| 00:01.57 | recurs|ve | WIMPy: diemos: bad eth0 |
| 00:01.59 | *** join/#asterisk Godfather_ (~godfather@42.Red-88-9-178.dynamicIP.rima-tde.net) |
| 00:09.10 | diemos | ugh, this is annoying -.- |
| 00:11.51 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 00:24.49 | diemos | oh cool, recurs|ve got it. |
| 00:28.32 | carrar | m00f |
| 00:29.25 | WIMPy | Is there a reason why the VPMOCT128 has hardwareDTMF disabled by default? |
| 00:31.01 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 00:40.36 | CheBuzz_Home | Why would I get a "No audio format found to offer" error? I have enabled ulaw, gsm, and ilbc, all of which my ATA and SIP provider support. |
| 00:51.27 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:01.54 | *** join/#asterisk Lantizia (~lantizia@erebus.seaquake.net) |
| 01:02.11 | Lantizia | Hey, odd question here... has anyone ever managed to port a number - to themselves? |
| 01:03.30 | Lantizia | Clarification: In the UK all the number ranges are owned by a particular company, call them company A. If you port to a new telecoms company (lets say B) then B negotiates with A but the range holder doesn't change. If you then port to company C then they need to talk to company A not B |
| 01:04.19 | Lantizia | So I guess I'm wondering if you can get agreement off the original range holder, might it be possible to port directly to yourself as an individual ? |
| 01:13.02 | WIMPy | What it that supposed to do? |
| 01:15.41 | *** join/#asterisk radic (~radic@dslb-094-216-233-062.pools.arcor-ip.net) |
| 01:18.08 | Lantizia | ? |
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| 02:28.34 | daxt | guys i am trying to compile dahdi and get the following error , "You do not appear to have the sources for the 2.6.16-xenU kernel installed" so how do i fix this ? |
| 02:29.06 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 02:32.53 | daxt | anybody home ? |
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| 03:15.35 | *** join/#asterisk FinboySlick (~shark@74.117.40.9) |
| 03:16.36 | FinboySlick | out of curiosity (nothing is broken or anything) what does this mean exactly? "utils.c: trying to reset empty pool" |
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| 03:48.15 | daxt | guyz i am trying to compile dahdi and get this error "You do not appear to have the sources for the 2.6.16-xenU kernel installed." |
| 03:48.20 | daxt | how do i fix this ? |
| 03:49.38 | WIMPy | Install them. |
| 03:49.39 | Corydon76-home | daxt: Install the sources for the 2.6.16-xenU kernel |
| 03:49.56 | daxt | kernel patch ? |
| 03:50.04 | WIMPy | As with any kernel module, you need to have configured sources of your running kernel. |
| 03:50.08 | Corydon76-home | No patch. Just the sources. |
| 03:50.24 | Corydon76-home | Actually, just the kernel headers will do. |
| 03:50.25 | daxt | i am doing it on CentOS |
| 03:50.35 | daxt | i guess the headers would do it |
| 03:50.37 | daxt | ya |
| 03:50.40 | daxt | exactly |
| 03:50.46 | daxt | can u tell me how to install it ? |
| 03:51.00 | daxt | i am an ubuntu guy |
| 03:51.04 | daxt | new to centos |
| 03:51.23 | daxt | in ubuntu make automatically got the kernel source from apt-get |
| 03:51.27 | Corydon76-home | but the path /lib/modules/`uname -r`/build generally has to point to it |
| 03:51.28 | daxt | but not in centos |
| 03:51.35 | Corydon76-home | yum is the apt-get of centos |
| 03:51.51 | Corydon76-home | roughly speaking |
| 03:51.57 | daxt | but i feel that yum is inferior to apt-get |
| 03:52.30 | Corydon76-home | You're welcome to feel a lot of things. However, you still have to use it. |
| 03:52.38 | daxt | yup :D |
| 03:52.40 | daxt | and i dont hate it |
| 03:52.43 | WIMPy | Or just use a distro you like. |
| 03:53.00 | daxt | unfortunately asterisk seems to hate ubuntu :( |
| 03:53.17 | Corydon76-home | Funny, because most of the core developers use Ubuntu |
| 03:53.27 | daxt | indeed |
| 03:53.45 | daxt | i tried many guides on the internet and installed asterisk and freepbx |
| 03:53.58 | daxt | and could not get a perfect installation |
| 03:54.02 | daxt | actually the issue is with freepbx |
| 03:54.31 | Corydon76-home | Yes, well, Asterisk experts don't use freepbx |
| 03:55.00 | daxt | so what should i use then ? |
| 03:55.02 | daxt | cli ? |
| 03:55.04 | Corydon76-home | FreePBX is the training wheels of the telephony world. |
| 03:55.21 | daxt | even asteriskgui did not work |
| 03:55.24 | Corydon76-home | Yep, edit your files directly or roll your own interface |
| 03:55.39 | daxt | i am new to asterisk |
| 03:55.45 | daxt | so what do u guys reccomend ? |
| 03:55.50 | WIMPy | vi |
| 03:55.55 | Corydon76-home | vim |
| 03:56.07 | daxt | i use vi and nano :D |
| 03:56.21 | Corydon76-home | func_odbc and a static dialplan |
| 03:56.37 | daxt | but now i want to get freepbx running for my boss |
| 03:56.50 | Corydon76-home | Abstract your data away from your logic |
| 03:56.51 | daxt | can somebody show me a perfect guide ? |
| 03:57.09 | daxt | <Corydon76-home> Abstract your data away from your logic <--- absolutely |
| 03:57.22 | Corydon76-home | Roll your own |
| 03:58.15 | daxt | will try as time permits :) |
| 03:59.19 | Corydon76-home | GUIs are designed to allow idiots to feel like they can alter stuff without repercussions. How many times has a firewall gotten screwed up because it has a GUI on it, and a boss messed with the settings? |
| 03:59.51 | daxt | no |
| 03:59.55 | daxt | he knows nothing |
| 04:00.04 | daxt | and that is why i have to give him the gui |
| 04:00.13 | daxt | so he will feel confident |
| 04:00.14 | Corydon76-home | No, you give him nothing |
| 04:00.24 | daxt | they he wont pay me :( |
| 04:00.31 | daxt | *then |
| 04:00.47 | Corydon76-home | Or you give him a GUI that does not actually do anything. Blinky lights, no functional knobx |
| 04:02.32 | daxt | that might work , put my neck on the line :D |
| 04:06.35 | Corydon76-home | That's the spirit! |
| 04:06.59 | daxt | alright , lets do it then :D |
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| 04:43.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 04:52.56 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 04:54.32 | v1s | what is the best way to dial if u have multiple lines. put like Dial(prov1&prov2&prov3) etc.. or Dial(prov1) Dial(prov2) in sequence. |
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| 05:00.34 | jaytee | v1s, the first way is correct |
| 05:01.11 | jaytee | it will dial all 3 simultaneously. the first to pick up wins |
| 05:02.37 | v1s | jaytee: thanks |
| 05:03.56 | jaytee | and to be more specific it would be Dial(SIP/prov1&SIP/prov2&SIP/prov3) |
| 05:06.53 | v1s | yes I know :) I was just trying to keep post short ;) |
| 05:07.29 | v1s | thanks again |
| 05:09.30 | jaytee | your welcome |
| 05:21.29 | drmessano | jaytee! |
| 05:23.51 | jaytee | hey drmessano |
| 05:25.02 | jaytee | so far I'm liking flowroute.com, thanks for the recommendation |
| 05:25.46 | drmessano | Awesome |
| 05:25.56 | drmessano | They're good people |
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| 06:22.45 | joeyjones | yeah, there seem to be quite a few decent companies |
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| 08:44.54 | trumee | guys. i did a timing test using dahdi_test. Results after 132627 passes. Best: 100.000 -- Worst: 98.868 -- Average: 99.996499, Difference: 99.996568 |
| 08:45.03 | trumee | is that bad? |
| 08:45.32 | trumee | this is all software based. |
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| 09:11.26 | ChannelZ | The average is good |
| 09:12.33 | ChannelZ | but that's a lot of passes :) |
| 09:18.46 | trumee | ChannelZ, yes i had it running for a day. |
| 09:19.13 | trumee | ChannelZ, i was worried that the worst was below 99. |
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| 09:19.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 09:53.30 | atan | Anyone know of a provider that offers Canadian DIDs in Nova Scotia outside of Halifax/Dartmouth? |
| 09:53.45 | atan | I don't even see any oddball names registered for rate centers outside of the city. |
| 09:53.54 | atan | Seems like Telus/Bell/Aliant |
| 10:41.00 | *** join/#asterisk dogonovmax (~Adium@78-106-142-111.broadband.corbina.ru) |
| 10:42.58 | dogonovmax | Hi! Anyone can you help me with Asterisk 1.8.1. I need receive after command Dial in dialplan SIP response in variable! |
| 10:47.53 | dogonovmax | My dialplan: |
| 10:49.16 | dogonovmax | exten => _X.,1,Dial(${bridgeid}/${EXTEN},,S(${CREDIT_TIME})ge:) |
| 10:49.16 | dogonovmax | exten => _X.,2,NoOP(${DIALSTATUS}) |
| 10:49.17 | dogonovmax | exten => _X.,3,NoOP($${HANGUPCAUSE}) |
| 10:49.17 | dogonovmax | exten => _X.,4,NoOP(${HASH(SIP_CAUSE,<slave-channel-name>)}) |
| 10:49.17 | dogonovmax | exten => _X.,5,Hangup |
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| 11:48.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:09.39 | *** join/#asterisk imox1234 (~imox1234@p4FC5C50B.dip0.t-ipconnect.de) |
| 12:13.08 | imox1234 | hello what means peer <guest> ? |
| 12:28.10 | ectospasm | imox1234: doesn't have a strict meaning in Asterisk. You can create a guest SIP peer, but typically that's not advisable |
| 12:28.58 | imox1234 | when I enter sip show channels I see that http://pastebin.com/Kc5SC8Pr |
| 12:29.41 | imox1234 | so much guests but with no activ channels. but whats thats I'm worried :-) |
| 12:30.26 | imox1234 | and I have set this in my sip.conf allowguest=no |
| 12:31.45 | ectospasm | This is just a guess, but <guest> means no peer has registered to that IP address. You'll notice that 212.227.67.201 has Peer lundl_275, not <guest> |
| 12:32.37 | imox1234 | ?? |
| 12:32.42 | ectospasm | imox1234: it depends, where do you have allowguest? You may need to put it in the peer sections, as well as [general] |
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| 12:33.17 | imox1234 | I set this in the generals |
| 12:33.19 | ectospasm | imox1234: I don't have any SIP peers on my 1.8 machine |
| 12:34.08 | imox1234 | what else? |
| 12:35.22 | ectospasm | imox1234: are you having any other symptoms? Like SIP peers not registering? |
| 12:35.35 | imox1234 | no |
| 12:35.54 | ectospasm | you may be chasing your tail then |
| 12:36.10 | imox1234 | this I has for a long time ago but this blocked this IP |
| 12:40.02 | imox1234 | now I see a channel guest with the same IP adress from one of my sip peers |
| 12:40.22 | imox1234 | very strange |
| 12:41.10 | ectospasm | imox1234: change the auth credentials, peer name, secret, etc. |
| 12:43.03 | imox1234 | hmm ok thanks |
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| 14:18.16 | lauris | hi |
| 14:18.45 | lauris | have a problem with phpagi-asmanagerc class |
| 14:19.04 | lauris | can't get it into loop mode to listen for incoming events on asterisk manager port |
| 14:20.50 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 14:26.08 | lauris | solved. |
| 14:27.33 | leifmadsen | thanks for letting us know how you did that ;) |
| 14:28.40 | cmnky | leifmadsen, ;) |
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| 15:24.58 | *** join/#asterisk path (path@gateway/shell/bshellz.net/x-wmwtsuholmhavulk) |
| 15:25.05 | path | hey guys |
| 15:25.17 | path | what does it mean 'Dead' when I run agi show commands ? |
| 15:25.25 | path | (first column) |
| 15:31.33 | leifmadsen | path: Dead AGI is when an AGI is run on a hung up channel (such as in the 'h' extension) |
| 15:32.08 | leifmadsen | newer versions of Asterisk (1.6.2+ I believe) don't distinguish between DeadAGI() and regular AGI() anymore. You just use AGI() everywhere |
| 15:32.38 | path | I see, thanks :) |
| 15:34.16 | leifmadsen | path: check out the Asterisk Gateway Interface chapter at http://ofps.oreilly.com |
| 15:34.19 | leifmadsen | it'll explain all that |
| 15:38.29 | path | thats nice! thank you |
| 15:44.02 | path | is it possible to download it as pdf? |
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| 16:02.55 | *** join/#asterisk Navion (~Bill@rivendell/users/navion) |
| 16:03.28 | Navion | Anybody know of an AoIP interface for asterisk? |
| 16:05.48 | Navion | How about 4W analog interface. Like a trunk interface? |
| 16:07.57 | jaytee | i heard a rumor that a few companies made them. I think one is called Digium. Pretty sure they make 4 port and even 24 port analog interface cards for Asterisk. |
| 16:12.41 | Navion | Thanks |
| 16:13.33 | Navion | Know of any multi-line SIP soft phones? |
| 16:14.06 | Navion | Like an operators console sort of thing. |
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| 17:06.25 | *** join/#asterisk ghostmediapro (~Trupsalms@71.23.182.164) |
| 17:07.45 | ghostmediapro | quick question room: could and modem or fxo/fxs card be used to talk my sip based pbx from softphone to landlines? |
| 17:09.13 | pabelanger | ghostmediapro: no directly, that's what Asterisk is for |
| 17:09.19 | pabelanger | s/no/not/ |
| 17:13.19 | ghostmediapro | i under stand that, i have build a working system before, but be for i attempt it again, i wanted to know, see my old system was sip and i used softphones or the computer to make the calls, now i want to tie asterisk now back into the phone line running through my house, so that i could pick up any phone and make and receive calls, when i'm not at a computer |
| 17:15.23 | pabelanger | ghostmediapro: yes, that is a pretty straight forward task. I'd suggest reading the book. |
| 17:15.25 | pabelanger | ~book |
| 17:15.25 | 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, or http://ofps.oreilly.com/ |
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| 17:18.14 | ghostmediapro | pabelanger: now could any idsn modem card do it, or dose it have to be fxo/fxs |
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| 17:19.53 | pabelanger | ghostmediapro: No, Asterisk (actually DAHDI) only support certain cards. If you are dealing with an analog line you'll need a FXO port. |
| 17:20.09 | ghostmediapro | pabelanger: i wouldn't now where to lock in the book |
| 17:20.38 | pabelanger | Start at the 'Table of Contents' |
| 17:20.47 | ghostmediapro | sip to fxo/fxs for pstn |
| 17:21.15 | pabelanger | read the section called 'Connecting to the PSTN' |
| 17:22.05 | ghostmediapro | last them as i am on the asterisk now site and can't find the link for the book, if you have it handy |
| 17:22.23 | pabelanger | ~book |
| 17:22.23 | 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, or http://ofps.oreilly.com/ |
| 17:22.28 | pabelanger | ghostmediapro: ^^ |
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| 22:40.45 | lordvadr | I'm having trouble getting usgmtime to work in the adaptivev cdr config. Does it not work? |
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