| 00:03.26 | Freeaqingme | anygivenname, has it ever occurred that there will never be a gui as powerful as the command line interface? |
| 00:08.08 | paulc | I got a lot of time for the command line. In fact, I'm using IRC with it now (via irssi). |
| 00:08.46 | paulc | Seems to me that a lot of the time, slapping a GUI on Asterisk restricts the power of what you can do with it. Fine for "just a PBX", but kills it for "the extra stuff" |
| 00:12.26 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-66-67-121-128.rochester.res.rr.com) |
| 00:28.54 | cusco | paulc: is there something you can do on irssi that you can't do on xchat besides the need for X server ? |
| 00:29.06 | cusco | can't even compare that |
| 00:29.28 | cusco | by the way, I had a look at edgebox, best gui I've seen |
| 00:29.50 | cusco | but as stated no gui will be as versatile as cli |
| 00:29.57 | paulc | cusco: Yeah, I liked X-Chat, used it before... but when I'm at work I can SSH in to my box at home and run irssi in a screen session. Works well. |
| 00:30.19 | cusco | paulc: you can ssh with X11 forwarding and use xchat too |
| 00:30.22 | cusco | so? |
| 00:30.26 | paulc | That said, X-Chat through SSH proxy could work.. but I kinda like irssi :) |
| 00:30.28 | paulc | yeah |
| 00:30.34 | paulc | 6 of one, half a dozen of the other |
| 00:30.36 | cusco | it could work |
| 00:31.00 | cusco | anygivenname: have a look at edgebox demo |
| 00:33.01 | cusco | paulc: my point was, irssi vs gui irc client *NOT* *COMPARABLE* *TO* asterisk silly command lines vs some asterisk gui front-end |
| 00:37.05 | paulc | Ah, yes.. sorry - that slipped through the cracks.. I'm fading, been a long day |
| 00:37.40 | paulc | my point was.. GUI for Asterisk tends to work for "we configure it this way, and we'll let you fill in the blanks to make it work" |
| 00:37.57 | paulc | versus.. full access to config files and rolling them yourself.. to infinity and beyond, etc etc.. |
| 00:44.44 | Freeaqingme | yeah |
| 00:45.04 | Freeaqingme | we've been developing a gui ourselves too so customers can also understand what's happening |
| 00:45.16 | Freeaqingme | (you dont want to have to explain to customers how asterisk's dialplans work) |
| 00:48.59 | cusco | Freeaqingme: no, you actually want to sell some assistance support and earn money off it :P |
| 00:49.13 | cusco | Freeaqingme: have you ever seen edgebox's gui? |
| 00:49.18 | cusco | I got amazed by it |
| 00:50.00 | Freeaqingme | well that would be most profitable indeed. But most of my customers are soho and got a few extensions and need a simple auto attendant menu |
| 00:50.52 | Freeaqingme | looking at some pictures now, looks good indeed |
| 00:55.29 | cusco | Freeaqingme: try the demo |
| 00:55.36 | cusco | admin |
| 00:55.40 | cusco | newy0rk |
| 00:55.47 | cusco | is password |
| 00:56.18 | cusco | https://demo.edgebox.com:8011/ |
| 00:56.35 | *** join/#asterisk Marvelous (~CE0@41.153.199.17) |
| 00:56.44 | cusco | or used to be |
| 00:56.45 | cusco | gah |
| 01:07.16 | Freeaqingme | cusco, what's your relation to the company behind it? |
| 01:07.55 | Freeaqingme | also, login fails, but you may have figured that already |
| 01:11.35 | cusco | used to work when I subscribed to the demo |
| 01:11.41 | cusco | wich you can do on their website |
| 01:12.07 | cusco | Freeaqingme: I just assisted some ... how do you call it... |
| 01:12.09 | cusco | presentation |
| 01:12.13 | cusco | on how it works |
| 01:12.14 | Freeaqingme | kk |
| 01:12.26 | cusco | from a guy that works with them |
| 01:12.45 | cusco | we implement asterisk in a IT helping services call centre |
| 01:13.16 | cusco | and these IT helping services might sell these edgeboxe's thats how I know them |
| 01:13.29 | cusco | anyway, they did an amazing job creating the gui |
| 01:13.51 | rogersja | anyone out there actively run asterisk on ec2? |
| 01:14.31 | Freeaqingme | rogersja, I wouldnt do that for anything business critical with their SLA's and recent outages |
| 01:14.43 | cusco | lol |
| 01:15.22 | cusco | tho I have used ec2 before bad publicity and also enjoyed their fast services |
| 01:15.44 | rogersja | i've only heard distant stories about it, sounded interesting, but wanting to see if its done en mass |
| 01:16.08 | cusco | I didn't use asterisk on it tho... |
| 01:16.15 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 01:16.15 | Freeaqingme | rogersja, I think most people will want to hook up a t1 line on it eventually |
| 01:16.28 | Freeaqingme | and getting a t1 cable into the clouds is a tough job |
| 01:16.48 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-66-67-121-128.rochester.res.rr.com) |
| 01:16.51 | rogersja | pft who needs t1 :p |
| 01:17.05 | cusco | rogersja: what would you preffer? |
| 01:17.55 | cusco | at least you would be connected directly to telco, otherwise you would have to get some number, comming from a t1 then trough some server u dunno until it get to you... |
| 01:18.53 | rogersja | direct to telco is so last year, lol |
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| 01:20.06 | rogersja | getting a high quality symetrical link is pretty easy nowadays |
| 01:20.17 | rogersja | at least it is up here |
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| 01:20.51 | cusco | what is high quality symetrical link ? |
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| 01:21.33 | rogersja | something like oc1 |
| 01:21.40 | Freeaqingme | rogersja, it is, but not backed by an ATM network |
| 01:21.41 | cusco | what is that? |
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| 01:21.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:21.57 | cusco | ow optical |
| 01:22.24 | rogersja | ya the providers up here are all for it |
| 01:22.27 | cusco | our E1 connection comes trough optical fiber |
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| 01:22.36 | cusco | unltil our server room |
| 01:22.53 | cusco | then we have what they call a 'MVR' that translates fiber into thos bnc connectors |
| 01:23.24 | cusco | we still call it a E1 PRI line, because thats what it says on the digium card it connects to |
| 01:23.27 | cusco | lol |
| 01:23.35 | rogersja | these sort of dc1 t1 type connections seem to be going away |
| 01:23.39 | rogersja | at least from what i've seen |
| 01:23.48 | cusco | what is dc1? |
| 01:23.52 | rogersja | even residential here is moving the 'fibre' |
| 01:24.05 | rogersja | sorry i meant ds1 |
| 01:24.54 | rogersja | mind you dc1 is long gone too :P |
| 01:25.25 | cusco | I read that ds1 is the digital signal (since we moved from analog) |
| 01:25.38 | rogersja | yep pretty much |
| 01:25.50 | cusco | besides voip, thats what everybody uses, right? |
| 01:25.56 | Freeaqingme | wtf |
| 01:26.01 | Freeaqingme | there's a windows build of asterisk |
| 01:26.30 | cusco | I mean, most internet services now ofer a phone port on the modem... and you get voip with portable number |
| 01:26.37 | cusco | connected to isdn |
| 01:26.52 | cusco | but besides that, ds1 is the most used |
| 01:26.53 | cusco | right? |
| 01:27.13 | rogersja | ds1 technically is just the signalling pattern |
| 01:27.15 | cusco | Freeaqingme: someone here already stated that they use it sucessfully |
| 01:27.26 | cusco | yes.. well... |
| 01:27.30 | Freeaqingme | cusco, on April 1st? :P |
| 01:27.31 | cusco | what else is there? |
| 01:27.34 | cusco | Freeaqingme: no, really |
| 01:27.43 | cusco | they got mad when I joked about windows |
| 01:27.43 | cusco | lol |
| 01:27.48 | Freeaqingme | whehe |
| 01:28.01 | rogersja | well where I am its all DOCSIS3 |
| 01:28.11 | rogersja | for residentail, and light business |
| 01:28.26 | cusco | well true, I have EuroDOCSIS3 at home lol |
| 01:28.29 | Freeaqingme | rogersja, qam256? |
| 01:28.55 | cusco | but phone lines placed by telephone company still run trough country ground |
| 01:29.21 | cusco | and maybe my internet provider offers me a good deal on phone, carried trough that docsis3 but its VOIP |
| 01:29.35 | rogersja | Freeaqingme: yes |
| 01:30.52 | rogersja | cusco, your telephone company likely uses fiber right beside that of your IP |
| 01:32.04 | cusco | ok I wouldn0t know but every building here has had normal phone line for years... |
| 01:32.07 | cusco | so.. |
| 01:33.00 | rogersja | of course they do, but you dont think they run 22 gauge copper lines accross the country and under the water do you |
| 01:33.40 | cusco | lol |
| 01:34.08 | rogersja | if you are in any sort of medium to large office building i doubt your copper telephone lines go anyfurther than a mechanical room in the basement |
| 01:35.17 | cusco | ko ok I got it |
| 01:35.38 | rogersja | Freeaqingme: the provider here offers 100Mbps down 5Mbps up in all areas, and 'fiber' to some places |
| 01:35.48 | rogersja | that is residential of course |
| 01:36.13 | Freeaqingme | heh |
| 01:36.21 | carrar | nice |
| 01:36.22 | rogersja | if you want a couple of oc192's they'll do that for you too |
| 01:36.24 | cusco | my isp offers 1Gb up and 1Gb down in residencial package |
| 01:36.28 | Freeaqingme | I'm enjoying my vDsl connection with an effective downspeed of 27Mb |
| 01:36.30 | cusco | expensive |
| 01:36.37 | cusco | at least they say they do |
| 01:36.54 | carrar | oh yeah, my isp offers 100Gb down and up!!! |
| 01:37.05 | cusco | residencial |
| 01:37.11 | Freeaqingme | lol, I'm doing 10GB / day |
| 01:37.20 | carrar | err sorry 100bits |
| 01:37.22 | cusco | and I'm on lousy 30Mb/2Mb |
| 01:37.28 | cusco | ahh |
| 01:37.36 | cusco | 100Mb x 10 |
| 01:37.36 | carrar | 200 on a good day |
| 01:37.59 | carrar | 300 baud or bust! |
| 01:38.47 | rogersja | anyways that seemed to have gotten a bit off topic |
| 01:39.53 | cusco | well nobody's talking about anything else neither... |
| 01:40.12 | cusco | http://icanhascheezburger.files.wordpress.com/2008/02/funny-pictures-spyware-cat-computer-monitor.jpg |
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| 01:45.18 | cusco | hi gruvfunk |
| 01:45.24 | gruvfunk | yo |
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| 01:46.39 | Micc | Why do I have to have a service contract to download the cisco 7960G SIP firmware? Does it cost money? |
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| 06:38.56 | sawgood | Is there a way to do work on the dialplan 'live'? By this I mean, could I do something like this from the CLI (so I could troubleshoot calls): |
| 06:38.59 | sawgood | exten => h,1,Set(CDR(userfield)=Hangupcause:${HANGUPCAUSE}) |
| 06:39.26 | sawgood | Then, while at the CLI, any completed call would in theory show the hangupcause |
| 06:39.47 | ChannelZ | dialplan add extension |
| 06:39.49 | sawgood | I understand what I put would write it to the CDR, but I could re-word it to something like a NooP |
| 06:41.42 | sawgood | ChannelZ: with your example, I would need to know the specific context I would want the exten to apply to right? |
| 06:42.01 | ChannelZ | yes.. same as if you put it in the dialplan from extensions.conf |
| 06:42.12 | sawgood | I was looking for a way to make the dialplan do something like this: |
| 06:42.30 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 06:42.39 | sawgood | "Hey Asterisk, until I tell you to stop doing it, please show me on the CLI what the hangupcause is for any call you terminate ... |
| 06:42.51 | jacc0 | hi all |
| 06:42.57 | ChannelZ | ...yes... it would do that |
| 06:43.20 | sawgood | But, what if I do not put the statement in the right context? because there are many different contexts? |
| 06:43.34 | ChannelZ | in this case "until I tell you to stop" would be "dialplan remove extension" or re-loading the dialplan/restarting asterisk |
| 06:43.37 | kaldemar | sawgood: use the Verbose app to output it in your dialplan. you can set a verbosity level for the print so that you can enable and disable it in CLI with setting the verbosity level. |
| 06:44.12 | kaldemar | sawgood: if you put it in the wrong context, then you obviously don't see it. your dialplan, your contexts... you should know the right one(s). |
| 06:44.13 | ChannelZ | There's no getting around your contexts |
| 06:45.23 | sawgood | I don't have a problem with the contexts ... its just say you log into an Asterisk box ... and you don't have the past knowledege of how the dialplan is all setup, but yet you need to troubleshoot NOW ... from the CLI ... so I was wondering what an approach to something like this might be |
| 06:46.00 | ChannelZ | I guess you could probably monitor Manager |
| 06:46.52 | kaldemar | if someone is troubleshooting, the first step would be to enable some verbosity anyway. |
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| 06:47.08 | sawgood | say the box is set at core set verbose 5 |
| 06:48.15 | sawgood | exten => h,n,NoOp(Hangupcause${HANGUPCAUSE}) |
| 06:48.30 | sawgood | how could I get this statement to apply to every context? |
| 06:48.37 | ChannelZ | It needs to go in all the contexts you need to see that from. |
| 06:48.48 | sawgood | ChannelZ: thank you! |
| 06:48.58 | kaldemar | sawgood: use app Verbose. |
| 06:49.35 | ChannelZ | I think he gets that, but doesn't want to have to add an 'h' extension to 15 contexts or whatever he has going |
| 06:49.49 | kaldemar | sawgood: that way they are not always printed, and you can reduce the output by using different verbosity levels. |
| 06:50.24 | kaldemar | if the dialplan is structured in a nice way, 1 h extension is enough. |
| 06:51.39 | kaldemar | nice way in this case would mean no unnecessary usage of Goto and common starting contexts. |
| 06:51.43 | sawgood | what if the context has NO h statements? Are these 'native' to the program? |
| 06:52.00 | kaldemar | if there is no h, nothing happens. |
| 06:52.02 | ChannelZ | eh? |
| 06:52.19 | *** join/#asterisk asterisk-Tester (~chatzilla@77.42.241.114) |
| 06:52.25 | sawgood | meaning, in my contexts ... there is no mention of h any any contexts now |
| 06:52.43 | ChannelZ | so nothing special happens |
| 06:53.12 | sawgood | So, wouldn't I have to have h,1 (before h,n) then? |
| 06:53.40 | ChannelZ | yes |
| 06:53.42 | kaldemar | of course you have to have h,1 |
| 06:53.50 | sawgood | I'll try it on an * box ... and I'll be right back |
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| 07:05.34 | sawgood | it worked! |
| 07:06.19 | ChannelZ | quick, change your pants! |
| 07:08.55 | asterisk-Tester | Hi, waifotsilence application does not wait for continuous silence, it's just the sum of the silences that should be equal to the one passed in, right ? |
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| 07:12.28 | ChannelZ | I believe it waits for a continuous stretch of silence the length of whatever parameters you gave it |
| 07:13.29 | kaldemar | asterisk-Tester: there is a separate option for how many times a silence of defined length should be observed before returning. |
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| 07:13.50 | kaldemar | i.e. iterations. see "core show application WaitForSilence" |
| 07:14.36 | ChannelZ | yes but the real question is are they contiguous or not? Does 1000,2 mean it will return after "blah blah blah blah [1 sec of silence] blah blah blah blah [1 sec of silence]" |
| 07:15.32 | ChannelZ | The way it's written it sounds like it would only return after 2 seconds in one place (otherwise why say 'Typically you will want to include two or more calls to WaitForSilence...' |
| 07:15.50 | ChannelZ | easy enough to test anyway |
| 07:16.22 | asterisk-Tester | kaldemar: i am getting the following output: http://pastebin.com/ENaS19e2 |
| 07:16.44 | asterisk-Tester | doesn't this indicate that it could have 0 ms of silence between other silence packets ? |
| 07:16.44 | kaldemar | and quite odd to have a parameter to multiply the defined length as the length can be written as is. |
| 07:18.02 | kaldemar | if you require 5000 ms of silence, nothing under 5000 ms counts. |
| 07:18.36 | ChannelZ | huh ok so the iterations can be discontiguous |
| 07:19.35 | asterisk-Tester | ChannleZ: Exactly, so like u said in ur example it does return with blah blah between the 2 seconds |
| 07:19.41 | asterisk-Tester | right? |
| 07:20.15 | ChannelZ | it would return if it heard 2 seconds of silence contiguously, OR 1 second of silence anywhere, twice |
| 07:20.44 | asterisk-Tester | ChannleZ: ok thx, just double checking :-) |
| 07:22.35 | ChannelZ | Which makes sense being there is an iteration parameter, just the application description is kind of misleading. You'd only need to call it more than once if you needed to detect differing lengths of silence |
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| 07:29.09 | ChannelZ | uh oh |
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| 07:33.00 | hugogee | greets all :D |
| 07:33.51 | hugogee | Must i install dahdi before asterisk on Openbsd? |
| 07:34.25 | kaldemar | the real question is do you need DAHDI? |
| 07:34.47 | hugogee | I would like to use a digium card. I believe dahdi are the libs/drivers? |
| 07:34.48 | kaldemar | i.e. are you planning to use the MeetMe app or IAX2 trunking. |
| 07:35.41 | kaldemar | you need dahdi for the card. it provides a driver and configuration tools for it. |
| 07:35.47 | hugogee | I am not planning on using MeetMe (sounds pr0n'ish) hehe. Nor iax2. |
| 07:35.48 | ChannelZ | you must build DAHDI first if you want Asterisk to build DAHDI support. |
| 07:36.08 | hugogee | so i will have to check it out w/ svn?? |
| 07:36.30 | ChannelZ | well.. or just get the release tar off asterisk.org |
| 07:37.54 | hugogee | ChannelZ: The release tar, is it a pkg for linux? |
| 07:37.58 | ChannelZ | or I guess you said openbsd |
| 07:38.03 | hugogee | ya. |
| 07:38.17 | ChannelZ | Sorry. I guess that would be separate |
| 07:38.45 | kaldemar | http://www.asterisk.org/downloads <-- compile from source |
| 07:39.01 | hugogee | Yeah not as much info for the platform. I know there are easier ways but i would rather sweat a little in the begining instead of when it breaks. :D |
| 07:39.20 | kaldemar | you might also need libpri, depending on your card. |
| 07:39.21 | ChannelZ | well assuming there is an openbsd port.. but I have no idea |
| 07:39.59 | hugogee | kaldemar: How do i find out the requirements for the card? |
| 07:40.23 | ChannelZ | is it a PRI card or an analog interface card? |
| 07:41.27 | hugogee | pri |
| 07:42.11 | ChannelZ | then you need libpri too |
| 07:42.29 | ChannelZ | but again, on BSD.. *holds up hands* |
| 07:43.48 | hugogee | diguims TDM410PLF |
| 07:44.08 | hugogee | well i will do some research. |
| 07:44.22 | hugogee | kaldemar: ChannelZ: thank you!! |
| 07:44.30 | ChannelZ | that's POTS.. |
| 07:44.54 | ChannelZ | so you don't need libpri |
| 07:46.02 | hugogee | Yes i will be using 2 POTS with the card. I am quite new to voip. |
| 07:46.29 | hugogee | But i will do my due diligence. :D |
| 07:46.49 | hugogee | ChannelZ what is PRI? |
| 07:47.16 | ChannelZ | T1, E1 etc |
| 07:47.40 | hugogee | oh. I better go do some more reading. Thanks guys. |
| 07:48.01 | hugogee | I am excited! |
| 07:48.19 | ChannelZ | good luck. Though I have to ask is there a reason you have to do this on openbsd vs linux? |
| 07:48.51 | hugogee | More secure and clean OS. my 2cents |
| 07:48.55 | ChannelZ | (just dunno how much support there is for asterisk on openbsd, how well they keep up with updates etc.) |
| 07:49.31 | hugogee | There is a port for 1.8.2. No gui but that doesnt bother me. |
| 07:50.00 | ChannelZ | well that's good. The key is the drivers though in your case |
| 07:50.12 | hugogee | There are only a few config files from what i have been reading. |
| 07:50.27 | ChannelZ | It's not too terribly difficult no |
| 07:50.33 | hugogee | Yes that is a good point. |
| 07:50.34 | *** join/#asterisk the_nation (~bla@adsl-75-19-113-197.dsl.yntwoh.sbcglobal.net) |
| 07:50.50 | ChannelZ | is there a port of DAHDI? |
| 07:50.59 | hugogee | sadly no. |
| 07:51.17 | ChannelZ | then you're screwed |
| 07:51.25 | hugogee | If i build it successfully all should be down hill from there. |
| 07:51.37 | ChannelZ | I don't think a linux hardware driver is going to build in openbsd |
| 07:51.38 | hugogee | It is possible. :D |
| 07:51.53 | ChannelZ | unless they have some kind of crazy vm-for-drivers kind of thing |
| 07:52.19 | kaldemar | there's a freebsd port though. |
| 07:52.30 | hugogee | well the source depends on certain libs. if they are available for openbsd then it should work. |
| 07:52.31 | kaldemar | don't know if that'll help some. |
| 07:52.51 | the_nation | morning. I have a working asterisk box running my GV. I basically copied/pasted a known working solution just to get going. I have just read over *: the future... and understand most of the dial plans ect. I do have a question on one of the parts of the plan and hope someone can explain it a little better. any experts :) |
| 07:52.55 | ChannelZ | the root of DAHDI is being able to talk to your hardware. |
| 07:53.02 | hugogee | Ya, the port should be a single command to install. DAHDI well that why im here. |
| 07:53.11 | ChannelZ | it's a device driver |
| 07:53.26 | hugogee | right. |
| 07:53.48 | ChannelZ | I don't know much about bsd but I assume they have an entirely different driver model |
| 07:54.10 | kaldemar | the_nation: just go ahead and ask your question. |
| 07:54.52 | the_nation | part of the google in context. first line exten => user@gmail.com, 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged) |
| 07:55.08 | hugogee | ChannelZ: let you know how it works. |
| 07:55.21 | the_nation | if matched leads to exten => user@gmail.com, n(bridged),Bridge(${DB_DELETE(gv_dialout/channel)}, p) |
| 07:55.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 07:55.28 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 07:57.42 | the_nation | any ideas? |
| 07:58.06 | kaldemar | the_nation: is the point of the database functions to limit the number of calls through google voice to one? |
| 07:58.58 | the_nation | I guess I really do not know, I was hoping for clarity |
| 08:00.01 | the_nation | thats really the only 2 instances I notice of it |
| 08:00.03 | kaldemar | so if a key gv_dialout/channel exists in asterisk's database, remove it and try to bridge the current channel to whatever gv_dialout/channel had in its value. |
| 08:01.11 | kaldemar | nothing more to in those two lines. the rest depends on other things in the dialplan. |
| 08:03.02 | the_nation | interesting. would you have any clues as to the location where the key would originally be set |
| 08:03.12 | the_nation | just to give me an idea to what to look for |
| 08:04.06 | the_nation | when I saw bridged, I assumed it was for use with a call back hack for GV, but after reading through the book, I was still a bit baffeled. |
| 08:05.09 | kaldemar | the_nation: look for a line that looks like Set(DB(gv_dialout/channel)=...) |
| 08:06.27 | the_nation | I've looked through extensions, gtalk, sip and jabber and those are the only 2 instances of it hmm |
| 08:07.55 | kaldemar | something must set that variable, otherwise the extension does nothing. |
| 08:09.38 | the_nation | no, those were just the 2 lines I was confused about. full context exten => superm1@gmail.com, 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged) |
| 08:09.38 | the_nation | exten => superm1@gmail.com, n, NoOp(Callerid ${CALLERID(name)}) |
| 08:09.38 | the_nation | exten => superm1@gmail.com, n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)}) |
| 08:09.38 | the_nation | exten => superm1@gmail.com, n, Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) |
| 08:09.38 | the_nation | exten => superm1@gmail.com, n, Dial(SIP/101, 180, D(:1)) |
| 08:09.38 | the_nation | exten => superm1@gmail.com, n(bridged),Bridge(${DB_DELETE(gv_dialout/channel)}, p) |
| 08:09.43 | kaldemar | ~pb |
| 08:09.43 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 08:09.57 | the_nation | redface |
| 08:10.00 | the_nation | sorry |
| 08:10.55 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 08:12.21 | the_nation | ok, see if this shines a little more light on it http://pastebin.com/ysgbsw6d |
| 08:12.30 | phix | hihi |
| 08:12.47 | phix | gmail? can you interact with google talk with asterisk? |
| 08:12.57 | the_nation | indeed |
| 08:12.58 | kaldemar | so if something has set gv_dialout/channel in the database (sounds like an incoming call should set it to CHANNEL(name) and connect to something so the channel stays up), bridge to it, else dial SIP/101. |
| 08:14.33 | kaldemar | the_nation: is there some other application that would set the value in the database via AMI for example? |
| 08:14.46 | kaldemar | the_nation: it's possible that it isn't done via dialplan at all. |
| 08:15.41 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 08:17.33 | the_nation | I'll try and do more looking around. the more I look at it, the more I think it is for initiating the web callback for GV and bridging it |
| 08:17.33 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:19.17 | the_nation | if you would'nt mind, check the following link and search for gv_dial http://forums.plugpbx.org/index.php?topic=112.0;wap2 |
| 08:23.52 | the_nation | looks like the web callback is the reason for that part of the dialplan not needed, but if its not hurting i wont change (: |
| 08:24.50 | hugogee | ChannelZ: You are correct about dahdi + openbsd. On the bright side i found a port for freebsd. :D |
| 08:31.31 | kaldemar | the_nation: it is set in [gv-outbound] |
| 08:32.57 | hugogee | nite folks |
| 08:33.02 | the_nation | nite |
| 08:33.39 | the_nation | I did read that, which is why I posted the link. that is the old/hack way to initiate a web callback through asterisk. Not the way it is done with gtalk native now |
| 08:34.22 | the_nation | no wonder I was soooo lost, that part of the plan was not included lol teach me to not write it myself! |
| 08:37.34 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 08:43.52 | the_nation | thanks again kaldemar, I'm sure you will see me again shortly as I dive in a little deeper (: |
| 09:03.52 | *** join/#asterisk glam (~glam@113.88.231.72) |
| 09:07.08 | *** join/#asterisk imcdona (imcdona@2001:470:e8f1:1:99ff:d879:c23b:b3ee) |
| 09:07.40 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ojpmfilvsitfhzwg) |
| 09:10.00 | asterisk-Tester | Any idea why I might be getting this notice in cli ? "Dropping incompatible voice frame on IAX2/ir-2550 of format slin since our native format has changed to 0x4 (ulaw)" |
| 09:11.24 | anygivenname | my asterisk isnt configured in a way that my sip client can connect to it....can anyone help ? |
| 09:16.33 | *** join/#asterisk e3eli3h (~e3eli3h@mail.fxcc.com) |
| 09:17.17 | *** join/#asterisk linuxgeek (~uppal@2001:470:1f06:123f::2) |
| 09:18.57 | linuxgeek | hello everyone |
| 09:19.50 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 09:20.07 | jacc0 | hi |
| 09:20.44 | anygivenname | sudo /usr/sbin/astman 127.0.0.1 gives connection refused |
| 09:21.03 | jacc0 | @anygivenname: start by filling sip.conf followd by making a dialplan in extensions.conf |
| 09:21.39 | anygivenname | jacc0: what should I do in sip.conf ? |
| 09:21.48 | jacc0 | al your sip clients |
| 09:23.41 | anygivenname | jacc0: seems I should change some settings in sip.conf but I dont know which exactly |
| 09:29.18 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:31.21 | anygivenname | can anyone help me with sip.conf & extensions.conf |
| 09:33.50 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 09:36.26 | *** join/#asterisk linuxgeek (~uppal@2001:470:1f06:123f::2) |
| 09:38.48 | ectospasm | anygivenname: be more specific, what doesn't work? |
| 09:38.56 | ectospasm | ~thebook | anygivenname |
| 09:39.10 | ectospasm | ~thebook |
| 09:39.10 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 09:39.40 | ectospasm | anygivenname: ^ that is a great reference, it can help you get started |
| 09:42.27 | anygivenname | ectospasm: I installed asterisk.....failing to connect to it using sip client |
| 09:43.17 | anygivenname | ectospasm: there r difinetley conf files that I need to change some lines in but I do not know |
| 09:43.22 | ectospasm | anygivenname: that could be from an entire host of configuration problems. |
| 09:43.35 | ectospasm | anygivenname: like I said, consult the book |
| 09:44.04 | ectospasm | Asterisk configuration is mighty confusing at first |
| 09:44.30 | ectospasm | if it's not configured properly it won't work |
| 09:44.44 | ectospasm | it took me two weeks to get a dialtone in Asterisk my first time. |
| 09:45.00 | ectospasm | ...but that was before the luxury of O'Reilly Asterisk books. |
| 09:48.42 | kaldemar | anygivenname: astman is not a sip client. |
| 09:52.36 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:03.44 | anygivenname | my sip client is SipDroid on Android & Xlite on Win PC |
| 10:13.50 | *** join/#asterisk Smirker (Smirker@14-202-69-225.static.tpgi.com.au) |
| 10:13.56 | Smirker | hey |
| 10:15.03 | Smirker | i've got an asterisk box that's working compeltely fine, dialing numbers and making calls etc |
| 10:15.37 | Smirker | makes around 10 calls per minute, all good |
| 10:16.03 | Smirker | i've got issues calling New Zealand, which my SIP provider has traced the issue back to their NZ carrier |
| 10:16.19 | Smirker | however-- they want to know why my Asterisk box is sending two invites |
| 10:17.01 | kaldemar | do you have a SIP trace of that? |
| 10:17.13 | Smirker | yeah |
| 10:17.19 | Smirker | summary is |
| 10:17.48 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:18.48 | Smirker | uploading an image |
| 10:19.50 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:20.26 | Smirker | http://www.extadi.com/sip.jpg |
| 10:20.33 | Smirker | is that normal behaviour? |
| 10:20.48 | Smirker | it seems to work fine for 99.99% of our calls, except to this one country |
| 10:23.08 | Smirker | it's an incoming call to our system, the top part is it inviting my asterisk box to the call |
| 10:23.24 | Smirker | at 8 sec, then at 21 sec i make the single Dial() command to the NZ number |
| 10:23.57 | *** join/#asterisk carloimperia (~carloimpe@host51-212-static.14-79-b.business.telecomitalia.it) |
| 10:24.49 | Smirker | is the second one maybe just establishing two way communication or something? |
| 10:26.07 | carloimperia | Hi I have a issue with asterisk 1.4.22.2 and thomson 2030 ip phone. Every hour I do not because, but the asterisk CLI print continuasly "Unregistered SIP <exten>" |
| 10:26.20 | carloimperia | To Stop I have to restart asterisk or all the phone |
| 10:26.34 | carloimperia | HAve you any ideas ?? |
| 10:26.44 | ectospasm | carloimperia: 1.4.22.2 is *really* old |
| 10:26.53 | ectospasm | try upgrading to 1.4.41 |
| 10:27.18 | ectospasm | note that Asterisk 1.4 is deprecated and will only be receiving security updates going forward. |
| 10:28.24 | carloimperia | Tnx I will try to upgrade |
| 10:31.01 | kaldemar | carloimperia: neither is probably the solution. |
| 10:31.30 | *** join/#asterisk sekil (~sekil@78.24.110.53) |
| 10:31.37 | kaldemar | carloimperia: asterisk says it unregisters something when the peer (in this case the phone) sends a message to unregister. |
| 10:34.55 | carloimperia | Yes kaldermar, I Know. But this message appear repetealy (every second) for every exten |
| 10:40.49 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 10:42.25 | kaldemar | carloimperia: has it always been like that? |
| 10:44.34 | *** part/#asterisk e3eli3h (~e3eli3h@mail.fxcc.com) |
| 10:53.10 | *** join/#asterisk oDesk (~oDesk@188.52.48.3) |
| 10:53.49 | Smirker | kaldemar: i figured it out. it's asterisk reinviting to join the two ends together instead of acting as an intermediatary. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
| 10:54.06 | Smirker | kaldemar: our NZ provider presumable has buggy hardware as mentioned in the artical |
| 10:54.14 | Smirker | article* :p |
| 10:54.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 10:57.20 | *** join/#asterisk Starscreamer (~Starscrea@webmail.mackellar.co.uk) |
| 10:57.30 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 11:01.02 | carloimperia | kaldemar: Yes |
| 11:01.04 | oDesk | there is a strange situation where i was able on 1.6 to REGISTER sip peer using Android Native application , but on 1.8 i can't do so, it doesn't need to register from outside network but using 3rdparty application, i was able to REGISTER the peer |
| 11:01.36 | oDesk | i can generate the logs if requested |
| 11:02.36 | oDesk | it REGISTER on local network |
| 11:03.02 | oDesk | and the 3rdparty app can REGISTER from WAN network but native app can't |
| 11:03.10 | oDesk | only Local REGISTRATION |
| 11:03.14 | oDesk | anyone ? |
| 11:03.51 | Smirker | set sip debug on |
| 11:03.57 | Smirker | check what's happening |
| 11:04.11 | oDesk | Smirker, i set .. Unauthorized 401 |
| 11:04.21 | Smirker | you have ssh access to the box? |
| 11:04.27 | oDesk | Smirker, yes |
| 11:04.50 | Smirker | record log of it working, and then of it not working |
| 11:04.51 | Smirker | compare |
| 11:04.56 | Smirker | maybe the issue will become apparent |
| 11:05.26 | oDesk | Smirker, can i do separate log files for each log ? |
| 11:06.22 | Smirker | i don't see why that matters |
| 11:06.31 | Smirker | just get both SIP logs and compare side-by-side |
| 11:06.50 | oDesk | Smirker, ok |
| 11:20.06 | *** join/#asterisk glam (~glam@113.88.231.72) |
| 11:25.07 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:32.14 | oDesk | Smirker, this line found into the failed try .. Authorization: Digest username=" 1033",realm="asterisk",nonce="1c2df183",uri="sip:dyndns:5060",response="c307d95de49b19e9283edfe98e0b82c9",algorithm=MD5 |
| 11:33.30 | oDesk | Smirker, also Contact: * <- failed side while it has user@ip, transport =udp into the good side |
| 11:34.08 | oDesk | Smirker, so the Authorization line seems the problem |
| 11:34.28 | oDesk | Smirker, how i can ignore this line by Asterisk 1.8 |
| 11:35.09 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 11:47.53 | *** join/#asterisk Amorsen (~Amorsen@94.127.50.7) |
| 11:55.54 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:57.28 | puzzled | hi |
| 11:57.48 | leifmadsen | o/ |
| 11:59.42 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 12:06.51 | MrTelephone | leifmadsen, whats going on buckshot? |
| 12:06.58 | leifmadsen | heh nuttin |
| 12:07.07 | MrTelephone | just woke up? |
| 12:07.30 | MrTelephone | Does asterisk setup at SCTE expo's? |
| 12:09.08 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 12:10.26 | leifmadsen | I've been up for a couple of hours |
| 12:13.19 | Smirker | oDesk: i'd suggest googling around. also check out that extra space in your digest username |
| 12:15.16 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 12:16.53 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 12:17.48 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:17.50 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:18.38 | *** join/#asterisk JerJer (~jj@asterisk/original-h323-guy/JerJer) |
| 12:19.46 | *** join/#asterisk oDesk (~chatzilla@188.52.48.3) |
| 12:20.20 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:20.38 | oDesk | Smirker: i've side by side comparison |
| 12:21.10 | oDesk | the space before the username doesn't exist on the client sw |
| 12:21.15 | oDesk | i can see this |
| 12:21.55 | oDesk | CSeq: 6256 REGISTER <- failed success -> CSeq: 1 REGISTER |
| 12:25.24 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:25.35 | oDesk | the Autorization uri='sip:domain:5060' <- failed success -> uri='sip:domain' |
| 12:25.58 | *** part/#asterisk billmania (~bill@38.98.130.98) |
| 12:27.07 | *** join/#asterisk bintut (~bintut@119.234.0.16) |
| 12:27.53 | bintut | waves |
| 12:34.58 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 12:39.39 | *** join/#asterisk coppice (~chatzilla@m180-219-222-171.smartone-vodafone.com) |
| 12:56.26 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 12:58.09 | *** join/#asterisk BeeBuu (b71b2e55@gateway/web/freenode/ip.183.27.46.85) |
| 12:59.22 | BeeBuu | how can i disable the multi-line of a sip user? |
| 12:59.50 | leifmadsen | eh? |
| 12:59.53 | leifmadsen | you mean on the phone? |
| 13:00.55 | BeeBuu | on sip.conf? |
| 13:02.52 | BeeBuu | i don't need the multiline function,how can i do that? |
| 13:03.10 | Freeaqingme | what do you mean by multiline function? |
| 13:04.51 | BeeBuu | a sip user talking,a income call can ring it |
| 13:04.51 | *** join/#asterisk glam (~glam@119.139.185.73) |
| 13:04.55 | leifmadsen | you can use GROUP() and GROUP_COUNT() in the dialplan to limit the number of calls to a device |
| 13:05.08 | leifmadsen | BeeBuu: that's typically a function of the phone to control how many calls it can receive |
| 13:05.27 | oDesk | waiting ^ |
| 13:05.31 | BeeBuu | how can i limit that? |
| 13:07.18 | leifmadsen | configure the device |
| 13:07.49 | leifmadsen | otherwise use the dialplan functions I mentioned |
| 13:09.06 | *** join/#asterisk dacqueries (~gdecaires@65.48.133.103) |
| 13:09.43 | dacqueries | hi can anyone tell me if having hardware and software echo cancellation enabled at the same time would/could cause kernel panics? |
| 13:10.12 | leifmadsen | it seems silly to have both running, regardless |
| 13:10.20 | dacqueries | yes indeed |
| 13:10.46 | dacqueries | but I've got several pbx set up in different locations and we're having random kernel panics seemingly once a month |
| 13:10.55 | m4xx | i read somewhere one would cancel out the other |
| 13:10.56 | dacqueries | and I'm looking at all of my options |
| 13:11.25 | BeeBuu | @leifmadsen: could you teach me the detail? |
| 13:12.08 | leifmadsen | BeeBuu: sure.... |
| 13:12.10 | leifmadsen | ~thebook |
| 13:12.10 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:12.37 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 13:13.07 | m4xx | when using priority labels, do you do Goto(context,exten,n(label)) or Goto(context,exten,label) ? |
| 13:13.10 | BeeBuu | thanks,i will check that slowly |
| 13:13.45 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 13:13.55 | *** join/#asterisk [Outcast] (~jfjones@westford-nat.juniper.net) |
| 13:14.11 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 13:14.21 | leifmadsen | m4xx: the altter |
| 13:14.24 | leifmadsen | latter* |
| 13:14.42 | m4xx | right on |
| 13:14.45 | leifmadsen | just the label name |
| 13:15.14 | m4xx | i'm not sure if it mentioned that in the book, perhaps i missed it or havent gotten to it yet |
| 13:15.26 | leifmadsen | it should have mentioned it and showed the format |
| 13:18.13 | *** part/#asterisk mcargile (~mikec@office2.vicidial.com) |
| 13:18.57 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 13:19.03 | leifmadsen | m4xx: http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Deeper.html#asterisk-CHP-6-SECT-3 |
| 13:19.56 | BeeBuu | @leifmadsen: is that call multi-line in the book? |
| 13:20.16 | leifmadsen | BeeBuu: no, it was in reference to m4xx as per the start of that line :) |
| 13:20.43 | m4xx | ah, havent gotten there yet |
| 13:21.00 | m4xx | any reason my book says chapter 10 and the link says chapter 6? |
| 13:21.11 | BeeBuu | thanks |
| 13:21.23 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:21.44 | E-bola | Cant you set a device hint via the asterisk console? |
| 13:21.48 | leifmadsen | m4xx: just the way the naming got done internally -- likely that chapter used to be chapter 6 until we added 4 chapters ahead of it ;) |
| 13:21.58 | m4xx | lol |
| 13:21.59 | m4xx | oic |
| 13:23.12 | leifmadsen | it's using the docbook names in the files there, which aren't necessarily matched up |
| 13:30.02 | leifmadsen | BeeBuu: while it is not directly related, some info about GROUP() and GROUP_COUNT() usage is in this chapter: http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html |
| 13:30.47 | BeeBuu | ok,let me check it too |
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| 13:54.00 | zkn | Hello, regarding features.conf.. is it relly not possible to transfer to extensions with more than one digit? |
| 13:54.21 | E-bola | I manupulate a custom hint with the following line: Set(DEVICE_STATE(Custom:KontorStatus)=NOT_INUSE) |
| 13:54.29 | E-bola | But how can i check what the state is? |
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| 14:09.44 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-203.telkomadsl.co.za) |
| 14:09.57 | jkroon | hi guys, i need some help with some CDR stuff. |
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| 14:10.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:10.07 | jkroon | three phones. say A, B, C |
| 14:10.40 | jkroon | A calls B, B puts A on hold and phones C, C then transfers so A talks with C, and B hangs up. |
| 14:11.21 | jkroon | that in my opinion is two calls, A -> B and B -> C. this is fine, and gets logged that way, what bugs me is the call durations. |
| 14:12.19 | jkroon | specifically, the duration of the call that * logs for the A -> B is the duration from B answering to making the transfer, and the B -> C call from the C answering till call termination. |
| 14:12.49 | jkroon | I would like to log the call for A->B as the duration from B answering till termination between A and C. |
| 14:12.55 | jkroon | is this possible - and how? |
| 14:13.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:15.01 | jkroon | ok, i'm getting the distinct impression i'm going to have to mess with Local/foo@bar/n type stuff :( |
| 14:15.23 | morbidwar | hello i have a little problem, i connected my gsm gateway to my asterisk but the asterisk do not get the CID, i connected an analogue phone to the gsmgateway and the phones recive the CID |
| 14:17.07 | jkroon | you need to match the cli settings between asterisk and the premicell. |
| 14:17.51 | morbidwar | the gsmgw doesn't have a menu, it's a analog one |
| 14:24.47 | *** join/#asterisk Strogg (~jean@unaffiliated/strogg) |
| 14:24.52 | Strogg | 'lo 'lo |
| 14:26.45 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:26.49 | Strogg | so.. just had a question which I'm having a hard time finding info for... I have an asterisk server that I setup (about 2 years ago) with some extensions that went to sip phones in our office. We moved to a new location and I want to change the target of my extension to a phone number + extension. I know how to dial a phone number via IAX2, but what's the syntax for dealing with extensions? |
| 14:27.20 | Strogg | ie: my extension on my asterisk box is 412. I want to forward that to 1-234-5678 ext 2805 |
| 14:30.34 | Strogg | hrmmm SendDTMF seems promising |
| 14:31.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:31.10 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:32.52 | Strogg | so.. I'm thinking 1) Dial number 2) Wait 2; 3) SendDTMF(2805). Does that sound reasonable? |
| 14:32.57 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 14:35.10 | pabelanger | Strogg: yes |
| 14:35.32 | Strogg | Awesome. /me tries |
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| 14:44.05 | jkroon | morbidwar, then you need to figure out what the appropriate cadence settings and fsk settings should be to make it work. |
| 14:44.15 | jkroon | there are only so many different possible options, so good luck. |
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| 14:45.29 | jkroon | the settings you're after in chan_dahdi.conf is cidsignalling, cidstart, callerid, usecallerid, and you will probably need to make sure the tonezone is correct in indications.conf and I had to explicitly set a cadence in chan_dahdi.conf. |
| 14:45.43 | *** join/#asterisk andyoutside (6161c2a5@gateway/web/freenode/ip.97.97.194.165) |
| 14:45.48 | morbidwar | thanks jkroon |
| 14:45.56 | andyoutside | Ok for dahdi whatis bpvio?l |
| 14:47.08 | jkroon | can anybody help with my CDR problems above? |
| 14:47.22 | jkroon | also got a bunch of other variants I just picked up now that I need to get sorted out. |
| 14:47.47 | leifmadsen | jkroon: you probably can't do that with CDRs -- that's what CEL was developed for -- more granular logging of call flows through transfers and such |
| 14:47.59 | leifmadsen | CDRs are designed more for just single end-to-end calls, and duration of that call |
| 14:48.01 | zkn | Does anyone know if there is a setting in Asterisk to tweak that will make it recognise DTMF tones pressed on Siemens Gigaset C470 IP phones ? |
| 14:48.42 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 14:49.32 | jkroon | leifmadsen, CEL ? can you point me at _something_ I can look at? |
| 14:49.44 | leifmadsen | look at the book |
| 14:49.50 | leifmadsen | there is information about it |
| 14:50.03 | jkroon | ~book |
| 14:50.03 | infobot | For more information about the Asterisk book, see ~thebook |
| 14:50.08 | jkroon | ~thebook |
| 14:50.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 14:52.07 | *** join/#asterisk g_r_eek (~g_r_eek@173.9.142.122) |
| 14:52.16 | zkn | my problem is that when I try to do a features.conf based callforwarding on a Siemens Gigaset C470 IP then nothing happens, whereas on eyeBeam softphone there are no problems at all.. what could I be overlooking here? |
| 14:52.24 | sunfone | Trying to understand the transport involved when a Linksys PAP2T accepts a "call waiting" call... |
| 14:52.44 | sunfone | I have a customer whose normal one-line conversations work fine... |
| 14:52.54 | sunfone | but when taking a call-waiting call gets no audio |
| 14:53.14 | sunfone | NAT enabled, PAP2T behind a D-Link wifi (Extreme N) |
| 14:53.23 | sunfone | Asterisk on public space |
| 14:53.24 | jkroon | leifmadsen, again, thanks. |
| 14:53.46 | sunfone | I think I get that it would be a second RTP stream |
| 14:54.12 | sunfone | and that it must be being blocked by the D-Link |
| 14:54.29 | jkroon | leifmadsen, is there still a pdf download available somewhere? |
| 14:54.33 | sunfone | But what is it that tricks the D-Link into accepting the *first* RTP stream? |
| 14:55.26 | leifmadsen | jkroon: only for purchase |
| 14:55.26 | leifmadsen | HTML for free |
| 14:55.28 | jkroon | ta. |
| 14:55.42 | jkroon | sunfone, d-link. |
| 14:55.51 | *** join/#asterisk GuillermoHurtado (~Guillermo@200.75.133.65) |
| 14:56.06 | jkroon | you will also experience "random 30 second call cut offs" |
| 14:56.30 | sunfone | jkroon: hmmm. Bummer. Any tricks on the D-Link to make it behave? |
| 14:56.53 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 14:56.54 | GuillermoHurtado | Hello I need some help with FFA of Digium |
| 14:57.06 | jkroon | sunfone, yes. chuck it in a bin. |
| 14:57.41 | GuillermoHurtado | where can I change the fax settings? everytime I do fax show settings I get the same |
| 14:58.15 | sunfone | jkroon: wish it were that simple... residential customer purchased it... may have to get one myself I suppose |
| 14:59.25 | Strogg | hrmmmmm my SendDTMF idea doesn't work.. hrmm when I do dial() it doesn't seem to go on to the next line in the dial plan |
| 14:59.32 | coppice | GuillermoHurtado: what do you want to change? |
| 14:59.42 | jkroon | sunfone, you can't mod the firmware, it's an actual Linux kernel bug, there is NOTHING you can do to fix it. period. |
| 15:00.20 | jkroon | leifmadsen, your write (re CDR records) "In some deployments, these records are used for billing purposes." - the issues mentioned above makes this impossible. |
| 15:00.27 | GuillermoHurtado | coppice I need to disable ecm and change the fax rates....I have tried to do it on res_fax_digium.conf but no luck |
| 15:00.36 | jkroon | sunfone, the other alternative is to get an iax enabled phone. |
| 15:00.55 | leifmadsen | jkroon: CDR is what it is |
| 15:01.08 | leifmadsen | "In some deployments" |
| 15:01.16 | leifmadsen | "some" deployments don't have transfers |
| 15:01.26 | coppice | its a strange world where people feel the need to disable ECM :-\ |
| 15:01.27 | leifmadsen | the statement is still valid |
| 15:01.34 | sunfone | jkroon: the D-Link is based on a Linux kernel? |
| 15:01.54 | jkroon | hehe, yea, if i could somehow prevent blind transfers, or ForkCDR() on blind transfers, and SIP 302 requests, as well I could maybe make it work as is. but still, CEL sounds interesting. |
| 15:02.30 | jkroon | leifmadsen, don't think your statement is invalid, just that based on what I've just seen I wonder how few accurate billing systems are out there. |
| 15:02.40 | jkroon | at least, ones based on * |
| 15:03.31 | GuillermoHurtado | coppice, forget about the ECM, what about the rates? |
| 15:03.47 | jkroon | fortunately the far majority of my clients use SIP/FXS gateways and I've told Dial() to ignore 302s for the moment, but I didn't realize blind transfers and even attended transfers causes similar issues. |
| 15:04.01 | coppice | why would you want to control the rates? |
| 15:05.19 | GuillermoHurtado | I need to set it to only negotiate 9600 because I have a Zhone MALC that only works wellwith that rate |
| 15:07.08 | coppice | people do love to use expensive broken junk, don't they |
| 15:07.24 | GuillermoHurtado | well, what can I say |
| 15:07.52 | *** join/#asterisk AndyML (~alauppe@unaffiliated/andymli7) |
| 15:12.37 | GuillermoHurtado | no help coppice? |
| 15:13.38 | coppice | sorry, I can't really help with not-very-free-fax-for-asterisk |
| 15:17.15 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 15:17.19 | jkroon | GuillermoHurtado, as they say "contact digium support" (personally I try and steer clear) |
| 15:17.44 | jkroon | rtfs and figure it out myself, but that's quite impossible with digium's res_fax. |
| 15:18.15 | coppice | use the genuinely free option. it works better |
| 15:19.09 | jkroon | leifmadsen, is cel part of ast 1.6.2 already? i can't find the mentioned config files and don't see the module files in the modules dir either? |
| 15:19.10 | coppice | people who I've helped move off the commercial FAX offering tell me Digium support consists of "Dunno. Have your money back" |
| 15:19.21 | leifmadsen | jkroon: probably 1.8 |
| 15:19.31 | jkroon | argh crap. |
| 15:19.38 | leifmadsen | "already" implies it would eventually be part of 1.6.2... |
| 15:19.55 | jkroon | still, looks like that might have been a solution. |
| 15:19.59 | leifmadsen | for sure |
| 15:20.09 | leifmadsen | it was developed in trunk after 1.6.2 was branched |
| 15:20.11 | jkroon | leifmadsen, implication not intended |
| 15:20.12 | GuillermoHurtado | well, I am using the free license and they dont offer support with that |
| 15:20.20 | leifmadsen | because it's free ;) |
| 15:20.25 | GuillermoHurtado | yeah |
| 15:20.28 | leifmadsen | support would be a net loss |
| 15:20.30 | jkroon | GuillermoHurtado, just use app_fax |
| 15:20.36 | GuillermoHurtado | I will have to spend $40 to get support |
| 15:20.37 | jkroon | trust me - you will safe yourself MANY headaches. |
| 15:21.02 | jkroon | the ability to read the code makes a HUGE difference. |
| 15:21.36 | coppice | GuillermoHurtado: why don't you want to use the genuinely free option? |
| 15:21.41 | jkroon | well, thanks leifmadsen, i'm off to go bang my head against Local/ /n channels and/or ForkCDR() |
| 15:21.46 | leifmadsen | :) |
| 15:21.47 | leifmadsen | enjoy |
| 15:22.10 | GuillermoHurtado | I am using 1.8 so res_fax, I will stick with that ! |
| 15:22.32 | coppice | but why use the digium fax module? |
| 15:22.43 | leifmadsen | you can use either option in 1.8 |
| 15:22.48 | leifmadsen | they both use app_fax |
| 15:23.00 | leifmadsen | it just uses either res_fax_spandsp or res_fax_digium |
| 15:23.07 | GuillermoHurtado | well I had problems with the spandsp module, so I tried the Digium module....what the heck |
| 15:23.15 | leifmadsen | fax sucks regardless |
| 15:23.23 | seanbright | leifmadsen: +1 |
| 15:23.32 | GuillermoHurtado | leifmadsen +100000 |
| 15:23.36 | coppice | if spandsp gives you trouble, you'll get plenty more with the digium option :-) |
| 15:23.43 | seanbright | damn... 1 upped again |
| 15:23.47 | coppice | what problems did spandsp give you? |
| 15:23.55 | seanbright | or, more specifically 99999 upped |
| 15:24.39 | GuillermoHurtado | with spandsp the box keeps rejecting the t.38 offers with 488 not acceptable |
| 15:25.01 | GuillermoHurtado | I tried everything with same result |
| 15:25.05 | coppice | that has nothing to do with spandsp. that rejection occurs before the FAX module is invoked |
| 15:25.34 | GuillermoHurtado | ummm |
| 15:25.59 | GuillermoHurtado | GuillermoHu: -1 |
| 15:26.19 | GuillermoHurtado | i played with the udptl.conf but no luck |
| 15:26.33 | GuillermoHurtado | any suggestions? |
| 15:29.35 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 15:33.39 | seanbright | anyone need a bug fixed? |
| 15:33.51 | Strogg | So.. I figured out how to dial extensions.. |
| 15:35.03 | Strogg | basically.. you want a line that looks like..... exten => 411,1,Dial(IAX/username:password@host.of.did.com/numbertodial,timeout,D(some numbers)) |
| 15:35.21 | Strogg | the D(2805), for example, dials 2805 after the call is answered |
| 15:35.40 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:37.13 | leifmadsen | yes, and you can add 'w' for wait |
| 15:37.20 | leifmadsen | D(wwww2805) |
| 15:37.25 | leifmadsen | w == 500ms i think |
| 15:37.30 | Strogg | Jah |
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| 15:50.03 | Chau | How may I retrieve the PSTN trunks status from Asterisk AMI? |
| 15:50.44 | mocker | The Audiocodes Mediant 2000 looks awesome. |
| 15:51.46 | \DSAFEW\ | Chau, dahdi show something something? |
| 15:52.39 | \DSAFEW\ | Chau, dahdi show status |
| 15:52.51 | \DSAFEW\ | Chau, which AMI client are you using? |
| 15:53.50 | andyoutside | Ok why would updating asterisk from 1.8.0 to 1.8.3.3 cause it to crash when a internal line calls the main number? |
| 15:54.10 | seanbright | andyoutside: because a bug was introduced between 1.8.0 and 1.8.3.3? |
| 15:54.13 | seanbright | what do i win! |
| 15:54.15 | seanbright | ? |
| 15:54.34 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 15:54.58 | coppice | is it true that scientific linux has solid long term funding from CERN? |
| 15:55.13 | seanbright | andyoutside: what kind of call? SIP? |
| 15:55.26 | Chau | DSAFEW, I am actually using Yeastar's MYPBX that is backed by Asterisk |
| 15:55.29 | \DSAFEW\ | coppice, I am CERN, this is true. |
| 15:55.45 | andyoutside | SIP phone with PRI(Dahdi to the tellphone company) |
| 15:56.07 | seanbright | have you tried latest 1.8? from SVN? |
| 15:56.18 | \DSAFEW\ | Chau, oh, so the upstream PBX probably interfaces with the POTS then, don't know if AMI actually manages upstream PBXs |
| 15:56.18 | Chau | DSAFEW, The MYPBX is voip appliances |
| 15:56.43 | \DSAFEW\ | Chau, is it asterisk based? |
| 15:56.47 | Chau | Yes. it is |
| 15:56.55 | andyoutside | no I was using wget and then tried a yum package for centos |
| 15:57.19 | Chau | DSAFEW, I can use connect to port 5038 and using AMI to communicate with the appliances |
| 15:58.03 | \DSAFEW\ | Chau, yeah, I'm not sure how inter-asterisk AMI works if it does at all |
| 15:58.23 | \DSAFEW\ | Chau, someone else in here might have tried that though |
| 15:58.37 | *** part/#asterisk Strogg (~jean@unaffiliated/strogg) |
| 15:58.51 | andyoutside | is there a diffrnece between wget and the svn? |
| 15:58.51 | Chau | DSAFEW, I can use "Action: SIPPeers" to retrieve a list of sip peers |
| 15:59.00 | andyoutside | also when was the last update |
| 15:59.07 | andyoutside | I did it on monday or so |
| 15:59.36 | Chau | DSAFEW, I amso can use "Action: ExtensionState" to retireve the sip peer status whether it is idle or in-use |
| 16:00.45 | seanbright | andyoutside: svn is the latest code base |
| 16:01.25 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 16:01.25 | Chau | DSAFEW, I may use "Action: DAHDIShowChannels" to get a list of pstn trunks but I can't retrieve the status of the trunk |
| 16:01.28 | \DSAFEW\ | Chau, so are you saying there's one asterisk server running? not two? |
| 16:02.00 | Chau | DSAFEW, yes. only 1 asterisk server is running |
| 16:02.59 | Chau | DSAFEW, there are 10 FXO ports available to the device. I only has 7 phone lines. |
| 16:03.21 | *** join/#asterisk Eitan88 (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 16:03.46 | Chau | DSAFEW, I want to know how to use AMI actions to retrieve the status of the FXO ports. |
| 16:04.13 | *** join/#asterisk pavi (~pavi@121.241.165.245) |
| 16:04.28 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 16:06.08 | pavi | I am using Adhearsion v1.0.3 and I have encountered errors during the time when I try to make outgoing calls |
| 16:06.17 | pavi | heres the error log http://dpaste.com/539522/ |
| 16:08.35 | \DSAFEW\ | Chau, oh, IDK, perhaps the core status has something about dahdi hardware status |
| 16:08.48 | \DSAFEW\ | Chau, AMI doesn't seem to have much for dahdi |
| 16:09.04 | pavi | with same sip and extension file my agi is working properly on local machine...but when I try the same on other machine...first its not showing the agi debug to ahn console..dial method is not working |
| 16:09.32 | Chau | DSAFEW, so is there any other way to retrieve the status? |
| 16:10.06 | pavi | \DSAFEW\, Chau any idea? |
| 16:10.44 | \DSAFEW\ | Chau, I just ran down a list of commands, not from AMI... you can use another program, or logs even |
| 16:11.08 | Chau | what ways? |
| 16:11.20 | Chau | DSAFEW, what program or logs you means? |
| 16:11.25 | *** join/#asterisk cj (~cjac@adsl-207-32-169-17.rockisland.net) |
| 16:11.37 | cj | I need to move my IRC client to a more stable location, obviously. |
| 16:11.59 | pavi | cj, hi can you help me out ? |
| 16:12.08 | cj | pavi: probably not, but I'll sure try! |
| 16:12.27 | andyoutside | seanbright: ok what is the best path/branch to get |
| 16:12.36 | \DSAFEW\ | Chau, well, with the console command dahdi show status, or with dahdi_tool |
| 16:12.54 | \DSAFEW\ | I'm sure those alarms are displayed in the asterisk logs |
| 16:12.59 | seanbright | svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk-1.8 |
| 16:13.40 | Chau | DSAFEW, you means i go asterisk log to retrieve the actual status? |
| 16:13.45 | Chau | DSAFEW, I need to check that |
| 16:14.27 | \DSAFEW\ | pavi, oh, something might be conflicting, are there extra things added in the .ael files or something? |
| 16:15.17 | pavi | cj, i am using adherasion and try to make outgoing call..when some one calls to did. everything ig working fine on my local machine, but on other machine ..i am unable to see debug agi console.. |
| 16:15.19 | andyoutside | ty |
| 16:15.28 | andyoutside | I was not sure if that or the trunk was better |
| 16:15.33 | m4xx | what's the proper format for setting an outbound caller id for originate? 5555551212? |
| 16:15.53 | Chau | DSAFEW, dahdi show status doesn't show the status |
| 16:16.40 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 16:18.34 | \DSAFEW\ | Chau, what? should say if it's ok |
| 16:19.01 | \DSAFEW\ | Chau, "alarms" |
| 16:19.12 | Chau | DSAFEW, what is "alarms"? |
| 16:20.34 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 16:22.05 | Chau | DSAFEW, "alarms" column show 'OK' for 10 FXO ports. However, I only has 7 phone lines plugin. |
| 16:22.33 | Chau | DSAFEW, how may i differentiate the status of another 3 ports? |
| 16:23.54 | *** join/#asterisk afrance (~afrance@65.117.122.45) |
| 16:25.34 | *** join/#asterisk ajeet (~ajeet@121.241.165.245) |
| 16:25.55 | afrance | Good afternoon! So I ran "yum update asterisk16" on a asterisk now install to 1) get it updated and 2) make cdr work, well those objectives were accomplished but now my calls are one-sided, are there some known changes that I can't seem to find? |
| 16:25.58 | ajeet | cj, this is ajeet previously typing as pavi |
| 16:26.54 | \DSAFEW\ | Chau, http://ofps.oreilly.com/titles/9780596517342/asterisk-DeviceStates.html#DeviceStates_id267818 |
| 16:27.09 | \DSAFEW\ | Chau, would the getvar and state help you? |
| 16:28.45 | cj | ajeet: ok. |
| 16:28.46 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:29.11 | cj | pavi: I don't know anything about adherasion, sorry. |
| 16:29.50 | m4xx | i was using '"My Name" <5555551212>' but for some reason this is no longer working, i'm wondering if i may have changed something or if it's my provider. |
| 16:30.27 | aidinb | you can't just PASS cnam |
| 16:30.28 | ajeet | cj, know anyone who is into adherasion here ? |
| 16:30.48 | aidinb | has to be set on the provider-side via a nation-wide database |
| 16:31.10 | m4xx | then i just want to set my number? 5555551212? |
| 16:31.18 | cj | nope |
| 16:31.31 | m4xx | + country code? |
| 16:31.34 | Chau | DSAFEW, I will try that first. |
| 16:31.44 | aidinb | pass it however they want it |
| 16:31.52 | Chau | DSAFEW, Is that anyway to contact you? I will leave this channel now |
| 16:32.15 | aidinb | e.g. flowroute takes +1xxxxxxxxxx or xxxxxxxxxx or 1xxxxxxxxxx |
| 16:33.01 | aidinb | or anything at all really |
| 16:33.53 | afrance | So no known issues with the latest 1.6 upgrade breaking sip trunks? Everything worked perfectly except for CDR before the upgrade now its as if nothing is being sent from my local pbx to the phone companies pbx. (SIP trunk to metaswitch) |
| 16:33.57 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 16:43.49 | afrance | hmmm... odd, problem has gone away. Thanks anyway. |
| 16:44.34 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-203.telkomadsl.co.za) |
| 16:52.41 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 16:52.55 | afrance | Whats the correct way of providing a ring tone to incoming callers? When people call in they are simply given silence. I've tried changing the MoH but that obviously doesn't work until the call has been picked up and placed on hold. |
| 16:52.57 | afrance | I've all attempted to add 'r' in the general options. Any other ideas? Currently the only way I've made it work is by creating a custom MoH and used it as the follow-me moh and enabled follow me. |
| 16:53.57 | paulc | afrance: start your dialplan with Ringing - it might help |
| 16:54.01 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 16:54.03 | paulc | depends on your connectivity + provider too |
| 16:55.17 | afrance | Paul: the connectivity and the provider is why i've determined there is no ring, they hand off a sip connection so the call is outside of their "control" within ms. (From what I've read about this issue anyway) |
| 16:56.48 | afrance | paul: are you familier with freepbx? I'm pretty sure I've tried that as well with no luck. |
| 16:57.18 | afrance | (the reason I ask is I can describe that option and see if we're talking about the same thing) |
| 16:58.32 | *** join/#asterisk Micc (Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 16:58.48 | Micc | is meetme known to hang asterisk on a busy server with no timing card? |
| 16:58.53 | afrance | On the inbound routes there is a "Signal RINGING" option. It says it sends rnging before answer but that seems more like an internal option to send to the destination |
| 16:59.28 | leifmadsen | Micc: what is a timing card? |
| 17:00.36 | Micc | leif, like a pri card. DAHDI. or even a DAHDI dummy. I didn't build chan_dahdi |
| 17:00.48 | leifmadsen | then what timing module are you using? |
| 17:01.43 | Micc | I guess none, but it seems to be working, but we had a hang this morning I forgot to do show locks, but I suspect the only thing thats changed is the meetme room |
| 17:01.44 | leifmadsen | if you didn't build DAHDI, then you're using a different timing module, and I know they can be less stable than res_timing_dahdi, so I suggest you install DAHDI and use that for timing |
| 17:01.55 | leifmadsen | you most certainly are using a timing module if you're using meetme() |
| 17:01.57 | leifmadsen | it is required |
| 17:02.01 | leifmadsen | it won't compile without it |
| 17:02.25 | leifmadsen | you may be using res_timing_pthread, in which case you should use a different timing module |
| 17:02.50 | Micc | oh, it probably is using res_timing_dahdi then, I just didn't build chan_dahdi, so it gives an error every time it starts saying no DAHDI device or something. |
| 17:03.13 | leifmadsen | I don't know, and you don't know... so someone should check that |
| 17:03.16 | leifmadsen | anyways it is not a known issue |
| 17:03.23 | leifmadsen | I suspect you're using a different timing module than you think you are |
| 17:06.45 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 17:07.49 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 17:07.56 | paulc | afrance: Freepbx is a different beast - #freepbx might be able to help you better. If you're able to edit your extensions.conf and try adding a "Ringing" in the right place, see if it helps |
| 17:09.07 | Kobaz | hmmmmm |
| 17:09.19 | Kobaz | i dont see any options to force a from ip in iax |
| 17:09.45 | Kobaz | asterisk is using the wrong source ip when i'm trying to connect to a local lan hostname |
| 17:13.03 | thehar | Sony offers identity theft protection, little news on PSN relaunch |
| 17:13.06 | thehar | whoops |
| 17:13.07 | thehar | MISTYPE |
| 17:13.08 | thehar | hehe |
| 17:13.13 | Micc | leifmadsen, I'm building res_timing_dahdi and res_timing_pthread, how do I know which one its actually using? |
| 17:13.52 | leifmadsen | timing testing |
| 17:13.57 | leifmadsen | errr |
| 17:14.00 | leifmadsen | *CLI> timing test |
| 17:14.07 | leifmadsen | Using the 'DAHDI' timing module for this test. |
| 17:14.52 | Micc | it says its using DAHDI. |
| 17:15.12 | Micc | It has been 1018 milliseconds, and we got 51 timer ticks |
| 17:15.46 | Micc | on another server it uses pthread and got 50 ticks in 1001ms |
| 17:16.19 | *** join/#asterisk StefanT (~stefan@92.63.214.181) |
| 17:16.29 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 17:16.38 | afrance | Thanks Paul, I'm looking into how to translate what I need to freepbx. I just needed that hint on where to start looking. |
| 17:16.39 | m4xx | is it me or does paquiao look like a cross between mike tyson and bruce lee |
| 17:16.56 | Micc | it doesn't hurt to run chan_dahdi on a server without a card, right? |
| 17:17.12 | leifmadsen | no |
| 17:17.14 | Micc | I had it disabled for some reason. |
| 17:21.32 | Micc | I remember back in the early days it was preferable to have a hardware timing source even if your only doing sip calls. |
| 17:21.35 | Micc | is that still the case? |
| 17:23.23 | leifmadsen | nope |
| 17:23.32 | leifmadsen | timing is only used for MeetMe() and IAX2 trunking |
| 17:23.48 | Micc | ok |
| 17:23.48 | leifmadsen | the kernel does the timing now, so hardware is not required |
| 17:29.37 | Micc | when I do ps I see asterisk is taking up 90% cpu, but there are 4 cores. load average is .9. does that mean I'm almost using up 100% of one cpu? |
| 17:30.06 | bdfoster | yes |
| 17:30.23 | Qwell | no |
| 17:30.24 | Micc | but isn't asterisk spreading the load on multiple cpus? |
| 17:30.43 | Micc | so the answer is yes/no. |
| 17:30.44 | Qwell | Load has noting to do with CPU usage. |
| 17:31.08 | Qwell | http://blog.scoutapp.com/articles/2009/07/31/understanding-load-averages |
| 17:31.26 | leifmadsen | is that 90% of all capacity, or 90% of 1 CPU? (< 25% capacity) |
| 17:32.09 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:32.41 | Qwell | Yes, it is < 25% |
| 17:32.47 | Micc | according the the understanding-load-averages, it is based on how many cpus there are, but it doesn't talk about how ps/top shows cpu usage. |
| 17:32.49 | leifmadsen | that's what I figured |
| 17:32.59 | Qwell | Micc: press 1, in top |
| 17:33.10 | Qwell | (the number 1) |
| 17:33.41 | Micc | ah! :) thanks |
| 17:34.14 | Micc | I'm such a noob sometimes even though I've been working with linux for almost 20 years. |
| 17:34.36 | Micc | ant AT&T system V before that. |
| 17:35.23 | Micc | I guess when I was learning all that stuff 20 years ago, there was no need for this kind of thing because there was not many dual cores/dual procs. |
| 17:35.38 | *** join/#asterisk twanny796 (~twanny@46.11.88.154) |
| 17:36.10 | Micc | I read all the man pages front to back in those days. I hope I never have to do that again. |
| 17:37.20 | Micc | So it doesn't seem like my hang/lock was due to a high load. So I've got to find something else that fits now. |
| 17:37.37 | Micc | I wish I had done a core show locks before restarting it. |
| 17:43.53 | *** join/#asterisk jkroon (~jkroon@197.170.2.171) |
| 17:46.28 | *** join/#asterisk jstapleton_ (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 17:46.33 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:50.32 | *** join/#asterisk \DSAFEW\ (~DSAFEW_@ip72-208-176-219.ph.ph.cox.net) |
| 17:54.12 | Micc | Ah! maybe I should have just checked the new asterisk version. |
| 17:54.14 | Micc | Resolve deadlocks related to device states in chan_sip |
| 18:01.10 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 18:01.45 | Jcook_5xData | anyone ever set up blf and presence for a polycom 601? |
| 18:08.41 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 18:11.50 | *** join/#asterisk imox1234 (~imox1234@p4FC5C74D.dip0.t-ipconnect.de) |
| 18:13.08 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 18:15.16 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-203.telkomadsl.co.za) |
| 18:17.41 | *** join/#asterisk zkn (~zkn@82.131.28.206.cable.starman.ee) |
| 18:24.53 | *** join/#asterisk michael-i (~michael@204.11.230.58.static.etheric.net) |
| 18:30.42 | jaytee | yay!!! the print edition of the book arrived! |
| 18:30.58 | russellb | :-D |
| 18:31.28 | thehar | wooooo |
| 18:31.29 | thehar | Asterisk |
| 18:31.48 | russellb | jaytee: once you read it, a review on amazon.com would be much appreciated. :-) |
| 18:32.33 | jaytee | russellb, sure but which one do I review? my print edition or my Kindle version :-) |
| 18:32.43 | russellb | both? heh |
| 18:32.49 | thehar | flip a coin. |
| 18:34.09 | jaytee | "This book is a must buy for any serious professional using Asterisk. I'd also recommend purchasing the Asterisk Cookbook as a companion book to this edition. Don't buy the Kindle version because page 9 is totally lame!" < jk, I'd leave out the last sentence. |
| 18:39.24 | jaytee | I've had the Kindle edition since last Saturday and I've already picked up on some new features in 1.6.2 and 1.8.x that I'd missed on the wiki. Like the larger page size of the print edition but the Kindle is great for traveling cuz it's light as a feather. |
| 18:40.10 | *** join/#asterisk satonio (~satonio@168.Red-213-96-115.staticIP.rima-tde.net) |
| 18:40.11 | satonio | hi |
| 18:40.20 | satonio | how can i see the incoming calls? |
| 18:40.20 | jaytee | o/ |
| 18:40.32 | michael-i | I have a minimal 1.8.3.2 system setup and was trying to add app_originate. The module is loaded but I still do not have a "channel originate ..." command. There isn't anything useful being reported in the logs. Any conf file required that I'm missing? |
| 18:40.57 | satonio | i have access to the machine with asterisk through ssh |
| 18:41.14 | jaytee | satonio, are you looking at the CLI? |
| 18:41.18 | leifmadsen | michael-i: that is done with res_clioriginate |
| 18:41.31 | leifmadsen | not app_originate which is the Originate() dialplan applicatoin |
| 18:41.53 | michael-i | leifmadsen: argh... :) that's what I get for celebrating cinco de mayo |
| 18:42.05 | satonio | jaytee, whatever, i just want a list of incoming calls in the last minutes or the active ones |
| 18:42.10 | jaytee | ocho de mustard is much better |
| 18:42.11 | michael-i | thanks! I knew it was something really stupid |
| 18:42.13 | satonio | i don't care if it's something in the web |
| 18:42.17 | satonio | or with cli |
| 18:42.49 | satonio | i have no idea about asterisk |
| 18:42.55 | satonio | can you help me? |
| 18:43.13 | itsbroken | Hello, I'm getting alot of hangupcause=0's on my incoming calls.. which looks like not defined. What is that supposed to mean? |
| 18:43.37 | leifmadsen | satonio: core set verbose 5 |
| 18:43.43 | leifmadsen | and now... go read some documentation |
| 18:43.45 | leifmadsen | ~thebook |
| 18:43.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 18:43.47 | jaytee | satonio, is your ssh session at a linux prompt or in the Asterisk CLI? |
| 18:44.00 | michael-i | satonio: a start would be "core show channels" |
| 18:44.00 | bdfoster | satonio, you wont help with that attitude |
| 18:44.12 | satonio | linux prompt |
| 18:44.18 | itsbroken | >.< |
| 18:44.23 | jaytee | type asterisk -rvvvvvvvvvvvv |
| 18:44.28 | satonio | bdfoster, it's not that i don't want to learn, it's just i need it fast |
| 18:44.30 | *** join/#asterisk neutr0n777 (~neutr0n77@173.228.121.4) |
| 18:44.37 | itsbroken | that's when you pay someone satonio |
| 18:44.43 | carrar | in 30 seconds or less |
| 18:44.44 | bdfoster | yup |
| 18:45.15 | jaytee | I should advertise....."Will work for medicinals" |
| 18:45.15 | carrar | http://www.osburn.com/asterisk-sign.png |
| 18:45.16 | leifmadsen | Fast, Cheap, and Good -- you can only have 2. |
| 18:45.34 | itsbroken | lol |
| 18:46.06 | jaytee | carrar, :-) hahahaha, almost forgot about that one. |
| 18:46.07 | carrar | thats back from 2008 |
| 18:46.13 | bdfoster | quantity, quality, reliablility |
| 18:46.14 | carrar | it's been that long ago |
| 18:46.26 | bdfoster | pick two |
| 18:46.33 | bdfoster | (another version) |
| 18:46.44 | jaytee | remember the guy with the voice modems that wanted to solder them all together to make an fxo channel bank? |
| 18:46.54 | carrar | oh yeah |
| 18:46.55 | bdfoster | heh... |
| 18:46.59 | carrar | ~soundcard |
| 18:46.59 | infobot | extra, extra, read all about it, audio is usually a codec issue. start with trying to set 'disallow=all' and 'allow=alaw' in sip.conf or the channel's config file if not using sip |
| 18:47.07 | carrar | or something like that |
| 18:47.44 | carrar | jaytee, he probably made one and is now rich!! :) |
| 18:48.08 | jaytee | hehe |
| 18:48.17 | jaytee | brb |
| 18:48.36 | coppice | jaytee: I wonder if he made this bank too big to fail? |
| 18:51.26 | bdfoster | heeeeeeeeeeeeeeeehehe |
| 18:51.42 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 18:53.35 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 18:53.56 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.244) |
| 18:54.03 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:54.21 | b0ot | Anyone know how to download the configuration file for a polycom phone? |
| 18:54.34 | b0ot | I have a ftp server setup that my sip phone can get the config from |
| 18:54.42 | bdfoster | satonio, I hope you arent being paid to administer it |
| 18:54.47 | b0ot | but right now I want to start with the current config from what I started with the web config |
| 18:54.50 | b0ot | and just tweak it |
| 18:55.15 | carrar | b0ot, you can get the files and instructions from polycom's web page |
| 18:55.26 | b0ot | carrar, I have been on the site for hours |
| 18:55.32 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 18:55.32 | b0ot | and haven't found anything |
| 18:55.44 | jaytee | b0ot, gimee a sec |
| 18:55.50 | b0ot | thanks |
| 18:55.50 | carrar | click on support |
| 18:56.01 | b0ot | carrar, http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip335.html |
| 18:56.07 | b0ot | I got to this but haven't found anything helpful |
| 18:56.33 | carrar | download the firmware you want to use |
| 18:56.45 | carrar | botorom/split file |
| 18:56.48 | carrar | and admin guide |
| 18:56.54 | carrar | start with the admin guide |
| 18:57.03 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 18:57.31 | carrar | note that you will need to read it |
| 18:58.03 | b0ot | carrar, I have been through that, and thats how i figured out hwo to setup the ftp server... but it doesn't tell you how to get the files actually off the phone |
| 18:58.05 | carrar | but you are at the correct page |
| 18:58.06 | b0ot | just how to put them on |
| 18:58.24 | carrar | What files "off the phone" |
| 18:58.38 | b0ot | all of the .cfg files |
| 18:58.39 | carrar | Phone will write data to your FTP server |
| 18:58.48 | b0ot | it wrote two log files |
| 18:58.49 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 18:59.04 | carrar | config files are on YOUR server |
| 18:59.05 | b0ot | but i want the actual .cfg files that are modified when I setup the phone on the web interface... |
| 18:59.17 | b0ot | nope... just log files |
| 18:59.34 | carrar | web interface is horried |
| 18:59.39 | carrar | never use that |
| 18:59.50 | b0ot | MAC-boot.log and MAC-app.log |
| 19:00.06 | jaytee | b0ot, http://supportdocs.polycom.com/PolycomService/support/global/documents/support/technical/products/voice/white_paper_configuration_file_management_on_soundpoint_ip_phones.pdf |
| 19:01.56 | jaytee | quite often trying to provision a Polycom phone using FTP AFTER setting it up with the web interface will cause the config to error out. |
| 19:02.40 | jaytee | or it won't overwrite any changes you set in the configs that were set in the web interface |
| 19:02.41 | b0ot | hmm |
| 19:04.03 | jaytee | best bet is to setup the config files using that whitepaper and the Asterisk book as a guide and make sure you have the DHCP server configuration passing the address of the boot server. |
| 19:05.20 | b0ot | well I have everything working |
| 19:05.37 | b0ot | the only issue is that the stupid phone doesn't do dtmf correct with my |
| 19:05.39 | b0ot | aa |
| 19:05.46 | b0ot | all the other phones do |
| 19:06.10 | jaytee | aa? auto-attendant? |
| 19:06.14 | carrar | good reason to use config files on a server |
| 19:06.16 | b0ot | yeah |
| 19:06.23 | jaytee | are the other phones polycom too? |
| 19:06.55 | b0ot | no |
| 19:07.57 | jaytee | Polycoms by default use RFC2833 for dtmfmode. Might want to check what you have in your sip.conf for the sip account that device is using. |
| 19:13.14 | b0ot | hmm |
| 19:13.16 | b0ot | alright |
| 19:14.19 | b0ot | I really thought that might be it... but it should be using rfc2833 |
| 19:17.15 | ruben23 | hi anyone have success on compiling asterisk with h.323..im getting error on doing it |
| 19:17.55 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:18.21 | jaytee | I don't use h.323 |
| 19:18.31 | Jcook_5xData | anyone ever set up blf and presence for a polycom 601? |
| 19:19.40 | b0ot | jaytee, I thought I read some places that polycom does some weird thing |
| 19:20.00 | carrar | polycom is weird! |
| 19:20.03 | b0ot | sometimes with AA |
| 19:20.45 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 19:21.03 | jaytee | Jcook_5xData, http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup#Setting_up_the_Polycom_Soundpoint_Phones_for_Presence_and_BLF |
| 19:21.33 | carrar | jaytee... google Ninja! |
| 19:22.03 | Jcook_5xData | jaytee, thanks I will give it a look. most guides I found have not worked yet |
| 19:22.06 | jaytee | b0ot, I have several Soundpoint 320 and 321 phones and haven't yet had a problem with DTMF on any of my IVR setups |
| 19:22.15 | b0ot | :( |
| 19:22.18 | b0ot | that's not good |
| 19:22.24 | jaytee | carrar, I bookmark everything for future reference |
| 19:24.39 | jaytee | b0ot, check the dtmf setting on the phone and check to make sure it matches what is in that device's account settings in sip.conf. |
| 19:24.46 | carrar | BOOKMARK THIS: http://www.hulu.com/watch/99944/saturday-night-live-snl-digital-short-on-the-ground |
| 19:24.56 | carrar | doh they moved it |
| 19:26.36 | carrar | guess this will have to do: http://www.youtube.com/watch?v=unQz4kFJ2Tg |
| 19:28.12 | jaytee | I'm not sure what hot babes in lederhosen have to do with Asterisk but I'm liking it! |
| 19:28.23 | carrar | heh |
| 19:28.32 | carrar | The girls in that video are also from there |
| 19:28.52 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-pxauucrbvuppeoex) |
| 19:29.24 | carrar | Washington (state) has many wonderful places :) |
| 19:39.58 | jaytee | I loved going to the Puyallup fair |
| 19:42.00 | *** join/#asterisk oDesk (~chatzilla@188.52.48.3) |
| 19:44.14 | *** join/#asterisk variable_office (~variable_@10.61.pool.border1.ch1.iswan.net) |
| 19:44.47 | oDesk | i've moved from 1.6.2 to 1.8.3 since then all my WAN clients went crazy, all config for sip peers are the same as were before, but only chaning to 1.8.3 caused that |
| 19:45.03 | variable_office | after my client sends an authorized invite, asterisk still responds with a 401 and I am trying to diagnose why this is. Is there a way to view the authorization process in detail from the asterisk cli so I can figure this out? |
| 19:45.34 | oDesk | variable_office: me too after upgrading to 1.8.3 |
| 19:46.43 | variable_office | its not as easy as versioning for me unfortunately |
| 19:51.37 | *** join/#asterisk _justdave (~dave@unaffiliated/justdave) |
| 19:57.09 | *** join/#asterisk oDesk (~egi@188.52.48.3) |
| 19:58.05 | oDesk | here is my pastiepin http://pastiebin.com/?page=p&id=4dc4002312b97 |
| 20:01.43 | oDesk | anyone plz ? |
| 20:11.20 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:16.21 | citywok | oDesk: post your configs |
| 20:28.19 | Jcook_5xData | jaytee, quick ? in the link you sent they talk about this 'reg.1.lineKeys="1"' where is that set at I check sip.cfg & phone1.cfg it not in ether |
| 20:30.58 | Jcook_5xData | jaytee, nm I found it |
| 20:34.29 | jaytee | Jcook_5xData, it's in the phone1.cfg but I don't use that particular file. I use a file like 0004f217135b-XXX.cfg where XXX is a 3 digit extension for the phone. in the master config file (mac-address-of-phone.cfg) there is a line called CONFIG_FILES= |
| 20:34.50 | jaytee | mine looks like this: CONFIG_FILES="0004f2434163-202.cfg, local-settings.cfg, phone1.cfg, sip.cfg" |
| 20:35.13 | Jcook_5xData | still not work if I show Hints I have 0 watchers - and sip show subscriptions list show nothing |
| 20:35.50 | Jcook_5xData | I have had work before that was many years ago I cant remember it being this hard |
| 20:35.51 | jaytee | any settings in the first file will take precedent over duplicate settings in files that come after in order from left to right. |
| 20:36.41 | jaytee | I put custom settings in the macaddress-xxx.cfg file and any overrides to sip.cfg in the local-settings.cfg file as Polycom recommends to never edit the sip.cfg file |
| 20:38.52 | Jcook_5xData | I am bad then I do, althou I can backups of my sip.cfg |
| 20:39.32 | Jcook_5xData | is there any place I can pay to get to work? |
| 20:40.39 | Jcook_5xData | I have to be missing some thing I like 3 setting |
| 20:42.14 | Jcook_5xData | Oh well I guess I work on it over the weekend. That for your help :) |
| 20:43.41 | *** join/#asterisk Buttons840 (~buttons@c-76-27-51-35.hsd1.ut.comcast.net) |
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| 21:03.23 | *** join/#asterisk aarcane (~aarcane@c-98-238-138-177.hsd1.ca.comcast.net) |
| 21:04.26 | aarcane | Hi, I've been thinking about using asterisk for a while, and am revisiting the idea. the thing I'm most interrested in knowing to put me on the right path is, How do I connect my home phones to asterisk, or where can I obtain inexpensive PBX phones to use instead, including wireless PBX phones. |
| 21:04.32 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:19.38 | paulc | aarcane: For your home phones, a lot will come down to how they're wired today. If it's daisy-chained (highly probable), all phones look like one "extension" to Asterisk. If you have "star configuration" wiring, it becomes easier. |
| 21:20.03 | paulc | aarcane: That's assuming you want to use your existing phones (corded or wireless). The alternative is to go with SIP phones all over the place. But that requires the right cabling etc. |
| 21:20.56 | sxpert | over here, elecrical code imposes network cables instead of phone cable now |
| 21:21.11 | sxpert | and it has to be setup in a star topology |
| 21:22.11 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:22.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:23.10 | paulc | sxpert: really? That's cool.. where's "here"? |
| 21:24.02 | sxpert | france |
| 21:27.57 | paulc | Progressive :) I got all excited a while back about being able to flood wire a house that had been framed but not built out.. deal fell through, but it was great to read up and plan stuff.. |
| 21:32.50 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
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| 21:35.02 | anygivenname | how can I have a GUI for Asterisk ? |
| 21:36.52 | seraphie | anygivenname: https://wiki.asterisk.org/wiki/display/AST/Asterisk+GUI |
| 21:37.17 | lpmusic | I recently upgraded asterisk (a minor version update) to 1.6.2.17.2 and I'm seeing an issue where DTMF digits aren't always recognized, even within the system and using out-of-band signally, http://pastie.org/1873204 , I also see this when I call inbound over multiple SIP origination services, so it's not just the SCCP driver, has anyone seen something like this? |
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| 21:42.12 | anygivenname | seaphie: the link does not lead to a gui of asterisk |
| 21:43.18 | sxpert | anygivenname: http://www.trixbox.com/ |
| 21:43.19 | seraphie | Look at the Download section. |
| 21:43.51 | seraphie | Also http://asterisknow.org |
| 21:44.09 | leifmadsen | anygivenname: sure it does... did you look at the Download section? |
| 21:46.50 | carrar | Where is the click go enable pbx icon!!! |
| 21:47.06 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 21:47.36 | anygivenname | download section is in: http://svn.asterisk.org/svn/asterisk-gui/branches/2.0/ |
| 21:47.56 | WiretapSeven | anygivenname, freepbx also works |
| 21:48.03 | carrar | no it doesn't |
| 21:48.19 | carrar | :) |
| 21:48.25 | seraphie | anygivenname: yes, that is the subversion repository and that is currently the best way to get the Asterisk GUI. |
| 21:48.27 | WiretapSeven | carrar, your reasoning? |
| 21:48.36 | seraphie | though we will be releasing packages soon. |
| 21:49.34 | carrar | I don't need a reason! I live in America!! :) |
| 21:50.56 | WiretapSeven | I see, you were just trolling |
| 21:51.10 | carrar | yes |
| 21:51.22 | carrar | for the last 10 years |
| 21:51.36 | carrar | or however long I've been here |
| 21:51.52 | leifmadsen | seraphie: hopefully Monday |
| 21:52.18 | leifmadsen | anygivenname: bye!!! |
| 21:52.21 | *** join/#asterisk anygivenname (~anygivenn@41.32.28.166) |
| 21:52.30 | carrar | anygivenname, try it without a gui |
| 21:52.39 | carrar | You'll like it! |
| 21:53.00 | anygivenname | carrar: extremely hard to configure it through .conf files |
| 21:53.16 | anygivenname | but the gui turned out to be a distro |
| 21:53.30 | anygivenname | I thought it was only an app interface |
| 21:53.31 | carrar | what part is extremely hard? |
| 21:53.53 | carrar | I find it rather extremely easy |
| 21:54.16 | anygivenname | all of it is hard for me...I've been for 2 weeks trying to make it work but never got it working |
| 21:54.33 | carrar | Did you read the book? |
| 21:54.39 | anygivenname | yes |
| 21:54.58 | carrar | hrmm |
| 21:54.59 | anygivenname | I'm not an expert like you guys here are |
| 21:55.07 | citywok | doesn't the book have a beginners tutorial? |
| 21:55.09 | anygivenname | I'm just a basic user |
| 21:55.10 | carrar | takes time |
| 21:55.25 | citywok | everybody here started out as a beginner at some point :P |
| 21:55.36 | anygivenname | I think gui will be better for me if I ever find it |
| 21:56.09 | carrar | You might like switchvox then |
| 21:56.40 | WiretapSeven | <-- not an expert |
| 21:56.52 | anygivenname | does it have a gui ? |
| 21:56.55 | carrar | yes |
| 21:56.55 | citywok | anygivenname: did you want a full distro of asterisk+gui, or did you want to compile * yourself and then install a gui? |
| 21:56.59 | michael-i | try askozia.com 's live cd to get a phone ringing in a few minutes |
| 21:57.44 | anygivenname | citywork: I dont need a distro of asterisk....i'm running ubuntu & want a gui of asterisk |
| 21:58.06 | carrar | perhaps go for a walk |
| 21:58.09 | carrar | clear the mind |
| 21:58.18 | carrar | whistle at some girls |
| 21:58.23 | citywok | anygivenname: it's more difficult to install and configure a GUI on your distro than it is to just use a full on * distro |
| 21:58.25 | carrar | then come back |
| 21:58.52 | citywok | anygivenname: if you want to do it on your ubuntu install i'd recommend simply learning the config structure. to get basic sip calls working you only need to modify sip.conf and extensions.conf |
| 21:59.45 | anygivenname | citywork: I have modified them according to my readings....but failed to let it work |
| 22:00.02 | citywok | anygivenname: if you are set on the GUI route, the asterisk gui as referenced above, or freepbx are both good GUI's, although i don't have an idea how to install the asterisk gui, i've only installed the freepbx gui on ad ebian box |
| 22:00.40 | citywok | anygivenname: if you post your configs and the errors you are getting there are quite a few people in here (myself included) who would be happy to help you resolve any issues. |
| 22:00.54 | citywok | ~pb |
| 22:00.54 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 22:01.25 | anygivenname | citywork: ok....will paste them |
| 22:09.43 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:15.10 | aarcane | paulc, far as I know, all the lines in the house go back to two lug-nuts at the phone box outside. there's another two lug-nuts that have wires from the telco, but are unused anywhere inside. |
| 22:16.14 | citywok | aarcane: that means your jacks are most likely all connected together on one continuous run through the house |
| 22:16.41 | citywok | it's easy to tell, pull a plate off and look to see if the phone wire is just run through the plate, coming in one direction and going out the other. |
| 22:17.21 | aarcane | either way, I just one to ring all the phones in the house, for now. nothing so fancy as separate lines for separate rooms or anything. |
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| 22:18.56 | citywok | aarcane: on a continuous loop if you connect multiple phones they should all ring |
| 22:18.59 | paulc | aarcane: easily doable then.. if you've got 2 pairs of wires throughout you can locate your FXO/FXS hardware anywhere.. otherwise it'll need to be near that demarc point |
| 22:19.20 | paulc | aarcane: Are you going to use an ATA with FXO/FXS interfaces? or get FXO/FXS card for your PC? |
| 22:19.28 | aarcane | citywok, paulc I think we're on the wrong level.. |
| 22:19.40 | aarcane | how do I physically connect my computer to my phone lines ? |
| 22:19.47 | paulc | You need hardware |
| 22:20.04 | aarcane | I'm guessing that it has to do with that fx0 and fxs stuff, but I can't find a list of approved hardware on the website, or anything similar. |
| 22:20.12 | paulc | either an ATA with FXO and FXS interfaces (FXO <--> Telco, and FXS <--> your house phone wiring) |
| 22:20.24 | paulc | Something like a SPA-3102 or an HT503 |
| 22:20.37 | paulc | OR.. (more expensive).. buy FXO/FXS card/modules and put in your PC |
| 22:21.19 | aarcane | which method produces better sound quality/is more reliable ? |
| 22:21.39 | WiretapSeven | aarcane, how long is a piece of string |
| 22:22.00 | aarcane | WiretapSeven, the measure of the distance between the endpoints. |
| 22:22.13 | paulc | oh touche! |
| 22:22.16 | WiretapSeven | grab yourself an SPA3102 or SPA3000 (older model), cut the line into the house, connect the FXO port to the telco side, and the FXS port to the house line side |
| 22:24.04 | paulc | aarcane: You can start off with the FXO port plugged in where you currently have a phone plugged in.. then plug that phone into the FXS port while you play and learn.. |
| 22:24.26 | paulc | aarcane: Then when you're ready, you can do some rewiring to get all your house phones fed from the FXS port |
| 22:24.26 | aarcane | WiretapSeven, okay, you're confusing me by saying "telco side". I want to cut the landline, and use a sip (or similar) service to make my existing phones ring.. |
| 22:24.46 | WiretapSeven | aarcane, 'telco side' is the line that runs back to your telco |
| 22:24.57 | paulc | aarcane: So you want no service from your Telco at all? Just pure SIP? In that case, something like a SPA-1001 will do the job nicely |
| 22:25.18 | WiretapSeven | you can ignore it completely, HOWEVER, the law in many countries can require that you have a telco physical line available for emergency dialling |
| 22:25.20 | paulc | or PAP2T-NA |
| 22:25.44 | paulc | WiretapSeven: Interesting.. care to name a few? |
| 22:26.05 | WiretapSeven | paulc, don't have a list off the top of my head, about 90% sure .au does though |
| 22:27.43 | aarcane | hrrm, okay, I think I get how most of it is supposed to work now.. but I'm still unsure about getting a pci(e) card or one of the SPA3102 boxes. I'm most curious to know if either of those is a more reliable option, or provides better sound quality than the other. |
| 22:27.52 | paulc | Interesting.. makes sense.. |
| 22:28.08 | paulc | aarcane: You don't need a SPA-3102 if you're not going to bring a telco line in |
| 22:28.21 | citywok | ~book |
| 22:28.21 | infobot | For more information about the Asterisk book, see ~thebook |
| 22:28.24 | citywok | ~thebook |
| 22:28.25 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 22:28.32 | citywok | anygivenname: ^^^ |
| 22:28.47 | paulc | aarcane: Use something like a PAP2T-NA instead. It will let you have 2 FXS devices connected. That could be "all your house phones on one port" or maybe "upstairs phones on one port, downstairs phones on another" depending on wiring |
| 22:29.22 | WiretapSeven | aarcane, the ATA is a pretty tidy option, theyre also rather stable |
| 22:29.56 | WiretapSeven | and yeah, I use a PAP2T-NA right now, am kinda regretting not getting an SPA3000/3102 as my changeover is not a smooth one |
| 22:36.23 | aarcane | okay, so.. after all your help, it looks like the changeover is a simple one, if I just buy a new box and hook it up. I just need to acquire a SIP service provider and go through the manual or book :) |
| 22:38.12 | WiretapSeven | aarcane, yep, and I really recommend you use * as an intermediary rather than just outright connecting your ATA to your service provider |
| 22:41.53 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:50.00 | aarcane | WiretapSeven, well, obviously :) |
| 22:50.56 | WiretapSeven | aarcane, some people don't and will have their ATA register directly with their ITSP |
| 22:54.23 | aarcane | well, wouldn't that not even use asterisk ? |
| 22:55.48 | WiretapSeven | correct |
| 22:58.44 | aarcane | well, I have two main purposes for wanting to install asterisk, instead of just using some cookie-cutter VOIP service. firstly, I want to be able to record incoming and outbound calls in accordance with local laws, and secondly, I want to be able to route incoming calls through an optional menu system. |
| 22:59.25 | WiretapSeven | yeah |
| 23:00.07 | aarcane | something along the lines of "We don't recognize your number, are you a human being? if so, please say your name now *beep* if not, please stay on the line", and send blacklisted numbers through an infinite loop of menu options. |
| 23:00.46 | WiretapSeven | lol |
| 23:00.54 | WiretapSeven | I plan to use mine to manage telemarketers |
| 23:01.10 | WiretapSeven | theyre not allowed to call businesses here |
| 23:01.10 | aarcane | exactly |
| 23:01.21 | WiretapSeven | so when it rings and they get hold music, they will likely hang up |
| 23:06.19 | aarcane | I hope to require them to press a random number every few minutes until they give up. |
| 23:06.49 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-252-104.pools.arcor-ip.net) |
| 23:09.48 | WiretapSeven | lol |
| 23:10.38 | WiretapSeven | 'may I please speak to the householder?' "one moment, transferring you now" *directs to infinite-hold queue |
| 23:12.51 | aarcane | but only blacklisted numbers. whitelisted numbers will ring straight through, and everyone else is on the screening queue |
| 23:13.36 | aarcane | I hate to look like an idiot that hasn't read the faq, but does asterisk work with google voice ? |
| 23:16.47 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:17.24 | *** join/#asterisk twanny796 (~twanny@46.11.88.154) |
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| 23:38.34 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:40.50 | WiretapSeven | aarcane, not reliably, or rather, google voice is not reliable |
| 23:41.48 | russellb | ~googlevoice |
| 23:41.48 | infobot | For information on setting up calls with google using Asterisk 1.8, see https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google, or see the External Services chapter: http://ofps.oreilly.com/titles/9780596517342/ch17.html |
| 23:52.48 | leifmadsen | I need to get that setup again |
| 23:52.53 | leifmadsen | it might even work if I'm lucky |
| 23:53.07 | leifmadsen | (I mean already work as I did set it up a month or two ago) |
| 23:53.29 | russellb | I should set it up, too ... but I'm lazy |
| 23:54.20 | carrar | well it is friday |
| 23:54.23 | michael-i | heading out for the weekend! thanks for the originate help again. have a good one everyone |
