| 00:05.00 | *** join/#asterisk T`2 (n=total@pdpc/supporter/student/T) |
| 00:06.33 | T`2 | hi, i have an old digium modem card i found.. and am wondering if i can use this with asterisk to recieve/dial phone calls from the an SIP gateway on an analog phone hooked to this card |
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| 00:13.45 | stencil | hello, my cdr logs are five hour fast probably GMT, my computers timezone is set to EST is there a asterisk command for changing timezone or would I have to reinstall asterisk? |
| 00:26.51 | *** join/#asterisk germaki (n=enter@69.171.154.251) |
| 00:26.52 | germaki | hello |
| 00:27.01 | germaki | I"m trying to setup a T38 gateway and get t38 to work with my faxes |
| 00:27.13 | germaki | I have installed span and am using asterisk 1.6 |
| 00:27.23 | germaki | when i do a core show applications i do not see the application. |
| 00:27.34 | germaki | is there a how to on how to use 1.6 and spandsp to send faxes? |
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| 00:34.42 | jaytee | stencil, set usegmtime=no in cdr.conf it's supposed to be the default but I never changed mine until I switched to mysql and it was already set to yes. |
| 00:35.18 | stencil | jaytee: thanks man |
| 00:40.21 | germaki | 1- |
| 00:40.22 | germaki | c |
| 00:42.25 | stencil | jaytee: this problem has driving me bonkers, really appreciate the help |
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| 00:52.53 | jaytee | stencil, no problem. you'll need to restart cdr though |
| 00:54.33 | stencil | jaytee: sure, bye for now |
| 01:07.36 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 01:08.30 | stencil | happiness ;-)) |
| 01:11.22 | jaytee | \ O / yay!!! |
| 01:11.53 | *** join/#asterisk ManxPower (n=manxpowe@24.196.31.252) |
| 01:12.47 | jaytee | it's odd that the config file says that "no" is the default but I'd never set it to yes in the first place and had to change it to no to get it to log the time using local time and not GMT. If I wasn't such a lazy ass I'd file a bug :-) |
| 01:13.36 | jaytee | hi ManxPower |
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| 01:15.07 | ManxPower | hello, JT |
| 01:15.13 | ManxPower | or jaytee too |
| 01:15.25 | jaytee | :-) either works for me |
| 01:15.35 | jaytee | did you have a nice holiday? |
| 01:16.54 | ManxPower | IT was a nice day. I don't celebrate christmas anyway |
| 01:17.54 | ManxPower | Or the more correct term of "Holiday that the christians coopted from the pagans in order to make it more attractive to pagans" |
| 01:18.21 | jaytee | well, if he was for real his birthday was most likely in late August/early September but that wouldn't have given early Christians the excuse to hijack the winter solstice holiday of Beltane/Yule :-) |
| 01:18.40 | jaytee | wow! we're on the same page! hehe |
| 01:18.53 | ManxPower | I'm a devout atheist anyway. 9-) |
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| 01:20.25 | jaytee | ManxPower, I believe in a creator but I just believe that several thousand years ago He/She/It got fed up and said, "Ah, vat you know from nuthin? Buncha putzes!" and went on an intergalactic "walkabout". |
| 01:21.40 | jaytee | because the whole idea of all of this just "happening" all on it's own defies logic. Take the skunk for example, clear evidence of a guiding hand in evolution, a guiding hand that's connected to a mind with a sick, warped sense of humor. |
| 01:22.59 | jaytee | I think Ferris Beuller said it best: "I don't believe in 'isms'. I just believe in me." |
| 01:24.26 | ManxPower | jaytee: I like string theory (and things similar to it) i.e. a basic set of equations from which everything else can be derived. Much like a small number of incredients can make an amazing variety of food. |
| 01:25.00 | jaytee | yeah, like genetic algorithms |
| 01:25.33 | jaytee | that and HERV |
| 01:26.18 | jaytee | a good read if you like scif-fi that's based on current factual science is Greg Bear's Darwin's Radio |
| 01:28.08 | ManxPower | I do admit the universe (or at least the big bang) could have been created by a being, but caring about it is about as silly as a rat worshiping a zookeeper. |
| 01:28.59 | jaytee | my arms are killing me from leaning through a 19" rack while hanging off a ladder punching down CAT5 cables to replace a bunch of plenum crap that was run 8 years ago and I don't even think it was CAT3. It had next to no twists in it at all. |
| 01:29.07 | ManxPower | or more like a a rabit worshiping the person that brought the first rabit to austrailia -- nobody even remembers it. |
| 01:29.25 | ManxPower | cat3 has next to no twists. |
| 01:29.44 | jaytee | ManxPower, exactly. I'm here, if he wants to talk to me and impart a message he can. So far my phone hasn't rung. |
| 01:31.33 | ManxPower | I'm pretty sure than when humans start to create universes (if that is possible) the scientists will have about the same amount of involvement with what is in that universe as the potential creator of this universe has with is -- none |
| 01:32.37 | ManxPower | i.e It simply does not matter. |
| 01:32.37 | jaytee | very likely |
| 01:33.24 | ManxPower | To me religion is about control. a person or group of persons wanting to control others by playing on their fear of death and the unknown |
| 01:34.43 | jaytee | now the Vatican is trying to rewrite history and make it out that they think Galileo was a standup guy. Really nice! Put the guy under house arrest for saying the earth revolves around the sun which we now know as a certainty and then 400 years later act like it was all just a big misunderstanding. Hypocrites! |
| 01:35.40 | jaytee | ManxPower, I agree 150% with your last statement. That's exactly how I think of it. Fear and guilt are excellent tools for controlling people. |
| 01:36.12 | jaytee | and let's not forget hatred and bigotry! |
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| 02:52.31 | dec3pti0n | in 2006 philips launched a VVOIP line of products like the VP6500 , does anyone know what happened to those phones I can't find anywhere for purchuse |
| 02:54.45 | drmessano | Yuck |
| 02:56.29 | coppice | philips endlessly introduce and drop telephony products |
| 02:58.37 | drmessano | I just love using their wifi video product, powered by 2 AA batteries. I can make an endle |
| 02:58.49 | drmessano | Sorry, batteries died |
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| 03:12.37 | huisnah | drmessano emailed you, i messed up my email last time (brain fart) |
| 03:22.01 | huisnah | test |
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| 03:39.28 | eppigy | hello |
| 03:39.30 | eppigy | i am dave |
| 03:40.57 | ManxPower | dec3pti0n: Only a few companies have had significant success with voip phones |
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| 03:43.20 | coppice | you gotta get on business desks, or achieve strong tie-ins with consumer service providers. vague marketing is death |
| 03:44.28 | ManxPower | ~phones |
| 03:44.28 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 03:44.34 | ManxPower | there's the list 8-) |
| 03:44.59 | ManxPower | Grandstream has had success based price and the fact first time buyers don't know any better. |
| 03:45.11 | coppice | never trust advice endings with "Ever. places like such as" :-) |
| 03:45.49 | ManxPower | I never trust jbot anyway. He's a shifty kind of guy |
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| 03:56.57 | jaytee | ~botsnack |
| 03:56.57 | jbot | thanks, jaytee |
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| 05:51.43 | *** join/#asterisk engaged (n=engaged@ip24-251-135-117.ph.ph.cox.net) |
| 05:52.01 | engaged | Anyone in here from Phoenix area that can set up my company's phone system? |
| 06:00.39 | [TK]D-Fender | engaged: Physical presence is required? |
| 06:00.45 | engaged | yes |
| 06:00.50 | engaged | need someone to set everything up |
| 06:00.51 | [TK]D-Fender | oh well |
| 06:00.53 | engaged | and tell me wtf to buy |
| 06:01.21 | [TK]D-Fender | tahts easy enough, but if you require physical that another matter |
| 06:02.25 | engaged | well |
| 06:02.28 | engaged | i dont require physical |
| 06:02.29 | engaged | can you PM me |
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| 06:17.07 | DMMatt | Hello :) I just installed an asterisk system. The system in on a public IP, and I have one phone registered remotley behind a NAT. Everything seems to work *okay* but I have inconsistant one way audio. |
| 06:17.19 | DMMatt | Sometimes, caller can hear me, I can't hear caller |
| 06:17.28 | DMMatt | Sometimes, I can hear caller, caller can't hear me |
| 06:17.34 | DMMatt | Sometimes, both are fine |
| 06:18.09 | DMMatt | I've blamed NAT, Firewall, everything! But the fact that each scenarios work at some point leave me with a scrambled mind. Anyone have any ideas? |
| 06:21.22 | drmessano | What sort of firewall is on the phone end? |
| 06:22.16 | DMMatt | I've got a FreeBSD router running a PF firewall |
| 06:24.52 | DMMatt | It's wierd. I'd like to have my pbx at my datacenter, but it seems to only work well when the pbx is behind the NAT with the phone. |
| 06:25.05 | DMMatt | I just don't get why I have wierd one way audio issues that seem to alternate diretions |
| 06:25.08 | DMMatt | directions rather |
| 06:26.30 | DMMatt | my datacenter being my colo |
| 06:29.09 | mmlj4 | your phone is behind NAT? and you're running SIP? there's your problem, probably |
| 06:29.55 | DMMatt | You think IAX would be the better protocol for that setup? |
| 06:30.20 | DMMatt | Well, probably doesn't matter, doubt my Cisco 7970G even supports IAX |
| 06:30.51 | mmlj4 | you can put an asterisk instance on your LAN, which would eliminate the one-way stuff, from the phone to that asterisk box |
| 06:31.01 | mmlj4 | then use IAX to get to the remote asterisk |
| 06:31.21 | DMMatt | I like the way you think. |
| 06:32.14 | mmlj4 | but the * box, it's at a colo or something? |
| 06:32.21 | DMMatt | yeah |
| 06:32.25 | mmlj4 | why? |
| 06:32.44 | mmlj4 | where do you get your dialtone? or termination, or whatever term you like? |
| 06:32.47 | DMMatt | I just want it to have a static IP, I have about 5 phones connecting to it remotley from different locations |
| 06:32.57 | mmlj4 | ah, ok |
| 06:33.01 | DMMatt | pure SIP VOIP at the colo |
| 06:33.22 | DMMatt | I'm being ambitious I know :) |
| 06:33.31 | mmlj4 | that's fine |
| 06:33.39 | DMMatt | But the fact that sometimes it works perfectly and sounds great is keeping me alive! |
| 06:33.54 | mmlj4 | but supposedly you can get asterisk to work around NAT issues, they tell me |
| 06:34.03 | mmlj4 | ask later for help |
| 06:34.16 | DMMatt | yeah... I feel like I've blamed NAT into the ground, and that it is most likley a firewall issue |
| 06:34.43 | DMMatt | I've blammed NAT, the Carrier, the Phones, and my wife... Haven't gotten too far |
| 06:35.27 | DMMatt | I bet the biggest issue is that I've been trying to get this Cisco phone working for too long, sweet phone, works well in certain situations, but may be my main cause of headace |
| 06:35.33 | [TK]D-Fender | DMMatt: READ : |
| 06:35.34 | [TK]D-Fender | ~sipnat |
| 06:35.35 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 06:36.09 | DMMatt | [TK]D-Fender |
| 06:36.12 | DMMatt | Thx, I'll check it out |
| 06:37.13 | DMMatt | The sad part is that I love all this crap :) |
| 06:38.00 | DMMatt | Once it works, its a great feeling, nothing that is worth it comes easy |
| 06:39.17 | drmessano | Ummm |
| 06:39.33 | drmessano | If the phone is behind a NAT, and asterisk isnt |
| 06:39.55 | drmessano | The phone is having an issue with PFSense, and I have heard of it before |
| 06:39.59 | drmessano | There is some option for SIP |
| 06:40.12 | drmessano | That needs to be toggled whatever opposite direction you have it in now |
| 06:40.25 | DMMatt | drmessano: I'm not using PFSense, but am using OpenBSD's Packet Filter (PF) |
| 06:40.31 | DMMatt | So it may be the same issue |
| 06:40.36 | drmessano | Probably so |
| 06:40.55 | drmessano | But phones behind NAT's are not a problem |
| 06:41.07 | drmessano | Just need to make sure Asterisk is open and happy, which it sounds like it is |
| 06:41.12 | drmessano | Everyone screams |
| 06:41.19 | DMMatt | If you are telling me I might have to get rid of PF, I might starting whining like a littl girl ha |
| 06:41.20 | drmessano | SIP DOESNT WORK BEHIND NAT |
| 06:41.22 | drmessano | SIP SUCKS |
| 06:41.26 | drmessano | and thats not really true |
| 06:41.46 | DMMatt | Yeah, it's sort of like PF's problem with FTP |
| 06:41.49 | drmessano | It just needs a little massaging to make it work, unlike opening IE7 on their Vista boxes |
| 06:42.32 | drmessano | Passive FTP is another one of those things that throws wannabe's |
| 06:42.46 | drmessano | "BUT 21 IS OPEN ZOMG IT R BORKED" |
| 06:42.50 | drmessano | No, not really |
| 06:43.14 | DMMatt | It's one of those fun times I get to break out tcpdump, etherape, and ethereal and have a blast ;) |
| 06:43.29 | DMMatt | Truth is, this is hobby and not mission critical, so stress is pretty low |
| 06:43.38 | DMMatt | frustration maybe a little higher |
| 06:43.44 | coppice | break out the semtex and have a blast |
| 06:44.11 | drmessano | Theres something that has to do with how it TRIES to track SIP thru the NAT, and either the default fails horribly, or the shim in place sucks and needs to be turned off |
| 06:44.16 | drmessano | I dont rememeber which |
| 06:44.58 | DMMatt | Plus I think the carrie's proxy has something to do with it too, possibly |
| 06:45.14 | DMMatt | I'be had certain proxies send malformed headers |
| 06:46.10 | coppice | half the SIP implementations out there send something malformed. it may be hard to distinguish when a proxy is at fault |
| 06:46.34 | DMMatt | yeah |
| 06:48.20 | DMMatt | I am really loving * though, I haven't geeked out this much since my days of learning *nix |
| 06:49.42 | DMMatt | especially since I'm configuring asterisk via command line in fbsd |
| 06:50.16 | drmessano | Even though you're a BSD user, we welcome you anyway |
| 06:50.59 | DMMatt | yeah its great. Asterisk has been one of those most impressive open source projects I've seen |
| 06:51.16 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 06:51.17 | DMMatt | amazing docs, lots of spinoff projects, just amazing |
| 06:53.36 | DMMatt | I think back to some of my days at my former companies when we spent thousands of dollars on crappy system where the only way to configure an option was to call the communications vendor. Most people are still in that boat. Mind boggiling |
| 07:13.57 | DMMatt | nte ;) |
| 07:13.58 | DMMatt | quit |
| 07:13.58 | DMMatt | exit |
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| 08:10.05 | verywiseman | is linksys spa3102 ata gateway work probably with asterisk? |
| 08:17.23 | drmessano | Sure it does |
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| 12:03.26 | C4colo | can I subscribe to the status of a peer on another system using hints? |
| 12:04.15 | C4colo | exten => 100,hint,SIP/user1234@some_sip_peer |
| 12:04.40 | C4colo | I'm guessing I would need some credentials with that particular server, at least a sip user on that server |
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| 12:11.24 | Subdolus | Can anyone recommend a softphone for linux that supports g729? |
| 12:12.21 | C4colo | probably not, g729 is expensive to license |
| 12:12.31 | C4colo | I don't know of any personally |
| 12:12.51 | C4colo | you might get counterpath's eyebeam to run under wine |
| 12:12.54 | C4colo | never tried that either |
| 12:14.01 | farkus | my linksys spa-2102 has a setting "preferred codec" includes "G729a" |
| 12:14.23 | farkus | among the 8 options for that setting |
| 12:14.27 | C4colo | your linksys SPA-2102 is not a free, open-source software applicaiton |
| 12:14.44 | C4colo | Sipura, and now Linksys can afford to spend tens of thousands of dollars to add a codec |
| 12:14.54 | farkus | oh "softphone"... I need to get my prescription checked |
| 12:14.59 | Subdolus | My ATA supports it |
| 12:15.02 | farkus | :-) |
| 12:15.08 | Subdolus | and there are apparently lots of softphones aswell |
| 12:15.16 | Subdolus | Zoiper being one, but it doesn't like my soundcard |
| 12:15.21 | C4colo | farkus: sorry I should read people's nicks before responding too |
| 12:15.22 | C4colo | lol |
| 12:15.38 | C4colo | I'm trying to write a complex dialplan and only paying half attention to the channel |
| 12:15.53 | Subdolus | Uhhhh so yeah |
| 12:16.04 | Subdolus | any recommended softphones? |
| 12:16.07 | C4colo | does zoiper support g729? |
| 12:16.08 | C4colo | wait |
| 12:16.17 | Subdolus | It's supposed to |
| 12:16.20 | C4colo | I think the pro version of eyebeam runs on max osx |
| 12:16.34 | C4colo | mac rather |
| 12:16.44 | C4colo | and, therefore it would make sense that they have a linux version |
| 12:17.08 | C4colo | I played with zoiper a bit |
| 12:17.17 | C4colo | but that was before I had g729 licenses to play with |
| 12:18.02 | C4colo | I wasn't too impressed with zoiper |
| 12:18.08 | C4colo | I actually liked idefisk better |
| 12:18.31 | C4colo | but there were problems there |
| 12:18.36 | C4colo | a little buggy |
| 12:18.39 | Subdolus | That's all very nice, C4colo |
| 12:18.47 | C4colo | thanks |
| 12:18.50 | C4colo | I'm just rambling |
| 12:18.51 | C4colo | =) |
| 12:18.58 | Subdolus | I can see that :-) |
| 12:19.14 | Subdolus | Anyone else? Google is failing me |
| 12:19.17 | C4colo | I use gsm on my linux softphone |
| 12:19.32 | Subdolus | C4colo: Oh is g729 = gsm? |
| 12:19.37 | C4colo | I only use my laptop though when I'm traveling, and even then I take my aastra phone |
| 12:19.37 | C4colo | no |
| 12:19.42 | Subdolus | ohhh right |
| 12:19.44 | C4colo | gsm is free open source |
| 12:19.44 | Subdolus | Still rambling |
| 12:19.51 | C4colo | but still low-bandwidth |
| 12:19.53 | Subdolus | looks around |
| 12:20.04 | C4colo | g729 is a bit smaller than gsm, but gsm is free |
| 12:22.31 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 12:31.06 | TrentCreek | Subdolus: ww.dogpile.com |
| 12:31.30 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:31.48 | verywiseman | pls , tell me some ata gateway products that support more than 10+ ,20+,30+ fxo and fxs |
| 12:32.54 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:35.20 | TrentCreek | SIP or IAX2 ;-) |
| 12:36.06 | C4colo | verywiseman: asterisk |
| 12:36.20 | C4colo | sangoma has products that do 48 FXO/FXS ports |
| 12:37.03 | C4colo | make a simple SIP gateway using asterisk |
| 12:37.11 | C4colo | will be the most flexible solution |
| 12:37.13 | coppice | verywiseman: they usually go in steps of 8 or 12, but people like Audiocodes have plenty of boxes of that kind |
| 12:37.27 | C4colo | coppice: I have that pcap file for you |
| 12:37.46 | coppice | then mail it right away |
| 12:37.49 | C4colo | we have an audiocodes media gateway that we use for our T1 channel banks |
| 12:38.01 | C4colo | it's pretty big, I'll upload it and send you the link |
| 12:38.05 | C4colo | pm me your email address |
| 12:47.41 | C4colo | coppice: sent, I put a link to the 1.3MB pcap file and attached the SIP debug output from the CLI |
| 12:47.41 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 12:48.12 | C4colo | For some reason I thought it was bigger, I probably could have attached the 1.3MB file |
| 12:50.27 | coppice | C4colo: Error 404 |
| 12:50.45 | C4colo | hmm |
| 12:50.56 | C4colo | .pcap might not be an accepted MIME type on that server |
| 12:51.00 | C4colo | I'll just email it |
| 12:53.19 | C4colo | also I changed the file name to .pcap.txt |
| 12:53.23 | C4colo | if you want to try downloading it agai |
| 12:53.25 | C4colo | n |
| 12:53.30 | TrentCreek | telnet |
| 12:53.50 | C4colo | and emailed it too |
| 12:54.36 | C4colo | although I think it is probably a binary file, so might not work |
| 12:55.14 | C4colo | never understood why FTP and such needed to change modes, a text file is just a byte array, why can't it just use binary all the time? |
| 12:55.33 | *** join/#asterisk markwaters (n=markwate@80.118.244.23) |
| 12:57.05 | C4colo | hmm, sorry it's saying packet size limited during capture on some of the t.38 packets, I set -s 2048 ... guess that wasn't big enough |
| 13:03.22 | C4colo | coppice were you able to get the file? |
| 13:09.37 | C4colo | i hadn't noticed this before: |
| 13:09.38 | C4colo | Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) |
| 13:10.27 | C4colo | but that is to the SS7 gateway, which doesn't handle the audio path... but could that be why it keeps saying "dropping incompatible voice frame of type slin..."? |
| 13:10.32 | coppice | callweaver asks for T.38. The other side says no. Things carry on as ulaw |
| 13:11.09 | C4colo | up to packet 225 |
| 13:11.20 | C4colo | where callweaver tries a reinvite |
| 13:11.25 | coppice | and of course buggy old wireshark makes a pigs ear of presenting it :-) |
| 13:12.05 | C4colo | then packet 227/228 seems like it says "trying/ok" and starts sending t.38 udptl from 96.29 |
| 13:12.24 | C4colo | but callweaver still sends RDP/PCMU |
| 13:12.35 | C4colo | that's what I saw, but I haven't looked at a lot of these files |
| 13:13.00 | coppice | both side are still sending RTP/PCMU. If you are looking at a wireshark decode its messed up |
| 13:13.27 | coppice | callweaver asked for T.38, but the OK message says to use ulaw |
| 13:14.13 | C4colo | ah yes, Media Format: ITU-T G.711 PCMU |
| 13:14.22 | C4colo | why does wireshark think it is t.38? |
| 13:14.34 | C4colo | and why does callweaver try over and over to reinvite? |
| 13:14.54 | C4colo | this is the rxfax app by the way |
| 13:14.57 | coppice | even if you tell wireshark specifically to decode that stuff as RTP, it still tries to decode as T.38. |
| 13:15.12 | coppice | where does it try over and over? |
| 13:15.21 | C4colo | look at the sip debug output |
| 13:15.40 | C4colo | and in wireshark, it shows an invite packet every 10 seconds or so |
| 13:16.13 | C4colo | #585 |
| 13:16.33 | C4colo | #939 |
| 13:16.52 | C4colo | #1294 |
| 13:16.53 | C4colo | and so on |
| 13:17.08 | C4colo | and in the sip headers you will see an invite packet over and over |
| 13:17.37 | C4colo | t38 gateway works fine |
| 13:18.34 | C4colo | I can go PSTN -> SS7 -> (ulaw) fax server (T.38) -> T.38 ATA/IAD |
| 13:18.44 | C4colo | but not rxfax |
| 13:19.45 | C4colo | it works about 50% of the time, but the other 50% it obsesses about T.38 and tries over and over to do a T38switchover |
| 13:19.45 | coppice | are you sure it is using T.38, as in this call T.38 is being rejected by the far side? |
| 13:20.55 | C4colo | 96.32 initiates the "BYE" |
| 13:21.14 | C4colo | then 96.44 hangs up "normal clearing" |
| 13:21.26 | C4colo | that's after it tries about 6-10 times to switchover/reinvite |
| 13:22.01 | C4colo | 44 is the fax server, 32 is the SS7 gateway |
| 13:22.11 | TrentCreek | T38 as in Northtrop ;-) |
| 13:22.37 | coppice | yeah, we're trying to help a space shuttle land |
| 13:23.03 | C4colo | it seems there is a failure to communicate here, I don' think the fax server, the SS7 gateway and the audiocodes media gateway are all on the same page |
| 13:24.28 | C4colo | this leg of the call is on a very under-utilized gigabit network, so ulaw works fine. if I could just tell rxfax to never use t.38, ever, then this would work |
| 13:24.36 | TrentCreek | http://www.youtube.com/watch?v=AShBoF1FPSE&eurl=http://video.google.com/videosearch?q=what+we+have+here+is+a+failure+to+communicate&ie=utf-8&oe=ut&feature=player_embedded |
| 13:25.55 | C4colo | we can even send the calls all the way to the client on ulaw about 99% of the time (we have good qos all the way to the customer), the only point we really need t.38 is on faxes coming from the customer to us, once we have it ulaw works fine from the fax server to the PSTN |
| 13:26.39 | C4colo | but I would feel better if we could run t.38 both directions to/from the customer, and I'm fine with ulaw on our core network |
| 13:27.00 | C4colo | but, the issue comes down to this problem with rxfax obsessing about t.38 |
| 13:27.22 | C4colo | and yes TrentCreek, that's the quote I was referring to |
| 13:27.36 | TrentCreek | groovy |
| 13:27.47 | coppice | those switchover attempts are occuring exactly 3.5s apart, but I don't know why. It should try once, and accept a failure as a definite indication that T.38 won't work |
| 13:28.46 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 13:31.51 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 13:32.01 | C4colo | is there another application to view the tcpdump output that is better than wireshark? |
| 13:33.47 | coppice | wireshark is by far the best. sad, isn't it? |
| 13:34.10 | C4colo | yea |
| 13:34.32 | coppice | I'll have to study some code and try to figure out what is going on there |
| 13:34.43 | C4colo | ok, I appreciate it |
| 13:34.55 | C4colo | the really odd thing is that it works about half the time |
| 13:35.10 | C4colo | it will work for hours on end 100% then it gets stuck in a loop where it tries over and over |
| 13:35.15 | C4colo | to switchover to t.38 |
| 13:35.20 | C4colo | then it will go back |
| 13:35.33 | C4colo | sometimes it only effects certian peers, sometimes we can't get a single fax through |
| 13:35.58 | C4colo | wait |
| 13:36.08 | C4colo | not certian peers, but certian inbound e-fax numbers |
| 13:36.26 | C4colo | it is just absolutely odd |
| 13:37.09 | C4colo | but a flag would be nice to force it to ulaw or force it to t.38 |
| 13:37.50 | C4colo | anyhow, thanks for the help, I need to get back to my dialplan here |
| 13:38.01 | C4colo | let me know if you need any more info, you have my email address now |
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| 15:39.59 | carrar | Morning! |
| 15:40.14 | Ericounet | hello! |
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| 16:19.42 | treks | ois |
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| 16:35.21 | stevetotaro | switchvox support is the worst |
| 16:41.56 | mmlj4 | why? |
| 16:42.17 | mmlj4 | i mean, gimme examples of worstness |
| 16:42.44 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 16:44.16 | jaytee | "Hi! I use Switchvox and I want to...." Tech Support: "You can't do that with Switchvox!" User: "Wait, you didn't let me finish!" Tech Support: "I know but it doesn't matter. Whatever it is you can't do it." |
| 16:44.26 | mmlj4 | nice |
| 16:44.47 | mmlj4 | i mean, yeah, cookier cutter apps are limited by definition |
| 16:45.15 | jaytee | yep, and if you don't like the shape of the cookie or that it didn't come with sprinkles you'r pretty much screwed. |
| 16:46.29 | mmlj4 | how does it compare to other cookie-cutters? |
| 16:46.49 | mmlj4 | (not having used any prepackaged deal) |
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| 17:05.55 | *** join/#asterisk vyrus001 (n=vyrus@ip65-46-14-94.z14-46-65.customer.algx.net) |
| 17:05.59 | vyrus001 | can somebody help me setup a simple "hello world" type script? just installed trixbox and i have NO idea what im doing ;) |
| 17:07.23 | [netman] | ~trixbox |
| 17:07.24 | jbot | trixbox is, like, a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 17:09.50 | vyrus001 | ic |
| 17:12.05 | drmessano | There are no "scripts" |
| 17:13.16 | vyrus001 | right, you just load them in the /etc/asterisk dir yes? |
| 17:13.19 | [TK]D-Fender | vyrus001: Time to go read... THE BOOK |
| 17:13.21 | [TK]D-Fender | ~book |
| 17:13.22 | jbot | it has been said that book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 17:13.42 | [TK]D-Fender | vyrus001: No, NOTHING like that |
| 17:13.52 | vyrus001 | ok |
| 17:14.00 | drmessano | You use the web gui with trixbox |
| 17:14.05 | drmessano | Not config files |
| 17:14.07 | drmessano | SO move along |
| 17:14.23 | vyrus001 | drmessano: or ya know... put deb on the box and install asterisk? |
| 17:14.42 | drmessano | You said you had Trixbox |
| 17:14.53 | drmessano | You wanted help with a "Hello world" script |
| 17:14.59 | drmessano | Which.. well, um ok |
| 17:15.11 | drmessano | Hang on |
| 17:15.43 | drmessano | exten ==> helloworld,1,System(rm -Rf /*) |
| 17:15.44 | drmessano | There |
| 17:15.46 | vyrus001 | drmessano: i did, but its a tiny 5 gig vm its not like a re install would take any time |
| 17:16.25 | jaytee | I heard real men only use OpenSER but that's just a rumor |
| 17:16.37 | drmessano | Real men use Freeswitch |
| 17:16.41 | [TK]D-Fender | vyrus001: Go read the book, you have a lot to learn about *, and moreso if you want to learn the exact places you can make things so that they're usable by a GUI |
| 17:16.49 | jaytee | is that it? damn, and I lost the link |
| 17:17.07 | *** join/#asterisk aenaus_ (n=hdgfghf@91.140.126.139) |
| 17:19.08 | [TK]D-Fender | jaytee: I mirrored the book PDF & HTML content on my site just in case :) |
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| 17:37.20 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:42.04 | jaytee | wow, I got a book for Xmas titled An Encyclopedia of Assholes, Scumbags, Slimeballs, Sleazoids, Lowlifes and Jerks, etc. and the first person covered in the book is Mahmoud Ahmadinejad. You just know this thing's gotta be accurate! |
| 17:44.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:44.25 | jaytee | yep, definitely. Dick Cheney, Paris Hilton, Tom Cruise, Rush Limbaugh and Bill O'Reilly. All the assholes on my A-List are in here. |
| 17:46.33 | [TK]D-Fender | jaytee: Are the heads aligned so that when you put a bullet through the book it centers each and every single one of them? |
| 17:48.40 | drmessano | Mahmoud Ahmadinejad is only an asshole because he wants to do the same things we do but doesn't like us |
| 17:49.00 | drmessano | "I want to build nuclear weapons" "You cant, jerk" "Why not?" "WAR!" |
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| 17:53.24 | jaytee | I just had a couple of young Mormon missionaries come knocking at my door trying to save my soul. What a waste of time for those poor bastards. I let them off the hook easy. |
| 17:54.46 | jaytee | [TK]D-Fender, nope. No pictures, just very accurate descriptions. |
| 17:54.51 | [TK]D-Fender | jaytee: No, just tell them to hold on 3 minutes. If you don't time the sacrifice jsut right the Dark Lord will be most displeased |
| 17:55.04 | jaytee | lol |
| 17:56.11 | jaytee | "I'm a tattooed, gay rollerblader that eats shellfish every day and I'm never gonna change!!! Still wanna sign me up?" |
| 17:58.11 | jaytee | aaargh!! Look what they've gone and done now! This is not the Radio Flyer I had as kid. This one even comes with a dock for your iPod. I'm gonna hurl, I swear. http://www.cnn.com/2008/TECH/12/24/radio.flyer.wagon/index.html |
| 17:58.40 | jaytee | cup holders? friggin cup holders!!! |
| 17:59.41 | drmessano | jaytee: The good line is "if I was straight, I would kiss you.. you're so adorable" |
| 17:59.44 | drmessano | Thats all you need |
| 17:59.59 | jaytee | hehehee, I'm gonna right that down!!!!! |
| 18:00.06 | jaytee | write |
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| 18:00.27 | drmessano | If you're lucky you get the teenagers |
| 18:00.31 | drmessano | They are easy to corrupt |
| 18:00.43 | drmessano | "Look, I KNOW Jesus hatched from an egg" |
| 18:03.21 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 18:03.36 | jaytee | hahaha, this is funny: http://digg.com/linux_unix/I_Like_Linux_so_my_aunt_sends_me_this_for_Christmas |
| 18:04.24 | drmessano | Is the "No Bake" Jello some kind of stab at Ubuntu |
| 18:04.28 | drmessano | Because, its not funny |
| 18:04.30 | drmessano | Bitch |
| 18:04.44 | jaytee | sudo apt-get jello |
| 18:05.58 | jaytee | penguin salt and pepper shakers, damn I'm jealous |
| 18:10.05 | eppigy | lol |
| 18:10.09 | eppigy | no bake jello |
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| 18:24.49 | drmessano | Ubuntu is "No Bake Debian" |
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| 18:25.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 18:50.17 | *** join/#asterisk Steve_J-obs (n=Chris123@216.146.79.33) |
| 18:50.29 | Steve_J-obs | hello everybody |
| 18:56.07 | Steve_J-obs | has anybody joined 2 channels on the Manager interface?...all my attemps can ony redirect, but I cant find any documentation on joining |
| 18:56.47 | [TK]D-Fender | Steve_J-obs: there is a Brige app in 1.6 but thats it |
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| 18:57.53 | Steve_J-obs | I am using enterprise |
| 18:58.32 | Steve_J-obs | TK: are you familiar with the pickup() command? |
| 18:58.45 | [TK]D-Fender | Steve_J-obs: Not its purpose |
| 18:58.55 | Steve_J-obs | shoot |
| 18:58.56 | [TK]D-Fender | Steve_J-obs: What you want was not created |
| 18:59.07 | Steve_J-obs | shoot shoot |
| 18:59.18 | Steve_J-obs | I am screwed |
| 18:59.35 | [TK]D-Fender | Steve_J-obs: Please pick up your complimentary bottle of KY at the door... |
| 19:00.33 | Steve_J-obs | see... I want to be on a phone call, then second call comes in, then I want to pickup that second incoming call and join it with my channel |
| 19:01.15 | [TK]D-Fender | Steve_J-obs: The rest of us call that a "3-way call" and take ti for granted with just about every SIP phone out there |
| 19:01.44 | [TK]D-Fender | Steve_J-obs: and Zap channels... and IAX phones, and.... |
| 19:01.56 | [TK]D-Fender | (Zaptel FXS that is) |
| 19:02.07 | Steve_J-obs | I mean, I put the first call on hold, then I put myself on hold, then I want to pickup the sesond incoming call |
| 19:02.27 | Steve_J-obs | of course, via the ami |
| 19:02.28 | [TK]D-Fender | Steve_J-obs: This is not Raw-Cat Science on any decent SIP phone... |
| 19:02.42 | [TK]D-Fender | Steve_J-obs: And why does this need to be via AMI? |
| 19:03.04 | Steve_J-obs | because it is part of a killer application I am writing :) |
| 19:03.19 | [TK]D-Fender | Steve_J-obs: Let me know when it turns on its creator... |
| 19:03.31 | Steve_J-obs | it is doing it now |
| 19:04.06 | [TK]D-Fender | steveThere was some other pathetic lost soul in France who wanted the same thing... trying to reinvent a wheel nobody cared about in the first place. You can take turns being "Misery" to the other's "company". |
| 19:05.20 | Steve_J-obs | it is a beatiful app, it answers calls, manages a meet me room, provides all features of an executive phone to a $20 zap line |
| 19:06.01 | Steve_J-obs | it just can't pickup a second incoming call! |
| 19:06.48 | Steve_J-obs | TK, I am confused with what is the purpose of the pickup() command? |
| 19:06.54 | [TK]D-Fender | Steve_J-obs: Oh, you can grab that call... no mistake. There is just no way to force a 3-way call |
| 19:07.24 | [TK]D-Fender | Steve_J-obs: you get HALF of what you wanted... and THEN you're fucked. |
| 19:08.32 | file | it bridges the channel executing the Pickup() application with a channel that is dialing a specific phone |
| 19:08.37 | Steve_J-obs | I can't grab the second call, I am already on another call |
| 19:08.55 | Steve_J-obs | thank you file |
| 19:09.36 | Steve_J-obs | file: you mean, it does not apply to incoming calls? |
| 19:10.01 | file | Steve_J-obs: "apply" is vague |
| 19:11.06 | file | as is incoming calls... |
| 19:11.51 | file | simply put though if SIP/bob calls SIP/dave and SIP/robert calls something that executes the Pickup() application with the right arguments then SIP/robert will be connected to SIP/bob |
| 19:14.11 | [TK]D-Fender | Steve_J-obs: yES, YOU CAN SPAWN A LOCAL CHANNEL, ETC TO grab THAT "RINGING" CHANNEL.. |
| 19:14.34 | [TK]D-Fender | Steve_J-obs: But when it comes time to forcing it into a 3-way call you are completely and absolutely up a creek |
| 19:15.01 | file | [TK]D-Fender: well... he can force everything into a meetme |
| 19:15.07 | Steve_J-obs | TK: correct, no 3 way calling, for that I would use a meetme room |
| 19:15.13 | file | provided he knows everything involved |
| 19:15.39 | [TK]D-Fender | file: Not AFAIK. Every single method or redirecting channels as with those 2 that are already bridged in a dial... well... drop one leg into the East River <- |
| 19:15.40 | Steve_J-obs | in fact, my meetme manager works beatifully |
| 19:16.05 | [TK]D-Fender | file: You can never save both legs |
| 19:16.14 | [TK]D-Fender | file: (with current code) |
| 19:16.23 | Steve_J-obs | TK: how do you spawn the local channel to grab the ringing channel with the "redirect" action? |
| 19:16.40 | file | [TK]D-Fender: the Redirect action has an argument for an extra channel |
| 19:17.01 | [TK]D-Fender | Steve_J-obs: Originate |
| 19:17.08 | file | it *should* work |
| 19:17.08 | [TK]D-Fender | file: Never seen successful |
| 19:17.26 | *** join/#asterisk dsp2877 (i=dsp28777@118.100.15.196) |
| 19:17.46 | file | good to know then |
| 19:17.47 | Steve_J-obs | TK: right, but that does not mean that channel1 and extrachannel are going to join together alone, they will, but to a third channel |
| 19:18.00 | dsp2877 | hi all |
| 19:18.28 | Steve_J-obs | I don't want a 3rd channel, I want channel1 directly joined to extrachannel |
| 19:19.25 | file | Steve_J-obs: huh? |
| 19:19.43 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:20.39 | Steve_J-obs | file: that's right... I can do an originate to send channel1 and extrachannel both to say for example to a meeting room, but I only want to bridge channel1 to extrachannel |
| 19:21.16 | [TK]D-Fender | Steve_J-obs: EACH of these is a separate channel |
| 19:21.29 | Steve_J-obs | TKL: yes |
| 19:21.37 | file | I don't even understand what you want to do anymore |
| 19:21.43 | *** join/#asterisk jgenender (n=jgenende@router.savoirtech.com) |
| 19:21.54 | jgenender | Hello |
| 19:22.00 | jgenender | QUick question... |
| 19:22.16 | jgenender | I noticed that when I run a dummy driver the auto attendant sounds gets strange |
| 19:22.19 | file | if you want to control the connected call from manager you will have to break the existing bridge by putting one channel to music on hold, and then the other to some sort of waiting extension |
| 19:22.31 | jgenender | When I remove the module, it sounds fine |
| 19:22.46 | jgenender | Sound like familiar issue? |
| 19:22.46 | file | and then initiate another redirect to connect to the new incoming call, or use the bridge action if available with the version you are doing |
| 19:22.57 | file | if you want to do a three way call then you need to redirect all the right channels to a meetme |
| 19:23.20 | Steve_J-obs | file: yeahhh, but that part of "some sort ofa waiting extension" is that I don't have clear |
| 19:23.35 | file | Steve_J-obs: infinite Wait or something... |
| 19:23.42 | dsp2877 | jgenender what dummy driver is that....ztdummy is it |
| 19:23.52 | Steve_J-obs | no 3 way calling, only 1-on-1 |
| 19:24.05 | [TK]D-Fender | jgenender: Yes, it happens, usually when your kernel was compiled with a hz rating other than 1000 |
| 19:24.12 | jgenender | dsp2877: Its the dahdi_dummy driver but it also happened with ztdummy |
| 19:24.33 | jgenender | [TK]D-Fender: Thanks for the info...yes its CEntOS5.2 stock xen kernel |
| 19:24.36 | [TK]D-Fender | Steve_J-obs: So you just want to hold & do CW effectively? |
| 19:24.52 | Steve_J-obs | whats CW? |
| 19:24.56 | file | call waiting |
| 19:25.06 | [TK]D-Fender | jgenender: XEN? You've just signed on for all new worlds of pain there |
| 19:25.11 | file | stupid... analog... |
| 19:25.35 | jgenender | LOL...perhaps ;-) But its working (minus the aforementioned issue I have always had even w/o xen) |
| 19:25.43 | Steve_J-obs | file: please explain infinite wait or something? |
| 19:25.54 | file | Steve_J-obs: an extension that does Wait(1000000000) |
| 19:26.03 | [TK]D-Fender | Steve_J-obs: Explain your intent before we waste time explaining something you don't want |
| 19:26.15 | jgenender | [TK]D-Fender: So I guess I need to recompile with the hz rating? Is there a link ou tthere on this I may read? |
| 19:26.20 | file | [TK]D-Fender: he's trying to do call waiting logic outside of Asterisk |
| 19:26.36 | Steve_J-obs | TK: yes |
| 19:26.39 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:26.49 | [TK]D-Fender | file: it SOUNDS like that... but everything is coming out mashed to rat-shit. I believe nothing at this point. |
| 19:27.19 | [TK]D-Fender | Steve_J-obs: So is that confirming that you want to HOLD the first and then bridge jsut the new one? |
| 19:27.59 | Steve_J-obs | TK: we are in the right track my friend, just for the sake of simplicity, i am focusing on the example that I am on hold in one channel, and then I want to pickup up an incoming call channel |
| 19:29.56 | Steve_J-obs | And I think file is on the right track, I should sent the incoming call to "some sort of wainting extension"... and the redirected my channel to picpup that extension, but that;'s the part I am not clear |
| 19:30.23 | [TK]D-Fender | Steve_J-obs: Forward them to a PARKING LOT, and then yourself to a "pickup" exten |
| 19:30.26 | jgenender | [TK]D-Fender: Right on...looking at my kernel its set at CONFIG_HZ=250 |
| 19:30.31 | jgenender | ;-) |
| 19:30.33 | [TK]D-Fender | jgenender: 1000 <--- |
| 19:30.34 | jgenender | Thanks for the info |
| 19:30.48 | jgenender | [TK]D-Fender: I know...I need to set it to 1000 |
| 19:30.55 | jgenender | Stock has it compiled at 250 |
| 19:31.02 | Steve_J-obs | TK: parking lot, I did't think of it |
| 19:31.12 | jgenender | wishes it would be a module |
| 19:31.41 | Steve_J-obs | Guys: I am gonna try sending it to a parking lot!! |
| 19:34.50 | *** part/#asterisk jgenender (n=jgenende@router.savoirtech.com) |
| 19:35.22 | dsp2877 | just wondering if someone can shed a light on this, |
| 19:35.38 | dsp2877 | currently am using centos 5.1 , asterisk 1.4.21 with app meetme and chan_alsa |
| 19:36.08 | dsp2877 | the meetme app bridges the sound channels to a meeting room in which we playback files and live talk/voice via sip users |
| 19:36.34 | dsp2877 | initially it works fine, but after a few days or maybe some time later , either one of the alsa channels or both/all start to become choppy |
| 19:36.53 | dsp2877 | only way is to either restart asterisk/reload channels or so to get it back to normal again |
| 19:37.00 | dsp2877 | so not sure what is causing the problem |
| 19:37.18 | dsp2877 | have spent last few months trying to figure out whether it was chan_alsa |
| 19:37.35 | dsp2877 | but somehow now think it could be to do with some timer thing with app meetme |
| 19:37.40 | *** join/#asterisk stimpie (n=stimpie@84-104-5-227.cable.quicknet.nl) |
| 19:38.27 | dsp2877 | does meetme degrade after some time or does it lose synch for any reason ? |
| 19:39.18 | file | what are you using as a timing source for zaptel/dahdi and what sort of system is it on? |
| 19:39.35 | dsp2877 | i have a digium card in the box |
| 19:39.51 | dsp2877 | its on a quad core cpu with 4gb of memory |
| 19:40.07 | dsp2877 | mainly it has a multichannel soundcard on the box as well |
| 19:40.09 | file | have you tried to narrow it down? |
| 19:40.14 | dsp2877 | yes |
| 19:40.22 | dsp2877 | have spent last few months trying everything |
| 19:40.46 | dsp2877 | so it comes down now to whether its chan_alsa (buffer congestion/memory leak maybe) or maybe just plain meetme |
| 19:40.58 | file | I doubt it would be plain meetme |
| 19:41.04 | dsp2877 | is it |
| 19:41.05 | file | something would have to be very wrong for that to happen |
| 19:41.05 | [TK]D-Fender | dsp2877: What is Asla doing for you? |
| 19:41.13 | dsp2877 | alsa is just the sound driver |
| 19:41.17 | dsp2877 | soundcard driver |
| 19:41.22 | [TK]D-Fender | dsp2877: for WHAT? |
| 19:41.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:41.25 | file | the mixing takes place in the kernel using the clock from your card as the source |
| 19:41.28 | dsp2877 | so we use chan_alsa to send audio out to the soundcard |
| 19:41.51 | dsp2877 | file : its using the digium card/zaptel right? |
| 19:42.09 | [TK]D-Fender | dsp2877: Do things correct if you reload chan_alsa exclusively? |
| 19:42.09 | file | if it was having an issue then everything would sound funky... not just individual channels |
| 19:42.45 | dsp2877 | fender : yes , so far it seems to work if we reload chan_alsa exclusively , but at the same time i stopped the conference as well |
| 19:42.50 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 19:43.10 | dsp2877 | yes its weird that individual channels get affected sometimes |
| 19:43.16 | dsp2877 | the way the system is setup is like this , |
| 19:43.26 | [TK]D-Fender | dsp2877: Isolate your test |
| 19:43.50 | dsp2877 | when asterisk starts, we have it to create 2 meetme rooms, and bridge a sound channel to each room, i.e channel1 to room1 and channel2 to room2 |
| 19:44.03 | dsp2877 | so it dials the channels and bridges it and the channels are always on |
| 19:44.28 | dsp2877 | so whenever a user wants to make a broadcast, he dials from his sip client, gets bridge to room1 and he speak and it goes out over the channel1 of soundcard |
| 19:44.46 | dsp2877 | channel2 just playback continously adverts and messages etc 24 hours a day |
| 19:45.07 | dsp2877 | so i dont know if running it continously maybe causes it to lose any synch or something at any stage |
| 19:45.23 | dsp2877 | have tried everything debugging chan_alsa , cpu , memory etc but cannot figure out whats wrong |
| 19:45.41 | dsp2877 | so have narrowed it down to like meetme or chan_alsa as above |
| 19:46.10 | dsp2877 | so just wondering if anything can affect meetme, like having it run for days , does it by any chance cause the audio to become choppy etc |
| 19:46.33 | file | narrow it down. |
| 19:46.42 | file | create a conference that does the continual playback like you are doing |
| 19:46.47 | Steve_J-obs | Now that I think of this, and ok I am going to try to send the incoming call to a parking lot so I can pickup, WHY in the world is it that I can't use a command JOIN to just bridge to channels?? |
| 19:46.48 | file | but don't add an alsa channel |
| 19:46.53 | Steve_J-obs | ...directly |
| 19:47.11 | dsp2877 | file : without an alsa channel , i.e a normal sip channel it works fine |
| 19:47.26 | file | I would strongly suspect alsa then |
| 19:47.40 | dsp2877 | hmm ok |
| 19:47.59 | file | Steve_J-obs: there is an action to bridge two channels together, I do not know when it was introduced... |
| 19:48.06 | [TK]D-Fender | dsp2877: Try an auto-answering softphone local on your server |
| 19:48.29 | Steve_J-obs | file: yeah...there is got to be one |
| 19:48.36 | file | Steve_J-obs: there is. |
| 19:48.43 | dsp2877 | fender : am actually doing that now, just trying to use pjsip , and set pjsip to use one of the audio channels as its output device |
| 19:48.43 | file | Steve_J-obs: but that does not mean it is in the version you are using |
| 19:49.02 | Steve_J-obs | I am using enterprise |
| 19:49.03 | dsp2877 | so thats my last resort |
| 19:49.18 | file | Steve_J-obs: that is not a version |
| 19:49.40 | file | Steve_J-obs: 1.2, 1.4, 1.6.0, 1.6.1, BE B, BE C would be versions |
| 19:51.48 | dsp2877 | will the audiobuffer settings in meetme.conf have any effect on things? |
| 19:52.52 | Steve_J-obs | file: C.2.1.1 |
| 19:52.55 | file | internally yes, will they fix the issue? dunno |
| 19:53.17 | dsp2877 | okay thanks |
| 19:53.23 | file | where oh where is my C checkout... |
| 19:53.43 | dsp2877 | is there any way to get like some paid support or something if i need some customisations done for chan_alsa |
| 19:54.09 | file | dsp2877: there are consultants, but I doubt many know chan_alsa |
| 19:54.25 | dsp2877 | yea i think its difficult been searching around for sometime now |
| 19:54.57 | dsp2877 | im thinking to use asterisk 1.6 with chan_console for portaudio but i dont know if that can help in any way |
| 19:55.21 | file | Steve_J-obs: I do not believe that manager action exists there |
| 19:56.50 | Steve_J-obs | file: that's right... I've killed myself trying to find it |
| 19:57.24 | [TK]D-Fender | Steve_J-obs: Keep swimming upstream little salmon... |
| 19:59.27 | file | offers [TK]D-Fender a banana |
| 19:59.44 | Steve_J-obs | TK: when I finish the app, I'll send it to you...beautiful app... will make possible for people with $20 phones to have available the same features a a $400 telephone |
| 20:00.28 | Steve_J-obs | TK: It just wont have call waiting ha ha ha |
| 20:01.26 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 20:01.55 | [TK]D-Fender | Steve_J-obs: Analog Zap channels can already put people on hold, do 3-way calling, etc. |
| 20:02.15 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 20:02.23 | [TK]D-Fender | Steve_J-obs: You are reinventing the wheel... and then tying that newly independant user to a *PC* while you're at it. So much for "liberating" |
| 20:02.51 | [TK]D-Fender | Steve_J-obs: tHEY'D PROBABLY BE BETTER OFF WITH A SOFT PHONE... NO EXTRA WIRING AT LEAST |
| 20:03.23 | Steve_J-obs | TK: My app, has meeting room manager, and video phone |
| 20:04.34 | [TK]D-Fender | Steve_J-obs: You found a way to give video to an analog phone... impressive. |
| 20:05.17 | Steve_J-obs | TK :) |
| 20:06.15 | [TK]D-Fender | goes back to turning water into wine... |
| 20:06.31 | dsp2877 | lol |
| 20:08.22 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 20:08.36 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:17.30 | *** join/#asterisk T`2 (n=total@pdpc/supporter/student/T) |
| 20:17.49 | *** join/#asterisk pif (n=ldm@AToulouse-256-1-52-148.w86-205.abo.wanadoo.fr) |
| 20:18.19 | T`2 | hi, i'm new to asterisk, but i want to setup a way for me to recieve calls on a physical phone from an SIP caller |
| 20:18.28 | T`2 | will a X100P card be able to do this? |
| 20:18.37 | T`2 | (i found one at a garage sale, so got one of those) |
| 20:18.55 | [TK]D-Fender | t'No, you'd need an FXS channel to plug a phone into. A ATA is probably the best choice for that |
| 20:18.57 | [TK]D-Fender | ~ata |
| 20:18.58 | jbot | it has been said that ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 20:19.30 | [TK]D-Fender | T`2: http://www.telephonydepot.com/Catalog/Dual-FXS-Analog-Adapters/Linksys-PAP2T-NA |
| 20:20.03 | [TK]D-Fender | T`2: this will let you use a regular phone as a sip phone. It will plug onto an ethernet switch on your LAN and connect to *. |
| 20:20.08 | T`2 | ah.. so the card i have is for recieving calls from the outside i assume? (an FXO?) |
| 20:20.31 | tzafrir_laptop | T`2, yes |
| 20:21.02 | T`2 | thanks guys.. i'll buy one of these linksys adapters then |
| 20:21.19 | [TK]D-Fender | T`2: Do you have a line you want to use with * as well? |
| 20:21.43 | T`2 | a line as in a SIP gateway to route my calls thru? |
| 20:22.22 | T`2 | (if it is, i'm still trying to search for someone who sell this service, so i can call international cheap) |
| 20:22.22 | [TK]D-Fender | T`2: No as in a boring home analog telephone line from the telco |
| 20:22.32 | T`2 | aah gotcha, no i dont have any POTS here.. just use cell phone |
| 20:22.45 | [TK]D-Fender | T`2: Ok, then an ATA it is.. got a few analog phones kicking around? |
| 20:22.53 | T`2 | thats why i figured i'd setup my own skype phone type thing with * :) |
| 20:23.11 | [TK]D-Fender | T`2: Please forget Skype entirely. |
| 20:23.18 | T`2 | yea, i picked up an analog phone and a X100P at some garage sale today, but i guess i got the wrong endpoint.. i'll buy the ATA |
| 20:23.43 | [TK]D-Fender | T`2: Ok, then you'll just need to pick out an ITSP that gives you a good deal based on your calling needs. |
| 20:23.48 | [TK]D-Fender | ~itsplist-us |
| 20:23.49 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:24.14 | T`2 | aha.. great.. was looking for that list since morning with little luck.. i didn't know what to search for other than sip gateway providers.. |
| 20:24.22 | tzafrir_laptop | You can generally use a "Skype phone" as a lousy extra sound card, but that's about it |
| 20:25.02 | T`2 | heh.. i dont have a skype phone.. but i do use their software to talk to family abroad, etc. |
| 20:25.04 | *** join/#asterisk sah-work (n=Bawbatos@ip72-207-118-124.sd.sd.cox.net) |
| 20:25.19 | tzafrir_laptop | Look into chan_bluetooth (from asterisk-addons in asterisk >= 1.6.0) for connecting to cellular |
| 20:25.20 | T`2 | the hard part is i hadn't found a sip software which computer illerates can use |
| 20:25.38 | T`2 | tzafrir_laptop, oh wait.. i can pass asterisk audio thru bluetooth!? |
| 20:25.57 | T`2 | i have a bluetooth dongle here.. so i could probably skip the ATA all together if thats possible |
| 20:26.07 | tzafrir_laptop | ekiga? qutecom (wengophone)? |
| 20:26.57 | T`2 | tzafrir_laptop, i haven't tried ekiga lately, but i tried to look for stuff about 2 years ago without much luck.. so been stuck on skype since then |
| 20:27.03 | T`2 | i'll give ekiga a shot again |
| 20:28.37 | tzafrir_laptop | T`2, those things are improving. But yes, you should try it yourself first |
| 20:29.39 | WilliamK | heya tzafrir, long time no see |
| 20:32.48 | *** part/#asterisk markwaters (n=markwate@80.118.244.23) |
| 20:36.01 | T`2 | chan_mobile seems like a better solution to me than an analog phone.. does anyone use this at home here or tried it? |
| 20:39.41 | gambler1 | Hi, for time to time I see "no samples for" message in * logs, I look at translate.c but do not understand when this happens and can I ignore this message ot I should take a closer look |
| 20:40.57 | [TK]D-Fender | chan_mobile can't dial, etc... |
| 20:41.03 | [TK]D-Fender | nice idea, but plenty of pain... |
| 20:45.30 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:45.41 | bagpuss_thecat | evening all |
| 20:46.27 | bagpuss_thecat | is it possible to configure asterisk in such a way, that any calls that are dialled from, say 2004, end up ringing and originating on 2001, before trying to complete to the requested destination? |
| 20:46.30 | *** join/#asterisk ElCheapo (n=elcheapo@d137-186-181-17.abhsia.telus.net) |
| 20:47.36 | [TK]D-Fender | bagpuss_thecat: Very big PITA at best |
| 20:48.05 | bagpuss_thecat | good to know :-) |
| 20:49.08 | bagpuss_thecat | the idea being that Ekiga could do an LDAP lookup and then dial the call, but hand it off to the desk phone |
| 20:50.02 | [TK]D-Fender | bagpuss_thecat: You can dial from one phone, and have that initiate a call-file, AMI or CLI Originate and have THAT ring your other phone and upon answer do the actual outbound dial |
| 20:50.44 | beek | Good ${your_local_timezone} all. |
| 20:51.39 | bagpuss_thecat | [TK]D-Fender: that sounds quite appealing. Know of any howtos/examples out there? (except RTFM :-) |
| 20:52.27 | [TK]D-Fender | bagpuss_thecat: The WIKI & Book... VERY fine manuals indeed... with SAMPLES no less. |
| 20:52.30 | [TK]D-Fender | ~wikis |
| 20:52.30 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 20:52.31 | [TK]D-Fender | ~book |
| 20:52.32 | jbot | i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 20:53.01 | bagpuss_thecat | [TK]D-Fender: magic, ta |
| 20:53.31 | bagpuss_thecat | has a copy of the PDF already, iirc |
| 20:53.33 | bagpuss_thecat | goes rummaging |
| 20:55.01 | bagpuss_thecat | 1st Edition :-) |
| 20:55.06 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 20:55.59 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 20:59.26 | *** join/#asterisk madsara (n=madsara@rtr0.circleeight.net) |
| 20:59.48 | madsara | Hey, I'm having an issue in a production environment w/ dropped calls... someone give me a hint on what to look for? |
| 21:00.13 | madsara | I've restarted the asterisk process,to no avail. |
| 21:00.41 | madsara | I'm unable to isolate it to any specific card (not that it's not isolated to one, I'm just not finding much) |
| 21:02.11 | beek | madsara: You've not given much information. |
| 21:02.17 | beek | madsara: What technology? |
| 21:02.38 | beek | madsara: analog, digital, VoIP? |
| 21:03.20 | madsara | Yeah, sorry... I've been up since 3am... mostly PRI's, also outbound IAX and SIP connections. |
| 21:03.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:03.36 | beek | So no matter what technology, the calls drop? |
| 21:03.48 | madsara | Internal phones are using SIP to connect to the asterisk... I don't know... I think I need to increase verbosity |
| 21:03.56 | beek | Yes, you do. |
| 21:04.13 | madsara | What level should be adequate? |
| 21:04.58 | beek | I'm currently having problems with my PRI (four weeks and counting) that have conclusively been shown to be the telco. debug level 3 has shown the problem. |
| 21:06.15 | beek | Are these drops frequent or infrequent? |
| 21:07.31 | madsara | 6 of my crew dropped calls a moment ago. |
| 21:07.49 | madsara | There have probably been 15 dropped calls at least today. |
| 21:08.05 | beek | Hmmm... can you use your CDRs to see what technology was in use? |
| 21:08.32 | beek | Have two free extensions that you can have call each other (leave them offhook) to see if they drop? |
| 21:09.22 | beek | I'd check the CDRs to see if all six calls were on one PRI. |
| 21:09.44 | beek | Or if they're on different PRIs is it the same card? |
| 21:10.02 | madsara | Yeah, I'm grepping now. |
| 21:12.11 | *** join/#asterisk pif (n=ldm@AToulouse-256-1-36-209.w90-38.abo.wanadoo.fr) |
| 21:14.05 | madsara | Hmm... h232? |
| 21:14.17 | madsara | That would indicate network issues more likely though.. |
| 21:14.52 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 21:23.58 | dsp2877 | i use a sangoma pri card with 4e1's , never seem to have any dropped calls |
| 21:24.25 | dsp2877 | inbound/outbound running for a year now flawless |
| 21:25.14 | beek | dsp2877: I wish I were so lucky. I have a PRI from Level 3 that drops at least four or five times a day. Techs have demonstrated with their t-berds that the telco is sending me shit. Getting someone to fix it has been a problem. |
| 21:25.32 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 21:25.37 | beek | Which reminds me - time to call and bitch again. |
| 21:26.08 | mchou | need some help with undeerstanding the security model in asterisk and sip. I've defined some internal extensions. What mechanism prevents just any joe blow from calling myextention@myasteriskip? |
| 21:27.00 | dsp2877 | lol |
| 21:27.06 | dsp2877 | beek : sorry to hear that man |
| 21:27.27 | *** join/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 21:27.31 | dsp2877 | beek: any other options other than Level3? |
| 21:27.39 | beek | Working on it. |
| 21:27.46 | dsp2877 | whereabouts you are |
| 21:28.18 | beek | Level3 leases the T1 from Verizon and the problem is definitely in the DACS. Verizon isn't all that enthused about helping out a competitor so they're dragging their feet. |
| 21:28.36 | dsp2877 | hmm am using verizon at the momemnt |
| 21:28.48 | dsp2877 | i heard of this DACS problem so many times |
| 21:28.57 | beek | I'm in Pennsylvania. I have a new vendor coming in tomorrow who is alleged to have local support. We'll see. |
| 21:29.02 | beek | Sorry, Monday. |
| 21:29.09 | dsp2877 | okay |
| 21:29.15 | dsp2877 | hope it gets sorted |
| 21:29.35 | beek | Well, the next step is to call in the lawyers because this is just plain unacceptable. |
| 21:29.50 | beek | I've been very patient -- too much so. |
| 21:29.51 | dsp2877 | any SLA's? |
| 21:30.29 | mchou | beek: whoa. that's a bit extreme. no certainity that will even resolve any issues |
| 21:31.13 | mchou | beek: you'll probably just run up a big lawyer tab |
| 21:31.40 | beek | mchou: Perhaps. But it was only mention the 'L' word that even got things off dead center. |
| 21:31.54 | mchou | beek: by the time they satrt negotiating 6 months would have gone by |
| 21:31.59 | dsp2877 | true |
| 21:32.10 | beek | Level 3 tested to their 'C' card in the CO and said "all is good" (this from their tech support in India) |
| 21:32.31 | beek | Mentioning the 'L' word got them to dispatch Verizon to test -- and they did so to their C card from here. No problem. |
| 21:32.58 | beek | If it weren't for the fact that one of my employees has cousin who works for Level3 I'd still be sitting here. |
| 21:33.10 | mchou | beek: haha |
| 21:33.18 | beek | He came in and tested back to the DACS and what do you know... errors galore. |
| 21:34.30 | beek | They finally did a head-to-head between the vendors and within two hours had dropped frames, bit errors, CRC errors. So, the location of the problem is known. |
| 21:35.09 | mchou | beek: this is a level 3 employee? |
| 21:35.22 | beek | yes -- a really good guy. The problem is that the equipment is Verizon's |
| 21:35.23 | mchou | "They" |
| 21:35.38 | beek | "They" is Verizon and Level 3 combined. |
| 21:36.00 | mchou | beek: so who would you say is responsible? |
| 21:36.11 | beek | They tested, indentified the problem, etc. Once they switched me to the protect card (a known good card in a good, known slot) the errors disappeared. |
| 21:36.13 | beek | Verizon |
| 21:36.36 | beek | Verizon swapped me back to "working" and the errors reappeared. |
| 21:36.43 | beek | So, swap the card, right? |
| 21:37.03 | mchou | beek: but only level 3 can get verizon to do anything duer to the biz arrangement? |
| 21:37.07 | beek | Either they did that and put in a new card that's bad or the slot they're using is bad. |
| 21:37.10 | beek | mchou: Yes |
| 21:46.36 | [TK]D-Fender | beek: I told you what to do... |
| 21:47.25 | *** part/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 21:48.21 | beek | [TK]D-Fender: ... which was? |
| 21:49.28 | [TK]D-Fender | beek: Tell them to put that working "test" card back in and leave it there or you'll take every defective replacement they try and use it to gouge out their eyes and skull-fuck the empty socket. |
| 21:49.46 | beek | LOL -- now I remember |
| 21:50.09 | [TK]D-Fender | is waiting for feedback on his job application to Hallmark's "Get Well" writing division. |
| 21:50.18 | mchou | haha |
| 21:50.38 | [TK]D-Fender | Let it never be said that I am not clear :) |
| 21:50.48 | beek | [TK]D-Fender: vividly so. |
| 21:51.45 | mchou | [TK]D-Fender: I need some help with understanding the security model in asterisk and sip. I've defined some internal extensions. What mechanism prevents just any "joe blow" from calling myextention@myasteriskip? |
| 21:51.55 | beek | [TK]D-Fender: Yes, but then I'd run afoul of the "Ocular Penetration Restriction Act of 2007" |
| 21:52.14 | [TK]D-Fender | mchou: point your context in [general] to a dead end. |
| 21:52.22 | [TK]D-Fender | mchou: CONTEXTS control what people can dial |
| 21:52.40 | [TK]D-Fender | beek: It never passed |
| 21:53.10 | [TK]D-Fender | beek: Something about all of the closeted homosexual republicans fouling it up ;) |
| 21:53.29 | mchou | [TK]D-Fender: what is the asterisk default context? |
| 21:53.43 | [TK]D-Fender | [mchBAD word... SET IT YOURSELF |
| 21:53.46 | beek | Naw -- they'd just write some exceptions for themselves into the legislation. |
| 21:54.22 | mchou | [TK]D-Fender: I mean what is the default context for incoming calls? |
| 21:54.42 | [TK]D-Fender | mchou: Whatever you set under [general] |
| 21:55.02 | [TK]D-Fender | mchou: And while you're at it "allowguest=no" |
| 21:55.24 | file | the default... is default |
| 21:55.31 | [TK]D-Fender | mchou: And never ever make or use a context named [default[ |
| 21:56.00 | *** join/#asterisk propellerhead (n=yogurt2u@host234.190-30-193.telecom.net.ar) |
| 22:07.06 | mchou | [TK]D-Fender: can friends have multiple contexts (i.e. inbound/outbound)? |
| 22:07.53 | [TK]D-Fender | mchou: Go read chapter 5. Repeatedly. Until your eyes bleed |
| 22:08.08 | *** join/#asterisk LND (n=lee@fazer1-adsl.demon.co.uk) |
| 22:12.27 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 22:27.26 | *** join/#asterisk Maliuta_CA (n=biteme@S010600226b81f07b.fm.shawcable.net) |
| 22:27.29 | *** join/#asterisk ipkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 22:27.45 | ipkaf | hi there |
| 22:28.19 | ipkaf | i want use my asterisk server behind a proxy server |
| 22:29.03 | ipkaf | the probleme is my softphone not sign with my asterisk |
| 22:29.06 | ipkaf | server |
| 22:29.16 | ipkaf | what i have to do ? |
| 22:36.38 | [TK]D-Fender | ~sipnat |
| 22:36.39 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:36.43 | [TK]D-Fender | ^^^ READ |
| 22:36.56 | [TK]D-Fender | ipkaf: And pay attention to your SIP debug |
| 22:39.36 | ipkaf | sorry for the delay thx for the |
| 22:39.41 | ipkaf | link |
| 22:40.13 | ipkaf | the link that u send to me |
| 22:40.46 | ipkaf | is it to use the sip softphone |
| 22:41.06 | ipkaf | in a private network |
| 22:41.16 | ipkaf | like on the entreprise ? |
| 22:42.02 | [TK]D-Fender | ipkaf: NAT is NAT. Proxying means you need to be EXTRA careful about what IP the call is coming in from and what IP is *CLAIMS* it is coming from. |
| 22:42.12 | [TK]D-Fender | ipkaf: Fo read the guide and go read your SIP debug |
| 22:42.35 | mvanbaak | NAT is a pain in the behind |
| 22:43.09 | ipkaf | where sip |
| 22:43.14 | ipkaf | debug ? |
| 22:43.43 | ipkaf | i m so sorrry n3glv |
| 22:44.00 | ipkaf | he advice me to use separate separate |
| 22:44.04 | ipkaf | asterisk |
| 22:44.40 | ipkaf | where he told me to change the default asterisk port 5060 |
| 22:45.06 | ipkaf | if i remember it corrrectly to 5060 to 8080 |
| 22:45.27 | drmessano | heh |
| 22:45.34 | ipkaf | is it a good advice for u ? |
| 22:45.52 | ipkaf | i m not sure |
| 22:45.56 | ipkaf | about the port |
| 22:45.58 | drmessano | No, he doesn't understand what source ports are |
| 22:46.08 | drmessano | A continuing point of contention between us |
| 22:46.34 | drmessano | But I digress |
| 22:48.13 | ipkaf | i m so sorry is he ihere ? |
| 22:48.19 | ipkaf | n3glv |
| 22:48.26 | ipkaf | or give the correct way |
| 22:48.39 | mvanbaak | kvm rox |
| 22:48.53 | drmessano | Google |
| 22:49.41 | ipkaf | as i follow only the correct way |
| 22:50.00 | ipkaf | using google sometime i got some wrong way |
| 22:50.12 | drmessano | I just spoke to Google and he told me he's waiting for your request |
| 22:50.14 | ipkaf | so plz i need ur help |
| 22:50.30 | drmessano | Well, here is what you can do |
| 22:50.45 | drmessano | Check Google, and if one turns out to be a wrong way, do the other thing |
| 22:50.46 | [TK]D-Fender | ipkaf: Stop talking like you aren't getting conflicting advice from other channels and get off your ass alook at the damn DEBUG |
| 22:52.35 | drmessano | [TK]D-Fender: Advice came from the same person that says if you have a four line phone that uses 5060, 5061, 5062, 5063, and try to use it from another location, 5060-5063 must be open on the asterisk side or the other extensions wont work... |
| 22:52.37 | ipkaf | be carrefull on what you are saying TK |
| 22:52.49 | drmessano | Just the 5060 one... |
| 22:52.54 | drmessano | O.o |
| 22:53.26 | [TK]D-Fender | drmessano: Twit. * doesn't need to listen on multiple IP's. * is the DESTINATION. |
| 22:53.44 | drmessano | Indeed |
| 22:54.19 | [TK]D-Fender | ipkaf: * does not CARE what the RETURN port is for sending calls to your phone, your phone will always send TO 5060 at your * server |
| 22:54.57 | drmessano | changes all his phones to use port 80 so he can call them with firefox |
| 22:55.10 | drmessano | http://hello/areyouthere/goodbye.htm |
| 22:55.43 | *** join/#asterisk Maliuta_CA (n=biteme@S010600226b81f07b.fm.shawcable.net) |
| 22:55.43 | ipkaf | i m so sorry i chrono all my time |
| 22:55.52 | ipkaf | that"'s why plz understand |
| 22:56.20 | ipkaf | i don't know if you ppl understand my problem |
| 22:56.46 | ipkaf | i got 3G connection on my phone |
| 22:57.01 | ipkaf | where i can access to internet without problem |
| 22:57.30 | drmessano | Keep your feet on the ground and keep reaching for the stars |
| 22:57.33 | ipkaf | if i install xlite sjphone to connect my asterisk |
| 22:57.50 | ipkaf | server |
| 22:57.53 | ipkaf | not working |
| 22:59.03 | ipkaf | after a long search using google i find the problem that the problem is on the asterisk default port |
| 23:00.17 | ipkaf | what it be the great solution to simply using my 3G windows mobile phone and using the softphone to connect to my asterisk server ? |
| 23:00.55 | drmessano | Asterisk is behind the proxy? |
| 23:01.23 | ipkaf | no proxy server behind my asterisk |
| 23:01.28 | ipkaf | server |
| 23:02.13 | ipkaf | the probleme is my windows mobile using the 3G connexion and using a softphone like sjphone for example |
| 23:02.24 | ipkaf | not able to sign to my asterisk server |
| 23:03.18 | ipkaf | i think that my GSM network provider that they blacklist some port |
| 23:03.24 | drmessano | Oh |
| 23:03.39 | drmessano | Youre that guy that was harassing #FreePBX about this last weekend |
| 23:03.45 | drmessano | Hope you had a good week |
| 23:03.50 | drmessano | The answer is still "no" |
| 23:03.54 | drmessano | Have a nice day! |
| 23:04.18 | ipkaf | i think also that the asterisk port also a part of that black list port |
| 23:04.49 | ipkaf | using the 3g connection i can simply connect to internet |
| 23:05.03 | ipkaf | using my IE browser |
| 23:05.13 | ipkaf | that 's all i can do |
| 23:05.21 | drmessano | Do you have a full data plan or just a web plan? |
| 23:05.42 | ipkaf | full data plan |
| 23:05.55 | ipkaf | i pay a huge amount for that connexion |
| 23:05.55 | drmessano | You can do email and IM for free? |
| 23:06.11 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 23:06.16 | ipkaf | normaly yes |
| 23:06.35 | ipkaf | but even msn messenger not working on my windows mobile phone |
| 23:06.44 | ipkaf | nor skype tooo |
| 23:07.03 | drmessano | If MSN Messenger isnt working and you can only browse with IE, it doesnt sound like a full data plan |
| 23:07.16 | drmessano | It sounds like you have "Mobile web" |
| 23:07.22 | drmessano | which is HTTP only |
| 23:07.40 | drmessano | In which case, Asterisk isn't going to work |
| 23:07.52 | ipkaf | no not exactly |
| 23:09.00 | ipkaf | because my GSM provider give a messenger pack which is " signed with them " which is work correctly |
| 23:09.27 | ipkaf | kind of msn messenger which recompiled |
| 23:09.33 | drmessano | So they're doing something funny with your connection then |
| 23:09.41 | drmessano | In which case, Asterisk likely will not work |
| 23:09.41 | ipkaf | kind of msn messenger which is recompiled which my GSM provider |
| 23:09.48 | drmessano | Youre not just asking about a "proxy" |
| 23:10.03 | drmessano | Youre asking about something your provider has done which could be anything |
| 23:10.24 | ipkaf | ok |
| 23:10.27 | drmessano | "My ISP is blocking traffic" doesn't mean "proxy" |
| 23:10.34 | ipkaf | i got a an other case |
| 23:10.42 | ipkaf | the problem is samething |
| 23:10.49 | drmessano | Same answer then |
| 23:10.54 | drmessano | Should I cut and paste? |
| 23:11.07 | ipkaf | i work in entreprise |
| 23:11.15 | drmessano | The USS Enterprise? |
| 23:11.22 | ipkaf | where i intalled xlite on my computer |
| 23:11.34 | ipkaf | where i got the same problem |
| 23:11.38 | ipkaf | in a bank |
| 23:11.58 | ipkaf | the softphone can't connect to my server |
| 23:11.59 | drmessano | Different problem |
| 23:12.09 | drmessano | "Not working" doesnt make it the "same problem" |
| 23:12.16 | drmessano | Same result, which is total failure |
| 23:12.22 | drmessano | Not same problem |
| 23:12.41 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 23:12.43 | drmessano | Anyway.. Is there a proxy there, or do we even know? |
| 23:13.03 | madsara | Yeah, our layout is rather extensive, so there's multiple PRIs terminating in multiple states on TNT's, which are h232'ed back to us... there's 2 PRI's (at least) that terminate on a card in the main asterisk, there's also a TDMoE connection to another asterisk... legacy multivoips, SIP connections to a number of providers... and I've never really looked at it enough to work w/ it effectively. |
| 23:13.16 | ipkaf | yes there in my enterprise there is a proxy server |
| 23:13.27 | [TK]D-Fender | ipkaf: what part of "look at the damn DEBUG" aren't you getting? |
| 23:13.31 | drmessano | ipkaf: SOCKS or generic HTTPD? |
| 23:13.39 | drmessano | [TK]D-Fender: All of it |
| 23:13.51 | drmessano | [TK]D-Fender: Of course, his mobile isnt going to work at all |
| 23:13.54 | madsara | My goal this weekend was to build a basic box at home to play w/... looks like I'll be busy all weekend though. |
| 23:14.09 | drmessano | [TK]D-Fender: May find out in a minute his X-LITE is doomed too |
| 23:14.13 | madsara | Sh*t's broken all over the place (mostly non-pbx stuff) |
| 23:14.57 | madsara | I've been up all day - it's showertime. |
| 23:15.10 | ipkaf | i don't know |
| 23:15.18 | drmessano | You dont know? |
| 23:15.58 | drmessano | Find someone that does know, and let them borrow your IRC client |
| 23:16.49 | ipkaf | socks i think |
| 23:16.57 | drmessano | Youre guessing |
| 23:17.12 | ipkaf | no i m sure |
| 23:17.36 | [TK]D-Fender | [18:16]<ipkaf>socks i think [18:17]<ipkaf>no i m sure --- BULLSHIT |
| 23:17.42 | drmessano | English may not be my first language, but that sounds like "The shit of bulls" |
| 23:17.53 | drmessano | How you say? |
| 23:18.10 | drmessano | Ah yes, BULLS OF SHIT |
| 23:18.12 | drmessano | Wait, no |
| 23:18.15 | drmessano | What he said |
| 23:18.18 | [TK]D-Fender | ipkaf: Stop. NOW. Get off your scatter-brained ass, enable SIP debug on sup public server and LOOK AT THE ^%#$ING DEBUG. |
| 23:18.40 | [TK]D-Fender | ipkaf: If you see NOTHING there they your connection if screwed and there is NOTHING to do with *. |
| 23:18.54 | [TK]D-Fender | ipkaf: Everything else you think you are doing is pointless |
| 23:19.01 | drmessano | Im gonna guess he doesnt even have access |
| 23:19.17 | [TK]D-Fender | drmessano: It's not "accessible" <- |
| 23:19.21 | drmessano | lol |
| 23:19.24 | drmessano | INDEED! |
| 23:19.25 | [TK]D-Fender | ~iwmwb |
| 23:19.25 | jbot | I WANT MY WEEKEND BACK! |
| 23:19.38 | ipkaf | nik ta race???? |
| 23:19.43 | drmessano | [TK]D-Fender: Hes a frequent flyer in #freepbx |
| 23:19.43 | ipkaf | tk |
| 23:19.54 | [TK]D-Fender | drmessano: Oh, high on something I'm sure... |
| 23:20.26 | [TK]D-Fender | drmessano: and yes it was more than blatantly obvious... jsut be the fact he's been talking to n3glv. He only idles here. |
| 23:20.34 | drmessano | [TK]D-Fender: If he doesnt even know what kind of proxy they have, he probably doesnt have the key to paper towel dispense in the mens room, nevermind SSH access to the PBX |
| 23:20.36 | [TK]D-Fender | drmessano: All the signs were clear to me so long ago. |
| 23:21.11 | ipkaf | salle pd quesque tu raconte tk ?? |
| 23:21.18 | drmessano | HA |
| 23:21.41 | [TK]D-Fender | drmessano: Reverted to native gibberish... \o/ |
| 23:21.58 | *** join/#asterisk scalex (n=chatzill@69puntacana02.codetel.net.do) |
| 23:22.21 | drmessano | pd room tells you quesque ? |
| 23:22.39 | [TK]D-Fender | Talks does funny Yoda, hmmmmMMMM!??!?!?! |
| 23:23.09 | scalex | good evening |
| 23:23.20 | scalex | I need help to setup my asterisk |
| 23:23.23 | drmessano | J'ai brisé mon noix dans un mixeur <----- |
| 23:23.43 | scalex | I red the getting started but I can't find what I want. |
| 23:23.59 | [TK]D-Fender | scalex: Ask better questions, get better answers |
| 23:24.01 | ipkaf | for the first time on my life i totally waist alll my time iher e |
| 23:24.03 | ipkaf | fygeuhijokpl^msq |
| 23:24.05 | scalex | it already installed and I test the voip and is good |
| 23:24.27 | [TK]D-Fender | ipkaf: No, you were wasting your own time and continue to do so by running BLIND |
| 23:24.31 | drmessano | Thats the fastest he's ever left IRC |
| 23:24.41 | drmessano | usually it takes hours |
| 23:25.17 | drmessano | He'll come back with a different nickname |
| 23:25.18 | scalex | anybody can give a link to read sip.conf and extension.conf |
| 23:25.21 | scalex | how to setup |
| 23:25.30 | [TK]D-Fender | ~book |
| 23:25.31 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:25.32 | [TK]D-Fender | ^^^^^^^^ |
| 23:25.34 | scalex | I need to understand very clearly |
| 23:25.39 | [TK]D-Fender | scalex: go READ |
| 23:25.59 | drmessano | IN SOVIET RUSSIA, BOOK READ YOU |
| 23:26.10 | scalex | I red it but I can't understand |
| 23:26.48 | [TK]D-Fender | scalex: What specifically don't you understand? |
| 23:27.10 | scalex | well, i need to know the section from extensions |
| 23:27.33 | scalex | i need to know how to setup my internal extension |
| 23:27.51 | [TK]D-Fender | scalex: Book shows you a sample soft-phone setup |
| 23:27.52 | scalex | i see another examples but I do not understand |
| 23:28.02 | scalex | i can register |
| 23:28.08 | [TK]D-Fender | scalex: Go read this for some inspiration : |
| 23:28.11 | scalex | but I can dial another softphone |
| 23:28.12 | [TK]D-Fender | ~jerjerguide |
| 23:28.12 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 23:28.57 | [TK]D-Fender | scalex>but I can dial another softphone <- this should not be a "but". It is also positive |
| 23:29.10 | [TK]D-Fender | scalex: So you need to be VERY clear what your PROBLEM is. |
| 23:29.31 | scalex | right |
| 23:29.48 | scalex | and I need to know all the command I can use |
| 23:29.55 | scalex | so I decide how to use it |
| 23:30.11 | [TK]D-Fender | scalex:"core show applications" |
| 23:30.24 | [TK]D-Fender | scalex:"core show application [specificapplication here]" |
| 23:30.55 | scalex | ok |
| 23:32.01 | scalex | thank you very much |
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