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| 00:42.17 | tech | ok so i brought up a brand new spaknin frepbx + asterisk box. |
| 00:42.25 | tech | i am getting 408 registration timeouts when trying to register. |
| 00:42.37 | tech | triple checked settigns and they are all right/ |
| 00:42.42 | tech | installing wireshark now... |
| 00:44.04 | tech | while that is installing - any ideas? |
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| 01:02.36 | tech | hrm no i got now peer. |
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| 01:25.39 | *** join/#asterisk gentoo_fun2 (seb@jet.bayhost.net) |
| 01:26.43 | gentoo_fun2 | hey guys, quick question. I used deny=0.0.0.0/0.0.0.0 and permit=192.168.1.0/255.255.255.0 in my sip.conf for asterisk |
| 01:27.08 | gentoo_fun2 | but I am still getting script kiddies from the outside trying to register |
| 01:27.25 | *** join/#asterisk eugeneoden (~goden@southeast-invest.oneringnetworks.net) |
| 01:27.34 | gentoo_fun2 | any idea whats next? |
| 01:27.58 | WIMPy | Yes: Having fun with RTP. |
| 01:28.41 | WIMPy | Do you have deny/permit on all peers? |
| 01:29.45 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-dybvcdlcjwcgfopr) |
| 01:30.09 | gentoo_fun2 | nope |
| 01:30.23 | gentoo_fun2 | the peer stuff is provisioned |
| 01:30.42 | gentoo_fun2 | id like not to touch that stuff |
| 01:31.13 | gentoo_fun2 | theres no root location to put this that superceedes? |
| 01:31.33 | WIMPy | I don;t think so. |
| 01:31.38 | gentoo_fun2 | like I cannot put the deny/permit b4 the #include for all the peers? |
| 01:31.54 | gentoo_fun2 | bleh |
| 01:32.17 | WIMPy | You can use a template. |
| 01:34.35 | gentoo_fun2 | template for? |
| 01:35.07 | WIMPy | your peers |
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| 01:45.10 | gentoo_fun2 | ehh idk, I cant fuck with it too much, its a live system |
| 01:45.18 | gentoo_fun2 | been using iptables to block the shit for now |
| 01:45.27 | gentoo_fun2 | i gota go on site to relearn how all the provisioning stuff works |
| 01:45.51 | WIMPy | If you can use iptables, that probably the better idea, anyway. |
| 01:45.57 | gentoo_fun2 | looks simple enough, but the conf files that the peers/phones use are generated |
| 01:46.07 | gentoo_fun2 | yea but obv i cant block the entire internets :) |
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| 03:19.56 | raden | anyone figure out the asterisk 1.8 no music on hold with aastra phones issue ? |
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| 03:30.27 | jplank | does anyone else have access to a VVX1500? |
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| 07:00.28 | *** join/#asterisk recluze (~recluze@178.238.133.90) |
| 07:00.59 | recluze | hi. I'm trying to insert a hook in the same place where the CDR is written. I basically want to generate a web call (using AGI) at this point with all the info sent to CDR... any hints for that? |
| 07:02.34 | Juggie | trap it in the hangup portion of your dialplan |
| 07:02.39 | *** part/#asterisk recluze (~recluze@178.238.133.90) |
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| 07:03.10 | recluze | oops... window got closed... did I miss a message? |
| 07:03.24 | Juggie | yes, trap it in the hangup portion of your dialplan |
| 07:03.58 | Juggie | however you need to make sure your dialplan allows both calls hungup by the callee and the caller to get trapped |
| 07:04.01 | Juggie | so be sure to test. |
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| 07:04.31 | recluze | the issue is that there are many places for the hangup |
| 07:04.42 | recluze | I want to record ring groups, extensions, ivrs. ... everything |
| 07:04.55 | recluze | so, I was wondering if there is one place where all the CDRs are written |
| 07:05.05 | recluze | otherwise, I will have to insert the hooks in all the places, right? |
| 07:05.18 | Juggie | right. |
| 07:05.26 | Juggie | check what cdr drivers exist. |
| 07:05.28 | recluze | oh |
| 07:05.40 | recluze | currently only file CDRs are used |
| 07:05.40 | Juggie | vague memory tells me there might be a cdr curl? |
| 07:05.47 | Juggie | i've been away from * for a while. |
| 07:05.47 | recluze | (no mysql etc) |
| 07:05.52 | recluze | :O CDRU curl? |
| 07:06.35 | recluze | CDR* |
| 07:06.47 | kaldemar | recluze: depending on your dialplan structure, a single hangup extension may suffice. |
| 07:07.19 | Juggie | recluze, perhaps the better question to ask is what is your goal specifically |
| 07:07.38 | kaldemar | if you want to do it from dialplan. |
| 07:08.49 | Juggie | see q. |
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| 10:49.02 | EmleyMoor | I am having trouble getting caller ID from my BT line into asterisk. There is reportedly a patch out there for chan_dahdi.c to fix this, but all links to it give a 404. Any ideasL |
| 10:49.07 | EmleyMoor | ? |
| 10:51.48 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 10:53.07 | EmleyMoor | (It worked in 1.2 and needed a patch in 1.4 - now on 1.6.2.9 which still reportedly needs a patch) |
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| 12:28.21 | as001 | Hi, can I use channel bank with Asterisk to combine 4 analog telephone lines ? I want to archive when 2 calls come at analog line 1 at same time call 2 goes to line 2 (to avoid busy signal for caller 2). Is this possible ? |
| 12:30.42 | WIMPy | Ask your Telco how to set up CFB. |
| 12:32.12 | as001 | CFB ? |
| 12:32.44 | as001 | can you tell me what is CFB ? |
| 12:33.47 | WIMPy | Call Forward when Busy |
| 12:35.29 | as001 | ok thanks |
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| 13:22.22 | benngard | i am a little bit lazy, can some1 tell me what i should type after mailcmd= when i change from exim to postfix |
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| 14:02.30 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 14:03.09 | MrTelephone | Is there a code update for comparing digest with endpoints that share the same source ip:port? |
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| 14:07.19 | recluze | hi. Is there an asterisk command that will allow me to register with an ITSP... I know it can be done using register strings but I need to do it manually |
| 14:09.26 | MrTelephone | I wouldn't say there is |
| 14:10.25 | recluze | I see an 'sip unregister' command ... so there should be an equiv. register one :( |
| 14:11.55 | MrTelephone | just add it to sip.conf and hit rleoad |
| 14:12.02 | MrTelephone | type reload |
| 14:13.25 | kaldemar | sip reload will only reload sip instead of all modules. |
| 14:17.51 | MrTelephone | I don't even see an unregister command in my console |
| 14:17.55 | MrTelephone | just sip show registry |
| 14:17.56 | zkn | "sip unregister <peer>" will unregister the peer from your system, I don't think you can force your system to register with your ITSP any other way than adding the register line to sip.conf (if the ITSP requires you to register) and running command "sip reload" in CLI |
| 14:19.12 | MrTelephone | "We need the users to use the same authentication user name until we support proper authentication by digest auth name" I found this in chan_sip.c. Is there any headway on this project? |
| 14:20.19 | recluze | my problem is that if I sip reload or reload any other way, my ITSP thinks I'm spamming them |
| 14:20.24 | recluze | since i have many trunks with them |
| 14:20.32 | recluze | so, I need to register one trunk at a time ... |
| 14:20.53 | recluze | so, any alternative strategy would be appreciated as sip reload doesn't work (it re-registers everything) |
| 14:21.22 | MrTelephone | they shouldn't be blocking legitimate clients |
| 14:22.32 | MrTelephone | I would deal with the ITSP instead of banging your head against the desk on this one. |
| 14:25.01 | MrTelephone | Even worse case scenerio you reload 200 times in 5 minutes I can't see it increasing a PIII 900mhz load by 1%. |
| 14:26.05 | *** join/#asterisk sourcode (~code@ppp-58-11-75-225.revip2.asianet.co.th) |
| 14:26.42 | MrTelephone | At least your reloads are legitimate. I'll tell you what is irritating is those idiots using brute force attacks trying to figure out sip accounts. |
| 14:28.41 | zkn | <MrTelephone: got to agree there |
| 14:29.58 | MrTelephone | I'm not understanding the logic there because they are not even remotely close to guessing the usernames. |
| 14:30.11 | recluze | :) |
| 14:30.22 | MrTelephone | It's just a waste of bandwidth |
| 14:31.31 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:32.20 | recluze | that's actually pretty good advice. I'm going to talk to the ITSP people to get my IP un-banned |
| 14:32.27 | MrTelephone | imagine if that is how magic jack works? a bunch of guys hacking sip accounts and charging 17 bucks a year to use them |
| 14:33.48 | zkn | i have been using iptables to drop all attempts to connect to TCP and UDP port 5060 by counting connection attempts but somehow the bots still managed to scan the system :/ |
| 14:33.55 | MrTelephone | recluze, I think they are just being a little too strict on the banning policies. |
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| 14:35.08 | MrTelephone | It's too hard to block everything. It's normal to get port scanned. When I changed my ssh ports to something other than 22 noone has even tried to brute force at all. |
| 14:35.46 | MrTelephone | Some guy had a shell script made up already that scanned the asterisk log files for failed auth attempts and banned the ip in iptables |
| 14:36.32 | zkn | fail2ban works the same way, although it doesn't work to well for me with Asterisk |
| 14:36.37 | MrTelephone | I downloaded a subnet list of all known .ru addresses and banned the entire country once for a laugh |
| 14:36.54 | zkn | ssh brute forces are blocked well with fail2ban |
| 14:36.54 | MrTelephone | wasn't fail2ban written for asterisk? |
| 14:37.20 | MrTelephone | are you getting massive port connection attacks? I haven't noticed anything like that yet |
| 14:38.56 | zkn | i guess my server's hostname is too obvious and attracts scanners or smth, i see attempts at least 2-3 times a week, some times even per night |
| 14:39.41 | MrTelephone | are you too big to change your default port? |
| 14:40.09 | zkn | i could try that.. |
| 14:40.56 | MrTelephone | It's a big job |
| 14:41.14 | MrTelephone | Reconfigure every client :( |
| 14:41.41 | MrTelephone | I just remember getting nailed on port 22 all the time and changed it and people don't even know it's ssh |
| 14:41.43 | zkn | it's a shame when they scan during an active call, then you can hear the call breaking up |
| 14:42.04 | MrTelephone | that would piss me off |
| 14:42.09 | zkn | oh yeah |
| 14:42.45 | MrTelephone | is your machine on a cable/dsl type connection then? |
| 14:45.07 | zkn | several machines in various countries |
| 14:45.15 | MrTelephone | I have a very overpowered xeon machine with asterisk on it servince maybe 100 people. I haven't experiences any jitter due to port scans or antyhing |
| 14:45.26 | MrTelephone | you sound massive |
| 14:46.32 | MrTelephone | zkn, magic jack? |
| 14:46.58 | zkn | well, it might be some config issue,too, i cannot say I'm an expert in Asterisk and know how to fine tune the system to get everything out of it, so when the bots start scaning then the effect is audible during calls for sure |
| 14:47.48 | zkn | what's magic jack? |
| 14:48.06 | zkn | not been using it... |
| 14:48.17 | MrTelephone | it's some phone company in north america that offers phone for 17 dollars a year. They give you some USB stick to plug in your computer |
| 14:49.35 | MrTelephone | zkn, do you run 'top' or something when you get scanned? |
| 14:49.42 | zkn | htop |
| 14:50.33 | MrTelephone | for 17 dollars a year this guy must have asterisk in his basement with a bunch of X100 cards hooked to analog phone lines to his neighbors house so he doesn't get a bill |
| 14:50.34 | MrTelephone | lol |
| 14:50.43 | zkn | usually it happes so fast than when I hear complaints about "call quality" and get to the CLI then everything is back to normal again, but /var/log/asterisk/messages reveals the truth |
| 14:51.53 | MrTelephone | I have a problem building served by a cable modem and they get blanks downstream which is very strange. I should check the logs as well. Are they experiencing stuttering or long blank periods? |
| 14:52.55 | MrTelephone | If you know for sure that its port scans try and switch your default ports and it might help quite a bit. |
| 14:52.57 | zkn | stuttering |
| 14:53.26 | zkn | just like packets are dropping or smth |
| 14:53.27 | MrTelephone | if you goto #romania you might even be able to talk to the people that are scanning you |
| 14:53.28 | MrTelephone | lol |
| 14:53.37 | zkn | lol |
| 14:53.48 | MrTelephone | The romanians got me bad a couple years ago |
| 14:54.02 | MrTelephone | fuckin kids get paid to goto school so they have too much time on their hands |
| 14:54.13 | MrTelephone | in north america if we arn't on welfare we are too busy to hack |
| 14:54.30 | MrTelephone | too busy working to pay for our computers to get reformatted at the shop |
| 14:55.25 | MrTelephone | so what kind of solution do you have for testing to see if the packets are dropping or just showing up late/out of order? |
| 14:56.42 | zkn | right now I have no real solution yet, i have been experimenting with IPtables and fail2ban but I think I need to read up on the security topic a bit to get an idea what else to try besides changing the ports |
| 14:57.37 | tzanger | hahaha |
| 14:58.11 | MrTelephone | If your on dsl or cable it might not be able to handle the sudden inrush of traffic? if that is the case there won't me much you can do? |
| 14:59.08 | MrTelephone | I'm starting a business going around spanking all the kids that are port flodding our production servers. |
| 14:59.23 | MrTelephone | s/flodding/flooding/ |
| 15:00.19 | MrTelephone | First thing I guess is to call their parents |
| 15:00.54 | MrTelephone | "while you were playing world of warcraft your kids hacked 3 banks" |
| 15:08.00 | *** join/#asterisk eugeneoden (~goden@southeast-invest.oneringnetworks.net) |
| 15:15.07 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:18.18 | benngard | little off-topic but i think and some know the anser, i am looking for an english word, when a trasfer a call to b and b dont answer, a is "getting the call back", in sweden we say a got a "transfer retur", whats the english word(s) for that? |
| 15:27.08 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 15:30.55 | *** join/#asterisk Aut0Exec (~root@65.75.65.130) |
| 15:32.33 | Aut0Exec | hi guys... I have a beginners question for you... is only 1 fxs tied to 1 sip configuration? for example... if i'm using a basic ATA and I use a phone splitter on the analog phone side to connect 2 analog phones. will both of them pick up the same sip configs? |
| 15:32.45 | *** join/#asterisk ripon (~ripon@78-86-161-207.zone2.bethere.co.uk) |
| 15:34.31 | kaldemar | Aut0Exec: yes |
| 15:36.03 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-190-64-53.nycmny.east.verizon.net) |
| 15:36.10 | KingDavidNYC | <PROTECTED> |
| 15:36.30 | Aut0Exec | oh i see... so to add more physical phones... I would have get a better ATA like a card? |
| 15:36.51 | Aut0Exec | but even the cards that I see only allow 2 fxo and 2 fxs ports? |
| 15:37.03 | Aut0Exec | so still a maximum of 2 physical phones on thoese cards right? |
| 15:38.20 | *** join/#asterisk LemensTS (~matthew@71.86.32.146) |
| 15:38.26 | Aut0Exec | I guess what would make it much easier in a basic office setup is to use ip phones going into a switch panel.. with 1 ATA connected from a fxo yes? |
| 15:38.38 | wdoekes2 | zkn: I've experienced audio stuttering when the romanians are doing brute force register attempts |
| 15:39.02 | wdoekes2 | my solution, iptables --hashlimit |
| 15:39.09 | zkn | what's with the romanians? :) |
| 15:39.13 | wdoekes2 | s/--/-m / |
| 15:39.40 | LemensTS | Good Morning/Evening/Night all, I have 2 pots lines going into a Digium card, and I can't get rid of echo. I was thinking about getting an ATA with FXO ports and converting them to SIP outside of asterisk, then bringing them into asterisk as SIP. Do those ATA's do pretty good job as ridding echo? |
| 15:40.02 | zkn | wdoekes2: will look into --hashlimit |
| 15:41.49 | ripon | hi everybody |
| 15:42.54 | MrTelephone | benngard, forward no answer |
| 15:43.51 | MrTelephone | zkn, most of the attacking hosts are .ro |
| 15:44.20 | ripon | any plain asterisk users who for a bit paypal magic , can offer an hrs or so tuition on asterisk monday or tuesday ? |
| 15:45.01 | ripon | preferbly in early eveing time- (uk) ? |
| 15:45.28 | MrTelephone | just ask quesitons here it is free? |
| 15:45.33 | zkn | MrTelephone: haven't noticed that yet, never really bothered investigating the IP addresses |
| 15:45.49 | KingDavidNYC | <PROTECTED> |
| 15:45.52 | zkn | but good to know |
| 15:46.09 | MrTelephone | how many packets are you actually getting?? how many/sec |
| 15:46.34 | wdoekes2 | KingDavidNYC: not any different from any other outgoing call |
| 15:47.59 | MrTelephone | KingDavidNYC, that is too vague of a description for people to help you. |
| 15:48.16 | ripon | can not receive incoming calls from did or make calls- local extension to extension,,, have contexts set up right ,, but suspect that when doing a "reload" the extensions.conf file isnt saving with newest dialplan |
| 15:48.40 | ripon | dialplan reload / reload dialplan |
| 15:48.41 | LemensTS | What happened to TKD_Fender? |
| 15:48.47 | ripon | doesnt work |
| 15:48.57 | benngard | MrTelephone: thx |
| 15:49.18 | MrTelephone | ripon, no errors in the output? |
| 15:49.46 | MrTelephone | double check you didn't mispell extensions.conf when you saved. I did that before and took me 35 minutes to figure it out |
| 15:50.05 | ripon | oh you mean like errors at line 100, or whatever ? no... |
| 15:50.24 | MrTelephone | I had trouble with musiconhold last week not loading the new settings so I actually had to unload the module and reload. |
| 15:50.35 | ripon | really? |
| 15:50.39 | MrTelephone | ripon, someones it will say extension already exists and it skips it |
| 15:50.46 | ripon | how do you unload a module and relaod ??? |
| 15:51.14 | wdoekes2 | hm? dialplan reload has never failed me |
| 15:51.20 | MrTelephone | me neither |
| 15:51.34 | ripon | ive googled dialplan reload ,,, quite a fewpeople it fails |
| 15:51.41 | wdoekes2 | ripon: dialplan show to see which one you got |
| 15:51.42 | MrTelephone | what version? |
| 15:52.09 | KingDavidNYC | wdoekes2: what about incoming calls? how does opensips knows how to foward the call to the asterisk server? I see an example in config.cfg that uses ASTERISK_IP, what is that, a variable? should I replace with the real IP or is there a table where I define this constant? |
| 15:52.28 | Aut0Exec | would a good basic office setup be 1 ATA going into a swith with IP phones? |
| 15:52.44 | wdoekes2 | KingDavidNYC: still too vague. I can guess where you're heading, but it's not immediately obvious |
| 15:53.16 | MrTelephone | aut0exec. what is the ata for? |
| 15:54.11 | ripon | Asterisk PBX 1.6.2.5-0ubuntu1.3 ,,, is the version and machine im running on |
| 15:54.19 | Aut0Exec | ata is to get my existing analog phone integrated |
| 15:54.21 | KingDavidNYC | wdoekes2: I just dont understand how the concept of forwardign from opensips to asterisk works... I am supposing asterisk works as a sip client |
| 15:54.45 | MrTelephone | auto0exec, how do you get your calls out? you have a voip provider? |
| 15:55.09 | wdoekes2 | if (method == "INVITE") { rewritehost("my.asterisk.server.com"); t_relay(); } |
| 15:55.29 | ripon | done dialplan show- its not in therer |
| 15:55.32 | wdoekes2 | you're not asking the right questions |
| 15:55.45 | MrTelephone | kingdavidnyc, you setup a peer in sip.conf with the ip address of your sip router. Asterisk does act like a client. |
| 15:55.47 | ripon | only the usual sample configurations.. not my custom one |
| 15:55.54 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-zbbymfpdzpllvifn) |
| 15:55.55 | Aut0Exec | MrTelephone, first off.. i havent done anything yet... i'm about to. and second.. I would like to use both my local line and a voip provider to make outgoing calls.. perhaps overseas calls going out using a dial code first.. like dial 8, then the number.. |
| 15:56.38 | ripon | is it something to do with writeprotect etc? - and thats why latest dialplans are not being saved? |
| 15:56.58 | Aut0Exec | MrTelephone, my question was only how a basic office would be setup (hardware wise).. if just 1 ATA would be sufficient going into a switch with ip phones? |
| 15:57.04 | MrTelephone | So you need a linux machine with asterisk and some kind of single port FXO card for the telco? I would say thats a good office setup compared to using some proprietary nortel/avaya system. If you have the time to set it up. |
| 15:57.22 | Aut0Exec | ok |
| 15:57.28 | *** join/#asterisk cyphorious (~cyphoriou@212-183-74-13.adsl.highway.telekom.at) |
| 15:57.37 | Aut0Exec | kewl |
| 15:57.42 | MrTelephone | autoexec. you just need a server and some kind of digium hardware to bring in your telephone line |
| 15:57.53 | KingDavidNYC | MrTelephone: anywhere I can see a sample of this in sip.conf? |
| 15:57.56 | MrTelephone | the ata and sip phones all use the same switch and register with your asterisk server. |
| 15:58.23 | Aut0Exec | MrTelephone, sounds good... when u say digium hardware.. u are refering to an ATA with fxo right? |
| 15:58.31 | kaldemar | no telephony hardware is needed on the asterisk box. an ATA with an FXO port will do. |
| 16:00.04 | MrTelephone | where do you get an ata with an fxo port? |
| 16:00.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 16:00.10 | MrTelephone | autoexec, yeah |
| 16:00.25 | Aut0Exec | i was thinking just a sipura 3000 or the likes |
| 16:00.33 | Aut0Exec | or linksys ATA |
| 16:00.36 | Aut0Exec | etc etc |
| 16:00.58 | KingDavidNYC | wdoeks2: thank you, that is the direction |
| 16:02.16 | MrTelephone | nice, i didn't even know they made fxo atas. That is all you need then |
| 16:02.23 | Aut0Exec | k |
| 16:03.06 | MrTelephone | I have a supirua 8 port fxs I don't need anymore. I wonder if it has any value left. |
| 16:03.29 | MrTelephone | KingDavidNYC. When I was playing around I just read some of the asterisk+openser wiki's |
| 16:04.06 | Aut0Exec | last nub question.... where is the softbutton configs for phones stored? like example... I have a nice cisco ip phone and I want to map the buttons.. where do I configure that? |
| 16:04.28 | KingDavidNYC | MrTelephone: Man, the truth has to be said, documentation and wiki on anything SER is AWFUL |
| 16:04.57 | MrTelephone | King. Put something liek this in sip.conf |
| 16:05.04 | MrTelephone | [openser] |
| 16:05.04 | MrTelephone | context=from-openser |
| 16:05.04 | MrTelephone | type=peer |
| 16:05.05 | MrTelephone | host=IP |
| 16:05.05 | MrTelephone | dtmf=rfc2833 |
| 16:05.05 | MrTelephone | canreinvite=no |
| 16:05.12 | MrTelephone | I'm going to get in trouble for that one |
| 16:05.17 | Aut0Exec | lol |
| 16:05.26 | KingDavidNYC | MrTelephone: thanks a lot |
| 16:05.45 | MrTelephone | extensions.conf you put [from-openser] then put your dialplan underneath |
| 16:06.09 | KingDavidNYC | MrTelefone: host=IP, which Ip, SIP-SERver-IP?? |
| 16:06.42 | MrTelephone | your sip server ip. there is no authentication so your allowing invites only from that IP |
| 16:07.07 | KingDavidNYC | MrTelephone: great |
| 16:07.11 | Aut0Exec | MrTelephone, where do i map my buttons for my ip phones? |
| 16:08.16 | MrTelephone | not sure, are you sure you can? You might only be able to use them for speed dial for people in your contact list. |
| 16:09.15 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 16:09.54 | KingDavidNYC | MrTelephone and wdoekes2: Thank you guys |
| 16:10.03 | MrTelephone | kingdavidnyc, the sip server forwards what you dial to asterisk. If you dialed 500 then you put exten => 500,1,Dial(SIP/asdkajdda) |
| 16:10.31 | MrTelephone | KingDavid, good luck with that. It's fun and frustrating at the same time trying to figure this stuff out |
| 16:11.14 | KingDavidNYC | MrTelephone: Thanks |
| 16:11.20 | Aut0Exec | thanks guys |
| 16:11.25 | MrTelephone | You might have to start using sip debug ip <sipserver> on asterisk to actually watch the messages. It teaches you a lot |
| 16:11.35 | *** join/#asterisk wesphillips (~wphillips@137.237.194.136) |
| 16:11.37 | MrTelephone | aut0exec. what did you want the buttons to do? |
| 16:11.52 | MrTelephone | Do you have one of those nice touch screen phones? |
| 16:11.59 | *** part/#asterisk wesphillips (~wphillips@137.237.194.136) |
| 16:13.42 | MrTelephone | The guy who made asterisk should be proud of himself. This software is amazing |
| 16:18.16 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 16:22.06 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-rzizrbpunvnszymn) |
| 16:26.28 | LemensTS | What do you recommend for ATA with 2 FXO ports |
| 16:27.42 | iprouteth0 | i have a linksys I like alot |
| 16:27.52 | LemensTS | What model |
| 16:28.10 | iprouteth0 | PAP2 |
| 16:28.27 | LemensTS | Thats 2 FXS ports |
| 16:28.37 | iprouteth0 | oh FXO |
| 16:28.40 | iprouteth0 | missed that |
| 16:28.49 | iprouteth0 | good question |
| 16:29.03 | iprouteth0 | have you looked at the digium cards? |
| 16:29.31 | LemensTS | Yea, was wanting to try an ATA. Ive used ZOOM ata with FXO ports before. |
| 16:29.37 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:32.59 | iprouteth0 | not sure I've seen any with 2 FXO |
| 16:33.04 | iprouteth0 | maybe cisco vg200 |
| 16:34.38 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 16:50.03 | LemensTS | try now |
| 16:50.09 | LemensTS | wrong window... |
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| 19:12.58 | jaytee | anyone have recommendations for a softphone for Android phones? |
| 19:16.19 | carrar | Is there a Bria client for Android? |
| 19:16.55 | carrar | looks like there is |
| 19:17.07 | carrar | http://www.counterpath.com/bria-android-edition.html |
| 19:18.47 | benngard | http://www.acrobits.cz/27/acrobits-mobile-voip-solutions <- i am using it on an iPhone, works fine, nut eating battery like a camel in the desrt :( |
| 19:19.10 | carrar | camel's are battery hungrey |
| 19:21.10 | *** part/#asterisk fabrianchi (~fabrianch@unaffiliated/fabrianchi) |
| 19:24.50 | *** join/#asterisk cyphorious (~cyphoriou@212-183-74-13.adsl.highway.telekom.at) |
| 19:27.44 | SunTsu | carrar: I use http://sipdroid.org/, which can even handle two sip accounts |
| 19:28.01 | carrar | bria can handle lots of accounts |
| 19:28.23 | carrar | least the version I have on my iPhone |
| 19:28.29 | carrar | works very nice |
| 19:29.17 | SunTsu | carrar: looks nice, yes |
| 19:31.10 | *** join/#asterisk ShadowHntr (~alex@wikipedia/Shadowhntr) |
| 19:35.05 | jaytee | just tried the 3CX free softphone. seems to work pretty good. |
| 19:36.51 | *** join/#asterisk Aut0Exec (~root@65.75.65.130) |
| 19:37.00 | ShadowHntr | i had trouble with everything but Blink |
| 19:37.06 | ShadowHntr | when i was testing |
| 19:37.08 | carrar | haha |
| 19:37.43 | Aut0Exec | hi guys I need some advice on a product... is the grandsteam ht503 good for an ATA? its only 40 bucks on amazon and i'm thinking of buying it now... I want to use it as an fxo and fxs to get my existing local line integraged into asterisk |
| 19:37.58 | carrar | ewe |
| 19:38.02 | Aut0Exec | :( |
| 19:38.04 | ShadowHntr | get a digium card |
| 19:38.07 | ShadowHntr | like a tdm400p |
| 19:38.11 | Aut0Exec | expensive man |
| 19:38.16 | Aut0Exec | :| |
| 19:38.18 | Aut0Exec | i'm a nub |
| 19:38.20 | Aut0Exec | just starting out |
| 19:38.23 | Aut0Exec | just messing around |
| 19:38.32 | carrar | waste more time with crappy hardware |
| 19:38.37 | carrar | time == $ |
| 19:38.40 | Aut0Exec | lol |
| 19:39.02 | Aut0Exec | i'm just learning now dude... when I get some real skills then I will get the good hardware |
| 19:39.09 | carrar | You will be glade you purchase the right hardware |
| 19:39.12 | Aut0Exec | right nowi'm just running debian/asterisk |
| 19:39.12 | carrar | purchsaed |
| 19:39.16 | Aut0Exec | virtualized in virtualbox |
| 19:39.26 | Aut0Exec | how much is the card? |
| 19:39.35 | carrar | you need ebaylessions? |
| 19:39.39 | Aut0Exec | lol |
| 19:39.41 | Aut0Exec | lulz |
| 19:39.46 | ShadowHntr | 85 dollars on ebay |
| 19:39.56 | Aut0Exec | k |
| 19:40.23 | Aut0Exec | 2 fxo, 2 fxs right? |
| 19:40.29 | ShadowHntr | customizable |
| 19:40.32 | carrar | whatever you need |
| 19:40.44 | ShadowHntr | i have one with one fxo and three fxs'es |
| 19:40.53 | Aut0Exec | ok nice |
| 19:41.05 | Aut0Exec | u have it as a home/office setup? |
| 19:41.17 | ShadowHntr | setting it up for a hobbyist space |
| 19:41.23 | Aut0Exec | oh ok |
| 19:41.36 | Aut0Exec | using all analog phones? |
| 19:41.53 | ShadowHntr | going to offer the functionality of analog phones if people bring them :) |
| 19:42.06 | Aut0Exec | what u need the 3 fxs ports for? |
| 19:42.13 | ShadowHntr | analog phones on the internal side |
| 19:42.43 | Aut0Exec | yeah well I was thinking to get that grandsteam... and plug that into my switch and then get like 2 or 3 ip phones.. |
| 19:42.57 | Aut0Exec | 1 fxs, and 2 or 3 ip phones |
| 19:43.02 | Aut0Exec | like 4 phones all together |
| 19:43.16 | Aut0Exec | 1 analog of course |
| 19:43.17 | carrar | all good till you said grandstream |
| 19:43.21 | Aut0Exec | hahaha |
| 19:43.28 | carrar | ~grandstream |
| 19:43.29 | infobot | [grandstream] the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 19:43.41 | ShadowHntr | ~polycom |
| 19:43.41 | infobot | [polycom] The Polycom Song by dialing sip:polycom@leifmadsen.com or ISN 7659*460. Polycom phone are devices that are favoured by much of the community and range in price from under $100 and upwards. |
| 19:43.44 | Aut0Exec | lol |
| 19:43.49 | Aut0Exec | ok how bout that linksys ATA? |
| 19:43.53 | ShadowHntr | ~linksys |
| 19:43.53 | infobot | methinks linksys is a tool of satan |
| 19:43.53 | Aut0Exec | better? |
| 19:43.59 | carrar | linksys works |
| 19:44.02 | Aut0Exec | lol |
| 19:44.14 | ShadowHntr | ~sipura |
| 19:44.14 | infobot | Sipura has been aquired by cisco, see ~cisco and ~linksys . |
| 19:44.26 | carrar | AudioCodes are nice |
| 19:44.29 | Aut0Exec | yeah linksys ATA replaces sipura dude |
| 19:44.58 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 19:45.03 | carrar | or get yourself a T1 card and connect it to a ADIT600 |
| 19:45.50 | Aut0Exec | man to be honest... I dont want to mess with cards right now... i'm just getting my feet wet |
| 19:46.01 | Aut0Exec | if the ATA box is gonna work then why not |
| 19:46.03 | carrar | then don't bother with analog |
| 19:47.43 | Aut0Exec | man u guys are high tech.. i'm coming off of watching http://asterisk.org/videos... and they recommend to start with that for beginners |
| 19:47.54 | carrar | haha |
| 19:47.57 | Aut0Exec | :( |
| 19:48.07 | Aut0Exec | come on dude.. dont laugh at me |
| 19:48.10 | Aut0Exec | i'm new to this |
| 19:48.21 | carrar | then just start with 2 sip phones |
| 19:48.30 | Aut0Exec | thats what i'm using now |
| 19:48.31 | carrar | or 2 free soft phones |
| 19:48.43 | Aut0Exec | but I want my analog phone integrated |
| 19:48.48 | Aut0Exec | local phone |
| 19:49.13 | carrar | Go buy a linksys ATA then |
| 19:49.23 | Aut0Exec | kk |
| 19:49.28 | Aut0Exec | better than the grandstream right? |
| 19:49.33 | Aut0Exec | :| |
| 19:49.39 | carrar | yes |
| 19:49.43 | Aut0Exec | lol k |
| 19:50.32 | carrar | http://www.cisco.com/en/US/products/ps10027/index.html |
| 19:51.00 | Aut0Exec | yeah thats the one |
| 19:51.04 | Aut0Exec | :) |
| 19:52.28 | Aut0Exec | carrar, you integrated your local line? |
| 19:52.44 | carrar | I don't have any local lines |
| 19:52.50 | Aut0Exec | ohh |
| 19:52.51 | Aut0Exec | ok |
| 19:52.53 | Aut0Exec | gatcha |
| 19:53.00 | carrar | Analog sucks |
| 19:53.04 | Aut0Exec | lol |
| 19:53.10 | carrar | seriously |
| 19:53.20 | Aut0Exec | well I live in the bahamas... we dont have digital carriers here |
| 19:53.23 | Aut0Exec | not yet at least |
| 19:53.30 | Aut0Exec | well not for mainstream use |
| 19:53.55 | Aut0Exec | so it will have to suffice |
| 19:54.10 | Aut0Exec | my goal is to integrate a voip solution with my local carrier. |
| 19:54.32 | carrar | then you want the Digium card |
| 19:54.37 | carrar | for best performance |
| 19:54.41 | ShadowHntr | well keep us appraised on your progress, it's cool to hear people in the community working out :) |
| 19:55.16 | Aut0Exec | ok |
| 19:57.23 | zkn | jaytee, get your Android phone up to day running Ginerbread - it has sip stack built in, no need to install separate 3rd party application |
| 19:57.30 | zkn | day=date |
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| 20:19.58 | philfine | Hello everyone, have upgrade debian to squeeze and now kernel (2.6.32) does not come with dahdi module ? Can anyone guide me on getting dahdi back to work ? |
| 20:21.23 | SiNGLer | did you install dahdi from source? |
| 20:21.29 | WIMPy | The same way your did it the last time. |
| 20:22.43 | philfine | Nop |
| 20:23.09 | philfine | Well, in last time there was lenny and it came with a module for the card |
| 20:23.28 | philfine | Now with 2.4.32 there is no such module compiled at least |
| 20:23.44 | philfine | Module was wctdm |
| 20:24.35 | philfine | So card doesn't even appear on lspci |
| 20:27.08 | philfine | Will reboot to older kernel to see if it comes back |
| 20:28.42 | philfine | On booting I have Module dahdi not found, missing /dev/dahdi |
| 20:30.53 | WIMPy | Linux has never had dahdi included. You need to install it yourself. |
| 20:35.44 | carrar | Just click on START |
| 20:41.30 | philfine | WIMPy: Your right, I have installed dahdi from source myself, but the kernel was there from the start I guess :S |
| 20:41.34 | philfine | Maybe not :S |
| 20:42.59 | SiNGLer | dahdi installation installs modules for kernel, so if kernel changed - modules needs to be recompiled |
| 20:43.39 | SiNGLer | strangest thing is that lspci does not show a card - it should show, because it does not depend on dahdi kernel module |
| 20:44.04 | SiNGLer | maybe you are missing some firmware for your pci controller? |
| 20:44.17 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-zmkpwgdgoouzgmmv) |
| 20:44.23 | carrar | Just start over with a clean install |
| 20:45.45 | philfine | The card manual claims Iwould need wctdm |
| 20:45.58 | philfine | Does it comes with dahdi ? |
| 20:46.03 | carrar | yes |
| 20:46.19 | carrar | wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.4.1+2.4.1.tar.gz |
| 20:46.52 | philfine | Ok |
| 20:46.52 | philfine | Maybe I should update dahdi too ? |
| 20:46.52 | philfine | I have 2.4.0 |
| 20:47.32 | carrar | I don't know what you are even trying to do |
| 20:47.39 | carrar | but soundsl ike you have a mess |
| 20:48.08 | carrar | You start changing your kenel you better very well know what you are doing and what will be effected |
| 20:48.41 | philfine | I didn't change kernel |
| 20:48.56 | philfine | And if I did I can assure you I would now pretty much what to do :D |
| 20:49.02 | philfine | know |
| 20:49.09 | carrar | I would install a latest reelease of linux fresh them |
| 20:49.17 | carrar | and not use a 2.4 kernel |
| 20:49.21 | philfine | Anyway, I have just upgraded debian |
| 20:49.33 | philfine | Didn't remembered that I had installed dahdi from source |
| 20:49.39 | philfine | Thought it was part of the kernel |
| 20:49.55 | carrar | if you don't remember then you probably use packages/rpm's |
| 20:50.14 | philfine | Sorry |
| 20:50.23 | philfine | 2.4.32 I meanth 2.6.32 |
| 20:50.34 | carrar | so you DID upgrade the kernel |
| 20:50.53 | philfine | I do not use 2.4 for more then 5 years or more :D |
| 20:51.30 | carrar | Well if I were you, I would install everything from source and compile |
| 20:51.42 | philfine | carrar: but your saying debian already comes with a package with dahdi modules |
| 20:51.57 | carrar | I don't use debian |
| 20:52.00 | carrar | so I do not know |
| 20:52.05 | carrar | but I highly doubt it |
| 20:52.35 | carrar | But if you installed asterisk from a package manager it probably automatically installed dahdi |
| 20:52.43 | carrar | since it's needed |
| 20:53.08 | WIMPy | WAS needed |
| 20:54.36 | carrar | confernce timing comes from dahdi I believe |
| 20:54.49 | carrar | meetme |
| 20:55.14 | philfine | It has a package dadhi-linux but doesn't seem to appear in modules list |
| 20:55.28 | WIMPy | Yes, but if you don't need MeetMe, you don't need dahdi any more. |
| 20:55.32 | carrar | haha |
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| 21:25.07 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
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| 21:25.50 | CRCinAU_ | yawns |
| 21:25.53 | CRCinAU_ | morning all |
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| 21:28.11 | CRCinAU_ | p3nguin: fyi: "Conferencing was tested and working with Cisco 7940/60, 7941/61, 7970/71, 7965 etc. and working fine. I am not sure if the WLAN phones work yet." |
| 21:28.19 | CRCinAU_ | regarding chan-sccp-b trunk :) |
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| 21:38.36 | p3nguin | crcinau_: But it only works in Asterisk 1.6.something branch, because they don't currently support 1.8 branch and 1.4 branch does not have ConfBridge. Does that sound about right? |
| 21:40.39 | CRCinAU_ | I believe so |
| 21:40.45 | CRCinAU_ | however I still haven't got it working ;) |
| 21:40.54 | CRCinAU_ | i'm checking out the trunk svn again to do more playing though. |
| 21:43.45 | p3nguin | I will never use 1.6.anything. |
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| 21:45.16 | p3nguin | The closest you will get me to it is that I might consider testing chan_sccp-b features on 1.6.2 branch in a test environment, but I will never put it on any of my production environments. |
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| 21:45.27 | CRCinAU_ | oh? |
| 21:47.37 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:50.00 | riddlebox | hello, what are people going to do with gizmo shutting down? Is there a service that is similar, that I can forward my google voice calls to, so that I get free incoming calls? |
| 21:50.52 | p3nguin | You can forward your Google Voice number to any phone number you want. I recommend forwarding it to a phone number you actually own/have control of. |
| 21:51.18 | WIMPy | You can have numbers, you don't own? |
| 21:52.37 | p3nguin | Well, I don't own any phone numbers. I have control of several, and I have several that I was granted use of. |
| 21:52.54 | p3nguin | And there's no comma in, "You can have numbers you don't own?" |
| 21:54.18 | WIMPy | Here, you own your numbers. |
| 21:54.46 | p3nguin | Can you explain how that works? |
| 21:55.09 | p3nguin | Is there a title of ownership or something? |
| 21:55.13 | WIMPy | What do you want to know? |
| 21:55.36 | p3nguin | I'm just curious how it works. |
| 21:55.50 | ripon | hi all - does anyone use siptosis?? |
| 21:56.39 | riddlebox | p3nguin, do you think google will eventually become a sip provider? |
| 21:56.52 | ripon | everytime i try to pass a call from asterisk to siptosis- this is what i get " SIP/siptosis-00000011 is circuit-busy" |
| 21:57.12 | WIMPy | The only point where it's of any importance if you want your number ported to another carrier. |
| 21:57.13 | p3nguin | riddlebox: I don't quite understand the term "sip provider." It really doesn't make sense to me. |
| 21:57.52 | p3nguin | riddlebox: In addition to what I told you seven minutes ago, you can also use Asterisk 1.8.3 and configure it to interconnect with Google Voice and Google Chat directly. |
| 21:58.24 | p3nguin | riddlebox: I've done it that way for a client and it seems to work quite well so far. |
| 21:59.44 | riddlebox | p3nguin, yes I understand that, and I am doing it as well, but right now calls go out of google voice, and show my google voice number, if someone calls it back, it is forwarded to gizmo which gives me a free DID for incoming, so then calls are totally free inbound and outbound |
| 21:59.47 | p3nguin | riddlebox: You configure chan_gtalk and res_jabber to work directly with Google and skip forwarding numbers completely. |
| 22:00.37 | p3nguin | riddlebox: If you would configure chan_gtalk and res_jabber properly, there will be no forwarding to anything. |
| 22:01.01 | riddlebox | p3nguin, ahh so you can setup the chat part to do it, I will look into that |
| 22:01.19 | riddlebox | didnt catch that |
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| 22:01.37 | p3nguin | riddlebox: Instead of forwarding the number to another number or gizmo, you uncheck any forwarding numbers, then check mark the one that says gtalk or google chat (I can't remember how it is labeled. |
| 22:01.54 | p3nguin | ) |
| 22:02.07 | riddlebox | p3nguin, yeah after you said it I logged in and saw that, now I have more to read about ;) |
| 22:02.12 | carrar | SIP provider is anyone who can provide you access to the PSTN via RFC 3261 |
| 22:02.17 | volga629 | which video codec is preferable use on asterisk ? |
| 22:02.26 | p3nguin | carrar: That would be an ITSP. |
| 22:02.38 | carrar | call it what you will |
| 22:02.42 | carrar | SIP |
| 22:02.48 | carrar | vs MGCP |
| 22:02.51 | carrar | or h.232 |
| 22:02.54 | p3nguin | I mean, you don't go around saying "http provider." |
| 22:02.59 | carrar | I do! |
| 22:03.07 | carrar | I need a new http provider |
| 22:03.09 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 22:03.31 | p3nguin | I need to call my http provider and see if they will give me higher speeds. |
| 22:03.53 | riddlebox | ohh that reminds me mine said they raised our speeds to 8mb |
| 22:03.58 | p3nguin | My IRC and FTP provider sent me a new modem recently. |
| 22:04.07 | p3nguin | riddlebox: Which one? |
| 22:04.13 | riddlebox | charter |
| 22:04.46 | carrar | well you don't pay for http/irc/ftp |
| 22:04.47 | riddlebox | I think they lied, I just hit 5mb |
| 22:04.49 | carrar | but if you did |
| 22:04.50 | p3nguin | riddlebox: Charter upgraded the 5 Mbit service to 8 Mbit service MONTHS AGO. |
| 22:04.54 | carrar | you could call them a provider |
| 22:05.00 | kaldemar | ripon: that line alone is not very informative. enable sip debug to see what is going on when you make a call. |
| 22:05.06 | p3nguin | riddlebox: Now they have increased the 8 Mbit service to 12 Mbit service. For free. |
| 22:05.22 | riddlebox | p3nguin, I see the billboards in st louis all the time, but everytime I check it I get 5mb |
| 22:05.47 | p3nguin | riddlebox: So now the lowest speed is still the 1 Mbit service, and the second level is 12 Mbit service. There is nothing in between. |
| 22:06.00 | riddlebox | wow did the rates go up too? |
| 22:06.10 | p3nguin | riddlebox: Nope. It was another free speed increase. |
| 22:06.12 | riddlebox | I cant remember what I pay a month |
| 22:06.18 | carrar | too much! |
| 22:06.19 | p3nguin | $45 |
| 22:06.24 | p3nguin | for inet |
| 22:06.28 | carrar | ouch |
| 22:06.40 | riddlebox | plus like $10 for taxes too lol |
| 22:06.45 | p3nguin | 12 Mbit for $45 is a bit high, but there's not much other choice. |
| 22:06.56 | carrar | 660 a year you could spend on something else! |
| 22:06.58 | riddlebox | I have always had good luck with their service |
| 22:07.25 | riddlebox | in 5 years we have only had 1 outage |
| 22:07.47 | p3nguin | You can get it for $29.99/mo. for 12 months when you bundle it. |
| 22:08.12 | p3nguin | The new speed is 12 down, 1 up. |
| 22:08.17 | carrar | Make your work pay for your internet! |
| 22:08.41 | riddlebox | we have dish network for tv service multi room dvr's are the way to go |
| 22:09.41 | riddlebox | plus I was tired of buying hardware for mythtv -- had 4 tuner cards and in one evening I lost my power supply on the pc and it fried all of the cards, $600 worth of cards |
| 22:09.50 | p3nguin | The current services are: 1 down, 128 k up. 12 down, 1 up. 18 down, 2 up. 25 down, 3 up. 60 down, 5 up. |
| 22:10.36 | p3nguin | I get 60 down, but I get more like 6-7 up. |
| 22:11.23 | riddlebox | wow thats nuts |
| 22:12.05 | p3nguin | Something like five years ago I talked with Charter Business about what speed of service I could get. The guy said the only thing he could give me was 5 down and 1 up. |
| 22:12.37 | p3nguin | I had to explain to him that he must not understand how services work, that I really would be much more satisfied with 1 down and 5 up. |
| 22:13.34 | riddlebox | yeah |
| 22:14.01 | p3nguin | He wasn't able to grasp how upload/download works, that when someone would download FROM me, it uses their 5 Mbit downstream, but it would be using my 1 Mbit upstream. |
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| 22:14.49 | riddlebox | those guys dont know how it all works, they just know how to hook it up |
| 22:15.09 | p3nguin | They don't even really know how to hook up stuff. |
| 22:15.45 | p3nguin | I'm not sure if he ever caught on or not, but I eventually told him that $250/mo. for 1 Mbit upstream and one static IP address was effing ridiculous, and I would not be accepting his offer. |
| 22:15.55 | riddlebox | look at their phone techs, they did a service cut for one of our customers, from AT&T, the guy installed their modem, then ran a feeder cable to the AT&T demarc, and wrapped his pairs directly to AT&T which then fed to us |
| 22:16.36 | riddlebox | notice I said wrapped, he didnt even use a punchdown |
| 22:16.44 | p3nguin | riddlebox: Are you saying he did it correctly or incorrectly? |
| 22:16.51 | riddlebox | incorrectly |
| 22:17.05 | p3nguin | Oh, there's no punch block in the outside demarc, is there? I've never seen one. |
| 22:17.31 | p3nguin | I always run into the screws, which each has half a dozen washers on it. |
| 22:17.32 | riddlebox | he was actually feeding dialtone back onto AT&T |
| 22:17.50 | p3nguin | I thought that's what you were saying, but I wasn't quite sure if I understood you right or not. |
| 22:17.55 | riddlebox | p3nguin, this was a business not a home, it is a 66 block inside |
| 22:18.23 | p3nguin | Oh, inside. Punch blocks inside are common, and I would have expected him to be able to use it properly. |
| 22:18.44 | riddlebox | I would expect him to if he is doing service cuts! |
| 22:19.11 | p3nguin | I told you they don't really know how to hook up stuff. |
| 22:19.41 | riddlebox | he didnt even understand what he did wrong, I told the customer and they called him back, I showed him, and told him that someone could be tapping into their lines, and he said ohh well they have unlimited calling |
| 22:19.43 | p3nguin | They honestly don't know anything. I fix more stuff than they do. |
| 22:20.00 | p3nguin | hahaha |
| 22:20.22 | p3nguin | That's a nice response. |
| 22:20.58 | riddlebox | man I was pissed, I told the guy how can you do this work if you dont understand wiring |
| 22:21.14 | p3nguin | No one that works for them understands anything. |
| 22:22.21 | riddlebox | yeah we actually get hired to work on their office pbx systems lol |
| 22:26.01 | p3nguin | They always want to run an eQA any time I call them to tell them something is wrong. If they run the eQA and it shows the numbers are fine, they think nothing can be wrong. |
| 22:26.20 | p3nguin | If the numbers are within the correct range, it's find and nothing else can possibly be wrong. |
| 22:26.30 | p3nguin | s/find/fine/ |
| 22:26.30 | CRCinAU_ | p3nguin: conferencing works :D |
| 22:26.49 | p3nguin | crcinau_: What version of Asterisk do you use? |
| 22:26.56 | CRCinAU_ | 1.6.2.17 |
| 22:27.00 | CRCinAU_ | with trunk |
| 22:27.03 | p3nguin | crcinau_: You have a 'conf' soft key? |
| 22:27.07 | CRCinAU_ | err with trunk for chan-sccp-b |
| 22:27.11 | p3nguin | crcinau_: not confrn |
| 22:27.11 | CRCinAU_ | confrn |
| 22:27.15 | p3nguin | crcinau_: oh |
| 22:27.24 | p3nguin | crcinau_: I thought they changed it to conf only. |
| 22:27.44 | CRCinAU_ | but now doing: sccp restart SEPblah causes asterisk to crash ;) |
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| 22:27.59 | p3nguin | crcinau_: So how did you get 'confrn' to finally show up on the phone soft key? It was showing ??? on the key. |
| 22:28.06 | CRCinAU_ | usign trunk. |
| 22:28.13 | CRCinAU_ | then compile with --enable-conference |
| 22:28.25 | CRCinAU_ | the usual make; make install |
| 22:28.45 | CRCinAU_ | then change the softkeymap a little to replace meetme with confrn |
| 22:28.51 | p3nguin | Okay, so the --enable-conference was the key to success on the button. |
| 22:28.56 | CRCinAU_ | seems to be |
| 22:29.14 | p3nguin | I can't compile it with --enable-conference because my Asterisk does not support it. |
| 22:29.23 | CRCinAU_ | upgrade ;) |
| 22:29.24 | CRCinAU_ | lol |
| 22:29.37 | p3nguin | I'm using the CURRENT version. There is nothing to upgrade. |
| 22:29.59 | p3nguin | Changing to a different branch that I have no desire to use would not be an upgrade. |
| 22:30.03 | CRCinAU_ | yeah, it is kinda annoying that there is 1.4.x, 1.6.x, and 1.8.x |
| 22:30.04 | CRCinAU_ | but meh |
| 22:30.20 | p3nguin | There's 1.4, 1.6.x, and 1.8. |
| 22:32.02 | p3nguin | I recently tried 1.8.2.4 for testing purposes, since I eventually want to change to the 1.8 branch, but it ended in failure. |
| 22:32.46 | CRCinAU_ | :| |
| 22:32.47 | p3nguin | I admit that I didn't spend a lot of time on it to see why it ended in failure. Maybe soon I can find out what's wrong and get it going. |
| 22:32.58 | CRCinAU_ | does chan-sccp-b support 1.8 yet? |
| 22:33.08 | CRCinAU_ | I don't recall seeing anything about it |
| 22:33.10 | p3nguin | Not that I can tell. |
| 22:33.29 | p3nguin | I tried compiling it against my installed 1.8.2.4, and it didn't work out. |
| 22:33.51 | p3nguin | It isn't listed as a supported version, but I tried anyway just to make sure. |
| 22:33.57 | CRCinAU_ | nods |
| 22:34.19 | CRCinAU_ | I think thats what makes it hard having 3 branches :\ |
| 22:34.36 | CRCinAU_ | eveything gets fragmented |
| 22:37.03 | p3nguin | I use 1.4 for anything of my own that is in production. |
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