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| 00:38.39 | Knew | What do you guys think about launching a competitor to this? http://business.fonolo.com/ |
| 00:40.15 | [TK]D-Fender | Knew: Pretty easy |
| 00:42.36 | carrar | Knew, is fonolo using Asterisk? Do you need help? |
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| 00:55.13 | EndEng | does asterisk need mysql? |
| 00:56.02 | [TK]D-Fender | EndEng: No. |
| 00:56.15 | EndEng | thx |
| 00:56.29 | drmessano | Nor do I.. screw MySQL |
| 00:57.09 | [TK]D-Fender | drmessano: I tried, but the CD broke and I got all sorts of lacerations... |
| 00:57.16 | drmessano | MySQL is like Access 97 w/replication |
| 00:58.55 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
| 01:02.03 | [TK]D-Fender | off to make some noise... later... |
| 01:02.11 | drmessano | later |
| 01:05.58 | Knew | carrar, I don't know if they are. But if I launch a competitor, I would use Asterisk. |
| 01:07.53 | *** join/#asterisk korcan (~johnynum5@adsl-76-234-140-170.dsl.sfldmi.sbcglobal.net) |
| 01:10.06 | Knew | I would be providing a hosted solution that allows users to go through phone numbers and enter account info securely with a web interface. Then when they're done, it would virtually queue them basically. The web interface would let them know their call position, etc. At the end of the call (when the user hangs up), the web interface would be a |
| 01:10.33 | Knew | great opportunity to to prompt them with a satisfaction survey on behalf of the company. |
| 01:11.04 | Knew | It would be sold to companies, which means I am competing with the business Fonolo solution: http://business.fonolo.com/ |
| 01:11.34 | Knew | Does this idea make sense? |
| 01:12.07 | Knew | I'd have to hire some Asterisk admins of course |
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| 01:28.06 | mmlj4 | hire me first? |
| 01:29.13 | drmessano | Show us some VC and we're yours |
| 01:30.39 | Knew | drmessano, that won't happen unless I pitch the idea. |
| 01:30.57 | Knew | At least I found out Fonolo exists. I would look silly saying there's no competitors. |
| 01:31.04 | mmlj4 | what is it? call a computer and have it wait on hold for you? |
| 01:31.09 | Knew | I am pitching to a VC btw. |
| 01:31.23 | mmlj4 | or...? |
| 01:31.27 | Knew | mmlj4, well |
| 01:31.54 | drmessano | I can't speak for everyone else, but I don't work for free.. So VC or "no me" |
| 01:31.56 | Knew | mmlj4, actually it just replaces the phone menus/dial by directory prompts and account info prompts. |
| 01:32.17 | Knew | drmessano, that's perfectly fine. |
| 01:32.44 | mmlj4 | I wonder |
| 01:32.50 | drmessano | You do? |
| 01:33.05 | mmlj4 | I wonder how they pull it off, I mean |
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| 01:33.30 | Knew | mmlj4: The caller goes through the phone menus to categorize their call from the company web site, enters account info securely (if needed), then the web app counts down the minutes until an agent can speak with them. |
| 01:33.45 | Knew | Once an agent is available, it patches them through. |
| 01:33.48 | drmessano | It's call return with a web front end.. 2 hours to code it, max? |
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| 01:34.25 | Letoric|home | Evenin folks. Anybody know of a way to set up some kind of indicator on a soundpointIP 670 if the asterisk database contains x value? |
| 01:34.27 | Knew | drmessano, yeah I'm not worried about implementation. I'm worried about competitors already doing this and selling the idea to businesses. |
| 01:35.04 | drmessano | Knew: Who cares if someone else is doing it.. do it BETTER.. Set yourself apart |
| 01:35.12 | Letoric|home | The goal is to be able to have a visible indicator that the autoattendant is on |
| 01:35.15 | Knew | drmessano, right. |
| 01:35.36 | mmlj4 | Letoric|home: set up and extension and call it if X exists... then set up a busy lamp for that extension |
| 01:36.04 | Letoric|home | ok, but what triggers asterisk to check if it exists? |
| 01:36.10 | Letoric|home | doesn't something have to 'start' the process? |
| 01:36.48 | Letoric|home | I know how to tell asterisk to check, but not how to tell it to check periodically |
| 01:36.58 | mmlj4 | or if you have an intelligent phone, send it SIP messages |
| 01:37.24 | Knew | My value proposition for companies is 1) save money on long distance minutes 2) improve customer experience 3) actually have a way to prompt the caller to do a satisfaction survey after they've hung up. |
| 01:38.09 | mmlj4 | so what is X exactly? or basically? |
| 01:38.23 | Letoric|home | it's 0 or 1, dependant on if the auto attendant is active or not |
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| 01:42.58 | mmlj4 | Knew: so it's doable... now how do you market that? |
| 01:43.31 | Knew | mmlj4, well I'd sell to businesses by going to tradeshows. |
| 01:43.43 | mmlj4 | who would bother using such a service? and how do you charge them? |
| 01:44.21 | Knew | mmlj4, I would charge the businesses that have incoming call volume to use it. |
| 01:44.37 | mmlj4 | oh, you'd charge the company to give their customers a web interface to wait on hold? |
| 01:44.46 | mmlj4 | that makes actual sense |
| 01:44.53 | Knew | mmlj4, yeah |
| 01:45.10 | Knew | mmlj4, only problem is I've got to be better than my competitor: Fonolo |
| 01:45.31 | Knew | Fortunately, they seem a bit confused about who their customer is, http://fonolo.com/ |
| 01:46.04 | Knew | Also, when the call is complete and they hang up, we have an actual way to ask them to fill out a survey. |
| 01:46.39 | mmlj4 | yeah, customers just love to be annoyed |
| 01:47.20 | drmessano | Slow down.. Add the survey later.. You don't want your startup to fail just yet |
| 01:47.21 | Knew | Well, if they like the call, they'll just close the window after they hang up. But an upset customer will fill it out. |
| 01:47.25 | drmessano | These things take months |
| 01:47.51 | drmessano | At least give yourself enough time to get some expensive lunches and bouncy balls |
| 01:48.12 | drmessano | THEN start adding shit that alienates the customers and brings the business to a dead stop |
| 01:48.15 | Knew | And some dot-com-era chairs :p |
| 01:48.20 | carrar | amd hermin miller chairs |
| 01:48.26 | mmlj4 | I had an aeron when I worked at Shell |
| 01:48.27 | carrar | heh |
| 01:48.39 | Knew | drmessano, well the customer isn't the caller, it's the business. |
| 01:48.51 | drmessano | YOUR customer is the business |
| 01:48.54 | Knew | mmlj4: Yeah, I have an Aeron at work |
| 01:49.02 | carrar | I'm sitting in my dotcom hermin miller chair right now |
| 01:49.07 | Knew | drmessano, right. |
| 01:49.48 | Knew | So I should just tell the VC "Fonolo.com is our competitor, but we're going to be way better"? |
| 01:49.53 | mmlj4 | the trick would be to get businesses to trust you with access to their customer DB |
| 01:50.14 | drmessano | Knew.. do you even know what you're doing here? |
| 01:50.32 | drmessano | You're asking questions like you're opening your first lemonade stand |
| 01:50.39 | Knew | drmessano, I'm young |
| 01:50.46 | drmessano | How young? |
| 01:50.48 | Knew | 21 |
| 01:50.55 | carrar | http://www.hermanmiller.com/MarketFacingTech/hmc/products/Aeron_Chairs/hero_aeron_work_1.jpg |
| 01:51.27 | Knew | And this is my first lemonade stand I guess, but that's okay because I know people smarter than me. |
| 01:51.43 | mmlj4 | smart enough, anyhow |
| 01:51.45 | Knew | carrar, yeah the office in Manhattan has that. I love those. |
| 01:51.46 | drmessano | 21? Go get me a coffee and a dirty magazine, and you can keep the change |
| 01:51.58 | carrar | yeah I love mine |
| 01:52.03 | tzanger | drmessano: haha |
| 01:52.10 | carrar | Free is the best price |
| 01:52.35 | Knew | drmessano, I'm the entitled generation. Give me $20 and you might get the magazine next week. ;) |
| 01:52.36 | mmlj4 | bah, I hate organized sports |
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| 01:53.44 | Knew | I was just trying to get a point of view on the idea from people that actually have seen the inside of call centers. |
| 01:54.03 | mmlj4 | seen and ran screaming to boot |
| 01:54.17 | genxweb | does any one have a link on how to use the new asterisk gui to configure a sip trunk. I am use to using freepbx but the box I am using this time can not have freepbx installed. |
| 01:54.26 | genxweb | the manual on the site is not found |
| 01:54.35 | drmessano | 404? |
| 01:54.50 | mmlj4 | go get vmware and put up a test box |
| 01:54.57 | Knew | So have you all heard of Fonolo and Lucyphone? |
| 01:55.12 | mmlj4 | ^C |
| 01:55.40 | drmessano | I called that chick, twas her number she swore, but her number was disconnected and her Facebook, 404 |
| 01:56.29 | mmlj4 | heh |
| 01:58.13 | mmlj4 | wow, that's an idea... route robocallers to a "we're sorry, but that number is disconnected or is no longer in service" context |
| 01:59.47 | genxweb | I have all numbers to my comcast line that is fowarded to my asterisk box get played a message telling the spammer off since I dont use that line and only have it for the discount comcast provides |
| 01:59.54 | carrar | genxweb, dump freebpx |
| 02:00.33 | genxweb | iam trying |
| 02:00.38 | genxweb | iam trying the asterisk gui |
| 02:00.45 | genxweb | but it is so confusing |
| 02:00.51 | carrar | don't use a gui |
| 02:01.01 | carrar | it's limiting |
| 02:01.26 | drmessano | Met this girl at a bar, thought I would score, looked for her Twitter, 404 |
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| 02:01.52 | carrar | People still use twitter? |
| 02:02.04 | genxweb | I would agree but I have 24 hours to finish this setup and learning straight asterisk to the extent i need is not gonna happen tonight. HAHA |
| 02:02.17 | carrar | better hire someone |
| 02:02.29 | drmessano | We dated for months, had lots of fun, asked her to marry me, 401 |
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| 02:46.06 | Micc | I've set jbenable=yes and jbforce=yes in general, but setting it to no on certain peers does not seem to turn it off. Is this the correct behavior? |
| 02:46.17 | Micc | Is jitterbuffer system wide no matter what? |
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| 02:51.10 | Knew | drmessano, that's because she knows you need to update your client to HTTP 1.1 |
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| 02:52.02 | drmessano | Knew: Sometimes you take what you can wget.. Know what I mean? |
| 02:53.43 | Knew | Women are deprecating marriage now because it is a silly, outdated institution. |
| 02:54.04 | Knew | Sorry, getting offtopic. :) |
| 02:56.09 | Micc | since I can't turn off the jitterbuffer per peer, how bad would it be to run a fixed 150ms jb on a system that handles 20-30 calls at a time? |
| 02:56.24 | Micc | with 265 sip peers. |
| 02:56.42 | Micc | I'm a little worried it may add too much cpu usage. |
| 03:00.21 | pabelanger | Micc: patches welcome :) |
| 03:00.52 | Micc | pabelanger, do you use the jitter buffer for anything? |
| 03:01.33 | pabelanger | Usually not, most of my installs have had large pipes of bandwidth |
| 03:02.39 | Micc | any experience using it with faxing or credit card machines or postage meeters? |
| 03:03.11 | pabelanger | Nope, but I don't imagine it will help much honestly |
| 03:05.56 | Micc | It seems to help some of my customers that could never get a credit card transaction through. |
| 03:06.35 | Micc | but I haven't done much testing with it yet. But I may have to code the turning off of jitter buffer per peer myself so I can continue my testing. |
| 03:06.49 | pabelanger | Possible, like I said, never tried. |
| 03:07.23 | pabelanger | Micc: Ya, don't see a reason why that should not work. Be sure to upload the patch to the tracker |
| 03:08.36 | Micc | sure, I'll share if I actually get around to doing the work. |
| 03:08.53 | Micc | I haven't done any asterisk dev work for years. |
| 03:09.00 | Micc | I could be a bit rusty. |
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| 03:47.51 | NotMikeReally | hi |
| 03:51.40 | NotMikeReally | i've got asterisk 1.6.2.11 and speex 1.1.12, as far as i can see this should support wideband speex 16000 right? 8000 works but not 16000 |
| 03:52.07 | NotMikeReally | if so, anyone have any suggestions for stuff to check? |
| 04:29.53 | p3nguin | pabelanger: Do you happen to have a good tutorial on when jitterbuffer should be used and exactly how to configure it based on the circumstances? |
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| 04:40.26 | NotMikeReally | are you just wondering what a jitter buffer is and what it's for? or something more specific? |
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| 07:40.16 | schmidts | good morning |
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| 08:10.09 | BeeBuu | i got "No application 'agentlogin' for extension",what's wrong? |
| 08:13.15 | Tim_Toady | missing module maybe |
| 08:13.23 | schmidts | what says core show application agentlogin ? |
| 08:14.32 | BeeBuu | your application is not registered. |
| 08:14.39 | Tim_Toady | check that chan_agent is loaded |
| 08:14.45 | BeeBuu | O... |
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| 08:15.48 | BeeBuu | i "module show like agent" can see the chan_agent.so |
| 08:16.29 | BeeBuu | chan_agent.so Agent Proxy Channel 0 |
| 08:16.35 | BeeBuu | 1 modules loaded |
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| 08:41.29 | akoma1s | Does Asterisk do any codec translation (to and back from slin) when bridging a DAHDI BRI channel (using alaw codec) with a SIP channel (again using alaw) ? |
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| 08:44.06 | akoma1s | anyone? :( |
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| 08:52.15 | TobSnyder | what VoIP Softphones do you use, which clients are recommendable? |
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| 08:56.14 | akoma1s | If we're talking about Linux, I'd suggest Twinkle. |
| 08:56.47 | akoma1s | It's old, it's ugly, it has few codecs, but it's dependable and has many PBX-suited features |
| 08:56.57 | TobSnyder | ok |
| 08:57.05 | TobSnyder | and for windows? |
| 08:57.49 | TobSnyder | I have tested twinkle and ekiga on my linux machine and I have also the feeling that twinkle is more stable |
| 08:57.51 | akoma1s | Not much experience there, Xlite did the job last time I needed Windows though |
| 08:57.54 | UQlev | TobSnyder: depends on protocol |
| 08:57.57 | TobSnyder | but on windows there are a lot of clients |
| 08:58.01 | TobSnyder | SIP |
| 08:58.13 | UQlev | TobSnyder: Phonerlite, zoiper |
| 09:00.31 | UQlev | TobSnyder: they work fine for me. xlite often hangs |
| 09:00.31 | TobSnyder | okay good to know |
| 09:00.31 | akoma1s | does it support call transfer? |
| 09:00.31 | UQlev | zoiper works perfect for linux and windows, sip and iax |
| 09:00.31 | UQlev | yes, both |
| 09:00.32 | akoma1s | ah sorry, just checked the feature list |
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| 09:00.32 | akoma1s | thanks for the recommendation, it sure seems impressive :) |
| 09:00.36 | UQlev | the only problem with zoiper I found incorrect work with DTMF if you need dial extension |
| 09:19.14 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 09:24.34 | *** join/#asterisk jekader (~jekader@212.56.212.178) |
| 09:24.50 | jekader | hi everyone! |
| 09:26.04 | jekader | so, I want to make an extension in asterisk, say 321. When one calls it, phones 201 ... 221 ring. When they answer, they're transfered to a conference. Where can I read some literature on this? |
| 09:28.50 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 09:30.01 | akoma1s | jekader: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 09:31.14 | akoma1s | another way is Dial(SIP/201&SIP/202&...), however that gets ugly quickly |
| 09:32.11 | *** join/#asterisk rossand (~aross@95.122.65.73) |
| 09:33.12 | akoma1s | here's an example queue member line that I'm using: |
| 09:33.14 | akoma1s | member => Local/spyros-home@queue-handler,2,Spyros - Home,SIP/spyros-home |
| 09:38.40 | *** join/#asterisk Black187 (~Black187@193.77.24.83) |
| 09:38.46 | Black187 | Hello |
| 09:39.06 | kaldemar | jekader: you need to make an extension that originates a call between the phones and a conference. originate is the key word. |
| 09:39.27 | Black187 | I have a question regarding dialplan - is it possible to make special dialplan if user makes a call transfer? |
| 09:39.55 | Black187 | So I would pass to the user some info regarding call transfer... |
| 09:42.13 | GhOnDiE | does anybody know any good uk suppliers for cheap fxo cards? |
| 09:42.54 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 09:47.11 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 09:49.11 | defswork | GhOnDiE, define cheap |
| 09:49.21 | defswork | I get all my stuff from voipon.co.uk |
| 09:50.37 | defswork | but I have a reseller account and discount |
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| 09:53.06 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 09:53.23 | GhOnDiE | lol well i been looking at voipon and they seem to have the best deal so far |
| 09:53.33 | GhOnDiE | wanted to upgrade my x100p card |
| 09:53.43 | GhOnDiE | so that i can at least do callerid |
| 09:58.36 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
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| 10:10.48 | *** join/#asterisk ptrahn4713 (~harald@ip-95-222-189-235.unitymediagroup.de) |
| 10:11.45 | ptrahn4713 | hi folks |
| 10:12.42 | ptrahn4713 | anybody here? |
| 10:12.54 | ptrahn4713 | not idleing? |
| 10:16.12 | schmidts | whats wrong with idleing? :D |
| 10:16.27 | schmidts | hello btw ;) |
| 10:16.32 | coppice | GhOnDiE: despite a refusal to integrate them into DAHDI, there are perfectly good UK caller ID patches around for the x100p |
| 10:20.20 | GhOnDiE | im using piaf at the moment and i cannot find the patch anywehere for zaptel source |
| 10:20.34 | GhOnDiE | all the sites i come across are no longer about |
| 10:23.51 | ptrahn4713 | hi |
| 10:24.25 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 10:24.45 | ptrahn4713 | missing the g.722 driver in my current asterisk distribution i'd like to ask where to get it from. |
| 10:25.08 | ptrahn4713 | i havn't fount it yet on asterisk server and hover driver seems to be gone |
| 10:25.25 | ptrahn4713 | any suggestions? |
| 10:26.49 | *** part/#asterisk rossand (~aross@95.122.65.73) |
| 10:28.58 | *** part/#asterisk ptrahn4713 (~harald@ip-95-222-189-235.unitymediagroup.de) |
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| 10:31.30 | coppice | G.722 is built into asterisk |
| 10:32.31 | jekader | kaldemar - thanks, will look into that! |
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| 11:00.32 | *** part/#asterisk AndyRomano (~Adium@mail-gw.helwacht.at) |
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| 11:57.46 | jekader | strange, can't make it work |
| 11:59.42 | jekader | gives out an error: pbx_extension_helper: No application 'Originate' for extension |
| 12:00.06 | jekader | maybe asterisk 1.4 doesn't have this application/ |
| 12:00.48 | jekader | yep :( |
| 12:00.49 | jekader | Originate a call. New in Asterisk 1.6.2. |
| 12:01.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 12:18.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:18.34 | leifmadsen | Corydon76-dig: so I have a suggestion for cdr_adaptive_odbc |
| 14:16.55 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:16.55 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta4 (2010/08/24), 1.6.2.11 (2010/08/10), 1.4.35 (2010/08/10), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
| 14:23.39 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:23.52 | *** join/#asterisk msetim (~setim@187.112.62.42) |
| 14:24.01 | msetim | hello guys |
| 14:24.37 | msetim | why when I got a 404 not found I received an AST_CONTROL_CONGESTION in the OriginateResponseReason? |
| 14:25.19 | Shane-S | anyone have phone recommendations aside from Polycomm, and/or a website that has good prices. I just need to get some reliable phone prices. |
| 14:25.33 | drmessano | What's wrong with Polycom? |
| 14:25.43 | Nugget | Whatever you do, don't buy Cisco phones. |
| 14:26.46 | Shane-S | drmessano: Nothing, just curious if there is another brand to look at :P |
| 14:26.51 | Katty | hi Nugget |
| 14:26.56 | *** join/#asterisk viperdudeuk (~jon@195.74.96.117) |
| 14:27.13 | Shane-S | anyone have a good site for pricing on them, google returns many sites. |
| 14:27.29 | Katty | telephonydepot.com |
| 14:27.48 | bougyman | Shane-S: we really like snom here. |
| 14:27.54 | bougyman | anything except the 8xx series, that is. |
| 14:28.15 | Shane-S | Katty: ty someone gave me that before...maybe I will be smart and bookmark that! |
| 14:28.19 | WIMPy | also likes Snom 320/360/370 |
| 14:28.31 | bougyman | i'm sad they dropped the 360 |
| 14:28.41 | Shane-S | bougyman WIMPy ty |
| 14:28.42 | bougyman | but they reduced the 370 price when they did, so it's not so bad. |
| 14:30.04 | Shane-S | bougyman: the 8xx looks nice, but they suck? |
| 14:30.22 | bougyman | Shane-S: unfortunately the software isn't quite right on the 8xx yet. |
| 14:30.34 | bougyman | for that matter don't use 8.x firmware on the 3xx series. |
| 14:30.36 | Shane-S | Darn...pretty colors impress me :P |
| 14:30.47 | bougyman | it's the 8.x firmware that's broken, and the 8xx series only runs on 8.x firmware. |
| 14:30.57 | bougyman | the 3xx series can still use 7.x firmware, which has been very solid. |
| 14:31.07 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-177-002.pools.arcor-ip.net) |
| 14:31.14 | Shane-S | ahh okay, well at least I can compare prices this way, thanks a bunch all |
| 14:32.12 | WIMPy | Yes. I sometimes have troube accepting calls with v8 :-( But the minibrowser has become sso much faster that it's actually usefull in v8. |
| 14:33.29 | WIMPy | You used to drink a cup of tea between each key press before which made it less usefull. |
| 14:34.16 | *** part/#asterisk minaguib (~mina@modemcable109.56-20-96.mc.videotron.ca) |
| 14:34.53 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:35.10 | WIMPy | But otherwise tehy behave the way you would exprect a phone to behave. Which unfortunaletly doesn't apply to al IP phones. |
| 14:35.28 | *** join/#asterisk neurosys (~neurosys@173.200.196.33) |
| 14:36.30 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
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| 14:38.41 | WIMPy | BTW: Is there any reason dahdi ohny supports multi port HFC cards and not the single port ones? |
| 14:38.53 | *** join/#asterisk ctaloi (~ctaloi@nat-66-218-1-160.usadatanet.com) |
| 14:39.47 | veryhappy | hello, im a beginner to asterisk, im just reading a good tutorial to asterisk, i have just a question where i couldnt still find the answer, what is the difference or the meaning of underscore POINT ( _. ) for what is the point? thank you |
| 14:42.47 | Nugget | sits down next to katty |
| 14:43.37 | *** join/#asterisk ManxPower (~manxpower@24.236.78.136) |
| 14:43.51 | ManxPower | Can anyone think of any way the last digit of this could be changed? |
| 14:43.54 | ManxPower | [Sep 9 12:23:20] VERBOSE[32181] logger.c: -- Called G0/011441922638094 |
| 14:43.54 | ManxPower | [Sep 9 12:23:21] DEBUG[32181] chan_dahdi.c: Sent deferred digit string: T01144192263809 |
| 14:44.08 | ManxPower | notice the last digit of the dialed number |
| 14:44.45 | Katty | hugs on Nugget |
| 14:44.47 | Katty | hi manx. |
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| 14:50.27 | veryhappy | hello |
| 14:50.39 | [TK]D-Fender | veryhappy: ... |
| 14:50.42 | [TK]D-Fender | veryhappy: [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:50.48 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:50.56 | [TK]D-Fender | veryhappy: Chapter 5 is your friend. |
| 14:55.25 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:55.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta4 (2010/08/24), 1.6.2.11 (2010/08/10), 1.4.35 (2010/08/10), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
| 14:57.41 | *** join/#asterisk alex_voip (~alex@201.161.45.81) |
| 14:57.56 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:58.17 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 14:58.32 | jeffik | ariel_: hi |
| 14:58.50 | ariel_ | hello jeffik |
| 14:59.14 | jeffik | areil_: still in FLA and working on boats? |
| 14:59.18 | *** join/#asterisk neurosys (~neurosys@173.200.196.33) |
| 14:59.33 | ariel_ | yes still in south florida and working on ships |
| 14:59.50 | jeffik | ships, sorry |
| 15:00.18 | alex_voip | hello just switched from 1.4 to 1.6.2.11 and now recordings generated with mixmonitor are way out of sync. Sip trunks and sip endpoints. Any ideas? |
| 15:00.36 | [TK]D-Fender | BRB |
| 15:01.23 | Katty | what should i make for lunch |
| 15:01.37 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-177-002.pools.arcor-ip.net) |
| 15:02.23 | *** join/#asterisk CunningPike (~CunningPi@204.239.12.183) |
| 15:09.16 | ManxPower | alex_voip, did you read all the UPGRADE* files included in the Asterisk source? |
| 15:13.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 15:13.57 | alex_voip | ManxPower, i did, but by your question i guess i missed something i'll take another look |
| 15:14.54 | Kobaz | eats Katty for lunch |
| 15:17.33 | Shane-S | here is a question, what is the most basic PoE Switch you use that can handle multiple phone sets. I am looking at a Cisco (really a Linksys), but I can't find the Polycom 321's draw or the total output on some switches. Do I go by their total power supply consumption? |
| 15:17.52 | Shane-S | I am looking at 24-48 port switches |
| 15:18.25 | *** join/#asterisk gr0mit (~tim@212.56.94.150) |
| 15:19.17 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 15:19.30 | [TK]D-Fender | Shane-S: You said 2-3 phones... how many aer you REALLY looking at? |
| 15:20.55 | Katty | Kobaz: that's highly inappropriate. |
| 15:21.14 | Shane-S | I am just getting a pricing idea for here IF they choose to let me look at a replacement system, would be about 55 phones, 5 barely used. |
| 15:21.14 | Qwell | Katty: O.o |
| 15:21.25 | Shane-S | I was looking at http://www.cdwg.com/shop/products/default.aspx?EDC=1487040 |
| 15:21.34 | Katty | Qwell: what should i make for lunch |
| 15:21.45 | Shane-S | I am just giving ball park numbers to them for now |
| 15:22.31 | Qwell | Katty: can't go wrong with a nice sandwich |
| 15:22.42 | Katty | hmm. sammiches are good. |
| 15:22.44 | Kobaz | Katty: hehe |
| 15:22.47 | Kobaz | Katty: aww |
| 15:22.47 | Katty | Qwell: what kind of sammich |
| 15:22.49 | Qwell | something with, say...avacado. |
| 15:22.53 | Katty | :>>> |
| 15:23.00 | Katty | omnomnom |
| 15:23.01 | Qwell | (I dislike avacado..) |
| 15:23.03 | Kobaz | Katty: there's nothing wrong with some good clean canabalism :( |
| 15:23.04 | Qwell | but hey! |
| 15:23.13 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:24.00 | Katty | Kobaz: i do believe i'd like to keep all my limbs for the time being. |
| 15:24.14 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:24.17 | Qwell | Katty: just a finger, or a toe or two. not too bad |
| 15:24.20 | Kobaz | yeah |
| 15:24.30 | Katty | :< |
| 15:25.44 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:26.00 | [TK]D-Fender | Shane-S: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=2787280&CatId=2800 |
| 15:26.27 | Kobaz | [TK]D-Fender: those aren't that great |
| 15:26.50 | Kobaz | [TK]D-Fender: they work... but the management part of it is kinda limited |
| 15:26.55 | [TK]D-Fender | Kobaz: Depends on your needs for "greatness". I run 2 of their predecessors and none of my clients have even had any issue with them |
| 15:27.11 | [TK]D-Fender | Kobaz: Never needed "management" personally |
| 15:27.15 | Kobaz | ah |
| 15:27.30 | Kobaz | yeah we use dell managed switches and adtran managed switches with the cisco style interface |
| 15:27.42 | [TK]D-Fender | Kobaz: If you're VLAN-ing QoS-ing and worse all at once maybe... |
| 15:27.50 | *** join/#asterisk gr0mit (~tim@eastrop.plus.com) |
| 15:27.56 | Kobaz | yeah vlans, lldp, and qos |
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| 15:33.50 | Katty | Qwell: chicken salad croissant |
| 15:33.57 | Katty | Qwell: sounds /really/ good |
| 15:34.21 | Qwell | so it does |
| 15:34.28 | Qwell | make me one too :D |
| 15:35.59 | Katty | kay |
| 15:41.06 | raden | Katty, !!!!!!!! :) |
| 15:41.31 | Qwell | nobody greets me with !s anymore. :( |
| 15:42.29 | Katty | raden: :>>>>>>>>>>>>>>>>>>> |
| 15:42.31 | Katty | hugs on raden |
| 15:42.44 | Katty | Qwell: do you feel neglected? |
| 15:42.49 | Qwell | nope |
| 15:42.54 | Katty | Qwell: sad? |
| 15:42.57 | Qwell | nope |
| 15:43.00 | Katty | Qwell: hungry? |
| 15:43.03 | Qwell | a little |
| 15:43.46 | raden | Hows Katty doing ? |
| 15:44.03 | Katty | tired, freezing cold, starving |
| 15:44.38 | Katty | how're you dear |
| 15:44.58 | Kobaz | you're always cold |
| 15:45.02 | Kobaz | you should bump the heat up |
| 15:45.13 | Katty | Kobaz: i have my space heater going. |
| 15:45.27 | Katty | Kobaz: and as soon as it hits 80ish in my office, i will no longer be cold. |
| 15:45.31 | Kobaz | heh |
| 15:45.34 | Kobaz | aww... 80 |
| 15:45.40 | Kobaz | it's a comfy 65ish in here |
| 15:45.44 | Katty | >.< |
| 15:46.12 | Katty | Kobaz: you have insaned. |
| 15:50.06 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:51.47 | *** part/#asterisk sunfone (~jeff@c-71-239-123-82.hsd1.il.comcast.net) |
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| 16:08.04 | raden | Katty, I doing better now was in hospital last 2 days |
| 16:09.08 | raden | everything I Have been through in life and a mesquitto bite almost gets me LOL |
| 16:09.42 | DogBoy | what happened with the mosquito bite |
| 16:10.03 | raden | Cellulitis |
| 16:10.21 | raden | very rapid |
| 16:10.58 | raden | 104 temp BP droped to 72 / 40 ( im normally 135 / 85 ) pulse was 140's -150's for 6 hours straight |
| 16:11.06 | raden | White blood count was over 30,000 |
| 16:11.12 | raden | all within a few hours |
| 16:12.19 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 16:12.20 | DogBoy | I'm constantly getting bitten by mosquitoes |
| 16:13.09 | raden | me 2 |
| 16:13.16 | raden | there all over the place |
| 16:13.20 | raden | just takes the right one |
| 16:13.34 | raden | when i was in the army I was always getting bit by weird crap never had issues |
| 16:13.48 | raden | been in places with all sortsa nasty diseases never got anything |
| 16:13.57 | raden | but yup a mosquito did it |
| 16:14.13 | *** join/#asterisk Tim_Toady (~moi@77.49.254.107.dsl.dyn.forthnet.gr) |
| 16:16.58 | Letoric | is there an easy way to see who has an active line open in asterisk? |
| 16:17.17 | Letoric | I can see that lines are open, but not sure without turning on debug, how to see who it is |
| 16:17.22 | [TK]D-Fender | Letoric: What is an "active line"? |
| 16:18.04 | Letoric | well, if I do core show calls it shows 1 active call |
| 16:18.23 | alex_voip | ok i've read UPGRADE* and, with my reading skills, i was unable to find any reason that going from 1.4 to 1.6.2 would cause MixMonitor to generate out of sync recordings |
| 16:18.25 | Letoric | I would like to be able to see WHO the active call is |
| 16:18.54 | Letoric | and also the device it terminates to, assuming that it isn't just a call in the IVR |
| 16:19.13 | [TK]D-Fender | Letoric: "core show channels concise |
| 16:19.48 | Letoric | sweet, thank you |
| 16:20.41 | *** join/#asterisk a_m_y (~IceChat7@116.50.246.68) |
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| 16:39.53 | Katty | raden: wow. |
| 16:40.12 | Katty | raden: you poor thing :< |
| 16:40.16 | *** join/#asterisk gr0mit (~tim@eastrop.plus.com) |
| 16:40.54 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:41.32 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
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| 16:56.16 | pabelanger | infobot: beer me |
| 16:56.16 | infobot | ACTION pours a pint of La Maudite for pabelanger |
| 16:57.38 | *** part/#asterisk Gohltron (~rooms@kobz-4d0012be.pool.mediaWays.net) |
| 16:58.05 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:08.20 | pabelanger | heh, seems if I manually load chan_iax2.so from the *CLI>, asterisk will use 99% CPU. (1.6.2) |
| 17:08.35 | bougyman | nice feature |
| 17:08.59 | anonymouz666 | what about unload ? |
| 17:09.41 | anonymouz666 | I am very afraid to move from 1.4 to 1.6.2.X because of things like that |
| 17:09.58 | drmessano | 1.6.2.x is 10x better than 1.4 |
| 17:10.14 | anonymouz666 | define "better" |
| 17:10.19 | drmessano | Features, stability, speed |
| 17:10.23 | anonymouz666 | stability? |
| 17:10.26 | anonymouz666 | are you sure? |
| 17:10.38 | drmessano | Why because it has A BUG? |
| 17:10.46 | drmessano | I bet it has others too |
| 17:10.48 | anonymouz666 | if I manually load chan_iax2.so from the console asterisk won't use 99% CPU |
| 17:10.49 | bougyman | 10 * 0 == 0 |
| 17:10.49 | drmessano | As does 1.4 |
| 17:10.57 | drmessano | ^^^ |
| 17:12.04 | anonymouz666 | I will use 1.4 until the bugs got fixed |
| 17:12.13 | drmessano | They'll never be fixed |
| 17:12.19 | drmessano | There will always be bugs |
| 17:12.23 | pabelanger | haha! Figured it out. Looks like you can load chan_iax2.so multiple times, cause asterisk to freak out. Lets see if I can fix it |
| 17:12.24 | drmessano | 1.4 has them too |
| 17:12.35 | anonymouz666 | every version |
| 17:12.41 | anonymouz666 | but there are bugs you can live with |
| 17:12.43 | drmessano | Yeah, every version does |
| 17:13.23 | drmessano | I never felt 1.4 was stable.. I moved from 1.2 and it was hell... gave me nothing but grief. 1.6.x has been night/day |
| 17:15.02 | anonymouz666 | running 1.4 on lots of customers... heavy load |
| 17:15.10 | anonymouz666 | no deadlocks or crashes |
| 17:15.12 | anonymouz666 | just works |
| 17:15.49 | anonymouz666 | but I agree, at some point I'll need to move ahead |
| 17:19.12 | drmessano | I guess it depends what you're doing with it.. My issues were not load related, but whatever |
| 17:21.54 | alex_voip | other than modifying res_monitor.c or running a custom script is there a way to increase the recording volume when using Monitor? |
| 17:23.25 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 17:24.16 | [TK]D-Fender | alex_voip: Shouldn't have to modify it.. that is what is going through your system. |
| 17:24.27 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:24.32 | [TK]D-Fender | alex_voip: What side is weak? |
| 17:25.12 | anonymouz666 | why not just use the mixmonitor app? |
| 17:25.19 | alex_voip | [TK]D-Fender, called channel is weak, mixmonitor is giving me out of sync recordings, monitor works but low volume |
| 17:25.22 | anonymouz666 | you can use v or V to control the volume |
| 17:26.11 | alex_voip | i was using mixmonitor with v to increase the volume but with 1.6.2 i'm getting out of sync recordings |
| 17:26.29 | alex_voip | monitor works but volume is low |
| 17:30.15 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:30.46 | [TK]D-Fender | alex_voip: What is this called channel? |
| 17:31.08 | *** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net) |
| 17:31.34 | alex_voip | [TK]D-Fender, sip trunk |
| 17:32.05 | [TK]D-Fender | alex_voip: they SHOULDN'T be weak. use VOLUME() at the source to jack them up |
| 17:33.32 | drmessano | alex_voip: which 1.6.2.x release are you on? |
| 17:34.12 | alex_voip | drmessano, 1.6.2.11 |
| 17:34.52 | alex_voip | [TK]D-Fender, ty hadn't thought of VOLUME() still 1.6 n00b |
| 17:36.43 | alex_voip | would still like to understand why mixmonitor results in out of sync recordings any thoughts on what to look at? |
| 17:36.43 | *** join/#asterisk imox1234 (~imox1234@p4FC5C51F.dip0.t-ipconnect.de) |
| 17:38.14 | [TK]D-Fender | alex_voip: You said this is only when using the option to increase gains. Is this correct? |
| 17:39.33 | alex_voip | [TK]D-Fender, no, but i haven't tried without it. This is 1.4 code with no modifications except for the known gotchas |
| 17:40.05 | [TK]D-Fender | alex_voip: I'm confused. Are you now saying you're using 1.4 code on 1.6? |
| 17:40.28 | alex_voip | [TK]D-Fender, sorry poor choice of words....1.4 dialplan |
| 17:41.02 | [TK]D-Fender | alex_voip: You are rather vague about the confirmed scope of this issue |
| 17:41.06 | [TK]D-Fender | alex_voip: Fix this |
| 17:49.52 | *** part/#asterisk KnickLighter (~meh@office.nlsecurity.org) |
| 17:51.55 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:52.20 | t_dot_zilla | anybody know why asterisk would be sending 503 service unavailable messages? |
| 17:52.39 | t_dot_zilla | or what would cause asterisk to send it? |
| 17:56.25 | alex_voip | [TK]D-Fender, Did not intend to be vague. I originally stated i switched from 1.4 to 1.6.2.11 and ASSuMEd that the same dialplan was a given. Got to takers on the original question so went a different route and rather than deal with the sync problem I started looking at Monitor rather than MixMonitor...the rest is history i misspoke and got my arse handed to me :P |
| 17:58.58 | [TK]D-Fender | t_dot_zilla: perhaps you could lok at the call that caused it |
| 17:59.17 | t_dot_zilla | i have a packet capture of the call but it reveal very little |
| 17:59.21 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 17:59.33 | [TK]D-Fender | t_dot_zilla: * SIP DEBUG & CLI. Not external |
| 17:59.43 | [TK]D-Fender | t_dot_zilla: external = useless |
| 18:00.08 | [TK]D-Fender | t_dot_zilla: 503 can be a response to many things for many reasons. |
| 18:00.16 | [TK]D-Fender | t_dot_zilla: like 404. |
| 18:00.51 | pabelanger | t_dot_zilla: https://ietf.org/rfc/rfc3261.txt |
| 18:01.04 | pabelanger | 21.5.4 503 Service Unavailable |
| 18:01.39 | anonymouz666 | it would be nice if all SIP stacks follow the RFC3261 :D |
| 18:02.17 | [TK]D-Fender | anonymouz666: RFC = NFC :p |
| 18:02.44 | anonymouz666 | but at least you understand the right behaviour on when the UAs should send the 503 |
| 18:03.05 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 18:03.34 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 18:28.36 | Letoric | Is there any way I could configure asterisk to monitor a call that we send to an outside line? IE, our oncall staff cell phones? |
| 18:28.57 | Letoric | the only thing I could think of was somehow initiating a conference call |
| 18:29.12 | *** join/#asterisk ahowlader (Adnan@180.234.35.175) |
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| 18:33.56 | *** join/#asterisk nitram (foo@superblob.com) |
| 18:34.23 | [TK]D-Fender | Letoric: Monitor sounds more like SPY in your request |
| 18:34.40 | [TK]D-Fender | Letoric: Which would be "core show application chanspy" |
| 18:35.30 | *** join/#asterisk lopb (~lopezjp@200.80.242.46) |
| 18:35.34 | lopb | hi |
| 18:36.04 | lopb | can somebody tell me where i can find good documentation about freepbx? Or it sould be better to configure the asterisk by hand? |
| 18:36.29 | lopb | wich is the best front end for asterisk? thanks |
| 18:36.35 | [TK]D-Fender | lopb: Perhaps you should ask in #freepbx and not here. Especially as it isn't supported in here. |
| 18:36.52 | [TK]D-Fender | lopb: and "best" depends on your needs |
| 18:37.33 | lopb | well, i'm not going to install it only, i would like to know |
| 18:38.00 | [TK]D-Fender | lopb: Huh? |
| 18:44.36 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 18:46.14 | lopb | [TK]D-Fender , never mind |
| 18:46.16 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 18:46.16 | lopb | :P |
| 18:46.37 | *** join/#asterisk mpe (~mpe@85.24.87.252.generic-hostname.arrownet.dk) |
| 18:46.50 | *** part/#asterisk mpe (~mpe@85.24.87.252.generic-hostname.arrownet.dk) |
| 18:47.11 | alex_voip | I looked into the VOLUME() function and a recent thread in Digium's Asterisk Support Forum suggests this only works when digital/analog conversion is involved. Documentation is sparse can someone confirm this? |
| 18:47.45 | anonymouz666 | no |
| 18:48.02 | anonymouz666 | VOLUME dialplan function can work even in SIP to SIP calls. |
| 18:48.17 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 18:48.49 | lopb | does anyone know how to set up an IVR, with the option to send fax (ffa asterisk module) ant to made the fax answer? |
| 18:51.21 | *** join/#asterisk imox1234 (~imox1234@p4FC5C51F.dip0.t-ipconnect.de) |
| 18:51.44 | [TK]D-Fender | lopb: lopb call rxfax. That is all |
| 18:51.56 | [TK]D-Fender | lopb: or receivefx.. whatever that FFA version of that app is |
| 18:52.24 | alex_voip | anonymouz666, do you happen to know at what level of verbosity i should see something in the log file when i use * and/or # to increase/lower volume in call once VOLUME is set? |
| 18:52.28 | [TK]D-Fender | lopb: And of course you're talking about FREEPBX dialplans which means you shouldn't even be asking that in here |
| 18:52.47 | [TK]D-Fender | alex_voip: what does */# have to do with volume? |
| 18:53.11 | [TK]D-Fender | alex_voip: that function sets the gain FIXED on the channel. It isn't a background listening changer. |
| 18:53.22 | anonymouz666 | [TK]D-Fender: in the source code these are used to increase/decrease the volume on the fly |
| 18:53.40 | anonymouz666 | but I don't know how this works |
| 18:53.40 | [TK]D-Fender | anonymouz666: Source code of what? |
| 18:53.46 | anonymouz666 | FUNC_VOLUME |
| 18:53.54 | [TK]D-Fender | anonymouz666: It shouldn't hijack DTMF... |
| 18:53.58 | [TK]D-Fender | anonymouz666: odd |
| 18:54.48 | anonymouz666 | it is - never worked for me these DTMFs - because I use a backport from FUNC_VOLUME to 1.4. |
| 18:55.30 | anonymouz666 | alex_voip: when I need to increase the volume I write it in dialplan like Set(VOLUME(tx)=2) |
| 18:58.07 | alex_voip | anonymouz666, i've set with Set(VOLUME(RX)=3) but it doesn't seem to affect the volume of the recording generated by Monitor |
| 19:00.02 | anonymouz666 | Monitor is a beast totally different.. as it doesn't use audiohooks. |
| 19:01.40 | alex_voip | anonymouz666, that would explain it i'll just modify res_monitor to call soxmix with -v thank you |
| 19:13.03 | lopb | ok thanks |
| 19:13.04 | lopb | cya |
| 19:30.30 | *** join/#asterisk DogBoy (~john@unaffiliated/dogboy) |
| 19:36.45 | *** join/#asterisk imox1234 (~imox1234@p4FC5C51F.dip0.t-ipconnect.de) |
| 19:39.45 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
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| 19:47.47 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 19:48.20 | wcselby | o/ |
| 19:49.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:49.48 | wcselby | So I've got an applicationmap in features.conf that calls a Macro...is there any way to pass arguments to the macro from the calling channel? i.e instead of just appname => *9,caller,Macro,macroname - can I do appname => *9,caller,Macro,macroname^CALLERID(number), or something similiar? |
| 19:50.28 | wcselby | and then in the macro the value of $ARG1 would be whatever CALLERID(num) evaluates to? |
| 19:55.43 | wcselby | taps channel |
| 19:55.46 | wcselby | this thing on? |
| 19:57.35 | Ikarus | yes sure |
| 19:57.36 | Ikarus | relax |
| 19:57.48 | wcselby | i'm relaxed, was tyring to make a funny |
| 19:57.51 | wcselby | sorry :) |
| 19:58.03 | wcselby | i think i may have found my answer |
| 19:58.07 | wcselby | gotta try something |
| 19:58.16 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 19:59.03 | niekvlessert | hello! anyone with e1-audiocodes-asterisk t-38 fax module and sending big faxes experience? |
| 19:59.49 | niekvlessert | the asterisk fax module should work good |
| 20:00.02 | niekvlessert | but audiocodes mediant - audiocodes ata works 100% |
| 20:00.20 | niekvlessert | but audiocodes mediant - asterisk works not nearly good |
| 20:16.55 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:22.03 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 20:24.12 | [TK]D-Fender | checkout time, later all |
| 20:37.15 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.29.191.201) |
| 20:37.24 | GhOnDiE | hello im having problems with dahdi on my server |
| 20:37.41 | GhOnDiE | keeps coming up with wcfxo: FATAL: Error inserting wcfxo (/lib/modules/2.6.18-194.11.3.el5/dahdi/wcfxo.ko): No such device |
| 20:37.45 | GhOnDiE | and i have no idea why |
| 20:39.56 | Letoric | Anybody know how to force the 'Dial' application to pass the true callerid? I'm using the option o that says it *should* do that, but it isn't working. |
| 20:40.13 | Letoric | Chon, not sure on the DAHDI issue, it's not my strong suit ;( |
| 20:40.41 | GhOnDiE | well i thought i would upgrade to all the latest stuff and now have a non working pstn line |
| 20:40.53 | GhOnDiE | should have stuck at 1.4 and zap |
| 20:41.13 | Letoric | I'm using 1.6.2.11 and it works great for me, outside of areas I struggle in due to lack of experience |
| 20:42.43 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:43.36 | *** join/#asterisk chuckp (~chuckp@c-76-106-198-76.hsd1.fl.comcast.net) |
| 20:45.31 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 20:45.55 | t_dot_zilla | which g722 does asterisk 1.6.1.20 use? |
| 20:46.15 | Qwell | What do you mean which? |
| 20:47.27 | t_dot_zilla | g722.1 (16kps, 24kps, or 32kps), or g722.1c (16kps, 24kps, or 32kps)? |
| 20:47.43 | t_dot_zilla | or g722.1c 48kps ? |
| 20:47.44 | Qwell | G.722 is G.722. G.722.1 is G.722.1. |
| 20:47.52 | Qwell | They are different things. |
| 20:47.56 | t_dot_zilla | ah |
| 20:48.19 | t_dot_zilla | do they transcode? |
| 20:49.08 | t_dot_zilla | are they related ? |
| 20:49.12 | *** join/#asterisk adyn (~adyn@unaffiliated/adyn) |
| 20:49.16 | pabelanger | GhOnDiE: How did you install DAHDI? |
| 20:53.58 | GhOnDiE | from source |
| 20:54.34 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:05.17 | *** part/#asterisk ^scott^ (~scott@stthom.org) |
| 21:10.43 | jamko | Any ideas as to why * would not be "hearing" the dtmf tones entered in VoiceMailMain? I have tried 2 different phones and * is completely ignoring what I enter for mailbox and password. |
| 21:11.58 | pabelanger | jamko: NAT? |
| 21:12.53 | ectospasm | jamko: turn on dtmf debug and see if Asterisk even sees those digits |
| 21:14.31 | jamko | no nat.. I will try dtmf debug. Normally asterisk shows the attempted mailbox and password at the cli, but here it is showing blank password, and the mailbox as my callerid, which is not what I am entering. The box is using realtime static. |
| 21:14.53 | pabelanger | inband DTMF or rfc2833? |
| 21:15.35 | jamko | I have rfc2833 specified |
| 21:16.58 | chuckp | Anyone ever run into issues setting your outbound CID to a toll free number? |
| 21:17.34 | ectospasm | chuckp: that would depend entirely on your provider, whether they allow that or not. |
| 21:17.36 | chuckp | I have tried both flowroute and sipstation and when I set my CID to a 888 # it just fails to connect on some numbers just blank air, when set to an areacode it works just fine |
| 21:17.44 | chuckp | ectospasm: okay |
| 21:18.18 | chuckp | strange that it would be rejected back, usually I only see that with PRI providers... where you are restricted to DIDs in your control |
| 21:18.26 | GhOnDiE | can i run zaptel-1.4 with asterisk 1.6? |
| 21:18.36 | GhOnDiE | i cant get dahdi to work with my card |
| 21:18.38 | ectospasm | GhOnDiE: no, and you wouldn't want to |
| 21:18.48 | ectospasm | GhOnDiE: what card? |
| 21:20.18 | jamko | ectospasm: what is the command for dtmf debug? using 1.6 |
| 21:21.10 | Letoric | you have to edit the logger.conf to turn it on |
| 21:21.21 | ectospasm | jamko: add "dtmf" to /etc/asterisk/logger.conf (the "console" line), and run "logger reload" in the Asterisk CLI |
| 21:21.30 | Letoric | yeah, what he said ;) |
| 21:21.34 | GhOnDiE | its an x100p |
| 21:21.52 | GhOnDiE | just had a look through dmesg and found this Invalid/unknown operating mode specified. Please choose one of: |
| 21:21.52 | GhOnDiE | 0: FCC |
| 21:21.52 | GhOnDiE | 1: CTR21 |
| 21:21.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:21.54 | ectospasm | GhOnDiE: I don't even think DAHDI has X100P drivers anymore |
| 21:22.32 | GhOnDiE | it still seems to have the drivers in the modules folder |
| 21:22.47 | ectospasm | what's the kernel module name? |
| 21:23.59 | GhOnDiE | wcfxo |
| 21:24.04 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 21:24.17 | GhOnDiE | how can i select the operating mode |
| 21:24.23 | GhOnDiE | i need to set to ctr21 |
| 21:24.39 | ectospasm | you should be able to "echo {0,1} > /sys/module/<module_name>/parameters/blah" where "blah" is something like opermode... |
| 21:25.21 | ectospasm | GhOnDiE: you'll have to do an "ls /sys/module/wcfxo/parameters" to see what module parameters are available |
| 21:26.21 | GhOnDiE | ok thanks |
| 21:26.28 | GhOnDiE | i think thats why the module is not loading |
| 21:27.49 | ectospasm | then, set /etc/modprobe.d/dahdi.conf to contain "options wcfxo blah=1" |
| 21:27.58 | ectospasm | and reload DAHDI |
| 21:28.05 | GhOnDiE | ahh cool thanks will try that |
| 21:28.33 | ectospasm | You may be able to get away with "echo 1 > /sys/module/wcfxo/parameters/blah" but that won't be permanent |
| 21:29.35 | GhOnDiE | i will do a full reboot |
| 21:29.51 | ectospasm | GhOnDiE: no need, just unload and reload the wcfxo driver |
| 21:30.05 | GhOnDiE | ive set options wcfxo mode=1 |
| 21:30.12 | GhOnDiE | hopefully that will do it |
| 21:30.55 | ectospasm | you could find out right away with "service dahdi restart" or "modrobe -r wcfxo && modprobe wcfxo" |
| 21:31.07 | ectospasm | ...be sure to stop Asterisk beforehand |
| 21:31.11 | GhOnDiE | ok |
| 21:36.05 | GhOnDiE | nope no joy |
| 21:36.23 | GhOnDiE | also tried options wcfxo opermode=1 |
| 21:36.30 | GhOnDiE | and a modprobe dahdi |
| 21:36.34 | GhOnDiE | and modprobe wxcfo |
| 21:36.42 | GhOnDiE | and modprobe wcfxo even |
| 21:37.32 | GhOnDiE | still getting wcfxo: FATAL: Error inserting wcfxo (/lib/modules/2.6.18-194.11.3.el5/dahdi/wcfxo.ko): No such device |
| 21:37.45 | GhOnDiE | when i do a service dahdi start |
| 21:39.10 | ectospasm | GhOnDiE: does dahdi_hardware report anything? |
| 21:41.53 | Katty | ecto |
| 21:41.54 | Katty | spasm |
| 21:41.58 | Katty | *hee* |
| 21:43.05 | GhOnDiE | pci:0000:00:0e.0 wcfxo- 1057:5608 Wildcard X100P |
| 21:43.09 | GhOnDiE | yeah |
| 21:44.04 | Katty | p3nguin: is it time to go yet?! |
| 21:48.48 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 21:51.41 | dadavoip | anyone familiar with the Originate command |
| 21:53.24 | p3nguin | katty: Nearly, I reckon. |
| 21:53.42 | p3nguin | I've already finished for the day AND went to the doctor. |
| 21:53.54 | p3nguin | (not trying to rub it in, though) |
| 21:54.30 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 21:55.02 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 21:55.03 | GhOnDiE | finally sorted my problem |
| 21:55.09 | GhOnDiE | needed to do an insmod |
| 21:56.53 | tzafrir_laptop | GhOnDiE, insmod? Of what, exactly? |
| 21:57.16 | GhOnDiE | insmod wcfxo opermode=1 |
| 21:57.27 | Katty | p3nguin: what'd the doctor say? |
| 21:57.27 | GhOnDiE | it needed to know what signalling type to use |
| 22:00.20 | dadavoip | did you know: |
| 22:00.32 | dadavoip | penguins mate for live |
| 22:00.37 | dadavoip | live |
| 22:00.40 | dadavoip | life |
| 22:12.27 | *** join/#asterisk rossand (~aross@114.Red-81-44-114.dynamicIP.rima-tde.net) |
| 22:12.37 | *** part/#asterisk rossand (~aross@114.Red-81-44-114.dynamicIP.rima-tde.net) |
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| 22:26.57 | jamko | Where does * pull the "fromstring" , "email subject" and "email body" from, when emailing voicemail messages, using voicemail with odbc storage and realtime static users? I have changed the default email body etc in voicemail.conf, reloaded *, but it still sends the default text that is no longer in voicemail.conf. It must be pulling from somewhere else, but where? |
| 22:29.04 | dadavoip | i think its in the voicemail.conf |
| 22:30.01 | dadavoip | are you using asterisk source? |
| 22:30.30 | dwayne | offers Katty 8 scoops of ice cream |
| 22:33.38 | xheliox | where's my friggin ice cream? :( |
| 22:33.54 | myster | jamko, have you looked in /etc/asterisk/vm_email.inc ? |
| 22:34.18 | myster | good luck |
| 22:34.52 | dwayne | banana splits for the whole room! |
| 22:34.59 | dadavoip | does anyone know how vicidial sends a call to the asterisk dialpan? |
| 22:35.10 | dadavoip | meaning what technology is envolved |
| 22:35.35 | dadavoip | i see it uses the manager interface to asterisk |
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| 22:49.22 | jamko | dadavoip: I compiled from source... |
| 22:49.29 | jamko | myster: I do not see that file |
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