |
| 20:20.02 | mercestes | tell them Mercestes D'Moriarty sent you. |
| 20:20.09 | mercestes | be sure to pronounce it right. |
| 20:20.15 | mercestes | ... just call them. |
| 20:20.35 | mercestes | anonymouz666: 50 bucks an hour. Yo ucalculate how many hours it will take you |
| 20:20.51 | mercestes | anonymouz666: 200 bucks for a person, 1000 for a business. |
| 20:20.56 | mercestes | or just contract |
| 20:21.00 | mercestes | for an hourly rate |
| 20:21.07 | Deeewayne | ls |
| 20:21.52 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:21.55 | mercestes | Deeewayne: Directory is /: ./porn ./more porn ./Stolen MP3's ./extra porn ./kiddie porn ./legal porn ./free porn ./mp3s I paid for |
| 20:22.48 | russellb | lol ... |
| 20:22.54 | hi365_m | merc: whats the last folder for? |
| 20:22.59 | *** part/#asterisk jacekowski (n=jacekows@aaot198.neoplus.adsl.tpnet.pl) |
| 20:23.24 | mercestes | hi365_m: 'mp3s I paid for' It's just a place holder in the theorhetical event that I paid for an Mp3 |
| 20:24.00 | hi365_m | lol. i wouldnt even stand a chance! |
| 20:26.17 | *** join/#asterisk angom (n=angom@red-corp-201.143.59.181.telnor.net) |
| 20:29.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:30.35 | Juggie | how is any of that appropriate for an asterisk channel? |
| 20:31.00 | JoseBravo | After I change the extension.conf I need stop and start the asterisk, or I can refresh using CLI? |
| 20:31.19 | Juggie | 'extensions reload' |
| 20:31.32 | Qwell[] | 'dialplan reload' |
| 20:31.53 | Juggie | oh yah, it changed in 1.4 |
| 20:32.20 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 20:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:36.01 | Ciber311 | anyone here know of a wireless ip phone that doesn't suck? |
| 20:36.43 | Bobthehunter | hm !faq RFC callerid |
| 20:37.20 | Bobthehunter | can there be a dash in callerid ? |
| 20:37.52 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 20:38.17 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 20:40.01 | mut | so hows video workin with asterisk |
| 20:40.16 | mut | two eyebeam phones call eachother and video conference just as easy as a call setup? |
| 20:40.29 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:40.37 | CunningPike | Bobthehunter: I'm not sure - I know my SPA-3000 strips it out |
| 20:41.09 | wnfaknd | what does SPA stand for? |
| 20:41.17 | SplasPood | mercestes: was there something specific that led you to suggest reinvite, or just a guess considering the issue description? |
| 20:41.27 | SplasPood | wnfaknd: Sipura.. |
| 20:41.41 | Caitriona | Probably sipura phone adaptor or something, it's just a product number though. |
| 20:41.51 | wnfaknd | ic |
| 20:42.20 | wnfaknd | anyone familiar with vicidial? |
| 20:42.43 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 20:42.56 | wnfaknd | im getting no responses on #asterisk-gui |
| 20:43.21 | Bobthehunter | hmm |
| 20:43.29 | Bobthehunter | that could be a problem somehow.. |
| 20:43.34 | Bobthehunter | on same client same calls |
| 20:43.44 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 20:44.09 | Bobthehunter | soe show wron pass on invite... they all have insecure=port,invite |
| 20:44.12 | Bobthehunter | and its intermittent |
| 20:44.16 | Bobthehunter | so that a real wonder |
| 20:46.47 | nosbig | Today seems to be the "Fix Asterisk" day over here at work... Just to confirm, 1.2.13 allows mp3 files for its native file playback, correct? |
| 20:47.06 | mercestes | nosbig: Yea |
| 20:47.43 | nosbig | OK. |
| 20:50.37 | Carp1 | Teliax is going to be better than NuFone, I can feel it. |
| 20:50.54 | Carp1 | Only problem right now is that I connect a call and it drops before it gets to my asterisk box :-\ |
| 20:51.46 | *** join/#asterisk clonaAway (n=clona@bjs2-dhcp111.studby.uio.no) |
| 20:52.49 | wnfaknd | Carp1, what sets teliax apart form nufone? |
| 20:53.14 | mercestes | Teliax is better and has better tech support. |
| 20:53.17 | Carp1 | I can talk to someone for support rather than no support |
| 20:53.32 | Qwell[] | yeah, teliax doesn't come on IRC and help people out |
| 20:53.57 | wnfaknd | but what makes them better, other than tech support, or is just the "better" tech support the main difference? |
| 20:54.53 | mercestes | nufone doesn't come on IRC and help them out AFAIK. |
| 20:55.09 | Qwell[] | mercestes: Jeremy was here yesterday helping... |
| 20:55.23 | mercestes | he a nuphone guy? |
| 20:55.26 | mercestes | I never knew |
| 20:55.27 | Qwell[] | owner |
| 20:55.40 | mercestes | ahh. |
| 20:55.47 | mercestes | He probably doesn't like me now then. |
| 20:55.53 | wnfaknd | lol |
| 20:56.57 | wnfaknd | so then it just comes down to better tech support from the teliax folks? |
| 20:57.07 | mercestes | Honestly, I'v enever used nuphone. |
| 20:57.14 | mercestes | I have used teliax and I had really good luck with them and their service. |
| 20:58.04 | hi365_m | anyone us axvoice.com? |
| 21:00.36 | nosbig | OK. So, I have mp3 files in a directory on the Asterisk server, pointed to by the default class in musiconhold.conf. |
| 21:00.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:01.08 | nosbig | When I put a call on hold, the console passes me two messages... |
| 21:01.26 | nosbig | Jan 30 15:55:57 WARNING[4860]: file.c:512 ast_openstream_full: File /usr/share/asterisk/mohmp3/11 does not exist in any format |
| 21:01.29 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 21:01.38 | nosbig | Jan 30 15:55:57 WARNING[4860]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/usr/share/asterisk/mohmp3/11': No such file or directory |
| 21:01.59 | wnfaknd | any good and free softphone software out there? |
| 21:02.01 | mercestes | nosbig: Sounds like a musiconhold.conf issue |
| 21:02.06 | mercestes | wnfaknd: sjphone |
| 21:02.16 | mercestes | well. |
| 21:02.21 | mercestes | heh, it's *free* and it works. |
| 21:02.25 | wnfaknd | hehehe |
| 21:02.26 | mercestes | :) |
| 21:02.29 | wnfaknd | thx |
| 21:02.59 | file | nosbig: you have to have format_mp3 from addons in order to use mp3 files without conversion |
| 21:03.20 | Bobthehunter | also.. |
| 21:03.25 | Bobthehunter | how you print out the URI ? |
| 21:03.37 | Bobthehunter | it shows on chan_sip.c:12126 handle_response_invite: Forbidden - wrong password on authentication for INVI |
| 21:03.54 | Bobthehunter | not sure why ..but that peer works all the time exept on 1 phone number |
| 21:04.04 | wnfaknd | mercestes, have you tried expresstalk? <Carp1> Teliax is going to be better than NuFone, I can feel it. |
| 21:04.04 | wnfaknd | <Carp1> Only problem right now is that I connect a call and it drops before it gets to my asterisk box :-\ |
| 21:04.04 | wnfaknd | * clonaAway has joined #asterisk |
| 21:04.04 | wnfaknd | * BSDTech has quit IRC ("Watch out for that blue w---") |
| 21:04.04 | wnfaknd | <wnfaknd> Carp1, what sets teliax apart form nufone? |
| 21:04.05 | wnfaknd | <mercestes> Teliax is better and has better tech support. |
| 21:04.07 | nosbig | At some point before, I didn't have to have that, IIRC. |
| 21:04.07 | wnfaknd | <Carp1> I can talk to someone for support rather than no support |
| 21:04.09 | wnfaknd | <Qwell[]> yeah, teliax doesn't come on IRC and help people out |
| 21:04.11 | wnfaknd | <wnfaknd> but what makes them better, other than tech support, or is just the "better" tech support the main difference? |
| 21:04.14 | wnfaknd | <mercestes> nufone doesn't come on IRC and help them out AFAIK. |
| 21:04.16 | wnfaknd | <Qwell[]> mercestes: Jeremy was here yesterday helping... |
| 21:04.18 | wnfaknd | <mercestes> he a nuphone guy? |
| 21:04.18 | nosbig | I'm not saying that you are wrong, but something odd is happening on my end... |
| 21:04.20 | wnfaknd | <mercestes> I never knew |
| 21:04.22 | wnfaknd | <Qwell[]> owner |
| 21:04.23 | mercestes | omfg |
| 21:04.24 | wnfaknd | <mercestes> ahh. |
| 21:04.24 | Bobthehunter | try voicemeup |
| 21:04.26 | wnfaknd | <mercestes> He probably doesn't like me now then. |
| 21:04.28 | wnfaknd | <wnfaknd> lol |
| 21:04.28 | mercestes | dude. |
| 21:04.30 | mercestes | stop |
| 21:04.30 | *** kick/#asterisk [wnfaknd!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Qwell[]) |
| 21:04.33 | Nugget | pasting is hard. let's go shopping. |
| 21:04.43 | Nugget | middle button to the frontal lobe. |
| 21:04.50 | mercestes | I was just sitting here thinking, "man, I wish I could just relive the last 30 minutes of my life." |
| 21:04.54 | Nugget | ha ha |
| 21:05.06 | nosbig | And what conversion could I do to have * play them natively? Convert them to wav files? |
| 21:05.17 | mercestes | nosbig: Gsm |
| 21:05.23 | mercestes | nosbig: ulaw even so I've read. |
| 21:05.47 | mercestes | and mp3 natively if you install the format_mp3 thingy tha tfile mentioned. |
| 21:06.35 | nosbig | Right... |
| 21:07.12 | nosbig | I have found the original mp3 files I was using, so I am going to reload those... If those work, then I really want to check what makes these different from other mp3 files. |
| 21:07.30 | *** join/#asterisk HushPe (n=HushPe@mail.kamar.co.nz) |
| 21:08.44 | Carp1 | Does anyone here use Teliax? |
| 21:08.55 | mercestes | I did...why? |
| 21:09.02 | Carp1 | Like their service? |
| 21:09.07 | mercestes | that company went out of business tho due to repetitive voice issues |
| 21:09.25 | mercestes | I'm kidding. |
| 21:09.30 | mercestes | they were alright. |
| 21:09.31 | Carp1 | Teliax went out of business? |
| 21:09.34 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:09.59 | mercestes | Carp1: No. I was joking |
| 21:10.08 | Carp1 | Lol, I thought so. |
| 21:10.10 | nosbig | Nope... Same results.... OK. I'll try the format_mp3 and/or converting to gsm... |
| 21:10.18 | nosbig | Thanks for the reassurance... |
| 21:10.23 | Carp1 | My call keeps getting dropped before it makes it to my box |
| 21:12.09 | fall0ut | so is asterisknow going to be free still? |
| 21:12.58 | Qwell[] | fall0ut: of course |
| 21:13.06 | Qwell[] | fall0ut: why wouldn't it be? |
| 21:15.35 | Carp1 | maybe my call isnt getting through becuase I dont have the proper port fowarded? |
| 21:15.39 | mercestes | Carp1: Wait. that's probably an incoming context issue where * has no match for it. |
| 21:15.56 | Carp1 | I did iax2 set debug |
| 21:15.59 | Carp1 | but it shows nothing |
| 21:16.03 | *** join/#asterisk AvoidingDeadlock (n=brian@64.241.37.140) |
| 21:16.05 | mercestes | Carp1 did you do set verbose > 3? |
| 21:16.15 | mercestes | Carp1: set verbose 99 for example? |
| 21:16.25 | Carp1 | I had it set at 30 |
| 21:16.32 | *** join/#asterisk wnfaknd (i=wnfaknd@cpe-24-30-183-141.socal.res.rr.com) |
| 21:16.33 | mercestes | Carp1: That should be enough. |
| 21:16.42 | Carp1 | i dont think its making it to asterisk |
| 21:16.51 | wnfaknd | sorry for that mess earlier |
| 21:16.54 | Carp1 | 18005331547 |
| 21:17.03 | Carp1 | try it, i think you'll see what i mean |
| 21:17.22 | mercestes | Carp1: ... Ok 1, is that your number? 2. Does it point to a DID, 3. Does that DID get routed to you? |
| 21:17.37 | mercestes | Carp1: this is really an issue to discuss with your 1-800# provider. |
| 21:17.53 | Carp1 | why is it working for you and not me? |
| 21:17.57 | potential1 | wtf are those noises on yoru phone? |
| 21:18.22 | Carp1 | thats the tt-monkey file on asterisk |
| 21:18.28 | Carp1 | it doesnt work for me, btu i seen it work for you |
| 21:18.35 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 21:18.52 | mercestes | lol |
| 21:18.58 | potential1 | :) |
| 21:19.03 | mercestes | lmao |
| 21:19.15 | olsen | hi, a friend was asking me, if is possible to write a sms client using SMPP (i know this doesn't have anything to do with asterisk) and receive sms messages through another cellphone |
| 21:19.23 | olsen | will i need to use a PBX for that? |
| 21:20.18 | mercestes | olsen: uhhh....no. But you woul dhave to be able to transmit sms messages to the carrier of whatever cell provider you wish to sms to, as well as recieve from that carrier |
| 21:20.29 | Carp1 | I am so confused |
| 21:20.37 | Carp1 | Why did it work for you and not me? |
| 21:20.41 | Carp1 | still doesnt work for me. |
| 21:21.01 | mercestes | Carp1: maybe it' snot a recieving issue and it's a dialing out issue |
| 21:21.23 | Carp1 | i dont really know what you mean. |
| 21:21.27 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 21:21.35 | olsen | i see |
| 21:21.36 | Carp1 | where did you dial the number from? |
| 21:21.41 | olsen | sorry, my connection dropped |
| 21:22.20 | olsen | what is a carrier? |
| 21:22.57 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 21:22.57 | mercestes | Sprint, Verizon, Cingular, AT&T, SBC, SWBC, TWTC, Teliax. |
| 21:24.08 | *** join/#asterisk y7n (n=gareth@dsl82-163-126-23.as15444.net) |
| 21:24.27 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 21:24.38 | Carp1 | mercestes: if you have time, maybe you could walk me through this problem? |
| 21:25.00 | JT | so it works for some? |
| 21:25.03 | y7n | I would like to perform a system command when a user hangsup. How would I do this? |
| 21:25.41 | Qwell[] | y7n: System() |
| 21:26.03 | y7n | do i add that to the end of the extensions profile |
| 21:26.04 | mercestes | y7n: h,1,System(mv / /dev/null) |
| 21:26.08 | mercestes | don't really do that |
| 21:27.01 | b11d | ghahgahaha |
| 21:27.04 | danp | i'm using 'example 3' from http://voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial to dial multiple phones with delay on some but i'm trying to figure out how to quiet down the CDR records...how can i only log the call that was actually answered? |
| 21:28.42 | codefreeze | danp: in an ideal world, the unanswered CDRs would be marked "NO ANSWER". |
| 21:29.35 | HushPe | is there a way to tell asterisk to delete voicemail that a. is < 2 secs (hangups), b. emailed successfully, so I don't need to delete them by hand |
| 21:30.15 | De_Mon | HushPe yes |
| 21:30.38 | HushPe | great :) that will be a great time saver! |
| 21:31.08 | HushPe | i'll investigate when i get a chance, trying to setup a SIP incoming/outgoing line with not too much joy so far |
| 21:32.27 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 21:32.53 | mercestes | ~book |
| 21:32.55 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:33.11 | HushPe | ok, it sort of works, it just drops my call after about 3 seconds LOL |
| 21:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:33.24 | Carp1 | mercestes: can you tell me how you made that call? |
| 21:33.29 | Carp1 | landline? cellphone? |
| 21:33.34 | mercestes | Carp1: telepathy |
| 21:33.36 | *** join/#asterisk nachoguy (i=boster@ivan.dreamhost.com) |
| 21:33.46 | Carp1 | Thats a VoIP service? |
| 21:33.52 | mercestes | Carp1: Yea. |
| 21:34.01 | mercestes | It's the best. |
| 21:34.02 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:34.14 | mercestes | I never get dropped calls...i *always* get who i wanna talk to. |
| 21:34.16 | Carp1 | I am so confused why it took your call and mine doesnt make it to the box |
| 21:34.23 | mercestes | because I used telepathy. |
| 21:34.27 | nachoguy | is there a way in asterisk to set a sip entry that is actually just a pointer to another sip entry (sip/1001 actgually goes to sip/pbx/1001) |
| 21:34.27 | mercestes | it never fails |
| 21:37.46 | JoseBravo | Im triying to listen MusicOnHold but don't sound anything, and in the CLI I see: Stopped music on hold on SIP/jose-09674910 |
| 21:38.42 | JoNate | telepathy...apparent communication from one mind to another without using sensory perceptions |
| 21:38.50 | JoNate | telepathy |
| 21:39.16 | danp | hmm |
| 21:39.41 | mercestes | JoNate: LOL...dude, you ruined it |
| 21:40.08 | JoNate | doh! |
| 21:40.53 | JoNate | I read somewhere about some engine or something, but I figured you were going down that path... |
| 21:41.14 | mercestes | yea. When he said "That's a voip service"...I couldn't stop myself |
| 21:41.24 | b11d | hahahahahha |
| 21:41.29 | b11d | i love backlogs |
| 21:41.51 | JoNate | evil |
| 21:41.58 | JoNate | at least now I know not to ask questions |
| 21:42.10 | b11d | exactly |
| 21:42.11 | b11d | that was easy |
| 21:42.17 | b11d | this channel is for IDLING ONLY |
| 21:42.18 | b11d | :) |
| 21:42.37 | jart | b11d: that's what i do |
| 21:42.39 | mercestes | idlign and religious debates |
| 21:42.43 | b11d | lol |
| 21:44.24 | danp | codefreeze: that does seem to be the case. here's the relevant part of my dialplan: http://pastie.caboo.se/36796. any way to check DIALSTATUS for the channels that didn't answer and make sure no CDR gets logged? |
| 21:44.57 | *** join/#asterisk AuPix (n=AuPix@adsl-04-85.abel.net.uk) |
| 21:45.28 | mercestes | danp: you want to not have CDRS for unanswered calls? |
| 21:45.39 | danp | yeah |
| 21:45.53 | mercestes | danp: You could move to cdr_mysql I suppose and delete from cdr where dispostion <> "ANSWERED" but.....I don't know why. |
| 21:46.04 | mercestes | danp: you could just select * from CDR where dispositon = "ANSWERED". |
| 21:46.15 | danp | yeah, that's definitely an option...i'm already using cdr_mysql |
| 21:46.20 | mercestes | danp: how do you expect to troubleshoot dropped/unanswered/misdirected calls? |
| 21:46.39 | mercestes | then....just select the calls you want by using disposition = answered. |
| 21:47.09 | Caitriona | it's an incredibly bad practice to discard CDRs for any reason. |
| 21:47.14 | b11d | yeah |
| 21:47.38 | Caitriona | Realisticly, you probably need them for auditing too. |
| 21:47.52 | danp | true, true |
| 21:48.14 | danp | i guess i was trying to approach it from the wrong angle. i should probably clean up my display of them instead of trying to clean up the data itself |
| 21:48.44 | Caitriona | Phone phreaking still happens, and with the phone company systems more hardened these days, PBX systems are high profile and usually soft targets. |
| 21:49.03 | mercestes | danp: Precisely |
| 21:51.59 | *** join/#asterisk curtn (n=curtis@ip-150.net-81-220-49.lyon.rev.numericable.fr) |
| 21:53.09 | sweeper | Caitriona: out-of-baqndd signalling ftw |
| 21:53.38 | hardwire | mog: sup |
| 21:53.51 | Caitriona | sweeper: lol, yeah |
| 21:54.17 | Caitriona | don't forget answer/disconnect supervision on analog lines too :P |
| 21:54.24 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:55.07 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:55.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:56.04 | sweeper | in a way, it's sad, because phreaking is the way many a telecom engineer was born |
| 21:57.12 | *** join/#asterisk ShakataGaNai (n=s@adsl-66-122-88-147.dsl.renocs.nvbell.net) |
| 21:57.24 | [TK]D-Fender | Ah.. the good 'ole 2600 days.... |
| 21:57.30 | rudholm | cheap long distance service did as much to kill phreaking as oob signalling :) |
| 21:57.51 | rudholm | [TK]D-Fender: how old are you? |
| 21:57.59 | mog | hiya hardwire |
| 21:58.07 | [TK]D-Fender | rudholm : plenty enough :) |
| 21:58.12 | hardwire | mog: sup pimp |
| 21:58.19 | rudholm | hmm |
| 21:58.35 | mog | chillin |
| 21:58.36 | mog | you? |
| 21:58.50 | hardwire | you still in the seventh circle of hell? |
| 21:59.05 | rudholm | [TK]D-Fender: old enough for blue boxing? |
| 21:59.31 | mog | i didnt know i was ever there? |
| 22:00.03 | [TK]D-Fender | rudholm : Yup, but I also started young :) |
| 22:00.49 | [TK]D-Fender | rudholm : I miss the days of column 4 DTMF hacking :) |
| 22:01.07 | [TK]D-Fender | rudholm : And 300 baud accoustic coupled modems :) |
| 22:01.16 | hardwire | mog: did you move? |
| 22:01.25 | mog | oh yeah |
| 22:01.26 | mog | i moved |
| 22:01.30 | mog | i have a house now |
| 22:01.32 | rudholm | [TK]D-Fender: I started engineering school in 1982, fwiw. |
| 22:01.33 | mog | no more hell apt |
| 22:01.39 | b11d | thats cool |
| 22:01.44 | hardwire | mog: wewt |
| 22:01.51 | b11d | the only thing i miss about living in an apartment was the nice old lady below me who would cook me dinner :) |
| 22:02.05 | hardwire | hmm.. app_conference still the prefered alternative to meetme? |
| 22:02.43 | [TK]D-Fender | rudholm : I started college in comp-sci in 1992 ..... |
| 22:03.16 | mut | i was 9 years old in 1992 |
| 22:03.16 | rudholm | [TK]D-Fender: ah, back when CS was an unaccredited letter :-) |
| 22:03.17 | mut | w00 |
| 22:03.18 | [TK]D-Fender | rudholm : Like I said... I started young :) Programmed on boxes no-one remotely normal knows about.... |
| 22:03.18 | mut | t |
| 22:03.34 | [TK]D-Fender | rudholm : Oh no... quite accredited here.... |
| 22:03.39 | rudholm | where? |
| 22:04.06 | [TK]D-Fender | rudholm : Montreal, QC. College Com-sci has been well recognized and gov't rated |
| 22:04.13 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:04.26 | rudholm | [TK]D-Fender: ah, yeah, it wasn't accredited in the US until somewhere in the early to mid 90s, iirc. |
| 22:04.45 | b11d | strange.. the comp sci field was already a few decades old by the 90's |
| 22:04.50 | b11d | unless it was more of an EE thing back then |
| 22:05.08 | rudholm | [TK]D-Fender: but obviously a reputable institution isn't going to hand CS degrees willy-nilly, accreditation or not. |
| 22:05.39 | b11d | I work on the "other side" of education, and I can assure you.. reputable or otherwise, they will hand out degrees if it suits them. |
| 22:05.47 | rudholm | hehe, true. |
| 22:05.54 | b11d | threaten a lawsuit here today and you can have your B.S. tomorrow :) |
| 22:06.09 | b11d | its, frankly, disgusting. |
| 22:06.13 | rudholm | [TK]D-Fender: CS was often part of Mathematics departments |
| 22:06.42 | b11d | it would have been so mint to work on UNIVAC systems back in the day |
| 22:07.15 | b11d | ahh.. I wish I still had my AN/UYK-20 |
| 22:07.33 | [TK]D-Fender | rudholm : As it was in my HS. I've been programming since age 6. Gone through the PET, VIC-20, Apple ][, 8080 and up. And the aforementioned handful of unnameables :) |
| 22:08.02 | b11d | name them. |
| 22:08.06 | b11d | for my sake.. im nostalgic |
| 22:08.29 | mercestes | b11d: Hey, think you can get me a hookup for a BS then? |
| 22:08.33 | b11d | no dice. |
| 22:08.47 | [TK]D-Fender | tool* |
| 22:09.15 | mercestes | b11d: I'll sue! |
| 22:09.52 | rudholm | [TK]D-Fender: I think I started on a Rockwell AIM-65 or an S-100 |
| 22:10.42 | HushPe | what am i missing if i can dial out to my isp's sip server, it rings the phone i want it to, BUT i can't speak or anything like that :( |
| 22:11.08 | [TK]D-Fender | rudholm : My earliest was some large blue metal framed monstrosity with a flip-cover protected on button, booted into BASIC, and had a single SD floppy integrated. |
| 22:11.44 | rudholm | hmm, S-100s were often blue |
| 22:12.06 | mercestes | [TK]D-Fender: I remember BASIC. :) |
| 22:12.13 | [TK]D-Fender | rudholm : could be... My father was an electrical enginerr, which is why I've seen so much tech... |
| 22:12.24 | mercestes | [TK]D-Fender: was it BASIC, pro-DOS or DOS-3.0? |
| 22:12.45 | rudholm | [TK]D-Fender: My father was a programmer, and I'm an EE --opposite progression :) |
| 22:12.55 | [TK]D-Fender | mercestes : this was BEFORE DOS. |
| 22:13.03 | rudholm | hehe |
| 22:13.10 | HushPe | i remember the first few basic programs i wrote, a bit after that day though... one played terrible PC speaker tones on a 286, and the other drew colors in lines vertical and horizontal on an apple iie :) |
| 22:14.42 | mercestes | [TK]D-Fender: Apple IIe disks had a DOS3.0 which was non-ms. But..heh, yea |
| 22:15.02 | [TK]D-Fender | I wrote 2D math graphing apps in HS, and a final exam was a telecom terminal complete with VT100 emulation done from scratch. its great when your teach is just a math teacher and you have to EXPLAIN your projects to them :) |
| 22:15.03 | Carp1 | [TK]D-Fender: I left NuFone |
| 22:15.06 | HushPe | i wrote it by hand each time :) |
| 22:15.28 | [TK]D-Fender | Carp1 : Congratulations. I have heard some intersting stories of their suckage. |
| 22:15.36 | Carp1 | lol |
| 22:15.39 | Carp1 | I went to Teliax |
| 22:15.45 | Carp1 | and still having a problem lol |
| 22:15.46 | *** join/#asterisk eltech (n=eltech@ool-457c93b6.dyn.optonline.net) |
| 22:15.51 | mercestes | HushPe: Hey, I wrote a program that did something like that. it was a Apple IIe drawing program. |
| 22:15.56 | Carp1 | now my call doesnt even make it to asterisk |
| 22:16.05 | [TK]D-Fender | ! |
| 22:16.06 | Carp1 | I had one person from here call and it went through just fine |
| 22:16.07 | mercestes | My call makes it to asterisk tho |
| 22:16.11 | Carp1 | I dont know what the problem is now |
| 22:16.17 | Carp1 | Yeah, you're the only one |
| 22:16.31 | Carp1 | I tried my cell, landline, friends cell, other friends cell, his landline, and his VoIP line |
| 22:16.38 | Carp1 | yours is the only one that went through |
| 22:17.12 | in-pt | hi |
| 22:17.30 | in-pt | any one had succefully used chan_sccp patch with asterisk1.4 |
| 22:17.34 | in-pt | ? |
| 22:18.10 | rudholm | [TK]D-Fender: back in 1985 I was a Lab Assistant at my university's Dept. of Fine Art's Computer Graphics Lab. There were no tools. We had art students writing C code. I wrote libraries and tools that they built on. Seems crazy in retrospect. |
| 22:18.10 | HushPe | what causes SIP to drop an incoming connection after say 5 seconds of picking up? |
| 22:18.13 | in-pt | the in built skinny support sucks.. |
| 22:19.06 | [TK]D-Fender | rudholm : Nah.... I did all my stuff from the ground up incl. assembler. Nothing like the joy of doing it yourself. |
| 22:20.27 | [TK]D-Fender | rudholm : in college I re-wrote my comm app in TP7, and and wrote Avatar/0+ emulation from scratch as well, and then started on a small 1-line script parser for macros. It then turned into an entire LANGUAGE. The worlds FIRST mid-level language! |
| 22:20.38 | rudholm | [TK]D-Fender: I built my own P-N junctions and went up from there. piker! |
| 22:20.49 | HushPe | i'm getting: WARNING[15514]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmissio |
| 22:20.51 | [TK]D-Fender | rudholm : Wanna see some serious bastardized code? :) |
| 22:22.59 | *** join/#asterisk tabari (n=tabari@host-85-27-49-12.brutele.be) |
| 22:23.23 | tabari | Hello all |
| 22:23.38 | rudholm | [TK]D-Fender: yeah, by 1985, things were relatively mature compared to the hardware I had in the 70s. |
| 22:24.49 | tabari | Is it possible to run asterisk with a user different from root under linux? |
| 22:25.01 | [TK]D-Fender | rudholm : http://www.pastebin.ca/333437 |
| 22:25.06 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 22:26.49 | *** join/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 22:27.08 | [TK]D-Fender | rudholm : the Language has 2 main parsers. First is the "#" one that deals with "define", and all the conditional jumps (assembly style) |
| 22:27.30 | *** join/#asterisk matt_ (n=matt@82-33-68-44.cable.ubr01.trow.blueyonder.co.uk) |
| 22:27.30 | *** join/#asterisk jahani (n=k@adsl196-174-86-206-196.adsl196-3.iam.net.ma) |
| 22:27.57 | rudholm | [TK]D-Fender: wow, how high level :-p |
| 22:28.13 | [TK]D-Fender | rudholm : anything starting alpha is assumed to be a function ad can handle multiple parameters, incl variables, etc. just no EXPRESSIONS. Thats why every command is on a single line. its actually parsed and processed as raw text real-time :) |
| 22:28.20 | rudholm | [TK]D-Fender: I'm used to wiring TTL circuits. :) |
| 22:28.33 | rudholm | I see |
| 22:28.42 | [TK]D-Fender | rudholm : Yeah... solder.. the ultimate programming language ;) |
| 22:28.51 | *** join/#asterisk aethon (n=dskinner@seymour.ofc.bluefrog.com) |
| 22:29.14 | rudholm | [TK]D-Fender: and that was after I built the P-N junctions into the TTL gates in the first place! |
| 22:29.52 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:30.13 | [TK]D-Fender | my start in telecom (aside from BBS-ing and blue-boxing), came from programming CRD / management software for a Vantage 25 system for a Bell tech who was a friend of my fathers. we hot-wired CID boxes into it, and parallel port driven hoook switches to allow for black-listing etc :) |
| 22:30.35 | b11d | goodnight lads |
| 22:31.34 | Bobthehunter | MMWARNING[3059706800] chan_sip.c: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
| 22:31.37 | Bobthehunter | waht CASUES this ? |
| 22:31.47 | rudholm | [TK]D-Fender: my start in telecom was when they cut my switch over from a SxS to a 1AESS |
| 22:31.57 | aethon | anyone here use screen in conjuction with watching the asterisk debugging output? |
| 22:32.04 | rudholm | aethon: yup |
| 22:32.06 | [TK]D-Fender | rudholm : Ok, you're now officially out of my lexicon :) |
| 22:32.11 | Bobthehunter | looks like its a redirect.. pstn -> me -> iax client ->me -> pstn |
| 22:32.28 | Bobthehunter | its slin |
| 22:32.31 | Bobthehunter | why would he send slin ? |
| 22:32.40 | rudholm | aethon: I leave my Asterisk consoles up in screen all the time |
| 22:32.41 | *** join/#asterisk angryuser (n=Miranda@d05m-212-194-99-75.d4.club-internet.fr) |
| 22:32.50 | angryuser | good night;) |
| 22:32.51 | aethon | rudholm: every experience the * daemon pause while scrolled back in screen? |
| 22:33.22 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.34 | rudholm | aethon: no, can't say that I have. scrolling back with ctrl-a <esc> shouldn't affect * |
| 22:33.37 | in-pt | anyone familiar with skinny channels of asterisk-1.4.0 ..i had configured callerid properly in the skinny.conf file, but on the phones i see "unknown number" all the time |
| 22:34.32 | angryuser | i have a little problem with my IVR menu, when is is called from ISDN lines, keypressing for choosing options has no effect, can somebody help me out? |
| 22:35.10 | [TK]D-Fender | angryuser : check your DTMF settings.... |
| 22:35.11 | angryuser | *it is |
| 22:35.15 | Moobius | angryuser: double check your dtmf settings |
| 22:35.24 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-183-223.ny325.east.verizon.net) |
| 22:35.32 | aethon | rudholm: if I have verbosity on 3 and ctrl-esc and scroll back and a call comes in, it will the the first sound file or two and then just sit there in silence. once i hit enter to exit scrollback mode, it will continue playing the prompts....if I set verbosity to 0 and scrollback, it works fine |
| 22:35.40 | angryuser | ok searching thank you |
| 22:36.16 | JT | Carp1: hrm, sounds very unusualy |
| 22:36.18 | JT | unusual |
| 22:36.32 | JT | Carp1: do you have log of the successful calls? |
| 22:36.37 | JT | i wonder what's different |
| 22:36.39 | rudholm | aethon: that is weird. I guess it's possible that * is responding to flow control or something, but that'd be dumb of it. |
| 22:36.40 | Moobius | aethon: are you running screen/dtach? |
| 22:36.48 | JT | with iax2 debug turned on would be especially good |
| 22:37.34 | Carp1 | I might have a log |
| 22:37.38 | Carp1 | I didn't LOG it thought |
| 22:37.46 | JT | actually... |
| 22:37.55 | Carp1 | I still see it on my CLI |
| 22:38.41 | aethon | rudholm: i am not detached from the screen, but I have....i usually leave this screen running all the time and reattach to it when I continue working on it (server is not production yet) |
| 22:39.09 | angryuser | about DTMF, they are intended for sip peer/friends? (i have a simple extension for DSA number's) |
| 22:39.13 | aethon | Moobius: sorry that was for you |
| 22:40.22 | JT | angryuser: what do you mean? |
| 22:42.40 | angryuser | i have a simple ivr menu for incoming call from ISDN like exten => 8008,1,Answer() ect ect 8008 is not a peer, or something, it is just a part of phone number |
| 22:43.04 | JT | 8008 would be known as an "extension" |
| 22:43.17 | *** join/#asterisk oys (n=eoyslebo@84.208.78.84) |
| 22:43.25 | JT | let me guess, the ivr doesn't work? |
| 22:43.40 | angryuser | yes ivr doesnt work;) |
| 22:43.55 | HockeyInJune | asterisk will call any number in the world? |
| 22:44.03 | angryuser | well if a call extension from inter it Does work |
| 22:44.04 | JT | well for incoming isdn, the extension matches against the MSN |
| 22:45.32 | angryuser | JT:msn working fine, if i call this extension from external with the full number, menu does not work(buttons have no effect) but from internal, it is working fine |
| 22:45.56 | JT | this is over isdn or sip? |
| 22:46.15 | [TK]D-Fender | angryuser : Pastebin it all please. |
| 22:46.19 | angryuser | From internal >sip, external ISND |
| 22:47.11 | HockeyInJune | holy shit, this like fucking phreaking undergroud |
| 22:47.23 | angryuser | [TK]D-Fender:whick conf. misdn extension, sip? |
| 22:47.37 | angryuser | which? |
| 22:47.41 | [TK]D-Fender | angryuser : All 3 |
| 22:47.47 | JT | sip.conf is most important |
| 22:47.56 | JT | actually, extensions may be |
| 22:48.03 | JT | if isdn dtmf doesn't work either |
| 22:48.13 | [TK]D-Fender | JT : Yeah.. the 8008 + "IVR" is a tip off ;) |
| 22:48.20 | JT | i've heard dtmf can be dodgy under misdn too |
| 22:48.22 | JT | heh |
| 22:48.34 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 22:49.04 | olsen | a friend asked me if i could receive sms messages from another cellphone, to the web, or whatever, is that possible? |
| 22:49.07 | olsen | over SMPP i think |
| 22:49.20 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 22:49.31 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:50.09 | *** part/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-183-223.ny325.east.verizon.net) |
| 22:50.09 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-183-223.ny325.east.verizon.net) |
| 22:50.59 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 22:52.06 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 22:54.00 | *** join/#asterisk Caitriona (i=6e57e5f1@wikipedia/Triona) |
| 22:54.14 | olsen | a friend asked me if i could receive sms messages from another cellphone, to the web, or whatever, is that possible? |
| 22:57.54 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 23:04.26 | angryuser | [TK]D-Fender: sorry for taking so long http://www.pastebin.ca/333475 |
| 23:05.13 | in-pt | [TK]D-Fender: do u have any knowledge about skinny channels of asterisk |
| 23:05.16 | angryuser | i ad only remote ssh access so i had to upload configs by ftp |
| 23:07.44 | *** join/#asterisk wltjr (n=wltjr@gentoo/developer/wltjr) |
| 23:08.17 | wltjr | looking for some info on VOIP phones, recommendations for mid range stuff, no cisco, considering primarily snom and polycom |
| 23:08.35 | wltjr | snom is nice, lots o lines, linux etc, but seem to have small displays, till you get to the 360 |
| 23:08.35 | [TK]D-Fender | angryuser : You can't run an IVR off 8008 like that. Go read up on the "s" standard extension. |
| 23:08.47 | wltjr | polycom has bigger displays but less lines :( |
| 23:08.53 | [TK]D-Fender | NEXT!@!@!# (c) BKW |
| 23:09.00 | angryuser | wltjr:Snom's working really fine for me |
| 23:09.01 | mog | eep |
| 23:09.05 | [TK]D-Fender | wltjr : Less "lines |
| 23:09.06 | [TK]D-Fender | "? |
| 23:09.10 | wltjr | I have no experience with snom stuff, but hear it's good, they run linux and are uk based, most others are asian :) |
| 23:09.13 | The_DoC^ | oh how I love my greenlee 45575's |
| 23:09.43 | wltjr | <PROTECTED> |
| 23:10.01 | wltjr | angryuser: which snom do you have? |
| 23:10.03 | [TK]D-Fender | wltjr : how many identities do YOUR phones need? |
| 23:10.10 | mog | snom is german based |
| 23:10.13 | mog | i thought |
| 23:10.22 | wltjr | [TK]D-Fender: they will have access to 11 channels, so not sure |
| 23:10.41 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 23:10.46 | wltjr | [TK]D-Fender: ~6 likely, main office/owner has a 7960, but don't want to poney up the $ for others phones :) |
| 23:10.47 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:10.57 | [TK]D-Fender | wltjr : And if you know what you're doing you can do at LEAST 5 calls per line key on even an IP 301 letting you handle *10* calls... |
| 23:11.05 | angryuser | [TK]D-Fender:but if i will use s,extension all incoming calls will be picked up by menu...? |
| 23:11.24 | wltjr | [TK]D-Fender: sure, I did that in the past with 7912's, one line phone, but inbound is a pain, outbound wasn't to bad |
| 23:11.30 | [TK]D-Fender | angryuser : While you're at it, learn to make MORE contexts, and use "GOTO" <- ****HINT**** |
| 23:11.31 | wltjr | er, 7910's I believe |
| 23:12.13 | angryuser | wltjr:i have couple snom's 360 with BLF wathing for 20 fxs/sip's |
| 23:12.26 | [TK]D-Fender | wltjr : I use a single line key in some cases jsut handling 5 calls. who the hell needs more that 3 anyways? hardly inconvenient... just scrolll down through them. 1-button to start a new call... easy as can be (well... ALMSOT :)) |
| 23:12.35 | [TK]D-Fender | ok, GTG for now, back in a few hours |
| 23:12.40 | Moobius | I've tried moving from 1.4 to the latest svn and the service just segfaults and auto-restarts... |
| 23:12.41 | wltjr | angryuser: yeah I like those allot, but those are at the top of the price range we are looking to stay within |
| 23:13.49 | JT | angryuser: scp is probably a better way to upload config files, fyi |
| 23:14.11 | wltjr | [TK]D-Fender: so you think polycom then? less lines but larger display |
| 23:14.41 | JT | wltjr: i think he's trying to say you can have more "lines" than you think |
| 23:14.46 | wltjr | angryuser: is your 360's speaker phone full duplex? |
| 23:14.47 | [TK]D-Fender | wltjr : forget the idea of less LINES! a 501 can handle 15 calls minimum. again, who needs that many? |
| 23:14.50 | angryuser | JT: yea buy i had session on, and ftp ready, no need to bother for 3 files |
| 23:14.52 | burnproof | guys good day, just want to ask, if i buy 15 g729 license how many call i can make simultaneously |
| 23:14.57 | [TK]D-Fender | wltjr : You read too much into the term "line key" |
| 23:15.07 | [TK]D-Fender | And Polycom DESTROYS Snom.... |
| 23:15.18 | wltjr | JT: yes I have done various mapping and etc, but don't like training users or etc, easier to say, press this button and label it for given lines |
| 23:15.19 | JT | burnproof: depends how many transcoding sessions your calls use |
| 23:15.28 | angryuser | wltjr:full duplex no problem, quality is exellent |
| 23:15.33 | wltjr | [TK]D-Fender: well the 7960 with 6 has worked out great |
| 23:15.38 | [TK]D-Fender | burnproof : How many angels can dance ont he head of a pin? :) Rephrase your question.... |
| 23:16.11 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:16.12 | wltjr | and I like the 4 lines on my cheese grandstream gxp 2000 or what ever it is |
| 23:16.12 | Moobius | on fc6 how do i keep a service from auto-restarting? |
| 23:17.22 | wltjr | the grandstreams are ok, volume sucks on both ends though, handset and speaker |
| 23:17.39 | *** part/#asterisk amdtech (i=amdtech@nat/digium/x-a5f8c860cacbb73c) |
| 23:18.10 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:19.26 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-32-84.vic.bigpond.net.au) |
| 23:19.35 | Moobius | how do i keep asterisk from being auto-restarted when it segfaults |
| 23:19.50 | JT | don't run the safe_asterisk script |
| 23:21.15 | Moobius | how do i stop it? its stuck in a loop right now. |
| 23:21.26 | wunderkin | killall -9 safe_asterisk |
| 23:21.34 | wunderkin | asterisk -vvvcng |
| 23:21.51 | Moobius | wunderkin: thanks. that did it. |
| 23:22.29 | angryuser | when i receive a call from misdn is there any global var's set, like number of extension called? |
| 23:23.04 | JT | the msn is ${EXTEN}, and it's a channel var, not a global |
| 23:23.14 | Moobius | angryuser: try ${CALLERID(name)}, ${CALLERID(num)} |
| 23:23.20 | JT | Moobius: wrong end |
| 23:23.25 | Moobius | oops. |
| 23:23.35 | JT | Moobius: that will give you the callING number/name |
| 23:23.39 | JT | not callED number |
| 23:24.46 | angryuser | thx JT: |
| 23:24.55 | JT | well bristuff sets the callED number as ${EXTEN}, don't see why misdn would be any different |
| 23:25.46 | angryuser | JT: going to write that script |
| 23:25.52 | *** part/#asterisk aethon (n=dskinner@seymour.ofc.bluefrog.com) |
| 23:26.03 | wltjr | man I hate deciding on phones based on price/feature balance, much easier when just going for high end or low end, mid sucks as far as making a decision |
| 23:26.25 | JT | angryuser: what script? |
| 23:26.59 | wltjr | anyone have experience with snom 300/320? do you like? would you recommend? |
| 23:27.26 | wltjr | kinda leaning toward polycoms despite lines, due to larger displays, and price |
| 23:27.28 | Bobthehunter | How you restore a channel you destoryed with zap destroy channel 1 |
| 23:27.30 | Bobthehunter | ;) |
| 23:27.36 | Bobthehunter | tought it would just reset it |
| 23:27.42 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 23:28.06 | angryuser | JT: it will evaluate a called number, and take actions, i am in obligation of use s, priority for IVR to get working:( |
| 23:28.07 | Moobius | wltjr: i'm using the 320. I'm having some dtmf weirdness, and it doesn't have a flash button. but other than that a pretty good phone. |
| 23:28.19 | *** join/#asterisk J4k3 (i=jsuter@157.sub-70-216-175.myvzw.com) |
| 23:28.32 | wltjr | Moobius: ok, ty for the info, good to know |
| 23:28.36 | Moobius | np |
| 23:28.50 | JT | angryuser: umm in your incoming isdn context, you just match different MSNs by having an extension or extension pattern to match the MSN |
| 23:29.41 | angryuser | JT: extensions=msn's |
| 23:30.06 | JT | yes |
| 23:30.13 | JT | you don't need a script for that bit |
| 23:30.13 | Bobthehunter | jt is this japper style ? |
| 23:30.17 | JT | japper? |
| 23:30.18 | Bobthehunter | jabber |
| 23:30.21 | Bobthehunter | lol |
| 23:30.25 | Bobthehunter | keyword offset +1 |
| 23:30.32 | Bobthehunter | darn keyboard |
| 23:30.35 | JT | not sure what the question is Bobthehunter |
| 23:31.32 | angryuser | JT: but i can't attach IVR to incoming msn=extension, not working;-( |
| 23:31.38 | wltjr | by the way did my first pri, and split voice data to a cisco on the other end was painless with sangoma a101, once I got wanpipe and zaptel compiled together, missed wanpipe dchannel patch for zaptel and caused some funkiness |
| 23:32.20 | angryuser | JT: as i said before IVR start to play files, but keys have no effect |
| 23:32.21 | k-man_ | how do i connect asterisk to my sip provider? |
| 23:32.25 | k-man_ | which file do i edit? |
| 23:33.35 | angryuser | k-man_: sip.conf |
| 23:33.35 | k-man_ | ah |
| 23:33.35 | k-man_ | thanks |
| 23:33.35 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.35 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.35 | JT | angryuser: i don't see why you couldn't get an ivr to work that way, it's just your way of doing it was not the proper way to do IVRs :) |
| 23:33.36 | angryuser | k-man_: there is one with the register => example |
| 23:33.36 | k-man_ | actually, let me rephrase... i just finished doing a make install... whats the first thing i should do? |
| 23:34.23 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-170-132.hsd1.ca.comcast.net) |
| 23:34.39 | oys | anyone: what is a "symbol lookup error"? asterisk won't start because of this. |
| 23:35.46 | angryuser | JT: i thought it is possible to attach IVR to any extension, not just s, |
| 23:35.46 | angryuser | JT: as fender says i am wrong;) |
| 23:36.22 | JT | angryuser: take a look at http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu |
| 23:36.36 | JT | you can make proorities jump to other context/priorities |
| 23:36.48 | JT | so you don't need a script |
| 23:36.53 | *** join/#asterisk kervel (n=kervel@d5152D246.access.telenet.be) |
| 23:37.48 | kervel | hello, i just bought a digium TDM10B card for hooking up an old fax ... i don't have the fax yet to test it, but when i load the driver, it tells me no modules are installed on the card. |
| 23:38.06 | kervel | however, there is clearly one module on the card. would this be a software issue, or should i return the card ? |
| 23:38.14 | kervel | i don't have the hardware to test it yet |
| 23:39.12 | angryuser | JT: is is not gonna work for me, i need some people calling some msns go directply to some internal sip's |
| 23:39.29 | JT | easy |
| 23:39.40 | JT | just set their extensions to go straight to sip |
| 23:39.41 | angryuser | JT: and only one exten, IVR |
| 23:39.52 | JT | don't understand you |
| 23:40.07 | JT | i don't think you're realising how flexible the extensions.conf dialplan can be |
| 23:40.23 | JT | from what you've told me so far, i have no idea why you'd need a script |
| 23:40.30 | JT | agi has a bit of a performance hit |
| 23:42.33 | angryuser | my 4 isdn lines are mixed on entry and exit, i have no like 2 lines for call out and 2 lines for in; so whatever i dvide the to different context's it wont change, i will search for your page a bit, maybe find answer |
| 23:43.31 | JT | hard to understand what you're saying, but i suggest you read up on extensions.conf |
| 23:43.50 | angryuser | JT: stupid me;) i unserstood |
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| 23:44.05 | *** part/#asterisk ArkyLady (i=ArkyLady@dsl-205-234-115-182.madisoncounty.net) |
| 23:44.17 | angryuser | JT: ok it will work, i think |
| 23:45.47 | angryuser | JT: thank you jt and fender taking you time on me, i dint saw the problem from you angle |
| 23:46.13 | JT | hope you are able to work it out |
| 23:47.04 | angryuser | good night, going to sleep;) |
| 23:54.29 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 23:54.30 | kink0 | hi |
| 23:54.42 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 23:55.35 | kink0 | I am seeking for direct dedicated T1's USA,Canada and Mexico , proper and mobile termination. |
| 23:58.20 | kink0 | good night, bye. |
