| 00:03.04 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:04.00 | manxpower | ruben23: Jitter is the DIFFERENCE in packet transit time |
| 00:04.28 | manxpower | idespinner: try it and see. other than your invalid Dial syntax. |
| 00:11.11 | Katty | hi |
| 00:13.08 | Katty | HI |
| 00:14.24 | jaytee | hi Katty |
| 00:14.27 | Katty | :> |
| 00:14.31 | Katty | hugs jaytee |
| 00:14.39 | jaytee | hugs Katty |
| 00:15.11 | wdbl | how many simulataneous calls (just talking) can I reasonably expect on a modern quad core xeon box with 8GB of RAM? |
| 00:16.41 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-145-5-54.mia.bellsouth.net) |
| 00:16.57 | Katty | while i don't have any definitive answer for you, it would probably depend on how you plan on getting the calls out to the world. |
| 00:17.15 | wdbl | an SIP VOIP provider |
| 00:20.35 | *** part/#asterisk kbr (~kbr@ASte-Genev-Bois-154-1-111-194.w83-199.abo.wanadoo.fr) |
| 00:25.27 | p3nguin | Yeah, the bandwidth will likely be your limiting factor. |
| 00:27.06 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 00:28.36 | *** join/#asterisk Gestahlt (~chatzilla@HSI-KBW-078-042-054-134.hsi3.kabel-badenwuerttemberg.de) |
| 00:28.41 | Gestahlt | Hi |
| 00:28.50 | Gestahlt | I dont get chan_lcr running with asterisk |
| 00:29.01 | Gestahlt | The module is loaded |
| 00:29.14 | Gestahlt | but i cant see anything in the CLI |
| 00:33.16 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 00:33.25 | *** join/#asterisk Slugs_ (~yeah@adsl-068-157-249-044.sip.pns.bellsouth.net) |
| 00:35.25 | Slugs_ | I've installed a fresh centos 5.4, latest asterisk, and when i try to start asterisk i get get caught in an endless loop of shutting down / restarting |
| 00:37.09 | p3nguin | slugs_: Which version? |
| 00:37.27 | p3nguin | "latest" is not very specific, given all the branches. |
| 00:38.11 | Slugs_ | <PROTECTED> |
| 00:38.12 | Slugs_ | Asterisk 1.6.2.6 |
| 00:39.13 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 00:40.53 | idespinner | manxpower, tried it before |
| 00:40.58 | idespinner | it was a no go |
| 00:41.29 | idespinner | this was a call created by an AMI originate command with extra variables passed.... |
| 00:43.48 | p3nguin | Yay, first day of spring is this Saturday. |
| 00:44.10 | Katty | :>>> |
| 00:44.51 | antiwire | That means mating season is coming up. |
| 00:49.44 | *** join/#asterisk n0cturnal (~n0c@203.161.121.158.static.amnet.net.au) |
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| 00:56.25 | shazaum | someone works with h323? |
| 00:57.51 | shazaum | rtp.c:2760 ast_rtp_raw_write: RTP |
| 00:57.51 | shazaum | Transmission error of packet 21286 to XXX.XXX.XXX.XXX:6064: Invalid argument |
| 00:58.19 | shazaum | someone saw something? |
| 00:58.43 | *** join/#asterisk Gestahlt (~chatzilla@HSI-KBW-078-042-054-134.hsi3.kabel-badenwuerttemberg.de) |
| 00:58.47 | Gestahlt | argh |
| 00:59.16 | manxpower | Slugs_: did you do a "make config" to install the new startup files? |
| 00:59.50 | manxpower | running "asterisk -cvvv" should give you a good idea what is making it blow up if it's not an init script issue. |
| 01:02.43 | Slugs_ | manxpower, would it be 'r' if im sshed? |
| 01:07.39 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 01:09.53 | *** part/#asterisk michaely (~michaely@CPE-124-190-32-229.sdcz1.lon.bigpond.net.au) |
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| 01:31.25 | shazaum | =/ |
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| 01:37.04 | Slugs_ | where is indications.conf suppoed to be located? |
| 01:37.20 | jaytee | /etc/asterisk |
| 01:37.29 | Slugs_ | ty |
| 01:49.48 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 01:50.34 | Deeewayne | salts Slugs_ |
| 01:51.19 | Slugs_ | melts |
| 01:51.40 | *** join/#asterisk slinksh0t (~slinksh0t@wsip-68-227-95-60.pn.at.cox.net) |
| 01:51.41 | Deeewayne | lol |
| 01:52.00 | Deeewayne | that was the most interesting 2 minutes of my day |
| 01:52.00 | Slugs_ | ;) |
| 01:52.20 | Slugs_ | sad day huh? |
| 01:53.22 | Deeewayne | yes, but tomorrow will be better |
| 01:54.41 | Slugs_ | y's that |
| 01:55.03 | Deeewayne | I'm traveling for work and will be home tomorrow |
| 01:55.12 | thehar | go find a hooker |
| 01:55.15 | thehar | that will be more exciting |
| 01:56.07 | *** join/#asterisk fofware (~chatzilla@186.125.41.133) |
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| 01:59.33 | Slugs_ | im travling too |
| 01:59.49 | Slugs_ | gots to love it |
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| 02:05.32 | joako_ | I think I am missing a sound file for the queue applicaiton but the call just drops, there is no message in the CLI. How do I resolve this? |
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| 02:23.59 | Slugs_ | . |
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| 02:34.01 | dlynes | Is there any way to see how many g729 channels you're using? |
| 02:34.03 | *** join/#asterisk fofware (~chatzilla@186.125.41.133) |
| 02:34.12 | dlynes | i.e. from the licensed codec? |
| 02:34.37 | dlynes | It used to be g729 show, but that's an invalid command now...show needs an argument now |
| 02:37.47 | *** join/#asterisk chendy_ (~chatzilla@58.250.10.227) |
| 02:40.22 | idespinner | coreg729 show? |
| 02:40.33 | idespinner | core g729 show*** |
| 02:40.37 | idespinner | just a guess |
| 02:40.43 | idespinner | alot of stuff has moved under core |
| 02:45.23 | ChannelZ | g729 show licenses should show you |
| 02:45.33 | ChannelZ | like "0/0 encoders/decoders of 1 licensed channels are currently in use" |
| 02:47.21 | *** join/#asterisk chendy_ (~chatzilla@119.137.95.48) |
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| 02:52.17 | dlynes | ChannelZ, yeah...that was probably in 1.4 |
| 02:52.31 | ChannelZ | it's in 1.6 |
| 02:52.47 | ChannelZ | 1.6.1 anyhoos |
| 02:53.15 | dlynes | ChannelZ, now i get File: G729-xxxxxxx.lic -- Key: G729-xxxxxx -- Host ID: xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx -- Channels: 30 (OK) |
| 02:53.22 | dlynes | ChannelZ, that's in 1.6.1.14 |
| 02:53.44 | dlynes | ChannelZ, I don't show any encoders/decoders |
| 02:53.58 | ChannelZ | it's the first thing it says, above "Licenses Found:" and what you pasted |
| 02:54.13 | dlynes | Oh...shit....competely missed it |
| 02:54.15 | dlynes | Thanks |
| 02:54.21 | ChannelZ | yeah it's butted up against the prompt :/ |
| 02:54.25 | dlynes | 2/1 encoders/decoders of 30 licensed channels are currently in use |
| 02:54.59 | dlynes | Maybe it was like that before, too, but it was just more noticable before |
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| 03:05.56 | *** join/#asterisk snowboarder04 (~un@obi.bemail.co.uk) |
| 03:07.25 | snowboarder04 | I'm using AsteriskNOW - I have my SIP trunk and extensions configured - I've configured an inbound route to divert all inbound calls to a ringgroup - however calls are only going to one extension (#100) |
| 03:07.34 | snowboarder04 | am I missing a confi step? |
| 03:07.36 | snowboarder04 | *config |
| 03:08.23 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 03:09.37 | xpot-mobile | anyone know what I am missing if .gsm files will play, but .wav will not? |
| 03:09.59 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 03:10.06 | b14ck | I'm trying to store the exit status of a System() command. Any idea how I can do that? The docs say that the result will be stored in SYSTEMSTATUS channel variable, but it doesn't store the exit code, it stores an *interpretation* of the exit code. eg: APPERROR instead of 179. |
| 03:13.01 | [TK]D-Fender | ~freepbx |
| 03:13.02 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:13.05 | [TK]D-Fender | snowboarder04: ^^^ |
| 03:13.13 | *** join/#asterisk sourcode (~code@ppp-58-8-118-184.revip2.asianet.co.th) |
| 03:13.18 | snowboarder04 | [TK]D-Fender: cheers |
| 03:13.52 | dlynes | Anyone come across some magic pill to make voip sound good on ADSL or cable? *winces in pain* |
| 03:14.12 | *** join/#asterisk sourcode (~code@ppp-58-8-118-184.revip2.asianet.co.th) |
| 03:14.14 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 03:14.17 | [TK]D-Fender | dlynes: Medium isn't relevent, and it sounds just fine over each in my experience |
| 03:14.18 | p3nguin | No problems with those here. |
| 03:16.03 | p3nguin | Even on really slow aDSL, VoIP still works just fine... just don't try lots of calls at once. |
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| 03:37.44 | dlynes | Yeah...my problem is it sounds fine until some random time of the day |
| 03:37.56 | dlynes | And it's never reproducable when we're on site |
| 03:38.23 | dlynes | And I'm getting way too many gray hairs... |
| 03:43.05 | b14ck | So, anyone know a way to get an exit code with Asterisk? |
| 03:43.06 | b14ck | :( |
| 03:43.15 | b14ck | I can't really think of a good way to do what I need without this. |
| 03:43.40 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 03:45.46 | [TK]D-Fender | b14ck: make an AGI to do your call and have the return the value |
| 03:46.53 | b14ck | That's not a bad idea, I was kinda hoping to avoid AGI though. |
| 03:47.00 | b14ck | But I guess that is the best way |
| 03:47.27 | hardwire | agi is the bomb |
| 03:47.33 | hardwire | YOU WILL EXPLODE WITH AWESOME |
| 03:47.37 | [TK]D-Fender | take off every zig |
| 03:47.52 | hardwire | doing verizon interop testing today |
| 03:48.00 | hardwire | I made it 1/3 of the way through all their test cases |
| 03:48.14 | b14ck | What happened to the other 2/3 of tests? :x |
| 03:48.28 | hardwire | I did 34 of their test cases today.. the rest of the 2/3 tomorrow |
| 03:48.47 | hardwire | 34 pcaps.. lots of monotony. |
| 03:48.51 | hardwire | anyhoot |
| 03:48.55 | hardwire | I just hope they don't sue us |
| 03:49.05 | hardwire | they are prone to stuff like that |
| 03:49.18 | b14ck | 'you failed test #21, you owe us 75,00$!' |
| 03:49.19 | b14ck | =p |
| 03:49.37 | hardwire | more like "Are you actually making money?" suesuesuesuesue |
| 03:50.42 | b14ck | What business are you in? oO |
| 03:50.59 | hardwire | Wholesale/prepaid/CC at the moment |
| 03:51.05 | b14ck | ah |
| 03:52.48 | hardwire | sigh |
| 03:53.05 | hardwire | most sources don't know that ak has more than one lata |
| 03:53.06 | hardwire | http://www.localcallingguide.com/lca_listexch.php?lata=832 |
| 03:54.42 | hardwire | anybody know where I can get rate center/lata information for the rest of the world? |
| 03:54.45 | hardwire | hehe |
| 03:56.52 | hardwire | I registered opendial.org a long time ago (unrelated to the gnu dialer project) so I could start making a community supported numbering plan database |
| 03:56.57 | hardwire | I need to start in on that |
| 03:57.06 | b14ck | Do you do any web design? |
| 03:57.12 | b14ck | or web development |
| 03:57.14 | hardwire | yeh |
| 03:57.23 | b14ck | what language / framework? |
| 03:57.23 | hardwire | I sort of have to do everything at the moment |
| 03:57.29 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 03:57.35 | b14ck | I hear that. |
| 03:57.40 | b14ck | I'm in the same position =p |
| 03:57.43 | hardwire | python/php and I've been using django and plone mostly over the last year or so |
| 03:57.47 | b14ck | oo |
| 03:57.49 | b14ck | Are you me? |
| 03:57.50 | b14ck | :x |
| 03:57.54 | hardwire | yup |
| 03:58.08 | b14ck | is your name chris? |
| 03:58.12 | hardwire | yup |
| 03:58.21 | b14ck | ... |
| 03:58.42 | b14ck | dude |
| 03:58.49 | b14ck | aim me =p |
| 03:58.50 | hardwire | vot comrade vot?! |
| 03:59.06 | hardwire | i will not aim me. |
| 03:59.17 | b14ck | doesnt show you online :x |
| 03:59.52 | b14ck | Ok, maybe there is some confusion. |
| 04:00.28 | b14ck | my aim: comradeb14ck |
| 04:00.34 | b14ck | send me a ping and i'll add you |
| 04:02.38 | hardwire | erm.. that's my address |
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| 04:55.12 | lmsteffan_ | Where can I get the zaphfc driver that I seem to need in order to use my ISDN card ? |
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| 05:10.28 | hardwire | lmsteffan_: http://blog.flemming.info/?p=51 |
| 05:10.36 | hardwire | lmsteffan_: http://www.voip-info.org/wiki/view/Asterisk+zaphfc |
| 05:11.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rohnsssarujkgmya) |
| 05:20.50 | sawgood | If someone has an Asterisk 1.6.x only box with CentOS 5.4 ... and they needed Asterisk to do 'something' other than what it normally does (they hire a 'programmer') to 'code' this into Asterisk ... In general terms, what type of 'programmer' would this be? |
| 05:20.51 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 05:20.51 | hardwire | sawgood: me |
| 05:21.08 | sawgood | hardwire: excellent ... |
| 05:21.08 | hardwire | "We need a 'hardwire'" |
| 05:21.08 | sawgood | so, what type of 'coding' do you do for Asterisk ... |
| 05:21.09 | sawgood | because I am looking for my on hardwire |
| 05:21.09 | sawgood | ha! |
| 05:21.09 | hardwire | I code the hell out of the Asterisk. |
| 05:21.28 | sawgood | hardwire: So I can talk to my staff about this ... what is the definition of 'code' |
| 05:21.30 | hardwire | Erm.. I do dialplan development and custom modules using AGI |
| 05:21.51 | hardwire | the only code is that it is somewhat cryptic to anybody that doesn't understand "dialplan logic" |
| 05:22.03 | sawgood | hardwire: So, I take it AGI is command line driven? |
| 05:22.11 | hardwire | you would sometimes refer to these people as a telephony integrator |
| 05:22.33 | hardwire | AGi is a scripted interface to the Asterisk dialplan through a myriad of scripting languages like Python and PHP |
| 05:22.43 | hardwire | It is driven by events |
| 05:22.46 | hardwire | like a call |
| 05:22.49 | sawgood | excellent answer ... |
| 05:22.57 | hardwire | AGI not AGi |
| 05:23.09 | sawgood | So, you are skilled Python and PHP? |
| 05:23.16 | hardwire | Like a ninja |
| 05:23.21 | hardwire | hi |
| 05:23.21 | hardwire | ya |
| 05:23.21 | sawgood | where are you located at? |
| 05:23.23 | hardwire | sailor |
| 05:23.27 | hardwire | Alaska :) |
| 05:23.36 | hardwire | buy your own travel tickets. |
| 05:23.37 | sawgood | My sister lived in Alaska for a long time |
| 05:23.52 | hardwire | I'm sorry to hear that.. Polar bears? |
| 05:24.10 | sawgood | Her ex-husband was in the US Army at the time ... stationed there |
| 05:24.16 | sawgood | back in the 1990's |
| 05:24.20 | hardwire | ah.. that's much simpler. |
| 05:25.15 | sawgood | So, other than Phyton and/or PHP ... what other 'languages' does it require to be a good telephony intregator? |
| 05:25.52 | hardwire | Simply having a good concept of how scripting languages operate will be a good start. Python and PHP are very popular.. lets not forget Perl. |
| 05:26.32 | hardwire | They aren't requisite to being somebody who programs an Asterisk dialplan to do things that Asterisk natively provides.. but having "logic" is useful :) |
| 06:12.28 | ChannelZ | heh :) |
| 06:13.09 | ChannelZ | oops |
| 07:00.45 | sawgood | If I have one CentOS box with a 'file' in a directory (which when I do file (filename) ... it tells me the type of file is a ELF 32-bit LSB executable |
| 07:00.57 | sawgood | If I move this file to another CentOS box ... and run it ... 'could it work' on the 2nd box? |
| 07:01.06 | sawgood | How do I 'learn' what scripts or other files this ELF 32-bit LSB executable uses? |
| 07:04.45 | ChannelZ | yeah it could work |
| 07:05.28 | sawgood | ChannelZ: basically the file is a PBX in a Flash process in the /usr/local/sbin directory |
| 07:05.53 | sawgood | I know it has to 'pull information' from other text files .... (it is a status screen in text) when you first log into the box |
| 07:06.06 | sawgood | shows me the Asterisk version, the kernal, and other misc. information in a text box |
| 07:06.09 | sawgood | kind of neat I think |
| 07:06.14 | ChannelZ | You can see an executable requires shared libraries with ldd |
| 07:06.28 | sawgood | ldd (filename)? |
| 07:06.35 | ChannelZ | yeah |
| 07:07.08 | sawgood | linux-gate.so.1 => (0x0052f000) |
| 07:07.10 | sawgood | <PROTECTED> |
| 07:07.14 | sawgood | what do you think this means? |
| 07:07.46 | ChannelZ | It means it's using two shared libraries, linux-gate and libc (which is pretty common) |
| 07:08.15 | sawgood | ok ... I'll try to scp the file from one box to another to see if it works |
| 07:08.40 | ChannelZ | they're kind of virtual libraries |
| 07:09.06 | ChannelZ | (linux-gate is anyway.. libc is actually 'real') |
| 07:09.28 | ChannelZ | So long as the other machine has a compatible version of libc it should generally work |
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| 07:16.53 | sawgood | the program 'ran' on the 2nd machine ... only to say it was a PIAF script not intented to be used on a non PIAF build of CentOS ... |
| 07:16.56 | sawgood | oh well |
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| 07:23.41 | ChannelZ | what is it you were trying to actually do? |
| 07:24.02 | sawgood | The ELF (is that the right term for the file)? |
| 07:24.16 | sawgood | The ELF is called 'status' ... and it is part of .bash_profile |
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| 07:24.59 | sawgood | when you log in ... on the console ... an ASCII text box appears with various build information on Asterisk, FreePBX, the Linux kernel, etc. |
| 07:25.15 | sawgood | just sort of neat ASCII information ... I got use to it, but I'm not using PIAF right now |
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| 09:11.52 | khussein78 | i installed linksys PAP2 ATA but when i press talk on the phone it take around 5 seconds to get line |
| 09:12.34 | khussein78 | what option should i change to immediately get dial tone |
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| 10:01.25 | benngard | khussein78: on "Regional" tab look for "Interdigit Long Timer:" |
| 10:01.52 | benngard | change that to 1 |
| 10:02.13 | benngard | it works at least on spa2102 guess it will do the trick for u to |
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| 10:09.55 | hluesea | <PROTECTED> |
| 10:09.58 | hluesea | how can i do that ? |
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| 10:13.50 | hluesea | anyone here :) |
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| 10:16.23 | c0rnoTa | нÑÑз |
| 10:16.26 | c0rnoTa | yeap |
| 10:17.12 | c0rnoTa | are you want to interrupt asterisk flow? |
| 10:20.19 | pisg | hi, i have problem, this is my extensions.conf and CLI http://pastebin.com/4f0siLvc, problem is i can not call, when i use X-lite and call to my GSM number nothing happen |
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| 10:21.12 | hluesea | c0rnoTa> i actually call a php script via system or like another way, and this php script take some results from other webpages etc. and its waiting the active call |
| 10:21.56 | hluesea | this script does not turned back me any value so i want to execute or work it once and i want to go on the flow |
| 10:22.03 | hluesea | how can i do that ? |
| 10:24.41 | Dovid | psig: the cli shows that its ringing. ur sending the calls to asterisk via h323 ? |
| 10:25.09 | c0rnoTa | hluesea: use AGI |
| 10:25.24 | c0rnoTa | hluesea: write result into variable |
| 10:25.49 | Dovid | look on voip-info.org for info on php+agi |
| 10:25.55 | hluesea | ok i am on it |
| 10:25.57 | Dovid | i have been using it for years |
| 10:26.15 | hluesea | thank you if i have trouble you are here :d |
| 10:26.42 | khussein78 | benngard, i changed it to 3 sec |
| 10:26.45 | c0rnoTa | hluesea: there was no problem :) |
| 10:26.56 | khussein78 | but this affect the time after i dial the number |
| 10:27.12 | pisg | Dovid: i use X-lite, and codec G711 aLaw and G711 ulaw |
| 10:28.00 | c0rnoTa | hluesea: "fputs(STDOUT, "SET VARIABLE 1C_EXTEN $local_ext\n");" |
| 10:28.30 | khussein78 | but when i open the phone it give 5 fast rings then open dialtone to dial the number |
| 10:28.30 | pisg | Dovid: i use realtime sip_buddies and user set context outline1 = exten => _X.,1,Dial(WOOMERA/g0/${EXTEN}) |
| 10:28.54 | hluesea | <c0rnoTa> 1C_EXTEN ? or these command is working |
| 10:29.19 | hluesea | i need a quick solution so i have a php script and it is working smoothly |
| 10:29.44 | hluesea | <c0rnoTa>this input solve the problem ? |
| 10:30.40 | c0rnoTa | 1C_EXTEN - asterisk variable |
| 10:30.56 | c0rnoTa | $local_ext - php variable |
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| 10:37.11 | pisg | Dovid: ? |
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| 10:43.23 | thereminbr | Hi, Im failing to route a call based on the CID |
| 10:43.30 | thereminbr | extensions.conf |
| 10:43.31 | thereminbr | [from_vono] |
| 10:43.31 | thereminbr | exten => user/05184021342,1,Playback(hello-world) |
| 10:43.32 | thereminbr | sip.conf |
| 10:43.32 | thereminbr | register => user:pass@troncodovono:5060/user |
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| 10:44.19 | thereminbr | I get Call from 'user' to extension 'user' rejected because extension not found. |
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| 10:58.42 | pisg | i use realtime sip_buddies and user set context outline1 = exten => _X.,1,Dial(WOOMERA/g0/${EXTEN}) when i try ring i see this in CLI and my GSM number not calling http://pastebin.com/GNfAWpTd |
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| 11:36.28 | asteriskATmarmuD | hi guys |
| 11:36.56 | asteriskATmarmuD | after 2 meetme() calls no dial() is ever executed... will post CLI in a sec |
| 11:36.59 | asteriskATmarmuD | any hints |
| 11:37.46 | asteriskATmarmuD | http://pastie.org/875247 |
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| 11:38.42 | pisg | i use realtime sip_buddies and user set context outline1 = exten => _X.,1,Dial(WOOMERA/g0/${EXTEN}) when i try ring i see this in CLI and my GSM number not calling http://pastebin.com/qhhKesPX |
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| 11:52.10 | moos3 | I have a question about ring groups |
| 11:54.15 | waa | I have installed asterisk 1.6.2.6 on ubuntu box and make samples created a system startup /etc/init.d/asterisk when I start using this script asterisk use 100% off my CPU but when I run asterisk manually it not occur, why? |
| 11:54.22 | *** part/#asterisk thereminbr (~theremin@201.21.230.64) |
| 11:54.59 | pisg | waa: /etc/init.d/asterisk using -c |
| 11:55.33 | waa | pisg, yes |
| 11:55.59 | waa | pisg, start-stop-daemon --start --oknodo --background --exec $DAEMON -- $ASTARGS -c |
| 11:58.39 | tzafrir_laptop | waa, huh? |
| 11:58.49 | tzafrir_laptop | that's plain wrong |
| 11:58.55 | tzafrir_laptop | chop off that -c |
| 11:59.24 | waa | but if I use -c manually it don't use all CPU |
| 11:59.45 | tzafrir_laptop | waa, more interestingly, consider using the upstart script instead |
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| 12:00.06 | tzafrir_laptop | the problem is that asterisk expects to have a console, but doesn't really |
| 12:00.09 | tzafrir_laptop | and shouldn't |
| 12:00.33 | tzafrir_laptop | upstart should give you a real monitoring script |
| 12:00.56 | tzafrir_laptop | (though I must admit that service dependecies there are not fully debugged, and testers are welcomed) |
| 12:01.34 | tzafrir_laptop | waa, basically, grab the init.d script from the asterisk package . This one should work well |
| 12:01.50 | pisg | i use realtime sip_buddies and user set context outline1 = exten => _X.,1,Dial(WOOMERA/g0/${EXTEN}) when i try ring i see this in CLI and my GSM number not calling http://pastebin.com/qhhKesPX |
| 12:04.45 | waa | tzafrir_laptop, I chop off -c form init script and is running well |
| 12:05.42 | waa | I will get init.d from ubuntu asterisk package |
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| 12:23.00 | ManxPower-work | ~answers |
| 12:23.01 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:25.22 | redax | hi, what can I do if my SIP provider uses 2 sip servers, and he balancing the calls. so you're never know which ip will the provider use for the next inbound call |
| 12:26.33 | redax | and host=sip.provider.com resolves the first one. but sometimes it cames from the other ip. |
| 12:30.40 | [TK]D-Fender | redax: try setting up a "user" with host=dynamic |
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| 12:33.55 | pisg | ManxPower-work: i use realtime sip_buddies and user set context outline1 = exten => _X.,1,Dial(WOOMERA/g0/${EXTEN}) when i try ring i see this in CLI and my GSM number not calling http://pastebin.com/qhhKesPX |
| 12:34.02 | redax | D-Fender; or shall I create 2 user with host=first_ip / host=second_ip ? |
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| 12:35.21 | [TK]D-Fender | redax: You could try that as well |
| 12:39.08 | ManxPower-work | pisg, I can't help you with Realtime |
| 12:41.12 | *** part/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 12:44.45 | pisg | ManxPower-work: why ? |
| 12:45.10 | ManxPower-work | pisg, because I have never used, and will likely never use it. |
| 12:47.12 | pisg | but problme is in ringing, and woodera look at this http://pastebin.com/qhhKesPX |
| 12:47.20 | pisg | in realtime i only have SIP users |
| 12:47.26 | ManxPower-work | And I have, and chances are never will, use WOOMERA |
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| 12:47.43 | ManxPower-work | I have never used WOOMERA and chances I never will use WOOMERA |
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| 12:50.24 | shader | why do all service providers (at least in the US) charge per "sip trunk"? i.e. simultaneous call? |
| 12:50.44 | ManxPower-work | shader, they don't |
| 12:50.57 | shader | I've been looking for one that doesn't |
| 12:51.00 | ManxPower-work | Virtually all providers have per min plans. |
| 12:51.03 | shader | having a hard time |
| 12:51.29 | shader | ManxPower-work: do you have an example? |
| 12:51.32 | ManxPower-work | vitelity, teliax |
| 12:51.45 | ManxPower-work | ~itsp-us |
| 12:51.53 | ManxPower-work | ~itsp |
| 12:51.54 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 12:52.06 | ManxPower-work | <PROTECTED> |
| 12:52.07 | infobot | rumour has it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 12:54.19 | shader | voicepulse and teliax come with four "channels", and you have to pay more/mo for more simultaneous calls |
| 12:54.32 | ManxPower-work | shader, no, that is only for the UNLIMITED plans. |
| 12:54.50 | shader | https://www.teliax.com/plans/3 |
| 12:54.59 | shader | 2500 minutes, 4 channels |
| 12:55.32 | ManxPower-work | shader, https://www.teliax.com/plans/4 |
| 12:55.45 | ManxPower-work | your plan 3 is an "unlimited" plan. |
| 12:56.02 | ManxPower-work | you're never going to get unlimited channels on an "unlimited" plan. |
| 12:56.43 | ManxPower-work | http://vitelity.net/?p=retailserv |
| 12:57.34 | shader | interesting |
| 12:57.50 | shader | I would have thought that unlimited meant no limit, not 2500 minutes ;) |
| 12:58.53 | ManxPower-work | there is no such thing as "unlimited". |
| 12:59.12 | ManxPower-work | It's just that some providers actually tell you what the limit is. |
| 12:59.23 | beek | shader: My Verizon Droid data plan is unlimited... as long as I don't go over 5Gb |
| 12:59.40 | ManxPower-work | beek, My Verizon EVDO service is the same |
| 12:59.59 | ManxPower-work | The first 5GB is $60/month, each additional 5GB is $250. |
| 13:00.24 | ManxPower-work | (based on what I remember from their "over 5GB" overage rates. |
| 13:00.46 | shader | ouch |
| 13:01.18 | ManxPower-work | shader, "unlimited" is a marketing term, nothing else. |
| 13:01.23 | shader | what are the pros and cons of IAX2 vs SIP? |
| 13:01.44 | slidesinger | shader: I Do get unlimited voice and SMS, but not data. |
| 13:01.51 | ManxPower-work | I use SIP because IAX2 caused me problems. |
| 13:02.08 | ManxPower-work | Anyone that tells you "IAX" is better for NAT, is an idiot or is not telling you the whole story. |
| 13:02.08 | [TK]D-Fender | slidesinger: You can trunk IAX2 to save on bandwidth for multiple simultaneous calls to a single host. |
| 13:02.22 | [TK]D-Fender | slidesinger: And IAX2 is more NAT friendly |
| 13:02.32 | [TK]D-Fender | slidesinger: However I advise using it only if NEEDED |
| 13:02.58 | ManxPower-work | IAX2 is more NAT friendly under *some* setups. |
| 13:03.23 | [TK]D-Fender | ManxPower-work: Friendly in general... but rarely needed. |
| 13:03.36 | ManxPower-work | [TK]D-Fender, Exactly. |
| 13:04.39 | ManxPower-work | slidesinger, start using 10,000 mins/month and see how long that "unlimited" lasts. |
| 13:04.43 | petern_ | hmm, i used an IAX2 trunk because... it seemed natural |
| 13:04.56 | ManxPower-work | ~trunk |
| 13:04.57 | infobot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 13:05.13 | ManxPower-work | petern_, you needed multiple calls in the same UDP packets? |
| 13:05.27 | ManxPower-work | 'cause that's what IAX2 trunking does, as compared to non-trunked IAX |
| 13:05.35 | petern_ | nope, i wanted to hook up two asterisk boxes |
| 13:06.32 | ManxPower-work | then you did not need to set it up as a trunk, an regular IAX2 connection would have been just fine. In fact, if you enable IAX2 trunking you actually use MORE bandwidth when you have less than 3 calls going between those same 2 hosts. |
| 13:06.52 | ManxPower-work | petern_, maybe you are mistaken and only set up an IAX2 connection, not an IAX2 trunk. |
| 13:06.59 | petern_ | you are right, i probably did :) |
| 13:07.19 | ManxPower-work | petern_, then stop confusing everyone and calling it "trunk". It is not a trunk. |
| 13:07.49 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 13:07.57 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:08.09 | moos3 | in a directory can i do just this extension,name ? |
| 13:08.49 | ManxPower-work | moos3, your question makes no sense. |
| 13:09.17 | redax | if I do Set(CALLERID(num)="123456" before dialout trunk |
| 13:09.27 | moos3 | ManxPower-work, I want to be able to search by my products name and then send you to the correct extension |
| 13:09.50 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:10.02 | redax | why does my CDR not contains the changed callerid ? |
| 13:10.02 | ManxPower-work | moos3, That is what the Directory application does. |
| 13:10.35 | ManxPower-work | redax, chances are your FreePBX is messing that up. |
| 13:10.49 | [TK]D-Fender | ReDdo NOT put quotes on that |
| 13:10.53 | [TK]D-Fender | redax: do NOT put quotes on that |
| 13:10.58 | moos3 | ManxPower-work, yeah I get that it uses voicemail.conf for it and is format is extension => password, name,email,,tz= can i just do extension,name |
| 13:10.59 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:11.14 | [TK]D-Fender | moos3: yes |
| 13:11.19 | moos3 | [TK]D-Fender, thanks |
| 13:11.24 | [TK]D-Fender | moos3: Actually.. you have to leave the space for the PW |
| 13:11.40 | moos3 | so extension => , name? |
| 13:11.40 | redax | ManxPower-work: although fpbx is installed, but I use my own contexts, |
| 13:11.46 | ManxPower-work | also if you want it to speak the name, you must record the name for each mailbox |
| 13:12.16 | ManxPower-work | redax, that does not matter. |
| 13:12.42 | ManxPower-work | the fact you said "trunk dialout" means you are in the FreePBX mindset. |
| 13:12.50 | redax | hehe |
| 13:13.14 | ManxPower-work | Once you follow [TK]D-Fender 's advice it might work, however. |
| 13:13.22 | redax | simpley I want to bypass calls from sip to mISDN |
| 13:13.34 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 13:13.37 | ManxPower-work | see what I mean. "bypass" makes no sense. |
| 13:14.00 | redax | I have to send last 3 digits to the isdn, meanwhhile I have to add the prefix |
| 13:14.30 | ManxPower-work | redax, does it work after you did what [TK]D-Fender told you to do? |
| 13:14.39 | redax | testing. 1min |
| 13:14.51 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:15.35 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:16.12 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 13:17.42 | slidesinger | If there is no such thing as a "SIP Trunk," what is the correct term to describe direct IP lines from outfits like Vitelity? |
| 13:17.52 | *** join/#asterisk LemensTS (~LemensTS@mail.zuckerfeather.com) |
| 13:18.02 | ManxPower-work | slidesinger, We call the "peers" |
| 13:18.20 | ManxPower-work | slidesinger, and you never have "direct IP lines" There is no such thing. |
| 13:18.51 | ManxPower-work | When you make a call a connection is set up. When you end the call the connection is torn down. |
| 13:19.00 | kombi | anyone using chan-sccp? just compiled the thing. phone is recognized but refuses to make calls... |
| 13:19.02 | ManxPower-work | There is no "direct" or "dedicated" or anything like that. |
| 13:19.56 | redax | MaxPower,D-Fender still the 3 digit callerid stored in CDR, |
| 13:20.09 | ManxPower-work | redax, I wish you the BEST of luck. |
| 13:20.41 | redax | not the rewritten by Set(CALLERID(num)=123456${CALLERID(num)}) |
| 13:20.54 | slidesinger | Thank you, I am still learning the terminology, not having a telecom background, but a networking one. So essentially it is a logical data connection that serves VOIP packets? |
| 13:21.04 | [TK]D-Fender | redax: CDR records who started the call.. not what it ended up with |
| 13:21.18 | redax | oh. |
| 13:21.33 | [TK]D-Fender | redax: You want to store alternative data? thats what UserField is for. |
| 13:21.44 | kombi | hmm, chan-sccp loaded, phone recognized, cli even shows "phone off hook" and such (hurray!) How can I debbug on the lowest possible level? |
| 13:22.07 | [TK]D-Fender | kombi: netcat |
| 13:22.33 | kombi | fender: that might be a little much low... |
| 13:23.23 | ManxPower-work | kombi, almost nobody uses SCCP/Skinny with Asterisk. Only the idiots and the people with no other choice. |
| 13:23.23 | slidesinger | Strike that, it should read a logical data connection that is dedicated for the purpose of carrying VOIP packets. I think. |
| 13:24.14 | ManxPower-work | slidesinger, What is this "logical data connection....dedicated to VoIP" you are talking about. There is no dedicated anything. |
| 13:24.15 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.87.110) |
| 13:24.29 | [TK]D-Fender | kombi: "How can I debbug on the lowest possible level?" <- maybe you shuold think about your questions a bit more |
| 13:24.34 | kombi | ManxPower-work: thank you so much..;) I converted all other cisco phones to sip, but now there is this chan-sccp that actually claims to work so I thought I give it a shot |
| 13:24.39 | ManxPower-work | When you make a call the connection is created, when you end a call the connection is terminated. This is not rocket science, this is like a frickin web site. |
| 13:25.27 | ManxPower-work | Do you have trunks to cnn.com or slashdot.org? No. The connection is created and torn down as needed, just like a SIP calls. |
| 13:25.38 | [TK]D-Fender | slidesinger: And on the theory of "connection" ..... typically SIP and RTP are all UDP hence stateless. There isn't even a "connected" state techniaclly. |
| 13:26.30 | petern_ | kombi, you could try on the mailing list? http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion |
| 13:26.31 | slidesinger | ManxPower-work: I see a registration of the two Vitelity peers and am trying to describe their function. |
| 13:27.03 | ManxPower-work | slidesinger, Correct. The registration says "this IP is associated with this user/pass". It means nothing else. |
| 13:27.18 | petern_ | kombi, i use chan_sccp successfully, out of choice because the phones behave "more nicely" |
| 13:28.14 | petern_ | kombi, there is, of course, "sccp debug" in the cli |
| 13:28.54 | kombi | petern_: I like to hear that! I think I just need to get the config right.. sccp_debug! <- that's my clue! thanks petern_, I'll delve into it |
| 13:29.16 | slidesinger | ManxPower-work: Rather than me continuing to go about this the hard way, are there docs that describe this part of the process? The peers provided by a VoIP carrier, that is. |
| 13:29.44 | ManxPower-work | The SIP RFC describes SIP. |
| 13:29.51 | ManxPower-work | I can look up the RFC number if you want. |
| 13:30.05 | file | slidesinger: let's see if this works... a VoIP carrier provides you with authentication credentials which are used in the authentication part of SIP, to authenticate the session that a SIP endpoint sets up |
| 13:30.09 | ManxPower-work | Also define "this part of the process" |
| 13:30.41 | file | slidesinger: a 'peer' in Asterisk is a configuration entry used to store that information and is used to know what host/username/password to use when establishing the session |
| 13:30.50 | kombi | petern_: Just so I got it right: In sccp.conf I define the device and refer to that name in extensions.conf, right? |
| 13:31.03 | slidesinger | I can look up the RFC's, thank you. |
| 13:31.05 | ManxPower-work | hands file not 1, not 2, but 3 glorious muffins |
| 13:31.33 | file | slidesinger: SIP is the protocol, usually used over UDP, to communicate with the VoIP carrier - the packets created as a result of this are no more special then my packets being sent for this IRC message right now |
| 13:32.18 | ManxPower-work | but...but...I thought SIP was SPECIAL! |
| 13:32.46 | file | slidesinger: if you really want to know SIP then I would suggest finding an overview of it |
| 13:32.52 | ManxPower-work | Next thing you'll be telling us is that God exists and that the Easter Bunny does not! |
| 13:32.59 | file | slidesinger: reading the relevant RFCs is just... not suggested |
| 13:33.57 | file | Disclaimer: the 'no more special' part assumes that QoS is not in use, if QoS is in use then they can be treated special to ensure they are delivered promptly |
| 13:35.00 | pisg | file: i have problem, extenions.conf is outline1 = exten => _X.,1,Dial(WOOMERA/g0/${EXTEN}) when i try ring i see this in CLI and my GSM number not calling http://pastebin.com/qhhKesPX |
| 13:35.13 | file | I don't know chan_woomera. |
| 13:35.54 | petern_ | kombi, pretty much |
| 13:36.03 | pisg | ;/ |
| 13:36.14 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:36.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:36.26 | file | malcolmd: hack the planet!!! |
| 13:36.49 | moos3 | [TK]D-Fender, question about using directories and searching for is Hosting in hosting services or serivces in it |
| 13:37.09 | slidesinger | In a traditional, PSTN connection, when the phone goes off hook, there is dialtone. Also, I have been told by the telecom guys I work with that there is always voltage. While it is a simplistic view, we can state that the line is either hot, if it works, or dead, if it does not. |
| 13:37.11 | slidesinger | In VoIP the mechanism is different, but the end result is the same. I had no need to learn how PSTN lines work, I do need to understand the VoIP equivalent. |
| 13:37.44 | file | slidesinger: in traditional PSTN there is an actual physical dedicated connection for each 'line' |
| 13:38.00 | kombi | petern_: would you maybe let me take look at the relevant parts of your config? |
| 13:38.05 | file | slidesinger: totally not true in VoIP |
| 13:38.12 | file | slidesinger: it's just data. |
| 13:38.32 | file | slidesinger: and on a SIP device when you go off hook the device itself is generating the dialtone locally |
| 13:39.00 | patrb | file: does that include softphones? |
| 13:39.06 | file | patrb: yes. |
| 13:39.33 | *** join/#asterisk saftsack (~oliver@p579DC7C3.dip.t-dialin.net) |
| 13:39.37 | ManxPower-work | Dialtone on IP phones is just there to make you feel better, it does not actually DO anything. |
| 13:39.40 | file | it's possible to have the SIP device immediately dial some sort of extension on a remote SIP server which then provides dialtone though, but that rarely happens |
| 13:39.56 | saftsack | hey, if i do a blindxfer and the destination is busy, then the call is dead. is there any possible thing to avoid this? |
| 13:40.05 | slidesinger | file: That's all I knew when I got into this conversation. So what I am looking for is an overview of everything that happens when you go off hook? |
| 13:40.10 | ManxPower-work | saftsack, yes. Fix your broken dialplan. |
| 13:40.25 | file | slidesinger: the device provides dialtone locally, no communication to the remote SIP server |
| 13:40.37 | saftsack | ManxPower-work: whats the reason for such a behaviour? |
| 13:40.44 | ManxPower-work | slidesinger, NOTHING happens when you go off hook. Stuff happens when you are done dialing. The phone collects all the digits then sends them to the server as a data packet. |
| 13:40.56 | ManxPower-work | saftsack, chances are you are running one of those PoS GUIs. |
| 13:41.15 | ManxPower-work | saftsack, you wrote your dialplan, you should know what it does. |
| 13:41.27 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:41.31 | saftsack | ManxPower-work: no it is a manually designed dialplan. maybe i forgot to add a parking extension? |
| 13:41.38 | kombi | thanks! |
| 13:42.22 | slidesinger | So the device that generates dialtone is the telephone instrument itself, whether a soft phone or a physical IP phone. |
| 13:42.25 | saftsack | furthermore i can not imagine what happens if the destination is busy. is the transfer-originator then called again or what? |
| 13:42.47 | saftsack | ManxPower-work: i have to say, that i don't use #1. i use the snom integrateds blind x-fer button |
| 13:43.03 | ManxPower-work | saftsack, if the destination is BUSY then the dialplan goes to the next priority in the dialplan |
| 13:43.12 | kombi | great stuff! what's with the SEP though? |
| 13:43.22 | petern_ | kombi, mac address of phone |
| 13:43.28 | [TK]D-Fender | saftsack: You transferred the call. It is GONE. Its up to your dialpla to decide what to do. |
| 13:43.40 | petern_ | kombi, well, default hostname, i guess |
| 13:44.01 | [TK]D-Fender | saftsack: Go read the CHANNELVARIABLES doc for a bit |
| 13:44.14 | ManxPower-work | saftsack, and "core show application dial" |
| 13:44.19 | saftsack | ManxPower-work, [TK]D-Fender where is this call? just after the Dial line of the original call? |
| 13:44.29 | [TK]D-Fender | ..... |
| 13:44.29 | saftsack | because then i could use DIALSTATUS :) |
| 13:44.42 | ManxPower-work | saftsack, the DIALED extension |
| 13:44.47 | [TK]D-Fender | saftsack: You transfered the call. it is executing whatever extension you transferred it to |
| 13:44.54 | saftsack | aaah ... sounds logically ;) |
| 13:44.55 | saftsack | thanks |
| 13:45.44 | ManxPower-work | saftsack, chances your transfer is completing just fine, it's that the call falls off the dialplan and hangs up because you don't handle a BUSY |
| 13:46.26 | petern_ | kombi, oh... that's a v2 config, not v3 |
| 13:48.20 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 13:49.06 | saftsack | is there a chance to decide whether it is a transferred call, or if it is a normal internal call? |
| 13:50.12 | [TK]D-Fender | [09:44]<[TK]D-Fender>saftsack: Go read the CHANNELVARIABLES doc for a bit |
| 13:51.17 | patrb | Does dialplan logic remind anyone else of gw basic? |
| 13:51.22 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:51.28 | *** join/#asterisk nickaugust (~anonymous@rrcs-71-42-53-182.se.biz.rr.com) |
| 13:52.22 | kombi | petern_: must look at it later, thanks so far! |
| 13:52.29 | saftsack | [TK]D-Fender: ok thanks :) |
| 13:53.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:56.06 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 13:56.23 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 13:56.26 | spenguin[work] | hey |
| 13:57.15 | spenguin[work] | if I wanted to make sure the dial status was BUSY or NOANSWER, then send to voicemail |
| 13:57.18 | spenguin[work] | can it be done |
| 13:57.26 | patrb | yes |
| 13:57.35 | spenguin[work] | liek how |
| 13:58.28 | [TK]D-Fender | spenguin[work]: "core show application gotoif" |
| 13:59.14 | spenguin[work] | ah well I know that :p |
| 13:59.15 | spenguin[work] | thanks |
| 14:00.08 | patrb | spenguin[work]: we use a standard macro that checks dial status and sends to different voicemail based on busy/noanswer |
| 14:00.29 | spenguin[work] | ok |
| 14:00.30 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 14:00.35 | [TK]D-Fender | patrb: Never said anything about needing to "standardize" anything.... |
| 14:00.49 | [TK]D-Fender | patrb: Not that it isn't usually the way to go.. |
| 14:01.03 | patrb | [TK]D-Fender: Thats fine, was simply stating how i did it |
| 14:02.12 | *** join/#asterisk moy (~chatzilla@74.12.130.209) |
| 14:02.34 | redax | [TK]D-Fender: I don't want to store the 3digit src in CDR but the real src which is <somemoredigit><3digit what I get>. that's what I want to store in CDR. is there a way to change the CDR src or callerid at all? |
| 14:03.09 | [TK]D-Fender | redax: Typically CDR is read-only, except for userfield. |
| 14:03.28 | [TK]D-Fender | redax: Go make some other log that you can chain to it by uniqueid, etc |
| 14:04.47 | redax | this is not the real callerid src. it's just the last3 digit. |
| 14:06.06 | [TK]D-Fender | redax: What isn't "real"? |
| 14:06.06 | patrb | redax: is something in your dial plan truncating the src? |
| 14:06.09 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:07.31 | redax | patrb, D-Fender; there's a panasonic pbx connected to the asterisk via ISDN. the panasonic wants/will send only the last 3 digits. that's what I have to transform to the real phonenumber, and do the call via SIP |
| 14:08.15 | [TK]D-Fender | ReDYour inbound call has that CID. It is STUCK with it. Your OUTBOUND call is separate |
| 14:08.26 | redax | patrb: nothing truncates, just I have to add a few digits like: Set(CALLERID(num)=36123456${CALLERID(num)}) |
| 14:08.58 | patrb | redax, so you can still identify the incoming 3-digit number by the last 3 digits? |
| 14:09.09 | redax | yes :D |
| 14:09.48 | patrb | redax: looking at my dial plan, I thought Id changed CDR fields before |
| 14:10.17 | redax | but I though I can rewrite the cdr.src field :/ |
| 14:10.21 | *** join/#asterisk saftsack (~oliver@IP-213157024107.static.heagmedianet.de) |
| 14:10.23 | Katty | hi |
| 14:11.42 | patrb | redax: I'm using this in one of my dialplans: Set(CDR(dst)=${EXTEN}) |
| 14:11.53 | patrb | redax: I imagine you could do something similar to change the src |
| 14:12.13 | redax | [TK]D-Fender: ok. thanks. I have 2records for 1 bypassed call. You're absolutly right |
| 14:12.58 | [TK]D-Fender | ----- All of the CDR field names are read-only, except for 'accountcode', 'userfield', and 'amaflags'. |
| 14:13.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:13.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:13.33 | patrb | [TK]D-Fender: Thanks for clearing that up :) |
| 14:13.44 | [TK]D-Fender | redax: ... and its not friggen BYPASSED. Its a simple stupid call. Call in. Call out. There is no "routing", "bypassing", "redirecting", etc |
| 14:14.34 | *** join/#asterisk aceio (~aioi@93-96-168-138.zone4.bethere.co.uk) |
| 14:14.48 | aceio | hi all |
| 14:15.31 | aceio | try to get cisco ip communicator to register |
| 14:15.47 | aceio | i am having no luck |
| 14:16.20 | aceio | any help will be wonderfull |
| 14:16.59 | [TK]D-Fender | aceio: Show us the actual problem and maybe we can |
| 14:17.03 | patrb | aceio: I think you'll need to provide some more information to get help.. |
| 14:17.04 | [TK]D-Fender | aceio: PASETBIN is your friend |
| 14:17.06 | [TK]D-Fender | ~pb |
| 14:17.07 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:17.48 | patrb | aceio: does the asterisk CLI give any output when the soft phone attempts to register? |
| 14:17.50 | aceio | okay sure |
| 14:18.25 | aceio | no |
| 14:18.43 | patrb | aceio: its most likely a networking issue then, not an asterisk configuration issue |
| 14:19.27 | aceio | hold on |
| 14:20.11 | aceio | i am getting messages on CLI now |
| 14:20.25 | redax | and may I change ${EXTEN} ? :D |
| 14:20.42 | aceio | i need to paste |
| 14:20.59 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 14:21.12 | patrb | redax: you should be able change ${EXTEN} like any other channel variable |
| 14:21.32 | [TK]D-Fender | redax: No |
| 14:21.55 | patrb | [TK]D-Fender: really? Well i'll be damned |
| 14:23.01 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 14:23.11 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:23.19 | aceio | http://pastebin.com/qVcBx7pe |
| 14:25.09 | aceio | <patrb> http://pastebin.com/qVcBx7pe |
| 14:25.29 | patrb | aceio: It looks like an issue with chan_skinny which I've never used |
| 14:25.35 | *** join/#asterisk Slugs_ (~0ca6b485@gateway/web/freenode/x-doqfwijsxzowcjmc) |
| 14:26.15 | ManxPower-work | You change EXTEN by using Goto |
| 14:26.58 | aceio | <patrb> i am trying to get the ip communicator to work with SIP |
| 14:28.40 | patrb | aceio: looking at the documentation of Cisco IP Communicator: Cisco IP Communicator version 2.1 supports Session Initiation Protocol (SIP) as well as the Cisco Unified Communications Manager Skinny Client Control Protocol (SCCP). |
| 14:28.40 | *** join/#asterisk kbr (~kbr@ASte-Genev-Bois-154-1-111-194.w83-199.abo.wanadoo.fr) |
| 14:28.59 | patrb | aceio: it sounds like you need to change the settings on your local machine IP communicator to use SIP instead of SCCP |
| 14:29.26 | shader | how would you make a simple test setup for asterisk? |
| 14:29.54 | [TK]D-Fender | shader: Very simply |
| 14:31.16 | shader | [TK]D-Fender: that's great. Now if only you can convey the instructions so simply ;) |
| 14:31.37 | shader | i.e, do I need to asterisk installations? |
| 14:31.41 | shader | or is there something simpler |
| 14:31.46 | shader | *two |
| 14:31.56 | patrb | shader: it depends on what parts of asterisk you want to test |
| 14:32.09 | aceio | <patrb> yes iam running version 7-0-3-0 Cisco IP Communicator |
| 14:32.14 | patrb | shader: the most basic test you could do, is one asterisk box, two soft phones |
| 14:32.22 | shader | ok |
| 14:32.36 | shader | can both softphones be running on the same computer? |
| 14:32.59 | patrb | aceio: go into the IP communicator settings and look for an option for SIP or SCCP and ensure that SIP is checked or enabled rather than SCCP |
| 14:33.53 | patrb | shader: I *think* both softphones can run on the same machine, but i've never tested that |
| 14:33.59 | ManxPower-work | shader, you do NOT want to run multiple SIP applications on the same machine. |
| 14:34.28 | [TK]D-Fender | shader: Who says you need 2 softphones? And no, you don't do that.. they'll fight over soundcard ersources, etc |
| 14:35.19 | patrb | [TK]D-Fender: he asked for a simple test, I told him a simple test would be 1 asterisk box and 2 softphones |
| 14:35.25 | [TK]D-Fender | shader: Calls should be on separate devices |
| 14:35.40 | [TK]D-Fender | patrb: My simple test only requires 1 |
| 14:35.48 | patrb | [TK]D-Fender: congrats |
| 14:35.58 | aceio | okay |
| 14:36.10 | [TK]D-Fender | patrb: My simple test is 33% simpler than yours \o/ |
| 14:36.14 | shader | woot! |
| 14:36.40 | shader | can anyone come up with a test that involves *no* softphones? |
| 14:36.45 | shader | ;) |
| 14:36.48 | patrb | shader: yes |
| 14:37.01 | patrb | shader: a simple rick roll script could do that |
| 14:37.03 | [TK]D-Fender | shader: Install *. The end. Who says has to talk to anything at all? |
| 14:37.17 | shader | [TK]D-Fender: lol |
| 14:37.17 | ManxPower-work | ~zeeek |
| 14:37.18 | infobot | from memory, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 14:37.19 | [TK]D-Fender | *it |
| 14:37.27 | ManxPower-work | This also applies to trying to test a PBX without any phones. |
| 14:37.54 | patrb | lol |
| 14:38.09 | [TK]D-Fender | shader: You could set up * with an ITSP that only takes messages. There. You have no phones. Its just a VM box. |
| 14:38.14 | benngard | phone - asterisk with meetme? |
| 14:38.17 | [TK]D-Fender | shader: * is what you make it to be. |
| 14:38.24 | [TK]D-Fender | shader: For me, Asterisk is a JUKEBOX |
| 14:38.29 | shader | really? |
| 14:38.35 | [TK]D-Fender | shader: And a coffee machine timer |
| 14:38.55 | patrb | [TK]D-Fender: you use a micro controller w/ that coffee timer? |
| 14:38.58 | Kobaz | is there a way to turn on dnd on a polycom 320/330 |
| 14:39.02 | Kobaz | from the phone itself |
| 14:39.03 | [TK]D-Fender | patrb: X-10 |
| 14:39.13 | [TK]D-Fender | Kobaz: Of course |
| 14:39.16 | gr0mit | does anyone here use AgileBill for voip billing etc? |
| 14:39.24 | Kobaz | this was really weird... i got a call this morning saying that a phone kept returning busy... and i had them reboot it, and now it works |
| 14:39.25 | [TK]D-Fender | Kobaz: Try pushing some buttons on it... |
| 14:39.31 | Kobaz | [TK]D-Fender: where in the menu is it... i can't find it... heh |
| 14:39.35 | Katty | hi |
| 14:39.40 | [TK]D-Fender | Kobaz: Keep lookin' blind boy! |
| 14:39.55 | Kobaz | menu...settings |
| 14:39.57 | patrb | [TK]D-Fender: I setup an arduino to turn an LED on and off when i get a call to my extension in asterisk...was a fun project :) |
| 14:40.08 | Kobaz | basic... (since they don' have a admin password) |
| 14:40.33 | Kobaz | there's no options anywhere in basic for dnd |
| 14:40.37 | [TK]D-Fender | patrb: Via X-10 I can blink my desk lamps, or whatever else I felt like doing... |
| 14:40.44 | Kobaz | gets the manual |
| 14:40.50 | [TK]D-Fender | Kobaz: Yes, there is |
| 14:40.57 | shader | [TK]D-Fender: how did you interface X-10 with asterisk? |
| 14:41.06 | [TK]D-Fender | shader: heyu2 |
| 14:41.08 | Kobaz | ooooh |
| 14:41.10 | Kobaz | it's under features |
| 14:41.28 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 14:41.35 | Kobaz | i found it :P |
| 14:41.48 | [TK]D-Fender | Kobaz: Imagine that... a call feature buried in a non-descript menu named "features" |
| 14:41.53 | Kobaz | haha |
| 14:41.59 | Kobaz | i thought it would be under settings |
| 14:42.07 | shader | [TK]D-Fender: so what purpose does * serve in that setup? |
| 14:42.36 | [TK]D-Fender | shader: phone home and coffe is ready when I walk in the door |
| 14:43.24 | shader | cool |
| 14:43.48 | *** part/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 14:43.53 | aceio | <patrb> look like i don't have that options |
| 14:45.19 | aceio | <patrb>i may have to downgrade from 7-0-3-0 |
| 14:45.41 | aceio | <patrb>2.1 |
| 14:46.51 | shader | [TK]D-Fender: so what was that one-phone test of yours? |
| 14:47.44 | [TK]D-Fender | shader: Use softphone to test IVR's voicemail, etc. Connect out to ITSP's, and so forth |
| 14:47.55 | [TK]D-Fender | shader: everything depends what you expect * to do for you |
| 14:48.36 | *** join/#asterisk dennisG (~root@84.30.136.208) |
| 14:48.58 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 14:49.05 | shader | true enough |
| 14:49.09 | Katty | i expect it to make my lunch. |
| 14:49.30 | [TK]D-Fender | Katty: Doable... |
| 14:49.45 | Katty | that's what she said. |
| 14:49.47 | [TK]D-Fender | Katty: There is also that pizza-ordering bash script out there... easily adaptable |
| 14:50.05 | [TK]D-Fender | Katty: Huzzah |
| 14:50.20 | Katty | my sister is having a laminectomy done :< |
| 14:50.48 | creativx | does that mean she is being laminated? |
| 14:51.01 | Katty | bits of her spine are being removed |
| 14:51.13 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 14:51.23 | Katty | apparently they have been putting extreme pressure on the spinal cord. |
| 14:51.32 | Katty | and causing severe pain |
| 14:52.13 | Kobaz | okay so |
| 14:52.16 | Kobaz | more polycrum |
| 14:52.42 | Kobaz | i'm using the idlebrowser thing to load up a microbrowser page to show some status info |
| 14:53.00 | Kobaz | so now... someone turns on dnd... the server doesn't know it's on dnd... and since i'm using idlebrowser, you cant see the phone status |
| 14:53.16 | Kobaz | so is there a way to like rpc query the polycom (or have the polycom post it's status) |
| 14:53.20 | [TK]D-Fender | Kobaz: Tell them to F-ing stop turning on DND |
| 14:53.27 | Kobaz | i can disable it |
| 14:53.32 | [TK]D-Fender | Kobaz: indeed |
| 14:53.41 | [TK]D-Fender | Kobaz: but I'd rather fix the user |
| 14:53.44 | Kobaz | rpc querying the polycom would be sweet |
| 14:54.07 | Kobaz | i've always wanted to get line status info... see what call is on line 1 |
| 14:54.38 | Kobaz | you can check for calls in * but you wont know what line it's on, on the phone |
| 14:55.17 | Katty | i think the idea of a phone having lines is kind of archaic |
| 14:55.42 | Katty | not to change the subject or anything |
| 14:55.53 | *** join/#asterisk torrancew (~torrancew@ip70-172-225-171.br.br.cox.net) |
| 14:55.56 | Kobaz | sure... show the 63 digit sip call id instead |
| 14:56.01 | Kobaz | 64 rather |
| 14:56.29 | torrancew | how might I change the reported CID for callers that don't broadcast CID info? |
| 14:56.46 | torrancew | say, have it show up as "Caller Unknown", or similar |
| 14:56.58 | *** join/#asterisk kbr (~kbr@ASte-Genev-Bois-152-1-19-61.w83-114.abo.wanadoo.fr) |
| 14:57.01 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 14:57.02 | Katty | if callerid = foo do something |
| 14:57.03 | Kobaz | torrancew: Set(CALLERID(name)=...) |
| 14:57.13 | Kobaz | and then there's (num) as well |
| 14:57.14 | Katty | ^- do something like that. |
| 14:57.40 | Katty | you could even run a bash script |
| 14:57.44 | torrancew | ok, so if both name and number come in as empty, i could do Set(CALLERID("")="...")? |
| 14:57.50 | Katty | and execute fender's pizza script |
| 14:57.52 | Kobaz | no |
| 14:57.57 | Kobaz | i just old you |
| 14:58.00 | Kobaz | (name) or (num) |
| 14:58.15 | torrancew | ah |
| 14:58.19 | Kobaz | Set(CALLERID(name)=...) Set(CALLERID(num)=...) |
| 14:58.22 | Katty | but you could check the number |
| 14:58.34 | Katty | which is obviously what everyone uses |
| 14:58.37 | Katty | for the most part |
| 14:58.41 | torrancew | right |
| 14:58.45 | Katty | i block Kobaz's calls by name |
| 14:58.46 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:58.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:59.08 | Katty | putnopvut: ohaider |
| 14:59.15 | ManxPower-work | 1) Name is not accepted by the telcos. |
| 14:59.20 | putnopvut | Katty: HOWDY |
| 14:59.22 | *** join/#asterisk s4msung (~s4msung@dice.s4msung.de) |
| 14:59.23 | Kobaz | GotoIf($["${CALLERID(num)" = ""]...) |
| 14:59.57 | Kobaz | or if you're using ael... it's much simpler to do: if ("${CALLERID(num)" == "") {Set callerid 0} |
| 15:00.03 | Katty | how does digium do their support? |
| 15:00.05 | Kobaz | etc |
| 15:00.11 | Katty | do they have certified partners who just... call in? |
| 15:00.15 | ManxPower-work | Katty, with money 8-) |
| 15:00.16 | Katty | or is it all Paid Cases |
| 15:00.40 | Katty | ManxPower-work: so there's no certified partner thing |
| 15:00.51 | ManxPower-work | Katty, you would have to call Digium. |
| 15:01.01 | Katty | ManxPower-work: i thought you knew EVERYTHING |
| 15:01.04 | Katty | ManxPower-work: way to let me down |
| 15:01.06 | Kobaz | You attacked Keira the Dread Knight hitting them for an Earth Shattering amount of Damage[2214] |
| 15:01.26 | fish-bulb | Katty: there are different support options, like subscriptions, reseller perks, stuff like that. talk to Digium Sales for information |
| 15:01.32 | Katty | LAMP DRAWS NEAR |
| 15:01.46 | Katty | fish-bulb: i don't want to talk to digium |
| 15:01.49 | Katty | fish-bulb: i want to talk to manx. |
| 15:02.55 | Katty | Kobaz: let's break out the minis and play some tabletop |
| 15:04.31 | Katty | wonders why there is a semi parked in front of her house |
| 15:05.14 | Kobaz | heh |
| 15:05.40 | Katty | Kobaz: actually i've been playing fable 2 for xbox, ever heard of it? |
| 15:06.10 | redax | ok. seems like, if anyone sets CALLERID(num) or CALLERID(name) it will not overwrite the cdr.src and cdr.callerid, BUT if you Set(CALLERID(all)=SomeName <1234>) then the CDR _WILL_ contain the modified callerid |
| 15:06.48 | Slugs_ | n somebody help me connect my asterisk to avaya pbx via h323 using freepbx? |
| 15:06.49 | Katty | i overwrite the callerid name on one incoming channel. |
| 15:07.00 | Slugs_ | can* |
| 15:07.01 | Katty | and just the name. |
| 15:07.27 | Katty | infobot: freepbx |
| 15:07.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:07.33 | Katty | ^- Slugs_ |
| 15:07.40 | Kobaz | Katty: nope |
| 15:07.46 | redax | if I modified _JUST_ the `num' then the old `num' was stored in CDR |
| 15:07.53 | Katty | Kobaz: bummer. it's a cute game. |
| 15:08.01 | Kobaz | Katty: speaking of board games... my ex girlfriend loved playing all the rails games... like eurorails and etc |
| 15:08.02 | redax | if I change the `all' then the modified will be Stored in CDR |
| 15:08.02 | Slugs_ | thanks |
| 15:08.17 | Katty | Kobaz: hrmm. never heard of rails before |
| 15:08.27 | Katty | Kobaz: but i'm not into board games really |
| 15:08.33 | Kobaz | it's the boardgame version of railroad tycoon |
| 15:08.37 | patrb | redax: very interesting, is that going to work for what you wanted then/ |
| 15:08.43 | patrb | redax: *? |
| 15:08.59 | redax | yep. this is perfect |
| 15:09.01 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:09.01 | Katty | Kobaz: haven't played that either |
| 15:09.06 | Katty | hi Defraz |
| 15:09.09 | ManxPower-work | redax, You are either 1) Wrong or 2) have discovered a bug |
| 15:09.11 | *** part/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:09.21 | patrb | hehe |
| 15:09.24 | redax | just didn't known that, differs if you set separately the num/name and if you set all |
| 15:09.26 | Katty | maybe he should pastebin a lil somethin somethin |
| 15:09.28 | Katty | and also the callerid |
| 15:09.32 | ManxPower-work | redax, it's not supposed to. |
| 15:09.44 | ManxPower-work | Therefore You are either 1) Wrong or 2) have discovered a bug |
| 15:09.58 | redax | ManxPower-work: if it's a bug, please don't fixit :D |
| 15:10.08 | patrb | redax: post hte bug and then never ever update your system :) |
| 15:10.12 | ManxPower-work | chances are you are using extra quotes and that's screwing it up. |
| 15:10.13 | patrb | redax: j/k |
| 15:10.24 | ManxPower-work | redax, IF it's not working as documented, it's a bug |
| 15:11.05 | redax | you're right some double quote stuff messed up the cdr. |
| 15:11.07 | *** join/#asterisk MACscr (~Mark@c-98-214-100-212.hsd1.il.comcast.net) |
| 15:11.07 | redax | look: |
| 15:11.17 | redax | "","3657558079","06706236818","alkozpontrol","""3657558079"" <3657558079>","SIP/079-090a2810" |
| 15:11.19 | redax | ... |
| 15:12.06 | redax | this is what changes the callerid: exten => _X.,n,Set(CALLERID(all)=${CALLERID(num)} <${CALLERID(num)}>) |
| 15:12.07 | MACscr | I need help debunking a myth. Ever heard of someone with a dynamic ip address through their isp having the ip change during the middle of a call and dropping the call? |
| 15:12.41 | ManxPower-work | Yup! You are using extra quotes |
| 15:12.59 | redax | no I don' use any quotes :D |
| 15:13.16 | patrb | MACscr: Sounds possible. The ISP can change your IP whenever they feel like it if you're dynamic |
| 15:13.26 | patrb | MACscr: however, most don't |
| 15:13.28 | ManxPower-work | redax, Um, your paste shows extra quotes |
| 15:13.37 | shader | MACscr: depends on the ISP I suppose |
| 15:13.47 | ManxPower-work | redax, so you are setting the callerid name to to be the same as the callerid num. |
| 15:13.52 | redax | ManxPower-work: yes, but check the Set(CALLERID... line |
| 15:13.57 | shader | In theory your ip address can change whenever you lease is up |
| 15:13.58 | MACscr | patrb: I understand that technically its possible, but I have never ever heard of it |
| 15:14.08 | ManxPower-work | redax, what you pasted does not match |
| 15:14.21 | ManxPower-work | now, pastebin the CLI output |
| 15:14.25 | patrb | MACscr: neither have I, sorry I cant help more :) |
| 15:15.06 | redax | geez. I have to turn off sip debugging, and call again |
| 15:15.08 | redax | 1min |
| 15:15.31 | shader | MACscr: in my experience, my isp only seems to change my ip address very infrequently |
| 15:15.37 | ManxPower-work | redax, I'm terribly sorry to interrupt your work on something not related to your problem. |
| 15:15.45 | *** join/#asterisk kartik (~koolkarti@117.199.119.61) |
| 15:15.54 | ManxPower-work | shader, that depends on the ISP and the technologu |
| 15:16.02 | redax | it's absolutly related to my problem :) and thank you for your help |
| 15:16.12 | shader | yep |
| 15:16.30 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:16.30 | Katty | hi kartik |
| 15:16.32 | Katty | hi bmoraca_work |
| 15:17.06 | *** join/#asterisk niros_ (~c075ec1d@gateway/web/freenode/x-pjxoazuubnoefwbc) |
| 15:17.09 | kartik | hello Katty |
| 15:18.07 | *** join/#asterisk huey23 (psyops@65.111.241.185) |
| 15:19.39 | redax | oh. ManxPower gone? |
| 15:19.41 | redax | geez. |
| 15:19.50 | Katty | he probably had work to do |
| 15:20.02 | Katty | not everyone can flutter about an irc channel all day like i do (= |
| 15:20.07 | *** join/#asterisk Jhirley (~Jhirley@mail.mmdlaw.com) |
| 15:20.15 | Katty | hi Jhirley |
| 15:20.35 | redax | no problem. just ready what he asked. |
| 15:20.37 | Jhirley | o/ Hello My Dear !! |
| 15:20.45 | shader | so are you the official greeter of #asterisk, Katty? |
| 15:20.47 | Katty | redax: we're volunteers--not paid to be here and help people with their issues. |
| 15:21.00 | Katty | hugs Jhirley |
| 15:21.02 | Katty | shader: no |
| 15:21.15 | Katty | shader: however i've been here for the better part of 4 or 5 years, so i know several of the regulars. |
| 15:21.19 | redax | Katty: I know. sorry, I don't wated bugging anybody |
| 15:22.32 | Jhirley | In this day and age i will take a little kindness from any source. Cheers to all . |
| 15:22.44 | spenguin[work] | hallo Katty |
| 15:23.01 | Katty | herroes mister penguin |
| 15:23.22 | p3nguin | katty used to be the little old lady at Walmart before coming here. |
| 15:23.29 | Katty | giggles |
| 15:23.46 | Katty | before i went on a kill spree and burned the building to the floor ;) |
| 15:24.01 | Jhirley | In that case , I want my happy face sticker ! |
| 15:24.36 | Katty | are you under the age of 5? |
| 15:24.56 | shader | what would you do if he said yes? |
| 15:25.02 | Katty | give him a sticker. |
| 15:25.06 | Jhirley | emotionally . |
| 15:25.14 | Katty | i think that's good enough. |
| 15:25.19 | Katty | gives Jhirley a smiley sticker. |
| 15:25.35 | Jhirley | no all the 5 year olds are over in the MCSE channels. |
| 15:25.59 | Katty | frowns |
| 15:26.02 | huey23 | i just installed the vmail.cgi, on the $context=""; line, what is it asking for here? i have tried many different ways to login but I believe this holds the key |
| 15:26.02 | Katty | i am an mcse |
| 15:26.06 | Katty | thankyouverymuch |
| 15:26.16 | shader | mcse? |
| 15:26.34 | Katty | microsoft certified systems engineer |
| 15:26.35 | Jhirley | look up, youngest mcse. |
| 15:27.04 | patrb | wow |
| 15:27.06 | patrb | 10 year old |
| 15:27.07 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:27.32 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:28.50 | Jhirley | There was a 9 year old a few years back. |
| 15:28.59 | huey23 | yea, now she's 10 |
| 15:29.07 | Katty | poor kid. |
| 15:29.12 | shader | time passes quickly |
| 15:29.34 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 15:29.41 | shader | he's back! |
| 15:29.44 | Jhirley | Mind you I am saying that the MCSE is with our merrit, just that kids can accomplish almost anything. |
| 15:30.29 | Jhirley | Just like the space program with enought time and money you can do anything. Well they have time on their side. Where as I ahve a "Honey do list" |
| 15:30.56 | Katty | you should put Change Katty's Oil on the list |
| 15:32.10 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:32.10 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:33.33 | coppice | Katty: you mean you aren't well oiled? |
| 15:33.42 | Katty | obviously. |
| 15:35.45 | aceio | <patrb> i got the ipc to work SIP now |
| 15:36.30 | [TK]D-Fender | [11:26]<shader>mcse? <- Must Consult Someone Else |
| 15:36.58 | aceio | <patrb> status message is now TFTP error: SEP002100D1EC17.cnf.xml |
| 15:37.23 | Kobaz | and |
| 15:37.25 | Kobaz | er |
| 15:38.12 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 15:38.43 | patrb | aceio: you probably dont need to tftp boot your softphone |
| 15:39.11 | patrb | aceio: Ive never used that software though |
| 15:39.55 | *** join/#asterisk AsteriskNoob (~jp@host86-150-212-129.range86-150.btcentralplus.com) |
| 15:40.05 | AsteriskNoob | Hello... Total noob here. |
| 15:40.28 | aceio | <patrb> i see |
| 15:40.46 | AsteriskNoob | I have a few Asterisk problems and would love some help... |
| 15:40.49 | petern_ | the cnf.xml file containers useful things like where the CM host is |
| 15:40.50 | p3nguin | aceio: Why do you keep quoting patrb with things that patrb didn't actually say? |
| 15:41.02 | petern_ | -ers |
| 15:41.17 | p3nguin | <AsteriskNoob> Hello... Total noob here. <--- quoting asterisknoob |
| 15:41.32 | p3nguin | asterisknoob: Hello. <--- talking TO asterisknoob |
| 15:41.48 | AsteriskNoob | Hello |
| 15:41.49 | Jhirley | Dude Just ask your questions or they will keep messing with you. |
| 15:41.54 | AsteriskNoob | Oh, OK.. |
| 15:42.18 | AsteriskNoob | I just set up an AsteriskNow session and I cannot get call xfers to work. |
| 15:42.30 | AsteriskNoob | FreePBX is just about useless. |
| 15:42.32 | p3nguin | ~freepbx |
| 15:42.32 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:42.32 | shader | Jhirley: but messing with people is half the point of irc... |
| 15:43.02 | ManxPower-work | ~asterisknow |
| 15:43.02 | infobot | from memory, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 15:43.06 | Jhirley | Noob: you may want to join asteriskNow and #freepbx also |
| 15:43.10 | AsteriskNoob | I just need to know how to get * features to work. |
| 15:43.26 | AsteriskNoob | Jhirley: Thank you... I'll go there too. |
| 15:43.29 | patrb | AsteriskNoob: enable it in features.conf :) |
| 15:43.31 | p3nguin | With that being said, are you trying to transfer on the phone or from the GUI? |
| 15:43.34 | ManxPower-work | AsteriskNoob, Um. you realize that Asterisk is not really a PBX, right? |
| 15:43.54 | patrb | err * features can go in extensions.conf |
| 15:44.00 | ManxPower-work | Asterisk is a TOOLKIT that lets YOU build a PBX, much like a library helps a programmer write applications. |
| 15:44.08 | p3nguin | patrb: hmm? Features in extensions.conf? |
| 15:44.32 | p3nguin | manxpower-work: But he's using AsteriskNOW, so someone already built it for him. :/ |
| 15:44.35 | AsteriskNoob | ManxPower-work: I thought it was actually a full PBX. What am I missing? |
| 15:44.35 | patrb | p3nguin: if he means like dial *97 for voicemail |
| 15:44.46 | Jhirley | What I always thought the main reason for this channel was to mess with [TK]D-Fender ? |
| 15:44.54 | ManxPower-work | AsteriskNoob, So if you want, for example Call Forwarding, then you write the dialplan to create that feature. |
| 15:44.58 | p3nguin | patrb: Okay, that's just an extension. |
| 15:45.21 | ManxPower-work | AsteriskNoob, you are missing all the configuration files that you would need to write and configure to build a PBX. |
| 15:45.47 | AsteriskNoob | ManxPower-work: Yes, I understand the scripting, but I can't seem to find a proper script. I have a simple dialplan for incoming calls ready to deploy, but the call transferring is crucial. |
| 15:45.50 | ManxPower-work | AsteriskNoob, People call Asterisk a PBX, but it's not a PBX in the traditional sense. |
| 15:46.11 | AsteriskNoob | ManxPower-work: Gotcha... It's much more flexible.. |
| 15:46.23 | Jhirley | AsteriskNoob: The #freepbx channel may be a better place for you to for now, but this channel will help you later on . |
| 15:46.23 | ManxPower-work | AsteriskNoob, since each PBX is unique to the needs of the users, asterisk does not come with working config files. The config files are SAMPLES trying to show as much as possible. |
| 15:46.47 | p3nguin | AsteriskNOW, on the other hand, is ready to go... and what's what he's using! |
| 15:47.04 | AsteriskNoob | Hehe.. I WISH it was ready to go.. |
| 15:47.05 | huey23 | i just installed the vmail.cgi, on the $context=""; line, what is it asking for here? i have tried many different ways to login but I believe this holds the key | i am using asterisk 1.4.28 |
| 15:47.13 | Jhirley | Think of Asterisk as the engine, and free pbx is the steering wheel and blinkers and stuff. |
| 15:47.17 | p3nguin | asterisknoob: It is. I've used it. |
| 15:47.32 | ManxPower-work | A simple dialplan for incoming calls would be exten => s-or-did-or-exten-or-whatever-calls-come-in-as,1,Playback(we-hate-customers.gsm) |
| 15:47.56 | patrb | lol |
| 15:48.01 | ManxPower-work | huey23, I expect it would be the VOICEMAIL CONTEXT |
| 15:48.08 | AsteriskNoob | I guess if I used FreePBX to configure it things would be smoother, but I'm a bit stubborn.. I want to understand the scripting so I can play a bit more with it. |
| 15:48.23 | p3nguin | asterisknoob: Then stop using AsteriskNOW. |
| 15:48.25 | AsteriskNoob | ManxPower-work: Hahaha |
| 15:48.27 | ManxPower-work | AsteriskNoob, you do not want to mix FreePBX and hand built stuff. |
| 15:48.47 | *** join/#asterisk saftsack (~oliver@213.157.24.107) |
| 15:48.50 | patrb | AsteriskNoob: if you really want to learn asterisk, ditch the gui |
| 15:48.56 | AsteriskNoob | Yeah. I figured that out the hard way.. But, having said that, I don't use FreePBX.. |
| 15:48.58 | ManxPower-work | pick one. FreePBX = get up and running quickly, but complicated, limited, and unsupported on #asterisk. |
| 15:49.28 | ManxPower-work | Asterisk = HUGE learning curve, not as complicated as it seems once you get a few years of experience with it, and people on #asterisk can help you. |
| 15:49.32 | Jhirley | AsteriskNoob: The #freepbx channel may be a better place for you to for now, but this channel will help you later on . |
| 15:49.58 | AsteriskNoob | How do I get the features responding? It says that they are available, but when I press ## nothing happens.. |
| 15:50.08 | VoIP-Penguin | ~freepbx |
| 15:50.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:50.23 | huey23 | ManxPower-work: i understand that but does that mean that it is "default"...my context says context=default |
| 15:50.23 | ManxPower-work | AsteriskNoob, To use, all GUIs look the same. Regardless of it is FreePBX, AsteriskNOW, AsteriskGUI, or Bob's Asterisk GUIfication |
| 15:50.31 | aceio | <PROTECTED> |
| 15:50.40 | ManxPower-work | huey23, do you have a [default] section of voicemail.conf? |
| 15:51.20 | AsteriskNoob | Builtin Feature Default Current |
| 15:51.21 | AsteriskNoob | --------------- ------- ------- |
| 15:51.21 | AsteriskNoob | Pickup *8 *8 |
| 15:51.21 | AsteriskNoob | Blind Transfer # ## |
| 15:51.22 | AsteriskNoob | Attended Transfer *2 |
| 15:51.22 | AsteriskNoob | One Touch Monitor *1 |
| 15:51.22 | AsteriskNoob | Disconnect Call * ** |
| 15:51.23 | AsteriskNoob | Park Call |
| 15:51.23 | AsteriskNoob | Dynamic Feature Default Current |
| 15:51.24 | AsteriskNoob | --------------- ------- ------- |
| 15:51.24 | AsteriskNoob | (none) |
| 15:51.24 | AsteriskNoob | Call parking |
| 15:51.25 | AsteriskNoob | ------------ |
| 15:51.25 | AsteriskNoob | Parking extension :700 |
| 15:51.26 | AsteriskNoob | Parking context :parkedcalls |
| 15:51.26 | AsteriskNoob | Parked call extensions:701-750 |
| 15:51.32 | ManxPower-work | some kick him |
| 15:51.34 | VoIP-Penguin | Don't do that. |
| 15:51.36 | AsteriskNoob | Sorry.. Hopefully this helps a bit. |
| 15:51.39 | AsteriskNoob | Sorry.. |
| 15:51.40 | VoIP-Penguin | ~pb |
| 15:51.41 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:51.41 | huey23 | ManxPower-work: unfortunately it is a "real-time" voicemail.conf...there is nothing in the voicemail.conf file |
| 15:51.41 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 15:51.49 | ManxPower-work | huey23, It sucks to be you. |
| 15:51.52 | VoIP-Penguin | PASTEBIN ^^^^^^ |
| 15:52.07 | huey23 | ManxPower-work: it's not that bad...i've lived with myself all my life |
| 15:52.07 | ManxPower-work | I suspect vmail.cgi may not even work with Realtime, but who knows. |
| 15:52.10 | *** join/#asterisk ShaunR (~shaun@staff.ndchost.com) |
| 15:52.15 | aceio | <PROTECTED> |
| 15:52.24 | ManxPower-work | aceio, contact Cisco. |
| 15:52.39 | ShaunR | whats a good, simple, robust SIP device that will allow a analog phone to plug into it. |
| 15:52.50 | VoIP-Penguin | shaunr: PAP2T |
| 15:52.58 | ShaunR | i had a iaxy unit but it died, also it was a pain in the ass to configure.. |
| 15:53.02 | aceio | yes cheers for that answer |
| 15:53.14 | huey23 | ManxPower-work: i am able to get to the web interface, i am just trying to figure out how to login |
| 15:53.40 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 15:53.45 | ShaunR | VoIP-Penguin: this? http://www.newegg.com/Product/Product.aspx?Item=N82E16833150031&Tpk=PAP2T |
| 15:53.54 | ManxPower-work | aceio, it's not like anyone here would know what the problem is. |
| 15:54.08 | AsteriskNoob | Can anyone send me a small script for activating call transfer? |
| 15:54.18 | AsteriskNoob | I simply can't seem to find one that makes sense. |
| 15:54.19 | VoIP-Penguin | shaunr: That's a PAP2T. |
| 15:54.22 | ManxPower-work | AsteriskNoob, there are none. |
| 15:54.28 | AsteriskNoob | Oh.. |
| 15:54.32 | AsteriskNoob | Hmmm. |
| 15:54.36 | ManxPower-work | Personally, I just press the fsckin' TRANSFER button on my phone. |
| 15:54.40 | VoIP-Penguin | shaunr: You should be able to find one on ebay for about $30 less. |
| 15:54.44 | ShaunR | VoIP-Penguin: ok i just saw the extra -na so i wanted to make sure.. |
| 15:54.58 | VoIP-Penguin | shaunr: -NA = North America |
| 15:54.58 | AsteriskNoob | So if it's active, ##EXT should transfer the call.. |
| 15:55.14 | *** join/#asterisk goofy03 (~kvirc@cha42-1-89-90-8-114.dsl.club-internet.fr) |
| 15:55.18 | goofy03 | hi |
| 15:55.28 | VoIP-Penguin | asterisknoob: That's a DTMF transfer... and it's TOTALLY not the same as a SIP device's transfer. |
| 15:55.29 | ManxPower-work | AsteriskNoob, no, if it's active, WHATEVER YOU CONFIGURED will transfer the call. Of course you also need the T and/or t options to Dial, |
| 15:55.29 | AsteriskNoob | Oh. I'm using free softphones. Phone coming tomorrow. |
| 15:55.49 | *** join/#asterisk hubbaba (~hubbaba@216.0.61.2) |
| 15:56.24 | AsteriskNoob | Ahh.. So if I add t to the Dial() app it should do it then... |
| 15:56.32 | AsteriskNoob | I'll give that a try. |
| 15:56.44 | goofy03 | i try to compile asterisk-1.6.1 from svn but i have many errors with app_nv_backgrounddetect-1.0.6_1-foras1.4.23.1.o is this module is always require too autodetect fax and forward it to hylafax ? |
| 15:56.49 | patrb | AsteriskNoob: yes, that allows the callee? permission to transfer |
| 15:56.51 | VoIP-Penguin | watches asterisknoob break his configs |
| 15:57.15 | AsteriskNoob | Hehehe VoIP-Penguin .. Likely. Thankj goodness it's not live yet.. |
| 15:57.29 | ManxPower-work | VoIP-Penguin, I already put him on /ignore |
| 15:57.58 | AsteriskNoob | Didn't mean to upset you ManxPower-work .... |
| 15:58.06 | ManxPower-work | ~answers |
| 15:58.07 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 15:58.20 | hubbaba | when using fxsgs over a T1, we're getting roughly a 4 second delay before the call is answered. We see Got event 18 (Ring Begin)... and Got event 2 (Ring/Answered)... The, 4 seconds later, we get Got event 18 (Ring Begin)... and our start extension executes. |
| 15:58.29 | ManxPower-work | Any n00b should be reading the Asterisk book |
| 15:58.44 | hubbaba | i have usecallerid=no in chan_dahdi and faxdetect is not being used |
| 15:58.50 | hubbaba | any ideas why it would take so long? |
| 15:59.13 | ManxPower-work | hubbaba, why are you using Ground Start instead of the common Loop Start? |
| 15:59.39 | hubbaba | because that is what our customer is requiring |
| 15:59.55 | ManxPower-work | pastebin the relevant configs as well as the cli output of a problem call |
| 16:00.19 | AsteriskNoob | ManxPower-work: The Asterisk book has been helpful and confusing at the same time.. Some good examples, others not so good, and certainly not written for a total noob. I've had a crash course in Ast over the last few days, but I'll crack it. |
| 16:00.38 | hubbaba | believe me, I'm be using PRI or e&m if I could. We're working a very old CTI based system. |
| 16:00.46 | VoIP-Penguin | Once you are no longer using FreePBX, we can help you. |
| 16:00.47 | AsteriskNoob | ManxPower-work: Thanks for the bit about the T option.. That's what I was missing. |
| 16:01.05 | Katty | formats her phone |
| 16:01.13 | Katty | when you format a phone... |
| 16:01.16 | VoIP-Penguin | Use XFS or JFS! |
| 16:01.17 | patrb | AsteriskNoob: there is a difference between the T option and the t option |
| 16:01.18 | Katty | what happens with the firmware |
| 16:01.26 | Katty | does it use the last firmware it recieved from ftp |
| 16:01.27 | patrb | AsteriskNoob: make sure you know the idfference |
| 16:01.33 | Katty | or whatever firmware came with the phone when it was shipped |
| 16:01.41 | VoIP-Penguin | core show application Dial |
| 16:01.45 | AsteriskNoob | patrb: Thanks. I'll make sure to use the right options. |
| 16:01.51 | AsteriskNoob | Thanks everyone.. |
| 16:03.23 | spenguin[work] | Katty: what phone |
| 16:03.31 | Katty | spenguin[work]: polcyom |
| 16:03.42 | Katty | spenguin[work]: it'll be back up in a minute |
| 16:03.51 | Katty | spenguin[work]: i can check the firmware version against my ftp bootrom |
| 16:04.37 | spenguin[work] | kk, is it the one which keeps dying after a while? |
| 16:04.46 | Katty | mmmmno? |
| 16:04.51 | Katty | i have nothing dying over here |
| 16:05.00 | Katty | i do however have quirky dtmf problems |
| 16:05.24 | Katty | debug shows it only recieves a portion of digits, even tho i know i put all the digits in from my phone |
| 16:05.27 | Katty | zoiper is fine |
| 16:05.46 | Katty | it's also sporatic, and not consitently failing |
| 16:06.13 | *** join/#asterisk Citrus2 (~citrus@wsip-98-173-200-235.sb.sd.cox.net) |
| 16:07.06 | goofy03 | i try to compile asterisk-1.6.1 from svn but i have many errors with app_nv_backgrounddetect-1.0.6_1-foras1.4.23.1.o is this module is always require too autodetect fax and forward it to hylafax ? |
| 16:07.22 | goofy03 | i'am on debian stable |
| 16:08.23 | AsteriskNoob | Thank you everyone. That t option was the missing piece. Works perfectly... |
| 16:08.32 | AsteriskNoob | And I'm off. |
| 16:09.15 | Katty | that's kinda like showing up, eating, and leaving |
| 16:10.02 | ManxPower-work | goofy03, Looks that file is not part of Asterisk |
| 16:11.10 | goofy03 | ManxPower-work: ok and it is the only solution to autodetect faxs ? |
| 16:12.03 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 16:12.11 | Chainsaw | goofy03: There is internal fax detection in Asterisk, at least as of 1.6.2 and probably earlier. |
| 16:13.09 | goofy03 | Chainsaw: and can i use it to redirect fax call too hylafax ? |
| 16:13.26 | Chainsaw | goofy03: Quite likely, it behaves the same. It jumps to the fax priority. |
| 16:14.00 | spenguin[work] | Katty: ah well you mentioned of some polycom phone which was acting funny - after 200 calls |
| 16:14.09 | goofy03 | ok then i must compile 1.6.2 in place of 1.6.1 ? |
| 16:14.26 | ManxPower-work | Some of our sales reps need this shirt: http://www.thinkgeek.com/tshirts-apparel/unisex/generic/aa00/ |
| 16:16.03 | Katty | spenguin[work]: yeah dtmf. |
| 16:16.23 | Katty | spenguin[work]: turns out it's not 200 calls.... it's more like 25% of them |
| 16:16.48 | Katty | spenguin[work]: but i never noticed, and most of us didn't, because when we dial extensions we put in 1 number. |
| 16:16.52 | Katty | spenguin[work]: not 4 digits. |
| 16:17.15 | Katty | spenguin[work]: i'm thinking perhaps digitmapping or some other ftp setting i have somewhere is messing it up--hence the back to factory default setting |
| 16:17.29 | spenguin[work] | hrm, Ive seen some dtmf issues, but those were cause the extensions were like 2222 or numbers like that |
| 16:17.37 | spenguin[work] | they were being punched in too quick |
| 16:17.41 | Katty | this particular one is 7744 |
| 16:18.08 | Katty | and i've been dialing it at a reasonable speed |
| 16:18.29 | *** join/#asterisk norrec (~norrec@76-201-85-140.lightspeed.frokca.sbcglobal.net) |
| 16:18.51 | Chainsaw | goofy03: As long as it's recent, 1.6.1 might work. |
| 16:18.55 | VoIP-Penguin | If your line is "up" (when you're in a call), then the digitmap won't be applied, will it? |
| 16:19.08 | Katty | no idea |
| 16:19.15 | VoIP-Penguin | It's not supposed to be. |
| 16:19.26 | Katty | i don't care |
| 16:19.33 | VoIP-Penguin | The digitmap is only for placing a call. |
| 16:19.35 | Chainsaw | It defaults to not doing it, but you can change that setting. |
| 16:19.46 | Chainsaw | So it might be applying a digit map even in that situation. |
| 16:20.05 | VoIP-Penguin | Eww, a setting for applying digitmap during a call? |
| 16:20.23 | Katty | i'm sure it has a reason for being there |
| 16:20.26 | Chainsaw | VoIP-Penguin: Yes, I believe if you were suitably evil, you can even have it apply the digit map to calls from the main screen (without hitting a line key at all). |
| 16:20.30 | VoIP-Penguin | Someone needs to be dismantled over that "feature." |
| 16:20.31 | Katty | so just because you don't get it doesn't mean you get to eww about it |
| 16:21.10 | norrec | so i'm running asterisk 1.6.0.24 and i have devices that support T.38 however not all my trunks support t.38 but i dont want to have to use different configs on my devices, is there a way for asterisk to convert from T.38 to, for lack of a better discription, inband (T.30 i believe it is) faxing based on the outbound trunk it uses? |
| 16:21.12 | *** join/#asterisk MoreAllLess (~Justo@cpe-76-169-252-172.socal.res.rr.com) |
| 16:22.05 | *** join/#asterisk farkus (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 16:22.22 | ManxPower-work | The Polycom digit maps work great if you have a well designed dialplan. |
| 16:23.43 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:25.43 | spenguin[work] | what defines if an agent is supposed to login to the realtime queue or not |
| 16:26.15 | VoIP-Penguin | management, probably. |
| 16:26.24 | spenguin[work] | I know its possible to have a user in queue but still recv calls |
| 16:27.24 | spenguin[work] | like in the queue_member_table, Ive got Agent/<EXT> |
| 16:27.35 | spenguin[work] | or SIP/<EXT> |
| 16:29.00 | *** join/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 16:29.00 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:30.13 | Chainsaw | It seems the digit maps can't do secondary dialtone twice, which is a shame. (I was hoping to respond appropriately to several international dialling codes) |
| 16:30.41 | Chainsaw | So *dialtone* 9 *dialtone* 0031 *dialtone* |
| 16:30.58 | Chainsaw | Wait for 9 numbers then send. |
| 16:31.32 | ManxPower-work | |9,0031| and exten => 0031,1,Playtones(dialtone) exten => 0031,n,WaitExten |
| 16:33.43 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 16:34.00 | VoIP-Penguin | I would rather use DISA because the dialtone won't cancel when you enter only part of the exten. |
| 16:34.14 | VoIP-Penguin | Use DISA so it will cancel, I mean. |
| 16:34.34 | VoIP-Penguin | It's annoying to hear a dialtone the whole time you're dialing numbers. |
| 16:37.38 | spenguin[work] | SO if a queue member is added like SIP/<EXT>, the user at that ext doesnt have to login to the queue |
| 16:37.46 | spenguin[work] | ready or not the calls would arrive |
| 16:37.57 | spenguin[work] | but if its AGENT/<EXT> |
| 16:38.02 | spenguin[work] | he needs to login |
| 16:38.13 | spenguin[work] | s/he |
| 16:38.19 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 16:38.34 | QbY | any dahdi experts available who want to consult.. |
| 16:38.55 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 16:39.47 | VoIP-Penguin | Well, neither Agent channels nor SIP channels have extensions... |
| 16:40.51 | Katty | sits and waits for bootrom to update |
| 16:40.56 | Katty | twiddles fingers |
| 16:41.33 | VoIP-Penguin | The users login from their devices (usually phones). If the queue member is a SIP device, then the phone device is an agent in the queue. |
| 16:42.14 | VoIP-Penguin | If the queue member is an agent channel and device, then the agent must be logged in to receive a call from the queue. |
| 16:42.22 | spenguin[work] | k |
| 16:45.46 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:46.02 | Katty | http://www.youtube.com/watch?v=32vpgNiAH60 <- pure awesome. |
| 16:47.20 | *** join/#asterisk aandrade (~aandrade@187.58.245.245) |
| 16:47.37 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 16:48.50 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 16:50.55 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 16:51.13 | Katty | mister monk |
| 16:51.27 | voipmonk | hello there |
| 16:52.20 | Katty | woo! finally. phones back up (= |
| 16:53.07 | gr0mit | root |
| 16:53.10 | Katty | waits for httpd to come up |
| 16:53.14 | VoIP-Penguin | pictures phones driving in reverse |
| 16:53.28 | patrb | hehe |
| 16:54.19 | Katty | anddddddddddd rebooting again >.< |
| 16:55.13 | Katty | looks like i was working on a slight older revision |
| 16:55.20 | Katty | or..still am. actually, just not this particular phone |
| 16:55.35 | Katty | for testing porposes |
| 16:55.42 | Katty | woo! |
| 16:56.11 | Katty | tests dtmf |
| 16:56.46 | *** join/#asterisk Z_God (~julius@wlan227220.mobiel.utwente.nl) |
| 16:56.49 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:57.10 | Katty | so far so good |
| 16:57.32 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:58.04 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:58.14 | Katty | hi Naikrovek |
| 16:58.20 | Naikrovek | hello |
| 16:58.36 | Naikrovek | anyone know of any polycom resellers in india |
| 16:58.52 | Katty | okay. |
| 16:58.53 | iam8up | did you check alibaba? |
| 16:58.57 | Katty | so it's still doing it. |
| 16:59.04 | Katty | but it's like the phone is..lagging |
| 16:59.05 | Naikrovek | alibaba? |
| 16:59.09 | Naikrovek | heh |
| 16:59.20 | Katty | cause the dtmf on debug will show 77, and then the next one will be like 774477 |
| 16:59.28 | Katty | or whatever i left out will get picked up |
| 16:59.30 | Katty | at the end |
| 16:59.30 | iam8up | http://www.alibaba.com/trade/search?SearchText=polycom&Country=IN&IndexArea=product_en&ssk=y |
| 16:59.32 | iam8up | polycom in india |
| 16:59.49 | Katty | Qwell: ping |
| 17:02.15 | *** join/#asterisk kbr (~kbr@ASte-Genev-Bois-152-1-19-61.w83-114.abo.wanadoo.fr) |
| 17:03.36 | spenguin[work] | timeout |
| 17:03.42 | *** join/#asterisk generalhan (~asd@about/windows/staff/generalhan) |
| 17:03.47 | Katty | what kind of timeout |
| 17:04.01 | spenguin[work] | dunno, i r dumb birdbrain |
| 17:04.04 | Katty | hrm |
| 17:04.38 | Katty | well this is what happens, i put in 7744. server recieves 77. then i put in 7744 again....then it shows 744744 or 744474 |
| 17:06.29 | spenguin[work] | is the server local or remote? |
| 17:06.46 | Katty | local |
| 17:06.55 | Katty | zoiper on same local network, seems fine with transmitting dtmf information |
| 17:07.23 | spenguin[work] | hrm, wash polycom phone in warm water, scrub gently |
| 17:07.36 | Katty | yesh |
| 17:08.43 | *** join/#asterisk eppigy (~eppigy@c-69-180-16-188.hsd1.ga.comcast.net) |
| 17:10.54 | Nugget | heh |
| 17:12.40 | spenguin[work] | why does this not work |
| 17:12.44 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 17:12.47 | spenguin[work] | exten => _[1-3],n,Queue... |
| 17:12.56 | spenguin[work] | exten => s,n,Voicemail ... |
| 17:13.13 | spenguin[work] | should it be exten => _[1-3],n,Voicemail.. ? |
| 17:13.46 | Naikrovek | iam8up: thanks |
| 17:13.47 | keith4 | anyone have an opinion on the Xorcom Astribank products? |
| 17:14.08 | iam8up | Naikrovek, that's $32.50 |
| 17:14.19 | Naikrovek | sends busy signal |
| 17:14.23 | Naikrovek | no comprende |
| 17:14.34 | iam8up | lol |
| 17:14.47 | spenguin[work] | $32 is cheap |
| 17:14.55 | spenguin[work] | liek real cheap |
| 17:15.01 | Naikrovek | for what |
| 17:15.01 | VoIP-Penguin | Well that's friggin' annoying... I was going to check dtmf in my log during the usual verizon voicemail failure... and it ACCEPTED MY PASSWORD for once. |
| 17:15.07 | spenguin[work] | for a polycom phone? |
| 17:15.07 | iam8up | for my help |
| 17:15.34 | iam8up | VoIP-Penguin, isn't dtmf awesome? |
| 17:15.38 | spenguin[work] | VoIP-Penguin: IT tends to be annoying like that |
| 17:15.43 | iam8up | DTMF + sip = hell |
| 17:15.59 | Naikrovek | inband DTMF yes |
| 17:15.59 | VoIP-Penguin | I think it's like the second time ever that it accepted the numbers. |
| 17:16.03 | Naikrovek | inband just works mostly |
| 17:16.11 | Naikrovek | erm |
| 17:16.15 | Naikrovek | out of band just works mostly |
| 17:16.53 | *** join/#asterisk Godfather_ (~Godfather@157.Red-88-11-88.dynamicIP.rima-tde.net) |
| 17:16.58 | VoIP-Penguin | But checking the log, it didn't even register all the numbers I typed. |
| 17:17.17 | QbY | What needs to be loaded to eliminate this: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented) |
| 17:17.35 | VoIP-Penguin | Don't call devices on dahdi channels. |
| 17:17.52 | VoIP-Penguin | or put devices on dahdi channels before calling them. |
| 17:24.08 | [TK]D-Fender | QbY: chan_dahdi.so is not loaded |
| 17:24.10 | Katty | http://pastebin.comm/uSc5cYEW <- if anyone has any thoughts on this DTMF issue, lemme know |
| 17:24.25 | QbY | [TK]D-Fender: .. i have it loaded now |
| 17:24.32 | QbY | i stopped everyhting, b rought up dahdi first |
| 17:24.41 | QbY | now at least calls are flowing |
| 17:25.11 | idespinner | relaxdtmf on zap? |
| 17:25.18 | Katty | it's set to yes. |
| 17:25.26 | Katty | i guess i should pastebin that stuff too tho |
| 17:25.44 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 17:26.26 | VoIP-Penguin | For some reason, I was under the impression that the dtmf problem was on sip rather than zap. |
| 17:26.27 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:27.19 | [TK]D-Fender | Katty: does it double direcft to * only? |
| 17:27.40 | Katty | syntax error near 'double' |
| 17:27.45 | *** join/#asterisk ruchir (~ruchir@204-133-215-130.dia.static.qwest.net) |
| 17:27.52 | Katty | directly? |
| 17:28.02 | ruchir | does anyone know how to convert odbc voicemail to wav file? |
| 17:28.28 | [TK]D-Fender | Katty: phone -> *. Does it double within that leg alone? |
| 17:28.36 | *** join/#asterisk dennisG (~root@84.30.136.208) |
| 17:28.42 | VoIP-Penguin | Isn't the phone a SIP phone? |
| 17:28.52 | Katty | well. |
| 17:28.54 | VoIP-Penguin | Why is chan_zap involved? |
| 17:28.57 | Katty | it goes through networking equipment |
| 17:29.06 | Katty | but i suppose i could put it on the same switch |
| 17:29.10 | Katty | otherwise, the answer is yes |
| 17:29.31 | Katty | tho zoiper seems to be fine. |
| 17:29.42 | Katty | which is on my workstation, on the same switch |
| 17:30.29 | VoIP-Penguin | Where is ZAPTEL coming into this problem? |
| 17:31.32 | Katty | the problem occurs before sending it out the pri, VoIP-Penguin |
| 17:31.40 | ruchir | odbc vm anyone |
| 17:31.42 | ruchir | ? |
| 17:32.06 | Katty | actually |
| 17:32.22 | Katty | [TK]D-Fender: does Read and SayDigits use dtmf? |
| 17:32.26 | Katty | [TK]D-Fender: or does it use something else? |
| 17:32.35 | [TK]D-Fender | Katty: Clearly yes |
| 17:32.37 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 17:32.40 | Katty | k |
| 17:33.16 | Katty | i dont' think the cli shows information about dtmf, just the debug log |
| 17:33.34 | Katty | i think |
| 17:33.35 | Katty | checks |
| 17:33.58 | ruchir | saydigits use playback |
| 17:34.10 | ruchir | senddtmf use dtmf |
| 17:34.46 | VoIP-Penguin | ruchir: The question was about "Read and SayDigits" in combination. |
| 17:35.18 | ruchir | hmm |
| 17:35.46 | Katty | yeah it shows the numbers |
| 17:36.14 | VoIP-Penguin | read() obviously has to read the dtmf tones in order to store them in the variable in order for SayDigits() to playback the sound files of the numbers it recognizes. |
| 17:36.41 | ruchir | right |
| 17:37.25 | Katty | [TK]D-Fender: i misunderstood your question |
| 17:37.44 | [TK]D-Fender | Katty: Verify the SIP>* leg first. |
| 17:37.49 | Katty | yeah i'm testing that now |
| 17:37.56 | Katty | 10 in a row, no issue |
| 17:38.17 | VoIP-Penguin | My DTMF problem never seems to occur between the phone and Asterisk, but does happen on a bridged call between the phone-asterisk-verizon. |
| 17:38.19 | Katty | i was under the assumption that the debug just showed what it recieved directly FROM the polycom |
| 17:38.26 | Katty | which isn't the case |
| 17:38.42 | Katty | i'm having no issue doing a read and playback |
| 17:38.46 | Katty | err saydigits |
| 17:39.09 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.32.97) |
| 17:42.04 | Katty | [TK]D-Fender: so if from phone to * is good, but asterisk -> telco is snickerdoodled... |
| 17:42.20 | Katty | [TK]D-Fender: and debug shows the dtmf stuff quirky |
| 17:42.29 | Katty | [TK]D-Fender: what's the next step? sangoma card? telco? |
| 17:42.42 | [TK]D-Fender | Katty: check for echo/gain issues |
| 17:42.59 | Katty | k |
| 17:43.26 | huey23 | has anyone had any luck successfully using a real-time voicemail solution and the vmail.cgi gui? |
| 17:45.54 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 17:47.03 | ruchir | huey23: we use either of them havent tried both |
| 17:47.25 | ruchir | u mean realtime vm users not odbc vm storage, right? |
| 17:47.56 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 17:47.57 | huey23 | correct |
| 17:48.18 | huey23 | the vm is still stored in the same dir structure as before, but the vm users are in the db |
| 17:50.48 | ruchir | i dont remember how vm.cgi used to check auth |
| 17:50.49 | huey23 | Fender would be proud, we finally upgraded from 1.0 to 1.4 but in doing so, someone thought that it would be easier to change things and add users |
| 17:50.59 | ruchir | but any case it shouldnt be difficult to modify just in case |
| 17:51.03 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:51.20 | huey23 | the gui is working...i am just having trouble logging in |
| 17:51.53 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:51.59 | huey23 | i imagine i have to make some changes in vmail.cgi to look at the db to get credentials from but i was wondering if it would even work before i wrapped my brain around it |
| 17:53.39 | ruchir | yeah |
| 17:54.21 | ruchir | is looking for way to convert odbc vm to wav file |
| 17:56.52 | *** join/#asterisk peterhup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 17:57.55 | Katty | [TK]D-Fender: will a debug show echo/gain issues? |
| 17:58.04 | Katty | [TK]D-Fender: the zoiper out to the telco seems to send dtmf fine |
| 17:58.19 | Katty | [TK]D-Fender: if it's echo/gain issues relating specifically to the polycoms, i'm not sure how to troubleshoot that |
| 17:58.32 | ManxPower-work | I see it, but I wonder if anyone else sees the problem with this number: 011 39 091 8689XXX |
| 17:58.59 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:59.51 | bmoraca_work | it's way long |
| 18:00.27 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:00.48 | ManxPower-work | bmoraca_work, 91 would be a city code and you don't put leading 0 on city codes when calling from outside that country |
| 18:01.16 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:01.18 | bmoraca_work | gotcha |
| 18:01.23 | VoIP-Penguin | What's the effective difference between playback(silence/1) and wait(1) if you are only trying to pause dialplan progression for 1 second (have 1 second of silence on the line)? |
| 18:01.28 | ManxPower-work | nobody in sales or tech support caught that |
| 18:01.36 | Katty | ManxPower-work: i doubt i would have |
| 18:01.40 | ManxPower-work | VoIP-Penguin, nothing |
| 18:01.40 | bmoraca_work | i don't do international dialing a lot...not a big thing for dental offices |
| 18:01.44 | DocAwesome | VoIP-Penguin: one of them sends audio |
| 18:01.49 | micols | how does "call waiting" look in asterisk log (/var/log/asterisk/full) ? |
| 18:01.51 | ManxPower-work | bmoraca_work, me neither |
| 18:02.04 | ManxPower-work | micols, depends on the technology used |
| 18:02.12 | micols | isdn30 card |
| 18:02.17 | *** join/#asterisk babbio (~somebody@host-78-13-24-238.cust-adsl.tiscali.it) |
| 18:02.24 | ManxPower-work | you can't do call waiting on ISDN with Asterisk |
| 18:02.51 | VoIP-Penguin | From a functional standpoint, should one method be preferred over the other? |
| 18:03.00 | micols | sorry, the stream is SIP.. |
| 18:03.02 | coppice | bmoraca_work: you haven't outsourced your basic flossing and polishing work to China? :-) |
| 18:03.06 | bmoraca_work | ISDN is "always available" anyway...call waiting is redundant |
| 18:03.21 | bmoraca_work | coppice: i'm THIS close: |-| |
| 18:03.27 | Slugs_ | Can somebody help me connect my asterisk to avaya pbx via h323? Ive set it up on avaya fine, im just working on the asterisk end. any help would be greatly appreciated |
| 18:03.31 | babbio | hi guys i'm having a problem with X100P board and dahdi module....i can't make asterisk works....please this is my first time with asterisk......somebody coud help? i will appreciate |
| 18:03.40 | ManxPower-work | micols, it would look just like another call. There is no such thing as call waiting in SIP. It's up to the phone on how it does a 2nd call. |
| 18:03.43 | leifmadsen | ~ask |
| 18:03.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:03.55 | [TK]D-Fender | Slugs_: FreePBX is NOT supported in here |
| 18:04.01 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 18:04.16 | bmoraca_work | micols: if the phone is allowed to take two calls, the second call will go through...if the phone is not, you'll get a "busy here" or a message about call-limit being reached |
| 18:04.20 | Slugs_ | im not using frr pbx |
| 18:04.23 | moos3 | how can I make the Directory search both first and last name |
| 18:04.25 | beek | [TK]D-Fender: Damn, you're good. He hasn't even mentioned that yet! |
| 18:04.26 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 18:04.32 | ManxPower-work | [TK]D-Fender, do you think we should just ban Slugs_? |
| 18:04.41 | Slugs_ | im using an installatoin w/ no free pb now |
| 18:04.47 | ManxPower-work | moos3, the same way you do anything in asterisk "core show application directory" |
| 18:04.57 | [TK]D-Fender | Slugs_: Then show us a failed attempt to process a call |
| 18:05.14 | Slugs_ | k |
| 18:05.33 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 18:06.12 | ManxPower-work | beek, he's been coming into the channel with FreePBX questions for WEEKS. |
| 18:06.28 | bmoraca_work | damnit |
| 18:06.29 | ManxPower-work | moos3, if it's not documented there, then it can't be done. |
| 18:06.32 | moos3 | ManxPower-work, heres what I have exten => s,1,Directory(sales,ef) but if I enter HOS for hosting solutions, I'm getting horizon and then insight |
| 18:06.51 | ManxPower-work | moos3, I'm happy for you. Are you using Asterisk 1.4.x? |
| 18:06.51 | moos3 | the other too are directory options tho |
| 18:06.57 | moos3 | no 1.6.x |
| 18:07.16 | ManxPower-work | Then I doubt me looking at "core show application directory" on my system is going to help you much. |
| 18:07.53 | moos3 | ManxPower-work, I'm call directory correctly? |
| 18:07.55 | micols | bmoraca_work: of course, but can you see if the phone supports call waiting in this log and if it is forwarded? http://pastebin.com/LgFt9KUC |
| 18:08.05 | peterhup | I am new to Asterisk, can anybody recommend a learning path for an experienced developer? |
| 18:08.08 | ManxPower-work | moos3, Oh, and you are not using 1.6.x 1.6.x does not exist. 1.6.1.x and 1.6.2.x, etc |
| 18:08.15 | ManxPower-work | ~answers |
| 18:08.16 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 18:08.23 | Slugs_ | http://pastebin.org/117121 |
| 18:08.42 | ManxPower-work | micols, Asterisk DOES NOT know if the phone support call waiting or not. |
| 18:08.47 | ManxPower-work | Asterisk just SENDS THE CALL. |
| 18:08.50 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 18:08.54 | Kobaz | babbio: what problems are you having |
| 18:08.55 | [TK]D-Fender | Slugs_: that isn't an entire call and has no background about what is what |
| 18:09.05 | moos3 | ManxPower-work, sorry 1.6.0.x |
| 18:09.09 | [TK]D-Fender | Kobaz: Did you get PM'd too? |
| 18:09.17 | bmoraca_work | micols: you don't CALL a PHONE in that snippet of log |
| 18:09.17 | Kobaz | yeah |
| 18:09.25 | Kobaz | 14:07 <babbio> hi kobaz do u remember of me?? i'm still having my problem with asterisk can u help me? |
| 18:09.28 | ManxPower-work | moos3, does "core show application directory" show you any options for searching both first and last names? |
| 18:09.34 | Kobaz | no, i don't remember everyone who asked a question |
| 18:09.45 | moos3 | yeah f |
| 18:11.14 | ManxPower-work | babbio*@* added to ignore list. |
| 18:11.34 | babbio | i'mtrying to build a simple aswering machine, this is my extensions.conf http://pastebin.ubuntu.com/397381/ |
| 18:12.01 | [TK]D-Fender | babbio: exten => s,n,Playback(/home/paramore_ignorance.mp3) <-- NEVER put the extension of the file in Playback() |
| 18:12.24 | Nugget | yeah, that's like crossing the streams |
| 18:12.43 | ManxPower-work | in the snow! |
| 18:12.46 | [TK]D-Fender | doesn't want ever atom in his body to explode at the speed of light |
| 18:12.54 | [TK]D-Fender | every* |
| 18:13.12 | Katty | hrmm. full log says nothing about echo/gain anything |
| 18:13.13 | vader-- | any of you guys use SIPXecs? |
| 18:13.25 | Katty | [TK]D-Fender: where and how do i go about looking for echo and gain issues |
| 18:13.45 | bmoraca_work | vader--: how's your TA924e going? |
| 18:14.00 | ManxPower-work | Katty, Echo must be removed where the call is coverted from PSTN to VoIP. |
| 18:14.48 | Katty | ManxPower-work: how i take echo off my call from Polycom -> asterisk -> pri -> telco? |
| 18:14.56 | Katty | ManxPower-work: at which stage, and how |
| 18:15.01 | ruchir | can someone point me in proper direction to convert odbc stored vm to wav file for playback on webpage? |
| 18:15.05 | *** part/#asterisk peterhup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 18:15.09 | ManxPower-work | Katty, remove it at the Asterisk/PRI point. |
| 18:15.13 | bmoraca_work | Katty: echo cancelling PRI card in asterisk...really the only place |
| 18:15.19 | Katty | ah okay |
| 18:15.34 | bmoraca_work | or use an Adtran TA900 and convert it to SIP there...works great and they have great echo cancellers |
| 18:15.45 | babbio | [TK]D-Fender: ok but the problem is another one....dahdi tell me that i have no active channel, http://pastebin.ubuntu.com/397385/ and this the the output of dahdi show channel and dahdi show status http://pastebin.ubuntu.com/397387/ |
| 18:15.46 | ManxPower-work | ruchir, use the wav49 format for voicemail messages |
| 18:15.51 | Katty | well i have echocancel=yes and echocancelwhenbridged=yes |
| 18:15.58 | ManxPower-work | Katty, do you have hardware EC? |
| 18:16.04 | babbio | why i have no channel in the dahdi show channels if i have a channel in the dahdi show status? |
| 18:16.16 | Katty | ManxPower-work: i'm pretty sure the card does have onboard echo cancelation yes |
| 18:16.23 | ManxPower-work | Katty, Digium or Sangoma? |
| 18:16.25 | Katty | ManxPower-work: i would have to go check tho |
| 18:16.27 | Katty | ManxPower-work: sangoma |
| 18:16.38 | Katty | ManxPower-work: i don't honestly remember which type of card is sitting in that one |
| 18:16.42 | Katty | checks |
| 18:17.16 | bmoraca_work | i honestly have started prefering to use the external PRI->SIP gateways...they have better quality from what I've seen. |
| 18:17.24 | ruchir | ManxPower-Work: how does it create wav file from mysql db record? |
| 18:17.30 | ManxPower-work | Katty, 1) I've never had echo with Sangoma and 2) I've never been able to totally remove echo using software EC 3) I've never solved an echo issue when using a Digium EC card. |
| 18:17.40 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 18:17.55 | ManxPower-work | ruchir, I have no idea, but if you want something to play in a web page, quickly then you'll use the wav49 voicemail format. |
| 18:17.58 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:18.21 | ruchir | playback is not major issue |
| 18:18.28 | Katty | well the card has a102 stamped on it, but it's not a dual t1 card |
| 18:18.30 | ruchir | issue is to convert db record to file |
| 18:18.38 | ruchir | so ew can use it for playback |
| 18:18.39 | Kobaz | music on answering machines is ghetto |
| 18:18.43 | bmoraca_work | ruchir: it just stores the file in a BLOB field. the file is identical to if it were stored on the filesystem itself. the point of using MySQL for voicemail is to centralize it and make it easily available to other applications that may or maynot reside on the server |
| 18:19.11 | [TK]D-Fender | babbio: ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings <----------------- |
| 18:19.27 | ruchir | bmoraca_Work: that means if i read the record and store in file as is and name it wav |
| 18:19.29 | [TK]D-Fender | babbio: I guess you didn't read the giant notice. Where is your chan_dahdi.conf? |
| 18:19.30 | ruchir | can i play it? |
| 18:19.33 | Katty | well let's assume this isn't an echo issue |
| 18:19.45 | Katty | how would i go about testing a gain issue |
| 18:19.54 | ManxPower-work | Katty, do you have echo? |
| 18:20.02 | Katty | no |
| 18:20.15 | ManxPower-work | Katty, Didn't I tell you weeks ago to use the toneduration setting? |
| 18:20.17 | Katty | i have absolutely ZIP echo problems |
| 18:20.23 | Katty | mhmm you sure did |
| 18:20.41 | ManxPower-work | Are you using it? |
| 18:20.41 | Katty | too bad it didn't help (= |
| 18:20.45 | Katty | mhmm |
| 18:21.18 | babbio | [TK]D-Fender: i dont have a chan_dahdi.conf but only a chan_dahdi_template.conf |
| 18:21.25 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 18:21.28 | ManxPower-work | Katty, Sangoma's support is very good. |
| 18:21.34 | Katty | ManxPower-work: yep, i've already contacted them |
| 18:21.36 | babbio | how should i set up the chan_dahdi.conf? |
| 18:21.38 | [TK]D-Fender | babbio: There is your problem. You don't have the IMPORTANT one. |
| 18:21.42 | Katty | ManxPower-work: they're refusing logs and whatnot |
| 18:21.48 | Katty | s/refusing/reviewing/ |
| 18:21.55 | [TK]D-Fender | babbio: Go look at the sample config and realize you'll have to INCLUDE that other file |
| 18:22.09 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:22.25 | vader-- | bmoraca_work i haven't had a chance to even play with it |
| 18:22.30 | vader-- | im trying to figure out where to start with ti |
| 18:22.42 | vader-- | im not sure if i want to do a regular asterisk setup |
| 18:22.47 | vader-- | or freepbx |
| 18:22.51 | vader-- | or i was looking at sipx |
| 18:22.56 | *** join/#asterisk Netgeeks-laptop (~chris@88.sub-75-208-163.myvzw.com) |
| 18:23.08 | ManxPower-work | vader--, Will you want help from people on #asterisk? |
| 18:23.13 | bmoraca_work | ruchir: yes, you should be able to do that |
| 18:23.22 | ruchir | cool |
| 18:23.40 | ruchir | so just writing record blob data to file |
| 18:23.48 | vader-- | haha i guess your asking that becuase if i go with freepbx or sipx #asterisk people won't help :-) |
| 18:24.09 | babbio | [TK]D-Fender:could u suggest to me some docs on configuring and using chan_dahdi.conf? |
| 18:24.13 | ManxPower-work | Exactly. It's an important aspect of your decision |
| 18:24.25 | [TK]D-Fender | babbio: They're included in the tarball |
| 18:24.42 | bmoraca_work | i just got pwned by my idiot inventory clerk. ordered the wrong f-ing routers. |
| 18:24.48 | vader-- | well one of my main issues i have now with my current asterisk system is the fact that im the only one who can work on it |
| 18:24.53 | vader-- | becuase it's all conf files |
| 18:25.09 | vader-- | if it were web based i could give some of the simple tasks to my coworkers |
| 18:25.11 | [TK]D-Fender | vader--: I don't think my local Toyota dealership appreciated when I expected them to fix my Malibu.... |
| 18:25.11 | vader-- | to handle |
| 18:25.35 | vader-- | ya but freepbx is just a gui for asterisk |
| 18:25.40 | vader-- | asterisk is still running underneith |
| 18:25.44 | bmoraca_work | vader--: you could also give them some simple tasks to fix within the config files as well...they're not that difficult to understand |
| 18:25.44 | ManxPower-work | vader--, It's not THAT hard to write some GUI pages for specific tasks |
| 18:25.45 | [TK]D-Fender | vader--: Then install a GUI on it, or train other admins |
| 18:25.59 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:26.45 | moos3 | ManxPower-work, directory(contextname,options) but why doesn't ef work? |
| 18:28.33 | ManxPower-work | moos3, pastebin the output of "core show application directory" so I can look it up. |
| 18:28.43 | *** join/#asterisk Netgeeks-laptop (~chris@173.11.68.155) |
| 18:28.50 | vader-- | i wonder how compatible my dial plans will be from my current 1.2.x box if i goto 1.6.x |
| 18:29.06 | leifmadsen | vader--: depends how you wrote them I suppose :) |
| 18:29.09 | [TK]D-Fender | vader--: Go read the changelogs. |
| 18:29.20 | leifmadsen | vader--: if you're using pipes as a separator, that's one thing to fix |
| 18:29.53 | ManxPower-work | vader--, Good thing there are those UPGRADE*.txt files to tell you that information |
| 18:30.00 | Chainsaw | vader--: I did the 1.2 to 1.6 jump. Expect to spend about a week. Deploy a second box for testing. |
| 18:30.08 | moos3 | ManxPower-work, http://pastie.org/875856 |
| 18:30.21 | [TK]D-Fender | Chainsaw: wow... thats a rather vague comparison.. |
| 18:30.42 | [TK]D-Fender | Chainsaw: So far about the only thing we see in common... is the VERSION FAMILY |
| 18:30.59 | ManxPower-work | moos3, looks to me like it doesn't support searching BOTH first and last name. |
| 18:31.06 | Chainsaw | [TK]D-Fender: Yes, and since you are refusing to answer with details, my answer will probably be rated more helpful. |
| 18:31.10 | [TK]D-Fender | Chainsaw: Especially as 1.6 isn't even a specific branch |
| 18:31.13 | moos3 | yeah thats why i'm doing just first name |
| 18:31.28 | vader-- | Chainsaw i play on deploying this new setup in a virtual enviroment and i have this Adtran box i can use while my production system is running |
| 18:31.32 | vader-- | i can do testing at night |
| 18:31.39 | vader-- | switch the PRI and FXS ports over |
| 18:31.42 | Chainsaw | [TK]D-Fender: For most of the migration paths, the 1.6 branch is irrelevant. |
| 18:31.52 | moos3 | ManxPower-work, what i dont get it doesn't seem to accept the f |
| 18:32.00 | [TK]D-Fender | Chainsaw: .... 1.6 isn't a specific branch <- You seem to forget this. |
| 18:32.18 | *** join/#asterisk s519 (~steve@87-194-151-213.bethere.co.uk) |
| 18:32.20 | Chainsaw | [TK]D-Fender: No, you're just hung up on the wrong things here. But that's okay. |
| 18:32.33 | ManxPower-work | Chainsaw, no, you're just ignoring important things |
| 18:32.46 | vader-- | are there any good guides out there to follow for setting up asterisk? |
| 18:32.50 | VoIP-Penguin | I think he means the brach of 1.6 versions is not important here. |
| 18:32.53 | vader-- | it's been 4 years since i setup my box |
| 18:32.54 | vader-- | hehe |
| 18:33.02 | ManxPower-work | VoIP-Penguin, I know. And he's wrong. |
| 18:33.23 | ManxPower-work | Hey, I'm having problems with CONNECTEDLINE in my Asterisk 1.6.1.5 |
| 18:33.40 | ManxPower-work | Since CONNECTEDLINE was added to 1.6.2, and not any earlier versions..... |
| 18:33.58 | VoIP-Penguin | I can see how that is a problem. |
| 18:34.07 | Chainsaw | Which won't be in an existing 1.2 dial plan. But, nice try. |
| 18:34.09 | ManxPower-work | The version is IMPORTANT |
| 18:34.28 | ManxPower-work | Chainsaw, no, but it will exist on the Wiki, etc. |
| 18:34.34 | [TK]D-Fender | Chainsaw: 1.6.0 is a branch, 1.6.1 is a branch, 1.6.2 is a branch. 1.6 by itself isn't specific enough and there are feature differences between the actual branches. My comparison has a compatability bottom-line difference in includes an important number of extra changelogs he'd have to read through |
| 18:34.43 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:35.06 | Chainsaw | [TK]D-Fender: Yes, and one day you'll learn to express your desire for further information in a polite way. So many more people will like you. |
| 18:35.13 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:35.45 | vader-- | im surprised though i can't find anyone using SIPX, it looks like a really good system |
| 18:36.14 | [TK]D-Fender | vader--: Of course it does... you can't find any users to contradict you :) |
| 18:36.14 | ManxPower-work | Chainsaw, you are welcome to /ignore anyone you want. |
| 18:36.18 | Chainsaw | vader--: There are a few gotchas. The jumping behaviour that your dial plan may rely on is now optional. |
| 18:36.24 | [TK]D-Fender | \o/ |
| 18:36.58 | Chainsaw | vader--: (Where it jumps based on success or failure of a command, I found I had to emulate it with a GotoIf in some cases) |
| 18:37.00 | ManxPower-work | moos3, irectory(vm-context[,dial-context[,options]]) |
| 18:37.00 | [TK]D-Fender | waits to see how many people effectively read the contents of various UPGRADE.TXT's to him line by line... |
| 18:37.23 | ManxPower-work | make sure you have the context or an extra , |
| 18:37.37 | ManxPower-work | [TK]D-Fender, I re-read them about once a month. |
| 18:38.34 | moos3 | so this exten => s,1,Directory(sales,ef) should be exten => s,1,Directory(sales,,ef) |
| 18:38.35 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 18:39.08 | [TK]D-Fender | moos3: Count your parameters <- |
| 18:39.14 | leifmadsen | heh, parameter fail |
| 18:39.35 | [TK]D-Fender | VERY FINE MANUALS! |
| 18:39.41 | ManxPower-work | moos3, you see how important it is to look carefully |
| 18:39.45 | moos3 | yeah |
| 18:39.51 | *** join/#asterisk Researcher (~shani@unaffiliated/unafilliate) |
| 18:40.10 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 18:40.37 | vader-- | it's a shame they don't have a config checker that will check for things that are old and outdated |
| 18:42.45 | vader-- | hmm my current asterisk system is running on debian, most of my new linux boxes are centos 5 |
| 18:43.38 | ManxPower-work | vader--, Yes, that silly README with all the required versions in it is just SO 1995 |
| 18:43.52 | vader-- | :-) |
| 18:44.25 | ManxPower-work | vader--, the standard CentOS 5 RPMs will work just fine. |
| 18:44.55 | vader-- | Manx power do you have freepbx installed anywhere? |
| 18:45.17 | ManxPower-work | vader--, If I told you that I'd have to kill you. |
| 18:45.24 | vader-- | hehe |
| 18:45.50 | *** join/#asterisk soman (~somnath@e82-103-205-134.elisa-laajakaista.fi) |
| 18:47.12 | benngard | g729 is about 8 k or am i totally wrong? |
| 18:47.55 | ManxPower-work | benngard, Plus packet overhead |
| 18:48.03 | VoIP-Penguin | benngard: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 18:48.05 | benngard | ManxPower-work: yes ofc |
| 18:48.18 | *** join/#asterisk slinksh0t (~slinksh0t@66.90.110.157) |
| 18:48.41 | *** part/#asterisk babbio (~somebody@host-78-13-24-238.cust-adsl.tiscali.it) |
| 18:49.13 | benngard | have to read the license for g729, the shit is working but need it for 1 single day |
| 18:50.02 | patrb | Who was asking about vmail.cgi and asterisk real time voicemail? I think I found a work around |
| 18:50.29 | benngard | and i dont wanna run something thats need a fee, even if it is for a single day |
| 18:51.13 | VoIP-Penguin | If you intend to transcode, you'll need the codec. And in order to use the codec, you should pay the licensing fee. |
| 18:51.22 | huey23 | patrb: i was |
| 18:51.22 | benngard | guees so |
| 18:52.02 | patrb | huey23: Even though you're using realtime config, you can keep a voicemail.conf file with legitimate information for the vmail.cgi script to use |
| 18:52.17 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 18:52.22 | benngard | i guess i need i codec like that on monday, we gonna run some extensions over gprs |
| 18:52.29 | patrb | huey23: I made a script to dump the voicemailusers table into voicemail.conf...you could do that nightly |
| 18:52.30 | huey23 | i guess all i would need would be the user, context, and pass? |
| 18:52.36 | patrb | correct |
| 18:52.45 | VoIP-Penguin | benngard: You mean run some devices over gprs? |
| 18:52.46 | huey23 | do you mind if i use that script? |
| 18:52.59 | patrb | huey23: nah, let me put it on pastebin for you |
| 18:53.06 | benngard | VoIP-Penguin: yes, and some other stuff |
| 18:53.06 | huey23 | ty ty |
| 18:53.10 | VoIP-Penguin | benngard: 'cause you said extensions, but extensions don't care about the tech. |
| 18:53.54 | benngard | no but i dont have much bandwith over for voip, some damn sql app will take the most of it |
| 18:54.38 | benngard | normally i always run alaw, but i dont think i will have that "space" on my gprs line |
| 18:54.42 | VoIP-Penguin | SQL uses TCP, so configure QoS appropriately for use with VoIP. |
| 18:55.04 | VoIP-Penguin | alaw is big, so you'll want something else. |
| 18:55.15 | benngard | g729 ;) |
| 18:55.26 | VoIP-Penguin | or go low-quality gsm. |
| 18:55.30 | patrb | huey23: its ugly..make sure to change the database users and passwords to suit your needs: http://pastebin.com/fDbbz9EY |
| 18:56.20 | benngard | i pay for a license, if i pay some days late, they have to live with it. i will order tomorrow, and pay as fast as i can |
| 18:56.26 | patrb | huey23: if you can script at all..that should get you started |
| 18:57.10 | benngard | but u can actually hear the difference between, alaw and g729 |
| 18:57.11 | patrb | huey23: oh yeah...my for loop is setup to only count to 151 (the total users)...you'll need to change that to suit your needs |
| 18:57.23 | huey23 | patrb: i cannot...but i can read that and tweak it a little...i'll give it a shot |
| 18:57.41 | huey23 | i have a lot less users that that |
| 18:58.07 | patrb | huey23: you could easily write a function to check the number of users in the table |
| 18:58.17 | patrb | huey23: then pass that as a variable to the for loop |
| 18:58.20 | VoIP-Penguin | benngard: You can hear the difference, but it's not THAT much of a difference to be worried about it. Compare alaw to gsm... there's a LOT of difference on those. |
| 18:58.25 | huey23 | i do have a voicemail.conf already, can i just add the [general] searchcontexts=yes to to top of that one? |
| 18:58.50 | patrb | huey23: yes |
| 18:59.09 | patrb | huey23: err my searchcontexts=yes is under [general] |
| 18:59.45 | huey23 | under that just have the users? |
| 18:59.52 | patrb | huey23: you'll want to dump the users under [default] |
| 19:00.02 | patrb | huey23: assuming you're using the default context |
| 19:00.06 | patrb | huey23: in your real time config |
| 19:00.41 | benngard | VoIP-Penguin: did some test calls, g729 asterisk alaw, was about as a cell phone, will work |
| 19:00.51 | leifmadsen | who was looking for the ISDN info in a BYE? |
| 19:01.58 | leifmadsen | M15800 |
| 19:02.01 | MuffinMan | [confirmed] [Asterisk] Channels/chan_sip/NewFeature 0015800: [patch] Fetching SIP headers from BYE sent by callee reported by sergee https://issues.asterisk.org/view.php?id=15800 |
| 19:02.52 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 19:05.58 | *** join/#asterisk generalhan (~asd@about/windows/staff/generalhan) |
| 19:06.16 | vader-- | do you guys run asterisk on 32bit or 64bit os? |
| 19:06.22 | VoIP-Penguin | yes |
| 19:06.50 | vader-- | ummm |
| 19:06.54 | vader-- | it was a choice |
| 19:06.55 | vader-- | hehe |
| 19:07.03 | vader-- | i guess i should reword it |
| 19:07.13 | vader-- | do you guys recommend a 32-bit or 64-bit OS |
| 19:07.18 | VoIP-Penguin | either |
| 19:07.31 | benngard | VoIP-Penguin: did some more test switching from alaw to g729 and back, dialing to pstn and listen, the difference is pretty big |
| 19:13.02 | Naikrovek | vader--: there are some modules that don't run on 64-bit machines i think |
| 19:13.15 | Naikrovek | as i recall digium has some things that don't run on 64-bit |
| 19:13.18 | Naikrovek | well |
| 19:13.19 | Naikrovek | they do |
| 19:13.22 | *** join/#asterisk kbr (~kbr@ASte-Genev-Bois-152-1-19-61.w83-114.abo.wanadoo.fr) |
| 19:13.25 | Naikrovek | but not with 64-bit asterisk |
| 19:13.42 | VoIP-Penguin | hmm |
| 19:13.54 | Naikrovek | wait for confirmation on that before you believe me |
| 19:14.52 | VoIP-Penguin | Any idea which modules aren't available (or don't work) with 64-bit asterisk? |
| 19:15.03 | *** join/#asterisk s14ck (~s14ck@190.72.134.63) |
| 19:15.05 | Qwell | Naikrovek: It was only Fax For Asterisk, and that has been fixed. |
| 19:15.12 | Naikrovek | ah there you go |
| 19:15.15 | Naikrovek | thank you qwell |
| 19:15.25 | Naikrovek | 64-bit ahoy! |
| 19:15.29 | VoIP-Penguin | I was just looking over my stuff and didn't see anything not working. |
| 19:17.10 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:17.28 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:20.43 | *** join/#asterisk brunner (~chris@99-1-221-215.lightspeed.tukrga.sbcglobal.net) |
| 19:20.46 | brunner | is it possible to get Caller ID w/ name on a PRI? |
| 19:23.43 | carrar | heck yes |
| 19:23.45 | carrar | if you ask for it and put a short delay before answering |
| 19:23.45 | carrar | brb |
| 19:23.46 | beek | brunner: wait(1) |
| 19:23.46 | brunner | got it, so one second should cover it? |