00:00.30 | afg_ch | asdx: I guess your Linux workstation will recognize the embedded USB soundcard as plug and play device and you can choose it as input/output sounddevice from the dropdown list in your preferred application. |
00:00.33 | asdx | lol |
00:00.45 | asdx | yeah |
00:01.37 | asdx | <PROTECTED> |
00:01.37 | asdx | <PROTECTED> |
00:01.40 | asdx | i will fix it |
00:02.05 | afg_ch | asdx: any nowadays the sound wizard will configure it for you at a comfortable soundlevel. |
00:02.07 | asdx | i just have to chage my ~/.asoundrc and put "card 0" on t |
00:02.08 | asdx | it* |
00:07.11 | tzafrir_laptop | asdx, full-speed? (USB1)? |
00:08.57 | asdx | input: USB HID v1.00 Device [Logitech Logitech USB Headset] on usb-0000:00:02.0-1 |
00:09.00 | asdx | yeah |
00:09.28 | asdx | guys what does this means: http://pastie.org/224203 |
00:09.39 | asdx | the numbers |
00:09.43 | asdx | 1001 |
00:09.46 | asdx | 2000 etc |
00:10.00 | asdx | ah the seconds |
00:10.12 | afg_ch | the cost of translation |
00:10.57 | asdx | cool |
00:11.08 | afg_ch | unlikely to be seconds. rather milliseconds but I am not sure on that. |
00:11.16 | asdx | yeah |
00:12.33 | afg_ch | be told that this value seems a bit high. I only see 11 for ilbc. The rest of the heard goes at about 1-2 |
00:13.00 | asdx | yeah this is in asterisk 1.6 |
00:13.40 | asdx | in asterisk 1.4 the value is lower |
00:14.08 | asdx | but asterisk 1.4 speaks about "milliseconds" and 1.6 in "microseconds" |
00:17.51 | asdx | feels like watching a good movie and eating pizza |
00:21.19 | afg_ch | my eyes are also heavy. I move my operations over to the sofa. |
00:21.38 | afg_ch | good night everybody! |
00:21.40 | asdx | heh |
00:21.44 | asdx | night dude |
00:31.03 | mvanbaak | my eyes start to be heavy as well |
00:31.08 | mvanbaak | Sun Jun 29 02:31:07 CEST 2008 |
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01:40.23 | Blacraft | hello can anyone help me, i can't keep a dialtone on my tdm410 fxs modules |
01:44.11 | Strom_C | what do you mean "keep a dialtone"? |
01:50.40 | Blacraft | if i reboot the server i get a dialtone for about 1 minute, then it goes away. |
01:56.04 | Strom_C | ok... |
01:56.06 | Strom_C | hmm |
01:56.45 | Strom_C | Blacraft: does simply unloading and re-loading the modules cause the dialtone to come back? |
01:58.54 | Blacraft | yes |
01:59.03 | Blacraft | ony for a bit though |
02:00.40 | Strom_C | Blacraft: I'd suggest you give Digium support a call |
02:02.00 | Blacraft | yeah ok thz |
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03:11.32 | asdx | how do i make sip work with tcp, transport=tcp? |
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03:14.43 | JayTee52 | asdx, you have to use * 1.6 for starters |
03:15.39 | asdx | JayTee52: i use 1.6 |
03:16.04 | JayTee52 | oh, well it's documented in the sip.sample.conf file and the docs |
03:17.05 | asdx | k |
03:17.37 | asdx | so it's tcpenable=yes |
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06:49.46 | ^^Johnny5 | can anyone help with asterisk and digium tdm04P boards... |
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07:13.45 | iuyto | hi |
07:14.09 | iuyto | i got sipura 3000 and asterisk installed on my system |
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07:15.33 | iuyto | my server asterisk is at home |
07:15.48 | iuyto | my sipura 3000 adaptor is on my work |
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07:16.40 | iuyto | sipura3000 connected with my work pstn line |
07:17.39 | iuyto | how to remote my pstn work line to work with my asterisk asterisk extension ,?? |
07:21.05 | TJNII | ~trixbox |
07:21.06 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
07:21.19 | TJNII | iuyto: ^^^^^ |
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07:27.42 | iuyto | thx TJNII |
07:28.15 | iuyto | the problem is not the asterisk installation |
07:28.29 | iuyto | i have allready install asterisk |
07:28.56 | iuyto | just wanna configure my sipura3000 work line to my asterisk box |
07:29.19 | TJNII | No, you installed trixbox |
07:29.25 | iuyto | eventough i try to install trixbox |
07:29.26 | TJNII | Which configures asterisk for you |
07:29.33 | TJNII | So we can't support it here |
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07:29.40 | TJNII | Because we don't know what trixbox did |
07:29.49 | TJNII | And since it did it for you, niether do you. |
07:30.16 | iuyto | ok |
07:30.24 | iuyto | thx TJNII |
07:30.32 | TJNII | Try #trixbox |
07:30.39 | iuyto | ok thx |
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07:33.51 | iuyto | noone there |
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07:47.16 | TJNII | iuyto: So you have the fxo line on the Sipura connected to a POTS line, and you want * to answer it? |
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07:55.20 | iuyto | ye |
07:55.29 | iuyto | yes TJNII |
07:55.29 | TJNII | And then what |
07:55.38 | TJNII | What do you want * to do with the call |
07:56.11 | iuyto | my sipura3000 is connected to my work pstn line |
07:56.22 | iuyto | my asterisk server is at home |
07:56.25 | TJNII | Through the fxo port, yes |
07:56.30 | iuyto | yes |
07:56.51 | TJNII | The server is physically at home or you are using asterisk@home (aka trixbox) |
07:57.02 | iuyto | i create an sip extension from asterisk server |
07:57.16 | TJNII | Are you using a gui or did you install from source? |
07:57.18 | iuyto | at home |
07:57.25 | iuyto | psycally |
07:58.03 | iuyto | yes from source |
07:58.11 | iuyto | and i compile the program |
07:58.23 | TJNII | Okay, than that is different. |
07:58.28 | TJNII | Then we can help you here. |
07:58.37 | iuyto | thx |
07:58.58 | TJNII | So you have the sipura at your work and asterisk at your home. You want work calls to go to the asterisk box |
07:59.18 | iuyto | no |
07:59.30 | iuyto | i just want to use my pstn work line |
07:59.36 | iuyto | from my home |
08:00.09 | TJNII | What hardware do you have at your home |
08:00.17 | TJNII | Any telephone adapters? |
08:00.26 | TJNII | Or do you want to answer the call on a softphone |
08:00.38 | iuyto | from softphone |
08:00.42 | iuyto | yes |
08:00.43 | iuyto | at home |
08:00.58 | TJNII | So you have the sipura at your work and asterisk at your home. You want work calls to go to the asterisk box which will then send it to an softphone. |
08:01.19 | iuyto | no |
08:01.47 | iuyto | i just want to use |
08:02.06 | iuyto | my work line to call outside the world |
08:02.20 | iuyto | from my softphone |
08:02.29 | TJNII | So what I said but the other way around |
08:02.43 | TJNII | Softphone -> asterisk -> sipura |
08:02.49 | iuyto | yes |
08:02.53 | TJNII | And the sipura will not answer the line |
08:03.03 | iuyto | that 's why i put sipura 3000 at my work |
08:03.40 | iuyto | yes it can answer the line |
08:03.46 | iuyto | using my sip extension |
08:03.55 | TJNII | Now, before we go too far, do you have NAT at home? Do you have a static IP? |
08:04.10 | iuyto | no NAT |
08:04.17 | iuyto | Static IP yes |
08:04.19 | iuyto | i have |
08:04.38 | TJNII | Cool. Then you won't have to deal with the one-directional auido headaches NAT causes. |
08:05.00 | TJNII | How far have you gotten with your configuration? |
08:05.43 | iuyto | sorry ?? |
08:05.52 | TJNII | What have you done so far? |
08:06.49 | iuyto | i just my asterisk |
08:06.51 | iuyto | server |
08:07.02 | iuyto | with my sip extension |
08:07.25 | TJNII | So you have the softphone connected already? |
08:07.30 | iuyto | which means i can call to my work using my asterisk server sip extension |
08:07.35 | iuyto | yes |
08:07.55 | TJNII | So you can make outgoing calls already? |
08:07.57 | iuyto | no |
08:08.24 | iuyto | from home softphone i can only join my work sipura3000 |
08:08.31 | iuyto | using my sip extension |
08:08.53 | TJNII | What do you mean ny "join my work sipura3000" |
08:08.57 | TJNII | You can call it? |
08:09.36 | iuyto | as sipura3000 got 1 FXS port and 1 FX0 port |
08:09.57 | TJNII | Right.... |
08:10.01 | iuyto | i configure my work sipura3000 to use my sip extension |
08:10.35 | iuyto | now i just want to connect my work pstn line on FXO port |
08:11.04 | iuyto | and remote using my work pstn line from home |
08:11.04 | TJNII | So you have the FXS port set up now? Or the FXO port? |
08:11.24 | TJNII | Yes, I'm clear on what you want to do, but I'm unclear on how far you've gotten |
08:13.40 | iuyto | i got i pc wich is running |
08:13.48 | iuyto | on asterisk server at home |
08:14.02 | iuyto | 1 softphone at home |
08:14.12 | iuyto | and 1 siprua3000 at work |
08:14.27 | iuyto | that's all |
08:14.58 | TJNII | And the softphone is setup successfully to * |
08:16.11 | iuyto | yes |
08:16.41 | TJNII | And you have the FXO line set up to connect to? |
08:16.42 | iuyto | i can make call from my home softphone to my work sipura3000 which is connected on fxs port |
08:16.52 | TJNII | s/connect to/connecto to */ |
08:17.44 | TJNII | Okay, so you can call a phone on the fxs port from the softphone, and now you want to set up the fxo port? |
08:17.56 | iuyto | yes |
08:17.57 | TJNII | Do I understand that correctly? |
08:18.13 | iuyto | yes it's correct |
08:18.23 | TJNII | Do you uunderstand dialplan contexts? |
08:18.38 | iuyto | yes |
08:18.56 | iuyto | not really |
08:19.03 | iuyto | no not really |
08:19.25 | TJNII | Okay, I need to go to bed, but I'll outline what you need to do. |
08:19.32 | TJNII | 1) Read chapter 5 of the book |
08:19.34 | TJNII | !book |
08:19.38 | TJNII | ~book |
08:19.38 | jbot | 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://tfot.leifmadsen.com, or see ~buybook |
08:19.50 | TJNII | That will give you an understanding of dialplans. |
08:20.02 | iuyto | yes |
08:20.05 | iuyto | i will |
08:20.09 | TJNII | 2) add a entry in sip.conf for the fxo port |
08:20.30 | TJNII | Shouldn't be too hard since you were successful with the fxs |
08:20.31 | iuyto | yes i have done |
08:20.51 | iuyto | wait i show u an example on what i have done |
08:21.12 | TJNII | mmmkay |
08:21.25 | TJNII | ~pastebin |
08:21.25 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
08:24.20 | iuyto | plz here http://pastebin.com/m77a5cc10 |
08:25.14 | TJNII | pastebin your dialplan. |
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08:26.43 | TJNII | And your entire sip.conf |
08:27.02 | iuyto | here |
08:27.34 | iuyto | http://pastebin.com/m2faba158 |
08:29.54 | TJNII | I only see one phone in that sip.conf |
08:29.58 | TJNII | There should be 3 |
08:30.35 | iuyto | why 3 ? |
08:31.00 | iuyto | my asterisk server LAN ip is 192.168.0.1 |
08:31.00 | TJNII | One for the FXO line, One for the FXS line, one for the softphone. |
08:31.08 | TJNII | Each device is independent. |
08:32.06 | iuyto | i create 6 sip extensions all work |
08:32.14 | iuyto | without any problem |
08:32.18 | iuyto | using my softphoen |
08:32.23 | iuyto | softphoen |
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08:33.02 | TJNII | There is no such thing as a sip extension. You have SIP peers which devices register as. |
08:33.14 | TJNII | You define extensions in your dialplan. |
08:33.48 | TJNII | Now in the sip.conf you posted, I only see one sip peer. So One device. |
08:33.59 | TJNII | You need a minimum of two for what you want to do. |
08:34.40 | iuyto | ok |
08:35.07 | iuyto | give the right way on what i have to do ?? |
08:35.15 | iuyto | on my asterisk server side |
08:35.24 | iuyto | and on sipura 3000 side |
08:35.29 | iuyto | ? |
08:36.16 | TJNII | You need a SIP.conf entry for the softphone, and a second SIP.conf entry for the sipura. |
08:36.21 | iuyto | have i have to have a static or dynamic ip type of connection where sipura3000 is connected ? |
08:36.40 | iuyto | ok |
08:36.47 | TJNII | You're going to set the sipura to look to your server's static IP. |
08:37.20 | iuyto | yes i configure one of my sip extension on sipura3000 |
08:37.23 | TJNII | I thought you said you could call the sipura from the softphone? Based on what you posted, that isn't possible. |
08:37.26 | iuyto | is it's enough ? |
08:38.20 | iuyto | no |
08:38.54 | TJNII | So you can register with the softphone _OR_ you can register with the sipura right now |
08:38.55 | iuyto | i can make call from softphone to join my sipura3000 |
08:39.03 | TJNII | That makes no sense. |
08:39.16 | TJNII | Go to the cli and type sip show peers |
08:39.23 | TJNII | Copy the output. |
08:40.03 | TJNII | Make sure to have your verbosity set to >= 5 |
08:40.20 | iuyto | wait |
08:42.36 | iuyto | plz see here |
08:42.38 | iuyto | http://pastebin.com/m7c3edc40 |
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08:43.00 | iuyto | where 1234 is my softphone |
08:43.21 | iuyto | and 0987654321 is my sipura box |
08:44.09 | TJNII | Please pastebin your dialplan. |
08:44.27 | iuyto | i post u all my sip.conf |
08:45.04 | iuyto | where i can find it ? |
08:45.13 | TJNII | Okay. |
08:45.24 | TJNII | You really need to read this: |
08:45.26 | TJNII | ~boot |
08:45.26 | jbot | from memory, boot is what you get when you act like a UnderNet user, or #debian-boot |
08:45.28 | TJNII | ~book |
08:45.29 | jbot | 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://tfot.leifmadsen.com, or see ~buybook |
08:45.35 | TJNII | ^^^^^^^ |
08:46.02 | iuyto | yes |
08:46.03 | TJNII | You can safely skip the beginning, since you already have the server up |
08:46.10 | TJNII | Read chapter 5 CAREFULLY |
08:46.16 | iuyto | what i |
08:46.18 | TJNII | It is key |
08:46.33 | iuyto | have allready buy this book from 75 dollars |
08:46.45 | iuyto | but it's in french |
08:47.30 | TJNII | Well, read the chapter on dialplans |
08:47.54 | TJNII | And things should start to make more sense for you. |
08:48.32 | iuyto | ok thx a lot TJNII |
08:48.36 | TJNII | Me, I need to go to bed. It is 10 till 4 in the morning, and if I sleep into the afternoon my GF will be pissed |
08:48.44 | TJNII | Good luck |
08:48.49 | iuyto | w |
08:48.51 | iuyto | ok |
08:48.53 | iuyto | thx a lot |
08:48.55 | iuyto | bye |
08:48.57 | iuyto | goood night |
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10:40.35 | unpaidbill | fuck nanpa in their dirty microsoft loving asses |
10:40.50 | unpaidbill | wtf are these assholes doing distributing the area code list in .mdb format |
10:41.04 | unpaidbill | fucking idiots |
10:42.55 | unpaidbill | also, i want to punch them all in the goddamn faces |
10:47.33 | tzafrir_laptop | directs unpaidbill to a near-by glass of water |
11:04.22 | unpaidbill | haha tza |
11:04.24 | unpaidbill | yeah. |
11:04.35 | unpaidbill | i had to install openoffice |
11:04.37 | unpaidbill | irritating |
11:05.12 | unpaidbill | and the info i got didnt have what i wanted |
11:05.18 | unpaidbill | so i am still digging through their damn site |
11:09.01 | tzafrir_laptop | unpaidbill, I have to agree. openoffice is irritating |
11:09.05 | tzafrir_laptop | ;-) |
11:09.07 | unpaidbill | yeah |
11:09.13 | unpaidbill | beats paying though |
11:09.47 | unpaidbill | do you have any idea where on the nanpa site they list the local numbers ranges for an areacode and exchange? |
11:10.02 | unpaidbill | i know i downloaded that info in CSV format .... 7 years ago |
11:10.06 | unpaidbill | but i cant seem to find it now |
11:10.30 | unpaidbill | and hawaii has one area code with exchanges broken into local/long distance, so sometimes you have to dial 1808 and sometimes you dial only 7 digit |
11:10.33 | unpaidbill | annoying as hell. |
11:11.13 | unpaidbill | i guess i can have my clec send it |
11:15.43 | *** join/#asterisk RoyK (n=roy@ip-194-54-149-91.dialup.ice.no) |
11:16.48 | Fiapo-CE | what the best linux-distro to install asterisk? |
11:16.57 | unpaidbill | whichever one you prefer |
11:17.44 | Fiapo-CE | is debian a good choice? |
11:17.50 | unpaidbill | if you like it |
11:18.25 | Fiapo-CE | yes, i like it... but i need recompile kernel etc? |
11:18.35 | unpaidbill | nope |
11:18.51 | Fiapo-CE | good, thanks! |
11:19.01 | unpaidbill | the most you will need to do is install the kernel headers |
11:19.11 | unpaidbill | which is just apt-get install blah |
11:19.22 | unpaidbill | and of course all your compiling utilities |
11:19.27 | unpaidbill | (if you want to use zaptel, that is) |
11:19.53 | unpaidbill | you'll need zaptel for MeetMe (conferencing), unless you're installing 1.6 i believe |
11:20.00 | unpaidbill | for the ztdummy module |
11:20.47 | Fiapo-CE | sure |
11:21.05 | Fiapo-CE | thanks for help |
11:23.02 | [netman] | Fiapo-CE: I love Debian, but I think CentOS based Asterisk system r better supported |
11:23.25 | [netman] | i.e. there r more information on the web related to that distro |
11:24.06 | [netman] | so I have installed my Asterisk over CentOS 5.2 |
11:27.21 | Fiapo-CE | [netman] i will test it |
11:27.25 | Fiapo-CE | I'm testing asterisk as active pbx.. |
11:27.30 | *** join/#asterisk droem (i=droemel2@p548EA431.dip0.t-ipconnect.de) |
11:27.37 | Fiapo-CE | with php i'm creating a .call and moving for outgoing directory |
11:27.54 | Fiapo-CE | I´m using php to integrate my actual system w/ asterisk to make the call |
11:28.07 | Fiapo-CE | sample: http://myserver/call.php?numer=XXX&session=xxxx&extensionline=XXXX |
11:28.18 | Fiapo-CE | I have 10 persons doing active calls, well asterisks make the call and give to extension.. |
11:28.35 | Fiapo-CE | this is the best way to do it? through .call file? |
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11:30.49 | unpaidbill | use the manager interface |
11:31.02 | unpaidbill | info here http://www.voip-info.org/wiki/view/Asterisk+manager+API |
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11:38.50 | *** join/#asterisk rolers (n=relas@port-92-195-88-113.dynamic.qsc.de) |
11:39.52 | rolers | hello. I've got 2 hfc-s cards (one in NT and one TE mode). The NT-card works fine. But with the TE-card I'm just getting the error: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
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11:50.38 | yangvnc | rolers: have you configured zapata.conf correctly ? |
11:50.49 | yangvnc | rolers: defined channels etc.? |
11:51.21 | yangvnc | rolers: maybe you could paste your configuration |
11:51.25 | yangvnc | ~pb |
11:51.26 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
11:52.04 | rolers | yangvnc: I think so: http://rafb.net/p/zw89dT86.html |
11:53.12 | yangvnc | rolers: I had problems with signalling=bri_cpe_ptmp try simply signalling=bri_cpe |
11:53.59 | yangvnc | rolers: and there were problems with SPANS, if you unplug the ISDN cable you should restart asterisk or reboot machine |
11:56.05 | rolers | yangvnc: My extenstion: exten => 35,1,Dial(Zap/g2/85695). I changed the value to bri_cpe, but there is still the same error. |
11:56.47 | rolers | incomming calls also don't work |
11:57.22 | yangvnc | try ZAP/g2/${EXTEN}) |
11:58.07 | rolers | yangvnc: nothing changed. |
11:58.22 | yangvnc | for incoming call you should specify extensions, if your number is 1234 you should make exten => 1234,1,Dial(SIP/<your phone> |
11:58.36 | rolers | yes I know |
11:59.28 | rolers | I'm running asterisk in verbosity level 3. On incomming calls just the capi-card reacts |
12:07.02 | rolers | yangvnc: pri_find_dchan: No D-channels available! Using Primary channel 6 as D-channel anyway! |
12:13.42 | tzafrir_laptop | rolers, It may mean that the span is down |
12:14.53 | tzafrir_laptop | try 'head -n 1 /proc/zaptel/*' . Is layer 1 up in those cards? |
12:16.20 | rolers | tzafrir_laptop: http://rafb.net/p/2LYqPy38.html and /proc/zaptel*:http://rafb.net/p/0W4c1m27.html |
12:17.54 | tzafrir_laptop | rolers, the second parameter in the span= lines should not be the same for all cards, IIRC . Though I don't think it is the issue here |
12:19.37 | tzafrir_laptop | The first card is actually connected as NT . But in zaptel.conf you set it to get timing from the remote side |
12:20.10 | tzafrir_laptop | Do you want to use it as NT (e.g.: to connect an ISDN phone) or as a TE (to connect to the telco)? |
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12:22.08 | rolers | tzafrir_laptop: I want to use the first card as NT and the first card works fine. |
12:22.23 | rolers | I want to use the second card as TE. |
12:22.55 | tzafrir_laptop | If you use that card as NT, better write in the span line span=1,0,<whatever> |
12:23.27 | tzafrir_laptop | So that card will provide timing to the remote side |
12:24.42 | rolers | tzafrir_laptop: ok I changed it. |
12:24.55 | rolers | Why doesn't the second card in TE-mode work? |
12:25.41 | tzafrir_laptop | can you try a loopback? |
12:25.50 | tzafrir_laptop | between the NT and TE cards? |
12:26.47 | rolers | How to do that? |
12:27.42 | tzafrir_laptop | Connect them |
12:28.33 | rolers | okay I'll do. But I have to restart asterisk? |
12:32.10 | tzafrir_laptop | actually, no |
12:32.31 | tzafrir_laptop | Just pretend that the NT is your telco :-) |
12:33.03 | tzafrir_laptop | But if you changed zaptel.conf, you do need to re-run ztcfg |
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12:37.24 | rolers | tzafrir_laptop: okay I've got a loopback. But TE-mode don't seems to work |
12:37.32 | rolers | *doesn't |
12:38.08 | tzafrir_laptop | doesn't seem to work == ? |
12:38.15 | rolers | Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) |
12:39.07 | rolers | I've got two extenstions in the local-context: |
12:39.07 | rolers | exten => 35,1,Dial(ZAP/g2/85) |
12:39.07 | rolers | exten => 36,1,Dial(ZAP/g1/86) |
12:39.24 | tzafrir_laptop | what is the output of: asterisk -rx 'zap show channels' and what is in your zapata.conf ? |
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12:41.58 | phlack | morning |
12:42.10 | phlack | i was just wondering could someone give me a quick help with my asterisk install |
12:42.18 | phlack | its hanging every time i do a reload |
12:42.51 | [netman] | I bet it's a bad zaptel configuration |
12:42.57 | phlack | i think you could be right |
12:43.18 | phlack | but unfortunately i don't know enough about it, I have no zaptel hardware installed but i installed the drivers for completeness |
12:43.43 | [netman] | don't install drivers for hardware u haven't |
12:43.47 | rolers | tzafrir_laptop: http://rafb.net/p/KiTFcg80.html |
12:43.59 | phlack | it was just for future proofing because i intend on getting the hardware |
12:44.14 | phlack | i just noticed the banner at the top about Contos 5.2 issues |
12:44.24 | phlack | ~centos52bug |
12:44.25 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
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12:44.47 | phlack | would that cause it? |
12:44.53 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
12:45.40 | Mike8861 | hello everyone |
12:45.54 | tzafrir_laptop | rolers, but what is in your zapata.conf? (for the groups) |
12:46.10 | phlack | just a second ill check |
12:47.05 | tzafrir_laptop | phlack, that specific bug is a build issue. If you successfully built zaptel it won't byte you at runtime |
12:47.21 | phlack | there is nothing enabled in the groups of zapata.conf |
12:47.30 | rolers | tzafrir_laptop: zapata.conf: http://rafb.net/p/WuYOA271.html |
12:47.40 | tzafrir_laptop | And it has now been fixed in SVN (the above bug report includes a patch) |
12:47.50 | phlack | ah ok |
12:47.57 | phlack | i compiled from source though |
12:51.26 | phlack | == Using TOS bits 0 |
12:51.34 | phlack | its hanging here during reload |
12:52.44 | phlack | == Using TOS bits 0 |
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12:52.51 | phlack | oh sorry didnt mean to paste that twice |
12:55.39 | phlack | that seems to be part of the MGCP |
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13:11.52 | icenicola | hello. how can i do auto provisioning in asterisk? what do i need? |
13:12.55 | Mike8861 | icenicola: do u mean auto provisioning a hard phone ? |
13:13.40 | icenicola | i want to plug a sip device and it will take configurtions from my server that runs asterisk |
13:15.08 | Mike8861 | well, i have not done it from asterisk. if i remember correctly |
13:15.09 | icenicola | can i auto provision a sip device? |
13:15.27 | Mike8861 | u can only provision a sip device that support auto provision |
13:15.32 | Mike8861 | which is usually a hardphone |
13:16.15 | Mike8861 | polycom, aastra, snom supports auto provision |
13:16.32 | icenicola | what about linksys and batm? |
13:16.51 | icenicola | i could search them |
13:16.53 | Mike8861 | check with the manufacturer if it support |
13:16.59 | icenicola | can u tell me please how to do it in asterisk? |
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13:18.02 | vk4akp | Anyone interseted in helping me with a compile error? |
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13:22.11 | trafim | Hi guys. Can anyone help me with my prob? I have dialplan like: incoming -> voicemenu (here customer chooses one of the languages) -> queue-decision (day and night queue for each language) -> queue. And now my task is to make member in night queue for every language to change every night (every night different guy will answer). |
13:22.16 | trafim | the only way i can think of is to create a bunch of queues for every phone numbe |
13:22.19 | trafim | r, and huge pack of gotofitime's, but i feel it's not efficient way. i'm not an |
13:22.22 | trafim | asterisk expert though, so i maybe missing the point. |
13:22.29 | trafim | aww, linebreaks. sorry. |
13:24.12 | trafim | anyway, can't figure out how to implement it without such a headache.. can anyone give an advice? |
13:24.26 | tzafrir_laptop | florz, ping |
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13:45.57 | vk4akp | make[3]: asnparser: Command not found |
13:46.02 | vk4akp | Any idea's? :( |
13:46.38 | riddlebox | vk4akp, what distro |
13:46.47 | vk4akp | Sabayon |
13:46.50 | lmadsen | trafim: I'd use a Local channel which then checks in your local AstDB (or you could hard code it into the dialplan too) which does the check to determine who the call should go to. |
13:47.35 | lmadsen | trafim: so your queue would have a single member, and that member would be a Local channel, then inside the Local channel configuration (which is just a context inside the dialplan), you could add your GotoIfTime()'s, or whatever other method you want. |
13:47.51 | lmadsen | trafim: just like you do your night and day menu's for the IVR, you would do the same kind of idea |
13:48.07 | riddlebox | vk4akp, http://www.google.com/search?hl=en&q=make%5B3%5D%3A+asnparser%3A+Command+not+found+asterisk&btnG=vkSearch |
13:48.40 | phlack | is it possible to run asterisk, im only using voip without zaptel and libpri or do i still need the ztdummy module for timing? |
13:49.00 | riddlebox | phlack, I did that for a long time |
13:49.00 | vk4akp | no |
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13:49.14 | vk4akp | you only need the ztdummy for the meetme conference. |
13:49.19 | riddlebox | phlack, you need ztdummy for conference |
13:49.27 | phlack | yeah i want conference so im gonna need it |
13:49.43 | riddlebox | phlack, yup |
13:51.33 | vk4akp | Looks like my problem is the new OS. |
13:51.42 | riddlebox | ok which codec uses the smallest bandwith, ulaw,alaw,or g.726? |
13:51.45 | phlack | so during the make menuselect, i can disable everything apart from the ztdummy |
13:51.52 | trafim | lmadsen: tnx, i thought i'll get no response at all. i'll go read about these local chan's now. |
13:51.56 | phlack | and still allow conferencing |
13:52.00 | vk4akp | G.729 |
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13:52.22 | vk4akp | G,729, followed by G.726 |
13:52.26 | riddlebox | vk4akp, dont you have to pay for that codec? |
13:52.56 | vk4akp | I believe it's free now for non commercial use. |
13:54.02 | coppice | its free for nothing |
13:57.15 | riddlebox | hrmm well I have been using u-law here at home for almost 2 years and havent had any issue, so I guess I will keep it |
13:59.56 | coppice | if u-law gives you no trouble don't change. it will most certainly give you better quality than G.726 or G.729 |
14:00.10 | trafim | lmadsen: and, if it won't disturb you, what would example of calling this local chan from queue look like? like, in queues.conf, there must be something like "member => Local/s@night/n" and somewhere in dialplan context [night] full of gotoiftime+dial? i got it correctly? |
14:11.05 | riddlebox | coppice, yeah thats kind of what i am thinking |
14:12.22 | coppice | wideband voice is far better, but the only broad support for it is in Skype |
14:14.50 | lmadsen | trafim: you got it pretty much, but abstract the context name a bit more: [queue_members], then member => Local/s@queue_members/n, then in [queue_members] you just determine what time it is, such as with GotoIfTime(), then you can have a bunch of GotoIfTime()'s that go to an extension in queue_members such as 'bob', 'nancy', 'george' depending on when you want to distribute calls to those queue members |
14:15.34 | lmadsen | so the 's' extension is just the check to determine which extension to actually send the call to based on your time requirements, then an extension like 'george' might be: exten => george,1,Dial(SIP/george) |
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14:16.14 | lmadsen | russellb: morning! |
14:16.31 | lmadsen | riddlebox: don't fix what ain't broke |
14:17.19 | trafim | lmadsen: thanks alot. |
14:17.38 | lmadsen | trafim: no problem, that should give you enough information to get going |
14:17.59 | lmadsen | trafim: thanks for asking a well phrased question :) |
14:18.26 | trafim | lmadsen: :) |
14:22.07 | riddlebox | lmadsen, thats how i feel abot it too |
14:22.12 | lmadsen | :) |
14:22.14 | *** join/#asterisk ManxPower (n=manxpowe@17.sub-75-203-110.myvzw.com) |
14:22.22 | lmadsen | all happy admins live by that mantra :) |
14:29.30 | coppice | I thought happy admins achieved that state by being fault tolerant |
14:30.24 | ManxPower | i thought admins chieved that state by killing the problem users? |
14:30.40 | ManxPower | and achieved too! |
14:31.12 | coppice | that sounds fault intolerant. not good for the karma |
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14:37.12 | ManxPower | It's always the user's fault! But I'll tolerate it. 8-) |
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14:38.59 | riddlebox | lol |
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14:40.24 | riddlebox | is the fax extension built in now, or do you have to add some 3rd party software |
14:42.25 | outtolunc | wonders if i drink some coffee this channel will appear to move faster <G> |
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14:46.36 | coppice | drink syrup of figs and everything will move faster |
15:03.46 | outtolunc | mmm coffee |
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15:06.15 | riddlebox | speaking of coffee, I need breakfast |
15:06.34 | coppice | at 11PM? :-\ |
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15:07.29 | mwalling | not everyone is on your side of the world... |
15:07.41 | mwalling | Sun Jun 29 11:07:41 EDT 2008 |
15:08.50 | coppice | well 11AM is also a little off for breakfast |
15:08.59 | outtolunc | only 8am hee |
15:09.03 | outtolunc | er here |
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15:14.41 | mwalling | coppice: never heard of brunch? |
15:16.12 | vk4akp | is ther ea newer svn release other then 1.4 ? |
15:19.18 | russellb | asterisk/trunk is the most bleeding edge |
15:19.23 | russellb | there is also asterisk/branches/1.6.0 |
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15:31.04 | vk4akp | Can I get 1.6 using SVN ? |
15:31.23 | vk4akp | I'm having trouble getting 1.4 to compile using a newer distro. |
15:31.45 | vk4akp | seems to be a common problem across lnewer linux distro's. |
15:31.56 | vk4akp | asnparser: doesn't exist. |
15:32.00 | [TK]D-Fender | vk4akp: just download off digium.com |
15:32.17 | [TK]D-Fender | vk4akp: And which distor would that be? |
15:32.24 | [TK]D-Fender | distro* |
15:32.24 | vk4akp | Sabayon. |
15:32.32 | vk4akp | But google shows others havign the same problem. |
15:32.38 | [TK]D-Fender | vk4akp: And whats the problem? |
15:33.02 | vk4akp | make[3]: asnparser: Command not found |
15:33.37 | vk4akp | asnparser -m CISCO_H225 -c cisco-h225.asn |
15:33.37 | vk4akp | make[3]: asnparser: Command not found |
15:33.52 | [TK]D-Fender | "If your version of OpenH323 requires ASNParser, then it is over a year out of date." <--- |
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15:34.11 | vk4akp | thats 1.4 |
15:34.22 | vk4akp | Asterisk 1.4 needs it |
15:35.00 | [TK]D-Fender | vk4akp: Only for chan_h323 apparently |
15:35.00 | mwalling | wow.. a gentoo distribution using old software? |
15:35.10 | mwalling | /o\ |
15:35.12 | [TK]D-Fender | vk4akp: So unselect it in menuconfig |
15:35.20 | vk4akp | hugh? |
15:35.29 | [TK]D-Fender | "make menuconfig" |
15:35.40 | vk4akp | You will have to dumb this down for me. I am no linux guru unfortunatly. |
15:35.40 | [TK]D-Fender | unselect h323. |
15:36.09 | vk4akp | I haven't had to do any of this b4 on older releases. Why has it become aproblem now? |
15:36.32 | [TK]D-Fender | vk4akp: What versions does this problem appear on? |
15:36.42 | vk4akp | How do I get this asnparser into the OS instead? |
15:37.06 | vk4akp | Sabayon 3.5 loop 3, with Asterisk SVN install of 1.4 |
15:37.28 | [TK]D-Fender | vk4akp: What other versions of *? |
15:37.45 | [TK]D-Fender | vk4akp: And don't jsut say SVN. EVERY versions is available through SVN |
15:37.46 | vk4akp | I don't know. |
15:37.52 | vk4akp | I am trying to install 1.4 |
15:37.59 | vk4akp | Its what I was using with Sabayon 3.4f |
15:38.07 | [TK]D-Fender | vk4akp: Stop using SVN and work of digium's download site |
15:38.15 | vk4akp | why |
15:38.22 | vk4akp | svn was simple last time. |
15:38.34 | mwalling | [TK]D-Fender: hes a gentoo user... have patience |
15:38.39 | [TK]D-Fender | vk4akp: Because you can't track down where the problem is when you can't even tell what version of * you're running. |
15:39.05 | [TK]D-Fender | genpoo |
15:39.11 | [TK]D-Fender | lights up the torches... |
15:39.27 | Mike8861 | HI [TK]D-Fender |
15:40.02 | mwalling | fwiw, * builds *perfectly* on slackware ;) |
15:40.06 | [TK]D-Fender | vk4akp: If you aren't a linux guru you should stick to a popularly supported distro. |
15:40.21 | vk4akp | http://www.asterisk.org/developers/get-source <<< This is what I've always used b4. |
15:40.21 | [TK]D-Fender | mwalling: Yup, Slack "just works" |
15:40.34 | mwalling | vk4akp: are you a developer? |
15:40.38 | [TK]D-Fender | vk4akp: Good, now stop. |
15:40.45 | vk4akp | what? |
15:40.51 | vk4akp | What the hell R U talking about. |
15:41.09 | [TK]D-Fender | vk4akp: http://www.asterisk.org/downloads |
15:41.20 | mwalling | welcome to irc, which is not aim. we spell words out here |
15:41.25 | [TK]D-Fender | vk4akp: SVN is the latest "to the minute" including the latest BUGS <- |
15:41.47 | [TK]D-Fender | vk4akp: FUCK SVN. Get over it, get off you ass and create a CONSISTENT working environment. |
15:41.49 | vk4akp | They are working on 1.6 |
15:41.56 | vk4akp | why would 1.4 now be a problem? |
15:42.04 | [TK]D-Fender | vk4akp: 1.4 is a SERIES! |
15:42.09 | vk4akp | I've installed it plenty of time sthis way b4. |
15:42.13 | [TK]D-Fender | vk4akp: not a specific versioN! |
15:42.23 | [TK]D-Fender | vk4akp: And its not working, so stop whining. |
15:42.36 | mwalling | b-e-f-o-r-e |
15:42.38 | vk4akp | Whining. Amazing. |
15:42.43 | Mike8861 | [TK]D-Fender , how do i get video ivr working on asterisk ? |
15:42.50 | [TK]D-Fender | Mike8861: no idea. |
15:42.50 | vk4akp | Trying to get around a problem is whining. OK right. |
15:43.03 | [TK]D-Fender | vk4akp: You don't even know what release you are working on. |
15:43.03 | mwalling | vk4akp: you're creating a problem for yourself |
15:43.08 | Mike8861 | [TK]D-Fender are u serious ? |
15:43.17 | mwalling | y-o-u |
15:43.19 | vk4akp | how can I know. |
15:43.20 | [TK]D-Fender | Mike8861: Yes. Who uses video for ivr? |
15:43.29 | russellb | video IVR is the same as audio IVR, actually |
15:43.36 | [TK]D-Fender | vk4akp: If you don't know you shouldn't be using SVN. |
15:43.38 | russellb | as long as the files are formats that asteirsk supports ... |
15:43.46 | russellb | and the caller has video support |
15:43.57 | russellb | works well .. |
15:44.06 | Mike8861 | i saw video on youtube, some taiwan people got nokia E61 with video IVR working |
15:44.12 | [TK]D-Fender | russellb: will * play multiple streams on mathing the video codec as well? |
15:44.18 | russellb | yes |
15:44.33 | russellb | it's ... magic |
15:44.47 | [TK]D-Fender | russellb: neato. How would you record both audio & video simultaneously for playback like that? |
15:44.58 | russellb | same, with magic |
15:45.09 | russellb | just using the Record app ... or any other recording stuff in asterisk |
15:45.10 | [TK]D-Fender | russellb: Sorry, all out of goats to sacrifice :) |
15:45.18 | russellb | calling into voicemail from a video phone will record audio and video |
15:45.24 | russellb | calling back in will play them both back |
15:45.26 | [TK]D-Fender | russellb: thought record only recorded a single format |
15:45.26 | russellb | video voicemail ftw |
15:45.41 | russellb | ok, the Record app may not work ... |
15:45.46 | russellb | but i know voicemail does |
15:46.01 | Mike8861 | can i post the youtube link over here ? |
15:46.02 | [TK]D-Fender | russellb: So use VM as a recording mechanism & move them out for IVR usage? |
15:46.07 | Mike8861 | it demo about vid ivr |
15:46.08 | russellb | ha |
15:46.14 | russellb | if record doesn't work, we should make it work |
15:46.17 | russellb | i haven't tried in forever |
15:46.18 | russellb | i don't remember |
15:46.38 | russellb | just checked, Record() supports video |
15:46.48 | Mike8861 | well, i have magic passion, i will make this magic to happen! |
15:46.52 | [TK]D-Fender | russellb: just the I recall Record using the file extension to specify the format which to me looks like it implies only a single format can be recorded at a time. |
15:47.09 | russellb | well, that's because we support audio transcoding |
15:47.19 | russellb | we don't support video transcoding, if there is a video stream, it will just record it in the native format |
15:47.21 | Mike8861 | @russellb: which AGI script do i need to use to push video ?? |
15:47.33 | russellb | um |
15:47.44 | [TK]D-Fender | russellb: and it will record it accordingly, transcoding audio if thats the case? |
15:47.50 | Mike8861 | @russellb i planning to push it in h.263 |
15:47.54 | russellb | nods |
15:48.06 | russellb | Mike8861: whatever AGI script you write, i guess? |
15:48.10 | [TK]D-Fender | russellb: I should try it sometime then. Also looking to go 1.6 shortly. |
15:48.17 | russellb | [TK]D-Fender: brave man :) |
15:48.27 | [TK]D-Fender | russellb: jsut at home. Little to lose |
15:48.29 | seanbright | [TK]D-Fender: i'd recommend sticking with 1.2 |
15:48.32 | seanbright | ducks |
15:48.40 | [TK]D-Fender | seanbright: that'd require DOWNGRADING ;) |
15:48.46 | russellb | ~fist seanbright |
15:48.46 | jbot | ACTION uses seanbright as a handpuppet |
15:48.49 | seanbright | haha |
15:48.52 | seanbright | nooooooooooooooo! |
15:48.55 | [TK]D-Fender | russellb: that looked... wrong... |
15:49.04 | russellb | ~punch seanbright |
15:49.05 | jbot | ACTION lets fly with a wild haymaker which catches seanbright right on the nose |
15:49.08 | russellb | that's what i meant! |
15:49.09 | russellb | :-p |
15:49.13 | seanbright | [TK]D-Fender: i made that unfortunate discovery last week |
15:49.13 | [TK]D-Fender | ... |
15:49.23 | seanbright | [TK]D-Fender: tried ~fist fight Qwell |
15:49.28 | seanbright | and all hell broke loose |
15:49.39 | Mike8861 | example : |
15:49.39 | Mike8861 | <PROTECTED> |
15:49.40 | Mike8861 | <PROTECTED> |
15:49.49 | Mike8861 | i need to have a script to play back video |
15:50.04 | Mike8861 | whichh command does the job ? |
15:50.05 | russellb | if you also have tt-weasels.h263, then it will work |
15:50.08 | seanbright | none of the apps support video playback i don't believe |
15:50.14 | russellb | stares at seanbright |
15:50.15 | seanbright | or i am lying |
15:50.21 | seanbright | i am lying. |
15:50.23 | seanbright | heh |
15:50.32 | [TK]D-Fender | Mike8861: apparently "playback" alone will do if you have a "tt-weasels.h263" from what russellb just said |
15:50.40 | russellb | yes |
15:50.51 | [TK]D-Fender | seanbright: ignorance != lying |
15:51.00 | russellb | touche. |
15:52.39 | seanbright | goes back to something else |
15:53.37 | Mike8861 | @russellb thanks |
15:53.47 | russellb | sure |
15:53.51 | russellb | seanbright: no! |
15:53.57 | russellb | jbot: make seanbright work on asterisk |
15:53.58 | jbot | make: *** No rule to make target `seanbright work on asterisk'. Stop. |
15:54.07 | russellb | heh |
15:54.09 | seanbright | is working on hoard |
15:54.15 | russellb | seanbright: w00t |
15:54.21 | russellb | jbot: make love |
15:54.22 | jbot | make: *** No rule to make target `love'. Stop. |
15:54.29 | russellb | jbot: make |
15:54.40 | seanbright | jbot: make war |
15:54.41 | jbot | make: *** No rule to make target `war'. Stop. |
15:54.44 | seanbright | ah well |
15:54.48 | russellb | i see. |
15:55.00 | seanbright | jbot: make something for god's sake |
15:55.01 | jbot | make: *** No rule to make target `something for god's sake'. Stop. |
15:55.10 | russellb | jbot: i hate you |
15:55.10 | jbot | You hate you? |
15:55.20 | russellb | jbot: die |
15:55.21 | jbot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
15:55.21 | seanbright | jbot: you hate you |
15:55.27 | Mike8861 | maybe video ivr is good for sex hotline |
15:55.36 | Mike8861 | it can increase sales ??? |
15:55.58 | seanbright | Mike8861: have you *seen* phone sex operators? |
15:56.30 | Mike8861 | seanbright: nope. |
15:56.47 | Corydon76-dig | There's a good reason why they're on the phone and not in a strip club |
15:56.50 | Mike8861 | seanbright: are they IVR or nerv network ?? |
15:57.00 | seanbright | has no idea |
15:57.12 | seanbright | has yet to resort to such measures |
15:57.14 | seanbright | :) |
15:57.17 | JayTee52 | picture a 400lb woman wearing spandex and a pound and a half of makeup with green eye shadow |
15:57.22 | coppice | ignorance is truly bliss in this case |
15:57.26 | Mike8861 | seanbright: i see |
15:57.29 | seanbright | holding a baby |
15:57.31 | seanbright | and ironing |
15:57.41 | seanbright | and smoking |
15:57.45 | JayTee52 | while smoking a cheroot |
16:01.48 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
16:03.30 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
16:06.32 | Mike8861 | hello Ebola |
16:06.53 | *** join/#asterisk smach (n=smach@bas15-ottawa23-1279684007.dsl.bell.ca) |
16:06.58 | Ebola | Hi |
16:07.18 | Mike8861 | its so quiet over here |
16:07.50 | Ebola | It being Sunday will have something to do with it. |
16:09.21 | Mike8861 | yeah, its monday midnight over here |
16:09.31 | Mike8861 | are u living in the US ? |
16:10.09 | hi365 | (how) can i use [zonemessages] (in voicemail) to set the default zone settings w/o having to add the settings to every mailbox? |
16:11.06 | Corydon76-dig | hi365: set tz=foo in the [general] section |
16:11.14 | hi365 | thanks |
16:12.29 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
16:16.14 | *** join/#asterisk mgdm (n=michael@serenity.mgdm.net) |
16:17.27 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
16:20.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
16:25.39 | Mike8861 | [TK]D-Fender: i wanna ask about Playback() |
16:25.56 | Mike8861 | exten => 123,2,Playback(tt-weasels) <-- the parameter means path to file ?? |
16:29.22 | gr0mit | Mike8861, yes |
16:29.22 | JayTee52 | yep, that's the path to the file allright and it's even documented in the book! |
16:36.03 | Mike8861 | it doesnt works. i got some error message from console |
16:36.10 | Mike8861 | <PROTECTED> |
16:36.26 | Mike8861 | can anyone help me to debug ? |
16:37.28 | [TK]D-Fender | Mike8861: maybe you should prove the files are THERE |
16:38.17 | Qwell | Mike8861: how is that an error? |
16:38.49 | Mike8861 | please hold on, i am pasting extension.conf to pastebin. |
16:38.59 | [TK]D-Fender | Mike8861: NO |
16:38.59 | Mike8861 | this is my first attempt to modify extension.conf |
16:39.15 | Mike8861 | [TK]D-Fender: No ?? |
16:39.23 | [TK]D-Fender | Mike8861: maybe you should prove the files are THERE <-- |
16:40.06 | Mike8861 | [TK]D-Fender: i am using a build-in blue claim river file, and the file is there, i proved it by "moh show files" |
16:40.26 | [TK]D-Fender | Mike8861: those aren't in the same place. |
16:40.35 | Mike8861 | [TK]D-Fender: oh.... |
16:40.39 | Qwell | also... |
16:40.44 | Qwell | ~freepbx |
16:40.45 | jbot | [~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 |
16:40.58 | [TK]D-Fender | Mike8861: go look in your lib folder under sounds and verify that the ones you are playing back are there |
16:41.10 | Qwell | Surely you aren't using freepbx...are you? |
16:41.15 | [TK]D-Fender | Qwell: He may have gone independant, so I was going to leave that alone. |
16:41.51 | Mike8861 | @Qwell, i got freepbx, but i am trying to do this without the help of gui |
16:42.11 | Qwell | the next time you use the gui, it's going to overwrite everything you've just done |
16:42.13 | [TK]D-Fender | Mike8861: You're frightfully close to "burn in Hell" territory. |
16:43.12 | Mike8861 | i will set it to permission to 666, so freepbx wont write this ? |
16:43.33 | [TK]D-Fender | Mike8861: You have no clue. Good luck with all that... |
16:49.43 | Mike8861 | [TK]D-Fender: I have changed the file, but its still not working to "exten => 123,2,Playback(you-sound-cute) " |
16:50.07 | [TK]D-Fender | Mike8861: and you haven't proved the file is there. |
16:50.30 | Mike8861 | [TK]D-Fender: you-sound-cute is under /var/lib/asterisk/sounds |
16:50.50 | [TK]D-Fender | Mike8861: PASTEBIN <---- and show us the ERRO (full CLI output) |
16:51.06 | [TK]D-Fender | Mike8861: And show us the file itself with mermissions, etc |
16:51.44 | Mike8861 | [TK]D-Fender: thanks, please hold, i takes some time |
16:57.41 | Mike8861 | [TK]D-Fender: the files are ready |
16:57.48 | Mike8861 | http://www.pastebin.sk/en/7225/ http://www.pastebin.sk/en/7224/ http://www.pastebin.sk/en/7223/ |
16:59.48 | [TK]D-Fender | Mike8861: BLIND |
17:00.00 | [TK]D-Fender | Mike8861: that isn't even executing the same code as your show from your dialplan. |
17:00.09 | [TK]D-Fender | Mike8861: -- Executing [123@from-internal:4] Playback("SIP/1025-b77c6b70", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack |
17:00.24 | [TK]D-Fender | Mike8861: Clearly whatever you are looking at in that other pastebin is NOT getting used at all |
17:00.53 | Mike8861 | what does that mean excatly, i am dialing from a extension 1025 |
17:01.00 | *** join/#asterisk af_ (n=getsmart@88-149-241-204.dynamic.ngi.it) |
17:01.17 | [TK]D-Fender | Mike8861: http://www.pastebin.sk/en/7224/ <-- this code is not being exewcuted at all |
17:01.42 | [TK]D-Fender | Mike8861: and the fact you think "but I diead it from extension 1025" matters just showes you have no idea what's going on. |
17:02.02 | [TK]D-Fender | Mike8861: http://www.pastebin.sk/en/7225/ <-- look at whats being CALLED <- |
17:02.17 | [TK]D-Fender | Mike8861: Nothing at all alike. |
17:02.22 | mwalling | !book |
17:02.24 | mwalling | ~book |
17:02.25 | jbot | 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://tfot.leifmadsen.com, or see ~buybook |
17:02.31 | mwalling | Mike8861: reaqd that, start fresh |
17:02.39 | [TK]D-Fender | mwalling: He's on FreePBX. There is no hope |
17:02.49 | Mike8861 | [TK]D-Fender: it has been called to s@macro-hangupcall:1] ???!!! |
17:03.13 | mwalling | [TK]D-Fender: if i can figure out how to build a svn-1.5.0 package for ubuntu, he can figure out how to write a dialplan ;) |
17:03.17 | Mike8861 | [TK]D-Fender: to solved this, i need to modify extension_addition.conf ? |
17:03.27 | [TK]D-Fender | Mike8861: Listen up. You made an exten (123) to try and play back thqat recording you mentioned. its not getting USED |
17:03.45 | [TK]D-Fender | Mike8861: FreePBX is not supported here and you have no Clue. this is no the channel for you. |
17:04.26 | Mike8861 | [TK]D-Fender: i am not using FreePBX to made this code, i swear i code it manually |
17:04.56 | [TK]D-Fender | Mike8861: yes and its NOT GETTING CALLED. You have no idea where to put it or how to set up FreePBX to get it EXECUTED. |
17:05.13 | [TK]D-Fender | Mike8861: And we are not here to figure out how to use that garbage to teach you. |
17:09.23 | JayTee52 | changes channel topic of #freepbx to "Abandon hope, all ye who enter here!" |
17:11.05 | mwalling | isnt it "dial here"? |
17:11.29 | ManxPower | how about just "Abandon all hope!"? |
17:17.50 | gr0mit | maybe we should have bot that looks for freepbx |
17:18.17 | gr0mit | any mention of the word should initiate a tyrade of messages |
17:18.33 | gr0mit | 'not supported - abandon hope, You Are On Your Own, etc! |
17:18.34 | [TK]D-Fender | gr0mit: Or maybe we should just KB users who don't get the picture and continue to press on in here.. |
17:18.39 | gr0mit | hehe |
17:21.02 | *** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com) |
17:21.55 | gr0mit | has had Bad Experiences with customers who insist on using Freepbx |
17:22.39 | gr0mit | to the extent that i will not touch any box that runs anything other than native Asterisk |
17:24.50 | [TK]D-Fender | gr0mit: I'm very close to that point myself. I make some exceptions for certain paid clients. |
17:25.09 | [TK]D-Fender | paying* |
17:33.37 | gr0mit | has had a client who has not paid |
17:34.02 | gr0mit | i took the project on before i realised it was freepbx |
17:34.13 | gr0mit | told him to abandon it if he wanted my help |
17:34.50 | gr0mit | persuaded me to help anyway - project was a disaster - he never got paid, and thus I never got paid. |
17:34.54 | gr0mit | grrr. |
17:35.23 | gr0mit | two lessons i learned . |
17:35.38 | gr0mit | 1) Avoid freepbx like the plague. |
17:35.49 | gr0mit | 2) Demand payment in advance. |
17:36.49 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
17:38.09 | gr0mit | wonders if the mention of the plague rattled Ebola's cage! |
17:38.10 | [TK]D-Fender | gr0mit: Words to live by. I do a quick free log-in inspection first, then quote. Payment upon initiation of work. |
17:38.31 | [TK]D-Fender | gr0mit: Never a dissatisfied customer. |
17:39.00 | gr0mit | do you stop work when the money runs out? |
17:39.13 | gr0mit | quoted by the hour |
17:39.55 | gr0mit | next time i will quote for the job |
17:40.23 | gr0mit | the project in question involved freepbx installs in 3 countries |
17:40.30 | gr0mit | shudders |
17:40.51 | *** join/#asterisk Wiedi (n=wiedi@newton-air.w.fruky.net) |
17:41.28 | *** join/#asterisk s519 (n=steve@87-194-151-213.bethere.co.uk) |
17:41.44 | [TK]D-Fender | gr0mit: I usually job quote, and it turns our rather accurate. Its always stated that if things go quicker that benifit gets passed on as follow-up support. |
17:42.24 | [TK]D-Fender | gr0mit: I discount certain "exploratory" work if it will expand my knowledge even if its likely to exceed my typical hourly rate. |
17:42.33 | [TK]D-Fender | (within reason) |
17:42.50 | gr0mit | well, the issue involved a lot of fiddling trying to bypass voip-blocking in certain ahem, Arab countries. |
17:46.24 | [TK]D-Fender | gr0mit: Yeah we've got a few people here who are trying they best to get themselves arrested :) |
18:00.16 | styelz | is there any benefit in using the -s command line option for asterisk ? |
18:00.39 | styelz | if you have zap hardware? |
18:01.34 | [TK]D-Fender | styelz: I don't even see that option listed. |
18:01.43 | styelz | sorry i mean -I |
18:02.03 | styelz | had s on the brain |
18:02.20 | styelz | <PROTECTED> |
18:02.43 | styelz | should i be using this if i have zap hardware ? |
18:02.49 | [TK]D-Fender | styelz: Never needed it personally... it "just works" |
18:02.56 | [TK]D-Fender | (without) |
18:03.00 | styelz | ok cool.. i pretend i didnt see it |
18:08.12 | [TK]D-Fender | ok, out for a while... |
18:26.14 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
18:31.47 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
18:39.27 | *** join/#asterisk James|TCC (i=James@87-194-161-247.bethere.co.uk) |
18:39.56 | James|TCC | does anyone know ho i could go about a prank call system in asterisk - sounds strange i know lol |
18:40.20 | James|TCC | basically we want to be able to from an IRC bot cause a call to go out to my mobile, ring for 20 secs, then hang up |
18:40.53 | James|TCC | its easy to do if i have an extension, but without makingn our irc bot a SIP compliant extension, im not sure how to proceed |
18:40.57 | plik | shell scripts and call files by the sound of things |
18:41.20 | James|TCC | oh and another thing, the irc bot is not on the same machine as the asterisk |
18:41.45 | plik | slightly more complexshells scripts and ssh keys |
18:42.03 | James|TCC | the bot is php, so SOAP is my method of choice, |
18:42.16 | James|TCC | do you know of any tutorials on making calls from a webpage, as its not far different |
18:42.42 | plik | google does, the book will help with call file syntax |
18:43.29 | James|TCC | ah cool |
18:48.37 | styelz | heh |
18:49.27 | styelz | er wrong square |
18:49.31 | plik | James|TCC: thinking about it, if you're using php you could prolly do something in AGI - again see the book for details |
18:49.50 | James|TCC | AGI? |
18:51.05 | plik | Asterisk Gateway Interface |
18:51.15 | plik | ~book |
18:51.15 | jbot | 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://tfot.leifmadsen.com, or see ~buybook |
18:51.24 | plik | or voip-info.org |
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18:54.06 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
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19:00.18 | *** join/#asterisk VoipMasta (n=fabio@201.139.138.246.cable.dyn.cableonline.com.mx) |
19:00.40 | *** part/#asterisk VoipMasta (n=fabio@201.139.138.246.cable.dyn.cableonline.com.mx) |
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19:10.30 | *** join/#asterisk adr3nalin3 (n=plasma@ip68-13-89-102.om.om.cox.net) |
19:12.38 | adr3nalin3 | hi guys, I am trying to verify that I have correctly installed a digium TDM800p. When I run ztcfg -vvvv I get 8 channels to configure at the bottom. Is there something else I need to do? Also when I go into the asterisk CLI and type 'zap show channels' all I see is a psuedo channel. |
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19:17.53 | *** part/#asterisk smurf (n=smurf@debian/developer/smurf) |
19:21.28 | styelz | adr3nalin3: try running genzaptelconf |
19:22.03 | styelz | then restart zaptel |
19:24.51 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
19:32.32 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
19:37.31 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
19:38.28 | adr3nalin3 | styelz: I have done so, but I am getting the same output |
19:39.47 | styelz | do you see anything in /proc/zaptel ? |
19:40.00 | styelz | and have /dev/zap ? |
19:40.23 | styelz | maybe cat /proc/zaptel/1 |
19:41.16 | adr3nalin3 | styelz: Yes when I cat /proc/zaptel/1 it shows 8 channels "1 WCTDM/0/0 FXSKS |
19:41.41 | adr3nalin3 | and I yes I do have /dev/zap |
19:42.21 | styelz | and they are set in /etc/zaptel.conf ? |
19:42.35 | styelz | k |
19:42.57 | adr3nalin3 | such as fxsks=1 fxsks=2 and so on? |
19:43.00 | adr3nalin3 | yes |
19:43.07 | styelz | what about /etc/asterisk/zapata-auto.conf |
19:43.16 | styelz | or is it zapata-channels.conf |
19:43.24 | styelz | are yo including the correct file |
19:43.33 | styelz | and are they set in there |
19:44.28 | styelz | you may have #include zapata-channels.conf in zapata.conf |
19:44.42 | tzafrir_laptop | ls -l /etc/asterisk/zapata*.conf |
19:45.14 | adr3nalin3 | the zapata-channels.conf looks like it is all setup |
19:45.18 | styelz | ok |
19:45.29 | styelz | does zapata.conf have #include zapata-channels.conf ? |
19:45.32 | styelz | it needs it |
19:45.37 | adr3nalin3 | checking |
19:45.41 | styelz | at the end somewhere |
19:45.48 | styelz | possibly |
19:46.34 | styelz | that should do it |
19:49.12 | adr3nalin3 | I have added that but I still get the same output. I have to go for now. thank you for all your help |
19:49.22 | styelz | ok good luck |
20:00.04 | mchou | anyone here ever subscribe to voipjet? |
20:05.17 | *** join/#asterisk xenonex (n=xenonex@89.218.236.247) |
20:08.33 | Strom_C | mchou: years ago |
20:09.29 | mchou | Strom_C |
20:09.45 | mchou | Strom_C: Do they give you a SIP address? |
20:10.17 | mchou | Strom_C: I couldn't find this on their pages..... |
20:10.37 | Strom_C | a SIP address for what purpose? |
20:11.12 | mchou | Strom_C: so ppl can call in to my SIP address (as an example) |
20:11.22 | Strom_C | ... |
20:11.29 | mchou | as opposed to a bona fide DID # |
20:11.29 | Strom_C | you don't need an ITSP for that |
20:12.29 | mchou | Strom_C: That's like saying I dont need the telco to get a phone # |
20:12.34 | Strom_C | no, wrong |
20:12.40 | Strom_C | you have an IP address |
20:12.59 | Strom_C | you can very easily have people call your box DIRECTLY |
20:13.28 | Strom_C | the whole point of the ITSP is for PSTN interconnect |
20:13.40 | Strom_C | if you just want people to call you directly via SIP, you don't need the ITSP |
20:14.03 | mchou | this is not enterprise level stuff....IPaddr is dynamic (via ISP) |
20:14.23 | Strom_C | did I ever claim this was enterprise-level stuff? :) |
20:14.29 | mchou | and dont really want to be messing with dyndns and crap |
20:15.06 | Strom_C | no ITSP that I know of is going to provide you with an address like that |
20:15.44 | Strom_C | if you want to allow unauthenticated SIP calls to your PBX, you must do that yourself, either via dyndns (or similar) or some box with a static IP address |
20:25.57 | *** join/#asterisk nirz (n=nir@bzq-79-177-133-58.red.bezeqint.net) |
20:27.28 | nirz | hello, i'm trying to set an iax trunk between asterisk appliance and other asterisk 1.2, from some reason i having problem to send the caller id ... do i need to make any modification because of the appliance ? |
20:51.47 | *** join/#asterisk Pctech37|Mac (n=Pctech37@unaffiliated/pctech37) |
20:52.13 | Pctech37|Mac | Can I join the project? |
20:55.36 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
20:55.50 | *** join/#asterisk RoyK (n=roy@ip-194-54-149-91.dialup.ice.no) |
20:56.17 | variable_office | I am trying to get chan_mobile to work with lg vx5800(chocolate) but it is not working; one thing I noticed is that it is coming up as a "handset" |
20:56.22 | variable_office | err headset |
20:56.44 | variable_office | and usable=no |
21:09.21 | variable_office | has anyone here even gotten chan_mobile to work? |
21:17.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
21:18.20 | florz | tzafrir_laptop: pong |
21:27.40 | *** join/#asterisk qwertyfcuker (n=psetti1@ool-18bd9928.dyn.optonline.net) |
21:28.09 | qwertyfcuker | whats up folks, what's the good word on asterisk 1.6? |
21:28.15 | qwertyfcuker | need SIP over TCP! |
21:28.37 | implicit | use openSER u can use it as a proxy to 1.4 |
21:28.51 | qwertyfcuker | yeah yeah i know the workarounds :) |
21:28.55 | implicit | tls, tcp, etc |
21:29.00 | implicit | i only use asterisk as a media server |
21:29.18 | implicit | do all routing, accounting, sdp mangling, normalization in openser |
21:29.20 | qwertyfcuker | you use openSER for telephony? |
21:29.26 | implicit | absolutel |
21:29.40 | implicit | i can do 10,000 calls per second on that thing |
21:30.02 | qwertyfcuker | wow |
21:30.12 | implicit | probably the most powerful sip proxy ever, its more like a SIP scripting language |
21:30.13 | qwertyfcuker | you done microsoft UM integration? |
21:30.16 | implicit | yeah |
21:30.18 | implicit | i have |
21:30.30 | implicit | but you will need another component too if you want to do REFERs properly |
21:30.33 | implicit | b2bua |
21:30.39 | qwertyfcuker | k |
21:30.45 | qwertyfcuker | how is the MS UM voicemail btw? |
21:30.57 | implicit | i don't know ive just done integration on a consulting basis |
21:30.58 | qwertyfcuker | asterisk's VM is absolutely awful |
21:31.08 | implicit | i wrote my own VM in python |
21:31.14 | qwertyfcuker | nice |
21:31.16 | implicit | and tied it into SEMS |
21:31.19 | qwertyfcuker | using tts? |
21:31.31 | implicit | u seems sems? |
21:31.35 | implicit | iptel.org/sems |
21:31.40 | qwertyfcuker | lemme have a look |
21:31.42 | implicit | but it's not really good if your not a programmer |
21:31.52 | implicit | asterisk is a much easier to use media server and for general purpose asterisk is good |
21:32.16 | implicit | i jsut don't use it for SIP routing cause it's not a sip proxy. it doesn't make sense to make new dialogs and deal w/ media etc when doing sip routing if you don't have to |
21:32.17 | qwertyfcuker | i'm a decent programmer - just getting into the telephony/voip side of things now |
21:32.42 | qwertyfcuker | there was some Mono AGI thingie that caught my eye the other day and I started playing w/ asterisk |
21:32.48 | implicit | ah cool |
21:32.53 | implicit | what are you trying to develop |
21:33.09 | qwertyfcuker | basically a call in timeclock solution for home health aides |
21:33.17 | qwertyfcuker | and any other type of industry really |
21:33.22 | qwertyfcuker | temp workers, etc |
21:33.28 | implicit | oh ok |
21:33.41 | qwertyfcuker | call from a location from a range of numbers, verify the # and clock the punches in and out |
21:33.51 | qwertyfcuker | then pump out a file for payroll |
21:33.57 | qwertyfcuker | pretty basic really |
21:33.58 | implicit | how are you integrating to PSTN? |
21:34.22 | qwertyfcuker | i've been playing with a sangoma a200 card w/ 4 lines in a hunt group |
21:34.34 | implicit | easier to just get it over SIP, isn't it? |
21:34.36 | qwertyfcuker | thinking about maybe using a SIP trunk provider instead |
21:34.39 | implicit | yeah |
21:34.40 | qwertyfcuker | definately cheaper |
21:34.47 | implicit | yep |
21:34.51 | qwertyfcuker | the POTS call quality seems much better though |
21:34.54 | implicit | do you ened a lot of phone #s? |
21:34.59 | qwertyfcuker | don't think so really |
21:35.19 | qwertyfcuker | SIP would suit my needs just fine |
21:35.35 | implicit | for 100 contiguous or so i can give you $50/mo |
21:36.04 | qwertyfcuker | wow that's probably more than i'd need for the foreseeable future |
21:36.05 | qwertyfcuker | lol |
21:36.09 | implicit | lol |
21:36.10 | qwertyfcuker | but a great deal |
21:36.15 | Strom_C | qwerty: your SIP calls will sound like PSTN quality if you use the right codecs |
21:36.38 | implicit | Strom_C: absolutely, if not better cause of impedence and grounding isseus of some hardware |
21:37.04 | qwertyfcuker | g729? |
21:37.12 | Strom_C | qwertyfcuker: ugh, no |
21:37.13 | implicit | g729 is toll quality |
21:37.15 | Strom_C | g711u |
21:37.15 | implicit | for sure |
21:37.20 | Strom_C | no, g729 is ass quality |
21:37.25 | qwertyfcuker | lol |
21:37.29 | implicit | Strom_C, absolutely not |
21:37.41 | Strom_C | implicit: it sounds like shit to me |
21:37.41 | qwertyfcuker | ulaw is probably fine |
21:37.46 | Strom_C | I even hate talking on my mobile phone |
21:37.51 | qwertyfcuker | this is true.... |
21:37.52 | implicit | what is your setup with it though |
21:38.03 | qwertyfcuker | basically employees would call a number to clock in |
21:38.04 | implicit | cause it soudns WAY better than my gsm phone |
21:38.06 | *** join/#asterisk dFence (n=chatzill@ings-d93226b8.pool.mediaWays.net) |
21:38.13 | qwertyfcuker | it's not like they have to have amazing call quality anyway |
21:38.22 | qwertyfcuker | they're not calling 1900spankalot |
21:38.39 | implicit | if you have other issues, like resizing rtp frames, messed up jitter buffers etc |
21:38.42 | implicit | yeah it will sound like ass |
21:38.58 | qwertyfcuker | well what would matter would be if IVR didn't work |
21:39.02 | Strom_C | implicit: heh, any lossy codec sounds like crap over the phone to me |
21:39.08 | implicit | otherwise g729 has a mos score of 4.1 |
21:39.12 | implicit | versus 4.4 for g711 |
21:39.23 | implicit | also disable VAD |
21:39.26 | Strom_C | that's because most people are ignorant :) |
21:39.28 | implicit | that's the biggest issue w/ bad quality of g729 |
21:39.41 | implicit | bad VAD implementations |
21:40.02 | qwertyfcuker | is sorry for starting a codec pissing contest |
21:40.02 | Strom_C | I'm just particularly picky about the subtleties of what it does to the audio |
21:40.02 | implicit | then don't use g711 |
21:40.02 | implicit | use g722 |
21:40.02 | implicit | haha |
21:40.02 | implicit | and call ip-ip only |
21:40.06 | implicit | so that you don't get cut off at 4khz |
21:40.09 | Strom_C | sigh...missing the point |
21:40.13 | *** join/#asterisk magic_hat (n=geoffdou@h-68-164-10-43.chcgilgm.dynamic.covad.net) |
21:40.35 | implicit | perhaps, but g729 itself especially if you do IP-IP is extremely good |
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21:41.09 | implicit | if your going to PSTN it's really hard to isolate the isseus to say what it really is |
21:41.15 | magic_hat | howdy folks. I'm seeing two entries in the logs for an autodialing app that I have. One is SIP response 503, service unavailable. The other is CONGESTION... everyone is busy. What's going on with these? |
21:41.20 | implicit | if you have the bandwidth then use g711 |
21:41.20 | Strom_C | heh, no, even just running it over my local LAN from my polycom phone to my asterisk box, I don't like the sound of g729 |
21:41.34 | implicit | try it without your * box |
21:41.39 | implicit | * sometimes resizes frames etc |
21:41.40 | Strom_C | tried that too. |
21:41.41 | implicit | makes it sound ass |
21:41.56 | implicit | have rtp go point-to-point |
21:42.20 | Strom_C | look, i know you think g729 is the holy grail of voice codecs, but I've tried it dozens of different ways and I don't like the vocoding. |
21:42.28 | implicit | i never relay RTP when unnecessary, i let it go point to point as much as possible |
21:42.52 | implicit | it's not that, it's just that people blame audio issues more on codecs than other problems that are generally teh real sources |
21:43.03 | qwertyfcuker | heh, i spoke to the folks at bandwidth.com and they told me the same thing implicit |
21:43.07 | Strom_C | I understand what you're arguing. |
21:43.12 | Strom_C | I don't like the vocoders. |
21:43.14 | qwertyfcuker | i was looking at a solution that involved MPLS and lots of NATing |
21:43.36 | qwertyfcuker | and they said it was better just to use a level 3 ISP w/ a T1 and let it go over the public internet to them |
21:44.10 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
21:44.35 | magic_hat | Everyone is busy/congested at this time (1:0/1/0)... does this mean my VOIP is busy? I've exceeded my allotted number of channels? Or the phone is off the hook o the other end? |
21:44.47 | Strom_C | magic_hat: congested circuit |
21:45.15 | magic_hat | Strom_C: Anything I can do about that? Or it's a problem at Teliax? |
21:45.32 | Strom_C | increase your timeout |
21:45.44 | Strom_C | it could be that teliax is just slow to respond |
21:45.52 | magic_hat | ah, okay. |
21:46.27 | beek | FWIW, I just placed a call through Teliax with no problem. |
21:46.51 | magic_hat | beek: Yeah, it's generally working. Just wondering what the occasional error messages mean. |
21:47.43 | qwertyfcuker | hah I just got my father in law to take my wife to sex and the city......PWNED |
21:48.17 | qwertyfcuker | nn fellas |
21:48.26 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
21:48.42 | magic_hat | Strom_C where do I set that timeout.... not seeing it in sip.conf |
21:49.22 | seanbright | is it just me, or does reading some of the long threads on the -users list make anyone else want to stab themselves in the eyes? |
21:56.23 | jblack | I'm not inclined to blame teliax. I've never seen a problem with them. |
21:58.26 | magic_hat | so... if not them, where's the congestion? |
21:59.22 | jblack | bad number, the number is busy? |
21:59.35 | jblack | Misconfiguration of your system? |
22:00.03 | magic_hat | jblack: system's working fine on most calls. But you're saying I can get an 'everyone is busy' if it's a bad number? |
22:00.05 | jblack | Bump up your verbose and debug settings on the console, and watch the call try to place. |
22:00.39 | jblack | pastebin it if you like |
22:04.17 | magic_hat | I'm doing random-digit dialing, so I know a bunch of #'s are going to come back bad. I just wasn't sure if the 'congestion' notice was that, or something else. |
22:05.38 | Strom_C | random-digit dialing? |
22:06.19 | magic_hat | yeah, for public opinion polling. take a list of valid exchanges, add a random 4 digits on the end, and you have a random sample. |
22:06.52 | Strom_C | I hope to God you're not calling mobile phones and people on the do not call list |
22:07.09 | magic_hat | there's an exemption for public opinion research. |
22:07.13 | implicit | it doesn't count as telemarketing |
22:07.17 | magic_hat | and we're a nonprofit, and exempt because of that, too. lol |
22:07.38 | Strom_C | die in a fire :) |
22:07.48 | jblack | I hope you get syphillis. |
22:08.29 | magic_hat | lol. we're doing some really important work that's in the public interest. and I already have syphilis :) |
22:08.41 | jblack | Then I hope you get herpes too. |
22:08.58 | magic_hat | we'll make sure you're on the call list! |
22:10.51 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
22:10.52 | Strom_C | I always answer the opinion polls out of curiosity, and I've NEVER answered one that wasn't full of ambiguous, badly-written, and loaded questions. |
22:11.17 | jblack | strom: Don't answer opinion polls. They establish a business relationship that can be used to call back and telemarket with. |
22:12.06 | Strom_C | jblack: the last one I received was three telephone numbers ago :) |
22:13.24 | Strom_C | but yeah |
22:13.35 | Strom_C | the opinion polls are always badly-written and terribly biased |
22:13.40 | *** join/#asterisk Fuzix (n=fuzix@250-118.citynet.ftth.internl.net) |
22:13.41 | Strom_C | so fuck that shit |
22:14.48 | jblack | Heh, you think opinion polls are used to survey what people think? |
22:14.54 | Strom_C | of course not |
22:15.01 | Strom_C | i'm not that naive :) |
22:16.29 | jblack | some day I'm going to make a survey, and call every mcdonald's in the land with one question "Are you comfortable with your participation in feeding america unhealthy food" |
22:17.49 | Strom_C | heh |
22:18.44 | jblack | Then, I'll follow up by calling every red cross with "Are you happy to personally profit from the practice of selling for $600 a pint blood that people gave you for free?" |
22:21.15 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
22:21.15 | *** mode/#asterisk [+o russellb] by ChanServ |
22:21.20 | *** join/#asterisk ^^Johnny5 (n=JeanCote@dsl-67-55-16-142.acanac.net) |
22:21.49 | JayTee52 | "wouldn't you like to be a Pepper too?" |
22:22.07 | magic_hat | Strom_C: we're actually pretty good at writing them to find out how people think about policy issues without biasing them. But yeah, there's a lot o'crap out there. |
22:22.48 | Strom_C | magic_hat: I'll reserve judgment until I actually read one of your surveys |
22:23.19 | magic_hat | lol if I hadn't busted the code that runs the dial-in number, I'd give it to you. |
22:23.39 | ^^Johnny5 | anyone know how to debug a digium tdm400P board with 4 pots (green) interface daughter boards? |
22:24.04 | ^^Johnny5 | was working and does not anymore...rrr |
22:24.07 | JayTee52 | the green modules are called FXS modules |
22:24.11 | Strom_C | ^^Johnny5: what's the actual problem you're having? |
22:24.12 | ^^Johnny5 | yes |
22:24.49 | Strom_C | magic_hat: i said "read," not "call" :) |
22:25.28 | ^^Johnny5 | the same boards (2x) were purchased in jan 08 and all of a sudden no dial tones on my pots anymore... I go in asterisk there is no zap* commands there... zapta.conf registers 1 board only rrr |
22:26.52 | Strom_C | ^^Johnny5: pastebin your zapata.conf and zaptel,conf files |
22:26.54 | ^^Johnny5 | I have setup a ssh route to the box from jcdirect.no-ip.org (my home) |
22:26.56 | russellb | if there are no zap commands, then chan_zap didn't get loaded, or failed to initialize ... |
22:27.03 | Strom_C | ~pb |
22:27.04 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
22:27.04 | russellb | zaptel itself probably isn't properly configured |
22:27.24 | implicit | or even better, dpaste |
22:27.33 | implicit | dpaste.com is faster than all of them |
22:27.40 | Strom_C | congratulations |
22:27.41 | implicit | it takes only 3.3ms to process |
22:27.53 | ^^Johnny5 | I am currently on a Win XP Pro using opera |
22:28.02 | implicit | congratulations |
22:28.05 | Strom_C | ^^Johnny5: just pastebin please |
22:28.58 | russellb | my pastebin is faster than your pastebin |
22:29.20 | Strom_C | i've got the fastest pastebin in six counties |
22:31.29 | *** join/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
22:31.36 | ^^Johnny5 | # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit |
22:31.36 | ^^Johnny5 | # Zaptel Configuration File |
22:31.36 | ^^Johnny5 | # |
22:31.36 | ^^Johnny5 | # This file is parsed by the Zaptel Configurator, ztcfg |
22:31.36 | ^^Johnny5 | # |
22:31.36 | ^^Johnny5 | # It must be in the module loading order |
22:31.38 | ^^Johnny5 | # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
22:31.40 | ^^Johnny5 | fxoks=1 |
22:31.42 | ^^Johnny5 | fxoks=2 |
22:31.43 | Strom_C | oh christ |
22:31.44 | ^^Johnny5 | fxoks=3 |
22:31.44 | Strom_C | PASTEBIN |
22:31.46 | Strom_C | not paste |
22:31.46 | ^^Johnny5 | fxoks=4 |
22:31.48 | ^^Johnny5 | # Global data |
22:31.50 | ^^Johnny5 | loadzone= us |
22:31.52 | Strom_C | russellb: kick plz |
22:31.52 | ^^Johnny5 | defaultzone= us |
22:32.06 | implicit | lol jut let him do it |
22:32.08 | implicit | it's alright |
22:32.19 | Strom_C | ^^Johnny5: PASTEBIN PASTEBIN PASTEBIN |
22:32.21 | Strom_C | ~pb |
22:32.22 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
22:32.27 | mchou | lol |
22:32.40 | implicit | http://www.dpaste.com |
22:32.44 | Strom_C | draws a circle on the whiteboard and bangs his head into the centre of it |
22:32.47 | ^^Johnny5 | never used pastebin I am there I must create an account ? |
22:32.50 | implicit | no |
22:32.53 | implicit | please just go to the site |
22:32.54 | implicit | and SEEEEEEEEEEEEEE |
22:33.09 | implicit | SEE |
22:34.04 | ^^Johnny5 | juste pasted in dpaste |
22:34.11 | implicit | ok now give the link to us |
22:34.25 | ^^Johnny5 | http://dpaste.com/hold/59834/ |
22:34.31 | implicit | perfect |
22:34.36 | implicit | but we already saw this in the channels |
22:34.42 | implicit | so we didn't need to see it agan |
22:34.53 | implicit | *channel |
22:35.32 | Strom_C | ^^Johnny5: ok -- now what happens when you run "ztcfg -vv" at the bash prompt? |
22:35.34 | ^^Johnny5 | I don't understand.... |
22:35.44 | implicit | ^^Johnny5: you are a nice guy, don't be scared |
22:35.58 | ^^Johnny5 | do you see the zaptel.conf contents in dpaste? |
22:36.07 | Veggen | Johnny: yes, we do. |
22:36.15 | ^^Johnny5 | how can I check/debug my board... |
22:36.27 | Strom_C | ^^Johnny5: follow my instructions |
22:36.34 | Strom_C | ^^Johnny5: what happens when you run "ztcfg -vv" at the bash prompt? |
22:37.34 | ^^Johnny5 | http://dpaste.com/59835/ |
22:39.07 | Strom_C | restart asterisk and see if your "zap" CLI command shows up |
22:39.07 | ^^Johnny5 | Besides this problem the PBX is working fine with XLite... |
22:39.20 | ^^Johnny5 | ok I will try |
22:39.26 | Strom_C | just asterisk |
22:39.28 | Strom_C | not the whole box |
22:39.35 | ^^Johnny5 | how is that done |
22:39.47 | Strom_C | connect to the asterisk CLI and type "restart now" |
22:39.50 | ^^Johnny5 | asterisk -r stop now |
22:41.51 | ^^Johnny5 | http://dpaste.com/59836/ |
22:41.58 | ^^Johnny5 | no zap* |
22:42.27 | Strom_C | pastebin your zapata.conf |
22:42.44 | adr3nalin3 | this problem above is the same problem I have with a tdm800p |
22:42.52 | adr3nalin3 | jumps in |
22:43.06 | Strom_C | adr3nalin3: wait till I fix this one |
22:43.24 | ^^Johnny5 | http://dpaste.com/59837/ |
22:43.42 | adr3nalin3 | im just gonna watch for now |
22:43.56 | Strom_C | ^^Johnny5: uhm...are you using trixbox? |
22:45.27 | russellb | i could have guessed that a long time ago :-p |
22:45.32 | ^^Johnny5 | yes stable ver 2.4 |
22:45.38 | russellb | ~trixbox |
22:45.39 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
22:46.14 | ^^Johnny5 | do youi suggest I use asterisk.... |
22:47.42 | russellb | almost anything but trixbox |
22:47.44 | russellb | but yes |
22:47.57 | russellb | using asterisk directly is just about all you'll get help with in this channel |
22:48.08 | ^^Johnny5 | can you tell me why? |
22:48.20 | russellb | ~trixbox |
22:48.21 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
22:48.21 | *** join/#asterisk d-k-t (n=dt@125.120.132.230) |
22:49.17 | ^^Johnny5 | yes I understand that part... |
22:49.20 | beek | ^^Johnny5: My first * box was Trixbox. It took me longer to figure out how to get it to do what I wanted (or to customize it) than it did just to load a fresh install of CentOS, then Asterisk, and configure it. |
22:49.47 | ^^Johnny5 | oh yeah well that hits me!! |
22:49.54 | ^^Johnny5 | thanks... |
22:51.15 | ^^Johnny5 | if I want to go asterisk then what about web interface like freePBX... is it ok? |
22:51.35 | Strom_C | no |
22:51.37 | Strom_C | ~freepbx |
22:51.38 | jbot | [~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 |
22:51.44 | beek | That's where things get complicated. |
22:51.55 | ^^Johnny5 | ok, thanks |
22:51.58 | Strom_C | learn to write your own dialplan |
22:52.00 | Strom_C | ~book |
22:52.01 | jbot | 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://tfot.leifmadsen.com, or see ~buybook |
22:52.29 | ^^Johnny5 | thanks for the pdf... and links |
22:52.30 | beek | freepbx is attractive because it seems to be a quick way to a working system but the code that it writes is spaghetti code. |
22:52.42 | beek | It's a bitch to work on. |
22:52.49 | Strom_C | spaghetti is a nice way of describing that mess |
22:53.56 | ^^Johnny5 | lol |
22:54.38 | beek | The other thing Johnny is that by learning to configure Asterisk by hand gives you the opportunity to load it on other platforms not supported by Trix/FreePBX, such |
22:54.50 | beek | as the little Asus router I have running * at home. |
22:55.58 | adr3nalin3 | could someone give me a hand with a problem similar to ^^Johnny5's? I am using a TDM800P. It appears that my channels are not configured. |
22:56.20 | Strom_C | adr3nalin3: pastebin your zaptel.conf and your zapata.conf |
22:59.26 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
22:59.55 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
23:00.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
23:00.45 | adr3nalin3 | Strom_C: zapata.conf, http://pastebin.com/m5fcf9562 The only thing I have changed in there is I added #include zapata-channels.conf |
23:02.11 | Strom_C | ok, and where are the contents of zapata-channels.conf? |
23:03.08 | *** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com) |
23:03.35 | adr3nalin3 | here --> http://pastebin.com/m31661c55 |
23:04.25 | Strom_C | ok, and what about zaptel.conf? |
23:05.02 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
23:06.44 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
23:06.49 | adr3nalin3 | zaptel.conf -> http://pastebin.com/m56f0907f |
23:07.19 | Strom_C | um, no |
23:07.30 | Strom_C | well, i suppose that will work |
23:07.50 | adr3nalin3 | gathered from reading http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
23:07.53 | Strom_C | but, just for simplicity's sake, change that to "fxsks=1-8" |
23:08.00 | adr3nalin3 | ok |
23:08.21 | Strom_C | just to clarify, you've got FXO (red) modules, right? |
23:08.28 | adr3nalin3 | correct |
23:09.12 | Strom_C | what happens when you run "ztcfg -vv"? |
23:09.55 | adr3nalin3 | http://pastebin.com/m6f812208 |
23:10.11 | Strom_C | ok |
23:10.14 | Strom_C | now restart asterisk |
23:10.47 | adr3nalin3 | ok |
23:10.58 | adr3nalin3 | done |
23:11.12 | Strom_C | does it appear to work now? |
23:11.21 | adr3nalin3 | same output |
23:11.27 | adr3nalin3 | ^^ from ztcfg -vv |
23:11.28 | Strom_C | ? |
23:11.38 | Strom_C | that response makes no sense |
23:11.46 | adr3nalin3 | I know... |
23:11.59 | Strom_C | now that you've restarted ASTERISK, do your channels appear to work now? |
23:12.14 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:12.53 | ^^Johnny5 | do you type in help in the asterisk prompt? asterisk -r , help |
23:13.07 | adr3nalin3 | they are all showing just as they did in the previous ztcfg -vv pastebin but it still says 8 channels to configure. Is that the expected output? I was expecting 8 channels configured |
23:13.18 | Strom_C | adr3nalin3: so wait |
23:13.24 | Strom_C | you didnt actually test them? |
23:13.33 | Strom_C | and just assumed from the output that it was broken? |
23:13.39 | Strom_C | TEST THEM |
23:13.41 | *** part/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
23:14.26 | ^^Johnny5 | how? |
23:14.45 | adr3nalin3 | the box is 150 miles away. I am trying to make sure it is ready to go when I arrive next week |
23:15.07 | Strom_C | adr3nalin3: um, ok? |
23:15.21 | Strom_C | log into asterisk and type "zap show channels" |
23:15.45 | ^^Johnny5 | do you have zap* commands when you type help? |
23:15.54 | adr3nalin3 | Strom_C: they DO show up now! |
23:15.59 | adr3nalin3 | thank you! |
23:16.06 | adr3nalin3 | they did not previously |
23:16.18 | ^^Johnny5 | thanks guys!! |
23:16.56 | ^^Johnny5 | it seems I have faulty boards since they were working for 6months at least... thunder! |
23:17.20 | Strom_C | ^^Johnny5: or freepbx fucked up. |
23:17.29 | Strom_C | I would blame trixbox before I blame the hardware |
23:17.34 | ^^Johnny5 | is there a way to check the boards... |
23:18.45 | ^^Johnny5 | I have used another hard disk and redone the config as previously... in jan 08 ... on one of the lines I here a tone very low.... with white noise |
23:19.26 | Strom_C | ^^Johnny5: you're using trixbox |
23:19.45 | Strom_C | that in and of itself is probably at least 97% of the problem |
23:20.48 | ^^Johnny5 | well it was working for a while... no auto updates either... this happened after they told me that thunder striked about 500 meters (3x=ft) away |
23:21.11 | Strom_C | ...you honestly think I'm so ignorant that I don't know what a meter is? |
23:21.23 | ^^Johnny5 | hum sorry |
23:21.53 | ^^Johnny5 | no offence in mind just respect... |
23:22.03 | Strom_C | ... |
23:22.12 | Strom_C | yeah, I believe that one </sarcasm> |
23:22.20 | Strom_C | call digium support, although they're not going to be much help if you're running trixbox |
23:22.21 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
23:22.42 | *** join/#asterisk xlogik (n=xlogik@c-71-232-176-24.hsd1.ma.comcast.net) |
23:23.08 | ^^Johnny5 | ok, then I will install what they require me to install right as per their web site... all dev stuff ? |
23:23.23 | Strom_C | no, you don't need to install the dev stuff |
23:23.40 | Strom_C | just go with whatever linux distro you like best and then a release version of asterisk/zaptel |
23:23.58 | ^^Johnny5 | which one do you sugggest... |
23:24.38 | ^^Johnny5 | linux distro |
23:24.47 | Strom_C | I like debian |
23:24.57 | Strom_C | but personal preference will dictate which one you choose |
23:26.16 | ^^Johnny5 | I have fedora 8 working on another machine here |
23:26.25 | ^^Johnny5 | is that ok? |
23:26.32 | Strom_C | are you listening to me? |
23:26.34 | Strom_C | IT DOESN'T MATTER |
23:26.39 | Strom_C | use whatever you like best |
23:26.47 | ^^Johnny5 | thanks... |
23:27.56 | adr3nalin3 | ~snom |
23:27.57 | jbot | rumour has it, snom is like all German products. High quality, but wacky engineering. :) |
23:29.35 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
23:31.54 | *** join/#asterisk moos3_ (n=richard@cpe-204-210-72-206.maine.res.rr.com) |
23:32.04 | moos3_ | is this a good card http://store.pbxeq.com/store/product.php?productid=17104&cat=283&page=1 |
23:32.38 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
23:34.24 | Strom_C | moos3_: i'd advise you not buy that one |
23:34.38 | JayTee52 | it's a clone knockoff of a Digium card. |
23:35.00 | moos3_ | Strom_C: what do you recommend then |
23:36.06 | Strom_C | moos3_: a Digium TDM410 |
23:36.36 | moos3_ | Strom_C: I would have i had that kind of cash |
23:36.58 | adr3nalin3 | moos3_: e4 technologies has digium cheap |
23:37.05 | Strom_C | lol |
23:37.13 | Strom_C | ancient crappy digium shit for cheap, maybe |
23:37.19 | *** join/#asterisk levi (n=user@166.70.39.219) |
23:37.20 | JayTee52 | buying cheap hardware is the easiest way to mess up your system |
23:37.41 | adr3nalin3 | Strom_C: all the stuff I have gotten from them has been up to date and worked great |
23:37.46 | moos3_ | yeah but I have to voip four offices for less then 1k |
23:37.53 | *** part/#asterisk levi (n=user@166.70.39.219) |
23:37.57 | adr3nalin3 | moos3_: good luck with that |
23:37.58 | moos3_ | thats for all the hardware |
23:38.00 | JayTee52 | what moron decided on 1K as a budget? |
23:38.04 | Strom_C | moos3_: lololololololololololololololololololol |
23:38.30 | adr3nalin3 | moos3_: I fucking hope that doesn't include phones... |
23:38.43 | moos3_ | Strom_C: i have read stuff tht zapmicro cards work great |
23:38.47 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
23:39.01 | moos3_ | no phones dont come out of the IT budget |
23:39.02 | moos3_ | :) |
23:39.06 | JayTee52 | I've read Lord of the Rings but I don't believe in hobbits |
23:39.13 | adr3nalin3 | I wouldn't go with anything except digium, I bought sangoma one time and returned it the next day. |
23:39.33 | adr3nalin3 | the drivers wouldn't even install |
23:41.01 | jblack | moos3_: On a budget of 1K, you'll have to go with voip. |
23:41.40 | Strom_C | on a budget of 1k, I'll give you some cans and string |
23:41.43 | adr3nalin3 | moo3_: sip trunking? |
23:42.28 | moos3_ | yeah |
23:42.30 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
23:42.43 | adr3nalin3 | moos3_: Do you already have the machines to use? 4 boxes will be > $1k |
23:42.54 | JayTee52 | I thought there wasn't any such thing as a sip trunk |
23:43.18 | adr3nalin3 | qwest has it, they call it IPLD. |
23:44.00 | moos3_ | yeah I have the boxes in place because they are old poweredges that we use to for staging |
23:44.21 | jblack | moos3_: Then blow the $1k on 4 dedicated dsl lines, and a voip provider. |
23:46.15 | moos3_ | i would do that i'm required by law to have dedicated hardlines |
23:46.46 | adr3nalin3 | moos3_: for 911> |
23:46.48 | adr3nalin3 | ? |
23:47.47 | Strom_C | moos3_: then ask for a budget increase, because you don't want to run your business on some shit clone of a discontinued card |
23:48.11 | adr3nalin3 | moos3_: or get one pstn and a red analog phone per office |
23:48.22 | moos3_ | lol |
23:48.43 | moos3_ | so the only aterisk cards that will work are nothing but digium |
23:49.05 | adr3nalin3 | digium wrote asterisk. |
23:49.22 | moos3_ | if there isn't any other way but digium then they will have to give me the increase |
23:49.33 | adr3nalin3 | I would be willing to bet their cards work the best |
23:49.53 | moos3_ | but if there is other cards that will work and my boss finds out my ass is gonna get crewed |
23:49.54 | Strom_C | moos3_: trust me -- being unreasonably cheap now will cause you no end of headache and wasted money in the future |
23:50.02 | jblack | I've heard sangoma cards are the best. |
23:50.28 | jblack | The rhini I "have" works also fine, now that it's set up. |
23:50.32 | jblack | rhino, that is. |
23:50.33 | adr3nalin3 | ^^ except there drivers don't always work. Kind of a big problem |
23:51.01 | jblack | afaik, All of the cards are a disaster if you care about free software. |
23:56.31 | *** join/#asterisk mwalling_ (i=mwalling@you.dontlike.us) |
23:58.12 | LiNeTuX|Home | Rhino cards are a pain in the butt. Even their own tech support has issues getting them working. |