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| 00:19.40 | igcewieling | Micc: Are you SURE SIP ALG and SPI is disabled on your NAT router? |
| 00:20.26 | igcewieling | Also make sure to disable any NAT features on the ATA and set nat=yes on Asterisk and don't forward any ports on the NAT router. |
| 00:22.27 | igcewieling | your nat router should then fixup the port for you when the packet is translated out |
| 00:22.41 | igcewieling | (using standard nat rules) |
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| 01:09.46 | Rahail | ~pastebin |
| 01:09.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:10.40 | Rahail | HI can some one help me understand this debug log |
| 01:10.56 | Rahail | when point a try to register with poin b via iax2 on point a i see this however this dont regsiter |
| 01:10.58 | Rahail | http://pastebin.com/Tq7T89Z0 |
| 01:11.02 | Rahail | what am I missing |
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| 01:17.12 | Rahail | hello |
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| 01:21.56 | Rahail | how can i fix that i keep seeing it on iax2 debug however i am not able to make point a register with point b |
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| 01:48.24 | Rahail | any one is good at iax2 when cna guide to this error |
| 01:52.23 | Rahail | http://pastebin.com/cVrdFAa4 |
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| 02:12.34 | igcewieling | Rahail: so few people use IAX2 you might want to search the mailing list archives or check the Asterisk Book |
| 02:13.57 | Rahail | i did still trying to find |
| 02:17.53 | igcewieling | why not use SIP? |
| 02:18.06 | Rahail | sip is block |
| 02:18.18 | Rahail | plus with iax2 trunk we save bandwith |
| 02:30.12 | Rahail | on mailing every on recomendation i try no luck |
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| 05:47.22 | igcewieling2 | andrewyager is collaborating with that nutjob caterwaul |
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| 07:48.11 | atan | I am trying to track down the source of "ERROR[27752]: config_options.c:495 aco_process_config: Attempting to process uninitialized aco_info" but so far I've only found AOC, not ACO... anyone know where I should be looking? |
| 07:49.35 | atan | Looking in config_options.c I see if (!info->internal) { ast_log(LOG_ERROR, "Attempting to process uninitialized aco_info\n");, but I'm not sure what exactly it's looking for |
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| 08:46.33 | Tuju | any idea why my INVITE packets are sent but i get no response packets from remote proxy? |
| 08:46.50 | Tuju | could it be that some response header is wrong and those are sent to wrong direction? |
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| 08:50.13 | bruce__ | hey guys |
| 08:50.51 | kaldemar | Tuju: are you sending from behind a NAT? |
| 08:50.52 | bruce__ | I have a question... do you need to install Asterisk in order for two SIP phones to talk to each other or can a simple SIP server work? |
| 08:51.06 | Tuju | kaldemar: no, all are public addresses. |
| 08:51.12 | bruce__ | something like opensips |
| 08:51.32 | Tuju | bruce__: you can call directly between phones too. |
| 08:51.37 | Tuju | no proxy required. |
| 08:51.45 | bruce__ | umm... |
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| 08:52.09 | Tuju | you just need to use more accurate sip urls when placing the call. |
| 08:52.19 | bruce__ | Tuju, how will someone talk to my phone though? |
| 08:52.29 | Tuju | what? |
| 08:52.29 | bruce__ | there needs to be DNS SVR recs and stuff |
| 08:52.34 | bruce__ | surely |
| 08:52.45 | Tuju | you asked is it possible, i said yes. |
| 08:52.56 | bruce__ | huh? |
| 08:53.32 | Tuju | i've done it, but i assume that you're thinking some other setup and the answer is no, it doesn't work. |
| 08:53.40 | atan | kaldemar, you keen fellow, might you have any idea about my "aco_process_config: Attempting to process uninitialized aco_info" when reloading a semi-simple Asterisk 11.3 install? |
| 08:54.09 | bruce__ | so do you need Asterisk for SIP-to-SIP calls? |
| 08:54.15 | Tuju | bruce__: no. |
| 08:54.38 | bruce__ | whats the purpose of asterisk then? |
| 08:54.40 | Tuju | take a eth cross-over cable, hook it between phones and dial. other will ring. |
| 08:54.43 | kaldemar | atan: no. |
| 08:54.54 | atan | kaldemar, drat, thanks anyhow :-) |
| 08:55.20 | Tuju | bruce__: i think mostly the internet and all bubblecum solutions it has, like NAT, DHCP, firewalls etc. |
| 08:55.38 | *** join/#asterisk RadJackson (~RadJackso@cpe-et000092.cust.jaguar-network.net) |
| 08:56.11 | Tuju | bruce__: if you make a question "does it work" and it's related to 10 zillion transistors and few hundred network configs and boxes, it's quite hard to figure out what you're actually thinking. |
| 08:56.11 | kaldemar | bruce__: it provides features that phones don't have. |
| 08:56.12 | bruce__ | what is bubblecum? |
| 08:56.21 | RadJackson | Hello, i am placing a .call file into /outgoing/ folder in order to make an automatic dial, sometimes it works, sometimes it says "Call failed to go through, reason (8) Congestion (circuits busy)" , what does it mean? sorry i'm an asterisk beginner |
| 08:56.41 | Tuju | bruce__: SIP-protocol is not stopping you. most likely the environment where you're trying it is. |
| 08:56.41 | kaldemar | RadJackson: define "placing" |
| 08:57.28 | RadJackson | mv mycallfile /var/spool/asterisk/outgoing |
| 08:57.29 | bruce__ | Tuju, umm... I want a SIP identify to be an email address... something like getonsip.com but I don't want to configure Asterisk |
| 08:58.16 | kaldemar | RadJackson: congestion can happen for many reasons. you'd need to take a closer look at the interface you use for dialing. what is the channel? |
| 08:58.20 | Tuju | that's already more complex than 'call between two phones' |
| 08:58.26 | bruce__ | ya |
| 08:58.31 | bruce__ | I'm trying to figure this out |
| 08:58.44 | Tuju | why not asterisk, it's like apache httpd, simple once you get it. |
| 08:58.57 | bruce__ | so the sip phone will look up DNS srv records right? |
| 08:59.05 | Tuju | <PROTECTED> |
| 08:59.15 | Tuju | i think it might, but those are not required. |
| 08:59.21 | bruce__ | kak |
| 08:59.23 | Tuju | you can use also sip.example.com |
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| 08:59.33 | bruce__ | oh I see |
| 08:59.34 | Tuju | it reads in sip rfc |
| 08:59.55 | bruce__ | so it automatically justs connects to sip.blah blah |
| 09:00.22 | Tuju | it's bit more hairy than that. |
| 09:00.29 | Tuju | install centos |
| 09:00.35 | Tuju | yum install asterisk\* |
| 09:00.40 | Tuju | edit sip.conf and extensions.conf |
| 09:00.42 | Tuju | you're done. |
| 09:00.54 | RadJackson | kaldemar , the channel is : Channel: SIP/TRUNK/NUMBERTODIAL |
| 09:00.55 | bruce__ | but do you need asterisk for that to work? |
| 09:01.02 | bruce__ | will something like opensips work? |
| 09:01.09 | Tuju | mostly because firewalls and NAT. |
| 09:01.27 | Tuju | it keeps record where the damn phone is that is supposed to ring when you call. |
| 09:01.28 | bruce__ | huh |
| 09:02.11 | kaldemar | RadJackson: you'd need to see sip debug of a failing call to know what's going on. then ask the other end why they fail your call. |
| 09:02.51 | Tuju | you may thank the asswipes who kept extending internet with ipv4 instead moving to v6 that is going to happen sooner or later anyway. |
| 09:04.06 | RadJackson | ok thank you kaldemar |
| 09:04.15 | kaldemar | bruce__: asterisk is not a requirement for SIP in any way. you may choose what ever server solution you wish. |
| 09:05.04 | bruce__ | kaldemar, so if I install opensips and connect to it with a sip phone and try and call other sip identity will it work? |
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| 09:07.19 | Tuju | bruce__: you're asking too generic questions. |
| 09:07.27 | bruce__ | ya I know |
| 09:07.38 | bruce__ | it's because I'm still learning about VOIP |
| 09:07.40 | Tuju | "if i join to amnesty, will it give world peace?" - well, it depends. |
| 09:07.41 | bruce__ | just messing around |
| 09:07.53 | bruce__ | ok what does this depend on? |
| 09:08.06 | Tuju | well that's nice. we're just trying to help you, that is, wasting our time. |
| 09:09.12 | kaldemar | bruce__: opensips is known to work. |
| 09:09.23 | bruce__ | kaldemar, ok sweet |
| 09:09.57 | bruce__ | kaldemar, you see I want something smaller than asterisk so I don't get pw3d |
| 09:10.12 | bruce__ | because I don't completely understand asterisk yet |
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| 09:13.09 | kaldemar | bruce__: configuring opensips is not necessarily any easier. |
| 09:13.23 | bruce__ | haha really? |
| 09:13.35 | bruce__ | does it have all that asteriskie stuff too? |
| 09:13.48 | kaldemar | your question makes no sense. |
| 09:14.02 | bruce__ | lolz |
| 09:14.22 | bruce__ | asterisk seems to be a shit ton of stuff right? |
| 09:14.33 | bruce__ | can opensips do everything that asterisk can? |
| 09:19.16 | kaldemar | bruce__: no. |
| 09:19.44 | bruce__ | then it does not have all the assteriskie stuff then |
| 09:29.27 | bruce__ | bye |
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| 11:04.19 | rampage21 | heya peeps |
| 11:04.45 | rampage21 | have any experience with e164 number conversion? |
| 11:04.50 | rampage21 | and snom ip phones? |
| 11:05.54 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 11:06.44 | WIMPy | ~ask |
| 11:06.45 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:07.52 | bulkorok | hi |
| 11:08.03 | din3sh | hey all |
| 11:08.06 | din3sh | what would this mean? |
| 11:08.08 | din3sh | DEBUG[24991] channel.c: Didn't receive a media frame from DAHDI/i1/271154676-918a within 500 ms of answering. Continuing anyway |
| 11:11.18 | rampage21 | I'm trying to have numbers with the country code in front (+) converted to 00 |
| 11:11.35 | rampage21 | I've been doing stuff with active directory and ldap |
| 11:11.48 | rampage21 | have an ou with gazillions of contacts |
| 11:12.07 | rampage21 | just the phone seems to have an issue with dialing the numbers because of the + in front |
| 11:12.40 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:13.25 | WIMPy | I'm very sure the phone won't care. |
| 11:13.40 | rampage21 | they do |
| 11:14.02 | rampage21 | someone here had the same issue and had to create a dial plan that did that |
| 11:14.09 | ectospasm | If you don't have a + in your dialplan patterns, it won't work |
| 11:14.16 | rampage21 | ya |
| 11:14.21 | rampage21 | how do I create such a dial plan? |
| 11:14.40 | WIMPy | Why do you set a dialplan at all? |
| 11:14.42 | ectospasm | exten => +NXXXNX.,... |
| 11:15.00 | ectospasm | Oh, you meant digit map. |
| 11:15.01 | rampage21 | ectospasm, is that done on the ip phone or asterisk server? |
| 11:15.06 | ectospasm | it doesn't need to go on that. |
| 11:15.11 | rampage21 | on what? |
| 11:15.14 | ectospasm | rampage21: that goes in extensions.conf |
| 11:15.17 | ectospasm | ...on Asterisk |
| 11:15.20 | rampage21 | oh |
| 11:15.25 | WIMPy | Both |
| 11:16.04 | WIMPy | On the phone I'd just delete it. Seems to be the only safe choice. |
| 11:16.24 | Onyx47 | hello, does AEL support return values in macros? I did find an example where you can use c-style "pointers" by defining a macro as macro myMacro(*return) and it works fine, but it's a bit of a pain if I only have one return value... |
| 11:17.56 | rampage21 | ectospasm, I'm lost |
| 11:18.04 | rampage21 | ectospasm, someone told me that they created it on the snom |
| 11:18.38 | ectospasm | I don't know much about Snom phones |
| 11:19.01 | rampage21 | grr |
| 11:19.05 | rampage21 | thanks though |
| 11:20.38 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 11:20.50 | WIMPy | What exactely is your issue? |
| 11:22.50 | din3sh | create what on snom phones? |
| 11:22.55 | din3sh | i have a couple of snoms |
| 11:23.29 | din3sh | hey WIMPy |
| 11:23.37 | WIMPy | Hi din3sh |
| 11:23.54 | din3sh | hwz it going? |
| 11:24.35 | WIMPy | Spending too much time trying to find out what to do first :-( |
| 11:25.58 | ectospasm | prioritize (-; |
| 11:26.16 | rampage21 | din3sh, awesome |
| 11:26.33 | rampage21 | they complain when they try to dial nums with +27 |
| 11:26.36 | rampage21 | in front |
| 11:27.01 | bulkorok | + is not a number... |
| 11:27.07 | rampage21 | yip |
| 11:27.14 | rampage21 | so it needs to be converted |
| 11:27.16 | rampage21 | right? |
| 11:27.18 | rampage21 | to 00 |
| 11:27.24 | bulkorok | so... tell them to dial numbers... |
| 11:27.32 | rampage21 | how? |
| 11:27.37 | WIMPy | Why do you want to convert them? |
| 11:27.55 | rampage21 | it won't dial |
| 11:27.58 | din3sh | WIMPy: debugging going on? |
| 11:27.59 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 11:28.04 | WIMPy | Just add a pattern that fits to your dialplan as ectospasm suggested. |
| 11:28.17 | bulkorok | well... open the door... shout out loud "don't use the + in the numbers you want to dial!" shut the door, and take a coffee :-) |
| 11:28.30 | rampage21 | WIMPy, how do I do that on Elastix? |
| 11:28.36 | din3sh | +?? |
| 11:28.48 | din3sh | why would you want to dial + from a snom phone? |
| 11:29.11 | WIMPy | rampage21: The same way as you add other patterns (routes? whatever). |
| 11:29.24 | rampage21 | http://wiki.snom.com/Features/Mass_Deployment/Setting_Files/XML/Dial_Plan/Syntax |
| 11:29.29 | rampage21 | like this kak |
| 11:29.38 | bulkorok | this is my opinion... no one would use a plus to dial to the "real" pstn wotrld and gets angry if it won't work |
| 11:29.44 | rampage21 | din3sh, cause it gets that from ad |
| 11:30.09 | din3sh | which ad? |
| 11:30.19 | rampage21 | active directory |
| 11:30.26 | rampage21 | via ldap |
| 11:30.37 | WIMPy | bulkorok: Why not? I only use numbers in that format as it's the only format that's not ambigous. |
| 11:30.57 | din3sh | oh ok |
| 11:30.57 | din3sh | lol |
| 11:31.11 | bulkorok | well not every carrier uses e.164 :-( |
| 11:31.21 | din3sh | + is not a number |
| 11:31.25 | din3sh | format your AD |
| 11:31.28 | din3sh | :p |
| 11:31.33 | WIMPy | din3sh: How does that matter? |
| 11:31.34 | bulkorok | :-) |
| 11:31.36 | rampage21 | lol |
| 11:31.52 | WIMPy | Noone says that phone numbers have to be numbers, do they? |
| 11:31.52 | rampage21 | I just spent ages adding a plus in front of the numbers because "it has to be like that" |
| 11:32.00 | din3sh | ok I tried to have a click to dial stuff on Lotus notes |
| 11:32.18 | din3sh | to detect numbers stating with + |
| 11:32.27 | din3sh | you have to have a regex |
| 11:32.27 | bulkorok | so... two options... make an extensions that turns + to 00 if that is what your carrier wants... or change your AD .-) |
| 11:32.37 | WIMPy | Not even the PSTN has the restriction to digits. Even though your provider most probably filters everythign else. |
| 11:32.53 | rampage21 | bulkorok, how do I do the converting to 00 |
| 11:33.12 | din3sh | what happens when you try to call? |
| 11:33.25 | din3sh | extension not found in context blahblah? |
| 11:33.44 | bulkorok | exten => +.,1,Dial/yourprovider/00{EXTEN:1} |
| 11:33.48 | rampage21 | all circuits are busy now |
| 11:34.10 | din3sh | strip off the + as bulkorok saying |
| 11:34.29 | din3sh | your asterisk box is dialing the +? |
| 11:34.32 | WIMPy | It even looks like the Snom can do a replace. I never tried that. I prefer to configure things on the server, not on the phone. |
| 11:34.55 | bulkorok | WIMPy: agree |
| 11:34.56 | rampage21 | WIMPy, how can I get the phone to conver it? |
| 11:35.05 | rampage21 | that's what I'm needing to do |
| 11:35.07 | rampage21 | on the phone |
| 11:35.33 | WIMPy | Read that wiki page you have been given and try it out. |
| 11:35.35 | din3sh | would simpler on asterisk though |
| 11:35.42 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:35.49 | WIMPy | But I'd recommend doing it on the server. |
| 11:36.01 | din3sh | what number is asterisk dialing? |
| 11:36.07 | rampage21 | that stuff on the wiki is not working |
| 11:36.07 | rampage21 | haha |
| 11:36.12 | bulkorok | ~elastix |
| 11:36.12 | infobot | [elastix] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 11:36.33 | bulkorok | if no body even mentioned that! |
| 11:36.54 | WIMPy | Or maybe better upgrade to real Asterisk. |
| 11:37.01 | bulkorok | mmh... interesting space between no and body... |
| 11:37.37 | din3sh | elastix aint that bad |
| 11:37.39 | din3sh | :P |
| 11:38.30 | WIMPy | hasn't heard anythign good about any GUI. |
| 11:38.47 | WIMPy | I'd like to know how usable the Switchvox thing is. |
| 11:39.02 | din3sh | but the scripts in elastix tend to complicate things |
| 11:39.16 | rampage21 | ag fuck |
| 11:39.30 | rampage21 | I'll give you $1 via paypal if you help |
| 11:39.41 | bulkorok | WIMPy: just because switchvox is from digium it isn't guarenteed that it works better than other guis I suppose |
| 11:40.13 | din3sh | I had a look at switchivox's gui |
| 11:40.40 | WIMPy | bulkorok: Sure. But I'd hope they made use of some of the features that have been added at lower level. |
| 11:41.00 | bulkorok | :-) |
| 11:41.06 | WIMPy | is more interested in the functionality than the configuration. |
| 11:41.14 | bulkorok | I see |
| 11:42.02 | din3sh | i have installed an elastix system |
| 11:42.18 | din3sh | but ended up writing my own php/mysql based gui |
| 11:42.19 | din3sh | lol |
| 11:42.49 | WIMPy | One of the many things I started but never finished :-( |
| 11:43.23 | din3sh | if u used realtime configs, its pretty straight forward |
| 11:43.40 | rampage21 | slaps din3sh with a large fish |
| 11:43.41 | din3sh | much much simpler than freepbx and elastix |
| 11:43.47 | WIMPy | Well, I do use genereted configs, but I wantd to make it multi-tennant with self-service. |
| 11:43.54 | rampage21 | slaps bulkorok with a large fish |
| 11:44.13 | din3sh | rampage21, up ur bid, $1 aint working |
| 11:44.28 | rampage21 | yaya |
| 11:44.33 | rampage21 | I'm too poor |
| 11:44.44 | din3sh | you're complicating things for urself |
| 11:44.45 | din3sh | :p |
| 11:44.50 | rampage21 | nsh |
| 11:44.56 | rampage21 | this is all the options I have |
| 11:45.08 | din3sh | am pretty sure stripping the + in your dialplan is simply |
| 11:45.30 | rampage21 | I can't figure it out |
| 11:45.47 | rampage21 | so din3sh can you give me the exact copy-n-paste |
| 11:46.20 | din3sh | mind you my configs are full of bugs |
| 11:46.23 | din3sh | lol |
| 11:46.37 | rampage21 | I need one that works |
| 11:46.53 | rampage21 | I've already tried rm -fr / but that did not work |
| 11:46.59 | WIMPy | does it with loopback switches. |
| 11:53.02 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:57.06 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 11:57.48 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 11:59.21 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 12:01.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.58) |
| 12:04.13 | *** join/#asterisk ctaloi (~ctaloi@50.56.202.179) |
| 12:04.27 | *** join/#asterisk evilman_home (kvirc@78-106-161-222.broadband.corbina.ru) |
| 12:05.21 | *** join/#asterisk James87 (~wiv@unaffiliated/james87) |
| 12:05.39 | James87 | does anyone know if it possible to 'capture' FSK signals that are send to my Asterisk (freepbx) ? |
| 12:06.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:14de:f2f8:77d5:8fce) |
| 12:06.30 | WIMPy | What kind/in what situation? |
| 12:06.44 | *** join/#asterisk vlad_sta_ (~vlad_star@109.95.84.114) |
| 12:07.09 | WIMPy | Mind you that we can only answer if Asterisk can do it. To find out if FreePBX can be configured to do so, you have to ask in #freepbx. |
| 12:07.20 | *** join/#asterisk janelleb (~jamii@ec2-184-72-157-132.compute-1.amazonaws.com) |
| 12:08.27 | James87 | customer has personal alamr boxes, they send some kind of FSK out |
| 12:08.56 | James87 | they use it for signaling |
| 12:09.17 | James87 | when i let asterisk answer the call i only hear beeps but i don't know what to do with it |
| 12:10.45 | WIMPy | Could be anything. If you're really lucky AlarmReceiver could fit. |
| 12:11.29 | James87 | not familiar with that, is it a software package? |
| 12:11.49 | WIMPy | It's an Asterisk dialplan Application. |
| 12:12.39 | James87 | ok, i'll google it, thnx! |
| 12:13.49 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:15.01 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 12:15.03 | _abc_ | hello |
| 12:15.29 | _abc_ | can someone point out how one deals with cisco tftp phone configs of the signed kind? SEPxxx.cnf.xml.sgn? |
| 12:15.34 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:15.37 | _abc_ | I can find nothing online about this. |
| 12:15.43 | _abc_ | And yes this is for asterisk use. |
| 12:19.58 | janelleb | Hi all, I've installed * 1.8 and chan_dongle successfuly (* detects the device), but I don't know how to initiate a call (i.e. make a call outside). How do I make a call? sorry I'm an Aserisk newbie. |
| 12:25.08 | itgrl | ~book |
| 12:25.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:25.11 | WIMPy | janelleb: You have to create some extension and call it. Did you read the |
| 12:26.46 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:27.32 | janelleb | If you mean the "Asterisk the Definitive Guide, 3rd edition" yep I'm working through the book now. I've created an extenion, but I don't know what command to do at CLI> to make a call. |
| 12:28.09 | WIMPy | Usually you use a phone to make a call. |
| 12:28.21 | WIMPy | From the CLI you can use 'channel originate'. |
| 12:29.14 | *** join/#asterisk [Outcast] (~anonymous@pool-96-237-60-251.bstnma.fios.verizon.net) |
| 12:29.19 | janelleb | WIMPy: is asterisk not suitable for a completely autonomous app? i.e. the only calls made will be scheduled and will only play a recording. |
| 12:29.23 | WIMPy | (channel originate doesn;t even need a dialplan) |
| 12:30.14 | WIMPy | See above. Other options are call files or AMI originate. |
| 12:30.49 | janelleb | WIMPy: ok thanks I had read about CLI> channel originate, I'm not sure what the other two are. I'll be back to IRC real soon. |
| 12:31.40 | WIMPy | The book should at least cover call files. |
| 12:31.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.114) |
| 12:43.08 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 12:43.15 | jeffspeff | good mornin asteriskers |
| 12:44.25 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 12:46.29 | *** join/#asterisk ipconfeng (~ipconfeng@pool-71-185-231-211.phlapa.fios.verizon.net) |
| 12:52.31 | janelleb | WIMPy: Does the following CLI output mean that I messed up in configuring channels? i.e. are there no channels on my asterisk right now? http://pastebin.com/pN1EUpP1 |
| 12:53.21 | WIMPy | No active channels, i.e. non carrying a call. |
| 12:54.21 | WIMPy | +e |
| 12:55.09 | *** join/#asterisk tamiel (~tamiel@ip-38.net-81-220-93.rev.numericable.fr) |
| 12:55.40 | *** join/#asterisk serafie (~erin@nat/digium/x-subuzpxnbjrmdiwe) |
| 13:01.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.114) |
| 13:08.06 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:10.37 | pabelanger | <caterwaul> ALREADY HAPPENING with SHITTY ops like Qwell banning and censoring people. FUCK YOU. He's an ENEMY. http://www.youtube.com/watch?v=GYYEzQDiDpc&feature=player_detailpage#t=2752s |
| 13:10.42 | pabelanger | look what you have done |
| 13:11.36 | tzanger | blinks |
| 13:11.43 | *** join/#asterisk afournier (~admin@46.255.181.29) |
| 13:11.43 | tzanger | looks at the channel name again |
| 13:20.29 | [TK]D-Fender | pabelanger, Yeah, I got much of the same yesterday |
| 13:20.39 | [TK]D-Fender | pabelanger, As did several others |
| 13:20.54 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 13:24.13 | leifmadsen | pabelanger: hawt |
| 13:25.55 | *** join/#asterisk aruntomar (~Thunderbi@49.248.153.139) |
| 13:30.10 | *** join/#asterisk garymc (~chatzilla@host81-139-176-231.in-addr.btopenworld.com) |
| 13:31.21 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mlfghuhyezpepeyc) |
| 13:31.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:33.03 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 13:33.49 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
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| 13:50.04 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:51.38 | igcewieling | andrewyager is collaborating with that nutjob caterwaul |
| 13:52.06 | [TK]D-Fender | who? |
| 13:52.12 | [TK]D-Fender | (1st guy) |
| 13:52.44 | [TK]D-Fender | There he is... |
| 13:52.46 | [TK]D-Fender | 00:10.18*** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 13:53.12 | [TK]D-Fender | 23:21.16*** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 13:53.50 | [TK]D-Fender | *** join/#asterisk andrewyager (~andrewyag@2-104-141-114.static-dsl.realworld.net.au) |
| 13:54.14 | [TK]D-Fender | *** join/#asterisk andrewyager (~andrewyag@1.144.72.51) |
| 13:54.30 | eirirs | same shit, different hosts, I guess? |
| 13:54.47 | [TK]D-Fender | as tm1000 informed me caterwaul had spun up a shitload of linodes worth of proxies and is abn-evading |
| 13:54.56 | eirirs | ah |
| 13:55.20 | [TK]D-Fender | Very probably the same guy |
| 13:55.42 | igcewieling | I changed my nick last night to try to avoid caterwaul and he /msg'd me a paste of andrewyager telling him I changed my nick |
| 13:55.55 | igcewieling | It takes a lot of energy to be that angry, eventually he'll run out. |
| 13:55.56 | [TK]D-Fender | 23:41.49*** join/#asterisk andrewyager (~andrewyag@101.171.130.32) |
| 13:56.21 | eirirs | maybe he just needs professional help |
| 13:56.24 | igcewieling | has anyone complained to Ircops? |
| 13:56.31 | Greenlight | What was the guy so angry about? |
| 13:56.51 | igcewieling | Greenlight: he got banned because he was asking about the pirate g729 codec |
| 13:57.01 | Greenlight | Oh same old then |
| 13:57.09 | Greenlight | Seen that argument so many bloody times in here |
| 13:57.26 | igcewieling | Qwell could have been a tad nicer about it, but I fully support his actions and Digium's policy. |
| 13:58.06 | Greenlight | And then the guy just flipped by the sounds of it? |
| 13:58.07 | [TK]D-Fender | He was rather quick on cutting him out... |
| 13:58.19 | WIMPy | Now we know how such things start. Sometimes they end by someone flying an airplane in to an office building. |
| 13:58.26 | [TK]D-Fender | but then the reaction he made created all the justification one could require |
| 13:58.39 | [TK]D-Fender | Greenlight, yes, completely flipped his shit |
| 13:58.56 | Greenlight | Sounds like I missed all the fun ^^ |
| 13:59.18 | igcewieling | I am against ANYTHING which might in any way cause issues with Digium's ability to sell g729 codecs. |
| 13:59.52 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:00.11 | Greenlight | You're against fatter internet pipes? :) |
| 14:00.40 | igcewieling | Greenlight: even if nobody wants to buy them I want Digium to be able to sell them. |
| 14:00.41 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:01.39 | igcewieling | Digium paid a massive amount of money for the right to sell g729 codec licenses (several years ago I heard from someone involved they expected it to take 10 years to recoup their initial license costs) |
| 14:02.08 | [TK]D-Fender | I am against idiots who upon being shut down start acting like an entitled whiny bitch swearing revenge against everything * related saying to thank Qwell for it.... |
| 14:02.22 | igcewieling | I have no problem criticizing Digium on a wide variety of topics, but g729 licenses is not one of those topics. |
| 14:02.46 | [TK]D-Fender | As I said he was out looking for * stuff to pirate and asked me for targets. The one he showed me first .... for a FREEPBX PAID MODULE. |
| 14:06.11 | igcewieling | That is like asking a cop where to buy weed. |
| 14:07.24 | WIMPy | I guess they only know sources for alcohol or cocaine. |
| 14:08.08 | igcewieling | WIMPy: [tk] is heavily involved in FreePBX |
| 14:09.35 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 14:10.00 | WIMPy | didn't even know they had paid options. |
| 14:10.19 | atan | When I reload Asterisk I see " ERROR[28088]: config_options.c:495 aco_process_config: Attempting to process uninitialized aco_info ", anyone know what that might be from or how I might resolve it? |
| 14:10.22 | igcewieling | there are commercial modules for some stuff |
| 14:10.36 | igcewieling | usually advanced features |
| 14:14.30 | Katty | morning |
| 14:14.49 | itgrl | morning |
| 14:15.03 | WIMPy | Good afternoon. |
| 14:15.14 | eirirs | evening! |
| 14:15.15 | eirirs | lol |
| 14:21.28 | chuckf | nighty night |
| 14:21.51 | WIMPy | Damn. That was a short day. |
| 14:22.05 | chuckf | but productive |
| 14:24.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.141) |
| 14:26.27 | atan | Can one use chan_skinny over a WAN similar to how one uses SIP, without any trouble? |
| 14:27.51 | acidfu | it's a tcp/ip connection |
| 14:28.23 | atan | I've _never_ used skinny before. I know nothing about it. Does the phone identify with a login, or does it look it up by MAC...? |
| 14:29.32 | acidfu | it doesn't use a login/password |
| 14:29.43 | acidfu | and it doesn't use the MAC adress from the ethernet frame |
| 14:30.05 | acidfu | the skinny protocol is sending a SEP-MAC-ADDRESS inside the tcp/ip connection |
| 14:30.35 | igcewieling | atan: I believe Asterisk's SCCP/Skinny does not support NAT, but you should confirm that. |
| 14:30.47 | atan | So it's all based on the mac of the device then, right? So on a public WAN, if Asterisk was not filtering connections... anyone who knows the MAC of the phone could spoof the connection if they wanted? |
| 14:30.48 | acidfu | I don't think either |
| 14:30.49 | igcewieling | It may have changed at some point to support NAT. |
| 14:30.59 | atan | igcewieling, thanks for the tip on that. VERY good to know :-) |
| 14:31.02 | acidfu | atan, yes |
| 14:31.03 | igcewieling | atan: Correct! |
| 14:32.55 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 14:33.30 | acidfu | atan, and if you want to try an alternative implementation you can try this one: http://gitorious.org/xivo/xivo-libsccp |
| 14:34.16 | atan | ty ty ty |
| 14:34.34 | igcewieling | you could also use phones from a company which actually wants your business. |
| 14:35.11 | acidfu | which means ? |
| 14:35.15 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 14:35.53 | Qwell | leifmadsen: caterwaul released a copy of your book, for free on the Internet. |
| 14:36.01 | Qwell | oh noes! |
| 14:36.10 | atan | igcewieling, was that a stab at Cisco? hah, I feel ya |
| 14:36.56 | atan | Qwell, you know much about aco_process_config? |
| 14:37.10 | Qwell | atan: no |
| 14:37.14 | atan | Crud :D |
| 14:37.22 | atan | Know where I could look? |
| 14:37.23 | igcewieling | acidfu: Cisco wants you to use CCM with their Cisco phones. They leave features out of their SIP firmware, they want to charge you for firmware upgrades, they do not document the SCCP/Skinny protocol |
| 14:37.40 | acidfu | ah, true |
| 14:38.01 | atan | ^ my biggest tickoff with Cisco is not allowing a sidecar on the SIP firmware, GAH! |
| 14:38.04 | igcewieling | atan: I like Cisco, but for PHONES they are user hostile if you are not paying for their expesive call manager platform. |
| 14:38.08 | acidfu | but it's fun to remove a CCM and use ASterisk instead when you already have a bunch of phones ;) |
| 14:38.38 | atan | What other nifty phones, perhaps older, are similar to the IP 79xx series from Cisco? |
| 14:38.43 | igcewieling | I have no interest in using products from a company which actively discourages it. |
| 14:39.13 | atan | I would have to guess Polycom has something, I just haven't had anything beefy from them I like yet :( |
| 14:39.19 | leifmadsen | Qwell: oh no! not for free?! |
| 14:39.20 | igcewieling | atan: Polycom, Cisco SPA series, Digium also has phones but they have not been out for all that long so community support is lacking. Digium does support their phones however. |
| 14:40.00 | atan | Who makes the Digium phones? Poly? |
| 14:40.22 | Qwell | atan: Digium. |
| 14:40.38 | atan | Oh really? It's not like Cisco conference phones from Polycom? |
| 14:41.09 | atan | Sick. I can get behind that! On eBay looking now (or is there a better source?) |
| 14:41.12 | atan | And do they use SIP? |
| 14:41.29 | Qwell | You could buy them from pretty much any Digium reseller. |
| 14:41.42 | Qwell | or Digium directly, for that matter |
| 14:41.47 | igcewieling | Digium seems to be highly motivated to get people to use their phones, so I imagine they have decent support for them. |
| 14:41.49 | atan | Oh this D70 looks awesome so far :D |
| 14:42.03 | atan | Well Digium hasn't done me any wrong so far, what's to lose!? |
| 14:42.21 | atan | Is it just the D40, D50, and D70 phones right now? |
| 14:42.24 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:42.26 | igcewieling | atan: you must have never used their older PSTN cards 8-| |
| 14:42.34 | Qwell | atan: I'll even give you the Qwell guarantee. If the phones eat any small children, I will personally refund your money. |
| 14:42.35 | atan | ...nope, all ITSP here! :-) |
| 14:43.54 | mmlj4 | heh |
| 14:44.05 | igcewieling | atan: if you are trying to decide which phone to standardize on, get a Polycom, and SPA, and a Digium and see which one you like. |
| 14:44.29 | atan | I would always assume the Digium phones will work with Asterisk without any issue :D hahaha |
| 14:44.57 | igcewieling | atan: A 10 year old Grandstream phone will work with Asterisk, that doesn't mean you want to use them. |
| 14:45.11 | mmlj4 | I hear little about aastra... are they junk? |
| 14:45.13 | igcewieling | It all depends on your requirements and expectations |
| 14:45.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:45.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:45.46 | atan | Do the D phones have a web interface for management? Do they pull configs from TFTP? |
| 14:46.11 | Qwell | atan: With DPMA (an Asterisk module), it can be configured right from Asterisk. |
| 14:46.36 | Katty | QWELL |
| 14:49.05 | [TK]D-Fender | atan, they support flat files, DPMA, and web config |
| 14:49.15 | Katty | fender bender. |
| 14:49.25 | [TK]D-Fender | Katty, Mew. |
| 14:50.57 | atan | Okay let the fun begin... where is the cheapest place I'll find a D70 to play with? |
| 14:51.34 | atan | And yes feel free to solicit me for money |
| 14:51.43 | Katty | d20 is as big a roll as you can get! |
| 14:51.53 | WIMPy | Digiums online store? |
| 14:52.03 | atan | They ship to Canada? |
| 14:52.10 | WIMPy | And in addition to the mentioned list, I'd also get a Snom. |
| 14:52.16 | Katty | rolls initiative |
| 14:53.37 | atan | Do other phones support visual voicemail? |
| 14:54.15 | WIMPy | the real question is how well Asterisk is documented in that area. |
| 14:54.25 | Katty | wait, asterisk is documented?! |
| 14:54.29 | Katty | leifmadsen must be doing his job. |
| 14:54.37 | Katty | puts book on head, attempts osmosis. |
| 14:54.38 | leifmadsen | pfft |
| 14:54.41 | leifmadsen | it's all a sham |
| 14:54.53 | Katty | a Sham WOW. |
| 14:54.53 | WIMPy | I tried to do it on the Snom, but I don;t know if there's a safe way to delete messages from other applications. |
| 14:55.00 | Katty | shamwows leifmadsen |
| 14:55.04 | *** join/#asterisk butthurtface (~Butthurtf@38.122.108.2) |
| 14:55.06 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:55.11 | leifmadsen | WOW! |
| 14:55.17 | Katty | leifmadsen: also, i've been drinking more water. |
| 14:55.22 | leifmadsen | Katty: well done |
| 14:55.26 | Katty | leifmadsen: and i pulled off my 108 squats last night without getting thirsty. |
| 14:55.27 | SuperNull | hey guys. |
| 14:55.27 | leifmadsen | I've been drinking water, and being sick |
| 14:55.31 | Katty | leifmadsen: ya'll might be onto something |
| 14:55.35 | leifmadsen | Katty: nice! I suck at squats |
| 14:55.43 | leifmadsen | hydrate constantly! :) |
| 14:55.47 | Katty | leifmadsen: i did too, until i realized what it was doing to my bum. |
| 14:55.56 | leifmadsen | oic |
| 14:56.02 | leifmadsen | approves |
| 14:56.07 | Katty | leifmadsen: what sort of plague did you aquire? |
| 14:56.15 | leifmadsen | Katty: not sure... some sort of child based headcold |
| 14:56.29 | Katty | leifmadsen: ah right :< chicken broth. |
| 14:56.36 | leifmadsen | water and coffee |
| 14:56.39 | Katty | leifmadsen: studies show it reduces the gunk your nose produces. |
| 14:56.46 | leifmadsen | oic |
| 14:56.50 | Katty | leifmadsen: due to immune system over re-acting |
| 14:56.55 | leifmadsen | aye |
| 14:57.09 | Katty | leifmadsen: feel better. |
| 14:57.18 | leifmadsen | do not demand of me! |
| 14:57.26 | Katty | YOU WILL FEEL BETTER OR ELSE |
| 14:58.34 | leifmadsen | or else I won't? |
| 14:58.42 | leifmadsen | I need to go to the driving range tonight... |
| 14:58.58 | Katty | burn some rubber? |
| 14:59.08 | leifmadsen | want to get some practice in before my first golf game of the season |
| 14:59.14 | leifmadsen | almost :) |
| 15:00.52 | mmlj4 | golf? bah |
| 15:01.18 | Katty | hey now, golf is hard. |
| 15:01.40 | Katty | it's hard to make it go somewhere, much less in the general direction you're aiming for. |
| 15:01.48 | mmlj4 | I like real sports, like curling |
| 15:02.22 | mmlj4 | frozen shuffleboard with really heavy pucks |
| 15:02.27 | Katty | mmlj4: i'll curl you in a minute. |
| 15:02.32 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:10.58 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-30.usfamily.net) |
| 15:11.57 | Maliuta | Katty: when did you move to Canadia? |
| 15:12.24 | Maliuta | Katty: didn't think yanks liked curing all that much :) |
| 15:12.42 | *** join/#asterisk carrar (~tim@osburn.com) |
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| 15:20.22 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:20.48 | igcewieling | Maliuta: Katty is....odd. |
| 15:21.02 | igcewieling | so is mmlj4 apparently 8-| |
| 15:24.21 | Katty | yes, yes she is. but life is just more fun that way! |
| 15:28.47 | *** join/#asterisk carrar (~tim@osburn.com) |
| 15:31.27 | Katty | hi carrar |
| 15:31.33 | carrar | hi hi Katty!! |
| 15:31.38 | carrar | How ares you |
| 15:31.46 | atan | Where on earth is Canadia? |
| 15:31.54 | carrar | it's north of canada |
| 15:32.09 | Katty | i hear it's canada's hat. |
| 15:32.12 | Katty | carrar: i am good :> |
| 15:32.16 | atan | Ahhh Alaska <3 |
| 15:32.19 | Katty | carrar: did you get the hello kitty car? |
| 15:32.27 | carrar | hahah not yet |
| 15:32.46 | carrar | I would need the HELLO KITTY MAN TRUCK |
| 15:33.08 | carrar | Special Ford F450 |
| 15:33.18 | carrar | or even a F250 |
| 15:33.33 | eirirs | hummer |
| 15:33.49 | Katty | carrar: no love for the tacoma? |
| 15:34.00 | Katty | carrar: or a sierra |
| 15:34.11 | carrar | no |
| 15:34.23 | carrar | well here Tacoma is city, yucky place |
| 15:34.33 | carrar | so I think of that whenI think of a car named that |
| 15:35.02 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.236.54) |
| 15:36.10 | Katty | ohisee. |
| 15:43.58 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 15:48.46 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 15:51.49 | rgsteele | So, does DUNDi just address routing SPOF's, or are there some provisions for the extensions provided by the DUNDi peer as well? |
| 15:52.10 | rgsteele | E.g., if a DUNDi peer who is authoritative for a set of extensions falls off the map, will one of the other peers become authoritative for those extensions, or will they all just update their caches and say "sorry, none of us can route to that peer any more" |
| 15:52.45 | rgsteele | I mean, it seems strictly routing related, but I want to make sure I'm not being dense. |
| 15:53.03 | *** join/#asterisk teff (~teff@host217-39-153-134.in-addr.btopenworld.com) |
| 15:53.22 | *** join/#asterisk janelleb (~jamii@ec2-184-72-157-132.compute-1.amazonaws.com) |
| 15:54.10 | janelleb | Hi all. Anyone with a link to a tutorial on SMS processing in Asterisk? i.e. execute this_code when a message is received containing this_text?? |
| 15:54.30 | *** mode/#asterisk [-bb caterwaul!*@* *!*nirv@*2001 19f0:1619:c9::c805:4072] by Qwell |
| 15:56.17 | igcewieling | janelleb: what country are you in? |
| 15:59.32 | janelleb | igcewieling: I'm not using an SMSC, actually using sms mobile |
| 16:00.29 | igcewieling | janelleb: Asterisk's SMS support only works with SMSC. Anything else you want to do with SMS you'll have to do yourself with dialplan apps (like curl), email, AGIs, etc. |
| 16:00.57 | igcewieling | mabe chan_dongle supports SMS? |
| 16:01.08 | *** join/#asterisk navaismo (~navaismo@189.241.51.199) |
| 16:01.21 | Qwell | chan_mobile does |
| 16:01.32 | janelleb | igcewieling: About "doing it myself" yep I know. That is why I'm asking about a tutorial link. |
| 16:01.58 | janelleb | igcewieling: chan_dongle does indeed Support SMS. |
| 16:02.52 | janelleb | Anyway... anyone here with a link to an article or tutorial on SMS processing in Asterisk? |
| 16:03.15 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
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| 16:30.21 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::3) |
| 16:30.25 | cusco | hi folks |
| 16:30.50 | cusco | sometimes we're having high load average, but low cpu on top/htop |
| 16:31.35 | Qwell | Lots of disk access? Recordings, maybe? |
| 16:31.48 | cusco | recordings are on the gateway |
| 16:32.46 | cusco | but full log is enabled with verbose 5 |
| 16:32.49 | cusco | and other stuff |
| 16:32.53 | cusco | so I will get a faster disk |
| 16:33.12 | Qwell | There are lots of reasons. I didn't say it was definitely the disk I/O. |
| 16:33.29 | cusco | but I believe it might be... |
| 16:33.54 | Greenlight | If you disable the items which you think are thrashing the disks, does the load drop ? |
| 16:34.35 | Greenlight | Also, load average is a strange metric at times espeically with asterisk |
| 16:35.20 | Greenlight | For instance, at present I've load average of 20, and CPU is 200% (out of max 800%). System is running perfectly though. |
| 16:37.03 | cusco | dual processor, 2 cores each |
| 16:37.19 | cusco | 20~40% |
| 16:37.25 | cusco | so not even one core |
| 16:37.34 | cusco | load avg 6.xx |
| 16:37.39 | cusco | but I must state |
| 16:37.42 | cusco | wr're using realtime |
| 16:37.48 | cusco | and mysql is on the same box |
| 16:37.55 | cusco | I will add a new disk for mysql alone |
| 16:38.06 | Greenlight | Ahh, quite possible that disk access is indeed the bottleneck for you then |
| 16:38.42 | cusco | and periodic scripts reading extra stuff from mysql, even from asterisk -vrx "core show channels concise" |
| 16:39.12 | Greenlight | You ran iostat? |
| 16:41.48 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 16:43.45 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.37.86) |
| 16:47.35 | igcewieling | How does the universe know to break stuff when I've been up half the night upgrading customers? How? How? |
| 16:48.14 | Greenlight | I believe it's called "sods law" |
| 16:48.56 | *** join/#asterisk anonymouz666 (~anonymouz@189-105-205-226.user.veloxzone.com.br) |
| 16:49.10 | igcewieling | this stuff has nothing whatsoever to do with any upgraded customers. 8-| |
| 16:51.24 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.116) |
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| 17:01.33 | cusco | Greenlight: iostat? |
| 17:01.35 | cusco | I know iotop |
| 17:01.48 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 17:02.15 | cusco | http://paste.debian.net/1730/ |
| 17:03.54 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:03.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 17:49.46 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:55.54 | jeffspeff | What's the difference between ChannelRedirect() and Goto() ? |
| 17:56.17 | pabelanger | jeffspeff: GoTo is for dialplans |
| 17:56.26 | pabelanger | channelredirects are for channels |
| 17:56.41 | pabelanger | also |
| 17:56.52 | pabelanger | *CLI> core show application goto |
| 17:57.04 | igcewieling | The difference might be obvious from the application docs |
| 17:57.14 | [TK]D-Fender | jeffspeff, ChannelRedirect(channel|[[context|]extension|]priority) |
| 17:57.25 | [TK]D-Fender | jeffspeff, this allows you to toss ANOTHER channel around |
| 17:57.32 | jeffspeff | ok, got it |
| 17:57.42 | [TK]D-Fender | jeffspeff, Goto is for the channel you are calling it from |
| 17:57.54 | jeffspeff | was just looking for something else in the wiki and skimmed across that |
| 17:58.09 | [TK]D-Fender | jeffspeff, It's hidden in the big print ;) |
| 17:58.31 | jeffspeff | lol |
| 17:58.53 | jeffspeff | at first glance it seemed it duplicated goto(); but now i see the difference. thanks |
| 18:01.05 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:05.47 | *** join/#asterisk izbushka (~izbushka@193.23.225.11) |
| 18:29.01 | *** join/#asterisk roentgen (~arthur@openvpn/community/support/roentgen) |
| 18:37.38 | *** join/#asterisk axelm7 (~axelm7@186.135.12.190) |
| 18:40.11 | axelm7 | hi from Argentina. One of my customers wants me to implement a high availability Asterix solution for a small call center (8 agents). I am interested in the dual server with Digium R800 failover switch. Anybody know of an alternative (i.e. clone) version of the R800? |
| 18:41.02 | *** join/#asterisk przerull (~philip@50.56.205.232) |
| 18:41.25 | butthurtface | HA asterisk… that's something I'd like to learn about! |
| 18:41.26 | przerull | I am experiencing ERROR[15058]: app_dial.c:2694 dial_exec_full: Could not stop autoservice on calling channel |
| 18:41.32 | przerull | itermitantly in production |
| 18:41.39 | przerull | what might be causing this? |
| 18:41.45 | pabelanger | przerull: are you using System() or Shell()? |
| 18:41.53 | przerull | neither |
| 18:42.00 | przerull | i'm on 1.8.10 |
| 18:42.22 | przerull | sorry 1.8.5 |
| 18:42.26 | pabelanger | I had an issue where something outside asterisk was blocking, and when asterisk when to stop the channel, the error would be generated |
| 18:42.45 | pabelanger | So, my System() was talking longer then I expected |
| 18:42.46 | przerull | i am making pretty extensive use of agi |
| 18:43.15 | pabelanger | so, likely something in your agi script is block when asterisk goes to do something with that channel |
| 18:43.20 | pabelanger | blocked* |
| 18:44.05 | navaismo | axelm7, You should choose that hardware Rseries work like a charm |
| 18:44.20 | navaismo | worth every dollar |
| 18:45.02 | pabelanger | xorcom has something too |
| 18:45.55 | przerull | hmmmm. an interesting thought. so basically what happens is a call comes in, agi, then we originate an outbound channel using originate, which later get's joined with the inbound leg in a meetme conference, so you're thinking that agi is blocking my channels while they are trying to be redirected |
| 18:46.43 | przerull | in my example the outbound channel that get's originated is a local channel which does the dial out to the actual endpoint |
| 18:47.29 | *** join/#asterisk DelphiWorld (~TayebMeft@openvpn/user/DelphiWorld) |
| 18:47.35 | DelphiWorld | 'lo everyone |
| 18:47.55 | DelphiWorld | would someone tel me how to originate a call to an SPA2102 without registration? |
| 18:47.59 | *** join/#asterisk cmendes0101| (~cmendes01@72.1.46.254) |
| 18:48.22 | igcewieling | DelphiWorld: registration has nothing to do with making calls |
| 18:48.28 | axelm7 | The customer has two PBXs. Main PBX is an old Ericsson box with incoming E1 (30 lines) and 4 analog backup lines. Secondary PBX is a TeleVantage 7 used only for call center and SIP calls. |
| 18:48.51 | DelphiWorld | igcewieling: lol i know. i dont want to register SPA2102 but i want to call it directly |
| 18:48.53 | igcewieling | DelphiWorld: host=theipofthesipdevice |
| 18:49.19 | DelphiWorld | igcewieling: got it? |
| 18:49.21 | axelm7 | I am trying to replace this will two asterix boxes, an R800 for the analog failover, but I still need to solve the E1 failover |
| 18:49.31 | axelm7 | R850? |
| 18:50.48 | axelm7 | There has to be a cheaper way to do E1 failover than spending USD 1000 on an R850 |
| 18:51.11 | axelm7 | maybe some E1 to Ethernet converter |
| 18:53.04 | navaismo | maybe google can help, or another asterisk box redirecting the e1 traffic via sip to the other pbx based on the master cluster mac addr |
| 18:53.07 | axelm7 | sorry TDM over IP |
| 18:53.08 | igcewieling | axelm7: you unplug the E-1 from server 1 and plug it into server 2 |
| 18:53.15 | igcewieling | that is the cheap way. anything else will be expensive |
| 18:53.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:54.00 | axelm7 | igcewieling, but that is not fun to implement ;) |
| 18:54.15 | igcewieling | fun == money |
| 18:54.18 | *** join/#asterisk miztic (~gerard@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 18:54.23 | axelm7 | fun == not my money |
| 18:56.26 | *** join/#asterisk h34d3r (~h34d3r@host120-124-dynamic.43-79-r.retail.telecomitalia.it) |
| 18:58.25 | axelm7 | let's suppose dual asterisks with TDM410 and TE410 cards. Plus an R800 and an R850 for the failover. What kind of software do you guys recommend for a call center? I need a simple IVR and ACD. |
| 18:59.21 | axelm7 | Currently 8 agents, might get bumped up to 30 agents if we get an ExxonMobil contract we've been after for about a year |
| 19:00.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:05.23 | navaismo | software? But you are using asterisk. Or are you talking about report tools and stuff? if so I recommed queuemetrics |
| 19:06.40 | [TK]D-Fender | axelm7, We recommend Asterisk. |
| 19:06.48 | [TK]D-Fender | axelm7, Then again, we might be biased... |
| 19:06.52 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 19:12.48 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:17.10 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 19:19.29 | jeffspeff | lol |
| 19:24.46 | *** join/#asterisk dgilmore (~dgilmore@fedora/dgilmore) |
| 19:26.05 | dgilmore | i have a sip trunk setup. the other end rejects my registration with 1 401 error and sends digest auth info, * seems to be just ignoring it and trying again over and over |
| 19:26.25 | dgilmore | is there some way to tell * to use digest when registering? |
| 19:27.13 | pabelanger | dgilmore: pastebin your SIP debug |
| 19:30.45 | igcewieling | does anyone have a moment to look at a SUBSCRIBE message? http://pastebin.ca/2371864 I'm not all that good with SIP, but it seems like the phone is trying to subscribe to itself. |
| 19:32.57 | dgilmore | pabelanger: not real easy to sanitise it |
| 19:33.40 | pabelanger | dgilmore: then you are likely in the wrong room for help. No body here cares about that information |
| 19:33.43 | igcewieling | dgilmore: you should not sanitize it |
| 19:34.36 | igcewieling | dgilmore: passwords are never sent in cleartext with sip and the IPs and telephone numbers should not be considered sensitive |
| 19:35.51 | igcewieling | dgilmore: I find it almost never takes more than 48 hours after a server is put on the internet for a bot to find it and try hacking sip. The IP is NOT secret information |
| 19:38.20 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 19:38.22 | _abc_ | Is there some way to sign SEPxxx.xml.cnf.sgn conf files generated manually and located on a tftp server? |
| 19:38.25 | _abc_ | for cisco voip phones |
| 19:38.54 | dgilmore | igcewieling: its more the digest auth info im worried about |
| 19:39.46 | igcewieling | dgilmore: the password is NEVER sent unencrypted over the internet and each authorization request is encrypted with a different key. If you want help you'll need to provide the information |
| 19:40.18 | *** part/#asterisk przerull (~philip@50.56.205.232) |
| 19:40.22 | igcewieling | dgilmore: go read up on nonce's and Digest authentication |
| 19:40.53 | dgilmore | igcewieling: well if it helps at all teh otehr end is openuc |
| 19:40.55 | [TK]D-Fender | _abc_, Yes, it's called "programming", and we highly recommend it. It's awesome |
| 19:40.58 | navaismo | igcewieling, the from and the to are the same |
| 19:41.31 | _abc_ | [TK]D-Fender: huh? |
| 19:41.40 | SuperNull | _abc_ cisco cool will create the encrypted version www.cisco.com 'MODEL # firmware' and then choose profile builder |
| 19:41.41 | igcewieling | navaismo: *nod* Any idea what might cause that. |
| 19:41.46 | SuperNull | i literally did this today. |
| 19:41.56 | SuperNull | cool = tool* |
| 19:42.02 | [TK]D-Fender | _abc_, Apologies I missed a rather important word in your request :) |
| 19:42.32 | _abc_ | It is not encrypted just signed. The phone will have no .xml.cnf with no .sgn I want to generate the files as I do now on an asterisk host and dump them on the linux tftpd |
| 19:42.50 | _abc_ | It's just some lame signature thing added in the xml |
| 19:43.00 | SuperNull | not sure what that does for ya |
| 19:43.02 | SuperNull | we dont use it |
| 19:43.05 | SuperNull | fuck we dont even encrypt it ;) |
| 19:43.06 | _abc_ | I know |
| 19:43.09 | _abc_ | I asked in cisco heh |
| 19:43.14 | _abc_ | *cisco |
| 19:43.18 | SuperNull | im in cisco too |
| 19:43.19 | _abc_ | #cisco ... |
| 19:43.52 | SuperNull | _abc_ just build a base file ex: spa504g.cfg on a server.. preload the boxes .. with the proper profile rule and your good. |
| 19:44.00 | _abc_ | basically I generate the SEPxxx.xml.cnf files by detecting cdp broadcasts in realtime |
| 19:44.13 | SuperNull | make sure dhcp hands out the tftp server of the one that has the .cfg file. |
| 19:44.21 | SuperNull | you dont need to |
| 19:44.28 | _abc_ | SuperNull: if you have option 150 set in the dhcp server the phones will not boot unless they find their SEPxxx.xml.cnf |
| 19:44.29 | SuperNull | those phones ask for 'model#.cfg' |
| 19:44.41 | igcewieling | SuperNull: I think he is using Cisco phones, not Linksys (SPA) phones. |
| 19:44.44 | _abc_ | they loop on loading that. I tested this today for over 4 hours with 5 phones of different kinds |
| 19:44.52 | SuperNull | which model is _abc_ ? |
| 19:44.58 | SuperNull | er |
| 19:45.05 | _abc_ | 7940 7911 7960 7906 7931 |
| 19:45.07 | SuperNull | damn i can tell its almost end of day. i can barely type |
| 19:45.08 | _abc_ | enough? |
| 19:45.08 | SuperNull | ahhhhhh |
| 19:45.12 | _abc_ | brb |
| 19:45.19 | SuperNull | yeah. i remember those sucking back on asterisk 1.2 ;) |
| 19:45.36 | navaismo | igcewieling, not really but take a look on the mailbos addr on phone |
| 19:45.44 | SuperNull | _abc_ is this for a wide area or a lan ? |
| 19:47.21 | SuperNull | also if you are using isc dhcp you can probably automate it to create the files since the files (at least that i know of) are based off the mac. obviously ISC dhcp knows the mac that requested the ip (and can execute a script when an ip is handed to it for creation) |
| 19:48.06 | _abc_ | it is for a lab which tests phones before deployment and after being bought sh and when they need to be serviced or when someone thinks they need to be serviced |
| 19:48.22 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 19:48.27 | igcewieling | navaismo: no mailbox address configured on the Polycom. The messages button is set to *97 with a method of contact. |
| 19:48.31 | _abc_ | SuperNull: yes, but this works now for non .sgn files |
| 19:48.40 | _abc_ | Now I have phones which want a .sgn file |
| 19:48.50 | igcewieling | I actually hope the customer cancels like they are threatening to. Hosted voip solutions REALLY SUCK. |
| 19:48.52 | SuperNull | no idea what the sgn file does. |
| 19:48.53 | _abc_ | Thus my initial question: HOW do I sign the files |
| 19:49.25 | SuperNull | igcewieling we had a customer setup freepbx off us.. then they fired their it guy and said 'your liable' for some reason management took that as a challenge.. it failed horribly |
| 19:49.31 | igcewieling | _abc_: these phones don't seem like such a good deal anymore do they? |
| 19:49.35 | SuperNull | ;) |
| 19:49.50 | SuperNull | i would laugh if you have to apply for a certificate to test these phones |
| 19:50.11 | _abc_ | igcewieling: I never said they were, this is not my call |
| 19:50.21 | _abc_ | igcewieling: when they work they are okay, other than that, eww |
| 19:50.25 | _abc_ | even with sip on them |
| 19:50.29 | SuperNull | hey |
| 19:50.34 | _abc_ | incidentally the phones which want the .sgn are SIP... |
| 19:50.36 | SuperNull | 6 years ago they were good |
| 19:50.46 | SuperNull | define want. |
| 19:50.53 | SuperNull | requesting via tftp ? |
| 19:51.03 | igcewieling | SuperNull: we are using a Bicom systems box for hosted (based on Asterisk). The problem is hosted customers seem to want to set up a buddy watch for every single phone ON every single phone and ring every phone. So 20 phones monitoring 20 phones and a call comes in and rings 20 phones. |
| 19:51.15 | _abc_ | tftpd RRQ "SEPxxx.cnf.sgn" octet in syslog good enough SuperNull ? |
| 19:51.35 | _abc_ | and I tried to copy a non sgn file to sgn, it took it and did not like it |
| 19:51.44 | igcewieling | that is not the problem in this specific case, but is the problem with almost all other problem reports on hoste |
| 19:51.46 | _abc_ | looped on |
| 19:51.46 | igcewieling | d |
| 19:51.51 | SuperNull | igcewieling xzhibit would be proud. 'yo dawg i heard you like ringing on your ringing on your ringing' |
| 19:52.45 | SuperNull | what does it do if it just doesn't get the file? |
| 19:53.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.148) |
| 19:53.18 | _abc_ | one more thing: in Asterisk extensions.conf if one has a call target SIP/trunkname/username , is this correct? or is it SIP/username@trunkname? |
| 19:54.03 | SuperNull | i think yes might be the answer _abc_ |
| 19:54.11 | _abc_ | yes to both ? |
| 19:54.14 | _abc_ | yay |
| 19:54.17 | SuperNull | might have changed in newer versions but i think both are identical |
| 19:54.26 | _abc_ | ok I thought so |
| 19:54.35 | SuperNull | damn guys, its getting toasty in this office |
| 19:54.46 | _abc_ | how toasty? |
| 19:54.51 | SuperNull | like 85 damn degrees. |
| 19:54.54 | SuperNull | servers.. |
| 19:54.54 | _abc_ | has 27C in his room and is prepping to go to bed soon |
| 19:54.57 | SuperNull | no ac in yet. |
| 19:55.01 | SuperNull | er |
| 19:55.05 | SuperNull | converts quickly |
| 19:55.13 | SuperNull | aprox 30C |
| 19:55.29 | _abc_ | okay, that is like my noon temperature here in Eastern Europe >:) |
| 19:55.48 | _abc_ | Don't the servers get premature aging for working without a/c? |
| 19:56.17 | SuperNull | sure, we took it out cause' the cold was coming in .. (old building) |
| 19:56.20 | SuperNull | stupid window AC. |
| 19:56.37 | SuperNull | heh. we have 2 servers for the office in here nothing big so .. |
| 19:56.44 | SuperNull | all the goodies are in the data centers. |
| 19:57.47 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 19:58.44 | _abc_ | would like to see more conduction cooled server racks, they do a lot for the din and for comfort |
| 19:58.48 | _abc_ | air cooling is dead |
| 20:01.32 | SuperNull | i need something decent for my home pc. it keeps blowing through fans (cats...) |
| 20:01.47 | SuperNull | right now the fan sounds like a bicycle with a baseball card in it |
| 20:02.35 | [TK]D-Fender | SuperNull, Considered cleaning them occasionally? |
| 20:02.53 | _abc_ | naaah |
| 20:02.56 | _abc_ | buy new |
| 20:03.01 | _abc_ | also AIR FILTERS |
| 20:03.11 | _abc_ | I used nylon stockings for that before, it works |
| 20:03.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.148) |
| 20:03.30 | _abc_ | I sprayed one with cooking oil in a spray can for better dust retention. Works |
| 20:06.31 | _abc_ | of course these are total hacks |
| 20:06.50 | _abc_ | a slightly better solution is a HEPA sheet filter meant for real server fans, cut with scissors |
| 20:15.41 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 20:17.48 | *** join/#asterisk garymc (~chatzilla@host86-174-137-60.range86-174.btcentralplus.com) |
| 20:26.48 | SuperNull | i do clean them, the case does have filter.. welcome to cat dander ;) |
| 20:27.01 | SuperNull | Antec 300 with the front filters.. |
| 20:31.24 | [TK]D-Fender | Compressor <- |
| 20:31.43 | [TK]D-Fender | Checkout time, BBIAB |
| 20:33.22 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 20:40.05 | *** join/#asterisk ikarugaRS (ikaruga@ARennes-651-1-366-31.w2-14.abo.wanadoo.fr) |
| 20:40.11 | ikarugaRS | hi everybody |
| 20:40.42 | ikarugaRS | is there some frensh people here ? |
| 20:40.58 | Kobaz | no s in french |
| 20:41.08 | Kobaz | salut! |
| 20:41.17 | ikarugaRS | Bonsoir =) |
| 20:42.29 | Kobaz | mais je ne suis pas francais, mais je parle un peu |
| 20:42.39 | mmlj4 | there is no "you" in team |
| 20:42.46 | Kobaz | oui |
| 20:43.04 | ikarugaRS | Je me présente, je suis francais et m'attaque à asterisk en vu d'en comprendre ces principes |
| 20:43.43 | ikarugaRS | i intro myself, i'm frensh and try to understand how asterisk work |
| 20:43.52 | Kobaz | l'attaque, hah |
| 20:44.11 | ikarugaRS | j'ai testé asterisk now |
| 20:44.39 | ikarugaRS | sans difficulté, configuré et hebergé sous centos6 |
| 20:44.58 | ikarugaRS | mais j'ai tenté une installation en mode console c'est une autre affaire... |
| 20:45.17 | eirirs | french? wtf |
| 20:45.29 | ikarugaRS | i manage to install asterisknow with centos without probleme |
| 20:45.53 | mmlj4 | ...and we're back to boring anglais |
| 20:45.54 | ikarugaRS | but when i try to install asterisk on centos 6, it's begin really a hell... |
| 20:46.00 | Kobaz | ici, c'est ne pas pour asterisknow |
| 20:46.32 | ikarugaRS | c'est pour cela que je suis la, je suis sur asterisk 1.8.5.0 |
| 20:47.03 | ikarugaRS | j'ai suivi des indications d'installation, je peux te fournir le lien en mp ? |
| 20:47.09 | Kobaz | asterisk soulemont |
| 20:47.25 | Kobaz | seulement |
| 20:47.49 | ikarugaRS | je n'arrive pas à assimiler le principe asterisk |
| 20:48.36 | Kobaz | quel a la probleme? |
| 20:49.03 | ikarugaRS | je suis au stade de asterisk -r |
| 20:49.23 | Kobaz | et....? |
| 20:49.29 | ikarugaRS | je tombe en xxxx*CLI> |
| 20:49.46 | ikarugaRS | impossible de me servir des commandes |
| 20:50.03 | ikarugaRS | ou d'acceder au menuselect afin de configurer asterisk |
| 20:50.11 | Katty | turns text message notifications up on loud and sits next to file |
| 20:50.16 | Kobaz | c'est bon... mais quel vous besoin du? |
| 20:50.34 | ikarugaRS | mettre en place 2 softphone en local |
| 20:50.36 | Kobaz | quel commandes? |
| 20:50.40 | ikarugaRS | pour en comprendre le fonctionnement |
| 20:50.58 | ikarugaRS | mettre en place un ivr afin de dirriger l'appelant |
| 20:51.08 | ikarugaRS | des rudiments |
| 20:51.28 | ikarugaRS | je ne sais pas vraiment à quoi sert le *CLI |
| 20:51.31 | Kobaz | vous avais lit le livre encore? |
| 20:51.33 | Kobaz | ~book |
| 20:51.34 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:51.56 | Katty | Kobaz is so handy. he's like a pocket translator! |
| 20:52.22 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:52.23 | ikarugaRS | je suis désolé de me tourner vers toi : / |
| 20:52.30 | drmessano | He's like Google, + |
| 20:52.34 | Kobaz | 2 softphone en local ne pas difficile |
| 20:52.46 | ikarugaRS | mais j'ai du mal avec ce que le livre parle |
| 20:52.48 | Katty | hugs drmessano |
| 20:52.58 | drmessano | Whoa Google, +.. Google+ <-- Weird |
| 20:53.03 | drmessano | hugs Katty |
| 20:53.03 | Kobaz | heh |
| 20:53.40 | Kobaz | voir le livre pour chan_sip, sip.conf |
| 20:53.58 | ikarugaRS | tous se gère sous asterisk donc ? |
| 20:54.06 | ikarugaRS | asterisk n'a aucun lien avec apache ? |
| 20:54.11 | Katty | drmessano: i hear you're in the market for a shot gun. |
| 20:55.09 | Kobaz | oui, toute de sip, dans asterisk |
| 20:55.32 | drmessano | I am? |
| 20:55.34 | ikarugaRS | ok je comprends un peu mieux |
| 20:55.45 | ikarugaRS | merci pour ton aide en tous cas |
| 20:55.59 | ikarugaRS | cela me permet de demystifier asterisk |
| 20:56.01 | drmessano | Oh god, what did I post on Facebook now |
| 20:56.08 | drmessano | There goes my shot at a CEO job |
| 20:56.10 | Katty | drmessano: an autographed 12 guage browning a5 shotgun, to be specific. |
| 20:56.29 | drmessano | Oh, that thing I shared. lol |
| 20:56.40 | Katty | yep. you'll just have to stay IT director FOREVER. |
| 20:57.06 | drmessano | I just want the money. I wouldn't be able to shoot it, because its a collectible.. and I wouldn't want to display it... |
| 20:57.22 | drmessano | Thats my approach to most contests |
| 20:57.28 | mjordan | All I saw in there was Asterisk 1.8.5.0 => old |
| 20:57.36 | drmessano | "Win an autographed..." "OOOOOOH.. EBAAAAY" |
| 20:57.41 | Katty | drmessano: i'm sure anyone with an NRA bumper sticker would love to bid on it. |
| 20:57.48 | Kobaz | je suis desole, je suis un peu occupé |
| 20:58.00 | drmessano | Katty, if I win it, the more the merrier |
| 20:58.08 | ikarugaRS | je recherche asterisk1.0.2 |
| 20:58.10 | Katty | drmessano: good luck! |
| 20:58.27 | ikarugaRS | car j'ai un support de cours dessus |
| 20:58.36 | drmessano | I also hope I win that Steve Jobs autographed iPhone 5 |
| 20:58.39 | ikarugaRS | il à l'air bien plus simple que la version 1.8.0.5 |
| 20:58.42 | drmessano | crosses fingers |
| 20:58.43 | Katty | drmessano: maybe chance be in your favor. |
| 21:00.08 | drmessano | Katty, Maybe I will win a chance to eat dinner with the Robertsons. That would be neato |
| 21:00.44 | Katty | drmessano: well if that falls through, you could probably call up twisted. |
| 21:00.49 | Katty | drmessano: you'd be short a T, but hey... |
| 21:00.52 | eirirs | :) |
| 21:01.04 | Katty | drmessano: close enough, right? |
| 21:01.08 | drmessano | lol |
| 21:01.30 | ikarugaRS | ce lien ? http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html |
| 21:02.22 | Kobaz | oui, la voir |
| 21:02.49 | drmessano | Short a T? |
| 21:02.51 | ikarugaRS | Kobaz en tous cas je te remerci pour le temps que tu m'as consacré, je te souhaite bonne continuation. |
| 21:03.16 | ikarugaRS | Thank all for answers spend a nice night |
| 21:03.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.116) |
| 21:04.08 | Katty | drmessano: possibly a tea too. |
| 21:04.27 | Kobaz | merci... je regret mais je dois partir maintenant |
| 21:05.25 | Katty | now it is time to run :< |
| 21:05.43 | drmessano | Did Nike+ tell you that? |
| 21:05.49 | drmessano | That thing is a bastard |
| 21:05.57 | Katty | no, my bum did. |
| 21:06.04 | Katty | and the hips weren't helping the conversation either. |
| 21:06.13 | *** join/#asterisk RyanTG (~Thunderbi@65.100.106.194) |
| 21:06.15 | *** join/#asterisk linuxgeek (~linuxgeek@2001:470:1f0b:1f6::1) |
| 21:06.41 | drmessano | Oh, if it was my Bum it would be urging me to walk to the store and get a cold drink. One in Brazil. |
| 21:07.17 | drmessano | My thighs would argue for Argentia |
| 21:07.20 | Kobaz | Katty:: oui courir vite |
| 21:08.44 | Katty | Kobaz: i'm sorry dear, but i don't speak french. |
| 21:08.48 | Katty | Kobaz: or read it, for that matter. |
| 21:09.13 | drmessano | I kiss French, in Russian |
| 21:09.52 | Katty | in the Russian Soviet Federative Socialist Republic, you mean. |
| 21:10.09 | Katty | or are we calling them the russian federation these days? |
| 21:10.33 | eirirs | FSSSR ? |
| 21:10.34 | eirirs | :P |
| 21:10.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:10.45 | Katty | speaking of french, and overly complicated country names. |
| 21:10.47 | Katty | hi fender! |
| 21:12.07 | [TK]D-Fender | yar |
| 21:12.41 | Katty | [TK]D-Fender: what do you have on the weekend agenda? |
| 21:12.50 | Katty | [TK]D-Fender: booze, bars, and bass? |
| 21:16.14 | [TK]D-Fender | Katty: Nothing particular. friends playing every weekend like always... |
| 21:16.15 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:16.28 | [TK]D-Fender | Katty: everything is optional. Probably start up the biking season, etc |
| 21:16.30 | axelm7 | is there an AsterixNow version that runs on Centos 6 instead of Centos 5? |
| 21:16.37 | ikarugaRS | yes |
| 21:16.47 | ikarugaRS | asterisknow 3.0.0 |
| 21:17.25 | ikarugaRS | 64 and 32 bit version |
| 21:17.36 | axelm7 | oops my mistake. I was using the wrong iso file |
| 21:17.40 | axelm7 | sorry |
| 21:18.18 | ikarugaRS | ; ) |
| 21:18.30 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:18.36 | ikarugaRS | very easy to hand |
| 21:19.10 | ikarugaRS | here is the link http://www.asterisk.org/downloads/asterisknow choice the good iso ; ) |
| 21:19.16 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 21:19.55 | ikarugaRS | there is a chan #asterisknow about this topic |
| 21:20.37 | axelm7 | yeah, I got the iso from a digium page and it was really outdated |
| 21:21.10 | axelm7 | this one http://www.digium.com/en/products/asterisk/downloads |
| 21:21.13 | ikarugaRS | lol i'm trying to install a manual asterisk one |
| 21:21.20 | ikarugaRS | erf :/ |
| 21:21.54 | ikarugaRS | 2.0.2 |
| 21:22.19 | Qwell | axelm7: outdated? |
| 21:22.27 | Qwell | oh. damnit. |
| 21:22.45 | ikarugaRS | i just done a centos 6.4 86 with asterisknow 3.0.0 work like a charm |
| 21:23.18 | axelm7 | I use Centos for all my Linux stuff so I was happy to see that AsteriskNow is built on Centos |
| 21:23.33 | ikarugaRS | axelm7 i need to understand asterisk befor the next week for my work : / |
| 21:23.42 | ikarugaRS | ^^ |
| 21:24.18 | *** join/#asterisk radic (~radic@dslb-178-002-210-214.pools.arcor-ip.net) |
| 21:24.37 | axelm7 | what brand of t1/e1 and fxo/fxs cards are most popular? Do most people use the Digium clones? |
| 21:24.50 | Qwell | axelm7: The clones are crap. |
| 21:24.53 | Qwell | Buy real hardware. |
| 21:25.01 | ikarugaRS | +1 |
| 21:25.08 | WIMPy | That question is too unspecific. Digital or analog makes a huge difference. |
| 21:25.50 | axelm7 | ok, let's consider FXO/FXS cards. OpenVox or Yeastar |
| 21:26.32 | WIMPy | With analog stuff you're likely to have some trouble. Better buy something with support. |
| 21:26.48 | axelm7 | I also have a Dialogic Springboard card left over from the old TeleVantage PBX. I don't know if that works on Asterix or not |
| 21:27.29 | axelm7 | Dialogic Springware D/41JCT-LS |
| 21:27.42 | axelm7 | plus two of these Intel DI/SI16-R2 o DI/SI32-R2 Vendor 12C7 Device 4143 |
| 21:28.05 | leifmadsen | axelm7: it's Asterisk. Not Asterix (which is a French cartoon). |
| 21:28.08 | WIMPy | Probably only vial CAPI if at all. Although I heard a roumor that they have some sort of Asterisk support. |
| 21:28.15 | ikarugaRS | axelm7 you see it ? http://forums.asterisk.org/viewtopic.php?t=70374 |
| 21:28.41 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 21:28.43 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:28.52 | ikarugaRS | i think it's talking about what you ask |
| 21:29.16 | axelm7 | yeah, I saw that post |
| 21:29.18 | ikarugaRS | no anwsers : / |
| 21:29.24 | WIMPy | Not really. It certainly won;t be supported bu dahdi. But that need not be an issue. |
| 21:29.28 | axelm7 | the same guy also asked on another board |
| 21:29.45 | axelm7 | an the answer was "not supported, get a Digium card" |
| 21:29.52 | axelm7 | *and |
| 21:30.03 | WIMPy | For digital cards I prefer ones which are supported by Linux instead of dahdi. |
| 21:33.24 | axelm7 | WIMPy, I don't understand why you say that lack of dahdi support is not an issue |
| 21:33.27 | ikarugaRS | analog display = analogic phone ? |
| 21:34.08 | WIMPy | There are many channels and drivers for Asterisk. dahdi is only one of them. But that's most for the digital stuff. |
| 21:34.20 | [TK]D-Fender | WIMPy: You mean BRI more specifically no? |
| 21:34.25 | WIMPy | doesn't know much about analog. |
| 21:34.34 | [TK]D-Fender | WIMPy: I'm unaware of T1/E1 that I wouldn't just use DAHDI for... |
| 21:34.36 | WIMPy | BRI or PRI. |
| 21:34.48 | axelm7 | so is my Dialogic Springware card supported by some other driver? |
| 21:34.51 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 21:35.03 | WIMPy | There's no difference between BRI and PRI except for the speed. |
| 21:35.04 | [TK]D-Fender | WIMPy: Can you show me one that sees a real amount of marketshare? |
| 21:35.53 | WIMPy | I don;t sell the stuff so I have no idea anout marketshare. And why should I care about that? I want something that's supported and works. |
| 21:38.08 | [TK]D-Fender | WIMPy: Yeah, I'm fine with that. What non-DAHDi T1/E! cards have you seen used with *? |
| 21:38.08 | igcewieling | Analog? Isn't that what you get when you are too cheap to go with a PRI, but your internet service sucks too much to go with SIP? |
| 21:38.21 | [TK]D-Fender | WIMPy: BRI I know there are tons via CAPI, etc.... |
| 21:38.39 | igcewieling | the problem with non-DAHDI / non-Digium cards is that you have far, far less community support. |
| 21:39.01 | ikarugaRS | axelm7 dont wanna use 3cx or x-lite ? |
| 21:39.16 | axelm7 | Yeah |
| 21:39.19 | ikarugaRS | may be less expenssive : / |
| 21:39.24 | WIMPy | [TK]D-Fender: HFC-E1 by whatever vendor. And CAPI support in Asterisk seems to have gone out of fashion, but mISDN has been the smoothest for me so far. |
| 21:39.34 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mlfghuhyezpepeyc) |
| 21:39.35 | axelm7 | but I need to support 8 FXO lines that come from another PBX |
| 21:39.47 | [TK]D-Fender | WIMPy: Can you link a store? |
| 21:40.01 | [TK]D-Fender | WIMPy: For a specific model or two you might trust... |
| 21:40.15 | WIMPy | You can get them from Junghanns and Swyx. |
| 21:40.24 | WIMPy | Probably others as well. |
| 21:40.44 | WIMPy | But they are only available as single or dual span. |
| 21:40.57 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 21:41.04 | [TK]D-Fender | WIMPy: http://shop.junghanns.net/index.php/interfaces/singlee1-pci.html?___store=english |
| 21:41.07 | ikarugaRS | axelm7 erf wish to be usefull but my knowledge on it is off for this moment on FX0 : / |
| 21:41.13 | [TK]D-Fender | $$ |
| 21:42.48 | WIMPy | Luckily there were still quite some cards on offer on ebay when I bought mine, both Digium and noname. |
| 21:42.51 | axelm7 | 700 euros? does it come with a weekend hooker? |
| 21:43.47 | igcewieling | axelm7: interfacing with the PSTN is not cheap. |
| 21:44.17 | WIMPy | Not if you buy new :-) |
| 21:44.51 | WIMPy | And I'm pretty sure the Dialogic stuff has been much more expensive. |
| 21:45.02 | axelm7 | for sure |
| 21:45.41 | axelm7 | so the chinese Digium clone analog cards suck big time? |
| 21:46.07 | axelm7 | can't believe they would be so bad |
| 21:46.38 | axelm7 | what about OpenVox? |
| 21:46.48 | WIMPy | Don't take any chances for the analog stuff. |
| 21:47.37 | igcewieling | axelm7: I believe Openvox and the clones are all based on very old designs which Digium moved away from for a reason |
| 21:47.42 | [TK]D-Fender | axelm7: If you have problems with them you'll tend to find that their support is crap or completely non-existant. |
| 21:48.05 | axelm7 | just like the 3000 dollar Dialogic cards |
| 21:48.25 | axelm7 | but 2700 bucks cheaper |
| 21:48.27 | WIMPy | And check that they have drivers for current Asterisk versions in the first place. |
| 21:48.41 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 21:49.08 | WIMPy | It's however best to avoid any analog stuff if ANY possible. |
| 21:49.41 | axelm7 | WIMPy, I have some legacy equipment I need to connect to unfortunately |
| 21:50.06 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:51.18 | *** part/#asterisk DelphiWorld (~TayebMeft@openvpn/user/DelphiWorld) |
| 22:01.20 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 22:01.36 | ikarugaRS | i say you good night all |
| 22:01.48 | ikarugaRS | hope you'll find a way out axelm7 |
| 22:01.58 | ikarugaRS | bye |
| 22:02.04 | *** part/#asterisk ikarugaRS (ikaruga@ARennes-651-1-366-31.w2-14.abo.wanadoo.fr) |
| 22:03.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.91) |
| 22:05.03 | *** join/#asterisk madhatt (~madhatt@23.31.65.29) |
| 22:05.25 | madhatt | hey all, does anyone here use Snom phones? I'm trying to understand why Snom put in google analytics code into the web UI |
| 22:06.01 | WIMPy | Oh. Where? |
| 22:06.13 | madhatt | you can read a short post I made about it here - http://madhatt.com/index.php/20-technology/476-does-my-snom-300-track-me |
| 22:07.47 | WIMPy | Lots of pictures? |
| 22:08.37 | _abc_ | madhatt: a) everyone hopes to be bought out by goopile b) before that, they can make some money by selling their user's souls |
| 22:08.44 | _abc_ | er online identities |
| 22:09.06 | madhatt | here is the picture that matters most I think, it shows clearly that they are using 2 kinds of analytic software on the phones Web UI http://madhatt.com/images/otherfiles/test.jpg |
| 22:10.36 | WIMPy | Looks evil. |
| 22:11.06 | madhatt | so I tweeted them and they came out and said that they don't track but I believe this could be seen as a big security risk. |
| 22:11.12 | madhatt | 1. I log into my phone to edit a value, |
| 22:11.19 | madhatt | 2. the phone sends info to google analytics |
| 22:11.39 | WIMPy | Interestingly enough that's most likely illegal. |
| 22:11.53 | madhatt | 3. a bad Snom employee could then (perhaps) get my public IP address which almost always would have a PBX sitting on it. |
| 22:12.00 | _abc_ | ghostery is what? I don't recognize that UI OS- |
| 22:12.08 | _abc_ | mac? |
| 22:12.17 | madhatt | gostery is a chrome app that block tracking code |
| 22:12.20 | madhatt | yes, I am on a mac |
| 22:12.43 | madhatt | I don't want to throw rocks but I sell these phones/pbxs for a living and now I'm not sure I want to pitch Snom anymore because of thisl |
| 22:12.45 | madhatt | *this |
| 22:13.09 | _abc_ | are you sure it is the phone and chrome doing that? |
| 22:13.13 | axelm7 | sure looks bad. it would be interesting to see a pcap |
| 22:13.36 | madhatt | pcap, I got one of those! give me a few |
| 22:13.40 | _abc_ | and yes if it's true that is a problem |
| 22:14.14 | _abc_ | goes to sleep a bit |
| 22:14.26 | axelm7 | I rephrase. It would be interesting for someone (not me) to check a pcap |
| 22:14.35 | madhatt | I can't see how chrome is doing is because the UA number 1331074 which is being used seems to be Snoms |
| 22:15.14 | madhatt | Well, I'll get the Pcap loaded with the article so anyone can look |
| 22:16.55 | madhatt | holy crap! so in capturing the data from this SNOM phone I see that the little fucker sends it's mac address via a SIP packet to IP address http://224.0.1.75/ |
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| 22:19.44 | igcewieling | that is a multicast address |
| 22:19.58 | madhatt | so why would the phone want to send a SIP packet there? |
| 22:20.20 | igcewieling | maybe it is trying to join a multicast group? Polycoms support multicast for paging and maybe MoH |
| 22:21.06 | madhatt | interesting… ok so that's not that big of a deal then? but the tracking still may be an issue |
| 22:21.19 | igcewieling | madhatt: I am speculating |
| 22:21.56 | madhatt | well, in the capture I can see that the phone is not sending anything to goolge but I think that is a non-starter as it would be my computer that actually sends the info off to google at the request of the javascript code on the Snom phone's web UI (I believe) |
| 22:23.35 | madhatt | hrmm. I'm leary of providing the pcap file until I can whittle it down to just the google analytics code, but I'm unable to find what IP or FQDN it uses |
| 22:26.15 | madhatt | this is all I can find about the mcast address |
| 22:26.16 | madhatt | How does a caller find its local registrar? |
| 22:26.16 | madhatt | The local registrar is either manually configured or discovered via DHCP (http://www.rfc-editor.org/rfc/rfc3361.txt) . Another more theoretical option is: the SIP client issues a multicast registration request to the sip.mcast.net standard multicast address, which all registrars (are supposed to) listen to (but in practice not all do). |
| 22:27.08 | madhatt | alright, so I'm out, just wanted to pop in and ask about this. I'll come back and provide an update once Snom contacts me, thanks for listening folks! |
| 22:27.12 | igcewieling | madhatt: no idea, we don't use Mcast with Polycoms and we don't use SNOM |
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| 22:44.28 | doctorray | I'm trying to originate multiple calls simultaneously with AMI; currently I'm sending them one at a time and I'm not super happy on the delay between the first and the last origination... Is there a way to give them to AMI all at once? Or would I need to generate call files instead? |
| 22:45.27 | WIMPy | There's no need to wait. |
| 22:45.54 | doctorray | so just dump them all at once? currently I'm waiting for acknowledgement from my command |
| 22:46.41 | WIMPy | No. As long as you can keep track if you want to. |
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