| 00:02.24 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 00:02.48 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 00:04.04 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:04.31 | *** join/#asterisk ctooley (n=ctooley@209.33.108.119) |
| 00:04.42 | ctooley | I'm seeing some odd IAX2 behavior. |
| 00:05.17 | [netman] | how odd? |
| 00:05.39 | [TK]D-Fender | [netman]: Even + 1 |
| 00:06.00 | [netman] | too bad |
| 00:06.10 | *** join/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
| 00:07.06 | *** join/#asterisk kisei (n=bit_frog@sion.redback.com) |
| 00:07.38 | *** join/#asterisk adorah (n=Michael@87.69.130.90) |
| 00:08.32 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:09.50 | *** join/#asterisk RipeR-81 (n=riper81@190.53.33.23) |
| 00:10.10 | RipeR-81 | anybody has configured net2phone ? |
| 00:10.31 | RipeR-81 | with asterisk as a sip trunk ? |
| 00:11.59 | glaz | for an unknown reason, my asterisk has stopped working and i haven't played with the config for at least 3 months... |
| 00:12.00 | [TK]D-Fender | RipeR-81: http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone |
| 00:12.13 | glaz | Unable to create channel of type 'IAX2' (cause 3 - No route to destination |
| 00:12.35 | glaz | and yes, I can traceroute my 'destination' |
| 00:13.04 | [TK]D-Fender | glaz: pastebin the call * and configs masking only passwords |
| 00:13.28 | glaz | the call * ? |
| 00:13.36 | glaz | the iax.conf ? |
| 00:14.08 | [TK]D-Fender | glaz: CLI output, "iax2 show peers, and your iax.conf masingk only passwords |
| 00:14.15 | glaz | ok |
| 00:15.28 | glaz | http://rafb.net/p/luxkhD21.html |
| 00:16.28 | [TK]D-Fender | glaz: You have no IP specified, and the other side has not registered to your * |
| 00:16.37 | [TK]D-Fender | glaz: domhome (Unspecified) (D) 255.255.255.255 0 (T) Unmonitored |
| 00:16.41 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 00:16.59 | glaz | lemme reload iax on the other side |
| 00:18.17 | *** part/#asterisk trapa (n=2264581@207.230.238.94) |
| 00:18.21 | jaytee | well, that corn-on-the-cob was pretty disappointing :-( |
| 00:18.28 | glaz | it "seems" working, but nothing happens |
| 00:20.06 | ctooley | and of course wasn't prepared with the pastebin before |
| 00:20.10 | ctooley | Now... |
| 00:20.24 | ctooley | http://www.pastebin.ca/1059555 |
| 00:20.36 | ctooley | IPs and MACs changed to protect the utterly guilty |
| 00:20.45 | unpaidbill | what is the proper way use AGI + MeetMe? what i'm doing is this: I originate a call to the application meetme, with data 9905|cdb (announce callers, create the meetme dynamically so i dont have to specify it in meetme.conf, execute an agi when a user joins). the user joins, the agi executes properly, plays an audio stream, then the agi exits and the line hangs up. i see that this is how it is supposed to happen.. I tried to keep the agi alive by doing |
| 00:20.59 | unpaidbill | but that appears to block audio |
| 00:21.06 | ctooley | So, call goes out, no response for a long time, and then an AUTHREQ comes in, followed by an immediate INVAL coming in. |
| 00:21.48 | unpaidbill | is there some way i can make the agi just sit there and wait till everyone hangs up without blocking audio? |
| 00:22.01 | JT | ctooley: MACs changed? that is paranoid |
| 00:22.17 | ctooley | JT eh, not my hardware |
| 00:22.34 | [TK]D-Fender | glaz: What qualifies as "seems working"? If you don't see the IP and a registration notice, then it has not worked. |
| 00:23.51 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:24.16 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ea5bc073bff9ed5d) |
| 00:24.32 | glaz | [TK]D-Fender: http://rafb.net/p/COaMup65.html |
| 00:24.43 | JT | ctooley: MACs are L2 and only accessible if directly connected |
| 00:24.45 | glaz | that's on the remote site. |
| 00:24.47 | JT | masking IPs makes sense |
| 00:25.12 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 00:25.32 | [TK]D-Fender | glaz: Proof of nothing. |
| 00:25.43 | glaz | [TK]D-Fender: allright, what do you suggest? |
| 00:25.46 | [TK]D-Fender | glaz: if on the receiving side you get no message, then its no good. |
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| 00:26.55 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 00:28.25 | glaz | I gotta go eat, i'll figure out after diner. thanks. |
| 00:28.38 | JT | ctooley: the MACs are still there btw. Sonicwal_22:3f:50 |
| 00:29.13 | ctooley | JT, yeah |
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| 00:47.43 | Pctech37|Mac | problem fixed |
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| 01:14.43 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 01:16.12 | TrentCreek | I was just trying to compile 1.4.21.1 on CentOS 5 x64 when it came across this problem: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 01:16.30 | TrentCreek | g++ is not on the server |
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| 01:17.34 | Qwell | so...install it. :D |
| 01:17.35 | [TK]D-Fender | TrentCreek: You have our permission ot install the rest of *'s prerequisites... |
| 01:18.05 | TrentCreek | i ASSume it's available in 64 bit? |
| 01:18.19 | TrentCreek | not in Yum |
| 01:18.46 | [TK]D-Fender | TrentCreek: use the carpet-bombing approach : "yum install gcc*" |
| 01:19.28 | TrentCreek | running now :-) |
| 01:19.32 | jaytee | TrentCreek, why are you going with 64 bit? |
| 01:19.33 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 01:19.58 | TrentCreek | more data bandwidth |
| 01:20.37 | TrentCreek | It installed! Thanks geetar (oops..not instrument) |
| 01:21.30 | jaytee | I run * on RHEL 5 64 bit and it works great but when I started looking at Lumenvox for a voice recognition enabled auto attendant I found out Lumenvox only supports * on 32 bit platforms at them moment. |
| 01:21.52 | TrentCreek | I have a compared AMD 64 to Intel's DUO 2 QUAD 32 bit...The AMD was almost as fats |
| 01:21.58 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 01:22.10 | TrentCreek | I will not be doing auto attendent |
| 01:22.12 | jaytee | mine is running on a Quad Xeon |
| 01:22.36 | jaytee | all the standard * stuff works fine and the add-ons too. |
| 01:22.57 | TrentCreek | yeah, but for now I am running it on VPS |
| 01:23.20 | TrentCreek | just want to be sure to have better performance |
| 01:23.26 | jaytee | probably a good idea to have my IVR and voice recognition running on a separate server anyways. |
| 01:23.42 | *** join/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
| 01:23.59 | TrentCreek | yeah..escially since this is only a VPS with minimum resources till I get it fully loaded down |
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| 01:25.23 | TrentCreek | I got the Asterisk Logo! It work, It works! |
| 01:25.32 | TrentCreek | cig time ;-) |
| 01:25.36 | jaytee | yeee haaawwww!!!! |
| 01:25.45 | jaytee | smoke em if ya got em! |
| 01:26.32 | TrentCreek | I just wish it was in the Packages in CentOS so I would not have to mess with trying to figure out how ot make it a service |
| 01:26.59 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 01:27.09 | TrentCreek | But for 64 bit..it was CentOS, or Ubuntu.. |
| 01:27.40 | TrentCreek | reviews indicated CentOS was better as a server |
| 01:27.42 | jaytee | TrentCreek, just type chkconfig zaptel on in the zaptel source directory |
| 01:28.07 | jaytee | and then chkconfig asterisk on in the asterisk source directory |
| 01:28.07 | TrentCreek | No zaptel..pure VOIP |
| 01:28.38 | jaytee | then just do the chkconfig asterisk on and then service asterisk start after that. Asterisk will start as a service then and when you restart the server. |
| 01:29.10 | TrentCreek | oh..and i was looking at a HOW-TO in the book..what a brain cracker |
| 01:29.25 | JT | TrentCreek: distro is a matter of preference |
| 01:29.31 | jaytee | here's a good howto to ref for CentOS 5 with *. http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 01:29.36 | JT | but personally i avoid rpm based distros especially for servers |
| 01:30.41 | TrentCreek | yes..but I googled... CentOS 5 vs Ubuntu ....More for CentOS as a server from what I saw |
| 01:30.42 | jaytee | I've run * on both Ubuntu and RHEL and CentOS. I don't mind using yum even though I prefer a debian based distro for most things like my desktop which runs Ubuntu |
| 01:30.43 | riddlebox | hrmm just got a ton of 3com phones and it looks like I might be able to upgrade them and use them as sip phones, but first I have to find the upgrade... |
| 01:31.08 | JT | TrentCreek: that's not a very good way to determine what distro to use |
| 01:31.13 | jaytee | I just prefer compiling from source rather than installing from third party repos. |
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| 01:31.42 | TrentCreek | JT: Why not? |
| 01:31.54 | TrentCreek | yes, compiling is good also.. |
| 01:32.16 | JT | TrentCreek: because that is not an objectivy comparison based on your own knowledge |
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| 01:32.21 | JT | popularity contests are stupid |
| 01:32.27 | tzafrir_laptop | I rather install from main repo than from third-party repo... |
| 01:32.48 | *** join/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 01:33.02 | jaytee | Ubuntu has * in it's repos but even in Hardy it's not a very recent version of 1.4 |
| 01:33.06 | JT | s/objectivy/objective/ |
| 01:33.20 | *** part/#asterisk sun_moon (n=RaviRaja@61.11.80.82) |
| 01:33.46 | TrentCreek | people were not just say "i perfer it because i like it better" |
| 01:33.48 | tzafrir_laptop | Ubuntu has Asterisk in Universe. It hopefully works |
| 01:34.00 | jaytee | it does |
| 01:34.10 | TrentCreek | they were saying things like "it seems to perfor better...etc etc |
| 01:34.18 | tzafrir_laptop | but it's not LTS |
| 01:34.19 | jaytee | I still prefer a compiled system. |
| 01:34.30 | JT | centos is like a rip off of red hat enterprise without the support |
| 01:34.33 | JT | imho pointless |
| 01:34.53 | TrentCreek | yes, and partially why the said it was better |
| 01:34.56 | [TK]D-Fender | jaytee: For common stuff except *, sure. devel packages? OK. Standard base utils? Sure. Specialized servers? No thanks... |
| 01:35.00 | tzafrir_laptop | JT, then likewise Debian has no support |
| 01:35.02 | JT | TrentCreek: they must be on drugs |
| 01:35.07 | JT | TrentCreek: that makes it *worse* |
| 01:35.14 | *** part/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
| 01:35.25 | JT | tzafrir_laptop: yeah, but it has more community support |
| 01:35.36 | jaytee | [TK]D-Fender, no thanks to what? package builds of asterisk? |
| 01:35.38 | JT | i wouldn't subject myself to rpm without support |
| 01:35.39 | JT | ;) |
| 01:35.49 | [TK]D-Fender | jaytee: indeed |
| 01:36.00 | jaytee | well, then we agree :-) |
| 01:36.34 | TrentCreek | why would it? RH is one of the oldest surviving distros |
| 01:36.51 | JT | TrentCreek: centos is NOT red hat. |
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| 01:37.01 | TrentCreek | true, but based on it |
| 01:37.05 | JT | it is a rip off of red hat, which affords you none of red hat's support |
| 01:37.11 | JT | you don't seem to be listening |
| 01:37.32 | jaytee | I've got RHEL 5 with support and I haven't really needed it.......yet. |
| 01:37.32 | JT | if you go to a commerial distro, you'd want the support to keep the business folks happy |
| 01:37.50 | JT | if you get rid of the support, it becomes pointless |
| 01:38.21 | TrentCreek | not if you are supporting yourself and with the help of good people ike on here... |
| 01:38.39 | jaytee | JT, in a large enterprise that cannot accept ANY or little downtime I'd agree with you. |
| 01:38.57 | JT | TrentCreek: then i'd want a distro that was well supported by the community, which is generally something debian based |
| 01:39.20 | TrentCreek | I was thinking debian, but |
| 01:39.25 | jaytee | and even a small business can get bitten on the ass if there's something seriously hosed with the OS and you have no one you can call for priority support. |
| 01:39.36 | TrentCreek | Everything I read was..dont go for it |
| 01:39.39 | JT | ubuntu is debian based, hot tip |
| 01:39.48 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.127.238) |
| 01:39.49 | JT | TrentCreek: you must've read a lot of rubbish to be honest |
| 01:39.59 | TrentCreek | actually...I think fedora based, isnt it? |
| 01:40.10 | JT | jaytee: true, but windows support is a falicy |
| 01:40.18 | JT | TrentCreek: ubuntu is debian based. |
| 01:40.39 | JT | jaytee: it will cost you hundreds of dollars to even speak to a real support person at microsoft |
| 01:40.41 | TrentCreek | ahh |
| 01:41.07 | TrentCreek | yes they will..even says "Please have your CC ready for a $500 charge minimum" |
| 01:41.35 | jaytee | JT, if you ever get a blue screen of death stop error on a Win server and search for it on the MS KB you usually get squat for results. If you search a pay for use site like Experts Exchange you get tons of hits, most of which link to MS KB articles. Ever wonder why that might be? |
| 01:41.46 | jaytee | TrentCreek, exactly! |
| 01:42.18 | JT | jaytee: can't you view most experts exchange stuff for free these days? |
| 01:42.19 | jaytee | and if you make good revenue supporting your buggy software by charging for support what incentive do you have to make the software less buggy? |
| 01:42.34 | JT | jaytee: does ms run experts exchange? |
| 01:42.41 | [TK]D-Fender | Funny thing is Novell supports netware 100% over the phone :) |
| 01:42.48 | [TK]D-Fender | Not that I'd want to USE it. |
| 01:42.57 | jaytee | JT, not that I'm aware of. I'm a member but I'm pretty sure they want you to sign up to see the solution to any search. |
| 01:43.07 | DigitalIrony | Debian is the way to go if you want to support yourself though |
| 01:43.23 | JT | jaytee: these days all the solutions seem to come up for free |
| 01:43.30 | JT | jaytee: you just scroll down past the bit saying you need to pay |
| 01:43.32 | TrentCreek | Strange! |
| 01:43.35 | jaytee | back in the early 90's Netware was about the best NOS you could run. OS/2 sucked ass. |
| 01:43.35 | TrentCreek | /usr/bin/ld: skipping incompatible /usr/lib/libidn.so when searching for -lidn |
| 01:43.35 | TrentCreek | /usr/bin/ld: skipping incompatible /usr/lib/libidn.a when searching for -lidn |
| 01:43.35 | TrentCreek | /usr/bin/ld: cannot find -lidn |
| 01:44.07 | JT | TrentCreek: your 64bit libraries aren't properly set up |
| 01:44.12 | TrentCreek | boo! |
| 01:44.32 | TrentCreek | suggestions? |
| 01:44.34 | JT | jaytee: do they pay people to add the solutions to EE? |
| 01:44.48 | jaytee | JT, I don't think so |
| 01:45.26 | JT | jaytee: i can't work out why anyone would add solutions to it then |
| 01:45.34 | JT | i hate experts exchange |
| 01:45.44 | JT | attempting to charge for communal information sharing |
| 01:46.17 | jaytee | I find it useful when I'm pressed for time and at 9.95 a month versus a single call to MS it's paid for itself many times over. |
| 01:46.45 | JT | i'm sure it's useful |
| 01:47.01 | jaytee | I think if you solve lots of problems and accumulate points you get discounted rates or free service for a period. |
| 01:47.03 | JT | i just debate the ethics of what they do, unless there's something i'm missing about how they oberate |
| 01:47.13 | JT | operate |
| 01:47.46 | jaytee | JT, it is a rather odd business model but the site does have a large database of solutions to problems and a decent localized search engine. |
| 01:48.14 | JT | if they pay the "experts", fair enough |
| 01:48.24 | jaytee | I use it for some MS stuff and use www.tek-tips.com for Nortel stuff. I use this forum and voip-info.org for * and voip stuff. |
| 01:50.23 | TrentCreek | JT: I think I got it...the 64 libraries were not installed... |
| 01:52.45 | TrentCreek | It worked! Woohoo |
| 02:01.39 | TrentCreek | i wonder why the make samples is not working |
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| 02:08.13 | TrentCreek | Now what?? service asterisk start |
| 02:08.22 | TrentCreek | Starting asterisk: Cannot find your TTY (9) |
| 02:11.04 | TrentCreek | got it...rnning |
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| 03:02.34 | TrentCreek | give it to me |
| 03:04.32 | jaytee | what? |
| 03:08.28 | TrentCreek | ChanServ gives channel operator status to russellb |
| 03:08.44 | TrentCreek | now time to start on asterisk2billing |
| 03:09.13 | jaytee | why would you want to be an op? you're kinda new to this aren't you? |
| 03:09.37 | russellb | o.O |
| 03:09.50 | TrentCreek | so I can boot |
| 03:09.52 | TrentCreek | haha |
| 03:10.10 | jaytee | ah, the drunk with power syndrome |
| 03:10.19 | TrentCreek | buwahahahaa |
| 03:10.24 | TrentCreek | kick jaytee |
| 03:10.30 | TrentCreek | gosh darm it did not work |
| 03:11.04 | *** kick/#asterisk [TrentCreek!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (like this?) |
| 03:11.24 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 03:11.37 | TrentCreek | hey hey..no power stealing |
| 03:11.42 | jaytee | well, russellb is an op because he's also one of the * devs and that means some of the code you've compiled he wrote |
| 03:12.22 | TrentCreek | well, I hope he included 64 bit CPU functions to take advantage of the extra power :-) |
| 03:13.17 | jaytee | I'm an op on another channel on the blitzed net and I rarely ever have to kick anyone except for the occassional troll named Fernando coming in and asking all the women present for A/S/L |
| 03:13.41 | TrentCreek | so he thinks they are all blitzed |
| 03:13.49 | jaytee | Trent, the code is the same, it's the compiler that makes the difference |
| 03:14.04 | jaytee | and the kernel |
| 03:14.42 | TrentCreek | ahhh , yes true..then I hope the GCC guys too advantage of the larger number handling |
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| 03:14.54 | russellb | oh man, you know what, i bet they didn't think of that |
| 03:14.59 | jaytee | lol |
| 03:15.00 | russellb | you should get on their development mailing list and remind them |
| 03:15.05 | TrentCreek | gosh darn it |
| 03:15.09 | jaytee | rofl |
| 03:15.45 | TrentCreek | yeah I think I will |
| 03:15.57 | russellb | ok, great |
| 03:16.11 | russellb | and please please please email me at russell@digium.com when you do, so I can watch the thread |
| 03:16.33 | TrentCreek | and I think I will rewrite * in NASM and become net hero |
| 03:16.37 | russellb | heh |
| 03:17.00 | russellb | i'm almost done with the rewrite in bash |
| 03:17.22 | jaytee | hey russell, tell the guys in hardware that the TE212 card is one sweet rock solid piece of hardware. |
| 03:17.22 | russellb | i have a blasted syntax error and all i get is > |
| 03:17.23 | TrentCreek | oh...want it to run slower, eh? |
| 03:17.42 | russellb | jaytee: will do, thanks :) |
| 03:18.27 | russellb | goes to bed |
| 03:18.36 | TrentCreek | nyte |
| 03:20.15 | jaytee | TrentCreek, if or when you move to * 1.6 and want to use the MeetMe application you can thank russellb for not having to load ztdummy for timing anylonger like you have to now in 1.2 and 1.4 |
| 03:21.12 | TrentCreek | Well,,,I am sure it is greta, but I don't use zaptel |
| 03:21.33 | TrentCreek | Will it use system timing? |
| 03:21.34 | jaytee | I know, you said that earlier |
| 03:21.46 | jaytee | MeetMe? not in 1.4 |
| 03:21.51 | jaytee | you'll need ztdummy |
| 03:22.43 | TrentCreek | I really have not read up much on MeetMe...I think I glanced over it and decided I would not need it at the moment |
| 03:22.57 | jaytee | MeetMe and it's counterpart Page are really nice apps. |
| 03:23.29 | jaytee | TrentCreek, it's great if you want moderated conferences or conferences with over 3 people. |
| 03:24.12 | TrentCreek | ahh |
| 03:24.14 | jaytee | and Page can create a dynamic conference that pulls everyone into the conference by dialing their phones. |
| 03:24.33 | TrentCreek | yeah..that is why I did not need it yet |
| 03:24.49 | TrentCreek | really dont have employee..hehe |
| 03:24.56 | jaytee | I just wrote a macro so that someone can call an extension that lets them enter multiple extensions and then runs the Page application to put them all in conference. |
| 03:25.42 | TrentCreek | now that sounds sweet. |
| 03:25.46 | *** join/#asterisk amerine (n=mturner@12.160.242.195) |
| 03:25.53 | jaytee | well, if you ever try to sell your services to some company that wants * it would help to know it cuz it's a great "selling" feature. |
| 03:26.41 | jaytee | and if you tried to do conferencing with over 6 people on a Nortel Meridian switch the Conference Bridge card and licensing would set you back thousands. |
| 03:26.52 | TrentCreek | ouch |
| 03:26.57 | jaytee | I'm sure Avaya's solution is just as pricey |
| 03:27.37 | TrentCreek | I should have sucked off info from my mom when I had the chance.. |
| 03:27.48 | jaytee | about what? |
| 03:28.03 | TrentCreek | She used to work for a telecom company and knew more than I ever could, and the shortcommings |
| 03:28.36 | jaytee | when you had the chance? has she passed on? |
| 03:29.13 | TrentCreek | no..she retired 8 years ago and now the info is outdated..I could have gotten info from her and expanded on that |
| 03:30.17 | jaytee | I kick myself for not making my grandmother teach me french, italian and latin when I was very little. She was a schoolteacher and spoke them all fluently. She taught me to read before I entered first grade but she never bothered with other languages. |
| 03:32.33 | TrentCreek | Yeah I know what you mean..all could have been useful. |
| 03:33.02 | TrentCreek | But, if you live in the SW US you can easily learn Spanish then getting Italian is simple |
| 03:33.12 | TrentCreek | almost identical |
| 03:33.21 | jaytee | any second language is a major plus on a resume especially if your fluent in it and can back up the claim. |
| 03:33.40 | jaytee | yeah, Italian and Spanish are so very very close |
| 03:34.36 | TrentCreek | Strange how Spanish and Italian came directly from Latin, but the syntax is totally different |
| 03:34.49 | jaytee | and if you learn latin it gives you a better feel for semantics in most european languages. |
| 03:36.06 | TrentCreek | i started learning some Latin and it is different... |
| 03:36.14 | TrentCreek | Bob Steve Killed |
| 03:36.21 | DigitalIrony | if you know english german would probably be easy to learn |
| 03:36.31 | TrentCreek | translates to Steve Killed Bob |
| 03:36.51 | TrentCreek | yes and no |
| 03:37.52 | DigitalIrony | The small bits and pieces of german I know sound and spell very closely to english, they syntax is a bit different but its alot and the same |
| 03:37.52 | TrentCreek | Though it is base on German...due to Britan being occupied by Latin, Frogs, and Germans...the end result is English |
| 03:39.01 | jaytee | the part I don't like about spanish and german is that the articles have gender. Why do I have to care if the pencil is masculine, feminine or neutral? it's a damn pencil!!! |
| 03:39.09 | TrentCreek | The structure is German, but 1/4 of the syntax is Latin based |
| 03:39.36 | TrentCreek | English that is |
| 03:39.52 | jaytee | I'm off to sleep, nite all |
| 03:39.58 | TrentCreek | nyte |
| 03:40.12 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:42.17 | DigitalIrony | wasn't that a movie |
| 03:42.28 | TrentCreek | knight |
| 03:42.36 | DigitalIrony | no |
| 03:42.41 | DigitalIrony | eXistenZ |
| 03:42.49 | TrentCreek | dun know |
| 03:43.23 | DigitalIrony | pretty sure it was a B movie along the same lines of the matrix mixed with VR games |
| 03:48.18 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 03:48.42 | [TK]D-Fender | eXistenZ is a schmuck trying to get an SPA-3102 to be smarter than he is (which isn't saying much), and has never installed or used * before. Oh yeah... and a cheap Matrix rip-off :) |
| 03:50.02 | drmessano | ~centos52bug |
| 03:50.03 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 03:56.33 | *** join/#asterisk d-k-t (n=dt@125.120.130.131) |
| 03:59.18 | TrentCreek | huh? |
| 04:01.11 | DigitalIrony | [TK]D-Fender: his questions wasn't involving G729 was it? |
| 04:01.46 | [TK]D-Fender | DigitalIrony: No, just how to get anonymous CID calls blocked on passthrough from the FXO to FXS ports on his SPA-3102 |
| 04:01.55 | DigitalIrony | ahh ok |
| 04:02.00 | [TK]D-Fender | DigitalIrony: not even VoIP related. |
| 04:02.28 | DigitalIrony | [TK]D-Fender: I had someone trying to get G729 to pass through a SPA-2102 and was complaining it wasn't working |
| 04:02.55 | [TK]D-Fender | DigitalIrony: Ok, fine, sure... |
| 04:03.03 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.29 | DigitalIrony | [TK]D-Fender: well i was hoping for some input :P cause SPA isn't something i really deal with |
| 04:06.13 | [TK]D-Fender | DigitalIrony: Like most of these problems, we'd need quality information to debug. |
| 04:06.22 | [TK]D-Fender | DigitalIrony: SIP debug, full CLI output, configs, etc. |
| 04:06.29 | TrentCreek | SPA is a good shpw on Animal planet |
| 04:06.39 | [TK]D-Fender | DigitalIrony: Because what you've said really hasn't said much of anything. |
| 04:07.40 | DigitalIrony | [TK]D-Fender: I can debug it when i have that....I'm sorry I could have been more clear. Any general information you might about about that model would be nice, so I can tell the person some things to try while i wait on them to send me ssh info...e-mail is slow |
| 04:08.05 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 04:08.13 | [TK]D-Fender | DigitalIrony: Ok. In general.... it works :p |
| 04:08.48 | DigitalIrony | [TK]D-Fender: good, thats basically what i wanted to know....didn't want to waste time figuring out it didn't work :P I appreciate it |
| 04:09.00 | [TK]D-Fender | DigitalIrony: And the only specific point concerning G.729 and that seris is that to my knowledge, it can only decode a SINGL G.729 stream at a time. |
| 04:09.39 | DigitalIrony | [TK]D-Fender: Thats fine, the customer only has one channel anyway. But if that were on an * server doesn't it only have two ports anyway |
| 04:09.41 | [TK]D-Fender | DigitalIrony: May or may not be relevant as you didn't state any of the circumstances. |
| 04:10.20 | [TK]D-Fender | DigitalIrony: the single G.729 affects a single port on 3-way call. |
| 04:10.40 | *** join/#asterisk Pctech37|Mac (n=Pctech37@unaffiliated/pctech37) |
| 04:12.39 | DigitalIrony | [TK]D-Fender: your right I didn't, because I don't really know, the customer was very....um brief in their description. It would appear that G729 is working on his asterisk server, so there isn't much I can really support him with, just trying to pick at people and see if this was a bug anyone else had and if their was an easy fix. Thanks for the help, leaving for lunch now |
| 04:13.05 | [TK]D-Fender | k |
| 04:14.43 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:18.47 | *** join/#asterisk CunningPike_ (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:20.32 | *** join/#asterisk erojasv (n=erojasv@190.43.15.224) |
| 04:24.40 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 04:30.22 | *** join/#asterisk xlogik (n=xlogik@c-71-232-176-24.hsd1.ma.comcast.net) |
| 04:34.35 | drmessano | Interesting |
| 04:35.34 | drmessano | I made the mistake in suggesting Asterisk should handle older modules made by itself a little better |
| 04:36.37 | drmessano | Now I see chan_zap.so and codec_zap.so get left behind, and thus conflict when upgrading to the newer dahdi |
| 04:37.13 | drmessano | I'm sure there was some well placed readme that instructed me to do that first.. surely. |
| 04:39.31 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 04:39.43 | drmessano | app_zapbarge.so, app_zapscan.so, app_zapras.so |
| 04:39.45 | drmessano | sweet |
| 04:40.17 | TrentCreek | what is sweeter is that I just compiled 1.4.21.1 in 64 bit |
| 04:40.23 | TrentCreek | yeah, boy |
| 04:40.41 | drmessano | Why is that so sweet? |
| 04:40.56 | *** join/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
| 04:41.05 | drmessano | m-a-k-e |
| 04:41.07 | TrentCreek | power! |
| 04:41.32 | TrentCreek | quadroople the calls! |
| 04:41.40 | drmessano | uh huh |
| 04:41.59 | TrentCreek | Just like in that movie.... |
| 04:42.13 | *** part/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
| 04:42.13 | TrentCreek | Lawnmower Man...call the whole world at the same time |
| 04:42.22 | drmessano | I'm just trying to work out how hard it would have been for the new zaptel to remove the old modules |
| 04:43.06 | drmessano | I guess it was just easier for *me* to do it |
| 04:45.10 | TrentCreek | sure |
| 04:46.15 | drmessano | Just like Redhat making it harder and harder to disable selinux |
| 04:47.31 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 04:48.20 | [TK]D-Fender | drmessano: Yeah, that blatant screen when you install it wasn't enough ;) |
| 04:49.09 | *** part/#asterisk amerine (n=mturner@12.160.242.195) |
| 04:49.38 | drmessano | Hmmm |
| 04:50.58 | drmessano | Except there isn't one |
| 04:51.17 | drmessano | Zaptel gives you the screen telling you it was installed |
| 04:51.36 | drmessano | Asterisk tells you about modules created with another version, but DOES NOT include the zaptel modules |
| 04:51.43 | drmessano | So, no screen |
| 04:53.38 | [TK]D-Fender | drmessano: I didn't say Zaptel..... |
| 04:53.44 | [TK]D-Fender | drmessano: when you instll the *OS* |
| 04:53.56 | [TK]D-Fender | "Do you want to install SELinux (yes/no)" |
| 04:54.01 | drmessano | Oh |
| 04:54.04 | drmessano | For CentOS |
| 04:54.26 | *** part/#asterisk Pctech37|Mac (n=Pctech37@unaffiliated/pctech37) |
| 04:54.30 | drmessano | No option for installing SELINUX |
| 04:54.54 | drmessano | Not in graphic mode or text mode |
| 04:54.56 | [TK]D-Fender | drmessano: Sometimes you avoid getting hit by a train, not by jumping off the tracks, but rather never getting ON them in the first place. |
| 04:55.09 | [TK]D-Fender | drmessano: Sure there is, I disable it every time myself. |
| 04:55.17 | drmessano | In CentOS 5? |
| 04:55.57 | [TK]D-Fender | drmessano: yup |
| 04:56.49 | drmessano | You'll have to show me a screenshot, because i've never seen it. I uncheck all the install options in the GUI and there's nothing prompting before that |
| 04:58.16 | [TK]D-Fender | Also easy to disable. |
| 04:58.29 | drmessano | It changed in 5.2 |
| 04:58.43 | [TK]D-Fender | drmessano: Guess anythings possible. |
| 04:59.00 | [TK]D-Fender | drmessano: since I haven't hard installed that one yet (just yum'd up to it on 2 servers though |
| 04:59.00 | drmessano | Well, considering I just installed it YESTERDAY, i'd say so |
| 04:59.10 | [TK]D-Fender | drmessano: http://sysdigg.blogspot.com/2008/01/how-to-disable-selinux-in-centos-5.html |
| 04:59.30 | drmessano | Yeah, that doesn't work anymore |
| 04:59.34 | [TK]D-Fender | "While installing CentOS 5 linux if you haven't paid much attention then chances are you probably have missed window where installation program ask to enable/disable SElinux." |
| 04:59.37 | drmessano | Needs to be disabled in grub.conf |
| 04:59.45 | [TK]D-Fender | drmessano: ew. |
| 05:00.23 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:01.18 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 05:01.39 | drmessano | Something has changed |
| 05:01.45 | drmessano | I never installed 5.0 |
| 05:01.51 | drmessano | Just 5.1 and 5.2 |
| 05:01.52 | drmessano | http://plainenglishsecurity.com/CentOS5.html |
| 05:02.17 | drmessano | http://plainenglishsecurity.com/CentOS521.jpg <--- Never seen that page |
| 05:02.28 | drmessano | Of course, I don't install X |
| 05:02.51 | drmessano | My install ends at "Congradulations" and the reboot prompt |
| 05:02.54 | drmessano | Err |
| 05:02.59 | drmessano | Congratulations |
| 05:03.02 | JT | stop using horrible distros that force SELinux on you :) |
| 05:03.34 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 05:03.34 | drmessano | It's not a big deal if they would stop changing how to disable it |
| 05:03.49 | JT | it should come disabled |
| 05:04.01 | [TK]D-Fender | drmessano: I've seen it in 5.0 & 5.1 personally. |
| 05:04.27 | drmessano | [TK]D-Fender: Yes, but if you look at that page, that's after the first reboot and continuing the GUI setup |
| 05:04.31 | drmessano | I don't install a GUI |
| 05:04.37 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 05:05.20 | [TK]D-Fender | drmessano: The exact point I'm not 100% sure of... I do install EVERYTHING, and then change the init runlevel once the install is finished |
| 05:05.25 | [TK]D-Fender | drmessano: just my method of working. |
| 05:06.19 | drmessano | My point is that the SELINUX prompt is only after continuing a GUI setup.. If one never installs a gui, after the first reboot you are greeted by a command prompt only.. not the welcome screen |
| 05:06.47 | drmessano | So therefore, it's only "obvious" to those that perform a full install |
| 05:07.15 | [TK]D-Fender | drmessano: Well as far as I can guess, its controlled through that simple config file. Tha means is not functional? |
| 05:07.37 | drmessano | No, not anymore.. |
| 05:07.54 | [TK]D-Fender | drmessano: Wow.... double whammy. |
| 05:08.05 | drmessano | You need to append selinux=no to the kernel directive in your grub.conf now |
| 05:08.20 | drmessano | Took me a couple hour google search to find that.. with 5.2 being so new |
| 05:09.08 | [TK]D-Fender | drmessano: I'm a linux newb myself (and functioning at my capacity throughout this conversation itself". This would be a good thing for me to keep in mind as well. |
| 05:09.30 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 05:09.50 | drmessano | From what it looks like, looking at these GUI pages, the drop down sets that one option in that one config file.. I wonder if the 5.2 setup has the same options now.. or if they removed that page.. They're trying to make it harder to disable so people actually FIX the security as it should be, not just shut it off |
| 05:10.24 | drmessano | Kinda like allowing apps in Vista UAC versus turning the shit off |
| 05:10.49 | drmessano | I'll load the 5.2 iso in a VMWare session and see how far it goes |
| 05:11.50 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 05:12.46 | the_5th_wheel | hi. if i have an agi script that works on 1.2, will it also work on .14? |
| 05:12.50 | the_5th_wheel | *1.4 |
| 05:13.00 | drmessano | I just yum upgraded to 5.2.. Now I get to reboot, recompile zaptel lol |
| 05:13.17 | drmessano | the_5th_wheel: Quite possibly not |
| 05:13.29 | drmessano | the_5th_wheel: 1.2 and 1.4 are very different |
| 05:13.31 | [TK]D-Fender | the_5th_wheel: Could be that some things changed from 1.2 to 1.4 Go compare the manuals and TEST it |
| 05:13.34 | drmessano | the_5th_wheel: Try it and see |
| 05:13.52 | drmessano | Either it works, or it wont |
| 05:14.01 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-96-246-91-9.nycmny.east.verizon.net) |
| 05:14.20 | the_5th_wheel | ok. i will test it. |
| 05:14.31 | VJFROMGT | is there a way to tell asterisk to randomly pick a trunk when there are many trunks in an outbound route? |
| 05:15.00 | drmessano | VJFROMGT: You just asked this in #asterisk and #freepbx |
| 05:15.01 | JT | trunk trunk route |
| 05:15.05 | drmessano | Is this a FreePBX question? |
| 05:15.07 | JT | #freepbx, VJFROMGT |
| 05:15.10 | drmessano | or an Asterisk Question |
| 05:15.23 | VJFROMGT | either answer will work for me |
| 05:15.32 | JT | ... |
| 05:15.36 | drmessano | Negative |
| 05:15.38 | JT | it's a freepbx question |
| 05:15.42 | drmessano | Only one answer will work |
| 05:15.49 | drmessano | Christ |
| 05:15.52 | JT | asterisk has no concepts of trunks and outbound routes |
| 05:16.22 | [TK]D-Fender | VJFROMGT: And when you get around to running your own configs, they'll do whatever YOU tell them to. |
| 05:16.41 | [TK]D-Fender | VJFROMGT: As long as FreePBX is running the show.... GOOD LUCK. This is not the place for you. |
| 05:17.09 | drmessano | [TK]D-Fender: Either answer will work for me |
| 05:17.17 | VJFROMGT | I have |
| 05:17.18 | VJFROMGT | [ALL] |
| 05:17.18 | VJFROMGT | include => outrt-001-D |
| 05:17.18 | VJFROMGT | include => outrt-002-M |
| 05:17.18 | VJFROMGT | include => outrt-003-G |
| 05:17.27 | drmessano | YAY, pastes too |
| 05:17.27 | the_5th_wheel | ok, while i play with my agi stuff, does asterisk 1.2 have a sip jitter buffer? |
| 05:17.36 | VJFROMGT | can asterisk randomly pick? |
| 05:17.43 | VJFROMGT | if yes, what is command? |
| 05:17.46 | [TK]D-Fender | VJFROMGT: ASTERISK does not pick anything! |
| 05:18.03 | VJFROMGT | ok |
| 05:18.09 | [TK]D-Fender | VJFROMGT: Its YOUR dialplan, it will match *1* exte, and no, not at rando=m. What it DOES is up to you. |
| 05:18.36 | [TK]D-Fender | the_5th_wheel: No. |
| 05:18.46 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 05:21.01 | VJFROMGT | so if dial plan has (see below) there is no way to randomly pick ? |
| 05:21.01 | VJFROMGT | exten => _7461269XXXXX,1,Macro(dialout-trunk,2,${EXTEN:4},,) |
| 05:21.01 | VJFROMGT | exten => _7461269XXXXX,n,Macro(dialout-trunk,9,${EXTEN:4},,) |
| 05:21.01 | VJFROMGT | exten => _7461269XXXXX,n,Macro(dialout-trunk,3,${EXTEN:4},,) |
| 05:21.15 | [TK]D-Fender | VJFROMGT: Go learn *. |
| 05:21.30 | frogonwheels | I've been trying to trace down some audio quality problems I'm having: |
| 05:21.40 | [TK]D-Fender | VJFROMGT: What you showed makes NO decision about what to exectute. |
| 05:22.03 | VJFROMGT | but asterisk always execute in that order trunk2, then 9 then 3 |
| 05:22.05 | [TK]D-Fender | VJFROMGT: And stop spamming useless dialplan chunks. |
| 05:22.22 | [TK]D-Fender | VJFROMGT: Yes, becase thats what you told it. |
| 05:22.43 | frogonwheels | 1 call connects fine. <1% CPU on a 233Mhz router. - and 20MB/s using ulaw. |
| 05:22.51 | VJFROMGT | correct, and my question is , ist there a comand that will tell it to pick oe at random? |
| 05:23.10 | [TK]D-Fender | VJFROMGT: Go look at the application and function listings yourself. |
| 05:23.11 | frogonwheels | but if a second call comes in, the quality of the incoming stream of the second call only is bad. |
| 05:23.18 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 05:23.23 | frogonwheels | choppy, tears etc |
| 05:23.39 | frogonwheels | - but the first call is still ok. |
| 05:23.57 | frogonwheels | I've tried win canreinvite on/off |
| 05:24.06 | frogonwheels | external phone provider is SIP. |
| 05:24.28 | frogonwheels | frogonwheels: if I then hang up the first call - the second call will continue fine. |
| 05:25.06 | [TK]D-Fender | frogonwheels: are both calls coming from the same source. |
| 05:25.09 | [TK]D-Fender | ? |
| 05:25.10 | JT | VJFROMGT: |
| 05:25.13 | JT | ~thebook |
| 05:25.16 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 05:25.23 | frogonwheels | yes. |
| 05:25.26 | JT | VJFROMGT: please take a look at the book, it will help a lot |
| 05:26.03 | frogonwheels | [TK]D-Fender: yes, both from the same source. |
| 05:26.26 | [TK]D-Fender | frogonwheels: and if you change which device you dial to bridge the call to? |
| 05:26.44 | [TK]D-Fender | frogonwheels: basically, could it be your endpoint? |
| 05:27.02 | frogonwheels | The endpoints are generally the same device - different ports. |
| 05:27.11 | frogonwheels | SIP 'PAP2T' ata. |
| 05:27.26 | frogonwheels | or sometimes over WLAN to my nokia mobile. |
| 05:27.40 | [TK]D-Fender | frogonwheels: Ok, so call 1 in from ITSP, out to PAP2 Port1. Next call in from ITSP out to port 2? |
| 05:27.52 | frogonwheels | yep |
| 05:28.02 | [TK]D-Fender | frogonwheels: SIP to ITSP? |
| 05:28.11 | [TK]D-Fender | nvm |
| 05:28.14 | [TK]D-Fender | Just read that |
| 05:28.22 | [TK]D-Fender | frogonwheels: Ok..... what codecs? |
| 05:28.26 | frogonwheels | ulaw |
| 05:28.31 | [TK]D-Fender | both legs? |
| 05:28.43 | frogonwheels | it's all I've got on at the moment. |
| 05:28.54 | frogonwheels | to avoid transcoding. |
| 05:29.01 | [TK]D-Fender | frogonwheels: and your internet connection is what exactly? |
| 05:29.09 | frogonwheels | even converted MOH to ulaw so I don't transcode. |
| 05:29.13 | frogonwheels | ADLS2+ |
| 05:29.19 | frogonwheels | ADSL2+ |
| 05:29.24 | frogonwheels | erm.. |
| 05:29.27 | [TK]D-Fender | frogonwheels: Might be an upstream bandwidth issue... |
| 05:30.22 | frogonwheels | possibly - 1021 Kbps upstream according to modem (bits) |
| 05:30.49 | [TK]D-Fender | frogonwheels: Ok.... near full synck... |
| 05:30.53 | [TK]D-Fender | eek. |
| 05:31.01 | [TK]D-Fender | frogonwheels: not sure what to say... |
| 05:31.31 | frogonwheels | The call seems to take 10KB/sec (bytes) bandwidth each way |
| 05:32.02 | frogonwheels | [TK]D-Fender: It might be my provider. I'm not sure they technicaly expect people to have 2 calls going |
| 05:32.57 | TrentCreek | problem is more than likely jitter |
| 05:33.01 | [TK]D-Fender | frogonwheels: That'd be more than sad. Allow it, but not be able to handle it... |
| 05:33.08 | frogonwheels | [TK]D-Fender: yeah |
| 05:33.12 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 05:33.22 | frogonwheels | [TK]D-Fender: but possibly they don't realise they allow it!? |
| 05:34.06 | frogonwheels | TrentCreek: how can I TELL that? |
| 05:34.26 | [TK]D-Fender | frogonwheels: Possible, but I wouldn't think they're incompetant like that. And it doesn't explain why it'd be degraded. |
| 05:34.38 | TrentCreek | Try this web site...run free test |
| 05:34.39 | TrentCreek | http://www.myvoipmcs.com/index.html |
| 05:34.41 | frogonwheels | [TK]D-Fender: yeah - and only one call degraded. |
| 05:34.41 | [TK]D-Fender | frogonwheels: Perhaps you should test with a better gateway |
| 05:35.08 | frogonwheels | [TK]D-Fender: you mean not run asterisk on my router ? ;) |
| 05:35.22 | [TK]D-Fender | frogonwheels: yeah :0 |
| 05:35.44 | frogonwheels | oh man , and spoil _all_ the fun. |
| 05:35.49 | *** join/#asterisk steliosk (n=Stelios@athedsl-106428.home.otenet.gr) |
| 05:36.17 | frogonwheels | [TK]D-Fender: but i've been running top -and asterisk doesn't even look like taking up even a couple of % cpu |
| 05:36.35 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 05:36.46 | frogonwheels | [TK]D-Fender: though I guess there are other bottlenecks to consider. |
| 05:36.54 | [TK]D-Fender | frogonwheels: Sanity check time <0 |
| 05:36.59 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 05:37.52 | frogonwheels | [TK]D-Fender: problem was that last time I tried connecting to my service provider from another machine inside the gateway - it wasn't all that successful |
| 05:38.02 | frogonwheels | [TK]D-Fender: NAT got in the way and all. |
| 05:38.07 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 05:39.06 | [TK]D-Fender | frogonwheels: bad router, or bad * config. |
| 05:39.10 | TrentCreek | did you try it out? It has a free online test |
| 05:40.05 | frogonwheels | TrentCreek: looking - doesn't seem to work under my mozilla |
| 05:40.32 | TrentCreek | are you on Linux? |
| 05:40.35 | frogonwheels | yep |
| 05:40.45 | TrentCreek | it uses java |
| 05:41.06 | TrentCreek | and I think you do not a sybomlic link to java |
| 05:41.46 | frogonwheels | ah ok - I'm doing it from my vista (puke) box |
| 05:41.54 | TrentCreek | you can try on a Winblows computer..since this only tests the network and not the computer |
| 05:43.27 | frogonwheels | hmm.. not so good QOS |
| 05:43.31 | frogonwheels | 3.5ms jitter |
| 05:43.39 | frogonwheels | 0% packet loss |
| 05:43.54 | frogonwheels | qos 19% - hmm. |
| 05:44.16 | frogonwheels | wonder if it actually sends RTP data - 'cause I have QOS handling on the router. |
| 05:44.53 | TrentCreek | guess have to see the FAQ |
| 05:45.10 | frogonwheels | thanks TrentCreek, [TK]D-Fender for your help. gg now - will read the FAQ |
| 05:45.18 | TrentCreek | I had a RoadRunner 10 Mbs down/1Mbs up and it still gave me bad jitter results |
| 05:46.04 | TrentCreek | it is measuring the whole network m but just parts |
| 05:46.14 | TrentCreek | not just parts |
| 05:46.17 | drmessano | I'm still just not convinced using Asterisk on a router |
| 05:48.29 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 05:49.56 | drmessano | [TK]D-Fender: There's something else factoring into the selinux as well |
| 05:50.43 | drmessano | [TK]D-Fender: 5.1 > 5.2 box, same kernel.. the former option to disable it still seems to be holding true.. has to be something compiled into the kernel or a package difference |
| 05:51.20 | [TK]D-Fender | drmessano: well... the kernel appears to be from the same series from what I've read, so must be part of the build process. |
| 05:55.14 | drmessano | CentOS, the OS that hacked Tuttle, Oklahoma, bitches |
| 05:55.31 | drmessano | CeNtOs OwNs U TuTtLe |
| 05:56.59 | JT | or is that Buttle? |
| 05:57.05 | [TK]D-Fender | drmessano: lol.. still remember that.. |
| 05:57.09 | *** join/#asterisk JCJC (n=JCJC@netblock-72-25-115-165.dslextreme.com) |
| 05:57.23 | Strom_M | JT: it's been confusion from the word 'go' |
| 05:58.19 | *** join/#asterisk ltd-- (n=z@patwk.transact.net.au) |
| 05:58.34 | *** join/#asterisk steliosk (n=Stelios@79.131.121.52) |
| 05:59.12 | drmessano | Tuttle |
| 06:00.03 | drmessano | That's almost as funny as when a friend and I used to go from IRC channel to IRC channel posting people's IP addresses we got from whois |
| 06:00.15 | drmessano | and people would freak out that we were "hacking" |
| 06:00.29 | drmessano | HOW DID YOU GET MY IP ADDRESS!!!!???!?!?!? ZOMGGGGG |
| 06:00.30 | Strom_M | drmessano: I think you missed the joke |
| 06:00.36 | drmessano | Completely |
| 06:01.39 | Strom_M | go watch "Brazil" |
| 06:02.21 | JT | Brazil is awesome |
| 06:02.35 | Strom_M | I want to talk to you about ducts. |
| 06:02.55 | Strom_M | JT: three years ago, one of the local cinemas showed a print of Brazil |
| 06:03.04 | JT | a print? |
| 06:03.11 | Strom_M | yes |
| 06:03.21 | Strom_M | i.e. prijected in a theater off 35mm :) |
| 06:03.27 | Strom_M | s/rij/roj/ |
| 06:03.41 | JT | "what do you blame the rise in terrorism on?" "bad sportsmanship" |
| 06:03.48 | JT | ah cool |
| 06:05.54 | Strom_M | yeah -- it was awesome |
| 06:06.18 | Strom_M | "My complication had a little complication." |
| 06:06.23 | [TK]D-Fender | Yay, just passed my retard upstream telco's bittorrent BW cap hours. |
| 06:10.31 | drmessano | ha nice |
| 06:10.53 | jblack | [tk] are you going to drop rst packets? |
| 06:11.04 | drmessano | you can forget the /etc/selinux/config file on CentOS 5.2 now |
| 06:11.15 | [TK]D-Fender | jblack: No, just throttled |
| 06:11.31 | [TK]D-Fender | drmessano: Oh? |
| 06:11.32 | drmessano | I just blanked out the one on my 5.1 > 5.2 box, added the grub.conf directive, and it took it |
| 06:12.00 | drmessano | by default, /etc/selinux/config is empty in 5.2 new installs |
| 06:12.04 | drmessano | and apparently is useless now |
| 06:12.25 | jblack | [TK]D-Fender: I think you misunderstood. You can fight against the most common type of throttling by firewalling off tcp rst packets. |
| 06:18.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 06:22.02 | TrentCreek | hey hey//I compiled * on CentOS |
| 06:22.48 | *** join/#asterisk nuonguy (n=john@c-24-6-187-202.hsd1.ca.comcast.net) |
| 06:23.48 | ThoMe | good morning ladys |
| 06:24.17 | drmessano | "ladies" |
| 06:24.32 | ThoMe | ups |
| 06:24.36 | ThoMe | drmessano: mh :) |
| 06:25.14 | *** join/#asterisk Hackbanger (n=hackbang@213.209.114.6) |
| 06:25.34 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:26.11 | drmessano | "oops" |
| 06:26.36 | drmessano | UPS is either Uninterruptible Power Supply or United Parcel Service |
| 06:27.12 | ThoMe | drmessano: Uninterruptable Power Supply :P |
| 06:27.17 | *** join/#asterisk admin0 (n=admin0@bb116-14-163-11.singnet.com.sg) |
| 06:27.20 | ThoMe | right right... |
| 06:27.34 | admin0 | like $10 for g729, does something like that also exist for g723.1 ? |
| 06:27.49 | drmessano | "Uninterruptible power supply" |
| 06:28.00 | drmessano | ible not able |
| 06:29.32 | creativx | idle not abel. |
| 06:29.45 | admin0 | i am on centos 5.1 64 bit os .. i have ipp installed, but cannot get it to work .. the module says " |
| 06:29.55 | admin0 | does not provide a license key on time of asterisk startup |
| 06:30.04 | admin0 | i have IPP non commercial key |
| 06:30.35 | drmessano | If you're using IPP, it sounds like you're compiling G729 yourself |
| 06:30.45 | admin0 | the docs point to 4.1 version of ipp, while there is nothing less than 5.0 |
| 06:30.57 | admin0 | tried the binary, tried compiling ..both cannot work |
| 06:31.10 | drmessano | Yeah, neither of those are supported here.. they're not legal |
| 06:31.15 | admin0 | "WARNING[20183]: loader.c:633 inspect_module: Module 'codec_g723-ast14-gcc4-glibc-x86_64-pentium4.so' does not provide a license key." |
| 06:31.30 | admin0 | yes .. for the g729, the $10 per channel is good .. the max i need is 2 channels |
| 06:31.37 | admin0 | does something like that also exist for the g723.1 ? |
| 06:31.46 | drmessano | No |
| 06:31.58 | [TK]D-Fender | admin0: There is no legal supported G.723 codec module for *. |
| 06:32.02 | drmessano | The only way to legally use G723 is the digium transcoder card |
| 06:32.15 | [TK]D-Fender | admin0: The only legit way is to but the TC400 transcoder card |
| 06:32.37 | [TK]D-Fender | hi-5's drmessano |
| 06:33.09 | drmessano | lol |
| 06:33.22 | drmessano | low-5's on the way back down, or something |
| 06:35.56 | TrentCreek | well then Fender...better get started for support |
| 06:39.10 | TrentCreek | then call it O723 |
| 06:41.50 | TrentCreek | pbx_dundi.c:445 reset_global_eid: No ethernet interface found for seeding global EID. You will have to set it manually. |
| 06:42.22 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 06:43.21 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:44.56 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-77-104.lns10.syd6.internode.on.net) |
| 06:52.19 | TrentCreek | nstalling Asterisk-addons (1.4.5 onwards) |
| 06:52.19 | TrentCreek | The instructions are exactly as above, with the exception that 'CFLAGS' has been replaced by 'ASTCFLAGS' in the 'Makefile' file. |
| 06:52.41 | *** join/#asterisk elver (n=Foo_Bar@ip134.cab48.lsn.starman.ee) |
| 06:52.48 | TrentCreek | I am glad they had that useful information AFTER I did the install |
| 06:53.48 | elver | I'm looking for a board that has a single RJ-45 and several RJ-11s, runs Linux (or BSD) and Asterisk and is capable of sending a VoIP stream to several regular phones and vice versa. And can route calls between the phones as well. Has anyone seen a board that can do this? Or does anyone have any knowledge of the kinds of chips that would be needed to control the RJ-11s as the hardware design companies I've approached have never worked with any |
| 06:54.13 | TrentCreek | www.ebay.com |
| 06:54.28 | drmessano | Sounds like you want a "PBX" |
| 06:54.44 | elver | I'm looking to get some modifications done and mass-produce. |
| 06:54.55 | elver | A board like that would be a great starting point. |
| 06:55.47 | drmessano | Well, generally a "board" doesn't run an OS |
| 06:56.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:56.16 | TrentCreek | uh oh...sounds like one of the forums where someone comes in late and asks "I need a help with a C++ program due in the morning..i'll be back later to get it..oh, and dont forget the comments" |
| 06:56.47 | drmessano | No, it sounds like someone wants to mass produce a PBX appliance without paying for R&D |
| 06:56.58 | *** join/#asterisk Bananaskin (n=mike@78-105-246-198.zone3.bethere.co.uk) |
| 06:57.26 | elver | I had a 2-hour meeting with an R&D company just yesterday. They don't know phone systems. |
| 06:57.27 | TrentCreek | OLD zaptel boards www.ebay.com |
| 06:57.30 | elver | I need some kinda starting point. |
| 06:57.49 | elver | System on chip with Linux, some flash, some ram, asterisk running on it, RJ-11s and RJ-45 |
| 06:57.49 | [TK]D-Fender | elver: www.voip-info.org <- go see what everyone ELSE is doing. |
| 06:57.50 | *** join/#asterisk bmg505 (n=leon@196-209-8-2-ndn-esr-2.dynamic.isadsl.co.za) |
| 06:58.01 | TrentCreek | why would someone deal with R&D company that knows nothing about the product you want? |
| 06:58.24 | elver | I didn't know that before I went for a meeting, now did I? |
| 06:58.25 | drmessano | I thought the whole point of R&D was to develop |
| 06:58.27 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 06:58.40 | drmessano | Not "I have this board....Lets make a case for it" |
| 06:59.02 | elver | I think you need a reality check. |
| 06:59.04 | TrentCreek | You mean like hiring someone to do a job, then finding out their first day of work they dont know anythign about doign the job? |
| 06:59.48 | drmessano | elver: It's not going to be as easy as a PBX-on-a-board.. if it was, I could get an EMERSON PBX at Wal-Mart |
| 07:00.06 | drmessano | You can start off with a micro-ATX board, add a telephony board, etc |
| 07:00.08 | unpaidbill | hey |
| 07:00.10 | unpaidbill | wow |
| 07:00.10 | TrentCreek | Or going to Subway, and after making a order, find out them they don't have hamburgers? |
| 07:00.16 | unpaidbill | shell crapped out, nm |
| 07:00.20 | elver | Getting an existing board as a basis would significantly reduce the unit price as well as reducing the R&D time by about 6 months. |
| 07:00.37 | admin0 | get asterisknow, load it in a cpu with intel board with inbuilt network, and get digium tdm cards .. and you get what you want .. a box with 1 network and several rj11 with web based control panel |
| 07:00.40 | drmessano | elver: good luck to you then |
| 07:01.07 | unpaidbill | asterisknow is good, but i wouldnt use it for anything production atm |
| 07:01.13 | JT | elver: *we* need the reality check? lol |
| 07:01.23 | JT | your expectations speak of unreality |
| 07:01.59 | elver | JT: I expect an ARM or MIPS based board that's got an RJ-11 socket. |
| 07:02.03 | elver | How is that unreal? |
| 07:02.05 | drmessano | JT: I am trying to develop a gaming console.. is there a sort of "Wii on a board" i can get to start my R&D with? |
| 07:02.05 | admin0 | unpaidbill, well there are other variations too. trixbox, freepbx, elastix etc |
| 07:02.14 | unpaidbill | i wouldnt use any of them, honestly |
| 07:02.21 | unpaidbill | asterisknow is the best of that bunch |
| 07:02.25 | JT | elver: a socket is a socket, you mean an RJ-11 with FXO or FXS functionality |
| 07:02.32 | unpaidbill | unless you dont want anything special/custom |
| 07:02.50 | admin0 | elver.. this is what you are looking for: http://trixbox.com/products/appliance |
| 07:02.50 | JT | elver: also it's unreal to expect asterisk to work in that function on ARM or MIPS |
| 07:02.58 | [TK]D-Fender | elver: There are boards like that. Go look at what OTHERS have done already. |
| 07:03.01 | JT | elver: have you done any research? |
| 07:03.16 | elver | [TK]D-Fender: where? |
| 07:03.23 | elver | JT: I'm doing research right now. |
| 07:03.26 | [TK]D-Fender | elver: www.voip-info.org <- go see what everyone ELSE is doing. |
| 07:03.27 | tzafrir_laptop | Asterisk should work well on ARM/MIPS . You just need to get it to build :-) |
| 07:03.47 | unpaidbill | i love asterisk on my amd geode :) |
| 07:03.50 | JT | elver: wouldn't it be a good idea to do research before meeting R&D companies? |
| 07:03.57 | drmessano | I'm still waiting for a link to my Wii-on-a-board |
| 07:03.59 | JT | tzafrir_laptop: what about zaptel? |
| 07:04.02 | unpaidbill | i can do 50 simultaneous calls (streaming audio without transcoding) |
| 07:04.06 | JT | unpaidbill: amd geode is x86 |
| 07:04.12 | JT | unpaidbill: not ARM or MIPS |
| 07:04.25 | unpaidbill | well, i was thinking embedded systems in general |
| 07:04.28 | unpaidbill | i didnt scroll up |
| 07:04.32 | unpaidbill | i'll shut up now |
| 07:04.33 | TrentCreek | ARM is up to 600Mhz |
| 07:04.33 | elver | Interesting. First you blame me for not meeting R&D companies that know about this stuff. Now you blame me for not knowing this stuff myself. |
| 07:04.41 | tzafrir_laptop | JT, while I haven't tried it myself, I know others have used Zaptel on ARM |
| 07:05.17 | JT | elver: i think you should know about basic asterisk platform compatibility before seeking out a suitable embedded board |
| 07:05.29 | drmessano | elver: Stop trolling and follow some links.. There is no pbx-all-in-one board out there, so you're going to have to do some research and some development here |
| 07:05.55 | drmessano | elver: [TK]D-Fender twice gave you a starting point |
| 07:05.56 | [TK]D-Fender | TrentCreek: You need to add "leg" and "thorax" for greater performance ;) |
| 07:05.56 | elver | JT: which is what I'm trying to learn here. |
| 07:06.07 | TrentCreek | oh yeah.... |
| 07:06.15 | [TK]D-Fender | elver: There is little to learn here about all of that. |
| 07:06.19 | TrentCreek | Zaptel can the the leg |
| 07:06.20 | unpaidbill | hey tell me how agi+meetme is supposed to work or asterisk sux |
| 07:06.21 | unpaidbill | thx |
| 07:06.25 | tzafrir_laptop | But someone without some Linux expeirnce should be willing to spend either more researerch time or research money on non-x86 platforms |
| 07:06.26 | [TK]D-Fender | elver: You've come to the wrong place. |
| 07:06.31 | JT | elver: you'll find if it's not x86, you'll have a hard time getting hardware support for asterisk unless you develop drivers yourself |
| 07:06.47 | tzafrir_laptop | TrentCreek, please report bugs |
| 07:06.52 | drmessano | We all run Asterisk on Compaq Deskpros.. We know nothing about MIPS, ARMS, FAPS, or WANG-DOODLES.. |
| 07:06.54 | admin0 | elver, to an end user, i don't see how your solution is going to sound better than the current appliances like asteriknow etc with the tdm card .. its all inside the box with a web gui and does what its supposed to do |
| 07:07.00 | TrentCreek | nstalling Asterisk-addons (1.4.5 onwards) |
| 07:07.00 | TrentCreek | The instructions are exactly as above, with the exception that 'CFLAGS' has been replaced by 'ASTCFLAGS' in the 'Makefile' file. |
| 07:07.03 | drmessano | Speaking of which, I need another 16MB of SDRAM |
| 07:07.21 | elver | admin0: let me worry about that. |
| 07:08.03 | tzafrir_laptop | TrentCreek, I know of no open issue of Zaptel specific to non-x86 platforms |
| 07:08.46 | TrentCreek | So it should work |
| 07:09.09 | tzafrir_laptop | Either that or nobody reports bugs |
| 07:09.24 | tzafrir_laptop | I suspect both |
| 07:09.25 | drmessano | elver: Try here ---> http://tinyurl.com/dabq4 |
| 07:09.50 | TrentCreek | so i have done all the research, so I will beat him to market..just 1 call to china and BAM they start working on my plans with ready to mass produce in 60 dauy |
| 07:10.46 | TrentCreek | Why am I working on this tiny Japanese keyboard, and dont even know Japanses? |
| 07:11.08 | unpaidbill | because you yearn to learn the kanji and get yourself some sweet asian tail? |
| 07:11.15 | unpaidbill | i dont blame you |
| 07:11.18 | unpaidbill | i have the fever too. |
| 07:11.19 | TrentCreek | hehe |
| 07:11.22 | drmessano | ^_^ |
| 07:11.38 | TrentCreek | Chimpokomon! |
| 07:11.55 | elver | drmessano: does stomping on people who know less about telephones than you really compensate for having a small dick? |
| 07:11.58 | unpaidbill | i dont know anything about that, but i do know hot japanese touristomon |
| 07:12.09 | unpaidbill | jesus elver |
| 07:12.24 | unpaidbill | do you really think that is going to work in your favor |
| 07:12.53 | elver | I know less than you guys here, but what the fuck is up with the attitude? |
| 07:12.53 | drmessano | Asian girls are like that.. You guys watch way too much hentai |
| 07:12.54 | unpaidbill | rule #1 of irc: if you get flamed, rtfm a lot, then come back and get flamed again, and rtfm some more |
| 07:12.57 | tzafrir_laptop | TrentCreek, you mean http://www.atcom.cn/En_products_IP04.htm ? |
| 07:13.05 | creativx | rtfm tilll you drop |
| 07:13.09 | unpaidbill | seriously |
| 07:13.32 | tzafrir_laptop | (which is the box designed by the astfin guys) |
| 07:13.40 | unpaidbill | i ask dumb shit all the time, if im ignored or shit on.. i assume my question was completely retarded and the answer i need is only a few pdfs away |
| 07:13.54 | unpaidbill | like my meetme agi question |
| 07:13.54 | TrentCreek | gosh darn it they beat me to a ALL IN ONE asterisk appliance! |
| 07:13.55 | unpaidbill | heh |
| 07:14.29 | drmessano | elver: You're a being a complete and total ass.. we've all posted links and given you info that "NO, THAT BOARD DOESNT EXIST EXCEPT IN UNICORN LAND", but it seems like youre just here to argue or with the hopes that someone is magically going to pull out the board you pay real money to develop out of their ass and hand you a link to it |
| 07:14.49 | drmessano | elver: Seriously, man.. |
| 07:14.55 | TrentCreek | The IP04 is a complete Asterisk powered IP-PBX with four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provide a solid, uniform platform for traditional PSTN communications as well as VoIP communications |
| 07:15.24 | jblack | Elver: What are you asking for help with? I can understand lack of experience. |
| 07:15.39 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:15.42 | JT | TrentCreek: damn they could've at least slipped "unified communications" into the blurb |
| 07:15.58 | elver | I think TrentCreek just pasted info about exactly what I'm looking for. |
| 07:16.02 | drmessano | JT: It lacks an SMTP connector |
| 07:16.15 | jblack | ok |
| 07:16.24 | elver | bows to TrentCreek |
| 07:16.35 | elver | Exactly what I was looking for. |
| 07:16.35 | JT | didn't tzafrir_laptop paste it? |
| 07:16.38 | TrentCreek | well...you know those chinese..lacking english ability |
| 07:16.50 | TrentCreek | yes he did |
| 07:17.04 | jblack | TrentCreek: Gotta love engrish.com :) |
| 07:17.36 | TrentCreek | haha...sounds like like Korean |
| 07:18.29 | drmessano | So, he wants a complete PBX then.. |
| 07:18.31 | *** join/#asterisk icenicola (n=pavlovni@83.244.78.241) |
| 07:18.44 | jblack | "I am a moody girl. It gets angry or laughs suddenly". |
| 07:18.56 | jblack | what can you say to that? "It puts the lotion on it's body?" |
| 07:19.01 | icenicola | Hello. any guy familiar with BATM A211N sip devices? |
| 07:20.33 | [TK]D-Fender | icenicola: I get all of *2* Google links looking that up |
| 07:20.44 | [TK]D-Fender | icenicola: I'd say your odds are bleak |
| 07:20.44 | jblack | http://engrish.com/recent_detail.php?imagename=does-it-end.jpg&category=Computer&date=2008-06-11 |
| 07:21.34 | creativx | haha |
| 07:21.51 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:23.05 | icenicola | Fender: what do u mean by my odds are bleak? |
| 07:23.36 | icenicola | i want to write a cfg file for auto provisioning |
| 07:24.04 | icenicola | i need help in writing one |
| 07:24.16 | TrentCreek | rentacoder.com |
| 07:26.19 | [TK]D-Fender | icenicola: It means GOOLGE doesn't know about your product so you're probably screwed getting any help with it. |
| 07:26.35 | [TK]D-Fender | GOOGLE* |
| 07:27.36 | icenicola | Fender: that's y i came here to ask :) |
| 07:27.47 | *** join/#asterisk lordmortis (n=lordmort@203.8.160.250) |
| 07:27.48 | TrentCreek | and one is in hebrew |
| 07:27.48 | unpaidbill | there's nothing like a diy pbx to make you want to go down the street, not across it! |
| 07:28.31 | [TK]D-Fender | icenicola: I'd say "goo luck", but I think you'll need a lot more than just luck |
| 07:28.42 | unpaidbill | although goo is pretty awesome! |
| 07:28.48 | icenicola | no problem with hebrew language |
| 07:29.10 | TrentCreek | seems Sunrocket used them |
| 07:29.24 | TrentCreek | but they are closed. |
| 07:29.24 | unpaidbill | i used sunrocket before.. they were not so hot |
| 07:29.26 | icenicola | Fender: thank u anyway |
| 07:29.38 | [TK]D-Fender | icenicola: If you're considering buying it, I'd say "don't", if you've already bought it I'd say "try to return it", and if you've already bought it and can't return it I'd say "At least you have a new door-stop" |
| 07:29.40 | TrentCreek | Hence why Sunrocket closed |
| 07:29.47 | unpaidbill | yeah trent, no doubt |
| 07:29.53 | icenicola | do u have by any change a cfg file regarding sunrocket? |
| 07:29.54 | unpaidbill | i got two uniden out of them though |
| 07:29.57 | unpaidbill | so i cant complain |
| 07:29.58 | icenicola | might help me |
| 07:30.21 | TrentCreek | you are better off buying a NEW and uNLOCKED box for less than $50 that is highly supported |
| 07:30.51 | icenicola | unfortunately, i am stuck with them now |
| 07:31.05 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 07:31.05 | drmessano | How many do you have? |
| 07:31.08 | Alpha_AI | Hello |
| 07:31.13 | TrentCreek | i got one that is two lines and supports two providers |
| 07:31.19 | icenicola | i just need a sample cfg that will apply so i can write my own |
| 07:31.23 | Alpha_AI | does anyone know where i can find a good speech recognition module for asterisk |
| 07:31.27 | drmessano | icenicola: How many do you have? |
| 07:31.28 | icenicola | around 80 devices |
| 07:31.32 | drmessano | Geez |
| 07:31.38 | Alpha_AI | ive heard of lumenvox but it costs heaps |
| 07:31.47 | TrentCreek | well..there is use...get the silver out of them |
| 07:31.49 | drmessano | I seriously, seriously doubt you'll find that data |
| 07:31.51 | icenicola | drmessano: I know :S |
| 07:32.05 | [TK]D-Fender | icenicola: Next time, do your research first. |
| 07:32.14 | drmessano | Why did you get 80 of them? |
| 07:32.21 | unpaidbill | alpha: you have three choices, the best and most expensive is not for asterisk, it is nuance. the second and not so expensive that has an asterisk app is lumenvox (these guys are good!) and the third and free is sphinx, which is completely free but you have to do it all yourself, and it's not exactly a walk in the park |
| 07:32.25 | TrentCreek | to learn a lession |
| 07:32.32 | icenicola | Fender: beleive me, i did my research |
| 07:32.45 | drmessano | icenicola: Apparently not |
| 07:32.53 | icenicola | did u find anything? |
| 07:33.05 | drmessano | icenicola: There's NO provisioning info, and you will likely not find any |
| 07:33.05 | icenicola | coz i find sth related to linksys but encryoted |
| 07:33.09 | unpaidbill | sphinx is good to start out with, because you are forced to learn a bunch of bullshit that you probably dont care about but will help you out in the long run |
| 07:33.26 | icenicola | encrypted* |
| 07:33.32 | drmessano | icenicola: A provisioning file is unique to the box |
| 07:33.42 | drmessano | icenicola: There is no such thing as a generic provisioning file |
| 07:33.43 | TrentCreek | icenicola: you may want to do a FCC search and see who makes it and see if it has an alternative model number |
| 07:34.12 | icenicola | neither a sample wher i can start from? |
| 07:34.21 | drmessano | icenicola: You need an XML config for that specific manufacturer and model.. |
| 07:34.27 | drmessano | icenicola: There is no "start" |
| 07:34.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:34.34 | drmessano | icenicola: You need a BATM config |
| 07:34.49 | drmessano | icenicola: Not just some random generic XML |
| 07:34.49 | unpaidbill | it's just you against the world, baby! (poolhall junkies quote) |
| 07:35.20 | drmessano | icenicola: That's what you're missing here.. Every manufacturer uses different provisioning methods.. |
| 07:35.50 | icenicola | ok, i got it |
| 07:36.29 | drmessano | icenicola: That's like asking me for a config for some random linux app you downloaded, and said "Any config will do" so I sent you one for MySQL |
| 07:36.42 | drmessano | It's that random |
| 07:37.04 | DigitalIrony | heh |
| 07:37.06 | icenicola | i understand now |
| 07:37.08 | DigitalIrony | don't be mean |
| 07:37.12 | drmessano | I'm not |
| 07:37.12 | DigitalIrony | lol |
| 07:37.14 | drmessano | Not at all |
| 07:37.21 | DigitalIrony | your right |
| 07:37.43 | icenicola | deltahtree used to have the BATM devices and had cfg files |
| 07:37.53 | icenicola | but i think they r closed, same as sunrocket |
| 07:38.14 | TrentCreek | have the Sunrocket A211N gizmo, and I was able to login via the "18," password (my firmware version was 4.60.xx). Once logged in, I upgraded to the 4.62.14 firmware and everything still works fine. |
| 07:38.30 | TrentCreek | QUOTED |
| 07:38.36 | icenicola | we have the admin password for the device |
| 07:38.54 | *** join/#asterisk Corydon76-dig (i=pink@pdpc/supporter/bronze/Corydon76-home) |
| 07:38.54 | *** join/#asterisk ltd-- (n=z@patwk.transact.net.au) [NETSPLIT VICTIM] |
| 07:38.54 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.115) [NETSPLIT VICTIM] |
| 07:38.54 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) [NETSPLIT VICTIM] |
| 07:38.54 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 07:38.55 | *** join/#asterisk kippi (n=kippi@untrust-gct.equinoxit.net) [NETSPLIT VICTIM] |
| 07:38.55 | *** join/#asterisk nn (n=nn@unaffiliated/nn) [NETSPLIT VICTIM] |
| 07:38.55 | *** join/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 07:38.56 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) [NETSPLIT VICTIM] |
| 07:38.56 | *** join/#asterisk cy3o3 (n=cy@it.was.otherkids.net) |
| 07:38.56 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 07:38.56 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
| 07:38.56 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) [NETSPLIT VICTIM] |
| 07:38.56 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) [NETSPLIT VICTIM] |
| 07:38.56 | *** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net |
| 07:38.56 | icenicola | it works fine so far |
| 07:39.03 | *** join/#asterisk Alpha_AI (n=Ben@iti142937-1.gw.connect.com.au) |
| 07:39.28 | icenicola | i did setup a tftp server on a linux server and all i want now is to write an xml file for those |
| 07:39.36 | icenicola | BATM devices |
| 07:40.36 | [TK]D-Fender | icenicola: And who said they COULD be provisioned? And who said by XML? delivered via TFTP? What bout format of the file, and naming conventions? Without a real manual you are completely screwed. |
| 07:41.07 | icenicola | i checked the data sheet for these devices |
| 07:41.19 | TrentCreek | yeah especially since there is zero info about it on the internet |
| 07:41.20 | icenicola | they support auto provisioning either using http of tftp |
| 07:41.35 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 07:42.01 | [TK]D-Fender | icenicola: Yay, 1 step down, a few more critical ones to go. |
| 07:42.16 | icenicola | sunrokcet and deltathree used them before |
| 07:42.22 | drmessano | I really don't think he understands |
| 07:42.31 | DigitalIrony | me either |
| 07:42.40 | icenicola | anyway guys |
| 07:42.50 | icenicola | thank u for the info |
| 07:42.54 | drmessano | Someone give him an XML for a polycom |
| 07:43.00 | drmessano | It's an XML, afterall |
| 07:43.19 | icenicola | it's ok guys, i will c what i can do |
| 07:43.19 | drmessano | 220, 221.. whatever it takes |
| 07:43.23 | icenicola | thanks a lot |
| 07:44.29 | drmessano | One of the best movie lines ever |
| 07:44.43 | Alpha_AI | unpaidbill, are u there? |
| 07:45.01 | unpaidbill | yes, i was retrieving pizza from the oven |
| 07:45.18 | unpaidbill | sphinx and lumenvox work much in the same way (at least last time i used lumenvox, 3 years ago) |
| 07:45.38 | unpaidbill | lumenvox is just much cleaner and much easier |
| 07:45.48 | unpaidbill | http://www.voip-info.org/wiki-Sphinx |
| 07:45.55 | unpaidbill | that will get you started with sphinx |
| 07:46.21 | unpaidbill | the first time you attempt this dont expect to have it working quickly or how you want it to work |
| 07:46.50 | unpaidbill | once it is working though, it's pretty nice. what it wont do: realtime recognition |
| 07:47.07 | drmessano | Flite is great too.. if you plan to use 1.4 forever |
| 07:47.23 | unpaidbill | if you need that, just go with nuance, or you'll have to do a lot of work yourself to have realtime recognition |
| 07:48.27 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:48.39 | dandre | Hello, |
| 07:49.01 | unpaidbill | speech recognition, not TTS |
| 07:49.41 | drmessano | Yeah, it's late.. I was thinking the other way |
| 07:50.30 | dandre | I had the opportunity to test a siemens gigaset c470IP but when I do a transfer the phone rings back . How can I fix the issue? |
| 07:53.12 | unpaidbill | how is flite compared to festival? |
| 07:53.24 | unpaidbill | i have used festival a bunch, never heard of or tried flite though |
| 07:53.53 | WildPikachu | seems this atcom phone is sending through a flash/hook every 1 min 20 seconds nearly on the dot on only outbound calls through my SIP PBX, anyone seen this? |
| 07:56.55 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:00.59 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:16.38 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 08:17.28 | ThoMe | is this the best solution for check for a available channel? exten => _0.,n,Chanisavail(IAX2/1 & IAX2/2 & IAX2/3 & IAX2/4 IAX2/5) |
| 08:17.34 | ThoMe | or can i this with a group? |
| 08:17.38 | Alpha_AI | unpaidbill, so for over the phone speech recognition, its best to use nuance? |
| 08:17.52 | ThoMe | with mISDN example: exten => _XX.,1,Dial(misdn/g:isdn-ext/${EXTEN}) |
| 08:17.53 | ThoMe | g = group |
| 08:17.55 | ThoMe | but for iax? |
| 08:18.18 | *** join/#asterisk liri (n=liran@bzq-179-128-126.static.bezeqint.net) |
| 08:28.45 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:28.50 | DigitalIrony | ThoMe: exten => _XX.,1,Dial(iax/gx/${EXTEN}) where x is the group number |
| 08:30.10 | DigitalIrony | Im sorry, that wrong |
| 08:30.20 | DigitalIrony | that would be for zap |
| 08:30.23 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:35.02 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:36.14 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 08:44.16 | *** join/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
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| 09:03.15 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 09:11.07 | dandre | I had the opportunity to test a siemens gigaset SIP/DECT c470IP but when I do a transfer the DECT phone rings back . How can I fix the issue? |
| 09:11.07 | dandre | I have read the logs but i don't understand them |
| 09:36.13 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:36.34 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 09:36.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 09:50.42 | *** join/#asterisk thomas (i=tm@tm.muc.de) |
| 09:50.44 | thomas | Re |
| 09:51.06 | thomas | is it posible IAX modems in a Group? |
| 09:51.16 | thomas | like misdn: misdn/g:bla ? |
| 09:58.34 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 09:59.23 | joobie | hey guys.. i have a polycom 320/330 ip phone that supports POE.. just wondering if anyone can recommend a cheap device that can put power over etherenet? I dont wantt o fork out for a full poe switch yet.. just want to test for 2 phones.. |
| 09:59.46 | joobie | i saw a few on dlink / linksys's website.. but not sure which model i need |
| 10:00.10 | dandre | In my DECT C470 testing, I have difficulties to have transfer functionality: I include full trace with sip debug set just before the effective transfer: |
| 10:00.10 | dandre | ext 43 is a sip phone that calls ext 51 (the DECT SIP Phone). Then ext 51 calls ext 50 (a Zap Phone) which answer the call and then ext 51 do the transfer. a short time after, ext 51 rings again . What is the trouble? |
| 10:00.23 | dandre | http://pastebin.com/d20f2d093 |
| 10:00.56 | gr0mit | aaah gday joobie |
| 10:01.32 | joobie | hey gr0m |
| 10:01.38 | joobie | sup |
| 10:01.55 | gr0mit | the sun shines here in UK |
| 10:02.04 | gr0mit | 22 degrees outside! |
| 10:02.11 | joobie | that would explain why the moon shines here:P |
| 10:02.13 | joobie | nice |
| 10:02.16 | gr0mit | hehe |
| 10:02.43 | gr0mit | considers joining the Flat Earth Society |
| 10:02.52 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 10:07.02 | *** join/#asterisk lordmortis (n=lordmort@117.102.157.107) |
| 10:13.07 | thomas | have: exten => 8709532,n,Chanisavail("IAX2/fax-IAX1") |
| 10:13.07 | thomas | exten => 8709532,n,Set(FREECHAN=${CUT(AVAILCHAN,,1&2)}) |
| 10:13.20 | thomas | i WOULD like from Chanisavail only "fax-IAX1" |
| 10:13.30 | thomas | but: NoOp("SIP/8709532-b6c01c98", "IAX2/fax-IAX1") in new stack |
| 10:25.48 | *** join/#asterisk ludan (n=daniele@192.167.215.122) |
| 10:28.22 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 10:29.08 | ludan | <PROTECTED> |
| 10:29.14 | ludan | i keep getting this msg, why? |
| 10:29.26 | tzafrir_laptop | ludan, do you use freepbx? |
| 10:29.52 | ludan | what is freepbx? |
| 10:29.54 | tzafrir_laptop | never mind |
| 10:30.17 | ludan | toll free number?? |
| 10:30.21 | ludan | that thing? |
| 10:30.27 | tzafrir_laptop | you get that message every time you run 'asterisk -rx' (or end a session of asterisk -r) |
| 10:30.47 | ludan | yes I use asterisk -r to reconnect |
| 10:30.57 | ludan | but i get a lot of this stuff once i'm connected |
| 10:31.14 | tzafrir_laptop | asterisk -r connects to the main asterisk process through something called a "unix-domain socket" |
| 10:31.26 | ludan | i know, through a socket |
| 10:31.56 | ludan | but i don't understand why i keep getting this message once i'm in |
| 10:32.08 | ludan | what does it mean? that i'm connect and quickly disconnected/ |
| 10:33.10 | ludan | uhm |
| 10:33.19 | liri | hey tzafrir |
| 10:33.31 | ludan | tzafrir_laptop: you know what I mean? |
| 10:36.06 | ludan | http://lists.digium.com/pipermail/asterisk-dev/2003-April/000512.html |
| 10:36.14 | ludan | it is like this but i've no crontab :S |
| 10:38.16 | *** join/#asterisk kombi (n=kombi@port-87-234-216-47.static.qsc.de) |
| 10:38.52 | tzafrir_laptop | that's old. It's from 5 years ago |
| 10:39.06 | kombi | what is wrong when I keep getting voicemail instead of the extension? |
| 10:39.14 | *** part/#asterisk RoyK (n=roy@ip-137-59-149-91.dialup.ice.no) |
| 10:39.28 | ludan | tzafrir_laptop: i know but the behavior is the same |
| 10:40.06 | tzafrir_laptop | liri, hi |
| 10:40.41 | seanbright | ludan: something or someone is connecting to asterisk via 'asterisk -r' |
| 10:40.49 | ludan | not possible |
| 10:40.51 | kaldemar | kombi: your dialplan. |
| 10:40.55 | seanbright | ludan: yes, it is possible |
| 10:41.00 | seanbright | ludan: cuz it's happening |
| 10:41.15 | seanbright | see how that works? something is happening, therefore it is possible. |
| 10:41.16 | ludan | i'm the only one logged into the machine |
| 10:41.23 | seanbright | ludan: than it's a script |
| 10:41.27 | seanbright | ludan: or the underpants gnomes |
| 10:41.36 | seanbright | but *something* is doing it |
| 10:41.51 | ludan | there are no processes doing funny business |
| 10:41.57 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 10:42.00 | seanbright | sigh |
| 10:42.12 | seanbright | ok, well how about this... |
| 10:42.17 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 10:42.36 | seanbright | ludan: the message " -- Remote UNIX connection disconnected" means that someone or something has connected to asterisk via the '-r' flag |
| 10:42.44 | seanbright | ludan: and that is all we can tell you |
| 10:42.49 | kombi | kaldemar: I have this new number from voovox, it actually arrives here, but only voicemail comes on. The line is "exten => 30873333896,5,GoTo(bureau,51,1)". bureau 51 is happily working, as is everything else.. |
| 10:42.54 | ludan | seanbright: thanks |
| 10:42.54 | tzafrir_laptop | Or directly to the astrerisk.ctl socket |
| 10:42.58 | seanbright | or that |
| 10:43.07 | ludan | what is the .ctl socket? |
| 10:43.22 | ludan | i mean, from where you can connect through it? |
| 10:43.42 | seanbright | tzafrir_laptop: you just had to go and confuse the guy, huh? ;-) |
| 10:43.45 | kaldemar | kombi: are you sure the call hits the right extension? |
| 10:44.08 | seanbright | ludan: it's what asterisk -r uses to communicate to the asterisk daemon process |
| 10:44.18 | seanbright | ludan: a special file, i think in /var/run somewhere |
| 10:44.33 | ludan | Your problem was that message "Remote UNIX connection" keep showing up, this is because scipt keep checking is asterisk working buy connecting at console with "asterisk -r". |
| 10:44.36 | kaldemar | kombi: check the incoming context of the voovox peer and start going through the dialplan. looking at the CLI prints with verbose 10 won't hurt either. |
| 10:44.41 | ludan | touche |
| 10:44.58 | seanbright | ludan: huh? |
| 10:45.09 | kombi | kaldemar: I cannot quite tell, all I see is "mISDN/1-u26 is ringing" and "mISDN-1u26 answered.." |
| 10:45.22 | ludan | seanbright: i read that on a forum |
| 10:45.58 | seanbright | ludan: ok, but you were very clear with us that no script or process was "doing funny business" |
| 10:46.16 | ludan | seanbright: do I know that asterisk_safe is a script running the server? |
| 10:46.19 | ludan | sorry eh |
| 10:46.25 | seanbright | safe_asterisk, you mean? |
| 10:46.29 | ludan | but i just did /etc/init.d/asterisk restart |
| 10:46.42 | ludan | and this script was called (but i didn't know) |
| 10:46.47 | kaldemar | kombi: "set verbose 10" in cli will show you more output. |
| 10:47.02 | kombi | kaldemar: got that on 1000 ..;) |
| 10:47.07 | ludan | seanbright: now i stopped asterisk and restart it again and the msg doesn't show up anymore |
| 10:47.16 | seanbright | ludan: ok, great. |
| 10:47.25 | kaldemar | kombi: are you using sip or iax? |
| 10:47.26 | ludan | seanbright: thanks, sorry for the disturb |
| 10:47.34 | seanbright | ludan: no sweat, that's kinda the reason we're all here |
| 10:47.37 | kombi | kaldemar: iax for voovox |
| 10:47.52 | ludan | seanbright: all right |
| 10:48.00 | kaldemar | kombi: pastebin your iax.conf and extensions.conf |
| 10:49.46 | kaldemar | kombi: and is bureau an external system with the voicemail and extensions to which you connect with BRI? |
| 10:49.48 | *** join/#asterisk angryuser (n=sdfsdf@78.115.250.180) |
| 10:51.37 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 10:53.13 | kombi | kaldemar: bureau is internal, http://pastebin.se/195106 |
| 10:54.53 | kaldemar | internal as in a phone directly connected to asterisk? |
| 10:55.05 | kombi | kaldemar: yip! |
| 10:56.01 | kaldemar | i don't see a context line anywhere |
| 10:56.04 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 10:56.58 | kaldemar | your inbound calls land to [default] without it, not to [voovox]. add context=voovox in iax.conf under [voovox]. |
| 10:57.31 | kombi | kaldemar: that must be it!!! I'll try that.. |
| 10:58.15 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 10:59.55 | nextime | hello, in * 1.4, using users.conf to add users from ajam, when i add a user i get automagically an hint and a dial command ( for users without voicemail ) in the dialplan. Is there a way to add some options to the dial command ( i need to add a timeout )? |
| 11:00.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 11:01.14 | kombi | kaldemar: hmm, still no luck. What's the pattern again to accept any number of any length? |
| 11:01.29 | kaldemar | kombi: _X. |
| 11:01.39 | tzafrir_laptop | any number except 1 |
| 11:01.48 | tzafrir_laptop | except 1 digit, that is |
| 11:01.49 | kaldemar | well that requires a minimum of 2 numbers |
| 11:02.05 | kombi | good enough for me anyway..;) |
| 11:04.10 | *** part/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 11:07.47 | thomas | tzafrir_laptop: aloha! |
| 11:08.07 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 11:10.48 | *** join/#asterisk LuisTorres (n=luistorr@bl9-248-212.dsl.telepac.pt) |
| 11:11.43 | kombi | hmm, still not.. this should be enough to make it work, now? -> http://pastebin.se/195107 |
| 11:13.56 | kaldemar | well, if voovox matches to that peer and bureau,51,1 works, yes. |
| 11:14.30 | kombi | strange... |
| 11:14.53 | kaldemar | a cli output of a call would be most helpful. |
| 11:15.06 | kombi | just a second.. |
| 11:15.17 | masus | The System (Asterisk 1.4) which we used effeciently before doesnt work after the agents logged in with "AGENTLOGIN", for example *reload* or *agent show* don't respond and locks the CLI and we get this message "The previous reload command didn't finish yet" Does Anyone have an idea. Thanks. |
| 11:21.18 | kombi | kaldemar: I stand corrected... call does NOT hit my box, I was deceived by a line saying "answering" when making the test call from that same box. Found out by calling from my cell in order to give you uncluttered CLI output.. The voicemail I hear is asterisk's vm, but their's..;) sorry about that |
| 11:28.07 | *** join/#asterisk ^shark_ (n=Anju_Kho@41.222.2.65) |
| 11:28.32 | tzafrir_laptop | thomas, hi |
| 11:29.05 | ^shark_ | hi friends, i am looking for * billing software for each of my phone extensions in the pbx |
| 11:29.43 | thomas | any ideas whiy this: [Jul 1 13:28:46] NOTICE[5550]: chan_sip.c:14035 handle_request_invite: Call from '2231475' to extension '8709532' rejected because extension not found. |
| 11:29.46 | thomas | ? |
| 11:30.00 | ^shark_ | when i get credit from the telco, i need to share this credit among phone extensions, what sort of software can let me do this? |
| 11:30.02 | thomas | have two sip accounts in the sip.conf |
| 11:30.06 | thomas | context=eingehend-freeline-fax |
| 11:30.07 | thomas | and |
| 11:30.10 | thomas | context=eingehend-freeline-fax-temp-aktion |
| 11:31.11 | kaldemar | the notice is quite self-explanatory |
| 11:31.55 | kaldemar | there is no exten line matching 8709532 in the context the call landed in. |
| 11:32.12 | thomas | kaldemar: em |
| 11:32.13 | thomas | mom |
| 11:33.05 | thomas | kaldemar: http://paste.keks.be/58 |
| 11:35.17 | thomas | kaldemar: if i add the ";" before register => 2231475:****@sipgate.de/2231475 |
| 11:35.20 | thomas | then works it |
| 11:36.04 | kaldemar | pastebin [eingehend-freeline-fax] and [eingehend-freeline-fax-temp-aktion] from extensions.conf and a cli output of a failed call with "set verbose 10" |
| 11:36.46 | thomas | +core ;) |
| 11:37.28 | thomas | kaldemar: http://paste.keks.be/59 |
| 11:38.16 | thomas | and http://paste.keks.be/60 |
| 11:38.38 | kaldemar | looks like if you register 2231475 at sipgate, calls from them land in [eingehend-freeline-fax-temp-aktion] that doesn't have 8709532 in it. |
| 11:39.07 | thomas | but only if i have two register-lines |
| 11:39.12 | thomas | if one then i have no problems. |
| 11:39.29 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:41.12 | thomas | kaldemar: i thin, because connect to sipgate.de - |
| 11:41.14 | thomas | :/ |
| 11:41.18 | *** join/#asterisk trnzmeta (n=bleh@123-243-201-39.static.tpgi.com.au) |
| 11:45.03 | ^shark_ | hi friends, i am looking for * billing software for each of my phone extensions in the pbx |
| 11:45.48 | seanbright | ~billing |
| 11:45.54 | seanbright | oh well |
| 11:49.02 | ^shark_ | has been looking around for days for * billing software but he can't find! He wants to credit his phone extensions with certain amount of credit but doesnt know how!? |
| 11:49.15 | seanbright | ~a2billing |
| 11:49.20 | seanbright | oh well |
| 11:49.21 | seanbright | :) |
| 11:49.57 | ^shark_ | is confused! |
| 11:50.01 | seanbright | me too |
| 11:50.06 | seanbright | goes to get his car washed |
| 11:51.47 | gr0mit | ^shark_, i looked at lots of things |
| 11:51.53 | gr0mit | we ended up writing our own |
| 11:53.20 | ^shark_ | gr0mit: is there any you think would just do the job for me? To credit each extension |
| 11:53.50 | gr0mit | well it is non-trivial |
| 11:54.10 | gr0mit | coz you have to know the rate for each call |
| 11:54.29 | gr0mit | and then set a cutoff to terminate the call X seconds later |
| 11:54.43 | *** join/#asterisk Unlockgod (n=ben@host81-148-207-112.in-addr.btopenworld.com) |
| 11:54.46 | Unlockgod | Hey there |
| 11:55.02 | ^shark_ | gr0mit: i would die to get my hands on that sort of software. |
| 11:55.05 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 11:55.30 | gr0mit | well for callshops there is a lot of stuff about |
| 11:55.48 | gr0mit | but i found it was very complex to configure |
| 11:57.07 | Dr-Linux|home | gr0mit: what module number should i buy, i need single span T1 card |
| 11:57.47 | gr0mit | well, I would probably use a Sangoma A10x |
| 11:57.52 | gr0mit | A101 i think |
| 11:58.22 | *** join/#asterisk awk (n=awk@security.web.za) |
| 12:00.18 | Unlockgod | is anyone able to help with trixbox problems? within asterisk please |
| 12:01.34 | *** join/#asterisk cjk (n=cjk@vodsl-11071.vo.lu) |
| 12:01.36 | Dr-Linux|home | gr0mit: I agree, but this company wants Digium |
| 12:01.45 | gr0mit | Dr-Linux|home, why? |
| 12:02.58 | cjk | hi; i programmed a button on my users phone that they can push if audio has drops. so basically they raise an alert during a call and at the moment 10% of all calls are rated bad. Any ideas where I should start looking? |
| 12:03.17 | *** part/#asterisk ctooley (n=ctooley@209.33.108.119) |
| 12:03.25 | Dr-Linux|home | gr0mit: because they are already using digium cards |
| 12:03.33 | gr0mit | cjk, look at packet loss! |
| 12:03.57 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 12:04.17 | ^shark_ | ~cdr |
| 12:04.17 | jbot | cdr is probably Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 12:04.18 | gr0mit | well i have not bought any Digium cards for a very long time. Had loads of IRQ related issues |
| 12:04.28 | *** join/#asterisk zeeqy (n=zeeqy@196-209-150-205-tbnb-esr-3.dynamic.isadsl.co.za) |
| 12:04.56 | ^shark_ | ~mysql cdr |
| 12:05.48 | zeeqy | hi all:...i was wondering if anyone successfully installed Asterisk on ubuntu server 8.04, I would really appreciate a little help |
| 12:06.00 | gr0mit | zeeqy, yes |
| 12:06.11 | gr0mit | we have done that |
| 12:06.20 | gr0mit | what is the prob? |
| 12:06.36 | zeeqy | i m running ubuntu server 8.04 the 64 bit version...thanks gr0mit |
| 12:06.41 | gr0mit | hmmm |
| 12:06.49 | gr0mit | never used 64 bit version |
| 12:07.21 | zeeqy | i tried on the ubuntu desktop edition but they say I must have the LAMP server installed which only comes with server |
| 12:07.43 | gr0mit | are you compiling from source? |
| 12:08.02 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:08.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:08.03 | gr0mit | you should not need AMP for asterisk |
| 12:08.04 | zeeqy | no |
| 12:08.25 | gr0mit | well, compiling from source is the most reliable way to build your box |
| 12:08.30 | thomas | is it not posible multiple sip accounts on the same provider to have? |
| 12:08.48 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:08.53 | gr0mit | thomas, depends on the provider! |
| 12:09.03 | thomas | gr0mit: hm. |
| 12:09.16 | zeeqy | gr0mit: here is what I want ...I can install asterisk core from Synaptic...but the reall problem is the sql and AMP portal integration |
| 12:09.25 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 12:09.27 | thomas | gr0mit: is it posible two connections with two ip-adresses? have multiple ips on my box. |
| 12:09.35 | thomas | gr0mit: externip for each sip account? |
| 12:09.54 | gr0mit | thomas, what exactly are you trying to achieve? |
| 12:10.15 | thomas | gr0mit: hm. i would like two connections to my sip provider |
| 12:10.24 | thomas | but if i have two accounts then i have an error |
| 12:10.37 | *** part/#asterisk ^shark_ (n=Anju_Kho@41.222.2.65) |
| 12:10.39 | thomas | any ideas whiy this: [Jul 1 13:28:46] NOTICE[5550]: chan_sip.c:14035 handle_request_invite: Call from '2231475' to extension '8709532' rejected because extension not found. |
| 12:10.43 | gr0mit | can you not ask them for two accounts? |
| 12:10.59 | thomas | gr0mit: if i have only one account then i have no problems |
| 12:11.16 | thomas | gr0mit: i have two accounts |
| 12:11.19 | gr0mit | so what error do you get? |
| 12:11.29 | thomas | gr0mit: http://paste.keks.be/58 |
| 12:11.32 | thomas | my config |
| 12:11.42 | thomas | gr0mit: my error: http://paste.keks.be/60 |
| 12:11.51 | zeeqy | gr0mit: there is a how to -on ubuntu site but its a draft and my installation stuck after a few instructions...here is the link : https://wiki.ubuntu.com/AsteriskOnUbuntu#head-d84bca9e76d0788bf6f2b2c7a711893b8247d40b |
| 12:12.43 | gr0mit | ok well this is a prob with your extensions.conf file |
| 12:12.56 | thomas | gr0mit: with my extentions.conf ? |
| 12:13.13 | gr0mit | the call comes in to 8709532 but looks like you don't have a matching extension for that |
| 12:13.18 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:13.32 | thomas | gr0mit: my extentions: http://paste.keks.be/61 |
| 12:13.59 | thomas | chan_sip.c:14035 <<the sip error or?! |
| 12:15.06 | zeeqy | gr0mit....I will wait...let thomas get some help first...thanks |
| 12:16.40 | thomas | gr0mit: ideas? ;) |
| 12:17.10 | gr0mit | thomas, am looking at your config |
| 12:17.12 | gr0mit | 1 sec |
| 12:17.14 | thomas | :) |
| 12:18.13 | kombi | with a b410p, do I still need ztdummy? |
| 12:18.20 | gr0mit | you sure you did a sip reload and a reload ? |
| 12:18.22 | JT | no |
| 12:18.23 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 12:18.27 | teknoprep | hey all |
| 12:18.33 | thomas | JT: you mean me? |
| 12:18.37 | thomas | gr0mit: jep. reload |
| 12:18.40 | thomas | gr0mit: restart.. etc |
| 12:18.41 | gr0mit | and sip reload |
| 12:18.43 | gr0mit | aah ok |
| 12:18.48 | thomas | ep |
| 12:19.02 | teknoprep | i am looking for a good desktop application for asterisk that allows for ... Channel Barge , monitoring of phone calls , recording , transfering , etc |
| 12:19.18 | teknoprep | other than hudlite as its the worst peice of software i have ever used |
| 12:19.40 | thomas | gr0mit: have now a solution... |
| 12:19.50 | gr0mit | thomas, good |
| 12:19.56 | thomas | gr0mit: mom |
| 12:20.03 | gr0mit | i think if you call not from your Asteirsk box it will work |
| 12:20.14 | kombi | JT: how do I use the card's timer I wonder.. It's in there but ztdummy is still loaded |
| 12:20.17 | gr0mit | i think the prob is you are trying to call from one extension to another |
| 12:20.22 | gr0mit | without handling it correctly |
| 12:20.45 | thomas | gr0mit: sip.conf: http://paste.keks.be/62 |
| 12:21.17 | gr0mit | well careful |
| 12:21.18 | thomas | gr0mit: extentions.conf: http://paste.keks.be/63 |
| 12:21.57 | gr0mit | coz your incoming calls from both numbers will noe arrive in the same context |
| 12:22.08 | gr0mit | but if it works for you. |
| 12:25.06 | zeeqy | gr0mit:...is thoma sorted out??? |
| 12:25.27 | gr0mit | i think so - he is very quiet! |
| 12:26.11 | zeeqy | ok, let me repeat my question....I can install asterisk core from Synaptic...but the reall problem is the sql and AMP portal integration |
| 12:27.07 | zeeqy | typically i want my ubuntu server to run AMP or asteriskNOW or any other front end gui |
| 12:27.08 | gr0mit | well, can you not just sudo apt-get install apache2 or wotever? |
| 12:27.31 | gr0mit | well if you want a gui you are on your own! |
| 12:27.32 | zeeqy | apache2 works ok... |
| 12:27.47 | gr0mit | this is the land of vi and emacs |
| 12:27.57 | thomas | gr0mit: better: http://paste.keks.be/64 |
| 12:27.58 | thomas | :-) |
| 12:27.58 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 12:28.00 | gr0mit | or in my case, joe |
| 12:28.17 | JT | kombi: all zaptel cards provide timing. |
| 12:28.24 | gr0mit | thomas, better ;-) |
| 12:28.29 | zeeqy | ok, its a silly question but how do u access the asterisk box via web...? |
| 12:28.30 | thomas | :-) |
| 12:28.44 | gr0mit | you don't on #a&a |
| 12:28.48 | gr0mit | i mena #asterisk |
| 12:29.00 | gr0mit | there are trixbox, freepbx forums |
| 12:29.07 | kombi | JT: I just wonder how I get it to work.. that b410p has been in that box for a while but ztdummy is still loaded |
| 12:29.17 | Unlockgod | does anyone know what the dialing rule is for any number? |
| 12:29.23 | Unlockgod | to go via the trunk |
| 12:29.24 | gr0mit | you will get short shrift here |
| 12:29.37 | masus | (Asterisk 1.4) which we used effeciently before doesnt work after the agents logged in with "AgentLogin()", for example *reload* or *agent show* don't respond and locks the CLI and we get this message "The previous reload command didn't finish yet" Does Anyone have an idea. Thanks. |
| 12:29.42 | zeeqy | i used trixbox and asterisk for quite some time but...not on a server like ubuntu |
| 12:30.04 | gr0mit | well you are on your own, zeeqy |
| 12:30.32 | Unlockgod | hi guys, anyone know why it would be saying all cuirvits busy? |
| 12:30.36 | JT | kombi: then stop ztdummy from loading, simple |
| 12:30.38 | Unlockgod | circuits* |
| 12:30.38 | zeeqy | ok, let me go without the GUI...m i still here ??? |
| 12:30.40 | thomas | gr0mit: what is a good GUI for asterisk? |
| 12:30.42 | gr0mit | if you want to install from source and configure using text files or databases, then ask here |
| 12:30.49 | gr0mit | thomas, there is'nt one |
| 12:30.53 | JT | wrong channel if you want a gui |
| 12:30.58 | thomas | gr0mit: hm. ok :) |
| 12:31.03 | thomas | gr0mit: the best gui are VIM? ;) |
| 12:31.05 | thomas | lol ;) |
| 12:31.13 | gr0mit | uses joe |
| 12:31.22 | thomas | i dont like joe |
| 12:31.24 | thomas | i like "notepad" |
| 12:31.27 | thomas | and vim ;) |
| 12:31.30 | JT | barfs |
| 12:31.34 | JT | notepad is utter rubbish |
| 12:31.39 | thomas | :P |
| 12:31.40 | gr0mit | it is very small, runs over IP-over-avian-carrier |
| 12:31.57 | zeeqy | gr0mit: my concern for asking about the GUI was how m i going to create extensions like we do that on asterisk iso CDs on Centos |
| 12:32.32 | gr0mit | use text files like extensions.conf and sip.conf |
| 12:33.09 | thomas | is asterisk.adsi better as asterisk.conf? |
| 12:33.16 | zeeqy | gr0mit: plz have a look at this howto:.. https://wiki.ubuntu.com/AsteriskOnUbuntu#head-d84bca9e76d0788bf6f2b2c7a711893b8247d40b |
| 12:33.56 | gr0mit | and? |
| 12:34.05 | kombi | JT: hmm, zaptel start gives me "No functioning zap hardware found in /proc/zaptel, loading ztdummy |
| 12:34.05 | kombi | ".. |
| 12:34.08 | zeeqy | the freePBX is included in this how to...which is typically used for such purpose |
| 12:34.23 | gr0mit | well go and install it then! |
| 12:34.32 | gr0mit | but we do not discuss freepbx here. |
| 12:34.43 | JT | kombi: oh, a b410p is not zaptel hardware |
| 12:34.44 | gr0mit | we are not experts on it, and cannot advise |
| 12:34.51 | JT | kombi: it uses awful misdn |
| 12:34.56 | gr0mit | sighs |
| 12:35.20 | kombi | JT: ..;) so I just don't start zaptel on boot? |
| 12:35.31 | zeeqy | great....thanks for your help and support...let me try...can always share with someone who will need some help if I get it right.... |
| 12:37.02 | *** part/#asterisk zeeqy (n=zeeqy@196-209-150-205-tbnb-esr-3.dynamic.isadsl.co.za) |
| 12:37.24 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582944.dsl.bell.ca) |
| 12:40.37 | kombi | JT: oh, oh.. from the KB: "The B410P does not provide Zaptel timing. Instead, use ztdummy or another Zaptel card to provide timing for Asterisk." |
| 12:42.26 | thomas | is it posible create a macro and give as parameter example "macro|bla|1,2,9" and then inside the macro a while loop ? |
| 12:42.51 | gr0mit | thomas, why not?? |
| 12:43.03 | thomas | gr0mit: and how? |
| 12:43.27 | thomas | example with "say a number" |
| 12:43.28 | thomas | moment |
| 12:43.31 | gr0mit | how what? pass parameters into a macro? |
| 12:44.01 | gr0mit | they get passed as ${ARG1}, ${ARG2} etc |
| 12:44.35 | gr0mit | so you call exten => 100,1,Macro(blah,1,2,9) |
| 12:44.38 | JT | fuck kombi is a know |
| 12:44.42 | JT | knob |
| 12:44.45 | JT | i just told him all that |
| 12:46.08 | thomas | gr0mit: have example: http://paste.keks.be/65 |
| 12:46.30 | thomas | gr0mit: but how i can count the args, i mean the parameter |
| 12:46.35 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 12:46.58 | thomas | gr0mit: in the example paste.keks.be ihave 4.. but the next time.. 3 or 8 |
| 12:47.26 | gr0mit | so you want a variable number of paramaters ? eeew |
| 12:47.32 | thomas | jep |
| 12:47.35 | gr0mit | why? |
| 12:47.42 | thomas | gr0mit: for tests. |
| 12:49.09 | gr0mit | can you not pass them in as a single string? |
| 12:49.20 | gr0mit | then it is easy to work out the length of the string |
| 12:49.54 | masus | is there another asterisk channel for asking another questions not for install or configure asterisk? |
| 12:50.08 | gr0mit | masus, depends on your question! |
| 12:50.20 | masus | gr0mit: (Asterisk 1.4) which we used effeciently before doesnt work after the agents logged in with "AgentLogin()", for example *reload* or *agent show* don't respond and locks the CLI and we get this message "The previous reload command didn't finish yet" Does Anyone have an idea. Thanks. |
| 12:51.03 | *** join/#asterisk Segnale007 (n=Segnale0@host163-249-dynamic.23-79-r.retail.telecomitalia.it) |
| 12:52.32 | gr0mit | masus, no idea sorry |
| 12:52.40 | masus | gr0mit: OK Thanks |
| 12:52.50 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:52.57 | masus | another irc channels maybe ? |
| 12:53.21 | gr0mit | masus, there will be others here who probably know a lot more than me. |
| 12:53.46 | masus | Yes i know . |
| 12:53.52 | *** join/#asterisk ManxPower (n=manxpowe@162.sub-70-223-245.myvzw.com) |
| 12:53.59 | masus | there are a lot of people here |
| 12:54.32 | gr0mit | so just ask again when ManxPower gets on line |
| 12:54.37 | gr0mit | oh, here he is ;-) |
| 12:55.56 | masus | hmm ok i'll ask time to time again .. Thanks |
| 12:56.09 | masus | maybe someone will know the answer |
| 12:56.33 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 12:57.03 | jblack | masus: Sounds like a bug to me. |
| 12:58.36 | masus | and here is my log file -> http://rafb.net/p/Ic1MYs30.html |
| 12:58.59 | masus | jblack: i don't know i'm newbie |
| 13:01.08 | ManxPower | looks like you have a denial of service attach |
| 13:01.24 | ManxPower | Also looks like you don't have a /etc/asterisk/inidcations.conf |
| 13:02.00 | masus | ManxPower: i'll take a look one moment please. |
| 13:02.47 | masus | ManxPower: you are right i havent indications.conf |
| 13:02.57 | ManxPower | what version of Asterisk do you have? |
| 13:03.01 | masus | 1.4 |
| 13:03.14 | ManxPower | No, what specific version of 1.4 |
| 13:03.18 | masus | Asterisk SVN-branch-1.4-r116799 |
| 13:03.43 | ManxPower | Too bad. I can't tell you if there are any significant well known issues with that version. |
| 13:03.46 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 13:04.07 | masus | which version is the best now ? |
| 13:04.11 | ManxPower | 1.4.21 was seriously broken, 1.4.21.1 was released yesterday to address the issue in 1.4.20 |
| 13:04.34 | ManxPower | Whatever version, I suggest using a released version |
| 13:05.41 | masus | hmm OK ManxPower Thank you for your Help. |
| 13:07.04 | ManxPower | masus: there can be some very bad undiscovered bugs in the SVN versions. |
| 13:07.08 | ManxPower | not often, but it happens |
| 13:07.37 | masus | i understand will try to install from tar.gz |
| 13:08.03 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 13:08.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.42 | anonymouz666 | ~centos52bug |
| 13:08.43 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 13:09.26 | masus | ManxPower: only one more question is it possible to authenticate agents from db like extensions |
| 13:09.39 | masus | Realtime |
| 13:09.42 | ManxPower | masus: I don't use agents |
| 13:10.01 | masus | ManxPower: Ok Thanks :) |
| 13:10.09 | thomas | grrr |
| 13:10.10 | thomas | servetux*CLI> stop now |
| 13:10.10 | thomas | servetux*CLI> |
| 13:10.15 | seanbright | thomas: stop now |
| 13:10.15 | *** join/#asterisk ludan (n=daniele@192.167.215.122) |
| 13:10.20 | thomas | seanbright: :) |
| 13:10.27 | ManxPower | or realtime. No need to have the complications of realtime for so few users (I have about 300 users) |
| 13:11.01 | masus | you have 300 Agents ? |
| 13:11.08 | ManxPower | no, 300 users |
| 13:11.13 | masus | and no need for realTime :) |
| 13:11.23 | masus | sorry my english is not very well |
| 13:11.23 | ManxPower | I don't use agents or queues |
| 13:11.53 | masus | ManxPower: Yes but you dont know the possibility ? |
| 13:14.32 | *** part/#asterisk Unlockgod (n=ben@host81-148-207-112.in-addr.btopenworld.com) |
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| 13:28.49 | daniev | hello. i have a doubt and your expertised help would be pretty useful. I will use an asterisk server with an E1. I've considering to buy Sangoma or Pika card. Which one can you recommend me? |
| 13:29.10 | ManxPower | Sangoma or Digium. |
| 13:29.50 | daniev | better sangoma than pika ? |
| 13:29.58 | gr0mit | i prefer sangoma |
| 13:30.04 | gr0mit | never used pika |
| 13:30.09 | ManxPower | I have never heard of Pika, so.... |
| 13:30.13 | gr0mit | hehe |
| 13:31.35 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 13:31.42 | Dr-Linux|home | anybody tried TE110P digium card? |
| 13:31.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:32.01 | daniev | i though pika would be a good choice |
| 13:32.05 | russellb | Dr-Linux|home: that card has been replaced |
| 13:32.05 | ManxPower | thousands of people, I'm sure. |
| 13:32.10 | russellb | but yeah |
| 13:32.15 | daniev | nobody used a pika card ? |
| 13:32.18 | ManxPower | daniev: It's not a good choice if nobody here uses it. |
| 13:32.24 | Dr-Linux|home | russellb: replaced with what? |
| 13:32.29 | russellb | TE120 |
| 13:32.49 | *** topic/#asterisk by Corydon76-dig -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.21.1 (2008/06/30) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for |
| 13:33.03 | Dr-Linux|home | russellb: I'm going to buy single span digium card, kindly suggest one |
| 13:33.10 | russellb | TE120 :) |
| 13:33.14 | russellb | or TE122 |
| 13:33.20 | ManxPower | Corydon76-dig: Did the maketing dweebs update the link on the web site for 1.4.21.1 ? |
| 13:33.30 | Dr-Linux|home | i'm already using a few two span cards so i'm confused |
| 13:33.31 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:33.31 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 13:33.32 | Dr-Linux|home | i see |
| 13:33.34 | russellb | if by marketing dweebs, you mean me, then yes |
| 13:33.43 | gr0mit | Sangoma A101 |
| 13:33.57 | ManxPower | russellb: So you are the only that always forgets to update the web site when a new version comes out? |
| 13:34.01 | Dr-Linux|home | what's the difference between TE120 and TE122 ? |
| 13:34.15 | gr0mit | Dr-Linux|home, 2 |
| 13:34.18 | russellb | i always update the web site |
| 13:34.18 | jaytee | the digit at the end? |
| 13:34.18 | daniev | ManxPower: thank you very much |
| 13:34.22 | russellb | it's the mirror that breaks all of the time |
| 13:34.30 | Corydon76-dig | ManxPower: it's not a matter of forgetting... sometimes the sync process needs to be kicked... hard... |
| 13:34.33 | russellb | Dr-Linux|home: hw echo can |
| 13:35.25 | Dr-Linux|home | http://www.voiplink.com/Digium_TE122_1_T1_E1_Port_Standard_PCI_p/digium-te122.htm |
| 13:35.40 | Dr-Linux|home | one difference is echo can... and other? |
| 13:35.55 | Dr-Linux|home | and from where i can buy it? |
| 13:36.06 | jaytee | www.telephonydepot.com |
| 13:36.13 | ManxPower | Corydon76-dig: then the sync process needs to be fixed, as every single time there's a new release, it's not on the web site until someone complains |
| 13:36.21 | jaytee | or www.voipsupply.com if you like paying a bit more |
| 13:36.29 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:37.39 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:37.49 | Dr-Linux|home | jaytee: what about this one: http://www.digiumcards.com ? |
| 13:38.15 | ManxPower | I don't think I'd buy from a company that uses Digium in their name, except for Digium itself. |
| 13:39.09 | Vec | Hi, I am trying to compile mISDN on ubuntu and I am getting this error "*** CFLAGS was changed in "/usr/src/mISDN-1_1_7_2/drivers/isdn/hardware/mISDN/Makefile". Fix it to use EXTRA_CFLAGS", any ideas ? |
| 13:39.26 | ManxPower | Vec: ask on #ubuntu? |
| 13:39.40 | thomas | how i can show the application what have the higherst cpuload? |
| 13:39.46 | Vec | ok. was not sure if it was ubuntu specific or misdn |
| 13:39.53 | thomas | like "top" |
| 13:39.54 | ManxPower | thomas: you can't, if you mean Asterisk Applications |
| 13:40.15 | thomas | ManxPower: hm. i would like show why my system to slow |
| 13:40.21 | ManxPower | Vec: It does not appear to be an Asterisk specific thing. |
| 13:40.21 | thomas | ManxPower: only if i dial with asterisk... |
| 13:40.31 | daniev | another issue. my telephony provider offers me three options: pri net5 / pri qsig / R2 Q 421. which one of these uses the sangoma cards ? |
| 13:40.37 | Vec | ManxPower : yeh no prob |
| 13:40.41 | daniev | sorry about my ignorance |
| 13:41.07 | ManxPower | daniev: you never want R2, you want PRI QSIG |
| 13:41.40 | jaytee | Dr-Linux|home, that is one of the ugliest websites I've ever seen |
| 13:42.17 | jaytee | and while they list the TE122 card the price looks too low for coming equipped with the hw echo cancellation. |
| 13:42.18 | daniev | ManxPower: Sangoma cards only works with Qsig? it seems like i heard about Net5 too? am i wrong? |
| 13:42.35 | russellb | the sangoma card has nothing to do with the signalling |
| 13:42.38 | ManxPower | daniev: ASTERISK only works with PRI QSIG |
| 13:42.39 | russellb | that is handled by asterisk |
| 13:42.48 | russellb | and you should get a digium card :-p |
| 13:43.23 | ManxPower | PRI is a high level protocol, not an issue with the lowlevel hardware drivers |
| 13:43.31 | daniev | ManxPower: ok. thank you |
| 13:44.13 | thomas | is devstate integrated on asterisk 1.4 ? |
| 13:44.41 | russellb | what do you mean by devstate |
| 13:44.44 | ManxPower | thomas: russellb would know, for sure, but I'm pretty sure it does not. |
| 13:45.17 | russellb | the dialplan function? |
| 13:45.18 | gr0mit | erm ManxPower i think you want Net5 - this is EuroISDN |
| 13:45.20 | thomas | russellb: sorry. i mean, can i set the LED of the snom phone without a patch? |
| 13:45.32 | russellb | manually? |
| 13:45.34 | russellb | no. |
| 13:45.37 | gr0mit | QSIG is mainly for pbx interconnects |
| 13:45.50 | russellb | svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4/ |
| 13:45.57 | russellb | i think that's the path ... there is a backport in there somewhere |
| 13:46.01 | ManxPower | gr0mit: maybe, those service names don't help much |
| 13:46.11 | Dr-Linux|home | oo |
| 13:46.15 | thomas | russellb: this: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ ? |
| 13:46.27 | Dr-Linux|home | anybody knows what's the warrenty for Digium TE120P ? |
| 13:46.40 | ManxPower | Dr-Linux|home: Digium does. |
| 13:46.42 | gr0mit | ManxPower, they are all european specs |
| 13:46.54 | russellb | yeah |
| 13:47.09 | Dr-Linux|home | http://www.digiumcards.com/Digium_TE120P.html |
| 13:47.12 | ManxPower | gr0mit: none of them say "EuroISDN" |
| 13:47.21 | ManxPower | Dr-Linux|home: that is not Digium |
| 13:47.23 | gr0mit | Net5 is euroisdn |
| 13:47.30 | Dr-Linux|home | okey but for the above card, how many years? 1 year, 2 year ? |
| 13:47.51 | Dr-Linux|home | oo i see |
| 13:47.54 | ManxPower | Dr-Linux|home: try looking for that card on DIGIUMs site. |
| 13:47.57 | Dr-Linux|home | let me find their site |
| 13:48.05 | Dr-Linux|home | ManxPower: okay thanks |
| 13:48.05 | thomas | russellb: ah, this patch is of you (from you?) |
| 13:48.06 | M1s3ry | :) |
| 13:48.14 | russellb | thomas: yes |
| 13:48.20 | thomas | aaeh, not patch, is a function, yes? |
| 13:48.31 | russellb | nods |
| 13:48.34 | russellb | it's a standalone module |
| 13:48.38 | creativx | hehe |
| 13:48.38 | russellb | you just drop it in the funcs directory |
| 13:48.41 | creativx | "blinky lights" |
| 13:48.41 | creativx | gg |
| 13:48.46 | thomas | russellb: ok, if is it ok, i can you say what i would like. |
| 13:48.58 | thomas | russellb: i would like via/per PHP set lamp on /off |
| 13:49.08 | thomas | russellb: i make this with a application devstate |
| 13:49.14 | russellb | yep, you can use the devstate function for that |
| 13:49.21 | thomas | russellb: is it posobile only with YOUR function, only? |
| 13:49.37 | russellb | i wouldn't say it's the _only_ way |
| 13:49.40 | russellb | but it's the easiest |
| 13:49.48 | thomas | emm |
| 13:49.49 | thomas | hmm |
| 13:49.58 | jaytee | I use Putty for linux to ssh into my * box but it won't let me copy and paste into a text file what's coming across the console. What do most of you gurus use? |
| 13:50.11 | thomas | i make this at the moment with login the console, then devstate bla.... |
| 13:50.21 | thomas | russellb: how i can this with you function, with php? |
| 13:50.24 | thomas | per AMI ? |
| 13:50.43 | russellb | AMI or AGI |
| 13:50.49 | russellb | depending on the application and how you want to trigger it |
| 13:50.52 | Dr-Linux|home | ManxPower: that's why i'm confused: |
| 13:50.53 | ManxPower | jaytee: I use Puppy |
| 13:50.53 | Dr-Linux|home | http://www.voiplink.com/Digium_TE122_1_T1_E1_Port_Standard_PCI_p/digium-te122.htm |
| 13:50.56 | ManxPower | ..er.. putty |
| 13:51.00 | *** join/#asterisk remibemol (n=remibemo@jem75-2-82-233-232-40.fbx.proxad.net) |
| 13:51.03 | thomas | russellb: emm. what is easier, AGI? |
| 13:51.12 | Dr-Linux|home | http://store.digium.com/productview.php?product_code=TE122B |
| 13:51.14 | russellb | AGI is easier, yeah |
| 13:51.18 | Dr-Linux|home | different prices |
| 13:51.20 | russellb | but that would be if something with a call triggers it |
| 13:51.23 | russellb | not something external to asterisk |
| 13:51.34 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:51.36 | jaytee | ManxPower, Putty on Windows or the linux version? If I select text from the console in Ubuntu I can't paste into a text file with it. |
| 13:51.44 | ManxPower | Dr-Linux|home: Gosh! I would never have found that non-digium web site! |
| 13:51.49 | thomas | russellb: moment, please. with a call, do you have a simple example? moment, i search my php... |
| 13:51.56 | ManxPower | jaytee: Windows, but no real difference. |
| 13:52.16 | russellb | thomas: sorry, no examples |
| 13:52.34 | jaytee | ManxPower, and you can copy from it? hmmm, must be something in my settings. |
| 13:52.37 | russellb | other than the basic ones on asterisk.org |
| 13:52.44 | thomas | russellb: i have this: |
| 13:52.44 | thomas | <PROTECTED> |
| 13:52.44 | thomas | <PROTECTED> |
| 13:52.46 | thomas | .... |
| 13:52.48 | ManxPower | jaytee: I would not use if if I could not copy/paste. see the putty settings |
| 13:53.10 | ManxPower | putty lets you pick the copy method and paste method |
| 13:53.35 | Dr-Linux|home | ManxPower: but her is the digium site, i can't find the model russellb suggest i.e. TE210 |
| 13:53.45 | *** join/#asterisk stream (i=stream@prostream.org) |
| 13:53.52 | stream | grettings everyone |
| 13:54.06 | russellb | Dr-Linux|home: http://www.digium.com/en/products/digital/ |
| 13:54.10 | daniev | signalling is an asterisk issue? always Qsig? excuse if i insist but i wanna make sure |
| 13:54.19 | Dr-Linux|home | ok thanks |
| 13:54.28 | stream | can anyone recommend any good cheap SIP providers? I only need like 3-4 trunks, one local #, cheap MRC |
| 13:54.44 | thomas | russellb: what you mean with "call trigger" ? |
| 13:54.47 | ManxPower | daniev: Asterisk uses Q.931, get whatever line supports that. |
| 13:55.10 | ManxPower | ~trunk |
| 13:55.10 | jbot | well, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 13:55.14 | ManxPower | ~itsp |
| 13:55.14 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 13:55.39 | daniev | ManxPower: ok. thank you. |
| 13:55.45 | ManxPower | daniev: The common usage for that is EuroISDN |
| 13:56.03 | stream | btw i am looking for a USA SIP provider |
| 13:56.18 | stream | broadvox seems to expensive |
| 13:56.23 | ManxPower | stream: then now you know |
| 13:56.24 | stream | i was hoping to keep it < $20-$30 /mo |
| 13:56.41 | daniev | ManxPower: i have to mention that i'm in latin america (colombia). We use the european standards. |
| 13:57.35 | ManxPower | stream: My boyfriend uses Vitelity and pays less than $10/month |
| 13:57.51 | stream | they will hand me a direct SIP trunk w/o a IAD? |
| 13:58.04 | thomas | russellb: huhu? :-) |
| 13:58.11 | ManxPower | no, because there is no such thing as a sip trunk. |
| 13:58.18 | russellb | thomas: i'm sorry, i'm busy with something else right now .. |
| 13:58.29 | thomas | russellb: oh, ok. sorry. please. |
| 13:58.31 | stream | how do you figure |
| 13:58.57 | Veggen | ...I have $20 a month and the posibility to call for the whole amount (i.e. it's sort of a "minimum usage fee". |
| 13:59.08 | Veggen | And I get iax :) |
| 13:59.09 | ManxPower | stream: I figure there is no such thing as a sip trunk. You have SIP connctions, but a connection is not a trunk. |
| 13:59.22 | elguero | unfortanely, some companies use the term sip trunking even though it is incorrect |
| 13:59.29 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 13:59.43 | ManxPower | electus: good thing there are people around to remind people it is not correct. |
| 14:00.13 | ManxPower | Even if there are people around that won't listen |
| 14:00.46 | stream | hah vitelity website wont load to sign up.. how lame |
| 14:00.46 | stream | https://portal.vitelity.net/main/signup/signup.php |
| 14:01.32 | ManxPower | oh well |
| 14:02.07 | gr0mit | daniev, we have an ISDN E1 running standard EuroISDN on my asteirsk box in Bogota |
| 14:02.54 | elguero | stream: we just started trying to use bandwidth.com and setup was very easy; it worked the first time... haven't put it in production yet so I can't say too much but pricing was pretty decent; |
| 14:03.23 | russellb | and bandwidth.com uses asterisk :) |
| 14:03.32 | elguero | russellb: yep!! |
| 14:03.33 | russellb | (and openser) |
| 14:03.44 | daniev | gr0mit: are you in bogota ? |
| 14:03.58 | stream | i use alcatel-lucent |
| 14:03.58 | daniev | gr0mit: using Qsig signalling ? |
| 14:04.03 | gr0mit | nope, Basingstoke, UK |
| 14:04.10 | jaytee | ManxPower, figured it out. In Ubuntu or probably any other linux distro Putty uses the "middle mouse button" which in my case is the scroll wheel. |
| 14:04.18 | daniev | ah ok |
| 14:04.20 | gr0mit | nope. Qsig is a PBX to PBX signalling system |
| 14:04.30 | gr0mit | we are running normal ISDN |
| 14:04.53 | gr0mit | where 'normal' is 'everywhere except Northe |
| 14:04.59 | gr0mit | North America and Japan |
| 14:05.07 | elguero | stream: also, at home I use Teliax (www.teliax.com)... I use the IAX protocol but they also do sip... and I use the pay as you go plan which I have found to be economical since I end up using my cell a lot |
| 14:05.28 | daniev | gr0mit: so, can i use Net5 signalling with asterisk? |
| 14:05.40 | gr0mit | yup. Net5 = EuroISDN |
| 14:05.47 | gr0mit | which is very well supported in Asterisk |
| 14:06.28 | daniev | gr0mit: ok. thank you |
| 14:13.01 | *** join/#asterisk gazzerh (n=garryh@86.54.113.226) |
| 14:13.36 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 14:14.32 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 14:14.43 | dandre | Hello, |
| 14:14.44 | dandre | In my DECT C470 testing, I have difficulties to have transfer functionality: I include full trace with sip debug set just before the effective transfer: |
| 14:14.44 | dandre | ext 43 is a sip phone that calls ext 51 (the DECT SIP Phone). Then ext 51 calls ext 50 (a Zap Phone) which answer the call and then ext 51 do the transfer. a short time after, ext 51 rings again . What is the trouble? |
| 14:14.44 | dandre | http://pastebin.com/d20f2d093 |
| 14:23.35 | stream | arg bandwidth.com cant service me they said they dont have e911 here |
| 14:26.01 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 14:29.02 | *** join/#asterisk mond0 (n=lvsyijnc@galactic.umflint.edu) |
| 14:29.11 | *** part/#asterisk mond0 (n=lvsyijnc@galactic.umflint.edu) |
| 14:32.36 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:32.54 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 14:33.28 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 14:35.18 | *** join/#asterisk Fuzix (n=fuzix@250-118.citynet.ftth.internl.net) |
| 14:35.33 | NovceGuru | stream: yeah they refuse to provide service without e911 |
| 14:36.05 | stream | lame |
| 14:36.11 | stream | ill try vitaleity |
| 14:37.01 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 14:37.46 | WildPikachu | Hi guys, beeing trying to solve this now for 1 week .... got 3 users with snom phones and about 20 with soft phones (x-lite) the snom phones work perfectly now, the x-lite soft phones all disconnect after 4-5 mins every time |
| 14:39.29 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:41.45 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 14:41.59 | UnixDog | ok have a issues with 1.4.21.1 |
| 14:42.25 | UnixDog | when I compile it for lol mem it runs fine till I get a call. then after the call it crashes |
| 14:42.55 | UnixDog | I need low mem because its running on a embedded platform |
| 14:43.50 | russellb | get a backtrace, post to bugs.digium.com |
| 14:45.11 | WildPikachu | setups a softphone to tes tthis |
| 14:46.20 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 14:48.07 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 14:48.25 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 14:48.35 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:48.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:48.49 | UnixDog | ok |
| 14:49.45 | tzafrir_laptop | russellb, are the .moduleinfo files and such intentionally included in the tarball? |
| 14:50.02 | russellb | i don't remember what .moduleinfo is |
| 14:50.44 | tzafrir_laptop | generated automatically at build (IIRC as early as menuselect) |
| 14:50.53 | russellb | oh, the XML tree? |
| 14:50.55 | russellb | yes |
| 14:51.05 | russellb | just to make building go a little bit faster, i guess |
| 14:51.57 | tzafrir_laptop | ok. I had to add a rule to remove them from the repackaged tarball, as make dist-clean removes them |
| 14:52.16 | russellb | ok, shouldn't hurt anything. |
| 14:52.48 | *** join/#asterisk aaawrekng (n=aaawrekn@c-76-121-222-172.hsd1.wa.comcast.net) |
| 14:52.54 | *** part/#asterisk aaawrekng (n=aaawrekn@c-76-121-222-172.hsd1.wa.comcast.net) |
| 14:52.55 | *** join/#asterisk aaawrekng (n=aaawrekn@c-76-121-222-172.hsd1.wa.comcast.net) |
| 14:53.10 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 14:53.21 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 14:53.42 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:55.38 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 14:55.54 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 15:05.30 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 15:07.32 | BCS-Satori | I am having an issue with "make" and Zaptel 1.4.11. During the "make" I recieve these errors: http://rafb.net/p/2pqhfr99.html Any Ideas? |
| 15:08.08 | [TK]D-Fender | BCS-Satori: Read the channel topic <-- |
| 15:08.15 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 15:08.28 | BCS-Satori | ah i see |
| 15:08.34 | BCS-Satori | ~centos52bug |
| 15:08.35 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 15:14.00 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 15:16.05 | *** join/#asterisk oilinki (n=oil@ppp-124-120-3-63.revip2.asianet.co.th) |
| 15:16.16 | thomas | russellb: hm. have now tested over two hours. i dont know what you mean with call the function.... |
| 15:17.00 | russellb | execute it in one of the ways you can execute a dialplan function |
| 15:17.06 | *** join/#asterisk spokra (n=spokra@host093-179-132.sea0.speakeasy.net) |
| 15:17.19 | russellb | exten => 1234,1,Set(DEVICE_STATE(Custom:foo1=IN_USE) |
| 15:17.32 | russellb | or, using the SetVar manager action |
| 15:17.50 | russellb | or setting it from AGI during a call |
| 15:18.08 | thomas | emm, a call from agi? hmmm |
| 15:18.23 | *** join/#asterisk oilinki (n=oil@ppp-124-120-3-63.revip2.asianet.co.th) |
| 15:19.42 | *** join/#asterisk I-MOD (n=I-MOD@c-68-62-216-5.hsd1.al.comcast.net) |
| 15:19.49 | WildPikachu | i wonder if callprogress will help fix my hangup problem on softphones |
| 15:20.00 | *** part/#asterisk I-MOD (n=I-MOD@c-68-62-216-5.hsd1.al.comcast.net) |
| 15:20.38 | [TK]D-Fender | WildPikachu: nope. "callprogress" is an analog thing. |
| 15:20.58 | [TK]D-Fender | WildPikachu: Softphone signalling is all digital. Describe the pieces involved with your problem |
| 15:21.04 | WildPikachu | dialing out over analogue phones, x-lite on 20 pcs now hangs up after 4-5 mins on a call :( |
| 15:21.08 | masus | ManxPower: switched to Asterisk 1.4.21.1 :) |
| 15:21.15 | WildPikachu | using a proper snom/granstream on 5 users works 100% |
| 15:21.58 | gr0mit | WildPikachu, have you got detect progress tones? |
| 15:21.58 | WildPikachu | i just set callprogress=yes and i'm 2 minutes past the longest time i've been on a call (out of more than 20 calls) |
| 15:22.17 | *** join/#asterisk oilinki (n=oil@ppp-124-120-3-63.revip2.asianet.co.th) |
| 15:22.22 | WildPikachu | well ... i have busydetect=yes, busypattern=500,500, busycount=10 |
| 15:22.32 | gr0mit | is'nt that more like 'random dropped calls = yes' |
| 15:22.43 | WildPikachu | callprogress you mean? |
| 15:22.43 | gr0mit | try turning it all off. then see |
| 15:22.46 | ManxPower | WildPikachu: NO! callprogress causes random hangups |
| 15:22.46 | gr0mit | yup |
| 15:22.56 | WildPikachu | ok, i'll disable it again, one sec :) |
| 15:23.06 | WildPikachu | (it was disabled before) |
| 15:23.06 | ManxPower | busydetect will also cause that. |
| 15:23.14 | ManxPower | busycount=10 will help. |
| 15:23.18 | *** join/#asterisk Firass-VC22 (n=Firass-z@method.vikcomm.wwu.edu) |
| 15:23.21 | gr0mit | any tone detection is heading for trouble |
| 15:23.23 | ManxPower | WildPikachu: I assume you can't use PRI? |
| 15:23.27 | WildPikachu | can I turn busydetect off aswell? |
| 15:23.31 | gr0mit | it is a fudge - use BRI or PRI |
| 15:23.32 | ManxPower | or SIP or IAX? |
| 15:23.38 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 15:23.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:23.38 | WildPikachu | ManxPower, nope ... got 3 normal analogue lines |
| 15:23.50 | WildPikachu | ManxPower, sip on all users |
| 15:23.53 | gr0mit | WildPikachu, consider getting service from a voip provider |
| 15:24.00 | ManxPower | WildPikachu: It sucks to be you, especially if you are outside the USA |
| 15:24.06 | WildPikachu | yep ;) |
| 15:24.11 | WildPikachu | south africa |
| 15:24.12 | gr0mit | where are you, WildPikachu ? |
| 15:24.17 | gr0mit | aah, well go for ISDN then |
| 15:24.20 | WildPikachu | one sec, i've disabled everything now |
| 15:24.21 | ManxPower | WildPikachu: don't expect to be able to totally solve your hangup issues |
| 15:24.23 | gr0mit | dirt cheap over there |
| 15:24.23 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 15:24.47 | WildPikachu | should i restart asterisk to be on the safe side? |
| 15:25.01 | WildPikachu | i've commented out everything, busydetect, busycount, busypattern |
| 15:25.16 | gr0mit | WildPikachu, can you replace your 3 analogue lines with a single BRI? |
| 15:25.24 | WildPikachu | not at present :( |
| 15:25.33 | gr0mit | technical reasons? |
| 15:25.54 | WildPikachu | yea |
| 15:26.00 | kippi | is there a good howto for jabber and asterisk? just playing around to see what i can do |
| 15:26.14 | ManxPower | kippi: no |
| 15:26.14 | WildPikachu | another of our asterisk boxes works 100%, just not this one ... really weird |
| 15:26.14 | gr0mit | ok. if this is a demo, you will fail at the first hurdle with analogue. |
| 15:26.18 | WildPikachu | ok, i'm on a test call right now |
| 15:26.21 | [TK]D-Fender | kippi: Instructions on the WIKI. Go read. |
| 15:26.46 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:26.52 | ManxPower | [TK]D-Fender: if kippi could read, he should have found that page already |
| 15:27.14 | gr0mit | WildPikachu, there are voip providers in ZA |
| 15:27.24 | WildPikachu | just got a hangup |
| 15:27.34 | WildPikachu | checks dmesg |
| 15:27.45 | WildPikachu | not a polarity switch at least, dmesg is empty |
| 15:27.46 | [TK]D-Fender | ManxPower: True. |
| 15:28.22 | ManxPower | [TK]D-Fender: isn't there a doc file for jabber in the doc dir? OR is that gtalk? |
| 15:28.27 | gr0mit | WildPikachu, so what does the consols say? |
| 15:28.38 | WildPikachu | <PROTECTED> |
| 15:28.42 | ManxPower | WildPikachu: your DIALSTATUS and HANGUPCAUSE would be handy |
| 15:28.45 | [TK]D-Fender | ManxPower: Both IIRC |
| 15:28.55 | WildPikachu | ManxPower, i can noop those, right? |
| 15:29.03 | ManxPower | [TK]D-Fender: I wish I could make my information as secret as the asterisk docs. |
| 15:29.06 | ManxPower | WildPikachu: of course. |
| 15:29.16 | WildPikachu | quickly changes his dialplan |
| 15:29.21 | gr0mit | so it looks like the hangup comes from the zap channel |
| 15:29.24 | gr0mit | which is odd |
| 15:29.25 | ManxPower | [TK]D-Fender: must be the whole "hiding in plain sight" issue. |
| 15:29.43 | WildPikachu | redials |
| 15:29.51 | WildPikachu | gr0mit, very ... one sec |
| 15:29.57 | gr0mit | is this prob with incoming or outgoing? |
| 15:30.00 | gr0mit | or both? |
| 15:30.04 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:30.10 | WildPikachu | only outgoing afaik |
| 15:30.33 | ManxPower | you're not doing something stupid like t/T/w/W on your Dial line, are you? |
| 15:30.51 | errr | how can I tell why asterisk is using 99.9% cpu ? |
| 15:31.01 | [TK]D-Fender | errr: What version? |
| 15:31.03 | dandre | Hello, |
| 15:31.15 | WildPikachu | ManxPower, i'm doing T |
| 15:31.21 | jameswf-home | is migrating some development machines to virtual box and has mixed feelings |
| 15:31.30 | errr | [TK]D-Fender: 1.4.19.2 |
| 15:31.30 | gr0mit | aaah - false DTMF detects? |
| 15:31.30 | ManxPower | WildPikachu: when troubleshiooting remove all the exra options |
| 15:31.45 | [TK]D-Fender | errr: Perhaps you should upgrade. |
| 15:31.57 | gr0mit | i wonder - what DTMF settings have you got in your SIP clients and sip.conf? |
| 15:32.00 | errr | [TK]D-Fender: I tried 1.4.21 but it crashed hourly |
| 15:32.04 | gr0mit | i'll bet you have inband |
| 15:32.08 | WildPikachu | ok, i must be doing something wrong .... it doesn't seem to hit my noops |
| 15:32.11 | WildPikachu | gr0mit, talking to me? |
| 15:32.14 | errr | I havent tried 4.21.1 though |
| 15:32.15 | gr0mit | yes, WildPikachu |
| 15:32.29 | WildPikachu | dtmfmode=rfc2833 |
| 15:32.33 | gr0mit | hmmmm. |
| 15:32.35 | gr0mit | ok. |
| 15:32.41 | [TK]D-Fender | errr: New version out yeterday.... |
| 15:32.44 | gr0mit | that rules that out |
| 15:32.44 | WildPikachu | defined for each user |
| 15:32.46 | jameswf-home | I hear Wild Pikachu taste like chicken |
| 15:32.47 | gr0mit | ok |
| 15:32.56 | errr | yeah I saw that. I hope it doesnt get worse :s |
| 15:33.09 | WildPikachu | one sec .... my noops after the dial() are not being hit, I shoved them in my ael ... *checks again* |
| 15:33.10 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 15:33.15 | dandre | I am testing a sip-dect phone and, when I do a transfer with that phone, after the transfer as been completed, it rings back. How can I trace to see wether the phone or my configuration is buggy? If I do the same with other sip phone, the result is fine |
| 15:33.41 | murdock_ut | I have a q quick question. If I update the version of zaptel on a system do I need to recompile * for that version? |
| 15:34.03 | WildPikachu | reads the ael docs to see if he must have another context or priority to catch hangups |
| 15:34.32 | ManxPower | WildPikachu: WHAT did I tell you about making things simple for testing?? |
| 15:34.36 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 15:34.42 | WildPikachu | its like 3 lines ManxPower :) |
| 15:34.45 | WildPikachu | removed all options |
| 15:34.46 | WildPikachu | everything |
| 15:34.51 | WildPikachu | just want to add a noop to check the hangupcause |
| 15:34.54 | ManxPower | but you are STILL using AE: |
| 15:34.56 | ManxPower | AEL |
| 15:35.21 | WildPikachu | ah, ok |
| 15:35.37 | WildPikachu | its like 10 lines long ... i'll write a quick traditional config |
| 15:36.40 | [TK]D-Fender | dandre: Enable full debug on everything and LOOK. |
| 15:37.21 | dandre | I have done it but I don't understand the debug trace |
| 15:37.42 | [TK]D-Fender | dandre: Then you should have provided it at the same time as asking your question. |
| 15:38.00 | dandre | http://pastebin.com/d20f2d093 |
| 15:38.21 | dandre | ext 43 is a sip phone that calls ext 51 (the DECT SIP Phone). Then ext 51 calls ext 50 (a Zap Phone) which answer the call and then ext 51 do the transfer. a short time after, ext 51 rings again . What is the trouble? |
| 15:38.34 | spokra | Hmmm guess i got the version combined.. 2.0.2 -> 2.1.0 upgrade problem: undefined method `find_full_template_path' when starting mongrel.. |
| 15:38.47 | *** join/#asterisk Docfxit (n=none@cpe-76-95-77-238.socal.res.rr.com) |
| 15:38.49 | gr0mit | WildPikachu, dumb question, but are your sure you have not got more than one place in your zapata.conf where you set callprogress= etc? |
| 15:38.56 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:39.02 | WildPikachu | greps |
| 15:39.08 | gr0mit | good lad! |
| 15:39.09 | ManxPower | spokra: Hate to tell you this but there is no Asterisk, zaptel, or pri version of 2.x |
| 15:39.13 | tzafrir_laptop | murdock_ut, if it's a minor version update: no |
| 15:39.29 | spokra | sorry wrong window!! |
| 15:39.31 | WildPikachu | gr0mit, all ;callprogress=yes and callprogress=no uncommented |
| 15:39.42 | gr0mit | ok -just checking, WildPikachu |
| 15:39.47 | WildPikachu | no problem :) |
| 15:39.52 | gr0mit | there is so often a 'DOH!' |
| 15:40.16 | murdock_ut | tzafrir_laptop: So as long as it's in the same series ie 1.4.x then I'm ok. |
| 15:41.03 | WildPikachu | gr0mit, /me nods |
| 15:41.26 | gr0mit | where in ZA are you, WildPikachu ? |
| 15:41.41 | WildPikachu | somerset mall, western cape |
| 15:41.48 | gr0mit | nice. |
| 15:41.55 | gr0mit | has a customer in Durban |
| 15:42.02 | WildPikachu | cool :) |
| 15:42.22 | gr0mit | they are using X-lite back to my hosted asteirsk in UK |
| 15:42.39 | gr0mit | so i know it works in the Southern Hemisphere ;-) |
| 15:43.06 | gr0mit | even with Telskum's lines |
| 15:43.53 | WildPikachu | yea |
| 15:44.00 | WildPikachu | i'm nearly done with my traditional extensions |
| 15:44.45 | [TK]D-Fender | dandre: I suspect you may be mixing up a blind transfer vs an attended one, and not completing it properly. |
| 15:45.06 | WildPikachu | ok ... now i want to add the noop to my traditional config |
| 15:45.12 | WildPikachu | searches voip-info |
| 15:46.21 | dandre | I don't know, I am just using the phone menu to do this transfer as stated in the doc |
| 15:47.04 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:47.47 | [TK]D-Fender | dandre: read it again a few times and trie the other options it my have for it |
| 15:48.09 | gr0mit | how's progress, WildPikachu ? |
| 15:48.21 | WildPikachu | testing right now |
| 15:49.12 | *** join/#asterisk datachomper (n=russ@75.146.194.59) |
| 15:49.21 | WildPikachu | waiting now for disconnect ... |
| 15:49.45 | gr0mit | drums paws on the floor of the kennel |
| 15:50.36 | datachomper | Is there a way to match zero or more characters in the extension names in extensions.conf? |
| 15:51.05 | WildPikachu | ooo, i pastebin |
| 15:51.16 | gr0mit | ok |
| 15:51.26 | datachomper | _[01+].5555555555,1,Playback(tt-somethingwrong) I tried this, but this only matches one or more ... |
| 15:52.01 | *** join/#asterisk TorrK (n=Torr@mail.stpaulinternet.net) |
| 15:52.01 | WildPikachu | http://pastebin.com/m1567ffb3 |
| 15:52.14 | WildPikachu | CANCEL .... weird ... the phone is unattended in speaker mode next to me |
| 15:53.10 | gr0mit | try muting both ends. |
| 15:53.21 | gr0mit | see if it is audio which is causing the prob. |
| 15:53.44 | WildPikachu | ah, good idea, i'm calling telkom's adsl helpdesk, my colleague was on the phone for 2.5hr to their music on hold ;) |
| 15:54.02 | *** join/#asterisk magic_hat (n=geoffdou@h-68-164-10-43.chcgilgm.dynamic.covad.net) |
| 15:54.25 | datachomper | I'm trying to match the same DID from multiple providers, sometimes they send +1NXXNXXXXXX others send just NXXNXXXXXX |
| 15:54.53 | gr0mit | datachomper, just do some manipulation |
| 15:55.06 | datachomper | gr0mit, howso? |
| 15:55.20 | magic_hat | hey folks. My VOIP account has a limit of 10 channels. I'm working with an AGI script that places outgoing calls in /var/spool/asterisk/outgoing. Can I make sure I don't overload the account by having the script put calls in there only when there's less than 10 in the directory? |
| 15:55.42 | gr0mit | e.g. exten => _+1X,1,goto(${EXTEN:2}) |
| 15:56.08 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:56.09 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:56.11 | gr0mit | <PROTECTED> |
| 15:56.13 | gr0mit | i menan |
| 15:56.26 | datachomper | gr0mit, genious, thank you |
| 15:56.44 | gr0mit | this will get anything beginning +1, and send it back in to the same context, minus the +1 |
| 15:57.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:57.15 | gr0mit | goes home. back later.. |
| 15:57.19 | WildPikachu | gr0mit, same cancel result |
| 15:57.23 | Docfxit | where can I upload a file for people to take a look at? |
| 15:57.36 | gr0mit | WildPikachu, wonder if it is a hardware issue? |
| 15:57.46 | WildPikachu | works fine with granstream & snom phones |
| 15:57.50 | WildPikachu | not x-lite |
| 15:57.51 | gr0mit | hmmmm |
| 15:58.09 | WildPikachu | tried 2 xlite versions (upgraded the one) |
| 15:58.21 | gr0mit | hands WildPikachu wireshark |
| 15:58.41 | gr0mit | but i thnik it is a zaptel issue. |
| 15:58.53 | gr0mit | gotta go now- bbl. after dinner! |
| 15:58.57 | WildPikachu | oki |
| 15:59.11 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 15:59.16 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-201-rrdg-esr-2.dynamic.isadsl.co.za) |
| 16:06.00 | WildPikachu | uk, forgive me, but i tried callprogress=yes and i've not been disconnected ... 5 mins and counting (disconnects normally < 5min) |
| 16:06.03 | WildPikachu | *ok |
| 16:06.11 | WildPikachu | prepares to be flamed |
| 16:06.13 | magic_hat | anyone.... My VOIP account has a limit of 10 channels. I'm working with an AGI script that places outgoing calls in /var/spool/asterisk/outgoing. Can I make sure I don't overload the account by having the script put calls in there only when there's less than 10 in the directory? |
| 16:06.30 | *** join/#asterisk oilinki (n=oil@ppp-124-120-3-63.revip2.asianet.co.th) |
| 16:08.33 | [TK]D-Fender | magic_hat: Its called "programming" You should try it some time. |
| 16:09.01 | WildPikachu | [TK]D-Fender, ever heard of callprogress=yes working for someone? |
| 16:09.18 | [TK]D-Fender | WildPikachu: that option also means "randomlydisconnectmycalls=yes". |
| 16:09.34 | magic_hat | [TK]D -- yeah, I would be writing one of these "programs" you mention to do this. I'm just wondering if the overall approach makes sense. |
| 16:09.35 | [TK]D-Fender | WildPikachu: and DISABLING it should not cause hangups. The reverse more likely. |
| 16:09.41 | WildPikachu | well ... its doing the opposite and i'm not joking :( |
| 16:09.57 | WildPikachu | progzone=za |
| 16:09.57 | WildPikachu | callprogress=yes |
| 16:10.01 | dandre | [TK]D-Fender: what dect sip phone could you recommand? |
| 16:10.01 | WildPikachu | no disconnects in 10 minutes |
| 16:10.09 | [TK]D-Fender | magic_hat: Method that makes sense is to actually scan *'s active channels before distributing new calls. |
| 16:10.18 | [TK]D-Fender | dandre: I have none to recommend. |
| 16:10.23 | magic_hat | Right, okay. I didn't know I could do that. |
| 16:10.32 | [TK]D-Fender | magic_hat: AMI <- |
| 16:10.37 | WildPikachu | [TK]D-Fender, the zapata.conf file says this tho ..."This feature can also easily detect false hangups." |
| 16:10.37 | dandre | ok |
| 16:10.40 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 16:10.41 | WildPikachu | unless i'm reading it wrong |
| 16:10.43 | [TK]D-Fender | magic_hat: Didn't look too hard I guess |
| 16:11.18 | [TK]D-Fender | WildPikachu: No... they probably overstated the easy by a factor of 100 |
| 16:11.40 | WildPikachu | i'm sure they did, but its working .... and yes, it was the very very last thing i tried after struggling for over a week with this issue |
| 16:11.48 | WildPikachu | very strange |
| 16:12.00 | fogo | My telco tells me I'm sending an additional "info device = 1" in the setup message on one of my PRIs. Any idea what could be causing this? |
| 16:12.23 | magic_hat | [TD]D: wow, you're crochety today! lol. I've never used AMI before, so I didn't think to check in those docs. |
| 16:12.32 | WildPikachu | [TK]D-Fender, if i disable it it definitly hangs up, i'll confirm shorly, its the only thing i changed |
| 16:13.23 | [TK]D-Fender | magic_hat: * has a very limited number of ways to process things and provide information. Dialplan, AMI, and CLI. |
| 16:13.38 | magic_hat | yeah, good point. |
| 16:13.51 | [TK]D-Fender | magic_hat: Leaving off 2 out of the 3 is not a sign of serious thinking. |
| 16:14.06 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 16:14.15 | WildPikachu | hands [TK]D-Fender a gun ... just shoot me :) |
| 16:14.54 | magic_hat | lol. i'm coming at this pretty fresh... I didn't need to do anything with AMI to get our office phones working. and that's the extent of my * knowledge. |
| 16:14.55 | nny_1 | quick Q.. company providing a PRI has asked some questions, and me being a PRI noobie, can't answer |
| 16:15.10 | nny_1 | I assume NI2 is the standard more or less from what i have read? |
| 16:15.14 | *** part/#asterisk datachomper (n=russ@75.146.194.59) |
| 16:15.31 | [TK]D-Fender | nny_1: your most common choice |
| 16:17.21 | masus | by all |
| 16:17.21 | nny_1 | k thanks.. that about does it.. I have descending sequential on the PRI with asterisk dialing out ascending sequential, immeidate in regards to immediate vs. wink signaling, and Return dialtone on seizure = yes and yield to glare =yes |
| 16:17.23 | *** part/#asterisk masus (i=masus@88.248.14.186) |
| 16:19.44 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:20.07 | WildPikachu | [TK]D-Fender, still going strong on the same call :) |
| 16:23.28 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 16:23.30 | *** join/#asterisk d-k-t-2 (n=dt@125.120.130.131) |
| 16:25.33 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 16:26.23 | WildPikachu | 25 mins |
| 16:30.18 | WildPikachu | ok, 30 minutes no problem |
| 16:30.22 | WildPikachu | hangs up |
| 16:36.46 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 16:39.09 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:44.38 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:45.01 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 16:45.24 | ddunavant | is there any way to tell a call queue set to ring all to stop ringing once one person has picked up, even though that person is listening to a macro? |
| 16:46.26 | [TK]D-Fender | ddunavant: doubt it |
| 16:47.02 | ddunavant | [TK]D: thanks, crud... |
| 16:48.27 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 16:51.20 | ddunavant | [TK]D: is there any way to call a group of people and have it once one of them pick up, have it stop ringing with the person who picked up listen to a macro? |
| 16:52.27 | sumasuma | ddunavant: not getting about your requirement |
| 16:52.33 | sumasuma | can you please explain a bit ? |
| 16:52.34 | [TK]D-Fender | ddunavant: the "M" option allows them to "reject" the call without answer "privacy mode". Since thats its main design concept, that'd be a "no" |
| 16:54.02 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:54.08 | ddunavant | [TK]D: alright, thanks again |
| 16:54.37 | gr0mit | WildPikachu, any success? |
| 16:54.52 | WildPikachu | yes .... callprogress=yes fixes the problem, was on a call for 32 minutes |
| 16:55.08 | gr0mit | how very odd! |
| 16:55.16 | WildPikachu | yes! ... i never tried it before |
| 16:55.19 | WildPikachu | till tonight |
| 16:55.24 | WildPikachu | then disabled it, now tried it again |
| 16:55.42 | WildPikachu | trying another call right now |
| 16:56.15 | gr0mit | another thought |
| 16:56.19 | WildPikachu | um? |
| 16:56.31 | gr0mit | have you got sulence suppression enabled on the X-lite phones? |
| 16:56.36 | gr0mit | silence i mean |
| 16:56.44 | WildPikachu | i think so |
| 16:56.49 | gr0mit | ha |
| 16:56.55 | gr0mit | disable that.... |
| 16:57.24 | gr0mit | asteirsk does not behave well with silence suppession |
| 16:58.01 | gr0mit | wonder if complete silence i.e. no ip packets was upsetting it |
| 16:58.12 | stream | any other voip sip providers you can recommend? |
| 16:58.38 | [TK]D-Fender | gr0mit: Yes, silence suppression can kill calls due to RTP timeout |
| 16:58.39 | WildPikachu | gr0mit, problem is the users are talking all the time, and it cuts out in middle of conversation |
| 16:59.08 | WildPikachu | i was disconnected when I tested this afternoon |
| 16:59.17 | gr0mit | well, I disable slience suppression as a matter of course |
| 16:59.25 | [TK]D-Fender | stream: Other than who? |
| 16:59.31 | gr0mit | it is a cause of Considerable Grief |
| 17:00.07 | stream | i called broadvox and broadband.com |
| 17:00.08 | WildPikachu | i just pressed mute on the phone with the worst problem, disconnects in under 2 mins ... and pressing mute i still see udp in both directions |
| 17:01.14 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 17:01.33 | gr0mit | WildPikachu, these all on the same subnet? |
| 17:01.38 | WildPikachu | yep |
| 17:01.40 | gr0mit | i.e. no NAT |
| 17:01.44 | [TK]D-Fender | WildPikachu: Could be call progress not sensing audio cutting off your call. |
| 17:01.46 | WildPikachu | i'm on a VPN at the moment into the office basically bridged |
| 17:01.54 | [TK]D-Fender | WildPikachu: You were warned |
| 17:02.06 | WildPikachu | [TK]D-Fender, yes i have been warned :) |
| 17:02.16 | gr0mit | wifey has just yelled those wonderful words : dinner! |
| 17:02.18 | gr0mit | bbl. |
| 17:02.19 | WildPikachu | [TK]D-Fender, 20-30 users wanting calls to work vs. me :( |
| 17:02.28 | WildPikachu | gr0mit, mine is about to |
| 17:02.41 | troy- | WildPikachu, just tell them they don't deserve a working phone. |
| 17:02.50 | WildPikachu | haha Trey-- |
| 17:02.56 | WildPikachu | * troy-, |
| 17:03.06 | WildPikachu | they were very angry :( |
| 17:06.16 | WildPikachu | does callprogress use the indications.conf config? |
| 17:07.27 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 17:07.41 | Superbartt | hmmfg, i'm having an slight annoying problem on my linksys phones... When they get ringed by the ring-all from an queue defined in queues.conf, the phone who doesn't pick up the call shows a missed call, would there be a way to prevent that? |
| 17:10.26 | kamh | hi |
| 17:10.38 | kamh | i have a question |
| 17:10.58 | kamh | if i use two IPPhones and one asterisk as a SIP registrar |
| 17:11.07 | kamh | and i make a call |
| 17:11.24 | kamh | who generates a ringback tone or busy tone or call waiting call? |
| 17:11.27 | kamh | asterisk? |
| 17:11.44 | kamh | ...signal |
| 17:14.12 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:15.07 | stream | signed up with vitelity |
| 17:15.09 | stream | good prices |
| 17:16.35 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 17:18.49 | WildPikachu | gr0mit, i'll see what the users say tomorrow |
| 17:20.39 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 17:21.05 | hrmphh | can someone recommend a backup to an ISDN PRI? we have towerstream for internet at 8mbps, is it possible to route over that to some voip provider? |
| 17:21.56 | *** join/#asterisk MihiNomenEst (n=argh@65-85-36-130.client.dsl.net) |
| 17:25.38 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 17:25.39 | *** mode/#asterisk [+o file] by ChanServ |
| 17:26.20 | *** join/#asterisk Porpes (n=Porpes@189-19-2-118.dsl.telesp.net.br) |
| 17:33.34 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:36.18 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 17:37.11 | WilliamK | can someone tell me howto fix this in the zaptel compile (just updated from svn) |
| 17:37.14 | WilliamK | make[1]: *** No rule to make target `zaptel-fw-oct6114-064-1.05.01.tar.gz', needed by `hotplug-insta |
| 17:37.31 | WilliamK | TE120 card |
| 17:38.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:38.37 | *** join/#asterisk implicit (n=bayan@dhcp-x221-142.mobile.uci.edu) |
| 17:39.03 | Superbartt | WilliamK couldn't you get a stable instead of cutting edge svn? ^^ |
| 17:39.38 | WilliamK | always use SVNs AFTER I read the patches |
| 17:39.43 | unpaidbill | <PROTECTED> |
| 17:40.03 | WilliamK | and if noone says anything about a compiler error, it'd never get fixed/resolved |
| 17:40.27 | Superbartt | yeah ofcourse, but if you don't have a clue how to fix it, just report it ^^ |
| 17:41.24 | WilliamK | Super, that's why I logged back into IRC - I've been around the asterisk community enough to know that there are ways of resolving issues |
| 17:41.36 | WilliamK | and yes I know about bugtracker |
| 17:42.13 | *** join/#asterisk hyegeek (n=hakimian@rw.aha.com) |
| 17:43.52 | tzafrir_laptop | http://svn.digium.com/view/dahdi?view=rev&revision=4490 <-- any chance this happen to help? |
| 17:44.07 | WilliamK | heya tzafrir - long time no c |
| 17:44.23 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) [NETSPLIT VICTIM] |
| 17:44.23 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) [NETSPLIT VICTIM] |
| 17:44.23 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) [NETSPLIT VICTIM] |
| 17:44.23 | *** join/#asterisk sack (n=sack@42.Red-83-44-211.dynamicIP.rima-tde.net) [NETSPLIT VICTIM] |
| 17:44.31 | tzafrir_laptop | WilliamK, that's a makefile. No C |
| 17:44.36 | WilliamK | same exact |
| 17:44.57 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 17:45.53 | tzafrir_laptop | Note that even if this actually works, I have no idea why |
| 17:46.00 | *** join/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 17:46.39 | WilliamK | this is make 3.80 under CentOS |
| 17:47.11 | tzafrir_laptop | centos4? 3? |
| 17:47.23 | WilliamK | 4 |
| 17:47.37 | iamhrh | is there a current 'best practice' for setting up a continuous stream for the music on hold? I've been reading about ways to connect to an icecast server - just wanted to make sure that using the "streamplayer" tool is what i want to use for that |
| 17:49.29 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 17:54.50 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:58.02 | gr0mit | anyone got an clever ways of grabbing the IP address of an IAX or SIP peer in the dialplan without resorting to chopping up parameters? |
| 17:58.51 | seanbright | with sip you can use SIPPEER(ip) i think |
| 17:59.06 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 17:59.16 | gr0mit | and for iax? |
| 17:59.20 | Mike8861 | hello all |
| 17:59.30 | seanbright | IAXPEER(ip) |
| 17:59.31 | seanbright | :) |
| 17:59.52 | seanbright | gr0mit: core show function SIPPEER |
| 17:59.56 | seanbright | gr0mit: core show function IAXPEER |
| 18:00.16 | Mike8861 | i got a question. I am dialing to a external IVR, is there any fuction asterisk can sent DTMF to external IVR when connected ?? |
| 18:00.33 | iamhrh | i believe there is a SendDTMF() command... |
| 18:00.33 | gr0mit | hmmm ok, so If it could be either, need to test for whether it is set or not |
| 18:00.45 | gr0mit | thanks, seanbright |
| 18:01.03 | iamhrh | yes there is, Mike8861: SendDTMF([digits]) |
| 18:01.04 | seanbright | Mike8861: there is always ExternalIVR() |
| 18:01.19 | seanbright | (kidding) |
| 18:01.40 | Mike8861 | iamhrh: thank you so much |
| 18:01.41 | iamhrh | err, excuse me...its SendDTMF(digits[,timeout]) |
| 18:02.00 | Mike8861 | SendDTMF(digits[|timeout_ms]) |
| 18:02.05 | Mike8861 | yup, i got that |
| 18:04.31 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 18:05.04 | *** join/#asterisk Pctech37|Mac (n=Pctech37@unaffiliated/pctech37) |
| 18:07.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:09.26 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:23.23 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 18:24.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:26.22 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 18:28.40 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 18:30.26 | *** join/#asterisk Segnale007 (n=Segnale0@host163-249-dynamic.23-79-r.retail.telecomitalia.it) |
| 18:30.54 | Pctech37|Mac | Ok, I installed asterisk and then asterisk-gui |
| 18:31.50 | Qwell | congratulations |
| 18:31.58 | Segnale007 | lol |
| 18:32.17 | WildPikachu | i gave a client my gxp2000 today :(, wonder if a snom 320 is a better phone for me |
| 18:32.18 | Pctech37|Mac | but when i login to the gui, i get asterisk restarting with check on top right after logging in. |
| 18:32.35 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
| 18:32.55 | *** join/#asterisk redax (i=redax@r6.hu) |
| 18:32.55 | Pctech37|Mac | nvm |
| 18:32.59 | Pctech37|Mac | lol |
| 18:33.02 | Pctech37|Mac | sorry |
| 18:33.21 | redax | hi, |
| 18:33.42 | redax | would you help, pls? where can I find the Asterisk book in PDF? |
| 18:33.48 | ViKing78 | Does anyone use Queuemetrics or something similar for monitoring call queues? |
| 18:33.58 | Qwell | ~book |
| 18:33.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:34.27 | redax | thanks Qwell, |
| 18:35.02 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 18:37.16 | Segnale007 | I bought it :D |
| 18:37.41 | Segnale007 | I found it very interested .. |
| 18:41.19 | Mike8861 | iamhrh: this line is not working, it got some syntax error |
| 18:41.24 | Mike8861 | exten => s,1,Dial(SIP/922720000,15) |
| 18:41.51 | Mike8861 | i cannot get into senddtmf unless the dial line got fix |
| 18:42.18 | Mike8861 | whats wrong with that syntax, it is not possible to dial out 922720000 with SIP trunk |
| 18:43.37 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.225) |
| 18:43.51 | Mike8861 | yup, senddtmf works, i do heard dtml on my softphone, incase, i am not sure if sendddtmf does send it out to the external ivr |
| 18:44.13 | Daejeo | whaT is the full form of "ENUM"? anyone knows? |
| 18:45.04 | Mike8861 | Telephone Number Mapping |
| 18:45.30 | Mike8861 | Daejeo: tElephone NUmber Mapping |
| 18:45.35 | Daejeo | what is E? |
| 18:45.45 | Daejeo | I SEE |
| 18:45.50 | Daejeo | :) |
| 18:45.55 | Daejeo | sorry caps was on |
| 18:45.59 | Mike8861 | Daejeo: can u help me with dialplan ?? |
| 18:46.07 | Daejeo | shoot |
| 18:46.26 | Mike8861 | whats my syxtax error |
| 18:46.42 | Daejeo | what is your current dial plan? |
| 18:47.08 | Mike8861 | exten => s,1,Dial(SIP/922720000,15) |
| 18:47.20 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 18:47.23 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 18:47.41 | Mike8861 | i need the dial plan to dial to 922720000 with a SIP trunk name SIP/trunk1 |
| 18:48.28 | *** join/#asterisk klathzazt (n=Jon@69.10.64.241) |
| 18:48.58 | Daejeo | i usually use freepbx |
| 18:49.34 | Daejeo | but let me have a look on an asterisk dial plan |
| 18:49.41 | *** join/#asterisk [netman] (n=netman@251.Red-83-63-246.staticIP.rima-tde.net) |
| 18:57.04 | Daejeo | do u want to use it for incoming call? |
| 18:57.48 | Daejeo | do you want to use it for dialing out? |
| 18:58.26 | Mike8861 | i want to use it to dial out |
| 19:06.03 | Daejeo | do u want to put 9 on the front of your dial pattern? |
| 19:06.54 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 19:07.06 | Daejeo | do u want to dial a local number? long distance? |
| 19:07.39 | Mike8861 | yes, outgoing trunk requires a 9 |
| 19:07.57 | Mike8861 | the point is, i dunno where to put the number in the command.... |
| 19:08.28 | Mike8861 | s,1,Dial(SIP/922720000,15) <= maybe the number are locate in the wrong parameter |
| 19:08.38 | Daejeo | first u need to add a variable |
| 19:08.59 | Daejeo | s,1,Dial(SIP/922720000,15) <<<< this is for incoming calls |
| 19:09.36 | Daejeo | [globals] |
| 19:09.38 | Daejeo | JOHN=Zap/1 |
| 19:09.39 | Daejeo | JANE=SIP/jane |
| 19:09.41 | Daejeo | OUTBOUNDTRUNK=Zap/4 |
| 19:09.51 | Daejeo | [outbound-local] |
| 19:09.53 | Daejeo | exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 19:09.54 | Daejeo | exten => _9NXXXXXX,2,Congestion( ) |
| 19:09.56 | Daejeo | exten => _9NXXXXXX,102,Congestion( ) |
| 19:10.25 | Daejeo | It will help u |
| 19:10.35 | Mike8861 | thanks, i will try |
| 19:11.13 | [netman] | ~centos52bug |
| 19:11.14 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 19:12.08 | Daejeo | s.1. Dial(SIP/3333) this kind of pattern we use for incoming calls |
| 19:12.30 | Daejeo | call will land at sip exten 3333 |
| 19:12.59 | Daejeo | then u can play with it |
| 19:14.26 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:15.01 | Daejeo | ${EXTEN:1} will strip off 9 |
| 19:15.17 | Daejeo | for example: u dial 912323233234 |
| 19:16.42 | shido6 | after 3 weeks of 2000 users accessing voicemail 24/7 ( 30 users simultaneously ) with verbose set to "3" asterisk became unresponsive, do you think it was all the scrolling ? does that fill up memory? |
| 19:16.51 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:16.59 | WildPikachu | gr0mit, i dont spose callprogress should work for me ... very strange it is |
| 19:17.12 | Daejeo | your trunk will dial trunk1/12323233234 |
| 19:18.00 | *** join/#asterisk khfj (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 19:18.05 | khfj | hi $ |
| 19:19.22 | khfj | i got sipura 3000 which is connected to my asterisk server on the port FXS and pstn port by FXO |
| 19:19.51 | khfj | by default i can only use my sip line how to use my pstn ? |
| 19:23.36 | Mike8861 | Daejeo: well, i have tried, but cannot make it to work |
| 19:23.44 | khfj | hello |
| 19:24.15 | khfj | by default i can only use my sip line how to use my pstn ? |
| 19:24.16 | Mike8861 | Daejeo: on my server: there is already trunks setup, it can call outside pstn network |
| 19:25.15 | Mike8861 | Daejeo: i would need to create a dialplan, when user dial 9001, it will dial a PSTN number 922720000, when connected, send a few DTML singal out to the PSTN |
| 19:25.45 | angryuser | have you heard orelly in europe is closed ? |
| 19:25.52 | angryuser | bankrupt |
| 19:25.58 | khfj | i don"t know if ppl understand my english |
| 19:26.06 | *** join/#asterisk cjk_ (n=cjk@d90-129-44-87.cust.tele2.lu) |
| 19:26.14 | Daejeo | speed dialing? |
| 19:26.16 | khfj | simply how to use my pstn line ? |
| 19:26.33 | khfj | from my sipura3000 |
| 19:26.40 | Daejeo | u just want to dial 9001? |
| 19:27.08 | angryuser | <khfj> red sipura manual, you have a prefix or thing like that |
| 19:27.27 | khfj | yeah |
| 19:27.32 | khfj | i can't find |
| 19:27.34 | khfj | it |
| 19:27.54 | khfj | what is term to make this option possible ?? |
| 19:28.07 | khfj | how the option is called ??? |
| 19:28.09 | Mike8861 | khfj: if i remember correctly 3000 is replaced by a new model, anyway, goto the vendor website to get infomation |
| 19:28.32 | Qwell | 3102 I think? |
| 19:29.03 | khfj | if i understand correctly i have to compose 3000 or 3102 to switch to my pstn line is it ??? |
| 19:29.13 | Mike8861 | @Qwell: indeed |
| 19:29.45 | khfj | on sipura3000 ??? |
| 19:29.51 | khfj | no not working |
| 19:30.31 | khfj | is it to switch between the pstn line and the sip line ? |
| 19:30.39 | Mike8861 | khfj: are u trying to use sipura as a GATEWAY to connect asterisk and PSTN ? |
| 19:31.00 | khfj | no |
| 19:31.06 | khfj | simple configuration |
| 19:31.38 | khfj | on my sipura3000 i connect my pstn line and my asterisk sip line |
| 19:31.51 | khfj | by default when i turn on the phone |
| 19:32.05 | Daejeo | _9XXXXXXXXXXXXXX,1,Dial(SIP/trunk1${EXTEN}) |
| 19:32.16 | khfj | i can only use my sip line |
| 19:32.21 | Daejeo | Mike8861: |
| 19:32.22 | Mike8861 | Daejeo: yes, when user dial ext 9001, i hope it can dial a PSTN number, and setout dtmf |
| 19:32.31 | Daejeo | yes |
| 19:32.39 | Daejeo | it is possible |
| 19:32.57 | khfj | my question is how to use my pstn line ? |
| 19:33.19 | khfj | how to switch to my pstn line to use on a same phone |
| 19:33.32 | khfj | ? |
| 19:33.58 | Mike8861 | khfj: please specify by "using your PSTN" line, and what excatly you want to accomplish, please provide more detail info |
| 19:34.50 | khfj | for example i connect British Telecom line on the FX0 port of the sipura3000 |
| 19:35.09 | khfj | and my sip extension line on the port FXS |
| 19:35.31 | khfj | the two line used by one phone |
| 19:36.00 | Mike8861 | Daejeo: thanks, do i replace _9XXXXXX with the real pstn number ? or just left that alone ? |
| 19:36.03 | khfj | when i turn on the phone i use my sip line |
| 19:36.14 | khfj | how to use pstn line ? |
| 19:36.43 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:36.51 | unpaidbill | ~astbook |
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| 19:36.58 | unpaidbill | ~book |
| 19:36.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:37.21 | unpaidbill | <-- training to replace fender |
| 19:37.26 | unpaidbill | i kid i kid |
| 19:38.17 | khfj | helo |
| 19:39.08 | khfj | is there any french iher ? |
| 19:39.10 | Qwell | unpaidbill: gonna have to do better than that |
| 19:40.18 | Mike8861 | Daejeo: here is the dialplan, http://www.pastebin.sk/en/7242/ |
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| 19:46.37 | thehar | Is anyone familiar with idle images on a Polycom IP 600 on the 2.2.2 software? |
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| 20:05.04 | jaytee | ~pb |
| 20:05.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 20:05.50 | Daejeo | User can also call our application from the following numbers directly without dialing any pin or extension numbers. |
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| 20:13.01 | jaytee | has anyone here used SIPTAPI to do click to call from Outlook to another * user? |
| 20:15.03 | unpaidbill | so sleepy |
| 20:15.07 | unpaidbill | oops, wrong window |
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| 20:19.42 | WildPikachu | gr0mit, ok ... i'm using a x-lite soft phone again now |
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| 20:24.01 | gr0mit | WildPikachu, working ok ? |
| 20:24.11 | WildPikachu | 2.5 mins so far, hold thumbs |
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| 20:26.07 | Qwell | Great_Anta_Baka: "Great"? A bit presumptuous, don't you think? :) |
| 20:26.42 | WildPikachu | past the 5 min mark gr0mit .... made over 20 test calls today, never got this far |
| 20:27.01 | gr0mit | cool¬ |
| 20:27.06 | gr0mit | what did you change? |
| 20:27.56 | WildPikachu | callprogress=yes :) |
| 20:28.03 | WildPikachu | fixed all my problems |
| 20:28.16 | anonymouz666 | really? |
| 20:28.16 | gr0mit | hmmmm |
| 20:28.18 | WildPikachu | and i'm not even in us, uk or whatever it supportes |
| 20:28.21 | WildPikachu | yes |
| 20:28.25 | WildPikachu | very very very very strange |
| 20:28.26 | anonymouz666 | callprogress creates problems ;) |
| 20:28.29 | gr0mit | i think it will have cancelled out anohther prpm |
| 20:28.32 | gr0mit | prb |
| 20:28.36 | WildPikachu | anonymouz666, fixes mine :( |
| 20:28.48 | WildPikachu | gr0mit, yea |
| 20:28.49 | gr0mit | WildPikachu, I think there has to be more to this |
| 20:28.57 | WildPikachu | oh, that silence thing was off |
| 20:29.09 | gr0mit | hmmm ok |
| 20:29.09 | WildPikachu | gr0mit, i'm free to test out again tomorrow evening .... i just must get my usres happy |
| 20:29.14 | WildPikachu | they were sooo angry today :( |
| 20:29.26 | gr0mit | how many users? |
| 20:29.39 | WildPikachu | 20-30 on x-lite, about 5 on proper phones |
| 20:29.55 | gr0mit | eeew - thats a lot of angre peeps |
| 20:29.56 | WildPikachu | a atcom had the same problem, MUCH WORSE than anything else, cut off after 30s |
| 20:29.59 | WildPikachu | thats also fixed |
| 20:30.11 | gr0mit | so 20-30 users on 3 analogue lines?! |
| 20:30.21 | WildPikachu | i tested that first this evening while i got my vmware->rdesktop->ssh forwarded device -> office pbx up ;) |
| 20:30.30 | WildPikachu | gr0mit, 3 analogue, 2 premicells |
| 20:30.37 | WildPikachu | mostly software support guys |
| 20:30.41 | WildPikachu | wait ... |
| 20:30.41 | gr0mit | ah ok |
| 20:30.48 | WildPikachu | there is more lines than that i think |
| 20:30.48 | gr0mit | dropped? |
| 20:30.55 | WildPikachu | all x-lite calls were dropped |
| 20:30.59 | WildPikachu | i confirmed with 5 users |
| 20:31.07 | WildPikachu | on 4-5 min mark, 100% reproducable |
| 20:31.17 | WildPikachu | the atcom, before 1.5 min every single call |
| 20:31.56 | WildPikachu | the snom's and grandstreams, no problem at all |
| 20:32.13 | gr0mit | WildPikachu, have you thought of using a local voip provider? |
| 20:32.28 | WildPikachu | gr0mit, i got management issues there, i have investigated it, yes |
| 20:32.43 | WildPikachu | problem is our dsl line is 4mbit, we have 2 of them, with 384k upload |
| 20:32.53 | WildPikachu | most of the day they maxed with oss iso downloads |
| 20:33.16 | gr0mit | best thing is to use one line just for voip |
| 20:33.22 | gr0mit | run G726 |
| 20:33.25 | WildPikachu | gr0mit, becomes very expensive :( |
| 20:33.48 | gr0mit | must be cheaper than telkom;s tariffs |
| 20:33.54 | WildPikachu | normal line is just about $20, then its $50 for the DSL portion, then its nearly $20 per phone number with 2 concurrent sessions |
| 20:34.03 | WildPikachu | excluding bandwidth |
| 20:34.09 | WildPikachu | which is about $10 p/Gbyte |
| 20:34.15 | WildPikachu | (all rough values) |
| 20:34.38 | gr0mit | ouch - expensive for phone numbers |
| 20:34.59 | WildPikachu | yep, i'm still in negotiations with them tho ... we're an isp wholesaler :) |
| 20:35.09 | gr0mit | ah ok |
| 20:35.16 | WildPikachu | looking to take a few trunks into our own pbx to supply to clients |
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| 20:42.57 | jshriver | greetings |
| 20:43.14 | jshriver | What would cause a busy signal when dialing in even if the line is open? |
| 20:44.09 | romanc | ~centos52bug |
| 20:44.10 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 20:44.11 | jshriver | Seems like I can call other asterisk servers, but when someone calls in using a pots line it's always busy even if it's open |
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| 20:51.30 | jaytee | jshriver, the SIP phone is busy? |
| 20:51.45 | jshriver | the phone line itself. |
| 20:51.51 | jshriver | pots phone coming in |
| 20:52.19 | jaytee | pots line to FXO port in * to "then what?" |
| 20:52.20 | jshriver | when I check it with a plain phone the line sounds fine and I can call in, but when I plug it into the asterisk box it's always busy. Seems like it just started that yesterday |
| 20:53.11 | jaytee | did this work before? |
| 20:54.13 | unpaidbill | hrm |
| 20:54.30 | jshriver | yeah |
| 20:54.40 | unpaidbill | has anyone come across a firefox addon that does any type of click-to-dial stuff and has the source available |
| 20:54.49 | jshriver | my boss left for vacation and I have about 1-2 hours ast experiance lol |
| 20:55.10 | jaytee | unpaidbill, no but I found one for Outlook that I just got working. If I come across a Firefox one I'll let you know. |
| 20:55.37 | jaytee | jshriver, s'ok. we all start someplace and everyone is a noob at some point. |
| 20:56.00 | jshriver | I tried asterisk -r -vvvvvvvvv to see if I could see anything |
| 20:56.08 | jshriver | at one point it said something about congestion |
| 20:56.13 | jshriver | but none of the phones where in use. |
| 20:56.29 | jaytee | jschriver, so you are dialing in from a pots line to * and then where is the call supposed to go? to a sip phone? to an analog phone on an FXS port? |
| 20:56.55 | jshriver | supposed to go to a sip phone then another, then mailbox I believe |
| 20:57.06 | jaytee | jshriver, do you know how to use pastebin to paste files? |
| 20:57.29 | jshriver | what I'm wondering is the main sip phone died, so I thought it was trying to ring that phone not seeing it, and just crapping out. So I took a spare sip phone set it up like the old main one. Didnt seem to help |
| 20:57.39 | jshriver | jaytee: yeah |
| 20:57.48 | jshriver | I'm not new, just new to * :) |
| 20:57.57 | ManxPower | if an ip is not listed next to the phone in "sip show peers" then the phone is not registered |
| 20:58.14 | jshriver | btw is there a command to see what phones are active/calls etc.. tried help but the list is huge |
| 20:58.20 | jaytee | ok, first let's make sure your zap channels are good to go. type zap show channels at the CLI to check the status. |
| 20:58.27 | jshriver | k |
| 20:58.36 | unpaidbill | jaytee yeah i think im going to write one |
| 20:58.45 | jshriver | listed 2 extensions |
| 20:58.52 | unpaidbill | just a simple regex to detect a phone #, that posts it to a cgi |
| 20:59.20 | jshriver | http://pastebin.com/m3ab5849b |
| 20:59.27 | unpaidbill | i'm hoping it isnt a pain in the ass to make ff extensions |
| 21:00.02 | jaytee | jshriver, are the pots lines part of a hunt group? |
| 21:00.21 | jaytee | or do you know which zap channel is taking the call when you test it? |
| 21:00.29 | jshriver | sorry dont understand hunt group? We have two cards in the box for 2 lines, but only using one of them |
| 21:00.46 | jshriver | tried putting the line on the other and didnt work and broke outgoing. So put it back to the original port1 |
| 21:00.54 | jaytee | jshriver, ok. pastebin your zapata.conf file please |
| 21:01.04 | jshriver | k will try, working remotely. |
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| 21:01.39 | jaytee | no problem, just take your time and then pastebin your extensions.conf too. I'm gonna run to the restroom and I'll be right back. |
| 21:02.01 | jshriver | http://pastebin.com/m359e12bf |
| 21:06.34 | unpaidbill | oh wow, a .xpi is just a zip archive |
| 21:06.35 | unpaidbill | hah |
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| 21:09.03 | jshriver | http://pastebin.com/m107f3888 |
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| 21:14.06 | jshriver | How do you set it to debug mode so I can see what's going on? I tried |
| 21:14.09 | jshriver | asterisk -r -vvvvvvvvvvvvvvvv |
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| 21:16.42 | bijit | any one play counter strike on slack? |
| 21:17.12 | bijit | sorry wrong irc |
| 21:17.21 | jshriver | lol |
| 21:17.29 | jaytee | jshriver, your zapata.conf point your zap channels to the default context but you don't have one in your extensions.conf |
| 21:17.39 | jshriver | let me check |
| 21:18.07 | jshriver | [default] |
| 21:18.11 | jshriver | include => demo |
| 21:18.32 | jshriver | sorry didnt copy the whole thing... I cut out the part that lists our other asterisk servers and phone names |
| 21:18.34 | *** part/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 21:19.23 | jaytee | jshriver, ok because I didn't see a [default] in what you pasted |
| 21:19.36 | jshriver | is it possible it's a HW issue? I checked the backups and ran diffs they are the same so no file has been changed... only that the main phone is down |
| 21:19.42 | jshriver | sorry about that |
| 21:20.04 | jshriver | that's the only thing I can think of that's throwing it off. |
| 21:20.09 | jshriver | but I might be wrong |
| 21:20.11 | jaytee | well, if the main phone is down you'll get a busy or fast busy because the call has no where to go |
| 21:20.25 | jshriver | where would I redirect that? |
| 21:20.29 | jshriver | or which file would have that. |
| 21:20.41 | jaytee | it would be in your extensions.conf |
| 21:20.47 | jshriver | ok looking |
| 21:21.18 | jaytee | can you find the section of it that has the [default] context and pastebin that? |
| 21:21.36 | jshriver | did above just says include => demo |
| 21:21.44 | jshriver | looks like my boss went with a lot of defaults |
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| 21:22.22 | jaytee | yeah, but I don't see a [demo] context either |
| 21:22.37 | jshriver | sorry looking |
| 21:22.43 | Marcums | hey i have a question about using asterisk to setup an auto-dialer |
| 21:22.51 | Marcums | set up* |
| 21:23.08 | jaytee | include => demo just means include the demo context within that context. if there's no match in the context it will then look in the include |
| 21:24.16 | jshriver | http://pastebin.com/m23645897 |
| 21:24.55 | jshriver | operator doesnt look right |
| 21:26.29 | Marcums | can anybody help me? |
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| 21:27.44 | jaytee | nope, it doesn't. it's just pointing to an IAX device not a SIP device. |
| 21:27.52 | jshriver | well the phones are IAX phones |
| 21:28.05 | jaytee | and what's the extension? 300? |
| 21:28.10 | jaytee | cuz that's commented out |
| 21:28.12 | jshriver | 500 300 was the old one |
| 21:28.18 | jshriver | 500-505 |
| 21:28.41 | jshriver | iaxy1 is a weird little doggle that connects via ethernet and lets you use a pots phone on the * server |
| 21:28.52 | jaytee | ok, but I don't see any matches for 500 as an extension except if the IAX call fails and then it goes to voicemail as user505 |
| 21:29.13 | jaytee | so iaxyl is like a SIP/ATA adapter |
| 21:29.18 | jaytee | only for IAX |
| 21:29.27 | jshriver | yeah |
| 21:29.42 | jshriver | rest of the phones are sip phones |
| 21:29.49 | jshriver | looks like iaxy1 is defined in iax.conf |
| 21:29.53 | jaytee | if you type IAX2 show peers at the CLI what do you get? |
| 21:30.09 | jaytee | any IAX phone or trunk is defined in iax.conf |
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| 21:30.38 | jaytee | if the extension was changed in extensions.conf but not in iax.conf that would break it |
| 21:30.54 | jshriver | ok checking now... |
| 21:31.24 | jaytee | I'm getting ready to leave for the day but I'll be back on in an hour or so. What timezone are you in? |
| 21:32.05 | jshriver | EST |
| 21:32.11 | jaytee | same here |
| 21:32.39 | jaytee | jshriver, you in Ohio? |
| 21:32.40 | jshriver | it's ok think I'm going to drive back to the office and work on it more. I think you're on to something.. the iaxy doggle thing is what broke, so that phone is gone and it's 0 operator.. so wondering when someone calls in, it tries to call that line and just dies |
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| 21:34.01 | jshriver | Yup |
| 21:34.09 | jaytee | Indianapolis here |
| 21:34.17 | jshriver | not to far, think about 1.5 hours I've been told |
| 21:34.24 | jshriver | not from here though semi-recent transplant :) |
| 21:34.26 | jshriver | ttyl |
| 21:35.03 | jaytee | but from looking at the demo context the call should make it as far as that and the call should be answered and then play the demo at least. |
| 21:35.55 | jaytee | are you getting that far and then entering the extension you want? |
| 21:36.04 | jaytee | or is it going busy before that? |
| 21:36.53 | jaytee | ok, I'll be on later |
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| 22:03.11 | *** join/#asterisk Docfxit (n=none@cpe-76-95-77-238.socal.res.rr.com) |
| 22:04.00 | Docfxit | Where is a good place to upload a file for all to see? |
| 22:05.08 | lesouvage | I have a persistent problem with sip nat traversal with a sip trunk that runs on every server I tried it on except for the server it should run on. I have paste sip.conf to http://www.pastebin.be/12539 . Did I missed something in what should be in sip.conf. (is also an aswer to Docfxit question) |
| 22:06.23 | lesouvage | www.pastebin.be is a nice place to paste a file for all to see |
| 22:06.32 | Docfxit | Great Thanks. |
| 22:08.37 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:14.12 | ManxPower | the 6060 port might do it |
| 22:14.47 | unpaidbill | wow that was easy as hell. i now have highlight -> right click -> dial in firefox! |
| 22:15.29 | ManxPower | They'll pry my hardphone from my cold dead hands |
| 22:15.56 | unpaidbill | haha |
| 22:15.58 | lesouvage | ManxPower: don't ask me why but that is the port they use for sip signaling. Is the rest as it should be (I checked, double checked and read lots on this subject on the internet but there is always a change of missing the obvious) |
| 22:16.38 | ManxPower | lesouvage: did you confirm the firewall is forwarding UDP 6060 and not UDP 5060? |
| 22:17.47 | ManxPower | lesouvage: do you use SRV? |
| 22:17.55 | lesouvage | ManxPower: I don't have access to the firewall myself, they are doing this for me. I ask them to portforward udp 6060 and udp 10000 to 20000. |
| 22:18.12 | ManxPower | lesouvage: come back when you can confirm it |
| 22:18.44 | lesouvage | ManxPower: but the sip looks ok to you? |
| 22:18.53 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 22:19.32 | ManxPower | lesouvage: other than the oddball incoming port number, np. |
| 22:19.59 | lesouvage | ManxPower: Thanks! |
| 22:20.12 | ManxPower | I expect that or the firewall is your problem |
| 22:24.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 22:26.16 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:28.31 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 22:30.42 | *** join/#asterisk Segnale007 (n=Segnale0@host163-249-dynamic.23-79-r.retail.telecomitalia.it) |
| 22:32.47 | *** join/#asterisk jshriver (n=jshriver@cblmdm72-240-213-73.buckeyecom.net) |
| 22:32.51 | jshriver | hi again |
| 22:32.54 | jshriver | exit |
| 22:33.19 | *** join/#asterisk jshriver (n=jshriver@cblmdm72-240-213-73.buckeyecom.net) |
| 22:33.47 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 22:43.24 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:44.10 | lesouvage | ManxPower: what does srv stands for? |
| 22:50.07 | lesouvage | ManxPower: is setting a SRV record something that is done on feature rich firewalls? |
| 22:50.28 | JT | dns... |
| 22:50.32 | JT | srv is googleable |
| 22:51.07 | jaytee | googleable...hehe...love new words |
| 22:51.08 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-55-174.lns10.syd7.internode.on.net) |
| 22:51.48 | *** join/#asterisk Moudmen (n=Julien@213.254.232.45) |
| 22:52.03 | Moudmen | can anybody plaese help me restore my g729 license ? |
| 22:52.18 | Moudmen | i have the .lic file from the old installation |
| 22:52.55 | drmessano | You need to talk to digium support |
| 22:53.29 | Moudmen | but in the README they said that keeping the .lic file will let me restore the license without having to talk to digium |
| 22:53.56 | Moudmen | This will help prevent you *** |
| 22:53.56 | Moudmen | *** from needing to contact Digium to request authorization to increment your *** |
| 22:53.56 | Moudmen | *** G.729 key and from needing to purchase a new G.729 key if you exceed the *** |
| 22:53.56 | Moudmen | *** maximum number of G.729 key increments allowed. |
| 22:54.00 | Moudmen | this is in the README |
| 22:58.38 | *** join/#asterisk hi365_m (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 22:59.05 | drmessano | Is it the same box? |
| 22:59.43 | drmessano | I mean, not that it MATTERS, but why are you AGAINST calling DIGIUM for SUPPORT? |
| 22:59.49 | Moudmen | yeah dr0ck |
| 22:59.53 | Moudmen | drmessano same box |
| 22:59.57 | Moudmen | im not against |
| 23:00.01 | Moudmen | but it's listed |
| 23:00.03 | Moudmen | :/ |
| 23:00.12 | drmessano | Listed? |
| 23:00.17 | Moudmen | in the README |
| 23:00.53 | drmessano | I'm not even sure what this conversation is about at this point. Digium support can help you. |
| 23:01.20 | drmessano | If there's not instructions in the README |
| 23:01.22 | Moudmen | okay okay, i already msgd them anyway, but it's via email, i wish they had live chat, i need to do this fast |
| 23:01.24 | drmessano | and you don't know HOW |
| 23:01.35 | drmessano | Ok |
| 23:03.00 | *** join/#asterisk unenough (n=unenough@CBL217-132-137-221.bb.netvision.net.il) |
| 23:03.23 | unenough | hi, i installed VoiceGlue but as a * newbie i can't figure how to direct incoming calls to it |
| 23:03.33 | unenough | what do I need to write in extensions.conf? |
| 23:04.06 | *** join/#asterisk Drunktard (n=sebas@201.198.239.167) |
| 23:04.25 | Drunktard | is there any good asterisk AMI perl module? |
| 23:07.35 | outtolunc | there is always asterisk-perl |
| 23:07.55 | outtolunc | but you could just use net::telnet for that matters |
| 23:08.59 | Drunktard | well i'm interested in parsing whatever happens in the console, thought it was already done |
| 23:09.18 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 23:09.21 | outtolunc | the console is the CLI not AMI |
| 23:09.45 | Drunktard | right, the AMI that is |
| 23:10.22 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 23:10.26 | Drunktard | i'm just trying to provide an interface to asterisk events, just know a few things about asterisk... |
| 23:16.49 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 23:20.25 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:24.08 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 23:28.44 | unenough | in extensions.conf, how do I redirect a call to an extension in another context? |
| 23:30.01 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:31.31 | *** join/#asterisk hi365_m (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 23:32.41 | unenough | anybody? |
| 23:33.13 | [netman] | unenough: goto |
| 23:33.37 | unenough | [netman], i'm a asterisk newbie, can you be more explicit? |
| 23:34.00 | [netman] | I'm asterisk newbie too... I'd have to check the documentation |
| 23:34.12 | [netman] | but I remember it's something like goto |
| 23:34.33 | [netman] | in dialplan functions |
| 23:34.49 | unenough | ok |
| 23:34.51 | unenough | thanks! |
| 23:35.53 | unenough | ...Goto(context,extension,priority) |
| 23:35.55 | unenough | thanks. |
| 23:36.52 | [netman] | it was an application, not function , sorry |
| 23:46.06 | jaytee | ~book |
| 23:46.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:46.37 | jaytee | the applications and functions are documented in the Appendices at the end of the book. |
| 23:47.25 | drmessano | I had my appendices taken out |
| 23:47.26 | rift0r | nice, my polycom 550 came in today |
| 23:47.32 | rift0r | is excited to play |
| 23:47.50 | jaytee | me too, I've got a 501 and a 550 to test and play with. |
| 23:48.06 | jaytee | I got Click2Dial from Outlook working with * today! psyched! |
| 23:48.11 | drmessano | I have a shipment of Budgetone 101s coming in from Tunisia |
| 23:48.27 | jaytee | lol, livin on the edge! |
| 23:48.50 | rift0r | drmessano how much |
| 23:49.31 | drmessano | Can't talk about it.. It's highly classified |
| 23:49.35 | drmessano | As a matter of fact |
| 23:49.45 | drmessano | I am redacting this chat window as we speak |
| 23:50.58 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 23:51.03 | *** part/#asterisk romanc (n=rchernob@owa.alicatscientific.com) |
| 23:51.12 | teknoprep | does asterisk work inside of xen with virtual extension CPU ? |
| 23:51.43 | teknoprep | now i know the hardcore nutjobs in here love bare-metal servers with asterisk... but i am asking a real question with the idea of a real answer |
| 23:52.13 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:52.15 | drmessano | HA.. that's a good way to get an answer.. |
| 23:52.21 | teknoprep | yup it is |
| 23:52.32 | teknoprep | because its the truth ? |
| 23:52.40 | teknoprep | and ppl can't handle the idea of being honest ? |
| 23:52.55 | drmessano | Asterisk does run much better on bare metal |
| 23:53.06 | teknoprep | ? |
| 23:53.13 | teknoprep | oh i read that wrong |
| 23:53.17 | teknoprep | i understand what you are saying |
| 23:53.24 | teknoprep | now i understand its not asterisk that is the problem |
| 23:53.40 | teknoprep | but the timing issue behind the guest kernel and the host kernel |
| 23:54.05 | teknoprep | i have asterisk running on a vmware server running.. vmware server 2.0 beta... with centos 5] |
| 23:54.28 | teknoprep | i then used the kernel that they have in there yum repo for proper vm timing... and it worked great |
| 23:54.38 | teknoprep | i was just wondering if the same scenario existed for xen |
| 23:55.13 | drmessano | You know, asterisk works really well on bare metal |
