| 00:06.38 | WiretapSeven | woot, found my old 48V telco PSU :P |
| 00:07.13 | *** join/#asterisk Yonn (Yon@c80-217-241-69.bredband.comhem.se) |
| 00:24.38 | WiretapSeven | damnit, it won't power my phone |
| 00:25.04 | WiretapSeven | suppose it is designed to power analogue phones |
| 00:34.55 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 01:48.09 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 01:49.00 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 01:53.44 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 01:55.50 | *** join/#asterisk gruvfunk_AFK (~chatzilla@user-160uac8.cable.mindspring.com) |
| 01:56.48 | l1nuxman | sup |
| 01:56.54 | gruvfunk | hey did you get it? |
| 01:56.59 | l1nuxman | ALMOST |
| 01:57.18 | l1nuxman | I'm just stuck trying to get an DHCP address from my router to the HT503 |
| 01:57.45 | l1nuxman | it won't assign |
| 01:57.58 | l1nuxman | and its in dynamic mode |
| 01:58.04 | gruvfunk | have you set the HT503 LAN port to DHCP? |
| 01:58.22 | gruvfunk | I think you want a static address, bro |
| 01:58.38 | l1nuxman | no I changed the set up a bit |
| 01:58.44 | gruvfunk | due to the fact that the HT503 will be connecting to the Asterisk PBX as a specific IP address configured in sip.conf |
| 01:58.45 | l1nuxman | I have a router feeding addresses |
| 01:59.02 | gruvfunk | well it doesn't sound like your issue is Asterisk |
| 01:59.16 | l1nuxman | not at the moment |
| 02:05.44 | *** join/#asterisk sourcode (~code@ppp-58-8-86-248.revip2.asianet.co.th) |
| 02:13.50 | gruvfunk | did you get it to work the way you intended? leaving the home phones plugged directly to POTS? |
| 02:15.11 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 02:20.56 | luckyaba | is Dahdi packaged wiht Asterisk or does that need to be installed as well? |
| 02:21.54 | rogersja1 | it is a separate tar ball |
| 02:22.12 | rogersja1 | and would need to be compiled before asterisk |
| 02:24.48 | luckyaba | gotcha |
| 02:24.50 | luckyaba | thanks |
| 02:37.01 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 02:37.05 | l1nuxman | ok I got somethign working :) |
| 02:37.20 | l1nuxman | except no phone calls are coming in |
| 02:37.35 | l1nuxman | I can press 123 on my phone and get a welcome message |
| 02:38.41 | gruvfunk | ? |
| 02:41.19 | l1nuxman | I can press 123 in my house and get a recording |
| 02:49.52 | l1nuxman | 1 => 1234,Household Mailbox,you@yourdomain.com ;1234 is the password right? |
| 02:50.01 | l1nuxman | I enter that when it asks for password |
| 02:50.16 | l1nuxman | but login incorrect |
| 02:57.36 | gruvfunk | what is that? |
| 02:57.44 | gruvfunk | I don't understand what that is |
| 02:58.00 | gruvfunk | oh voicemail? |
| 02:58.20 | gruvfunk | yes, 1234 |
| 03:08.23 | gnutsy | gruvfunk: ok figured out a much simpler way for extension rollovers with VM time out :) |
| 03:08.48 | gruvfunk | yeah? |
| 03:09.17 | gnutsy | I wass trying to do it the hard way: examine for any failure status after each dial then conditionally dial next... |
| 03:09.53 | gnutsy | this meant macros and a gotoif for every possible failure. |
| 03:11.23 | gnutsy | All needed was to ad a gotoif after each sequential dial to check DIALSTATUS for NOANSWER ! |
| 03:11.32 | gnutsy | exten = s,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?vmail) |
| 03:13.42 | *** join/#asterisk cosmo83 (~phani@1.23.36.49) |
| 03:14.54 | cosmo83 | Hi Guys, Iam trying to setup a MIM call recording setup for two of my PRIs. Everything seems to work fine. Here is my extensions.conf http://pastebin.com/TT386YgN . The problem iam having is that iam not able to record CDR. Can anyone help ? |
| 03:15.41 | gnutsy | now any failure will rool over to the next extension, and only if it actually finds & rings one to full time does it goes to VM |
| 03:16.26 | gruvfunk | perfect! |
| 03:17.01 | gnutsy | it works soooweet! :) |
| 03:18.22 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 03:20.36 | *** join/#asterisk OldMonk (~raju@122.176.227.88) |
| 03:21.37 | OldMonk | hi. how do i make asterisk send a different indication if the called SIP channel is busy? at the moment it's sending a ring, so the called party (also SIP) can't distinguish |
| 03:23.23 | gruvfunk | cosmo83 maybe post your cdr.conf |
| 03:23.58 | gnutsy | OldMonk: check indications.conf |
| 03:24.06 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 03:24.44 | OldMonk | gnutsy: thanks |
| 03:26.33 | OldMonk | gnutsy: hmm, indications.conf seems to have distinguished tones already set. however, here the the calling party hears a ring, while the called party gets an incoming call indication |
| 03:27.23 | OldMonk | to technically it /is/ ringing, but the called party is on another call |
| 03:27.43 | gruvfunk | so you want to give them busy tone instead? |
| 03:28.02 | OldMonk | gruvfunk: yes, some tone that indicates that the called party is already on a call |
| 03:29.53 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 03:30.22 | gruvfunk | i'm thinking: check PEERSTATUS, and then feed them the Busy application |
| 03:31.12 | gruvfunk | or a ChanIsAvail rather |
| 03:31.47 | gruvfunk | scratch that |
| 03:31.54 | OldMonk | ok :) |
| 03:33.25 | OldMonk | ChanIsAvail should do it: test for AST_DEVICE_BUSY |
| 03:34.35 | OldMonk | oh... "ChanIsAvail is not a solution to tell you conclusively whether the channel is busy or not, it is primarily to tell you whether it would be possible to send a call there. Whether that call would end up being accepted or not is entirely up to the peer that we send the call to, and they could easily reject the call even though they do not appear to be 'busy'." |
| 03:35.06 | gruvfunk | was just going to say same :) |
| 03:36.07 | OldMonk | let's look up 'busylevel' in SipPeer instead... |
| 03:40.33 | OldMonk | gah, that needs call-limit, which is deprecated |
| 03:42.16 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 03:42.35 | l1nuxman | how come I get an error shown here? http://pastebin.com/w8Srmmp5 |
| 03:44.13 | rogersja1 | because that extension does no exist in that context, either explicitly or by pattern matching |
| 03:44.43 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:44.54 | OldMonk | get rid of the @ht503fxo in _9X |
| 03:46.24 | OldMonk | [i presume that's an extension] |
| 03:46.41 | gruvfunk | that's his FXO pots port to the pstn |
| 03:47.06 | OldMonk | ah ok, my bad |
| 03:47.47 | l1nuxman | should I change the 'n' to ht503fxs ? |
| 03:47.54 | OldMonk | so how is the call coming to extension 503fxo anyway? |
| 03:47.55 | rogersja1 | OldMonk, and in anycase, that is his outgoing plan. op was referring to inbound |
| 03:48.05 | OldMonk | rogersja1: got it |
| 03:48.23 | rogersja1 | l1nuxman, no change the 's' |
| 03:48.28 | OldMonk | l1nuxman: replace "s" with hto503fxo |
| 03:48.41 | OldMonk | s/hto/ht/ |
| 03:49.20 | gruvfunk | l1nuxman you have configured your HT503 to send calls to 'ht503fxs' which is not configured extension in asterisk as OldMonk says |
| 03:49.39 | gruvfunk | either tell the HT503 to send calls to the 's' extension |
| 03:49.55 | gruvfunk | or, see OldMonk's mod |
| 03:51.26 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:51.50 | OldMonk | configuring the device to send to "s" does make more sense, though |
| 03:52.26 | OldMonk | unless you plan to have multiple, heterogenous devices |
| 03:52.55 | rogersja1 | generally, one would not send a call to 's' |
| 03:53.14 | rogersja1 | s is a special extension that is used when all other matching fails |
| 03:53.27 | gruvfunk | in this case there is no matching, all calls are to be handed off |
| 03:53.33 | gruvfunk | from the HT503 ATA to Asterisk |
| 03:53.47 | gruvfunk | but, many ways to skin a cat - just trying to get l1nuxman up and running |
| 03:54.08 | l1nuxman | OldMonks suggestion worked :) |
| 03:54.22 | OldMonk | cool, you may pay me now ;) |
| 03:54.27 | l1nuxman | hahaa |
| 03:54.28 | gruvfunk | lol |
| 03:54.40 | l1nuxman | what's the charge |
| 03:55.08 | OldMonk | very nominal -- typical fee is half your kingdom and your daughter's hand in marriage |
| 03:55.39 | l1nuxman | lmao haah |
| 03:55.51 | l1nuxman | that's pretty steep |
| 03:56.59 | rogersja1 | gruvfunk, again matching refers to an explicit match or by pattern, so yes he is matching |
| 03:57.17 | gruvfunk | not from ATA to * he's not |
| 03:57.42 | gruvfunk | pots -> fxo ata -> asterisk -> fxs ata |
| 03:58.24 | gruvfunk | whatever extension name he configures in HT503 ATA should be the same name as the extension configured in asterisk in the right context |
| 03:58.37 | gruvfunk | I'm not arguing your point, avoiding the use of the 's' extension as a default |
| 03:58.54 | l1nuxman | gruvfunk, its like this |
| 03:59.22 | gruvfunk | call it harry, and it will ring the harry extension |
| 03:59.36 | l1nuxman | POTS->router->(ATA->FXS,Asterisk) |
| 04:00.07 | gruvfunk | not sure I follow that last comma , |
| 04:00.07 | l1nuxman | wait |
| 04:00.59 | l1nuxman | POTS->router,Asterisk->Router,FXS->ATA |
| 04:01.15 | gruvfunk | ? nah not that |
| 04:01.55 | l1nuxman | everything is centralized through the router |
| 04:02.20 | l1nuxman | it works :D |
| 04:02.33 | gruvfunk | the router is not handling any calls |
| 04:02.39 | l1nuxman | no |
| 04:02.49 | gruvfunk | only passing ethernet traffic (which carries the voip legs) |
| 04:02.57 | l1nuxman | connects the FXO to Asterisk |
| 04:03.01 | gruvfunk | so i'd remove it from your flow |
| 04:03.22 | l1nuxman | sorry bad diagram |
| 04:03.38 | l1nuxman | now I want to figure out how to do multiple users :) |
| 04:03.53 | l1nuxman | "Who do you want to leave this message for?" |
| 04:03.57 | gruvfunk | so your POTS comes in to your FXO port on the ATA, which talks to Asterisk - in parallel, you have the POTS line ringing both the native copper in the house and the FXS port on the ATA |
| 04:04.12 | l1nuxman | "PRess #1 for Joe,#2 for Mary" |
| 04:04.28 | gruvfunk | and then somehow you got Asterisk picking up voicemail if nobody answers |
| 04:04.47 | l1nuxman | yea |
| 04:04.50 | l1nuxman | ;) |
| 04:05.47 | l1nuxman | wait I wanna figure out to email these messages now :) !! |
| 04:08.13 | rogersja1 | l1nuxman, you can add your additonal voicemail boxes to voicemail.conf like you have with your [1] vm box |
| 04:08.57 | gruvfunk | and input the appropriate email addresses in there too |
| 04:09.03 | l1nuxman | I think I have to configure it in extensions.conf too though |
| 04:09.18 | l1nuxman | and how does the email get sent? |
| 04:09.22 | gruvfunk | magic |
| 04:09.23 | l1nuxman | using what smtp? |
| 04:09.24 | rogersja1 | then have asterisk answer the call before sending it to VM in your dialplan, and have the caller select from a menu which box |
| 04:09.43 | rogersja1 | you can use sendmail if you wish |
| 04:09.57 | gruvfunk | right, you'll need a simple IVR and add your recordings as desired |
| 04:10.30 | l1nuxman | cool stuff |
| 04:10.46 | l1nuxman | well what does asterisk use by default to send email? |
| 04:11.07 | gruvfunk | does it matter much? |
| 04:11.17 | l1nuxman | nope, but I'm curious how it works |
| 04:11.21 | gruvfunk | ah |
| 04:13.46 | l1nuxman | well I got no email :P |
| 04:14.13 | WiretapSeven | l1nuxman, it uses the system MTA |
| 04:14.43 | gruvfunk | you got a linux box, you likely have email |
| 04:15.59 | l1nuxman | lol gotta install mail |
| 04:21.48 | l1nuxman | hmm I wonder why it didn't work. I sent an email to my address but no cigar. I'm running vmware linux and behind a router. Does that affect my sendmail? |
| 04:22.00 | l1nuxman | maybe I'll go #postfix |
| 04:22.18 | gruvfunk | if you're using sendmail, see voicemail.conf for a needed edit |
| 04:22.47 | gruvfunk | ; You can override the default program to send e-mail if you wish, too ;mailcmd=/usr/sbin/sendmail -t |
| 04:23.52 | gruvfunk | also, yes, if you expect mail to flow to that virtual machine, you need to ensure the SMTP port is allowed through |
| 04:24.25 | gruvfunk | (you send you sent an email to your address, from where?) |
| 04:26.19 | gruvfunk | s/send/said/ |
| 04:26.46 | gruvfunk | and are you considering DNS too? |
| 04:28.59 | l1nuxman | oh actually. I think it doesn't work because my ISP blocks port 25 isn't that right? |
| 04:29.11 | l1nuxman | I have a domain and a host, maybe that could work somehow |
| 04:30.34 | l1nuxman | hmmm like have it connect to my host smtp and send from there |
| 04:32.39 | gruvfunk | time for bed, adios amigos |
| 04:36.24 | OldMonk | later |
| 04:56.47 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 04:58.43 | *** join/#asterisk engrxyz (~puitpyitr@host81-150-217-173.in-addr.btopenworld.com) |
| 04:59.47 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 05:01.55 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 05:01.59 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 05:05.02 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 05:09.49 | l1nuxman | anyone still here? Do you guys know if you're accessing voicemail outside calls will still come through? Or will it be a busy signal? |
| 05:21.42 | *** join/#asterisk Marvelous (~CE0@41.32.59.52) |
| 05:21.53 | Marvelous | hey every one |
| 05:22.24 | Marvelous | hello |
| 05:25.52 | *** join/#asterisk sourcode_ (~code@ppp-58-8-82-200.revip2.asianet.co.th) |
| 05:26.04 | Marvelous | hey all |
| 05:30.31 | WiretapSeven | Marvelous, if you have a question just ask it, don't wait for a regreeting |
| 05:31.57 | Marvelous | when i dial extension and he is not active peer it gives me service unavaliable i need it to route to voicemail |
| 05:32.32 | Marvelous | hen i dial extension and he is not active it gives me service unavaliable i need it to route to voicemail |
| 05:34.49 | Marvelous | i need context syntx to route if was busy or unreachable |
| 05:36.34 | Marvelous | any help |
| 05:37.02 | WiretapSeven | a bit beyond me I'm afraid |
| 05:37.08 | WiretapSeven | be patient, someone will be able to help |
| 05:37.15 | ChannelZ | Dial(Whatever) |
| 05:37.20 | ChannelZ | Voicemail(Whatever) |
| 05:38.28 | Marvelous | okie |
| 05:39.10 | ChannelZ | You can also put a timeout on the Dial so if he's there and just doesn't answer, it will eventually give up and go to voicemail |
| 05:39.19 | ChannelZ | Dial(SIP/Somedude,15) |
| 05:39.46 | ChannelZ | VoiceMail(100@default,u) |
| 05:41.38 | *** join/#asterisk darkdrgn2k (~darkdrgn2@70.51.26.226) |
| 05:41.53 | ChannelZ | l1nuxman: That totally depends on your channel config.. too vague of a question |
| 05:42.10 | darkdrgn2k | how can i re-compile a chan module to use with asteriskNOW |
| 05:42.47 | darkdrgn2k | i tried pulling the same version tarball and compiling it (as is) there.. but it doesnt want to seem to load it |
| 05:43.29 | ChannelZ | were they built on the same architecture? |
| 05:43.40 | ChannelZ | were they configured pretty similarly |
| 05:43.40 | darkdrgn2k | i686 |
| 05:43.56 | darkdrgn2k | i have no idea what asteriskNOW is configured as.. |
| 05:44.03 | darkdrgn2k | but it seems it was also build on i686 |
| 05:44.17 | darkdrgn2k | Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-24 20:43:18 UTC |
| 05:44.36 | gnutsy | darkdrgn2k: mix n match never plays well in Linux. Just compile and install your asterisk. YOu dont have the configure options the same there will be all kinds of strange issues |
| 05:44.50 | ChannelZ | indeed |
| 05:45.02 | darkdrgn2k | hmm |
| 05:45.09 | gnutsy | not worth the pain |
| 05:45.23 | darkdrgn2k | ugh i dont want to rebuild the whole dam thing.. |
| 05:45.29 | darkdrgn2k | just wanan hack one of the chans! |
| 05:45.58 | ChannelZ | If you're hacking it kind of suggests maybe you shouldn't be running a turnkey system |
| 05:46.28 | darkdrgn2k | granted. its not production.. |
| 05:46.34 | darkdrgn2k | well it is but its my home systems :) |
| 05:46.35 | *** join/#asterisk AlecTaylor (~AlecTaylo@unaffiliated/alectaylor) |
| 05:46.47 | darkdrgn2k | only thing i can see differnt when i ldd both modules |
| 05:46.54 | darkdrgn2k | (the stock one and the compiled stock one) is |
| 05:47.00 | darkdrgn2k | libpthread.so.0 => /lib/libpthread.so.0 (0x005a1000) |
| 05:47.00 | darkdrgn2k | vs |
| 05:47.03 | darkdrgn2k | <PROTECTED> |
| 05:47.10 | darkdrgn2k | oops |
| 05:47.24 | darkdrgn2k | i mennt libpthread.so.0 => /lib/libpthread.so.0 (0x00e4b000) |
| 05:47.39 | darkdrgn2k | asterisk now doesnt have a souce repo does it? |
| 05:48.03 | gnutsy | you'll spend more time chasing issues like that, than just ./configure -> make install |
| 05:48.14 | gnutsy | what distro |
| 05:48.17 | darkdrgn2k | but then all the paths are gonna be scewed |
| 05:48.19 | darkdrgn2k | asteriskNOW |
| 05:48.45 | gnutsy | ./configure --<your path requirements> |
| 05:49.14 | gnutsy | Isnt that centos 5? |
| 05:49.24 | darkdrgn2k | looks like it |
| 05:49.32 | darkdrgn2k | asterisk16-1.6.2.11-2_centos5 |
| 05:49.41 | darkdrgn2k | damit its 11-2 :-S argh |
| 05:50.02 | gnutsy | there are some rpms out there, but you may end up in the same boat your in now depending oin the builder. |
| 05:50.36 | darkdrgn2k | :-S what if i just rebuild the rpm from the srpm |
| 05:51.06 | gnutsy | same same, still need to configure it how you want it. |
| 05:51.21 | darkdrgn2k | <PROTECTED> |
| 05:51.42 | darkdrgn2k | baa so much work just to play with this dam unistim phone :-S |
| 05:52.01 | gnutsy | whats wrong with the channel your trying to hack? |
| 05:52.10 | darkdrgn2k | well wanna see what i can do to enhance it |
| 05:52.20 | gnutsy | what channel |
| 05:52.24 | darkdrgn2k | first thing i wanna do is shortten some of the text that gets sent to the phone cause i got a smaller display :-P |
| 05:52.26 | darkdrgn2k | chan_unistim |
| 05:52.38 | l1nuxman | what's 'n' do? |
| 05:52.57 | darkdrgn2k | l1nuxman: no? |
| 05:53.10 | darkdrgn2k | l1nuxman: ooo.. its like... next |
| 05:53.50 | l1nuxman | ah ok |
| 05:55.45 | gnutsy | if you have the time to hack the channel driver, what's the issue with with setting some configure options and building? |
| 05:56.04 | darkdrgn2k | gnutsy: dont want to break the prebuilt environment |
| 05:56.41 | darkdrgn2k | i was hopeing to just drop the module in.. and when im done playing remove it and put the original back |
| 05:57.45 | gnutsy | building the srpm is the best bet for that then |
| 05:58.06 | darkdrgn2k | now just to find the dam thing |
| 05:58.51 | gnutsy | sorry I dumped that OS after having too many issues just like that. |
| 06:03.00 | gnutsy | maybe ask here #centos |
| 06:04.13 | Marvelous | i can't find addmailbox |
| 06:04.20 | Marvelous | root@host [~]# locate addmailbox |
| 06:04.21 | Marvelous | root@host [~]# updatedb |
| 06:04.21 | Marvelous | root@host [~]# locate addmailbox |
| 06:04.35 | Marvelous | root@host [~]# find / -name addmailbox |
| 06:05.25 | ChannelZ | I don't even know what that is |
| 06:06.12 | Marvelous | i need to create mailbox |
| 06:06.45 | ChannelZ | what mailbox? email mail? |
| 06:06.54 | Marvelous | no |
| 06:06.54 | ChannelZ | there's nothing 'standard' called addmailbox |
| 06:07.01 | gnutsy | home depot has nice ones ;) |
| 06:07.13 | Marvelous | voice mailbox |
| 06:07.27 | Marvelous | voicemailbox |
| 06:07.49 | ChannelZ | /etc/asterisk/voicemail.conf |
| 06:09.09 | gnutsy | http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 06:09.25 | Marvelous | :) thanks |
| 06:09.38 | Marvelous | it's my first time to install asterisk |
| 06:09.58 | gnutsy | what distro? |
| 06:10.20 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:10.27 | Marvelous | centos |
| 06:10.43 | Marvelous | root@host [~]# cat /etc/redhat-release |
| 06:10.44 | Marvelous | CentOS release 5.6 (Final) |
| 06:10.44 | Marvelous | root@host [~]# |
| 06:10.49 | gnutsy | try this: http://www.freepbx.org/ |
| 06:11.00 | darkdrgn2k | or grab asterixNOW |
| 06:11.09 | gnutsy | same-same |
| 06:11.19 | Marvelous | it's webserver + voip server |
| 06:11.21 | darkdrgn2k | gnutsy: kinda :-P asteriskNOW is a quick 15 min install |
| 06:11.56 | gnutsy | sure,as long as you don't do anything custom ;) wink wink |
| 06:12.07 | darkdrgn2k | LOL |
| 06:12.44 | ChannelZ | well if you're doing FreePBX then you add the mailbox through that mess |
| 06:14.39 | gnutsy | messy yes, newb freindly ? more so than learning from scratch. Depends on the goal. Speed to deployment, or educational value |
| 06:15.23 | ChannelZ | I'm not going to have that dopey argument again |
| 06:15.29 | gnutsy | lol |
| 06:15.34 | gnutsy | me either |
| 06:16.18 | Marvelous | i love to learn the basics before learn wizard |
| 06:16.31 | darkdrgn2k | marlowe: freepbx is not a wizard |
| 06:16.49 | darkdrgn2k | marlowe: their dialplans will make your head spin |
| 06:16.51 | ChannelZ | No, it's a little more like a retarded court jester |
| 06:17.08 | darkdrgn2k | marlowe: but you can do almost anything right of the bat w/o coding it |
| 06:17.18 | gnutsy | CHannelZ - then your probably sad that Digium now likes freepbx as their default front end |
| 06:17.19 | darkdrgn2k | ChannelZ: lol |
| 06:17.57 | ChannelZ | no gnutsy I don't run any front end so no sweat off my back |
| 06:18.22 | ChannelZ | And I take that back, that was rude to actual retarded court jesters |
| 06:18.35 | Marvelous | okie guyz but i will learn config from scratch by yours help and google hand |
| 06:18.52 | darkdrgn2k | Marvelous: good luck :) |
| 06:18.53 | Marvelous | u give the keys and google give the rest |
| 06:19.03 | Marvelous | :) |
| 06:19.11 | darkdrgn2k | Marvelous: asterisk is like C++.. you cant do anything good the first few months of class |
| 06:19.30 | Marvelous | asterisk like Shell Scripting |
| 06:19.42 | darkdrgn2k | naaa |
| 06:19.47 | Marvelous | once i know how to config it i will write shell scripts to config it for me fast |
| 06:19.47 | ChannelZ | I set my business up on Asterisk, with no GUI, in about a week.. having never touched VoIP before (besides Skype which doesn't really count.) |
| 06:19.56 | ChannelZ | To say you can't do anything yourself is absurd |
| 06:19.57 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 06:20.06 | darkdrgn2k | ChannelZ: one phone doesnt count :-P |
| 06:20.08 | ChannelZ | It's a matter of motivation |
| 06:20.18 | darkdrgn2k | ChannelZ: but yes your are right. you can get the basics done quite quickly |
| 06:20.35 | ChannelZ | darkdrgn2k: I have 4 hard lines (POTS, so I had to deal with Zaptel at the time) and a dozen internal extensions.. |
| 06:20.35 | darkdrgn2k | ChannelZ: IF you understand a thing or two about phone systems LOL |
| 06:20.40 | darkdrgn2k | ChannelZ: like what a trunk is. |
| 06:21.00 | Marvelous | exten => 1000,1,Dial(SIP/1000,30) |
| 06:21.04 | Marvelous | exten => 1000,2,VoiceMail(1000@default) |
| 06:21.04 | Marvelous | exten => 1000,3,PlayBack,(vm-goodbye) |
| 06:21.04 | Marvelous | exten => 1000,4,HangUp() |
| 06:21.20 | darkdrgn2k | FLOOD! |
| 06:21.25 | Marvelous | no |
| 06:21.25 | Marvelous | :D |
| 06:21.31 | Marvelous | my lil config |
| 06:21.33 | ChannelZ | close.. use pastebin in the future |
| 06:21.38 | Marvelous | to have ur correction sir |
| 06:21.44 | Marvelous | okie |
| 06:21.46 | darkdrgn2k | yeh i think it was 1 line away from a flood ;) |
| 06:21.49 | ChannelZ | you have an extra , after Playback |
| 06:21.59 | ChannelZ | and it probably won't even get there anyway |
| 06:22.04 | gnutsy | ChannelZ: I don't use a front end personally either, but for customers to make small changes, they need one. |
| 06:22.08 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 06:22.21 | darkdrgn2k | gnutsy: small changes with freepbx? |
| 06:22.36 | darkdrgn2k | it rewrites the univese every time you make a "small change" |
| 06:22.43 | ChannelZ | god help them |
| 06:23.22 | Marvelous | <PROTECTED> |
| 06:23.23 | Marvelous | <PROTECTED> |
| 06:23.24 | Marvelous | :D |
| 06:23.42 | gnutsy | yes and the key was what you said "it rewrites" they dont have to. God help us all if the unlearned start modding extensions.conf |
| 06:23.47 | ChannelZ | As I said, you have an extra , after Playback |
| 06:24.27 | Marvelous | so what is the correction sir |
| 06:24.42 | ChannelZ | .....remove the extra , after Playback....... |
| 06:24.47 | Marvelous | okie |
| 06:25.08 | darkdrgn2k | 1000,3,PlayBack(vm-goodbye) <-spoon fed |
| 06:26.59 | Marvelous | [PBX]: New message 1 in mailbox 1000 |
| 06:27.04 | Marvelous | :D seems good |
| 06:27.17 | darkdrgn2k | one could even say its marvelous... |
| 06:27.23 | gnutsy | lol |
| 06:27.26 | gnutsy | nice |
| 06:27.59 | ChannelZ | unless you hate voicemails |
| 06:28.05 | Marvelous | okie guys am soo thank full for u all |
| 06:28.43 | darkdrgn2k | i would LOVE to pipe my dam maridian extension in the office into a asterisk box JUST so voicemails can get emailed to me as attachments |
| 06:29.22 | Marvelous | i need to understand how to make groups |
| 06:29.29 | darkdrgn2k | ringgroups? |
| 06:29.42 | Marvelous | like family,office |
| 06:29.54 | ChannelZ | that do what |
| 06:29.55 | Marvelous | only ppl in office able to call office extentions |
| 06:30.00 | darkdrgn2k | you identify the ppl in your family and stick a label to them |
| 06:30.05 | Marvelous | and family call family only |
| 06:30.07 | darkdrgn2k | haha have fun with that :-P |
| 06:30.16 | ChannelZ | well you can separate them all by context |
| 06:30.21 | Marvelous | :D STICK !! |
| 06:30.32 | darkdrgn2k | you have to essentially create a seperate dialplan for every "group" |
| 06:30.33 | Marvelous | better |
| 06:30.36 | gnutsy | read 1st : http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 06:30.37 | ChannelZ | and I guess only put whatever devices in whatever context |
| 06:31.06 | Marvelous | okie could i call family and office in same time |
| 06:31.07 | Marvelous | ? |
| 06:31.18 | darkdrgn2k | depnd is you configure that in the dialplan |
| 06:31.24 | ChannelZ | if you connect them but then why do they need to be separate in the first place? |
| 06:31.28 | darkdrgn2k | asterisk has no inherit "groups" |
| 06:31.45 | Marvelous | i need to be the only one to call family and office |
| 06:31.57 | Marvelous | super user :D |
| 06:32.01 | darkdrgn2k | Marvelous: create 2 sip servers and trunk them together... |
| 06:32.03 | gnutsy | outbound rules |
| 06:32.04 | darkdrgn2k | :-D |
| 06:32.08 | darkdrgn2k | thats what i did |
| 06:32.50 | ChannelZ | you make 3 contexts. family, office, and jesus |
| 06:33.11 | Marvelous | mmm |
| 06:33.12 | ChannelZ | family has whatever extensions dial other family, whatever that is.. office, the same |
| 06:33.13 | Marvelous | :D |
| 06:33.20 | ChannelZ | jesus includes both family and office |
| 06:33.24 | Marvelous | :D |
| 06:33.36 | Marvelous | i will use jesus as my context |
| 06:33.40 | ChannelZ | Your phone's default context is jesus. Others are family or office as appropriate |
| 06:33.43 | darkdrgn2k | remember to assign the contexts to the right extensions |
| 06:33.56 | Marvelous | :D sure |
| 06:34.05 | Marvelous | u know |
| 06:34.07 | darkdrgn2k | man... you the joker or something... |
| 06:34.08 | Marvelous | am soo happy |
| 06:34.11 | darkdrgn2k | smiling allot |
| 06:34.17 | Marvelous | just soo happy |
| 06:35.09 | gnutsy | sounds like the good ol FOSS smile hard at work |
| 06:36.05 | Marvelous | sure |
| 06:37.02 | darkdrgn2k | *sigh* doesnt look like it worked |
| 06:37.12 | darkdrgn2k | compiling the srpm and then copying the file over |
| 06:37.44 | gnutsy | compiling on Centos5 as well? |
| 06:37.48 | darkdrgn2k | yep |
| 06:37.48 | ChannelZ | You seemed to show earlier that some of the core system libraries linked against were of different versions |
| 06:37.53 | Marvelous | i have good way for centos |
| 06:38.05 | Marvelous | it's soo cool |
| 06:38.23 | darkdrgn2k | i might have to install anoter copy of asteriskNOW and see what i can do at that pont |
| 06:38.25 | Marvelous | install asterisk via yum with no errors fast like rocket |
| 06:38.26 | Marvelous | asterisk.org/downloads/yum |
| 06:38.43 | Marvelous | write the repos |
| 06:38.55 | Marvelous | yum install is soo cool |
| 06:39.04 | darkdrgn2k | rm -rf / is cooler |
| 06:39.06 | ChannelZ | so use the entire distribution that you built, not just the one module which is still mismatched |
| 06:39.18 | darkdrgn2k | yeh but i copeid onmly over the module.. |
| 06:39.25 | darkdrgn2k | didn tyr installing the whole rpm yet |
| 06:39.26 | Marvelous | cat /etc/issue > /dev/sda;reboot faster darkgrgn :P |
| 06:39.28 | darkdrgn2k | ill leave that for tomorrow |
| 06:40.06 | gnutsy | lib compatibility is one of the reasons I left that distro in the dust. |
| 06:40.31 | darkdrgn2k | Marvelous: dd if=/dev/zero of=`mount | grep \ /\ | awk '{print $1}'` |
| 06:40.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:40.52 | Marvelous | :D |
| 06:41.21 | Marvelous | wb ChannelZ |
| 06:41.30 | ChannelZ | Cable barf I guess |
| 06:41.59 | AlecTaylor | I'm building an Internet Radio site, with call-in functionality. The website will stream the audio, and to join the conversation they'll use an embedded webclient (which'll grab mic input). Can asterisk do the job for me? - http://lists.digium.com/pipermail/asterisk-users/2011-April/261986.html |
| 06:42.38 | darkdrgn2k | ok nite all |
| 06:42.43 | Marvelous | if 10.000 active call need good resources |
| 06:43.13 | gnutsy | nite dark |
| 06:44.50 | ChannelZ | later |
| 06:47.35 | Marvelous | nite darkdragon |
| 07:31.55 | *** join/#asterisk OldGrumpy (Whacko@p5B3135C5.dip.t-dialin.net) |
| 07:33.03 | *** join/#asterisk Marvelous (~CE0@197.195.117.235) |
| 07:36.18 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 07:39.48 | *** join/#asterisk imox1234 (~imox1234@p4FC5C250.dip0.t-ipconnect.de) |
| 07:41.24 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:41.56 | Marvelous | cable problems |
| 07:42.58 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:55.51 | *** join/#asterisk sphenxes (~sphenxes@85-127-213-25.dynamic.xdsl-line.inode.at) |
| 07:55.58 | Marvelous | i need tollfree account |
| 07:56.41 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 07:57.08 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:15.12 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 08:16.28 | atan | looks at Marvelous and wonders about the toll account |
| 08:17.22 | atan | `providers |
| 08:17.25 | atan | ~providers |
| 08:17.25 | infobot | hmm... providers is http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
| 08:17.29 | atan | hmm, what was it. |
| 08:17.31 | atan | !providers |
| 08:17.34 | atan | `sip |
| 08:17.39 | atan | ~sip providers |
| 08:17.48 | atan | Yeah, I would totally forget it wouldn't I. |
| 08:18.09 | atan | Marvelous, no need to pm :-) |
| 08:18.22 | atan | Marvelous, are you going for quality or for cheap? |
| 08:19.05 | Marvelous | free |
| 08:21.02 | tzafrir | ~itsp |
| 08:21.02 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 08:22.17 | Marvelous | ~itsp-us |
| 08:22.41 | Marvelous | itsplist-us |
| 08:22.50 | Marvelous | ~itsiplist-us |
| 08:23.02 | Marvelous | ~itsplist-us |
| 08:23.02 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 08:25.39 | atan | 'free toll-free' I believe voip.ms does this on their value route |
| 08:26.31 | Marvelous | ~help |
| 08:26.58 | *** join/#asterisk imox1234 (~imox1234@p4FC5C250.dip0.t-ipconnect.de) |
| 08:27.53 | Marvelous | ~rssfeeds |
| 08:28.30 | Marvelous | ~rssfeeds flush |
| 08:28.54 | Marvelous | ~rssfreeds update |
| 08:29.51 | Marvelous | ~uptime |
| 08:35.38 | *** join/#asterisk zxd (~zxd@95.211.21.34) |
| 08:35.46 | zxd | hi., how do I disable logging in asterisk? |
| 08:35.47 | *** join/#asterisk imox1234 (~imox1234@p4FC5C250.dip0.t-ipconnect.de) |
| 08:37.13 | zxd | comment out lines in ogger? |
| 08:37.15 | zxd | logger.conf |
| 08:38.06 | Marvelous | login or log |
| 08:38.07 | Marvelous | ? |
| 08:41.39 | Marvelous | asterisk.conf |
| 08:41.42 | Marvelous | <PROTECTED> |
| 08:41.42 | Marvelous | [options] |
| 08:41.42 | Marvelous | ;verbose = 3 |
| 08:41.42 | Marvelous | ;debug = 3 |
| 08:42.12 | Marvelous | ;console => notice,warning,error |
| 08:42.12 | Marvelous | ;console => notice,warning,error,debug |
| 08:42.12 | Marvelous | ;messages => notice,warning,error |
| 08:42.13 | Marvelous | ;full => notice,warning,error,debug,verbose |
| 08:42.26 | Marvelous | that all i know |
| 08:42.31 | Marvelous | make some search too |
| 08:42.44 | Marvelous | logger.conf |
| 08:42.51 | Marvelous | asterisk.conf |
| 08:42.55 | Marvelous | read the files |
| 08:43.34 | Marvelous | ~itsplist-us |
| 08:43.34 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 08:45.27 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:02.12 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 09:14.01 | *** join/#asterisk xPhilosx (~asdfa@64-139-76-219-Pima.hfc.comcastbusiness.net) |
| 09:14.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:20.08 | *** join/#asterisk AlecTaylor (AlecTaylor@unaffiliated/alectaylor) |
| 09:21.06 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 09:45.58 | AlecTaylor | Can asterisk 1) broadcast an audio conference stream 2) have a proxy in between the audio conference and guest caller (where I can screen the call) and 3) provide a web-interface to call in and listen in (without explicitly dialing) |
| 09:51.55 | ectospasm | not out of the box, but you could program it to do that. Asterisk can be the proxy, and you could have a webserver that does a one-to-many stream of the audio |
| 09:52.29 | ectospasm | basically, you have your audio source dial into the Asterisk MeetMe or ConfBridge conference bridge |
| 09:53.30 | ectospasm | ...and you could screen guests |
| 09:53.52 | ectospasm | ...and have the web service listen to the conference, itself muted |
| 09:54.19 | ectospasm | ...the web server then could broadcast to anyone who wants to listen to the conference via the web |
| 09:54.29 | ectospasm | exactly how to do that is outside of what I know how to do |
| 09:55.50 | AlecTaylor | Great to know it's possible though |
| 09:56.00 | AlecTaylor | Where should I ask the specifics of how to do this? |
| 10:04.21 | ectospasm | break it down into pieces |
| 10:04.30 | ectospasm | Start on the piece you perceive as being the hardest |
| 10:04.41 | ectospasm | ...and approach it that way |
| 10:05.14 | ectospasm | you may need to write extensive dialplan, AMI, or AGI code to get this running |
| 10:05.36 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 10:09.07 | AlecTaylor | Hmm, okay |
| 10:09.13 | AlecTaylor | What language is Asterisk written in? |
| 10:10.29 | ectospasm | download the source and see (-; |
| 10:10.52 | ectospasm | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 10:11.04 | ectospasm | now, that link may be dead, they've been down for a couple of days. |
| 10:11.22 | ectospasm | doesn't work for me |
| 10:12.50 | ectospasm | (this is due to the worst tornado disaster to strike the Southern US in nearly 80 years) |
| 10:13.04 | WiretapSeven | and digium only had one datacentre? |
| 10:13.47 | AlecTaylor | There was a tornado? |
| 10:13.50 | ectospasm | no, digium.com |
| 10:13.59 | ectospasm | There were several tornadoes in the area |
| 10:14.14 | ectospasm | ...digium.com is up |
| 10:14.33 | *** join/#asterisk knorkeknie (~hans@p5496E50C.dip.t-dialin.net) |
| 10:18.02 | *** join/#asterisk rportugal (~rportugal@217.129.164.233) |
| 10:24.14 | cusco_ | ola rportugal |
| 10:35.53 | knorkeknie | hey there, is ther a way to use sangoma A200/Remora FXO/FXS Analog AFT card as a timing device? |
| 10:39.11 | ectospasm | knorkeknie: if it uses DAHDI, maybe, but I'd be surprised if it did |
| 10:40.00 | knorkeknie | ectospasm, no it doesnt... so my choice will be ztdummy? |
| 10:40.39 | ectospasm | knorkeknie: no |
| 10:40.49 | ectospasm | the base dahdi driver provides timing |
| 10:41.22 | ectospasm | (it has the features of dahdi-dummy folded into it) |
| 10:41.33 | ectospasm | ...no need for the extra -dummy driver anymore |
| 10:42.02 | knorkeknie | ectospasm, ok thx im gonna try that |
| 10:42.45 | AlecTaylor | http://lists.digium.com/pipermail/asterisk-users/2011-May/261994.html |
| 10:43.00 | AlecTaylor | ^I've rewritten my problem case succinctly |
| 10:43.34 | ectospasm | then someone will answer, I'm sure. |
| 10:52.08 | AlecTaylor | Hopefully! |
| 11:02.52 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 11:34.33 | *** join/#asterisk engrxyz (~puitpyitr@host81-150-217-173.in-addr.btopenworld.com) |
| 11:37.20 | fauxalliance | AlecTaylor, as 'succinct' and to the point as that was.. certainly no substitute for google.. have you looked at http://web-meetme.sourceforge.net/ |
| 11:38.22 | fauxalliance | AlecTaylor, and the screen-shots look real pretty |
| 11:38.50 | *** join/#asterisk imox1234 (~imox1234@p4FC5C250.dip0.t-ipconnect.de) |
| 11:53.29 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 12:24.23 | Exten | whats the thing with x-lite DTMF and asterisk ? |
| 12:27.30 | fauxalliance | Exten, SIP? probably DTMF signalling |
| 12:28.38 | fauxalliance | in x-lite Go to menu -> Advanced System Settings -> DTMF Settings... jiggle the settings a bit.. some say force inband.. i prefer RFC2833 when properly configured end to end. |
| 12:29.08 | Exten | it should be RFC, my asterisk box is RFC |
| 12:29.17 | Exten | and also inband |
| 12:29.21 | Exten | in the xlite |
| 12:30.02 | fauxalliance | there are bugs to beware..."The Almighty X-Lite DTMF Problem" circa 2003 |
| 12:30.28 | Exten | lol ok |
| 12:30.31 | Exten | bye bye xlite |
| 12:31.05 | fauxalliance | Exten, good job! find a softphone that is RFC2833 COMPLIANT ;) |
| 12:31.18 | fauxalliance | or was that an Asterisk issue ;) |
| 12:32.42 | Exten | oh cmmon, x-lite became so heavy .. |
| 12:33.18 | fauxalliance | i use Twinkle for GNU/Linux... works well |
| 12:36.43 | Exten | im on a win box... |
| 12:36.43 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:38.29 | fauxalliance | Exten, Zoiper then |
| 12:42.08 | Exten | Me not like zopier |
| 12:42.14 | Exten | lol, zoiper |
| 12:42.28 | Exten | its a bit ugly. |
| 12:42.57 | fauxalliance | funny, i found x-lite f'ugly |
| 12:43.09 | Exten | i like the old x lite |
| 12:43.39 | russellb | have you tried Blink? that's what I have been using (at least on mac, but it supports windows and linux too) |
| 12:44.26 | fauxalliance | PJSUA works quite well (at a console) |
| 12:44.52 | russellb | blink is built on the same SIP stack |
| 12:45.04 | Exten | oh, Blink looks pretty |
| 12:45.07 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 12:45.12 | fauxalliance | takes a look |
| 12:45.50 | Exten | the old x lite was also bugly - the "spaceship look" was better then the "ajax animation- modern black" thingie |
| 12:45.58 | Exten | though |
| 12:49.21 | jaytee | yay! just got my Kindle edtion of Asterisk: The Definitive Guide |
| 12:50.08 | fauxalliance | perhaps I should get a copy for the coffee table |
| 12:50.40 | jaytee | I ordered the print edition too but it won't be here till tuesday |
| 12:51.19 | jaytee | I love the instant gratification of buying books for the Kindle. 1-Click and BAM! it's there. :-) |
| 12:54.15 | russellb | jaytee: yay! :-) |
| 12:55.48 | jaytee | russelb, it's weird how it took Amazon a week longer to offer TDG than it did the Cookbook. |
| 12:56.45 | jaytee | now I can get up to speed on all the new stuff in 1.8 |
| 12:58.01 | jaytee | just built another mini-itx Atom box to use for testing. Got 1 in production already and another ready to go to a customer to use as a front end for a Voiceguide/Dialogic HMP 3.0 system. |
| 12:59.11 | jaytee | the new box I just built is the first one I've used a solid state drive on. it's a quick little bugger and the total for the parts including case was $235.00 |
| 13:00.23 | *** join/#asterisk Corydon76-home (red@c-69-137-80-31.hsd1.tn.comcast.net) |
| 13:00.23 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:01.02 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 13:05.22 | Exten | lovley. my asterisk box exists for 5 days and it has a 13mb file of trying to register extensions/ hacking tries |
| 13:06.56 | russellb | Exten: install fail2ban |
| 13:07.03 | jaytee | yeah |
| 13:07.18 | jaytee | probably most of those attempts are from China |
| 13:07.25 | russellb | how to install it is discussed in the security chapter of Asterisk: The Definitive Guide. You can read it for free here: |
| 13:07.28 | russellb | ~book |
| 13:07.28 | infobot | For more information about the Asterisk book, see ~thebook |
| 13:07.32 | russellb | ~thebook |
| 13:07.32 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:07.51 | jaytee | the rest are from the Russian Federation and a few from here in the US |
| 13:08.25 | jaytee | I block most pacific rim subnets in iptables permanently and let Fail2Ban handle the rest |
| 13:09.14 | jaytee | and I use strong passwords to prevent dictionary attacks |
| 13:12.00 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 13:46.13 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
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| 13:54.44 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:03.56 | knorkeknie | ectospasm, dahdi works... thx for help |
| 14:08.31 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 14:12.34 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 14:14.54 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 14:33.18 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 14:43.06 | *** join/#asterisk hairyraven (~nobody@95.72.33.212) |
| 14:45.51 | *** join/#asterisk tamiel (~tamiel@ip-32.net-81-220-18.versailles.rev.numericable.fr) |
| 15:11.13 | *** join/#asterisk sol0 (75e42e2a@gateway/web/freenode/ip.117.228.46.42) |
| 15:12.39 | sol0 | hello, I'm going to use asterisk for VOIP plug-in in an user application named Kontact for desktop environment for linux kernels named kde.. |
| 15:13.11 | sol0 | and I want someone who can mentor me fr use of asterisk during this open-source project. |
| 15:21.30 | *** join/#asterisk Kylix (~trek@ppp-165-71.27-151.libero.it) |
| 15:21.54 | Kylix | I'm havin a problem with asterisk 1.8.3.2-2 on openWRT |
| 15:22.42 | Kylix | I use asterisk and chan_datacard; when calling the chan_datacard device, everything is OK. Just when hanging up, asterisk gets killed |
| 15:22.55 | Kylix | I have to manually start asterisk ... |
| 15:23.07 | *** join/#asterisk wonderworld (~ww@port-92-201-77-40.dynamic.qsc.de) |
| 15:23.39 | Kylix | the device I'm using is an ADSL modem with OpenWRT, 300 Mhz processor speed, 32 MB RAM |
| 15:24.29 | Kylix | practically once I've installed openwrt I've lost the modem part as there are no openwrt drivers |
| 15:25.42 | l1nuxman | how to customize what you want asterisk to say in your IVR ? |
| 15:26.03 | l1nuxman | like specific words |
| 15:26.17 | rogersja | l1nuxman: you can use a text to speech processor |
| 15:26.49 | l1nuxman | otherwise, I'd have to record my own voice in which way? |
| 15:28.03 | rogersja | asterisk plays nicely with festival, vestec, cepstral, and lumenvox |
| 15:28.29 | rogersja | or yes you can record your own prompts, or have them done professionally for a nominal fee |
| 15:28.53 | rogersja | you can get have your custom prompts done in the official voice of Asterisk, Allison Smith |
| 15:34.33 | l1nuxman | oh cool |
| 15:35.13 | l1nuxman | lol it says festivals voice is rough |
| 15:35.18 | l1nuxman | is there one that isn't |
| 15:35.34 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 15:38.33 | rogersja | I have only used cepstral, these are licenced 'add-ons' so you'll need to purchase a licence to use. |
| 15:39.30 | rogersja | you can sample festival voices on their webpage, they are getting more realistic, but still not quite as real sounding as the others |
| 15:40.24 | rogersja | if you don't need some form of dynamic speech, you can always record your own prompts |
| 15:41.20 | AlecTaylor | How can I setup asterisk to join and listen to conference calls through a web-interface? - http://lists.digium.com/pipermail/asterisk-users/2011-May/261994.html |
| 15:41.55 | rogersja | l1nuxman, you can have allison smith record your prompts if you want, the rate last time i checked was $12USD per 15 words |
| 15:42.31 | rogersja | l1nuxman: http://store.digium.com/productview.php?product_code=IVRPROMPT |
| 15:43.07 | *** join/#asterisk darkdrgn2k (~darkdrgn2@70.51.26.226) |
| 15:43.11 | darkdrgn2k | hey |
| 15:43.18 | darkdrgn2k | im trying to compile the asterisk16 SRPMS |
| 15:43.23 | darkdrgn2k | but i keep getting File not found: /var/tmp/asterisk16-1.6.2.17.3-root/usr/lib/asterisk/modules/codec_speex.so |
| 15:43.27 | darkdrgn2k | about 6 of them |
| 15:43.31 | darkdrgn2k | doesnt look like there are any other errors |
| 15:45.44 | rogersja | AlecTaylor: are you using asterisk to host your conference? |
| 15:47.01 | rogersja | eitherway, there is no native ability for asterisk to play live audio over the web. as far as I am aware of. |
| 15:50.07 | *** join/#asterisk Marvelous (~CE0@41.153.56.13) |
| 15:50.13 | rogersja | AlecTaylor: you may be able to use the ICES() application in asterisk to stream to an icecast server |
| 15:50.59 | rogersja | from the icecast server to the web is up to you, you would need to configure the icecast, and build your webinterface. all of this is outside of asterisk |
| 15:51.21 | AlecTaylor | Darn |
| 15:51.37 | AlecTaylor | I thought there'd be an official Asterisk project for something similar |
| 15:51.58 | AlecTaylor | If you can think of FOSS projects which can do something akin to what I'm trying to do, please reply on the mailing-list |
| 15:52.41 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 15:52.48 | rogersja | well there is the ices application, so there is a great chance someone is using it, do some google searches for asterisk and icecast |
| 15:54.40 | rogersja | darkdrgn2k, why not simply checkout asterisk from svn, and compile? |
| 15:55.25 | *** join/#asterisk guilhermebr (~Guilherme@187.113.38.198) |
| 15:56.49 | darkdrgn2k | loader.c: Module 'chan_unistim.so' was not compiled with the same compile-time options as this version of Asterisk. <- any one know how to get around this error |
| 15:59.02 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 16:00.53 | rogersja | darkdrgn2k: follow these instructions http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html |
| 16:01.11 | rogersja | from ~thebook and you'll likely not have issues |
| 16:01.21 | rogersja | ~thebook |
| 16:01.21 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 16:03.06 | darkdrgn2k | rogersja: i wanna just throw in a single module into it not rebuild the whole ting |
| 16:04.07 | rogersja | use make menuselect to add your new module, then make make install |
| 16:04.25 | darkdrgn2k | rogersja: i really dont want to replace the WHOLE asterisk installation thought :-S |
| 16:04.54 | rogersja | if you mean you dont want to replace your config files, just dont do make samples |
| 16:05.15 | rogersja | this will leave your /etc/asterisk directory alone and not replace your .conf files |
| 16:05.46 | darkdrgn2k | yeh but if im hacking a module. i dont want to be reinstalling asterisk every time i make a change |
| 16:06.17 | darkdrgn2k | "was not compiled with the same compile-time options as this version of Asterisk." ??? |
| 16:06.19 | darkdrgn2k | oops |
| 16:06.22 | darkdrgn2k | "replace the hash with the one that is in the modules that came with the rpm." |
| 16:06.23 | OldGrumpy | well, as the message says, you need to match compile-time options |
| 16:06.33 | OldGrumpy | heh |
| 16:06.34 | OldGrumpy | :) |
| 16:06.46 | darkdrgn2k | hmm |
| 16:06.58 | darkdrgn2k | but i used the SRPMS of the rpm to compile |
| 16:07.02 | darkdrgn2k | how could they be differnt |
| 16:10.51 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 16:15.51 | OldGrumpy | same compile flags? |
| 16:20.46 | *** join/#asterisk sol0 (75e46909@gateway/web/freenode/ip.117.228.105.9) |
| 16:21.45 | l1nuxman | anyone know how to change default voice in festival? |
| 16:25.08 | l1nuxman | wow the default voice festival uses rogersja is hooorrrrible |
| 16:26.22 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 16:28.49 | *** join/#asterisk knorkeknie (~hans@p5496FAF7.dip.t-dialin.net) |
| 16:29.44 | knorkeknie | hi there, im trying to use odbc to connect to a mysql database... console shows error: load_odbc_config: Limit should be a number, not a boolean: '0'. Disabling ODBC class 'asterisk' |
| 16:30.02 | knorkeknie | in res_odbc.conf i have limit => 0 |
| 16:30.18 | knorkeknie | some hints whats wrong? |
| 16:31.56 | rogersja | knorkeknie: perhaps if you comment that line, it might connect :P |
| 16:33.35 | knorkeknie | rogersja, ok can try that, so this setting is useless? just trying to "copy" the example from asterisk: the definitive guide |
| 16:34.33 | knorkeknie | ... commenting it leads to connecting to database |
| 16:34.35 | knorkeknie | <PROTECTED> |
| 16:34.45 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 16:34.50 | rogersja | you might find that limit and pooling are helpfull with MS SQL or Sybase |
| 16:35.36 | knorkeknie | ah ok, think this setting is to limit query-results, isnt it? |
| 16:36.31 | rogersja | it limits the connection to one execution at a time |
| 16:37.11 | knorkeknie | oh.. ok |
| 16:37.22 | knorkeknie | so im fine without limit ;) |
| 16:42.30 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 17:02.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:19.20 | Kylix | I'd need a little help with asterisk 1.8 on openWRT |
| 17:19.25 | Kylix | mips platform |
| 17:32.57 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 17:38.14 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 17:41.09 | knorkeknie | reading asterisk: the definitive guide... why ther are in sip.conf the phones defined in hexcode (eg [0000FFFF0001]) ? |
| 17:49.28 | ChannelZ | I think it's just a means to show you that device names are NOT "extensions" and can be anything |
| 17:49.48 | ChannelZ | They can just as well be [Bob] and [Fred] |
| 17:51.23 | knorkeknie | ChannelZ kk understand that |
| 17:56.31 | Kobaz | the problem is many people do [1000] and secret=1000 |
| 17:56.36 | Kobaz | or something similar |
| 17:57.03 | Kobaz | which is nice and easy to set up, but then there are the evildoers who scan for easy to guess sip accounts |
| 17:57.13 | Kobaz | and use them to place expensive international calls |
| 17:58.12 | Kobaz | so if you use stuff like the mac address for the sip account, then it's much much harder to guess (security by obscurity) |
| 17:58.43 | Kobaz | and then the next thing to fix is don't use passwords that are easy to guess... use a random string or something suficiently long |
| 17:59.26 | Kobaz | dd if=/dev/urandom bs=1024 count=1 | md5sum |
| 17:59.33 | Kobaz | that's a realy good way to generate a password |
| 18:00.02 | OldGrumpy | not that really but sufficient for you ;) |
| 18:00.31 | *** join/#asterisk xchg (~xchg@94.229.33.133) |
| 18:01.24 | knorkeknie | kobaz, yea ok passwords is clear, but one handy thing is in patternmatching, if i have a sip-phone 101 i can do a _ZXX => Dial(SIP/${EXTEN}... so how to use this with mac-adresses... mappings in a database ? or how do you do that? |
| 18:01.42 | xchg | Hi. Maybe offtopic, but do I need voip gateway to use IP phones, or it's enough to make them able to reach each other by IP protocol? |
| 18:09.43 | l1nuxman | when a message goes into voicemail does the caller ID get saved somewhere too? |
| 18:09.55 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:13.30 | rogersja | knorkeknie: simplest method might be to use globals for your devices in extensions.conf |
| 18:13.42 | rogersja | if you dont have that many that is :P |
| 18:14.01 | rogersja | if you plan on having hundreds of devices, you might want to look into ARA |
| 18:15.33 | rogersja | xchg, its a bit off topic unless one of your devices is asterisk ;) |
| 18:16.01 | rogersja | l1nuxman, it certainly can be, if you are using ARA |
| 18:16.54 | knorkeknie | rogersja, planing to have about 700 devices .... so ara is asterisk realtime architecture i guess? |
| 18:17.24 | rogersja | knorkeknie: correct, using ARA would make management and maintenance much easier |
| 18:17.35 | xchg | rogersja: eeh, yes it is :D |
| 18:17.38 | Kobaz | knorkeknie: you would need a mapping |
| 18:18.36 | knorkeknie | rogersja, yea ok, i started this today |
| 18:18.52 | rogersja | l1nuxman: if you are using filesystem storage, take a look in your voicemail directory, you should find a .txt file that contains the envelope data to go along with the recording |
| 18:20.24 | knorkeknie | ok thx so long, wife is calling for dinner ;) |
| 18:20.35 | rogersja | knorkeknie, if you dont want to use ARA, you could employ func_odbc, and querie a simple table in a database. but this would cause you to have to change things in 2 places, anytime an extension or device was modified |
| 18:22.22 | *** join/#asterisk WannaBeGeekster (~fholmes@adsl-99-97-195-110.dsl.hstntx.sbcglobal.net) |
| 18:56.56 | *** part/#asterisk neurosys (~neurosys@c-65-34-188-197.hsd1.fl.comcast.net) |
| 18:57.48 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
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| 19:03.07 | Jcook_5xData | Is there a service I can use that I can use to check if I am receiving caller ID |
| 19:05.36 | Freeaqingme | check if the caller id string is empty? |
| 19:06.17 | l1nuxman | I"m having that problem now :P |
| 19:07.47 | l1nuxman | my caller id is callerid="FXOPort" <ht503fxo> |
| 19:08.00 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 19:09.31 | Jcook_5xData | Freeaqingme, yes. ATT says they are sending it, but all I have is a cell phone and it works but one time is hardly a test |
| 19:10.40 | Jcook_5xData | I will give the number if some people want to call it 1-616-245-3388 |
| 19:10.49 | Freeaqingme | l1nuxman, what kind of fxo card do you use? |
| 19:11.05 | l1nuxman | Freeaqingme, HT503 Grandstream |
| 19:11.14 | Freeaqingme | I can call, but can you then not answer it Jcook_5xData ? ;) |
| 19:11.53 | Jcook_5xData | yes It goes to a closed message then hangs up it Company phone system |
| 19:12.02 | Jcook_5xData | we are closed today |
| 19:12.11 | Freeaqingme | well, then it does get answered, but okay |
| 19:12.12 | Freeaqingme | hold on |
| 19:13.06 | Jcook_5xData | no name :( |
| 19:13.08 | Freeaqingme | Jcook_5xData, just called |
| 19:13.17 | Freeaqingme | oh, that makes sense, we dont do name stuff in the netherlands |
| 19:13.30 | Freeaqingme | but you should have a number |
| 19:14.16 | Jcook_5xData | Michigan USA - yes I received the number but boss want name a number -- hhhmm |
| 19:14.54 | Freeaqingme | Jcook_5xData, some countries just dont use named clid |
| 19:15.17 | Jcook_5xData | you in USA |
| 19:15.23 | Jcook_5xData | I am |
| 19:15.26 | l1nuxman | Freeaqingme, I though that maybe adding this line would fix my callerID problem? |
| 19:15.29 | l1nuxman | <PROTECTED> |
| 19:17.07 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 19:17.13 | cusco_ | I'm looking for small machine like dockstar to run some linux in it cheap.. any sugestions?:p |
| 19:20.15 | rogersja | cusco_: how does that relate to asterisk? |
| 19:20.59 | cusco_ | sorry .. Im looking to get a small device to put asterisk in it |
| 19:21.32 | rogersja | try a linksys wrt |
| 19:21.54 | rogersja | virtually any device you can run linux on, you can run asterisk on |
| 19:23.13 | cusco_ | yea im looking for a small pretty one.. with preference with mips |
| 19:30.58 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 20:01.12 | *** join/#asterisk m4xx (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 20:04.06 | m4xx | I'm trying to find some information about auto provisioning my cisco sap 504g ip phone. From what little i've been able to find you can do with with tftp and dhcp although i've yet to find anything about the dhcp part of the configuration. Can someone point me in the right direction? Or does anyone know if it's covered in the book? I didn't see anything in the table of contents. |
| 20:04.40 | Jcook_5xData | m4xx, dhcpd and option66 |
| 20:06.31 | m4xx | thanks =] |
| 20:08.06 | Freeaqingme | Jcook_5xData, I'm about to look at the same for grandstreams. Got any keywords in advance for me? ;) |
| 20:11.06 | Jcook_5xData | Freeaqingme, mostly a Polycom man but if ftp and option 66 work with it here quick how to: http://thevoipcentre.co.uk/wordpress/2010/11/dhcp-option-66-with-asterisk/ |
| 20:11.14 | Jcook_5xData | m4xx, http://thevoipcentre.co.uk/wordpress/2010/11/dhcp-option-66-with-asterisk/ |
| 20:11.27 | Freeaqingme | Jcook_5xData, cool, tnx ;) |
| 20:11.57 | Jcook_5xData | Freeaqingme, not sure if this help http://www.voip-info.org/wiki/view/Grandstream+GXP2000+Firmware+Archives |
| 20:12.37 | Jcook_5xData | Voip-info.org is great if you never been there check it out |
| 20:12.47 | Freeaqingme | yeah, got it open all day long |
| 20:12.49 | Freeaqingme | for weeks |
| 20:13.02 | Jcook_5xData | lol been there :) |
| 20:13.07 | Freeaqingme | thought its a bit outdated here and there |
| 20:14.45 | Jcook_5xData | yea, I guess the appliances have taken over the hackers.. |
| 20:15.22 | Freeaqingme | or the hackers got more experienced and didnt feel the need to maintain it |
| 20:15.43 | Freeaqingme | I've been on forums where the general level got higher and higher, until none of the frequent visitors cared to check it out |
| 20:16.08 | *** join/#asterisk m4xx (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 20:19.24 | Jcook_5xData | cool what the site |
| 20:19.48 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 20:20.04 | Freeaqingme | Jcook_5xData, some dutch forums |
| 20:20.17 | Freeaqingme | it's funny, now a younger generation is taking over, and level is rising again |
| 20:20.21 | Freeaqingme | history repeats itself |
| 20:21.39 | Jcook_5xData | will I guess it is as good it will get. till tomorrow. will off home. good luck |
| 20:21.51 | Freeaqingme | cool |
| 20:21.51 | Freeaqingme | u2 |
| 20:21.53 | Jcook_5xData | thanks for your help |
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| 22:47.40 | [hC] | anyone here have experience trying to update a polycom (ip430 in this case) from bootrom 3.1.3 to anything newer? In my case it seems to pull the bootrom, says downloading, then updating, then like 2 seconds into updating it reboots and cycles through the same process. (never updating) |
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