| 00:16.54 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 00:17.20 | roxdragon | hi |
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| 01:56.59 | p3nguin_ | Anyone here using chan_sccp-b that is familiar with the conf? |
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| 04:44.16 | b0gatyr_ | hello, not that I would like to use asteriskGUI but I'm wondering if you make a change using the config files would it reflect on the gui as well or is only one way? |
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| 06:14.13 | rethus | how can i check on asterisk, if a agi-connection is established? |
| 06:17.09 | ferdna | guys i can get my system to work |
| 06:17.11 | ferdna | i need help |
| 06:17.29 | ferdna | i had to remove the echo cancelation card, for it to work |
| 06:17.36 | ferdna | i can make outgoing calls but i cannot receive |
| 06:19.19 | WIMPy | ~ask |
| 06:19.19 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 06:23.10 | ferdna | WIMPy, heck... dont really know what to ask... |
| 06:23.16 | ferdna | problem is that i cannot receive calls |
| 06:23.33 | *** join/#asterisk _zoom_ (~zoom@41.218.33.112) |
| 06:23.37 | WIMPy | That is rather vague. |
| 06:23.41 | ferdna | i know |
| 06:23.55 | ferdna | but how can i ask a specific question when i dont know what is going on |
| 06:24.10 | ferdna | also, the asterisk gui manager doesnt work properly... |
| 06:24.26 | WIMPy | What are you connected to and how? What does the CLI show when a call should be received? What error does the caller get? |
| 06:24.44 | _zoom_ | hi, guys am looking for sip client (ubuntu) support both tls and srtp, any suggestion? |
| 06:25.03 | WIMPy | If you're using freepbz, you should ask in #freepbx. |
| 06:25.27 | ferdna | WIMPy, i have an pci-e 8 line card, with 4 FXO ports |
| 06:25.42 | rethus | how can i check on asterisk, if a agi-connection is established? |
| 06:26.05 | WIMPy | NFI about FXOs. |
| 06:26.24 | ferdna | WIMPy, i have 1 pots connected to port 1 in my card... |
| 06:26.29 | ferdna | NFI, means what? |
| 06:26.39 | p3nguin_ | ~nfi |
| 06:26.39 | infobot | hmm... nfi is No Fucking Idea |
| 06:26.46 | ferdna | oh got it |
| 06:26.56 | WIMPy | is not a historian. |
| 06:27.44 | ectospasm | ferdna: not being able to receive inbound calls but outbound calls work point to dialplan as the issue. |
| 06:29.00 | ferdna | ectospasm, awesome... thanks... let me look into that... i will create a simple dialplan |
| 06:32.18 | ectospasm | essentially, you need to tell Asterisk what to do with an inbound call, using dialplan |
| 06:32.18 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 06:33.37 | ferdna | ectospasm, i will remove the gui and just install asterisk |
| 06:34.41 | ectospasm | ferdna: that's my option, but I'm experienced at this. I remember it taking me two full weeks of work before I was able to get a call into the system. |
| 06:34.47 | ectospasm | That was during the pre-1.0 days |
| 06:34.58 | ectospasm | ~thebook |
| 06:34.58 | infobot | it has been said that thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 06:35.12 | ectospasm | ferdna: ^^^ that book will be your friend |
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| 07:30.41 | ferdna | ectospasm, thanks for the link |
| 07:30.45 | ferdna | :) |
| 07:30.51 | ectospasm | no problem |
| 07:30.52 | ferdna | that is the same book that i have |
| 07:31.09 | ectospasm | yeah, it's quite old, but it still applies for the most part. |
| 07:31.23 | ferdna | i see |
| 07:31.38 | ferdna | so, purchasing a new book... the 1.6 books will it make an improvement? |
| 07:31.42 | ectospasm | it's a really good introduction to Asterisk, all phases |
| 07:32.14 | ectospasm | I don't know of any comprehensive books on 1.6.x, there probably won't be one |
| 07:32.58 | ectospasm | If you don't need to go into production soon, I suggest you try 1.8, that's the next LTS release, and it's already in the release candidate state. |
| 07:33.19 | ectospasm | They're probably waiting to make the full release at Astricon, but I don't know for certain. |
| 07:33.50 | ferdna | awesome |
| 07:33.53 | ferdna | thanks for the advice |
| 07:34.36 | ectospasm | There probably won't be another O'Reilly book until 1.8 is released. |
| 07:41.07 | ferdna | cool |
| 07:41.24 | *** join/#asterisk BANSAL (~bansal@117.199.116.254) |
| 07:54.26 | ferdna | gn |
| 07:54.33 | ferdna | thnx 4 da help |
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| 08:26.58 | dandate2 | will a call center style amplifier work with a cheap ATA adapteR? |
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| 12:14.10 | DND | guys how can i check the users connected on iax2 and the coded they are using? |
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| 12:30.09 | rethus | i search for a way to get error-messages back to php if i use phpagi. Has somebody a idea how i can do this? |
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| 12:34.49 | Sabrina22 | hi |
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| 12:48.27 | rethus | i have installed the module konference. |
| 12:48.49 | rethus | konference list mychannel show the users into this channel. |
| 12:49.11 | rethus | but in column Users are only numbers (1,2,3 and so on. |
| 12:49.17 | rethus | can i set this to another value? |
| 13:01.36 | *** join/#asterisk russo (~russo@etc.nedos.net) |
| 13:03.36 | rethus | can i set caler-id for an incomming sip-call? |
| 13:03.54 | russo | hey all, i got a quick question. I have a small asterisk server which is just running some sip trunks for me to dial via different sip providers. I have several mobile phones which (and computers) which i would like to use with my asterisk server. The problem/question i have is, how can i make it so that an incoming call rings on all interfaces (i.e. ringall) and then another device can join the call after its ongoing, i.e. like when you have multiple PTSN ph |
| 13:04.27 | russo | i was looking into conferences etc and call files, but I'm just not sure what the simplest way of doing this is |
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| 13:05.47 | russo | oops, sorry for the drop :) |
| 13:05.55 | russo | did i miss something? |
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| 13:12.16 | iscario | hi, i would like to know how i could originate a call from an asterisk server to any iax client at the time i decide it (my question is in fact : which channel am i suppose to use?). I guess i could create a kind of virtual iax account (whose client would be on the server itself) but it seems to be a bit heavy , and their is probably a better way to do this ? |
| 13:49.32 | *** join/#asterisk gain_ (~gain@host33-194-dynamic.9-87-r.retail.telecomitalia.it) |
| 13:56.39 | tzafrir | iscario, run something at the time you decide: either through cron or through atd |
| 13:57.21 | tzafrir | originate a call through a command: asterisk -rx 'originate <...>' |
| 13:57.36 | iscario | yes, i know that tzafrir, the problem is not to launch any application, my problem is to have a channel available to do that |
| 13:58.04 | iscario | know what i mean ? |
| 13:58.22 | tzafrir | originate IAX/123 extension 1234567@trunk-context |
| 13:59.31 | iscario | yes, it suppose that IAX/123 has a softphone on, right ? |
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| 14:27.07 | imox1234 | hello, i have a problem with my 1und1 sip accounts. i dont get any call from the accounts. when the call is incommong on asterisk i get this error |
| 14:27.08 | imox1234 | [Sep 26 16:19:08] NOTICE[4277]: chan_sip.c:19987 handle_request_invite: Sending fake auth rejection for device "+4915253537516" <sip:+4915253537516@arcor.de;user=phone>;tag=SDv0cn501-0029ba71 |
| 14:27.56 | imox1234 | i have nothing changed on my conf or asterisk version. i think i have this problem till 2 weeks or so |
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| 15:16.42 | dandate2 | will a call center style amplifier work with a cheap ATA adapteR? |
| 15:20.51 | dandate2 | specifically the Cisco SPA8000 8-Port Telephony Gateway SPA8000-G1 with plantronics s12 amps |
| 15:23.32 | p3nguin | Does the S12 plug into an FXS jack? |
| 15:24.01 | p3nguin | Replacing a standard telephone? |
| 15:24.55 | dandate2 | i would have to have a standard telephone in between |
| 15:25.17 | dandate2 | so FXS connects to standard telephone which connects to plantronics S12 amp + headset |
| 15:25.45 | dandate2 | i tried this once using a linksys pap2-na and a very old telephone and the sound was very distorted |
| 15:25.53 | nemith | dandate2: should work fine |
| 15:26.31 | nemith | you may need to modify some settings, turn off vad, turn off echo suppression (or play with it) |
| 15:26.37 | nemith | i am not really familar with the SPA |
| 15:26.42 | dandate2 | k |
| 15:26.45 | nemith | i've done it for VIC-FXS cards |
| 15:30.38 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 15:38.03 | drmessano | Commodore makes FXS cards? |
| 15:40.54 | *** join/#asterisk Howie69 (~Howie69@69.60.103.136) |
| 15:41.00 | Howie69 | Good morning |
| 15:41.50 | Howie69 | I have a rather annoying problem that pops up every once in a while (like this morning) that I have had in 5 different versions of asterisk |
| 15:41.58 | Howie69 | Was hoping someone here could help |
| 15:42.09 | Howie69 | I have an extension forwarded, but refuses to unforward |
| 15:42.19 | Howie69 | I even destroyed the extension and recreated it |
| 15:42.32 | p3nguin | What does that even mean? |
| 15:43.03 | WIMPy | Sounds like gooey. |
| 15:43.32 | Howie69 | WIMPy: you talking to me or p3nguin ? |
| 15:44.01 | WIMPy | Yes |
| 15:44.03 | p3nguin | That's a foreign language. |
| 15:45.00 | drmessano | Howie69, is this a FreePBX install? |
| 15:45.14 | Howie69 | drmessano: yes, but been running fine for years |
| 15:45.16 | Howie69 | about 3 years |
| 15:45.19 | drmessano | ~freepbx |
| 15:45.19 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:45.41 | Howie69 | And all of the problems are from phone commands, not gui |
| 15:45.52 | Howie69 | *72 to forward, *73 to unforward |
| 15:45.58 | WIMPy | What are 'phone commands'? |
| 15:46.14 | Howie69 | *73 in the asterisk console shows it claiming to unforward |
| 15:46.17 | p3nguin | That has nothing to do with "phone commands," I'm quite sure. |
| 15:46.21 | drmessano | No, I know what your problem is, and you're asking in the wrong place |
| 15:46.22 | WIMPy | And whay are they non-standard? |
| 15:46.25 | drmessano | ^^^ |
| 15:46.37 | Howie69 | drmessano: ok, I'll try them |
| 15:46.39 | drmessano | If you would like it fixed, go to the correct channel |
| 15:49.55 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
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| 16:14.52 | Flyser__ | Hi, I am going to replace my PCI Fritzcard (fcpci) ISDN card with a newer one, as the driver is discontinued and not supported on newer kernel versions. The only purpose of this card will be to act as a SIP/ISDN gateway at home. Are the HFC-S cards still recommended for this purpose or are there better choices? |
| 16:15.26 | *** join/#asterisk Daejeo (~chatzilla@117.203.129.112) |
| 16:18.27 | WIMPy | Flyser__: You need either misdn1/chan_misdn or misdn2/LCR/chan_lcr. The later has worked best for me so far. |
| 16:21.10 | Flyser__ | chan_lcr? for the fritzcard or for the HFC-S cards? |
| 16:21.23 | WIMPy | For HFC cards. |
| 16:23.23 | Flyser__ | so you would recommend a HFC card? |
| 16:24.27 | WIMPy | There isn't much choice I think. Either you go for Digium or for some HFC variant. |
| 16:26.01 | Daejeo | wimpy ice cream? |
| 16:26.44 | Daejeo | ah.. wimpy ice cream from Sweden |
| 16:26.59 | Daejeo | :) |
| 16:27.20 | WIMPy | Ah. Nice. Any swedish girls around then? ;-) |
| 16:28.32 | WIMPy | I like the HFC based stuff as it's not limited to Asterisk. Tho strictly speaking they can't be used with Asterisk any more. Only via LCR. |
| 16:29.08 | WIMPy | The wcb4xxp dahdi driver only supports multi-port cards, not single ones. |
| 16:33.54 | *** join/#asterisk ectospasm (~ectospasm@188.72.223.139) |
| 16:48.32 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 16:48.35 | [sr] | hi friends |
| 16:49.01 | WIMPy | Hi [sr] |
| 16:49.09 | [sr] | how r u WIMPy ? |
| 16:49.19 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 16:49.47 | p3nguin | Hmm, what was my quit/part reason? |
| 16:50.25 | WIMPy | [18:31] p3nguin has left IRC (Read error: error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number) |
| 16:53.32 | [sr] | WIMPy: i think i'll abandon LCR... witht he DAHDI progress.. meanwhile i still LOVE the LCR console, dahdi should have something similar |
| 16:55.06 | WIMPy | It's on top of my list to try the exact opposite. Use LCR as PBX instead of jut channel for Asterisk. |
| 16:56.10 | WIMPy | It's time to get some basic functionality working. And as I understand LCR can do all you expect when on it's own. |
| 16:56.32 | WIMPy | Just need to find a few spare minutes to try... |
| 16:57.13 | [sr] | WIMPy: i understand what you are saying, but you're just gonna use LCR, and SIP, how are you goind to do about it? |
| 16:57.24 | [sr] | or you'll just use ISDN phoneS? |
| 16:59.00 | WIMPy | Not entirely sure, but I still have ISDN phones everywhere so I'd like them to work again as they used to. |
| 16:59.41 | [sr] | i see your point |
| 17:00.01 | WIMPy | The trouble is that I need to use dialplans in both LCR and Asterisk then, which would be some extra hassle. |
| 17:00.12 | [sr] | my scenario is diferent, in fact i never had an ISDN phone, and don't know if im able to buy one here, never saw anyone that sell them |
| 17:01.19 | WIMPy | I still love my good old Eurit 40. Best phone I ever had. |
| 17:02.57 | WIMPy | And in fact the only one so far with a sensible user interface for handling multiple calls. |
| 17:09.39 | [sr] | i see |
| 17:09.59 | [sr] | have to see if i can get one just for tesitng porpose |
| 17:10.02 | [sr] | testing |
| 17:13.07 | GlobeTrotterz | Hi Guys, I am running asterisk 1.4. Recently I have started experiencing dropped calls with this error: [2010-09-26 10:01:38] NOTICE[22614]: app_queue.c:2353 wait_for_answer: No one is answering queue '1111' (34/0/0 |
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| 17:13.51 | [sr] | WIMPy: saw it's pics and manual of that Eurit 40, good old 80's style :) |
| 17:14.15 | WIMPy | Yes, but it works in the most optimal way I've seen so far. |
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| 17:30.48 | [sr] | want's an ISDN phone! |
| 17:31.03 | WIMPy | Are you sure? |
| 17:31.36 | WIMPy | You're using Asterisk. |
| 17:31.37 | [sr] | yes, for testing :p |
| 17:32.56 | [sr] | damn it'«s getting cold here... there it goes the summer... |
| 17:55.54 | p3nguin | Wednesday was the first day of autumn. |
| 17:56.03 | p3nguin | (here in the US) |
| 17:58.14 | [sr] | p3nguin: here too... 23-09 |
| 17:58.34 | [sr] | well thursday in my case |
| 17:58.50 | p3nguin | There went the summer. :) |
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| 18:43.54 | iscario | hi, is it possible to use MixMonitor to record a confBridge conference call ? |
| 18:52.10 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 18:56.14 | *** join/#asterisk [cannibalera] (~cannibale@201-25-255-240.fnsce703.dsl.brasiltelecom.net.br) |
| 19:15.27 | *** join/#asterisk MasterXP (~IceChat7@190.167.50.162) |
| 19:15.52 | MasterXP | Hello everybody |
| 19:16.39 | [sr] | I had no idea there was an version for win32 of asterisk |
| 19:17.10 | MasterXP | yeah, the win32 version has long time |
| 19:17.29 | MasterXP | i download once, but never use |
| 19:17.46 | [sr] | ya.. very old version |
| 19:18.22 | MasterXP | yeah |
| 19:19.11 | MasterXP | [sr] do you know a page or a link, where i can read about Asterisk Manager Interface |
| 19:19.23 | [sr] | no idea |
| 19:19.30 | MasterXP | because i have a little problem when dial DAHDI Channels |
| 19:19.44 | MasterXP | wow, i don't found nothing |
| 19:20.03 | *** join/#asterisk plundra (404@article.se) |
| 19:20.59 | *** join/#asterisk |Physis| (|Physis|@201009163167.user.veloxzone.com.br) |
| 19:22.18 | MasterXP | please, some one can answer me, just a question about Asterisk AMI ? |
| 19:22.56 | WIMPy | No question, no answer. |
| 19:23.21 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 19:23.35 | MasterXP | ok |
| 19:23.35 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:24.44 | MasterXP | i want to call a number by dahdi and playback a file |
| 19:25.13 | MasterXP | the problem is that when the dahdi channel answer the audio start to play |
| 19:25.35 | MasterXP | and i want to play when the other end(movil phone, or outside phone) takes the call |
| 19:25.54 | MasterXP | someone have idea? or a link where i can read about this? please. |
| 19:26.45 | ChannelZ | These POTS lines? |
| 19:26.59 | WIMPy | Obviousely :-( |
| 19:27.15 | MasterXP | Action: Originate |
| 19:27.15 | MasterXP | Channel: Channel: DAHDI/g0/18299869786 |
| 19:27.15 | MasterXP | Context: default |
| 19:27.15 | MasterXP | Application: Playback |
| 19:27.15 | MasterXP | Data: /var/spool/asterisk/myaudiofile |
| 19:27.20 | WIMPy | Looks like it can't find out when the call is answered. |
| 19:27.28 | MasterXP | thats the code |
| 19:27.35 | WIMPy | ~pb |
| 19:27.35 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 19:27.57 | MasterXP | yeah WIMPy |
| 19:28.11 | WIMPy | It's a dahdi (configuration) / ancient technology problem. |
| 19:28.17 | ChannelZ | POTS lines don't really get any indication that the remote end has answered. DAHDI will return 'answered' as soon as it's done dialing. |
| 19:28.38 | ChannelZ | You can use 'callprogress' which is basically DAHDI trying to listen to the audio and figure out what is happening, but it's not reliable |
| 19:28.54 | MasterXP | ChannelZ, i don't know that :( |
| 19:29.06 | MasterXP | them there is no way, to do this? |
| 19:29.10 | ChannelZ | don't know what |
| 19:29.34 | WIMPy | Use something that's less outdated. |
| 19:30.14 | MasterXP | can you recommend me something? what i want to do |
| 19:30.20 | ChannelZ | Theoretically PRI does call progress 'correctly' |
| 19:30.59 | [sr] | what about BRI |
| 19:30.59 | [sr] | ? |
| 19:31.13 | MasterXP | yeah, what about BRI? |
| 19:31.37 | WIMPy | It's the same except for capacity. |
| 19:32.17 | MasterXP | What i want to do is a Autodialler, that dial a number and play a audio file... but dont want to do with CALLFILES |
| 19:32.32 | MasterXP | any ideas WIMPy and [sr]? |
| 19:32.39 | [sr] | not me |
| 19:32.43 | WIMPy | We told you. |
| 19:33.12 | ChannelZ | Cheaper than putting in a T1/E1 or whatever would be to get an ITSP and do it over the net |
| 19:33.46 | MasterXP | yes you told me 'callprogress', well lets read about this |
| 19:33.54 | MasterXP | mmm what ITSP means ChannelZ? |
| 19:33.59 | p3nguin | ~itsp |
| 19:34.00 | 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. |
| 19:34.21 | ChannelZ | VoIP |
| 19:34.53 | ChannelZ | You call a SIP channel at your ITSP who actually terminates the call, presumably via PRI which provides good call progress indication |
| 19:35.36 | MasterXP | ohhhh, i understand |
| 19:35.40 | MasterXP | thats a good idea |
| 19:35.47 | MasterXP | the only problem its |
| 19:36.10 | MasterXP | that in my country, Dominican Republic, there is not cheaper SIP Providers :( |
| 19:36.49 | WIMPy | The good thing about the internet is that it's international. |
| 19:36.50 | ChannelZ | well you'd have to look around then, maybe doing PRI yourself is cheaper. How many phone lines do you have right now? (how many simultaneous calls are you interested in making?) |
| 19:37.22 | MasterXP | 3 simultaneous call |
| 19:37.51 | MasterXP | righ now i have T1, 24 channels, but for auto dial just 3 simultaneous call is fine |
| 19:37.54 | WIMPy | Then you should look at two BRIs. |
| 19:38.39 | WIMPy | You have T1? |
| 19:38.40 | ChannelZ | well maybe you just have a config problem with your T1 that you're not getting what you need already |
| 19:38.46 | ChannelZ | Or your Telco is hosed |
| 19:39.47 | MasterXP | mmm, righ now in the test lab |
| 19:39.58 | MasterXP | i testing with SANGOMA FXO cards |
| 19:40.21 | MasterXP | but in the real world, there will be in Digium TE122 |
| 19:40.56 | MasterXP | with t1 PRI lines |
| 19:41.13 | WIMPy | That will make the difference you're looking for. |
| 19:42.24 | MasterXP | you say that, if i test example righ now in a asterisk with a T1, its posible that works? |
| 19:42.51 | WIMPy | yes |
| 19:43.26 | MasterXP | ohhh, let me try then righ now |
| 19:43.46 | MasterXP | i'll be back in ten 15 minutes to tell to you the results |
| 19:43.54 | MasterXP | i hope that works :) |
| 19:45.19 | ChannelZ | wanders off to find lunch |
| 19:51.25 | *** join/#asterisk NickNick (~nicholas@93-97-188-195.zone5.bethere.co.uk) |
| 19:54.10 | iscario | hi i try again^^, is it possible to use MixMonitor to record a confBridge conference call ? |
| 20:35.13 | MasterXP | ChannelZ WIMPy , i tested with a T1 Line and works perfect!!! |
| 20:35.20 | MasterXP | Thanks :) |
| 20:38.20 | *** join/#asterisk darkskiez (~dz@62-50-207-127.client.stsn.net) |
| 20:39.33 | ChannelZ | good to hear |
| 20:40.40 | MasterXP | thanks very much, really :) |
| 20:54.57 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 21:12.18 | sshock | newbie here; I want to set up asterisk and I have a gizmo account |
| 21:12.43 | sshock | what makes more sense? have asterisk register with gizmo server, or |
| 21:13.14 | sshock | setup gizmo to forward all calls to me@mydomain.com ? |
| 21:13.47 | sshock | I assume the latter takes more work, but maybe there is some benefit to doing it that way? |
| 21:15.32 | MasterXP | mmm |
| 21:15.33 | sshock | dumb question: is a "sip trunk" pretty much the same thing as saying "sip connection" ? |
| 21:16.15 | MasterXP | Gizmo is a VOIP Provider, right? |
| 21:16.26 | MasterXP | you wil use to take out calls ? |
| 21:16.27 | sshock | yes |
| 21:16.28 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-150-41.cinci.res.rr.com) |
| 21:16.39 | sshock | both in (DID?) and out (terminator?) |
| 21:16.50 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 21:17.21 | sshock | gizmo will call out to PSTN for 1cent/min |
| 21:17.22 | MasterXP | ok, just make a sip trunk |
| 21:17.34 | MasterXP | and you can make and receive calls from gizmo |
| 21:17.43 | dandate2 | i need to run 8 phones through an ATA (no computers) over a DSL with ulaw, how fast would my DSL need to be to handle? |
| 21:18.15 | sshock | ok, I think I saw a page on voip-info for how to set up a connection with gizmo |
| 21:19.03 | MasterXP | ok |
| 21:19.09 | sshock | but I think it would also be neat to have "sip:myname@mydomain.com" work, but I'm not sure how that would fit in |
| 21:19.15 | dandate2 | there would be no computers at all on its network, just the ATA |
| 21:19.23 | WIMPy | dandate2: About 640k without compression. |
| 21:19.32 | dandate2 | ok |
| 21:19.46 | dandate2 | does anyone know if these cisco ATA devices will do |
| 21:19.49 | dandate2 | PPOE to the DSL |
| 21:20.23 | dandate2 | wimpy: thats 640k upstream right? |
| 21:20.38 | WIMPy | Both directions, obviousely. |
| 21:20.43 | dandate2 | duh heh |
| 21:21.02 | dandate2 | whats the most basic DSL speed? 60k down/12kup ? |
| 21:21.06 | MasterXP | both up and down |
| 21:21.35 | WIMPy | o.O |
| 21:22.10 | dandate2 | i havent used a DSL in 10 years and i remember it being just about 2x faster than dialup on the upstream |
| 21:22.29 | dandate2 | the math is a bit confusing to me, i think 5.3k or 56k i dont know the difference anymore |
| 21:22.49 | sshock | another dumb question: (do I / should I) have to open up port 5060 on my firewall to use asterisk? |
| 21:23.03 | WIMPy | Don't know about your area, here you can get from 1000/128 to 50000/10000. |
| 21:23.07 | dandate2 | the asterisk machine will need 5060 to be open |
| 21:23.13 | sshock | ok |
| 21:23.21 | sshock | will * let me "register" from an external sip phone ? |
| 21:23.23 | dandate2 | as well as the RTP port 10-20k |
| 21:23.28 | dandate2 | right |
| 21:23.33 | dandate2 | if your asterisk is behind a firewall / router |
| 21:23.41 | dandate2 | you wont beable to connect to it without open ports |
| 21:23.54 | WIMPy | Depends on the router. |
| 21:23.56 | dandate2 | a simple experiment would be to get putty an ssh application, try to remote control your machine without open ports |
| 21:24.10 | WIMPy | No! |
| 21:24.27 | WIMPy | We're talking UDP here, not TCP. |
| 21:24.46 | dandate2 | ahh never knew the difference really |
| 21:25.02 | WIMPy | Well, SIP can be TCP, but RTP is UDP. |
| 21:25.15 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 21:25.16 | sshock | yeah, I've never seen it use TCP, only UDP |
| 21:25.20 | *** join/#asterisk ttwhy (~tekkno@p4FECF7C7.dip.t-dialin.net) |
| 21:25.37 | Nugget | you have to explicitly make it use tcp |
| 21:26.09 | sshock | that explains it |
| 21:26.20 | sshock | Ok, tell me if this idea makes sense or is stupid: |
| 21:26.53 | sshock | I'd like if someone calls my gizmo number it connects to asterisk which answers and says "Press 1 for joe; press 2 for sally" |
| 21:27.22 | sshock | I'd also like to have "joe@mydomain.com" and "sally@mydomain.com" be valid so anyone can call those directly |
| 21:27.24 | WIMPy | Why don't you get a 2nd number? |
| 21:27.55 | sshock | I'd like to be able to "register" as joe or sally from anywhere (home, on vacation, etc.; ip phone, softphone, etc)g |
| 21:28.02 | sshock | a 2nd number? you mean from gizmo? |
| 21:28.18 | WIMPy | for example. |
| 21:28.23 | sshock | gizmo doesn't allow new accounts ever since google acquired them :( |
| 21:28.47 | WIMPy | Ok, but there are enought ITSPs out there. |
| 21:29.34 | sshock | but I've heard using gizmo I'll be able to make and receive calls 100% for free, by somehow marrying it with google voice |
| 21:30.02 | sshock | I already know how to have my google voice number forward to my gizmo number; that part was easy |
| 21:30.33 | sshock | and I have seen some tutorials on getting * to make calls via google voice somehow; I'm still a ways out from figuring all that out though |
| 21:31.25 | sshock | so anyway, I'd like to have mydomain.com be a "sip server" (is that the right term?); can * do this, or do I need to use something like SER ? |
| 21:31.58 | WIMPy | Asterisk will do. |
| 21:32.16 | sshock | ok, good |
| 21:32.23 | sshock | I've been reading a lot the last couple days, but I just haven't been able to wrap my head around everything yet... |
| 21:33.34 | sshock | I think I'm getting to the point where I just need to install it and start playing with it before I can understand much more... |
| 21:33.50 | WIMPy | Or you reed the |
| 21:33.54 | WIMPy | ~book |
| 21:33.55 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:34.23 | sshock | yeah, I've been reading parts of that... |
| 21:34.39 | sshock | one main question I have is, should I go with something like FreePBX, or just configure * directly? |
| 21:35.05 | sshock | I already have a debian server that I'm using for other stuff (web server, mail server, jabber, etc.) |
| 21:35.16 | WIMPy | I'd go for pure Asterisk. |
| 21:35.42 | sshock | yeah, I'm leaning towards that I think; |
| 21:35.54 | sshock | I mean, part of me thinks it would be nice to have something simplify things for me, |
| 21:36.03 | sshock | but the other part of me wants to get in deep and dirty |
| 21:36.10 | sshock | and understand how things work |
| 21:37.07 | sshock | here's another dumb question, not really about asterisk: |
| 21:37.36 | sshock | are there IP phones out there that can register to multiple sip accounts simultaneously? |
| 21:38.02 | bougyman | yes |
| 21:38.05 | bougyman | most |
| 21:38.10 | sshock | ok, good |
| 21:38.24 | sshock | I want my phone at home to be able to receive calls for both joe and sally |
| 21:38.50 | sshock | where at my office I'll have the phone only register for joe |
| 21:40.10 | sshock | Has anyone here been successful with getting * to dial out via gizmo via google voice so calls are free, and if so, what are some of the gotchas I should watch out for? |
| 21:40.36 | sshock | I read a couple articles on it, and I assume it is still possible (fingers crossed) |
| 21:45.28 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 21:49.04 | sshock | Ok, question about "allowguest" |
| 21:49.23 | sshock | I need to set that to "no", or else anyone on the Internet would be able to connect to my * box and make calls from it, correct? |
| 21:50.34 | sshock | I can't believe "yes" is the default. Or did I read it wrong here? |
| 21:51.18 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 21:53.04 | WIMPy | You need it if oyu want to be reachable for anyone via ULR. |
| 21:53.06 | WIMPy | URL |
| 21:53.33 | sshock | oh, ok; it sounds like I read it wrong then |
| 21:54.35 | sshock | so if I set it to "no", I'll only be reachable via my gizmo number that my * is trunked to ? |
| 21:55.14 | WIMPy | Only by authorized (configured) peers. |
| 22:00.29 | sshock | ok |
| 22:00.33 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 22:00.58 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 22:01.59 | sshock | so if I set it to "yes", that allows people to call me by URL, but it doesn't allow them to use my * to call out, right? |
| 22:02.30 | WIMPy | Depends on your dialplan. |
| 22:02.48 | WIMPy | -> Contexts |
| 22:02.55 | sshock | I thought I read somewhere that * allows anonymous phones to call out, and that is what worries me. I don't want to enable that by accident. |
| 22:03.32 | sshock | I only want registered users (that have authenticated with a password) to be able to make outgoing calls. |
| 22:04.32 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:05.39 | sshock | is there a context called "from-trunk"? |
| 22:05.55 | WIMPy | If you configure one. |
| 22:06.21 | sshock | ok, and so then I would make it so anything in the "from-trunk" context would be limited |
| 22:06.51 | sshock | they can only make calls to my defined users, not back out to anyone in the world |
| 22:07.13 | WIMPy | That would make sense. |
| 22:09.48 | sshock | so given that "allowguests" is yes by default, and * listens on port 5060, |
| 22:10.21 | sshock | then is it true that if I install asterisk, set it up to connect/trunk to my ITSP (gizmo), and open up port 5060 on my firewall |
| 22:11.08 | sshock | and if I don't do any extra work with contexts, then by default I am opened up so that anyone in the world could abuse my * box to make outgoing calls from it |
| 22:11.53 | WIMPy | Well, unless you do some configuration it's more likely that you can't even call out yourself. |
| 22:12.15 | *** join/#asterisk Faithful (~Faithful@180.194.1.116) |
| 22:22.08 | sshock | ok |
| 22:24.24 | sshock | I'm also trying to decide whether to buy IP phones or buy an analog<->digital router so I can keep using my existing phones... |
| 22:35.36 | sshock | or there's that PCI card thing I guess |
| 22:41.42 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 22:42.08 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 22:48.01 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 23:01.52 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 23:17.11 | *** join/#asterisk Boter (563d44b0@gateway/web/freenode/ip.86.61.68.176) |
| 23:17.19 | Boter | hello everybody |
| 23:17.46 | Boter | i have one question... just installed and configured asterisk, so I can make calls with softphone to other softphone |
| 23:18.04 | Boter | how can i call landline number? |
| 23:19.57 | *** join/#asterisk Faithful (~Faithful@180.194.1.124) |
| 23:23.10 | *** join/#asterisk [cannibalera] (~cannibale@201-25-255-240.fnsce703.dsl.brasiltelecom.net.br) |
| 23:23.36 | *** part/#asterisk [cannibalera] (~cannibale@201-25-255-240.fnsce703.dsl.brasiltelecom.net.br) |
| 23:26.32 | Boter | okay nobody here :) |
| 23:27.44 | drmessano | ~itsp |
| 23:27.44 | 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. |
| 23:29.16 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 23:34.28 | *** join/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 23:34.53 | WIMPy | Is there a length restriction for voicemail numbers? |
| 23:36.57 | R-Guy | Anyone know if there is a Spanish Asterisk IRC channel? |
| 23:42.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:44.06 | *** join/#asterisk psykon (~joel@72.186.121.75) |
| 23:48.05 | drmessano | R-Guy, press 2 for Espanol |
| 23:49.28 | R-Guy | drmessano: Tried asterisk-es, but no one home. Maybe just a slow day?? |
| 23:50.24 | R-Guy | I have a couple of telecom terms in Spanish that I am hoping to get some help with. |
| 23:50.35 | drmessano | I dunno.. you could always ask in here, in English, and see what happens. I could try google translate for you, but if I tell you to "roll your dialplan into a donkey and left it", that may be a problem |
| 23:51.21 | R-Guy | I can do the literial translation myself, need someone that can help in the telecom context. |
| 23:51.46 | drmessano | Ok, well, I know Taco Bell isn't the Mexican phone company |
| 23:52.15 | R-Guy | Not much of a restaurant either. |
| 23:52.32 | R-Guy | I'll try again in the morning. |
| 23:56.08 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 23:58.03 | carrar | taco taco |