| 00:00.06 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 00:00.10 | hardwire | jesus |
| 00:00.22 | hardwire | the debian teem is really behind on source packages for 1.2.4 |
| 00:00.28 | hardwire | they are always really behind however |
| 00:00.35 | hardwire | they always end up overpatching |
| 00:00.36 | _Sam-- | lol |
| 00:00.40 | tronix | heh... Gentoo's only up to 1.2.1, last I checked. |
| 00:00.46 | hardwire | yeh |
| 00:00.53 | Qwell[] | with ~x86 maybe |
| 00:00.53 | hardwire | I am going to up to 1.2.4 as soon as debian hits it |
| 00:00.58 | Qwell[] | 1.0.8 otherwise |
| 00:01.01 | hardwire | I am for the most part trying only to use the packages |
| 00:01.11 | _Sam-- | i have a debian 1.2.1 package |
| 00:01.16 | hardwire | using unstable dsc's and diffs on stable |
| 00:01.21 | _Sam-- | if you want the apt source let me know |
| 00:01.26 | [av]bani | debian team is awesomely out of date |
| 00:01.36 | Qwell[] | EVERY package of * is out of date |
| 00:01.38 | hardwire | _Sam--: yeh.. I just compiled it for debian stables base |
| 00:01.47 | Qwell[] | (and is also broken) |
| 00:02.11 | _Sam-- | Package: asterisk |
| 00:02.12 | _Sam-- | Versions: |
| 00:02.12 | _Sam-- | 1:1.2.1.dfsg-3(/var/lib/apt/lists/ftp.de.debian.org_debian_dists_unstable_main_binary-i386_Packages) |
| 00:04.18 | malverian[work] | Damn... after all that work getting this piece of crap to compile correctly I get an error trying to use these OGI diphones. |
| 00:04.24 | malverian[work] | Maybe it only works with festival 1.95 and up... |
| 00:04.40 | masked | tronix brilliant, module loaded |
| 00:04.47 | tronix | masked: yay! |
| 00:05.13 | tronix | masked: also, make sure you modprobe wcfxo too |
| 00:05.21 | tronix | (sometimes it's easy to forget and only load zaptel) |
| 00:05.22 | masked | so that; Zapata Telephony Interface Registered on major 196 idicates that it is using the card? |
| 00:05.27 | masked | oh ok |
| 00:05.38 | litage | are there any other options/settings you can specify in cdr_manager.conf besides "enabled=" ? |
| 00:05.45 | masked | Found a Wildcard FXO: Wildcard X100P |
| 00:05.46 | masked | :) |
| 00:05.47 | masked | neat |
| 00:05.49 | tronix | sweet |
| 00:05.55 | masked | thanks tronix you've been a great help |
| 00:06.03 | tronix | you're welcome; my pleasure |
| 00:07.25 | _Sam-- | can someone explain what this means: "The Rhino channel bank is fully interoperable with Asterisk(tm) PBX. |
| 00:07.25 | _Sam-- | With the "Auto-Config" feature, you can just plug it in, and the channel bank is ready to go. |
| 00:07.26 | _Sam-- | " |
| 00:07.36 | _Sam-- | ready to go where? |
| 00:07.44 | Qwell[] | _Sam--: To the supermarket |
| 00:07.49 | _Sam-- | lol |
| 00:08.04 | Qwell[] | it's hungry...it's a rhino...come on |
| 00:08.05 | _Sam-- | its 'good to go' to taco bell |
| 00:08.18 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 00:09.11 | _Sam-- | like how could there be such a thing a zero config fxs channel bank |
| 00:09.23 | _Sam-- | or, an auto-configged rather |
| 00:09.36 | denon | why not? |
| 00:09.38 | _Sam-- | wouldnt asterisk need to know what extensions are on the thing? |
| 00:09.48 | denon | fxs ports dont have extensions |
| 00:09.50 | Qwell[] | _Sam--: It didn't say Asterisk was auto-config'd |
| 00:09.53 | denon | you point fxs ports TO extensions |
| 00:09.55 | Qwell[] | it said IT was auto-config'd |
| 00:10.01 | denon | er .. extensions TO fxs ports I mean |
| 00:10.14 | _Sam-- | i see. |
| 00:10.33 | _Sam-- | the CHANNEL BANK is ready to go |
| 00:10.33 | _Sam-- | sure |
| 00:10.36 | _Sam-- | what good does that do |
| 00:10.47 | RoyK | echo 220V > /proc/acpi/processor/CPU0/power |
| 00:10.50 | denon | a lot, if you're used to configuring CBs with crappy and undocumented dip switches |
| 00:10.55 | _Sam-- | i see |
| 00:11.01 | troyb | hey slePP |
| 00:11.27 | [av]bani | _Sam--: "dhcp" + "tdmoe" |
| 00:11.41 | [av]bani | i could see pnp channel bank happening |
| 00:11.45 | denon | you know .. tdmoe isnt really .. well .. |
| 00:11.47 | *** part/#asterisk Aldo (n=aleyva@200.62.180.209) |
| 00:11.52 | _Sam-- | ive never worked with a channel bank that is why i have no clue |
| 00:11.57 | denon | IAX2 fits the niche better, I think |
| 00:11.59 | [av]bani | denon: compared to what... sip? |
| 00:12.14 | [av]bani | tdmoe is very simple, easy to implement (=cheap) |
| 00:12.31 | denon | you mean from the device perspective? |
| 00:12.33 | denon | or networking? |
| 00:12.37 | [av]bani | both |
| 00:12.41 | [av]bani | pri interfaces are expensive |
| 00:12.51 | denon | 'cause iax is cheaper than tdmoe, with regards to using existing network infrastructure |
| 00:12.54 | [av]bani | 100 or even 1000 ether parts are dead cheap |
| 00:14.07 | fugitivo | a sip gateway is expensive |
| 00:14.25 | _Sam-- | the way most channel banks would interface with asterisk is through a t1 card? |
| 00:14.33 | denon | bbl: food |
| 00:14.33 | [av]bani | likely |
| 00:14.46 | *** join/#asterisk outtolunc (n=me@adsl-69-110-61-148.dsl.pltn13.pacbell.net) |
| 00:14.58 | Err | does anything other than asterisk support TDMoE? |
| 00:15.11 | [av]bani | Err: tdmoe is used in wireless |
| 00:15.31 | Qwell[] | _Sam--: generally |
| 00:15.44 | Err | used in wireless what? links between asterisk boxes? |
| 00:16.10 | [av]bani | in generic voice applications |
| 00:16.18 | [av]bani | various hardware vendors use tdmoe for voice products |
| 00:16.24 | Qwell[] | If you can do TDMoE over wireless, why not PoE? |
| 00:16.24 | _Sam-- | what is the purpose of interfacing with the channel bank over a t1 as opposed to over regular ethernet or something else? |
| 00:16.40 | Qwell[] | _Sam--: You want to write an ethernet driver for *? |
| 00:16.46 | [av]bani | _Sam--: t1 is legacy |
| 00:16.50 | justinu | cuz most traditional channel banks support T1 |
| 00:16.52 | justinu | not ethernet |
| 00:17.01 | litage | is there a way to search through the asterisk-users mailing list without manually going through each month? |
| 00:17.05 | [av]bani | t1 is 1960s technology |
| 00:17.08 | justinu | yep |
| 00:17.14 | [av]bani | so theres alot of stuff around that uses it :) |
| 00:17.27 | hardwire | _Sam--: I wish they would poost the individual diffs going into asterisk packages in debian |
| 00:17.27 | wunderkin | litage, google, site:lists.digium.com |
| 00:17.39 | hardwire | it would make it easier to submit patchwork and get ahead of the game |
| 00:17.47 | masked | tronix so will asterisk use that card as a timing device by default now? |
| 00:18.05 | _Sam-- | could channel banks have greater density if they interfaced over fca or something? |
| 00:18.17 | [av]bani | _Sam--: "T3" |
| 00:18.31 | *** part/#asterisk rene- (i=rene@201.144.60.114) |
| 00:18.47 | justinu | OC-48 |
| 00:18.52 | justinu | oh noes!!!!!!!1!!! |
| 00:18.54 | [av]bani | OC768 |
| 00:18.57 | [av]bani | O NOES |
| 00:18.57 | _Sam-- | lol |
| 00:19.10 | Err | heh, and then an add/drop so that you can deal with it :-) |
| 00:19.21 | coppice | [av]bani: T1 actually started in the 1950s. its amazing they could make it work with the available components back then |
| 00:19.34 | [av]bani | coppice: they had to, there wasnt enough copper to go around... |
| 00:19.38 | justinu | i liked N carrier better |
| 00:19.46 | justinu | barbed wire fence was the layer1 technology |
| 00:20.22 | _Sam-- | man now i know why everyone just gets new phones instead of channel banks. |
| 00:20.27 | Err | there were several multiplexing schemes pre-T1 - the real problem was the poor audio quality over long hauls |
| 00:20.29 | _Sam-- | the cost per port its just cheaper to get new phones |
| 00:20.46 | [av]bani | _Sam--: exactly, its also easier to deal with switches than miles of rj11 |
| 00:20.51 | justinu | yeah... FDM had notorious issues with static, howling, etc... |
| 00:21.41 | Err | FDM equipment is also *very* expensive, as you need tons of very narrow-band filters |
| 00:21.41 | _Sam-- | outtolunc: if you look at 24 port FXS |
| 00:21.41 | _Sam-- | and you need a t1 card |
| 00:21.41 | justinu | basically captive broadband radio stystems, as I understand it |
| 00:21.41 | outtolunc | i get mine for $100 |
| 00:21.43 | Qwell[] | That's a good $2k right there |
| 00:21.49 | outtolunc | (or less) |
| 00:21.52 | coppice | T1 was a *lot* more expensive than FDM in the early days. |
| 00:22.09 | _Sam-- | so to set up a 24 port FXS channel bank is about 1500-1700 for 24 ports |
| 00:22.11 | Err | justinu: actually, it's a ton of narrow-band receivers - they didn't have DSPs, so there was no way to decode directly from the broadband signa |
| 00:22.15 | Err | +l |
| 00:22.19 | coppice | the last of the FDM kit was all DSP, and very stable. It worked greate just before it was scraped :-) |
| 00:22.23 | _Sam-- | and you still end up with crappy phones...if you want crappy phones, you can get them for 85 bucks each from grandstream |
| 00:22.26 | [av]bani | you also lose a lot of the functionality of voip with fxs |
| 00:22.38 | justinu | Err: interesting |
| 00:22.43 | [av]bani | a crappy gxp will give you more functionality than fxs+channelbank |
| 00:22.49 | _Sam-- | 85 * 24 = about the same cost |
| 00:22.55 | _Sam-- | as the 24 port FXS channel bank |
| 00:23.01 | tronix | masked: sorry, was in another window. :) I'm not familiar with the X100P (other than it's popular) but think so |
| 00:23.17 | Err | I have a hard time believing that there were DSP systems capable of decoding multiple channels from a broadband source in the 50s |
| 00:23.28 | masked | tronix if it is infact a timing device, there shouldn't be any more configuation needed? |
| 00:23.37 | tronix | masked: not for the timing aspect, no. |
| 00:23.48 | masked | tronix thanks |
| 00:23.48 | outtolunc | sam i think you forgot the cost of the switch in there |
| 00:23.52 | justinu | well, i thought they used radio demuxing techniques |
| 00:23.53 | tronix | masked: think you'd want to review /etc/zaptel.conf though to make sure sane settings. also |
| 00:23.56 | _Sam-- | a 24 port switch is a 100 bucks |
| 00:23.57 | outtolunc | if you want to be exact |
| 00:23.59 | tronix | masked: what country you in? |
| 00:24.09 | tronix | masked: dumb question. sorry. :) Oz, I see |
| 00:24.17 | tronix | masked: reason why I asked was because there may be a few |
| 00:24.24 | tronix | masked: settings you will want to set differently for Oz. |
| 00:24.31 | [av]bani | Err: they didnt use DSPs, at least not in the traditional sense |
| 00:24.40 | Err | justinu: they did - many receivers *is* radio demuxing :-) In some sense it's a broadband signal, that's then re-received into a bunch of individual signals... |
| 00:24.44 | tronix | masked: the default settings are generally US-centric, I'm afraid. most are fine, but just a few tweaks. |
| 00:24.46 | _Sam-- | my network works fine, thanks! |
| 00:25.02 | _Sam-- | my internet speed is 2mbps...i dont think my switch is holding me up |
| 00:25.12 | masked | tronix orright, in the zaptel.conf? |
| 00:25.22 | [av]bani | _Sam--: GbE is only $150 :) and $50 a port |
| 00:25.24 | tronix | masked: I don't recall Oz-specific settings offhand, but i've seen them in various Google searches |
| 00:25.31 | masked | ok |
| 00:25.37 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 00:25.42 | _Sam-- | i have gbe between some computers that actually move files around |
| 00:25.52 | _Sam-- | but for general users browsing web and emailing, they wouldnt even notice the diff |
| 00:25.58 | [av]bani | its nice when the disk is the bottleneck, not the network :) |
| 00:26.02 | masked | tronix i assume i have to reload the modules after altering the conf? |
| 00:26.03 | coppice | [av]bani: yes they did. the last FDM kit, built to link the new digital systems into old FDM ones were 100% DSP. At least the ones we built were |
| 00:26.14 | tronix | masked: it's just 'cause your telcos do technical things a little differently. which is normal for a lot of countries. |
| 00:26.18 | tronix | masked: aye |
| 00:26.20 | [av]bani | coppice: DSP as in silicon |
| 00:26.21 | Err | coppice: how recent was that? |
| 00:26.32 | Err | was this to interface to back-woods FDM links? |
| 00:26.37 | [av]bani | coppice: no silicon DSPs in the 1950s i think... |
| 00:26.38 | X-Rob | ooh, I heard australia mentionde |
| 00:26.41 | masked | tronix yeah thanks for the pointer |
| 00:26.42 | X-Rob | Morning all. |
| 00:26.46 | masked | mornin' X-Rob |
| 00:26.47 | tronix | masked: welcome |
| 00:26.59 | coppice | the last of the FDM kit was built in the late 70s and early 80s |
| 00:27.01 | Err | ...because my parents' phone system had FDM coming into it into the 80s, but that was not the norm |
| 00:27.01 | Qwell[] | masked: When you get echo problems in a few minutes, talk to X-Rob :p |
| 00:27.07 | masked | haha ok |
| 00:27.08 | [av]bani | coppice: maybe valves :) |
| 00:27.22 | Err | oh yeah, I'd buy that - FFT was around by then, along with other important DSP breakthroughs |
| 00:27.31 | _Sam-- | anyone have verizon FIOS (fiber) for their home yet? its available at my house |
| 00:27.33 | [av]bani | i could see them doing some real basic digital AD/DA and TDM with valves |
| 00:27.43 | Qwell[] | _Sam--: Get it |
| 00:27.44 | tronix | morning, X-Rob |
| 00:27.58 | Err | [av]bani: heh, you just described the early TDM T1 systems :-) |
| 00:27.59 | _Sam-- | i want to see if my dsl reseller can resell it to me before i sign up |
| 00:28.07 | justinu | vavles == tubes |
| 00:28.14 | [av]bani | Err: "yay my TDM takes 15 minutes to warm up" |
| 00:28.14 | justinu | for those who don't know :P |
| 00:28.15 | Qwell[] | _Sam--: They have access to the coper, not the fiber |
| 00:28.18 | Err | haha, yeah |
| 00:28.19 | coppice | Err: FFT goes back 100 years, although it was "reinvented" by Cooley and Tukey more recently :-) |
| 00:28.21 | _Sam-- | bani what do you think...if my dsl provider can resell verizon dsl, can they resell fios? |
| 00:28.37 | [av]bani | _Sam--: no idea, ask them? |
| 00:28.43 | Qwell[] | Verizon is gonna keep their fiber closed until they're forced not to |
| 00:28.48 | _Sam-- | i believe it |
| 00:28.51 | X-Rob | And 'lo everyone back. |
| 00:29.03 | Err | coppice: well, there's a difference between someone having discovered the equation and someone putting it in a useful application, right? :-) |
| 00:29.07 | _Sam-- | you think it would be a bad idea to try to run any type of commercial server/service off a home fios connection? |
| 00:29.26 | Qwell[] | _Sam--: I don't think it's disallowed... |
| 00:29.31 | Qwell[] | in fact, I bet 80 is wide open |
| 00:29.37 | X-Rob | masked, I realise I'm coming in late in the convo, what hardware are you using? |
| 00:29.45 | coppice | Err: well, people did put them to use, but with a computer who drew a salary for the work :-) |
| 00:30.03 | _Sam-- | ty as always for the info |
| 00:30.38 | tronix | masked: one last comment... the card-related tweaks would generally be done in /etc/zaptel.conf (for hardware side) and /etc/asterisk/zapata.conf (for asterisk side, its interface to the zaptel) |
| 00:30.41 | masked | X-Rob x100p |
| 00:30.48 | X-Rob | coppice, possibly not as 'fast' as they are now 8) |
| 00:30.54 | masked | thanks tronix |
| 00:30.55 | X-Rob | masked, ick. You'll have echo problems. |
| 00:31.11 | [av]bani | _Sam--: one way to find out! |
| 00:31.27 | masked | X-Rob it was $20... |
| 00:31.28 | _Sam-- | i dont want to put a hurtin on your isp business! |
| 00:31.38 | X-Rob | masked, yeah. You got ripped 8) |
| 00:31.42 | tronix | X-Rob: curious why that's so? something about the technical implementations by the Oz telcos? |
| 00:31.42 | masked | lol |
| 00:31.46 | X-Rob | Sorry. |
| 00:31.47 | tronix | X-Rob: re: echo issues |
| 00:31.50 | X-Rob | Unless you're _really_ close to the exchange. |
| 00:32.09 | masked | nah think i should be about 4km from it |
| 00:32.20 | coppice | X-Rob: those early DSP engines were pretty fast, as long as they didn't run out of fingers :-) |
| 00:32.27 | masked | yep |
| 00:32.27 | X-Rob | tronix, we have 600ohm exchanges. US has 900ohm. That gives an impedance mismatch and, end result == massive echo |
| 00:32.28 | masked | 4km |
| 00:32.30 | Qwell[] | "_really_ close" is far less than 4km |
| 00:32.39 | tronix | X-Rob: ahh! that's interesting. thanks |
| 00:32.42 | masked | heeh |
| 00:32.50 | ljam | anyone use ALERT_INFO to change the ring tones on the 7960? |
| 00:32.57 | _Sam-- | qwell i know you probably have no idea, but do you have any idea what backbone networks verizon peers with? :) |
| 00:32.59 | justinu | anyone have an explanation as to why some places us a-law, and some places use u-law? |
| 00:33.07 | Qwell[] | _Sam--: a bunch, I'm sure |
| 00:33.07 | justinu | it must have a slightly different sound to it |
| 00:33.07 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:33.15 | X-Rob | masked, look, it's a good thing to play with. Expect echo. You _may_ be able to get close to beating it with asterisk-trunk and the kb1 patches |
| 00:33.21 | coppice | X-Rob: the US uses 600 ohm. the difference is what they do about complex impedance |
| 00:33.26 | Qwell[] | _Sam--: Especially considering Verizon now owns gtei... |
| 00:33.26 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 00:33.33 | X-Rob | coppice, ok, well then we use 900ohm then. |
| 00:34.04 | X-Rob | There is a difference, I know this from looking at the zaptel code. |
| 00:34.07 | _Sam-- | they probably route all their cheap residential internet traffic (like me) over some cheap interconnect |
| 00:34.40 | _Sam-- | but hell it is worth a shot |
| 00:34.51 | _Sam-- | 30mbps down / 5 up |
| 00:34.57 | masked | X-Rob can u point me to a sample conf for the x100p in .au? |
| 00:35.10 | masked | is that all _Sam--? :P |
| 00:35.27 | _Sam-- | i think they have a more expensive package for 45 down :) |
| 00:35.51 | coppice | X-Rob: there are differences for lots of places, but its mostly the phase shift. only a few places are 900. most are 600. if you don't get the complex impedance right, though, the mismatch is terrible |
| 00:35.52 | X-Rob | masked, zaptel.conf == 'fxsks=1', 'loadzone=au', 'defaultzone=au' |
| 00:35.54 | Qwell[] | _Sam--: 2, 15, 30 |
| 00:35.57 | Qwell[] | 30 isn't worth it |
| 00:36.03 | Qwell[] | 30 jumps up to like $200/mo |
| 00:36.05 | _Sam-- | its funny...im in a pretty rural, but developing area...and we couldnt even get dsl here last year |
| 00:36.07 | [av]bani | X-Rob: US has 600ohm |
| 00:36.11 | _Sam-- | now you cant get dsl, just fios |
| 00:36.25 | justinu | well, you skipped a whole generation |
| 00:36.38 | mattwj2005 | what is the netsec version of * all about? |
| 00:36.46 | mattwj2005 | what is the difference? |
| 00:36.49 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:36.54 | X-Rob | [av]bani, yes, coppice has corrected me. |
| 00:36.58 | [av]bani | yay! |
| 00:37.19 | X-Rob | I would like to rephrase my previous statement to 'The exchanges are different' |
| 00:37.23 | _Sam-- | 30 may be worth it if i need the 5 up |
| 00:37.30 | X-Rob | If I dumb it down enough it's correct enough 8) |
| 00:37.32 | _Sam-- | but you're right 30 is the cap |
| 00:37.41 | _Sam-- | and its 180 a month, if you commit for 12 |
| 00:38.06 | Qwell[] | _Sam--: 15 is about $50, $40 in some markets |
| 00:38.06 | tronix | _Sam--: certainly much cheaper than my T-1, and with better speeds in both directions |
| 00:38.24 | _Sam-- | tronix: your t1 has a guarentee if it breaks that someone is going to fix it, quick. |
| 00:38.26 | _Sam-- | i hope. |
| 00:38.27 | Qwell[] | _Sam--: Convince your whole neighborhood to get it, and do a bandwidth sharing scheme |
| 00:38.34 | tronix | _Sam--: point. |
| 00:38.38 | Qwell[] | Verizon gets their money, you get your pr0n...it's win/win! |
| 00:38.44 | masked | X-Rob, ok so will that config allow the fxs and fxo ports to work? |
| 00:38.44 | blitzrage | Qwell[]: lol |
| 00:38.45 | _Sam-- | lol |
| 00:39.14 | _Sam-- | tronix, how much you pay for a ptp t1? |
| 00:39.18 | _Sam-- | for the line and internet charges |
| 00:39.18 | Qwell[] | You all know I'm right |
| 00:39.50 | tronix | masked: the x100p is a fxo card. you have to 'speak' in the "opposite" protocol, so that means you have to enable FXS for the card. that's why the fxsks=1, so it can talk with your FXS device you'll hook up to it |
| 00:39.53 | justinu | what if you don't download porn |
| 00:39.55 | _Sam-- | i have a couple t1s that i pay too much for, but i like the isp and their bgp |
| 00:39.56 | justinu | do you lose? |
| 00:40.02 | Qwell[] | justinu: You lose anyways |
| 00:40.07 | tronix | _Sam__: hmm my T-1 at home runs my company about um.. I want to say $330/mo? |
| 00:40.07 | mattwj2005 | netsec?? |
| 00:40.13 | MikeJ__ | Qwell[]!!!! |
| 00:40.23 | _Sam-- | 330 is a pretty good deal, at least compared to what i pay :) |
| 00:40.24 | Qwell[] | MikeJ__: You dirty socks asshole! |
| 00:40.29 | justinu | Qwell[]: why? |
| 00:40.29 | MikeJ__ | yes? |
| 00:40.36 | Qwell[] | hi :D |
| 00:40.37 | _Sam-- | i pay 425 but my isp is good |
| 00:40.42 | Qwell[] | justinu: dunno |
| 00:41.06 | tronix | _Sam--: heh, in my case, my company *is* the isp with a nice backbone, so it's definitely good to have 2ms latency for ssh to my office gear. :-) |
| 00:41.26 | masked | tronix oh yeah, thats ryte. but this card has two ports on the back, one is called line and other phone. |
| 00:41.29 | _Sam-- | our office and my home dsl are on the same provider, so im about 10ms away |
| 00:41.42 | masked | tronix or otherwise this is a dodgy clone with a modem backing on it |
| 00:41.44 | AndyCap | mattwj2005: http://www.voip-info.org/wiki/view/Asterisk+security |
| 00:41.46 | *** part/#asterisk DarkFlibble (n=darkflib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com) |
| 00:41.47 | [av]bani | we cheat, we use LADS to home |
| 00:41.53 | tronix | masked: ahhh! does sound like a fxo+fxs card. do mind that I know nothing about the card other than it's popular. :-) |
| 00:41.58 | _Sam-- | that is dry alarm type circuits? |
| 00:42.01 | tronix | _Sam--: sweet |
| 00:42.03 | [av]bani | yeps |
| 00:42.04 | _Sam-- | LADS? |
| 00:42.13 | [av]bani | 'local area data set', dry copper |
| 00:42.19 | _Sam-- | those are still distance sensitive though right? |
| 00:42.22 | [av]bani | yep |
| 00:42.23 | masked | tronix they could be wrongly indicated, i only knew of it as a fxo card until it arrived today |
| 00:42.25 | [av]bani | totally |
| 00:42.35 | _Sam-- | i had one of them when ihad my isp |
| 00:42.40 | _Sam-- | i lived like three blocks away from the office |
| 00:42.49 | _Sam-- | told the phone company we were running an alarm circuit |
| 00:42.49 | [av]bani | i live pretty close to the office so i leech 2mbps :) |
| 00:42.54 | [av]bani | for $15/mo |
| 00:42.56 | _Sam-- | they setup the dry pair and i bought some pairgain i think |
| 00:43.00 | masked | tronix so should i enable fxo for the fxs port and see if it works? |
| 00:43.14 | [av]bani | we dont tell them what we are running, dont need to |
| 00:43.17 | [av]bani | they dont need to know |
| 00:43.20 | [av]bani | :) |
| 00:43.24 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 00:43.31 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 00:43.34 | tronix | masked: here's what i'd suggest... do a search for voip-info.org and x100p. there's config examples there i think. that, in addition to what X-Rob suggested. |
| 00:43.39 | _Sam-- | it was kind of odd...i owned some video stores..and iw anted a circuit from the video store to my house (but int he video store was housed all the isp equipment) |
| 00:43.45 | _Sam-- | they did ask what i was doing |
| 00:43.47 | masked | ok tronix thanks |
| 00:43.57 | [av]bani | they dont have any right to know, you can lie :) |
| 00:43.59 | X-Rob | masked, xrobau@gmail.com |
| 00:44.03 | [av]bani | you're running candy canes out your butt |
| 00:44.07 | X-Rob | and register your phone number with e164.org |
| 00:44.17 | justinu | i got fava beans comin' out my ass!! |
| 00:44.17 | masked | X-Rob yeh im gunna do that shortly |
| 00:44.22 | tronix | [av]bani: "I'm the last honest burglar remaining... that's why I want a burglar alarm line, you know..." :-) |
| 00:44.36 | [av]bani | we're using them to power our torture devices in our remote office |
| 00:44.46 | _Sam-- | i dont think they have to provide dry copper to anyone |
| 00:44.52 | _Sam-- | its not a tarriffed kind of service? |
| 00:44.53 | [av]bani | yes they do, if they can |
| 00:45.03 | _Sam-- | ok then i never knew |
| 00:45.04 | [av]bani | it's tariffed here |
| 00:45.12 | [av]bani | they cant play favorites |
| 00:45.12 | tronix | they did change some stuff around a while ago for this kind of reason |
| 00:45.32 | *** join/#asterisk andio (n=andio@port-195-158-165-243.dynamic.qsc.de) |
| 00:45.42 | [av]bani | well when people started going wifi they realised, a few $/mo for lads was better than $0/mo if they customer went wifi |
| 00:45.46 | [av]bani | so.... |
| 00:45.48 | tronix | the ILEC here (whom I also used to work for) got tired of losing money to people getting smart about these circuits so they got the rules changed. |
| 00:45.49 | mattwj2005 | thanks andy |
| 00:45.51 | mattwj2005 | :) |
| 00:46.10 | Qwell[] | tronix: Getting smart about the circuit? |
| 00:46.15 | _Sam-- | at the time i did this is was probably 1998 or so, so maybe things really are/were different then |
| 00:46.28 | *** join/#asterisk iq (n=iq@71-214-5-12.omah.qwest.net) |
| 00:46.34 | [av]bani | tronix: the ilec here tried to change the rules, but the ISP associations raised a stink and the tariff stayed |
| 00:46.38 | _Sam-- | then a cable modem came along and there was no need for that anymore |
| 00:46.53 | [av]bani | tronix: also helped that the ILEC got fined a few tens of millions of dollars recently for anticompetetive behavior |
| 00:47.14 | tronix | Qwell[]: people were ordering the lines as 'burglar alarm circuits' and saving huge bundles of money instead of ordering them as DSL lines |
| 00:47.14 | [av]bani | so the PUC wasnt really sympathetic to their claims LADS was killing them :) |
| 00:47.37 | _Sam-- | it was the same exact circuit that my t1 for ISP at that time was implemented over |
| 00:47.38 | [av]bani | the savings over DSL isnt so great, its the savings over T1 |
| 00:47.40 | andio | i upgraded from 1.0.9 to 1.2 and i noticed that the agi command "stream file" doesn't work any more. instead i only can use "control stream file", but that one can't be interrupted by entering digits. what happened to "stream file" function? |
| 00:47.48 | [av]bani | at the time we were using LADS, dsl wasnt available |
| 00:47.49 | _Sam-- | except they were charging 500 a month for the t1 pair, and 15 a month for the dry copper |
| 00:48.12 | [av]bani | one thing though, lads is ultra reliable. because the ILEC cant fuck anything up |
| 00:48.17 | [av]bani | its just a pair of wires punched down |
| 00:48.19 | blitzrage | anyone ever change the ring tones per line, or via Asterisk, on the 7960 ? |
| 00:48.24 | [av]bani | none of their crapola in between |
| 00:48.25 | justinu | ~lads |
| 00:48.37 | [av]bani | in the 10 years we've had LADS circuits in, none of them have ever gone down |
| 00:48.46 | [av]bani | while our T1, DS3, etc have had regular outages |
| 00:49.05 | justinu | what's lads? |
| 00:49.11 | [av]bani | hell, we've totally lost all local dialtone but the LADS stayed up :) |
| 00:49.41 | andio | anyone got an idea why "stream file" doesn't work as expected any more in 1.2.x versions? |
| 00:49.42 | tronix | justinu: basically 'raw' lines (used to be known as burglar alarm lines) between two points; you could run DSL or whatever over them |
| 00:49.49 | _Sam-- | <[av]bani> 'local area data set', dry copper |
| 00:49.53 | _Sam-- | lads |
| 00:49.58 | *** join/#asterisk btoe1 (n=nick@adsl-71-131-254-137.dsl.sntc01.pacbell.net) |
| 00:50.03 | [av]bani | different ilecs call them different things |
| 00:50.08 | [av]bani | in qwest territory its LADS |
| 00:50.14 | *** join/#asterisk AJMn (i=AJay@63.231.252.9) |
| 00:50.43 | _Sam-- | the distance/speed limit of the circuit is about the same as dsl? 1mbps at about 18,000 feet? |
| 00:51.04 | AJMn | anyone got there hands on a UTstarcom F3000 yet? |
| 00:51.15 | justinu | thx |
| 00:51.17 | _Sam-- | they are out now? i have an f1000 |
| 00:51.22 | btoe1 | Hi, n00b q: Is Asterisk a good starting point if all I want is to have a FreeBSD box act as an answering machine, and convert the msgs to voicemail? Seems the setup and equipment are expensive and overkill for this need. |
| 00:51.42 | btoe1 | (sorry, convert the msgs to email attachments) |
| 00:51.47 | AJMn | voipsupply is taking preorders... im sure someones got one somewhere :P |
| 00:51.49 | _Sam-- | my guy at voipsupply.com says the f3000 is going to be a big seller, but i cant see it anyplace. |
| 00:52.12 | tronix | btoe1: sure, you don't really need hardware if you take the inbound calls via a VOIP provider |
| 00:52.26 | AJMn | Sam did u look at the other flip phone voipsupply has? price is huge!!! big has a cool pop3 interface. |
| 00:52.28 | _Sam-- | AJMn: it is just a nicer F1000? |
| 00:52.39 | btoe1 | the calls come in through my home analog phone line, so I guess I just need a modem? |
| 00:52.48 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 00:53.07 | tronix | btoe1: in that case, yes, you do, 'cause it's analog stuff, and you need hardware to convert from analog to digital |
| 00:53.08 | btoe1 | I saw the syst3m video that used a Sipura 3000(?) box, for $100, plus a whole separate machine. |
| 00:53.13 | _Sam-- | i dont see the f3000 on voipsupply |
| 00:53.19 | _Sam-- | er |
| 00:53.21 | AJMn | Sam the 3000 is a flip phone... |
| 00:53.23 | _Sam-- | i guess i need to look harder |
| 00:53.24 | _Sam-- | i see it now |
| 00:53.28 | AJMn | http://www.voipsupply.com/product_info.php?products_id=1193 |
| 00:53.31 | *** part/#asterisk outtolunc (n=me@adsl-69-110-61-148.dsl.pltn13.pacbell.net) |
| 00:53.31 | AJMn | theres the link same |
| 00:53.33 | AJMn | sam |
| 00:53.45 | btoe1 | tronix: thanks, btw. |
| 00:53.59 | justinu | so if you want LADS, do both ends have to be on the same CO? |
| 00:54.05 | _Sam-- | i wouldnt buy another utstarcomm phone just yet....the f1000 while it works ok, it just works ok |
| 00:54.22 | tronix | btoe1: welcome. hardware doesn't have to be too bad. voipsupply.com has various stuff, or there's digium's direct, or various hardware |
| 00:55.12 | _Sam-- | i need to get setup as a dealer with atacomm |
| 00:55.25 | *** part/#asterisk andio (n=andio@port-195-158-165-243.dynamic.qsc.de) |
| 00:55.27 | _Sam-- | voipsupply has dealer pricing...but their dealer prices are like atacomms regular prices |
| 00:55.33 | justinu | lol |
| 00:56.12 | Jabroni | guys i have a question regarding to mysql() app, im trying to fetch 2 columns of a table, and asigning them to a variable, but for some reason it just passes 1 variable |
| 00:56.27 | Jabroni | heres the link for the code im using http://pastebin.ca/39347 |
| 00:56.55 | tronix | justinu: don't think so. the telco can do their internal CO-to-CO routing |
| 00:57.02 | btoe1 | thanks again |
| 00:57.28 | Qwell[] | btoe1: Why not just like...get an answering machine? |
| 00:57.32 | Qwell[] | They can be had for $20 |
| 00:57.45 | AJMn | _Sam-- whats atacomm's website? |
| 00:57.59 | Qwell[] | atacomm sucks more than voipsupply |
| 00:58.03 | AJMn | lol nevermind |
| 00:58.10 | _Sam-- | fair enough |
| 00:58.11 | Qwell[] | crap selection, crap prices... |
| 00:58.14 | AJMn | then who do you suggest for wifi phones? |
| 00:58.19 | _Sam-- | their prices are way cheaper than voipsupply |
| 00:58.26 | _Sam-- | at least on every item ive every looked at |
| 00:58.27 | Qwell[] | _Sam--: For very few things. |
| 00:58.32 | SocialD | yo! |
| 00:58.33 | btoe1 | Qwell[]: sure, but I'd prefer to get msgs as email, because I want to hook machine off my fax machine, which is in an inconvenient place. |
| 00:58.35 | justinu | i like atacomm |
| 00:58.40 | _Sam-- | which is mainly gxp2000 and some asterisk cards |
| 00:58.52 | SocialD | j00 |2 |=uck1|\|9 31337 |
| 00:59.00 | justinu | voipsupply shipped me a non-working 7960g, a non-working power brick, and took a month to complete the RMA process. |
| 00:59.02 | Qwell[] | hell, I think they only sell one or two cisco phones |
| 00:59.03 | justinu | i was not amused. |
| 00:59.16 | Qwell[] | justinu: I wouldn't buy from voipsupply either :P |
| 00:59.26 | _Sam-- | voipsupply has muffed 2 of my orders in the past...i ordered via overnite some gxp2000s.... |
| 00:59.36 | _Sam-- | they shipped them out overnite...but the next day the package was returned back to them |
| 00:59.40 | _Sam-- | because they put 2 labels on the box |
| 00:59.45 | justinu | lol |
| 00:59.50 | _Sam-- | that is 1 of 2 |
| 00:59.51 | Jabroni | personally ive had good experiences with voipsupply |
| 01:00.03 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:00.09 | Ariel_ | hello everyone |
| 01:00.11 | _Sam-- | they fixed the problem and it was a non-issue |
| 01:00.18 | _Sam-- | and i was pleased with how they handled the problem |
| 01:00.29 | Qwell[] | _Sam--: They sure as hll better have reimbursed the overnight shipping |
| 01:00.34 | Jabroni | they order processing is really good |
| 01:00.59 | _Sam-- | i always call in my orders....so anyway, the 2nd muff... |
| 01:01.04 | oatis | Hi, my asterisk server is behind my router at the office and I would like to use it from home too... what ports do I need to forward in the router at the office to the asterisk server so I can do this? |
| 01:01.06 | oatis | Im using SIP |
| 01:01.10 | _Sam-- | i call in an order for (guess) more gxp2000s...and they say they have them... |
| 01:01.14 | justinu | he said muff |
| 01:01.15 | Jabroni | i tried ordering onces from voxilla store just to save a few bucks... was a nightmare.. ended up ordering again with voipsupply |
| 01:01.21 | justinu | oh, speaking of that |
| 01:01.24 | _Sam-- | so three days later i dont have the phones, and i dont have a tracking number...so i call them up... |
| 01:01.32 | *** part/#asterisk btoe1 (n=nick@adsl-71-131-254-137.dsl.sntc01.pacbell.net) |
| 01:01.33 | _Sam-- | "OHHH...we are waiting on another shipment" |
| 01:01.37 | justinu | i forgot to mention that it took the 7960g's over 2 weeks to get to me after ordering them |
| 01:01.38 | _Sam-- | OK...you TOLD me you had them here. |
| 01:01.44 | justinu | the powerbricks came next day |
| 01:01.53 | X-Rob | _Sam--, speaking of GXP-2000's, you got the new firmware yet? |
| 01:01.56 | X-Rob | It's _hawt_ |
| 01:01.59 | X-Rob | and it doesn't suck!! |
| 01:02.01 | X-Rob | *amazed* |
| 01:02.06 | _Sam-- | X-Rob: got it, i wrote it! (kidding of course)) |
| 01:02.10 | AJMn | Who do you suggest to order from then guys??? VOIPSUPPLY SUCKS AND SO DOES atacom.... /!?! NOW WHO!? :P |
| 01:02.12 | _Sam-- | but yeah its a big step! |
| 01:02.16 | mzo_ | how much are those phones now? |
| 01:02.24 | Qwell[] | mzo_: $300 |
| 01:02.25 | tronix | AJMn: maybe ask "who sucks less?" :-) |
| 01:02.26 | Qwell[] | or so |
| 01:02.30 | justinu | they wouldn't give me a tracking number for the 7960g's even tho they claimed they shipped. |
| 01:02.31 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 01:02.37 | AJMn | Tronix Whos sucks least? lol |
| 01:02.37 | mzo_ | i mean the gxps, i can't afford ciscos :P |
| 01:02.39 | Qwell[] | of course, "those phones" isn't much of a qualifier |
| 01:02.41 | Ariel_ | voipsupply is getting better |
| 01:02.42 | _Sam-- | justinu: sounds like my gxp situation. |
| 01:02.49 | _Sam-- | they probably had to wait to get the phones they told you they had in stock |
| 01:02.53 | Jabroni | X-Rob havent found any issues with that beta firmware ? |
| 01:03.06 | justinu | _Sam--: i'm sure, but they should have admitted it... fucktards |
| 01:03.07 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 01:03.09 | X-Rob | Jabroni, not yet, but I've only got it on one phone so far |
| 01:03.10 | AJMn | whats better... Zyxel P2000w or UTStarcom F1000? |
| 01:03.15 | X-Rob | it's been up for a couple of days w/o a crash. |
| 01:03.17 | _Sam-- | the gxp2000s are selling for about 85 bucks. |
| 01:03.21 | Ariel_ | oatis, 5060/61 and rtp what you setup in the rtp.conf file |
| 01:03.26 | oatis | If my asterisk server is behind a router do I just have to forward UDP port 5060 to it so I can access from outside? |
| 01:03.28 | mzo_ | are they good? I have a few budgetones, i'd like to replace one. |
| 01:03.40 | slan | X-Rob: Which is the new Grandview firmware? 12? |
| 01:03.42 | oatis | Ariel, oh thank you |
| 01:03.43 | _Sam-- | they are good for what they are...everyone has an opinion of the phone. |
| 01:03.48 | mzo_ | yeah for $85? |
| 01:03.52 | _Sam-- | i use them fine and have no complaints for what they are |
| 01:03.56 | mzo_ | i mean my budgetones are abused. I killed them tons :P |
| 01:04.09 | X-Rob | slan, http://www.grandstream/BETATEST/GXP2000 |
| 01:04.16 | Ariel_ | AJMn, I have used the UTSTarcom but found the Zyxel hard to get working on asterisk |
| 01:04.19 | X-Rob | was release a couple of days ago. Huge amount of changes. |
| 01:04.30 | X-Rob | uh |
| 01:04.32 | slan | X-Rob: Thanks. Is this version 11 or 12? |
| 01:04.35 | _Sam-- | i have the utstarcomm...its just ok |
| 01:04.41 | X-Rob | No, it's not 12 or 12. |
| 01:04.42 | _Sam-- | it stops working every now and then and have to reboot it |
| 01:04.43 | X-Rob | 11 or 12 |
| 01:04.43 | Qwell[] | cisco 7920 ;] |
| 01:04.47 | _Sam-- | and the sound quality is marginal |
| 01:04.53 | mzo_ | i have a zyxel, it's weird |
| 01:04.54 | X-Rob | slan, http://www.grandstream.com/BETATEST/GXP2000 |
| 01:04.55 | X-Rob | even |
| 01:05.03 | mzo_ | zyxels need way better firmware |
| 01:05.06 | X-Rob | I posted to -users about how to upgrade |
| 01:05.16 | justinu | zyxel is pretty suck ass |
| 01:05.30 | mzo_ | yeah, i got one of the phones free and it just sits there looking pretty. |
| 01:05.36 | _Sam-- | i think the ustarcomm F1000 - like the grandstream gxp2000 - can be a good phone when the firmware gets there |
| 01:05.36 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 01:05.39 | Ariel_ | _Sam--, yes but the others are hitachi and Zyxel are hard to get working with asterisk |
| 01:05.55 | mzo_ | the zyxel was tedious having to through their own gui and then the handset and having to reboot hte phone after EACH CHANGE |
| 01:05.59 | AJMn | Ariel_ I have 2 Zyxel's and work fine.. would like to see how much better sound quaility is from a F10000 |
| 01:05.59 | Ariel_ | _Sam--, they have a new one out 3.8 which now has a web gui |
| 01:06.05 | oatis | Ariel, _just_ UDP? or do I forward the TCP ports too? |
| 01:06.10 | Ariel_ | udp |
| 01:06.13 | _Sam-- | Ariel_: where could i get that? |
| 01:06.16 | _Sam-- | ive checked their site just today |
| 01:06.22 | Ariel_ | from your vendor |
| 01:07.07 | mzo_ | heh my xp100s are sharing irqs with strange things. :P |
| 01:07.23 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-222-26-17.msy.bellsouth.net) |
| 01:07.24 | slan | X-Rob: Thanks. I looked at the release notes and looks much better. Ver is 1.02 vs 1.01 series before. |
| 01:07.28 | Ariel_ | mzo_, ok disable strange things |
| 01:07.34 | X-Rob | yeah |
| 01:07.41 | Zodiacal | if i make a change to zapata.conf , how can i reset it so i don't have to reboot? |
| 01:07.46 | mzo_ | oh, haha, i know it works fine, i just noticed it a few days ago, that it's sharing an irq with the onboard sound card. |
| 01:07.50 | Zodiacal | whats the best way |
| 01:07.58 | Ariel_ | service zaptel restart |
| 01:08.02 | Zodiacal | ariel Thank You! |
| 01:08.09 | _Sam-- | Ariel_: my vendor isnt ip-phone-forum.de, but they have the firware :) |
| 01:08.14 | *** part/#asterisk tainted- (n=identd@adsl-71-129-32-116.dsl.irvnca.pacbell.net) |
| 01:08.28 | *** join/#asterisk oatis_ (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net) |
| 01:08.54 | AJMn | mzo_ want to sell your extra Zyxel? ;) |
| 01:08.59 | oatis_ | Ariel, my router doesnt seem to have rtp port forwarding, only udp and tcp. is that going to be a problem? |
| 01:09.11 | Ariel_ | rtp uses udp |
| 01:09.17 | Jabroni | rtp port is not a type of protocol |
| 01:09.17 | mzo_ | it's the older one, not the one everyone likes. It's crap ;P save your money |
| 01:09.25 | mzo_ | it's not worth buying, really |
| 01:09.36 | Jabroni | rtp is like saying http/ftp etc, which uses udp, as ariel said |
| 01:09.56 | slan | X-Rob: Do you know where the new firmware has to be in Asterisk for sftp? What name and under which directory (boot?) ? |
| 01:09.56 | oatis_ | oh, okay, wasn't aware of that.. thanks you guys |
| 01:10.08 | X-Rob | slan? |
| 01:10.13 | Ariel_ | set the range like from 10000 to 11000 in the rtp.conf file located in the /etc/asterisk and then forward those ports |
| 01:10.18 | slan | X-Rob: yes? |
| 01:10.28 | X-Rob | You unzip it to your web server's root. Then point the update at http://your.web.server/ |
| 01:10.56 | slan | X-Rob: So no subdirectory involved at all? Just the file? |
| 01:11.02 | X-Rob | there's 4 files. |
| 01:11.07 | X-Rob | stick them all somewhere web accessable |
| 01:11.15 | X-Rob | then tell the phone to get them from that location |
| 01:11.21 | X-Rob | There's no limits. You can put it in a directory if youw ant. |
| 01:11.40 | oatis_ | Ariel, would 10000 to 10100 be to few ports? my router won't let me add a range value spaning longer than 100 |
| 01:11.41 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 01:11.42 | slan | X-Rob: Thanks a lot for that info. I've been wanting to upgrade from 1.09 on 13 phones! |
| 01:11.45 | AJMn | Ohhh Ver 1 ... ya thats a piece of crap |
| 01:11.46 | AJMn | haha |
| 01:12.03 | Ariel_ | oatis, that should be fine |
| 01:12.19 | AJMn | ok guys.. im using Broadvoice right now .. Yuck I know.. Who would be a better company to go through? yet cheap like Broadvoice? |
| 01:12.19 | _Sam-- | slan: that is BETA firmware |
| 01:12.22 | oatis_ | K, thnx |
| 01:12.22 | _Sam-- | which means there may issues |
| 01:12.31 | _Sam-- | like the phone display no longer lights up when people call |
| 01:12.51 | *** join/#asterisk heath__ (n=heath__@12-215-33-205.client.mchsi.com) |
| 01:12.59 | _Sam-- | but i think it by far does more good than bad in my opion |
| 01:13.05 | Ariel_ | _Sam--, I found that the F1000 sound best with g729 as the codec. |
| 01:13.17 | _Sam-- | thank you i just got the new firmware, have to give it a shot |
| 01:13.18 | slan | _Sam--: Yes. I'll be careful, test out on just one first! |
| 01:13.47 | X-Rob | _Sam--, turn on 'use backlight all the time' |
| 01:13.48 | Jabroni | X-Rob do u use tftp for provisioning the phones ? |
| 01:13.50 | X-Rob | everyone does 8) |
| 01:14.03 | X-Rob | Jabroni, nah. Bugger that. Why bother with tftp when the phone understands http? |
| 01:14.04 | _Sam-- | i dont want to burn out my display! |
| 01:14.12 | X-Rob | The _point_ of tftp is that it's an extremely simple protocol to implement |
| 01:14.23 | X-Rob | _Sam--, ALl my phones have the lights on all the time. |
| 01:14.23 | litage | which is the best site to use for searching the asterisk mailing lists? gmane.org, mail-archive.com or asteriskguru.com ? |
| 01:14.26 | _Sam-- | hence the T in tftp? |
| 01:14.36 | X-Rob | In 5 years, I'll tell you if any of the LED's have failed. |
| 01:14.41 | Ariel_ | google |
| 01:14.42 | Jabroni | well the question is that if you have a cfgxxxxx.txt example |
| 01:15.49 | *** join/#asterisk oatis__ (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net) |
| 01:15.50 | X-Rob | _Sam--, Average lifespan of a blue LED is 100,000 hours before falling to 80% of original brightness. |
| 01:16.06 | X-Rob | eg, about 11 years. |
| 01:16.09 | masked | X-Rob is the 'phone' port on the x100p useless? |
| 01:16.13 | X-Rob | masked, yes |
| 01:16.22 | X-Rob | it's a single port FXO card. |
| 01:16.24 | _Sam-- | grandstream usies the budget blue LED, its only good for 20,000 hours! |
| 01:16.45 | masked | X-Rob yeh thats the impression i got from the docs |
| 01:17.03 | X-Rob | masked, the reason why it has a phone socket on it, is that that card is actually a winmodem. |
| 01:17.13 | masked | yeah i gathered that now |
| 01:17.30 | masked | X-Rob so you'd recommend a tdm now? |
| 01:17.47 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 01:17.52 | X-Rob | masked, yeah. The X100 is good for playing with, but if you actually want to use it, get a TDM4xx |
| 01:18.22 | masked | well it was just planned to be used for the incoming line at my parents place |
| 01:18.45 | masked | but if u really think its no good, ill scrap it, an ata might be an even better choice in the end.. |
| 01:19.04 | [av]bani | ... |
| 01:19.09 | Ariel_ | at parents use a good ata like sipura 1001 |
| 01:19.47 | X-Rob | wtf. |
| 01:20.06 | _Sam-- | aireal to update the f1000, i put the phone in local tftp mode then start the fwupgrade.exe? |
| 01:20.21 | Ariel_ | yes |
| 01:20.21 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 01:20.21 | _Sam-- | or does the windowx machine need to be running tftpd |
| 01:20.24 | [av]bani | utstarcomm is a good wifi phone? |
| 01:20.30 | masked | Ariel_ will look inti it |
| 01:20.47 | masked | ahh speaking of the wifi phones |
| 01:21.09 | masked | i ordered one then it turned out it wasn't available, im thinking of getting a senao 680 |
| 01:21.34 | AJMn | ok guys.. im using Broadvoice right now .. Yuck I know.. Who would be a better company to go through? yet cheap like Broadvoice? |
| 01:21.40 | _Sam-- | [av]bani: there is some firmware out for it that i didnt know about, maybe it will help the f1000 |
| 01:22.52 | *** join/#asterisk hack8086 (i=ircap8@116-37-112.adsl.terra.cl) |
| 01:23.50 | oatis__ | is it possible to register multiple extentions for a user? |
| 01:24.13 | _Sam-- | Ariel_: do you know any easy way to get the ip of the f1000 phone if the firmware utility doesnt locate it? |
| 01:24.21 | *** part/#asterisk hack8086 (i=ircap8@116-37-112.adsl.terra.cl) |
| 01:24.24 | oatis__ | can I just do regexten=1,2,3 ... ? |
| 01:24.28 | _Sam-- | i guess i could log into my wireless router but thats still a pita |
| 01:27.46 | *** join/#asterisk stephen_d (n=stephen@70.53.220.101) |
| 01:28.24 | *** join/#asterisk sack (n=sack@113.Red-81-34-163.dynamicIP.rima-tde.net) |
| 01:29.14 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:29.15 | *** join/#asterisk Three (n=vircuser@70.53.220.101) |
| 01:29.33 | X-Rob | heh |
| 01:29.39 | X-Rob | one person |
| 01:29.59 | *** join/#asterisk Three (n=vircuser@70.53.220.101) |
| 01:29.59 | stephen_d | i bet you 50 that hes gonna come bck |
| 01:30.06 | *** mode/#asterisk [+b *!*=vircuser@70.53.220.*] by drumkilla |
| 01:30.06 | *** kick/#asterisk [Three!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by drumkilla (drumkilla) |
| 01:30.07 | X-Rob | too late |
| 01:30.24 | Ariel_ | _Sam--, wifi-settings > Network Parameter > IP |
| 01:30.24 | [av]bani | just pm an admin to have him banned from freenode |
| 01:30.26 | _Sam-- | ariel thank you...updating phone now (too me a while to find the 888888 code for local tftp server) |
| 01:30.36 | drumkilla | [av]bani: already banned him from here, at least |
| 01:30.45 | [av]bani | drumkilla: better to ban from freenode totally :) |
| 01:30.50 | [av]bani | much more satisfying |
| 01:30.52 | drumkilla | yup |
| 01:31.00 | [av]bani | those russian criminals a few weeks ago, are banned |
| 01:31.17 | _Sam-- | that is a nice firmware updater. |
| 01:32.01 | Ariel_ | _Sam--, did you get it updated takes a while |
| 01:32.03 | [av]bani | should get the channel registered and only allow registered users in |
| 01:32.06 | _Sam-- | hope mine isnt in german when its done :) |
| 01:32.08 | [av]bani | that would block 99.9% of the abusers |
| 01:32.12 | masked | Ariel_ _Sam-- so you are happy with the f1000? |
| 01:32.31 | Ariel_ | masked, it's ok. real desk phones are better but it works. |
| 01:32.58 | Ariel_ | I like my Aastra 480i CT with it's 2.4ghz wireless phone better. |
| 01:33.03 | _Sam-- | masked: i wasnt so thrilled with it that is why i am trying some new firmware.....i think its just an ok phone that works just ok...the buttons are small for many people, the sound is just ok, and the phone needs to be rebooted now and then to make or receive calls |
| 01:33.07 | masked | mum likes her cordless phone, but its a 2.4g, doesn't go well with the wifi |
| 01:33.14 | _Sam-- | i would rather buy a plantronics cs50 usb and use a softphone for wireless |
| 01:33.51 | masked | Ariel_ you dont get interference with your wifi? |
| 01:33.59 | Ariel_ | _Sam--, I actually think it's too small |
| 01:34.05 | Ariel_ | no |
| 01:34.27 | masked | the panasonic phone here causes havok to the wifi and the phone line |
| 01:34.33 | _Sam-- | the web interface for the f1000 is always available when the phone is on? |
| 01:34.50 | Ariel_ | _Sam--, don't know but I would guess it is |
| 01:35.03 | _Sam-- | the menus seem quicker |
| 01:35.29 | Ariel_ | the change log says they fixed some sound issues with asterisk |
| 01:35.38 | _Sam-- | arieel do you know the username and password combo for the web interface? |
| 01:35.47 | tronix | wonder if it'd help any to put the wifi networking gear on a separate channel? |
| 01:36.13 | masked | tronix i've tried changing channels |
| 01:36.15 | stephen_d | <PROTECTED> |
| 01:36.29 | _Sam-- | he pressed alt-f4 |
| 01:36.31 | masked | tronix and all i have noticed is that on channel one, the phone wont work.. |
| 01:36.39 | masked | sorry.. |
| 01:36.49 | masked | the wifi wont work on channel 1 if the phone is being used |
| 01:37.06 | masked | otherwise, whenever the phone is in use, the wifi drops out no matter what channel is being used |
| 01:37.13 | [av]bani | hmm.. any particular reason why one wouldnt want to use qualify=yes ? |
| 01:37.39 | tronix | masked: was afraid of that given how certain wireless technologies works. was a long shot. |
| 01:37.49 | Jabroni | could someone please check my code http://pastebin.ca/39347 for some reason it wont get 2 fields from the mysql table using the mysql() app... it justs gets the 2nd one (im using the syntax from voip-info.org) |
| 01:38.03 | _Sam-- | that firmware is better...the sound quality is better. |
| 01:38.18 | _Sam-- | there is nothing that tells the default username/pass for the web interface |
| 01:41.18 | JamesDotCom | admin / psw |
| 01:41.32 | Ariel_ | _Sam--, admin psw |
| 01:41.33 | JamesDotCom | _Sam--: ^^ |
| 01:41.39 | _Sam-- | brilliant! |
| 01:41.51 | _Sam-- | now i just need some Chinese language packs! |
| 01:41.57 | tronix | :-) |
| 01:42.00 | JamesDotCom | haha |
| 01:42.14 | _Sam-- | thank you both |
| 01:42.43 | _Sam-- | damn and all this time ive used those terrible little keys on the phone keypad to configure this thing |
| 01:43.01 | JamesDotCom | haha, yeah, they're pretty terrible to configure through the handset |
| 01:43.47 | _Sam-- | especially wep keys :) |
| 01:44.02 | Ariel_ | it's so small that sometimes I take it home with me and forget that I need to charge it. |
| 01:45.14 | _Sam-- | it sounds 10x better...there used to be some background static when the volume was above 3 or 4 |
| 01:45.16 | _Sam-- | that is gone |
| 01:45.33 | _Sam-- | i probably had a really old firmware |
| 01:46.02 | _Sam-- | im surprised that g729 is the best sounding codec |
| 01:47.44 | slan | X-Rob: I'm ready to tftp the new Grandstream firmware. How does tftp know _where_ on the server to look? |
| 01:47.45 | *** join/#asterisk flujan (i=flujan@201-0-85-111.dsl.telesp.net.br) |
| 01:48.04 | _Sam-- | tftp has a root |
| 01:48.07 | _Sam-- | like /tftproot |
| 01:48.13 | tronix | slan: tftp server has a default location defined to it |
| 01:48.13 | flujan | hi all, I`m new to asterisk and I`m trying to configure a soft phone |
| 01:48.23 | tronix | slan: it's a tftp server configuration issue |
| 01:48.25 | Jabroni | ive just sent the new firmware to a gxp2000 |
| 01:48.26 | justinu | g729 shouldn't sound better than g711 |
| 01:48.29 | JamesDotCom | anyone here worked with audiocodes mediant series before? |
| 01:48.33 | flujan | when i attemp a call i receive the message: 404 not found |
| 01:48.38 | Jabroni | its been around 20 mins, and the phone is still on the logo screan :S |
| 01:48.43 | Ariel_ | justinu, correct but it does on the small phone. |
| 01:48.47 | _Sam-- | justinu: on this particular wifi phone ariel had suggested that it did |
| 01:48.51 | _Sam-- | thats why i was suprised |
| 01:48.53 | Ariel_ | flujan, ok ask away |
| 01:48.54 | _Sam-- | i havent actually tried |
| 01:48.55 | slan | tronix: I don't have a tftp server (I think). Just another computer with the files. |
| 01:49.04 | flujan | I`m following the instructions in the O`Reilly asterisk book. |
| 01:49.05 | fifer | What do people do to back up an existing asterisk install before doing a major upgrade (other than the /etc/asterisk stuff) that allows for a quick rollback? |
| 01:49.07 | Jabroni | slan u can use http |
| 01:49.22 | tronix | slan: it's a piece of cake to set up a tftp server. just need a tiny bit of software. you can grab 'em for any OS out there |
| 01:49.30 | tronix | slan: windows, linux, solaris, macos x, whatever |
| 01:49.41 | flujan | how can I debug it, to see if asterisk is receiving the call request? |
| 01:49.50 | fifer | I changed my destination in the makefile, but that just helps if there is a make issue |
| 01:49.52 | Jabroni | _Sam-- how much time did it took ur gpx2000 to upgrade ? |
| 01:49.53 | Ariel_ | fifer, copy directorys |
| 01:49.54 | _Sam-- | slan: you will have no probelm with tftp, its 't' for trivial |
| 01:49.54 | justinu | sip debug? |
| 01:49.55 | [av]bani | slan: you might find http upgrade easier |
| 01:49.58 | tronix | slan: you only need a tftp server software + IP running on some machine, could be a laptop or desktop, doesn't matter. |
| 01:50.08 | _Sam-- | if you want to do http, i will put the files somewhere for you |
| 01:50.17 | _Sam-- | Jabroni: it should take about 5 minutes max |
| 01:50.24 | slan | Jabroni: I'll try http. |
| 01:50.30 | _Sam-- | if it is taking longer your probbably in a reboot loop, or the phone needs to be power cycled |
| 01:50.32 | [av]bani | on LAN it takes about 45 sec :) |
| 01:50.33 | fifer | as in /usr/lib/asterisk and /usr/sbin/asterisk? |
| 01:51.01 | slan | _Sam--: I have the files on this machine so I'll have a go. I'll just use my dnydns/mysubdir and see what goes. |
| 01:51.02 | Jabroni | _Sam-- doing a power cycle now |
| 01:51.25 | _Sam-- | slan: if your computer is on a 192 network, and your phone is on a 192 network...then you probabyl donyt want dyndns |
| 01:51.29 | Ariel_ | fifer, look at the /etc/asterisk/asterisk.conf it will tell you all the locations |
| 01:51.33 | _Sam-- | although it would probably work too |
| 01:51.46 | fifer | Arial_: Good point! |
| 01:51.59 | slan | _Sam--: Oh yes. I wasn't thinking. How to I mix a 192 dotted addr and subdir name? |
| 01:52.02 | [av]bani | _Sam--: pretty crazy gs added the nat router function to the latest firmware. i would much rather have had minibrowser |
| 01:52.17 | slan | _Sam--: Just 192.x.x.x/subdir? |
| 01:52.25 | _Sam-- | exactly |
| 01:52.29 | flujan | can someone help me? I`m using the x-lite program |
| 01:52.40 | fifer | Arial_: what about the sbin folder? |
| 01:52.58 | slan | _Sam--: Thanks for your help. Should take only a couple minutes which usually means an hour <g> |
| 01:53.16 | fifer | it may have only safe_asterisk, not sure |
| 01:53.17 | _Sam-- | yeah..it should take about 5 minutes |
| 01:53.21 | _Sam-- | since you have to reboot the phone twice |
| 01:53.24 | _Sam-- | or it reboots twice rather |
| 01:53.32 | _Sam-- | less if you are super-bani |
| 01:53.40 | _Sam-- | but 5 is safe |
| 01:53.54 | slan | _Sam--: We shall see. |
| 01:54.01 | _Sam-- | what tftp server? |
| 01:54.16 | Ariel_ | flujan, xlite is fairly easy to setup |
| 01:55.05 | Ariel_ | if you go to the asterisk boxes cli do: sip show peers |
| 01:55.06 | flujan | Ariel_, i setup this following the book |
| 01:55.18 | Ariel_ | fine people miss steps all the time |
| 01:55.42 | flujan | i will try again and will back in a couple of minutes... |
| 01:55.44 | flujan | :) |
| 01:55.52 | dfgas | what folders do i delete to start over |
| 01:56.00 | dfgas | for amp and asterisk |
| 01:56.11 | Ariel_ | drastic move |
| 01:56.32 | Ariel_ | dfgas, look in the /etc/asterisk/asterisk.conf |
| 01:56.35 | _Sam-- | maybe next time you wont use amp :) |
| 01:56.39 | Ariel_ | for amp then you have many other locations |
| 01:57.04 | Ariel_ | amp work just fine. you just need to learn it's ways. |
| 01:57.28 | dfgas | amp is not working with asterisk right |
| 01:57.36 | _Sam-- | im sure its fine, i just dont like it...it will sneakily overwrite your files when you're not looking when you install it |
| 01:57.45 | _Sam-- | and i dont think its full featured enough |
| 01:57.52 | _Sam-- | but i guess it beats what else is out there. |
| 01:58.13 | Ariel_ | _Sam--, hummm never stopped me using the custom.conf files don't get changed |
| 01:58.14 | *** join/#asterisk kn0x (i=atlantic@71.194.235.251) |
| 01:58.36 | *** join/#asterisk r_evolution (i=_evoluti@12.155.106.12) |
| 01:58.45 | r_evolution | ... |
| 01:59.01 | r_evolution | some days... |
| 01:59.04 | r_evolution | are better than others... |
| 01:59.10 | _Sam-- | alls i remember is early on my asterisk days, i had some stuff in /etc/asterisk...i installed amp...my stuff was gone. |
| 01:59.15 | Qwell[] | r_evolution: That's what consultants are for |
| 01:59.16 | wunderkin | id agree with you there if any of them were good |
| 01:59.21 | r_evolution | haha @ qwell |
| 01:59.28 | r_evolution | well here's one for you qwell... |
| 01:59.29 | _Sam-- | at least, all the configuration i had done was gone |
| 01:59.44 | r_evolution | how do you react when a tech comes up to you and tells you they've got a linux user on the phone |
| 01:59.48 | r_evolution | who cant setup their own dial-up |
| 01:59.49 | r_evolution | :-\ |
| 01:59.54 | Ariel_ | _Sam--, yes that it will change your orginal ones |
| 01:59.57 | r_evolution | linspire... etc. |
| 02:00.00 | Qwell[] | Are you getting paid to support them? |
| 02:00.09 | r_evolution | am i? |
| 02:00.10 | r_evolution | no |
| 02:00.11 | r_evolution | is the tech |
| 02:00.12 | r_evolution | yes |
| 02:00.13 | Qwell[] | was he? |
| 02:00.16 | slan | _Sam--: Did reboot but firmware did not update. Checked automatic updates and the url/subdir is correct. Guess I'll look for a Linux tftp server. |
| 02:00.16 | Qwell[] | Then you fire his ass |
| 02:00.17 | libila | asterisk can't start up properly because it keeps saying /dev/zap/channel permission denied. I think it might be that asterisk crashed or something, and now it's locked up but I tried restarting and the channels are still locked. |
| 02:00.21 | r_evolution | haha |
| 02:00.28 | r_evolution | nah it's not their fault... they dont use linux |
| 02:00.30 | r_evolution | win32 |
| 02:00.31 | _Sam-- | slan: there are plenty fine tftp windows server |
| 02:00.31 | _Sam-- | s |
| 02:00.40 | Qwell[] | Then you need a tech that does Linux, if you support it |
| 02:00.43 | slan | _Sam--: Using Slackware linux here. |
| 02:00.50 | Ariel_ | libila, udev |
| 02:00.56 | *** part/#asterisk QbY (n=Kelvin@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 02:01.01 | libila | no, thats working correctly... |
| 02:01.03 | r_evolution | well it is "supported" in that ppp connections are accepted |
| 02:01.04 | libila | or at least it was |
| 02:01.10 | Qwell[] | otherwise, you give the standard "sorry, that isn't a supported operating system", and refund his money |
| 02:01.13 | _Sam-- | if you are running slackware, do you already have a webserver running? |
| 02:01.14 | r_evolution | but the humor of a *nix user being incapable of setting up the dial-up |
| 02:01.23 | _Sam-- | and did you set up /etc/inetd.conf for tftp? |
| 02:01.34 | Qwell[] | not funny at all, imo |
| 02:01.41 | r_evolution | why? |
| 02:01.45 | r_evolution | it's funny in a sad way |
| 02:01.50 | Qwell[] | how so? |
| 02:02.07 | Qwell[] | just because he uses Linux, means he knows everything about it? |
| 02:02.15 | r_evolution | I expect people who use an operating system that requires a bit more configuration to be a little more capable |
| 02:02.15 | Qwell[] | So, why don't you have a Windows user install Exchange |
| 02:02.21 | Qwell[] | "Oh, you don't know how? HAHA" |
| 02:02.21 | r_evolution | ^ |
| 02:02.27 | _Sam-- | you dont need any configuration to run linux anymore |
| 02:02.31 | _Sam-- | look at knoppix/morphix |
| 02:02.44 | *** join/#asterisk MatsK (n=mk@6.80-203-84.nextgentel.com) |
| 02:02.47 | Qwell[] | Did you end up helping him? |
| 02:02.51 | r_evolution | yes |
| 02:02.57 | r_evolution | Would you expect me NOT to? |
| 02:03.12 | _Sam-- | you still remember how to setup pppd? |
| 02:03.14 | Qwell[] | No, I would expect you to, if you claim you support Linux |
| 02:03.17 | _Sam-- | damn that has been a long ass time |
| 02:03.21 | r_evolution | just because it irritates me that he doesnt want to spend the time to learn an OS which deviates from the norm |
| 02:03.42 | Qwell[] | r_evolution: So, like when Windows came out...everybody knew it? |
| 02:03.55 | r_evolution | no but most people were willing to spend the time to get to know it |
| 02:03.56 | _Sam-- | god i remember the trumpet winsock support days |
| 02:03.58 | Qwell[] | sounds like you're just trying to be an elitist |
| 02:04.10 | r_evolution | i cant be an elitist until i know everything :) |
| 02:04.24 | rob0 | pppd was a horror! |
| 02:04.31 | _Sam-- | trying to tell clueless windows 3.1 users how to setup trumpet winsock |
| 02:04.31 | r_evolution | i'm just an easily-irritated-after-11-hours-of-work-ist |
| 02:04.32 | Qwell[] | Then you can't bash a <insert software> user for not knowing how to use it 100% |
| 02:04.36 | _Sam-- | i used to just do housecalls instead |
| 02:04.51 | r_evolution | you take away all my fun qwell |
| 02:04.55 | Qwell[] | r_evolution: Do you know how to setup ldap? I sure as hell don't |
| 02:05.00 | Qwell[] | nor do I know how to setup ppp |
| 02:05.02 | r_evolution | not yet :-D |
| 02:05.12 | Qwell[] | It's not something I care to know, and it's something I'm paying somebody to do for me |
| 02:05.13 | slan | _Sam--: I don't have initd.conf on this machine. It must be a SysV init thing and Slack uses BSD. |
| 02:05.23 | _Sam-- | inEtd.conf |
| 02:05.33 | _Sam-- | most linux should have it, but maybe you dont need it for tftp anymore |
| 02:05.37 | r_evolution | you really just kill all my mean humor qwell |
| 02:05.40 | _Sam-- | used to have to tell inetd how to handle tftpd |
| 02:05.40 | r_evolution | you really do |
| 02:05.44 | Qwell[] | r_evolution: Because it's unfounded |
| 02:05.48 | Qwell[] | and you know I'm right |
| 02:05.56 | r_evolution | haha... i admit nothing |
| 02:06.03 | Qwell[] | You don't need to. ;] |
| 02:06.08 | *** join/#asterisk rene- (i=rene@201.144.60.114) |
| 02:06.11 | Qwell[] | bbl, time to lose the [] |
| 02:06.19 | *** part/#asterisk rene- (i=rene@201.144.60.114) |
| 02:06.27 | r_evolution | bb... tomorrow... time to lose the work clothes |
| 02:06.39 | r_evolution | erm |
| 02:06.40 | r_evolution | fuck |
| 02:06.42 | r_evolution | long day |
| 02:06.43 | r_evolution | *bang* |
| 02:06.57 | *** join/#asterisk MoR4euZ (i=kvirc@port-83-236-3-151.dynamic.qsc.de) |
| 02:07.07 | slan | _Sam--: Sorry I just found it. Luckily the Slack init files are very well commented so I should be able change them properly. It will take a little more than an hour though <g> |
| 02:07.16 | *** join/#asterisk bert1 (n=admin@adsl-220-179-181.mob.bellsouth.net) |
| 02:07.19 | rob0 | slan: $ grep tftp /etc/inetd.conf gives me "# tftp dgram udp wait root /usr/sbin/in.tftpd in.tftpd -s /tftpboot -r blksize" |
| 02:07.20 | _Sam-- | it should take about 5 minutes |
| 02:07.24 | rob0 | ah you found it |
| 02:07.37 | _Sam-- | if it is going to take an hour i will put the files on a web server for you |
| 02:08.16 | slan | _Sam--: Thanks but I need to learn this. I've learned a _lot_ of Linux from using Slack for a few years but there are still holes to plug up. |
| 02:08.29 | _Sam-- | if you insist..i have the files ready for ya. |
| 02:08.47 | _Sam-- | there isnt a whole to learn from getting tftp running on linux, but its admirable. |
| 02:09.09 | libila | Ariel_: http://rafb.net/paste/results/bYCYQm64.html Those kind of errors don't come from not having udev setup? Plus I have it setup, and it was working. |
| 02:09.23 | slan | _Sam--: You are very gracious. First I have to choose the flavor of tftp server - I got a lotta google hits on it. |
| 02:09.30 | _Sam-- | you could save some headeach but just installing the solarwinds tftp server on a windowx machine |
| 02:09.37 | tronix | slan: try tftp-hpa if you have a choice. :) |
| 02:09.40 | _Sam-- | sorry i cant type |
| 02:09.47 | tronix | (if linux) |
| 02:09.52 | *** join/#asterisk MoR4euZ (i=kvirc@port-83-236-3-151.dynamic.qsc.de) |
| 02:09.54 | tronix | otherwise, if windows, solarwinds is decent |
| 02:09.56 | masked | X-Rob do i have to enable this card with zttool before it'll work with asterisk? |
| 02:10.06 | slan | tronix: I'll try tftp-hpa thanks |
| 02:10.19 | masked | cos asterisk/chan_zap can't find the channel |
| 02:10.28 | *** join/#asterisk sack (n=sack@244.Red-81-38-35.dynamicIP.rima-tde.net) |
| 02:10.50 | _Sam-- | slan: if you want a project, setup your http server and upgrade via http |
| 02:10.56 | _Sam-- | that learnnig is more valuable than learning to setup tftp |
| 02:11.17 | slan | tronix: I see tftp-hpa is for bsd inits. It should to the trick. |
| 02:11.20 | tronix | masked: maybe put up on pb (pastebin) the output of 'grep -v ^\; /etc/zaptel.conf' and 'grep -v ^\; /etc/asterisk/zapata.conf' |
| 02:11.26 | _Sam-- | but i guess you gotta start someplace, sorry to push ya too hard. |
| 02:11.42 | tronix | tronix: i'm running it fine on my Gentoo Linux box. hpa = H. Peter Anvin, one of the early Linux people (Slackware guy?) |
| 02:11.43 | Jabroni | _Sam-- have u figured out a way to get the transfer button working on the GPX2000??? using the asterisk transfer option rather than the phone transfer |
| 02:11.56 | Jabroni | since using phone transfer = no MoH :( |
| 02:11.57 | _Sam-- | i use both, asterisk transfer and phone button |
| 02:11.59 | _Sam-- | they both work |
| 02:12.00 | tronix | err slan |
| 02:12.13 | tronix | darn I'd be bad in bed with a woman, screaming out my own name. :-) |
| 02:12.29 | _Sam-- | phone transfer = person on line 1, put on hold from hold...call person on lin2 "hey joe is on line 1 want to talk?" press transfer |
| 02:12.42 | _Sam-- | put on hold from hold button |
| 02:12.56 | Jabroni | yeah but for the person calling there wont be music on hold |
| 02:13.01 | _Sam-- | bs |
| 02:13.04 | _Sam-- | they are on hold |
| 02:13.06 | _Sam-- | they are hearing moh |
| 02:13.30 | _Sam-- | and if you use asterisk transfer, you can call park them, where they hear moh |
| 02:13.38 | _Sam-- | or you can blind transfer #extension |
| 02:13.43 | Jabroni | let me check again with this new firmware.. the old firmware did that do me.. the phone parked the call, rather asterisk |
| 02:14.02 | _Sam-- | call your phone, put yourself on hold on line 1 |
| 02:14.06 | _Sam-- | that is what your caller(s) will hear |
| 02:14.17 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 02:14.20 | _Sam-- | then you pick up the phone on line 2 and call whoever you want to transfer to |
| 02:14.36 | _Sam-- | that is an attended transfer from the phone |
| 02:15.07 | masked | tronix: zaptel.conf http://pastebin.com/533176 |
| 02:15.21 | _Sam-- | there is also one other way to do it from the phone, but that is the method i use (i never even do it that) |
| 02:15.35 | _Sam-- | i do: call comes in on line 1...put them on hold on the phone..(they hear moh)... |
| 02:16.05 | _Sam-- | on line 2 i call whoever i want to transfer to ..."hey do you want to talk to joe from so and so"..."yes".."ok"...then back on line 1 hit #extension-to-transfer-to and blind transfer them |
| 02:16.28 | masked | tronix: zapata http://pastebin.com/533177 |
| 02:16.40 | tronix | masked: okay. comment out the line that says loadzone=us and i'm guessing you also want channels=1 |
| 02:17.34 | masked | tronix with the s? |
| 02:18.03 | tronix | masked: err let's see |
| 02:18.18 | tronix | masked: yes |
| 02:18.27 | tronix | masked: you'll see the 'channels' line commented out |
| 02:18.57 | tronix | it can be for a single channel. the keyword is just 'channels' cos that's the way code is written |
| 02:18.57 | masked | oh i thought i'd done that... odd. |
| 02:19.00 | tronix | no biggie. |
| 02:19.03 | [av]bani | http://69box.atlantic.net/daily/show.wmv <- lollerskates |
| 02:19.39 | masked | tronix: asterisk still doesn't like me |
| 02:20.25 | tronix | masked: you've rmmod wxfxo and then zaptel, then modprobe'd zaptel then wxfxo? |
| 02:20.42 | masked | nope |
| 02:20.52 | masked | kernel doesn't have module unloading :P |
| 02:20.55 | tronix | ahh. :) |
| 02:21.03 | tronix | ouch. |
| 02:21.17 | tronix | makes life much easier for initial debugging. :) |
| 02:21.25 | masked | yeh i usually have it on |
| 02:21.32 | masked | just this box i dont, for some reason.. |
| 02:21.35 | tronix | heh |
| 02:21.50 | masked | ahh well, reboot will do. |
| 02:21.57 | tronix | heh. cool. sorry :( |
| 02:22.18 | masked | lol |
| 02:22.23 | masked | not ur problem |
| 02:22.39 | Err | I didn't know you could build a system without the ability to unload modules |
| 02:22.48 | Err | I can't offhand think of any reason why this would ever be wanted |
| 02:23.11 | masked | that name plays tricks on my head |
| 02:24.22 | masked | excellent |
| 02:24.28 | masked | it works now, thanks again tronix |
| 02:24.54 | [av]bani | Err: security |
| 02:25.08 | masked | oh wait |
| 02:25.12 | masked | no it doesn't |
| 02:25.27 | tronix | seeing particular error messages? how are you testing? |
| 02:25.29 | Err | [av]bani: heh, by the time somebody has the privileges to unload modules, does it really matter? :-) |
| 02:25.39 | masked | chan_zap.c:923 zt_open: Unable to specify channel 1: No such device or address |
| 02:26.22 | [av]bani | Err: yes, just because they can unload modules doesnt mean they can do anything they want |
| 02:26.50 | tronix | masked: hmm... maybe pb output of ztconfig -vvv ? |
| 02:27.18 | masked | lol unknown keyword ' channels' |
| 02:27.27 | masked | lol mustn't have liked the space |
| 02:28.07 | tronix | :) |
| 02:28.14 | masked | reboot! |
| 02:28.18 | tronix | hahaha |
| 02:29.02 | tronix | masked, save this url: http://www.voip-info.org/tiki-index.php?page=Asterisk+X100P+Echotraining |
| 02:29.02 | masked | boots up quick tho, ill give i tthat |
| 02:29.11 | tronix | you might find it useful. |
| 02:29.36 | tronix | i've got a tdm400p w/1 fxo and 1 fxs module so i'm not much good with X100P tips but that's what I found re: echo |
| 02:29.52 | masked | tronix i have it bookmarked ;) |
| 02:29.56 | tronix | sweet |
| 02:30.22 | masked | wttttttt |
| 02:30.33 | masked | line 223: Cannot get number of tones chanel 1 |
| 02:30.33 | masked | line 223: Cannot init tones chanel 1 |
| 02:30.39 | tronix | typo there |
| 02:31.04 | masked | yeah but not on my behalg |
| 02:31.05 | masked | f |
| 02:31.09 | tronix | hmm |
| 02:31.21 | tronix | grep chanel /etc/asterisk/* |
| 02:31.23 | masked | uhm, odd thing is it was working okay before |
| 02:31.40 | masked | nada |
| 02:31.41 | masked | nothing |
| 02:31.45 | tronix | hmm. |
| 02:31.51 | slan | _Sam--: I'm back from trying to download tfpt-hpa from some site called Softpedia. It's designed to drive me crazy. |
| 02:31.52 | masked | it just be in the code |
| 02:32.04 | masked | must* |
| 02:32.09 | tronix | lsmod|egrep "wcfxo|zaptel" |
| 02:32.14 | tronix | both loaded? |
| 02:32.24 | masked | yeh |
| 02:32.34 | tronix | where you seeing that error message? |
| 02:32.42 | masked | ztcfg |
| 02:32.47 | tronix | hmm. |
| 02:32.57 | masked | could it be cos there is no line plugged in? |
| 02:33.09 | tronix | maybe... or grep chanel /etc/zaptel.conf |
| 02:33.21 | masked | nothing |
| 02:33.26 | tronix | I haven't been using asterisk for very long, but do recall |
| 02:33.35 | tronix | it would be unhappy about starting up if nothing was plugged in |
| 02:33.47 | tronix | though I'm not sure how you're getting that 'chanel' error message. |
| 02:34.05 | masked | ztcfg.c |
| 02:34.10 | tronix | heh. okay. |
| 02:34.22 | drumkilla | masked: i committed a fix for those typos, hehe |
| 02:34.25 | tronix | :-) |
| 02:34.27 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 02:34.42 | masked | drumkilla care to tell me what causes them? |
| 02:35.25 | drumkilla | masked: what hardware is this? |
| 02:35.30 | masked | x100p |
| 02:35.30 | tronix | x100p |
| 02:35.35 | tronix | what he said. ;) |
| 02:35.41 | masked | i have a stunt double |
| 02:35.45 | tronix | hahaha |
| 02:35.58 | _Sam-- | slan: when you're ready to give up, i will give you a url for http upgrade :) |
| 02:36.00 | drumkilla | probably not the right card ... |
| 02:36.14 | masked | but? |
| 02:36.21 | drumkilla | but that's it |
| 02:36.24 | tronix | drumkilla: yeah. it's what he has right now... might as well as make it a learning experience. |
| 02:36.26 | rajiv | how many RENs does the digium tdm400p support ? |
| 02:36.29 | drumkilla | if it was a digium card, i'd tell you to contact tech support |
| 02:36.47 | masked | it was working earlier |
| 02:36.51 | masked | well |
| 02:36.55 | masked | it wans't giving this error |
| 02:37.01 | masked | ztcfg -v was healthy |
| 02:37.05 | tronix | masked: what changed in between? |
| 02:37.17 | masked | uhm |
| 02:37.56 | masked | i made linux26 and reinstalled thinking it might post a 2.6 related module entry to /etc/sysconfig/modules, but it did the alias and install stuff again |
| 02:38.01 | *** join/#asterisk flujan (i=flujan@201-0-85-157.dsl.telesp.net.br) |
| 02:38.04 | masked | thinking that is more 2.4 related |
| 02:38.14 | masked | so i reformatted the modules file completely after that |
| 02:38.23 | *** join/#asterisk flujan (i=flujan@201-0-85-157.dsl.telesp.net.br) |
| 02:38.35 | masked | just leaving module entries |
| 02:38.42 | masked | well, names. |
| 02:38.56 | *** join/#asterisk flujan (i=flujan@201-0-85-157.dsl.telesp.net.br) |
| 02:38.57 | masked | it may have borked zaptel.conf, but it still seems healthy |
| 02:39.30 | flujan | Ariel_, thanks for your previous help... now Asterisk is working marvelous... :) |
| 02:39.45 | flujan | so, yeat another doubt about the dial command. |
| 02:39.53 | flujan | The 4 parameter is a URL |
| 02:40.49 | flujan | does X-lite support this? I do a call and pop-up a url. how can i achieve this behavior? I tried SendURL also |
| 02:40.53 | masked | oh i loaded ztdummy for some reason |
| 02:40.56 | masked | that might be why |
| 02:41.49 | masked | ok building kernel with module unloading support :P |
| 02:41.50 | *** join/#asterisk klictel (n=klictel@modemcable119.206-200-24.mc.videotron.ca) |
| 02:42.00 | tronix | :) |
| 02:42.24 | _Sam-- | i never heard of a kernel that didnt support rmmod? |
| 02:42.33 | flujan | is there other client that support this behavior I`m seeking? |
| 02:42.46 | _Sam-- | what kernel specific option would make it so you cant remove a module? |
| 02:43.06 | masked | _Sam-- it actually by default doesn't support unloading |
| 02:43.14 | masked | you have to specifically enable module unloading |
| 02:43.21 | rob0 | CONFIG_MODULE_UNLOAD=n |
| 02:43.26 | tronix | _Sam--: could be a distro-related kernel config option |
| 02:43.28 | masked | its been this way through most of 2.6 |
| 02:43.53 | rob0 | I would think distro kernels would be CONFIG_MODULE_UNLOAD=y |
| 02:43.54 | _Sam-- | my 2.6 kernels are all stock debians that is probably why |
| 02:44.16 | _Sam-- | i have a bunch of 2.4s that i compiled, i have to check to see |
| 02:44.55 | _Sam-- | not in 2.4 -- config_module_unload |
| 02:45.38 | _Sam-- | but my 2.4 kernels definite have rmmod |
| 02:46.11 | _Sam-- | guess that is why i never heard of it |
| 02:46.21 | rob0 | (I am looking in a 2.6.x config) |
| 02:46.49 | _Sam-- | debian stock: CONFIG_MODULE_UNLOAD=y |
| 02:48.35 | *** join/#asterisk kc5cqm (n=kc5cqm@cpe-68-206-116-214.stx.res.rr.com) |
| 02:49.05 | kc5cqm | help: my /dev/zap stuff is owned by root... fc3/udev. How do I specify group/mode for stuff in udev? |
| 02:50.04 | flujan | how can I open popup windows with asterisk and x-lite? |
| 02:50.50 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-222-26-17.msy.bellsouth.net) |
| 02:50.54 | _Sam-- | flujan: who said you can? |
| 02:51.03 | _Sam-- | (you may be able to, im not sure really) |
| 02:51.07 | slan | _Sam--: Really waste of time. Finally d/l tftp-hpa, compiled, compile didn't go well, then I found that Slack has tftp v.41 already! |
| 02:51.18 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 02:51.36 | kc5cqm | nevermind... /etc/udev/rules.d and permissions.d ...pretty self explainatory |
| 02:51.42 | masked | bah ok so i modifed my kernel source now it wont compile |
| 02:51.45 | masked | lol reboot! |
| 02:51.49 | masked | im gunna go get some lunch |
| 02:51.55 | tronix | :) |
| 02:52.08 | _Sam-- | slan: sounds like you're really motoring ahead! |
| 02:52.09 | masked | well once i've tried this.. |
| 02:52.13 | kc5cqm | is there some way to refresh udev? |
| 02:52.23 | flujan | _Sam--, this is my question... Can I open popup windows with asterisk and x-lite? If not, which client support this feature? |
| 02:52.47 | slan | _Sam--: So how do I start the tftp server and specify the directory? |
| 02:53.19 | masked | nope same error |
| 02:53.19 | masked | bbs |
| 02:53.25 | slan | _Sam--: Man page unclear as usual for those that don't already know the program. |
| 02:53.40 | heath__ | flujan: ajax->php->manager |
| 02:53.57 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 02:54.26 | flujan | heath__, just with this features i can accomplish this? I will try it with ragi and rails. |
| 02:54.32 | _Sam-- | slan if you uncomment the inetd.conf lines for tftp... |
| 02:54.40 | *** join/#asterisk da_monumental_1 (n=da_monum@cpe-065-191-084-026.nc.res.rr.com) |
| 02:54.43 | _Sam-- | then it should probably work (depending on your tftp lines in your inetd.conf> |
| 02:54.50 | slan | _Sam--: already uncommented it. |
| 02:54.53 | flujan | heath__, but I should present something like this to my boss. :( |
| 02:55.04 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 02:55.24 | *** part/#asterisk da_monumental_1 (n=da_monum@cpe-065-191-084-026.nc.res.rr.com) |
| 02:55.25 | _Sam-- | you need to restart inetd |
| 02:55.45 | flujan | i must receive a call and Dial to one operator. When the operator receive the call, the browser must open with a specific URL. |
| 02:56.16 | slan | _Sam--: inetd.conf has this (partial) /usr/sbin/in.tftpd in.tftpd -s /tftpboot -r blksize |
| 02:56.24 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net) |
| 02:56.36 | slan | _Sam--: Is /tftpboot the dir I want? |
| 02:57.15 | Err | that's the root directory that the tftp server will serve from |
| 02:58.06 | _Sam-- | but it wont work unless inetd is restarted |
| 02:58.11 | _Sam-- | if you made changes to inetd.conf |
| 02:58.24 | slan | _Sam--: Just source it? |
| 02:58.33 | _Sam-- | most people kill -HUP inetd |
| 02:58.34 | _Sam-- | i think |
| 02:59.35 | _Sam-- | you may have /etc/rc.d/init.d/inetd as well |
| 02:59.43 | _Sam-- | i dont know slackware anymore |
| 02:59.58 | _Sam-- | in which case if you have that, you can do /etc/rc.d/init.d/inetd stop |
| 03:00.01 | _Sam-- | and then start |
| 03:00.06 | _Sam-- | or restart |
| 03:00.16 | *** join/#asterisk blackremedy (n=ur3rdeye@240M06.oasis.mediatti.net) |
| 03:02.37 | *** join/#asterisk rene- (i=rene@201.144.60.114) |
| 03:02.50 | slan | _Sam--: It's in /etc/rc.d/rc.inetd and I just did ./rc.inetd restart |
| 03:03.15 | rene- | 420 people in room, i wonder what they are up to |
| 03:03.18 | _Sam-- | i just checked...most of the inetd restart scripts just do kill -HUP anyway :) |
| 03:04.03 | rob0 | killall -HUP inetd is the same thing, yes. |
| 03:04.25 | Err | that sounds more like a reload than a restart target; I would expect restart to actually kill it, and then re-run it |
| 03:04.45 | rene- | anyway is it possible for an agent in a call to put his call into park and be available to get a new call from the acd and be able to switch back and forth between calls |
| 03:05.15 | slan | _Sam--: I think it needed to start tfpt as a server/daemon. Going to man. |
| 03:05.37 | rob0 | oh you're right Err, it kills and starts it again. I just do "killall -HUP inetd" though. :) |
| 03:06.07 | slan | _Sam--: There is no sftp in ps so maybe I need to start it. |
| 03:06.17 | _Sam-- | slan: based on the line in your inetd.conf i dont think you need to run it as a daemon |
| 03:06.28 | brockj49464 | Any way to track hook switch changes from a SPA2100? |
| 03:06.30 | Err | it won't run, unless there's a client connected |
| 03:06.45 | slan | _Sam--: OK but I booted the phone and it didn't boot again twice. |
| 03:07.40 | rob0 | slan: did it DHCP an address? |
| 03:07.45 | Err | slan: you might run tcpdump on the server and see if you can see its tftp requests |
| 03:07.58 | rob0 | check logs first |
| 03:08.05 | _Sam-- | where does tftp log? |
| 03:08.11 | slan | Err: Yes. I'll tcpdump and try again. |
| 03:08.17 | Err | I wouldn't bother with logs, but sure - you could do that :-) |
| 03:08.48 | Err | tftp almost certainly logs to syslog, so it'll be dumped wherever syslog is told to put it (who knows - depends on the system) |
| 03:09.14 | slan | _Sam--: Nothing fresh in /var/log |
| 03:09.27 | rob0 | It *would* be interesting to know if it tried to dhcp |
| 03:09.39 | _Sam-- | im sure it did |
| 03:09.40 | rob0 | nothing in logs suggests that it didn't |
| 03:09.56 | _Sam-- | slan: does your phone have an IP on the display? |
| 03:10.31 | Err | note that if you're using tcpdump (as opposed to ethereal, which would really be a more user-friendly approach), you'll need to use -s1500 -vvv as options to see exactly what's going on - and probably -e, to see MAC addresses so you can determine what packets are coming from the phone during broadcasts |
| 03:11.02 | *** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 03:11.16 | _Sam-- | there is no need for ethereal or tcpdump really...he's upgrading firmware on a phone that dozens of people here have upgraded fine |
| 03:11.20 | *** join/#asterisk axscode (n=axscode@210.213.106.188) |
| 03:11.26 | slan | _Sam--: 192.168.0.120 |
| 03:11.29 | _Sam-- | so in my opinion its pure config problems...on the phone, and on the tftp server |
| 03:11.45 | _Sam-- | right if your phone has an IP, then it got DHCP info |
| 03:11.52 | _Sam-- | which means it can do tftp if you set it up right |
| 03:11.56 | Err | tcpdump will show what's failing in the TFTP, if it's really what's dying |
| 03:12.01 | slan | _Sam--: The tftp server due to my inexperience with it. |
| 03:12.01 | _Sam-- | and if your tftp server is on the right net |
| 03:12.11 | rob0 | can you ping 192.168.0.120 ? |
| 03:13.35 | slan | Ping ok |
| 03:13.52 | *** join/#asterisk kc5cqm (n=kc5cqm_@cpe-68-206-116-214.stx.res.rr.com) |
| 03:13.58 | kc5cqm | hello |
| 03:14.22 | slan | Err: I tried tcpdump with your parameters but there's an awful lot going on the screen |
| 03:14.37 | kc5cqm | anyone here set up a digium wildcard under fc3? |
| 03:14.54 | _Sam-- | slan: if you cant setup tftp, nothing from tcpdump will be of much value. |
| 03:14.58 | Qwell | kc5cqm: no, but it's the same for any distro |
| 03:15.03 | Err | slan: well, if the box already has an IP address, you can filter on that - add "host 192.168.0.120" to the end |
| 03:15.33 | Err | _Sam--: that's not true - because he'll see the tftp requests, and no responses, so he'll know that it's requesting to the right machine and that the server isn't set up right |
| 03:15.51 | Err | if he doesn't see requests, or they're going to the wrong server, then he knows that the DHCP server isn't set properly |
| 03:16.10 | rob0 | check netstat and make sure that inetd is listening on 69 |
| 03:16.18 | _Sam-- | if you are going to hand hold through all of that, you are a better man than myself. |
| 03:16.34 | slan | _Sam--: Looks like the phone booted twice but no change in firmware date. |
| 03:16.55 | rob0 | I'm interested because I will be setting up tftpd on Slackware eventually. :) |
| 03:17.31 | _Sam-- | there really isnt much to it. |
| 03:17.33 | Err | tftpd is actually one of the easiest servers to set up ever - the man page tells you everything you need to know |
| 03:17.48 | _Sam-- | the t in tftp the first t...is there for a reason. |
| 03:18.10 | Err | heh, that actually refers to the on-wire protocol, but it stands for the server as well :-) |
| 03:18.19 | rob0 | You *do* have to have a /tftpboot or whatever directory, I guess. |
| 03:18.28 | Err | no - you can set that directory to whatever you want |
| 03:18.36 | slan | rob0: I have /tftpboot with the files in it. |
| 03:19.23 | rob0 | ok |
| 03:19.41 | rob0 | yes that's why I said "or whatever directory". |
| 03:20.33 | slan | _Sam--: It doesn't seem that tftp is not talking to the phone so I suspect misconfiguration of tftp. Now into the second hour but that's par for the course. |
| 03:20.59 | rob0 | test with a manual tftp client |
| 03:21.19 | _Sam-- | and double check the phone |
| 03:21.24 | Err | is there a firewall involved? is there anything in hosts.allow or hosts.deny? |
| 03:21.28 | _Sam-- | to make sure you are pointing to the right adress for the tftp server |
| 03:21.37 | _Sam-- | and that you tell the phone to check for updates |
| 03:21.53 | slan | Err: The firewall basically allows everything on this subnet. |
| 03:21.58 | _Sam-- | and make sure the path to the files in the phone is right |
| 03:22.13 | Err | basically, or it does? :-) |
| 03:22.15 | _Sam-- | if you are using /tftpboot then maybe dont use a subdir |
| 03:22.56 | kc5cqm | Qwell, can't seem to make it work although ztool and ztcfg show fine |
| 03:23.03 | kc5cqm | I keep getting : Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 03:23.04 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 03:23.09 | slan | _Sam--: The GV web config doesn't have any way to specify the subdir. But I'll try it again with the files in / |
| 03:23.13 | Qwell | kc5cqm: is the module loaded? Is it configured properly? |
| 03:23.34 | kc5cqm | its loaded |
| 03:23.37 | kc5cqm | brb...phone |
| 03:24.05 | Err | slan: whatever directory is specified in the inetd.conf file will be the *root* directory that tftpd serves - so the file /foo.txt would reside in /tftpboot/foo.txt (just like apache's http root, or any other file-sharing service) |
| 03:24.15 | _Sam-- | i know for http you can (obviously) specify subdir no problem |
| 03:24.16 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 03:25.06 | _Sam-- | slan: so what Err is saying...is put boot55.bin, boot55a.bin, etc....in /tftpboot |
| 03:25.11 | _Sam-- | not in / |
| 03:25.28 | Err | yes |
| 03:26.10 | slan | _Sam--: That is where they were. |
| 03:26.18 | slan | _Sam--: And are now. |
| 03:26.50 | _Sam-- | then they are good to go...as long as your tftp server uses /tftpboot as the root |
| 03:27.07 | _Sam-- | which based on the line from your inetd.conf which i dont remember...may or may not be the case |
| 03:27.38 | rob0 | tftpd/tftp work for me, and nothing was logged. |
| 03:28.30 | _Sam-- | mine doesnt log either (just checked) |
| 03:28.41 | _Sam-- | i log *.* to a custom file |
| 03:28.46 | _Sam-- | and it didnt show at all |
| 03:29.10 | rob0 | I log *.* to tty12 :) |
| 03:29.14 | slan | _Sam--: inetd.conf specifies /tftpboot |
| 03:30.32 | _Sam-- | slan: good luck on your journey. my journey leads me now to my bed. |
| 03:30.45 | kc5cqm | Qwell, back |
| 03:30.46 | _Sam-- | you are in capable hands around here, im positive. |
| 03:30.47 | rob0 | I used /var/lib/tftp and put my resolv.conf file in there, retrieved it from another machine. |
| 03:30.50 | slan | _Sam--: Sam thanks vy much for trying to help. I'll play with it some more. |
| 03:31.02 | kc5cqm | how can I tell if I have the module configured correctly? Loading zaptel and wcfxo |
| 03:31.10 | Qwell | kc5cqm: Do you get errors loading them? |
| 03:31.14 | kc5cqm | although it's using fxs_ks signaling |
| 03:31.30 | _Sam-- | slan: no shame either in using the http ugprade URL i gave ya earlier :) |
| 03:31.31 | kc5cqm | no errors on load, and zttool detects if the card is plugged in or not to pots |
| 03:31.35 | _Sam-- | that is my site, those are my files |
| 03:33.15 | masked | mmm subway, eat fresh. |
| 03:33.18 | kc5cqm | Qwell, var/log/messages shows no errors...just "found a wildcard fxo: generic clone" etc... |
| 03:33.46 | elg | anyone know where I can get ahold of sipura's spc binary (linux preferably, but windows ok too) |
| 03:33.59 | slan | _Sam--: I didn't get the url. Again pls? |
| 03:34.00 | kc5cqm | maybe the card is configed/loaded ok and its just my asterisk config |
| 03:34.22 | slan | _Sam--: Been working much too hurridly - make mistakes. |
| 03:34.40 | _Sam-- | no worries...take your time, take a deep breath...its just a phone. |
| 03:34.48 | _Sam-- | it will work fine still even if you cant update it. |
| 03:35.02 | slan | _Sam--: Unless I brick it <g> |
| 03:35.06 | kc5cqm | Qwell, zttool shows "OK" not "RED" (unless I unplug the line..which it goes "RED" |
| 03:35.07 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 03:35.35 | _Sam-- | best of luck...look out for the reboot loop! |
| 03:35.40 | _Sam-- | <remember this is beta firmware> |
| 03:35.53 | slan | _Sam--: I've heard about the reboot loop here. |
| 03:36.10 | _Sam-- | bani is the resident loop expert as evidenced by the tiki page |
| 03:37.02 | slan | _Sam--: Could you give me the upgrade url again? |
| 03:37.21 | _Sam-- | check your messages if they are in a different window |
| 03:37.24 | _Sam-- | ive sent it three times now |
| 03:38.17 | slan | _Sam--: I guess that's what the beep was all about. I'll find it somehow. |
| 03:38.25 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 03:38.35 | _Sam-- | what irc client do you use |
| 03:39.09 | _Sam-- | slan: http://voipserv.com/firmware |
| 03:39.16 | axscode | I just wonderin if someone use astbill in here? |
| 03:40.07 | ManxPower | ~docs |
| 03:40.09 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:40.09 | ManxPower | ~mailinglist |
| 03:40.11 | jbot | hmm... mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 03:40.11 | slan | _Sam--: Sam thanks for the url. Also I just found the new window that Xchat creates with the 3 url sends. Thanks much. |
| 03:40.29 | _Sam-- | sure thing...you will get it upgraded im sue. |
| 03:40.31 | _Sam-- | sure too |
| 03:40.39 | _Sam-- | im definitely not sue. |
| 03:40.41 | _Sam-- | gnight! |
| 03:40.42 | iCEBrkr | Hi Sue, nice to meet ya |
| 03:40.46 | rob0 | :) |
| 03:40.53 | rob0 | 'Night Sue! |
| 03:40.58 | _Sam-- | bastids! |
| 03:41.04 | rene- | is it possible for an agent to park a call and make himself available in the call queue |
| 03:41.04 | slan | _Sam--: Gnight Sam. Youv'e been great. |
| 03:41.22 | _Sam-- | if the call is parked using call parking, the agent should be available for more calls |
| 03:41.32 | iCEBrkr | rene-: Won't work well, cuz if the call is parked and unanswered it's gonna bounce back to him |
| 03:41.40 | axscode | hi guyz... do you happen to know what would be in the extensions.conf if im going to use another sip proxy? myphone -> myasterisk -> sip-proxy-> sip-proxy-member |
| 03:42.12 | _Sam-- | iCEBrkr: somehow i know thats what the docs say, that the parked call will be sent back to the person who parked the call after the timeout... |
| 03:42.17 | _Sam-- | but ive seen something different |
| 03:42.23 | ManxPower | axscode, don't think of it as a proxy, think of it as just another sip device. |
| 03:42.24 | masked | omg dont tell me newt has deps? |
| 03:42.31 | _Sam-- | mine timeout back to the 1st priority of the extension they called in on |
| 03:42.41 | _Sam-- | not to the person who parked the call |
| 03:42.48 | iCEBrkr | _Sam--: Yea, that's the default behavior of "parking" I'm not sure where the call would go in that scenerio |
| 03:42.57 | iCEBrkr | Ahhh |
| 03:43.10 | iCEBrkr | _Sam--: That's a nice option |
| 03:43.17 | axscode | ManxPower: so how would I put that on extensions.conf ? im a bit confuse |
| 03:43.17 | _Sam-- | i dont know how it works that way really |
| 03:43.18 | ManxPower | exten => 9NXXXXXX,1,Dial(SIP/${EXTEN:1}@sipconfentry) or something like that |
| 03:43.26 | _Sam-- | but i saw it do it a few times yesteday |
| 03:43.32 | *** join/#asterisk jgomata (n=jgomata@red-corp-201.143.78.76.telnor.net) |
| 03:43.32 | iCEBrkr | haha |
| 03:43.36 | ManxPower | that would be a _ before the 9 |
| 03:43.55 | axscode | ok.. hmm I did that... but where would I put the sipconfentry? |
| 03:44.03 | ManxPower | 1.0 and 1.2 parking timeouts work different |
| 03:44.03 | *** part/#asterisk jgomata (n=jgomata@red-corp-201.143.78.76.telnor.net) |
| 03:44.09 | axscode | is that the name of the outgoing trunk? |
| 03:44.32 | ManxPower | axscode, the [whatever] section of sip.conf |
| 03:44.57 | axscode | manx.. my problem is im using astbill... |
| 03:44.59 | _Sam-- | ManxPower: i am on 1.2 and my call parking times out back to the 1st priority of the extension the caller dialed |
| 03:45.13 | _Sam-- | and i dont think i have anything special configured |
| 03:45.30 | ManxPower | _Sam--, *nod* In 1.0.x it timed out to exten => s in the context it was in |
| 03:45.37 | _Sam-- | i see! thanks! |
| 03:45.51 | ManxPower | axscode, Perhaps you should learn asterisk before you learn astbill. |
| 03:47.35 | kc5cqm | damn thing just won't create a zap channel |
| 03:47.39 | rene- | Manx: so if i was in 1.2 and my caller reaches my queue without dialing anything in my ivr, that means that he after his parked call would time out be back at s,1? |
| 03:48.03 | kc5cqm | I guess you get what you pay for with these cheapie digium ripoff cards |
| 03:48.25 | kc5cqm | suprized digium didn't sue them for using their name |
| 03:48.33 | blitzrage | anyone get a username as the callerID num in 1.2.1 when using fromuser= in sip.conf? |
| 03:48.39 | rene- | _Sam-:: if my parked call hasnt reached timeout, can i go back and forth between my two calls? |
| 03:48.52 | iCEBrkr | kc5cqm: WTF are you bitch'n about? |
| 03:49.44 | rene- | are you talking about the one port fxo clones or the chinese tdm4xx clones |
| 03:50.04 | rob0 | I think the x101p clones are made in and sold from China, so suing them would not be trivial nor likely to do any good. |
| 03:50.04 | *** join/#asterisk bmg505 (n=leon@dsl-146-24-189.telkomadsl.co.za) |
| 03:50.52 | kc5cqm | iCEBrkr, the cheapie $10 FXO cards |
| 03:50.57 | Supaplex | FUD! works for me. |
| 03:51.10 | kc5cqm | yeah, x101p |
| 03:51.16 | iCEBrkr | kc5cqm: You mean the Intel 553 Voicemodem things? |
| 03:51.29 | kc5cqm | is that all they are? |
| 03:51.35 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 03:51.46 | iCEBrkr | That was my first 'FXO' card. |
| 03:51.47 | iCEBrkr | $10 |
| 03:51.49 | iCEBrkr | ebay |
| 03:51.56 | kc5cqm | managed to get it working? |
| 03:51.58 | iCEBrkr | yup |
| 03:51.59 | rob0 | I have one ... I admit it :) |
| 03:52.02 | iCEBrkr | Worked just fine |
| 03:52.09 | rob0 | it works as well as can be expected |
| 03:52.17 | kc5cqm | interesting |
| 03:52.36 | kc5cqm | it's just frustrating there are no error messages... |
| 03:52.42 | iCEBrkr | I didn't have money to be blow'n on tinkering with this stuff. So the $10 card it was... |
| 03:53.08 | kc5cqm | what I'm really trying to do is plug packet8 into my asterisk box |
| 03:53.13 | iCEBrkr | But now, I have 3 Sipuras and started purchasing TDM cards at work |
| 03:53.16 | elvisthedj | Hey, was there ever an option on the dial command that didn't bridge the call until the callee pressed # or something? I've got an extension that forwards to the cell, but i'd like to screen and either accept or send to VM.. the only thing i can do right now is parkandannounce, then call back and pick up the parked call. |
| 03:53.23 | iCEBrkr | kc5cqm: I didn't think Packet8 worked with Asterisk |
| 03:53.36 | kc5cqm | iCEBrkr, it doesn't...hence the analog card |
| 03:53.50 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 03:54.08 | rob0 | I have a PSTN line in my x101p clone. |
| 03:54.37 | kc5cqm | thats what this is... x101p |
| 03:54.53 | masked | yer i just got a x100p in the mail today |
| 03:54.57 | masked | having a little trouble with it now |
| 03:55.51 | rob0 | I had no problem getting it to work. But disconnect supervision seems broken, as described on the wiki. |
| 03:55.51 | Supaplex | call the OEM nehhehahahaa |
| 03:55.57 | rob0 | hahaha |
| 03:56.14 | Supaplex | disconnect supervision is the only real issue I had with mine |
| 03:57.01 | kc5cqm | iCEBrkr, did you use the fxo_ks or fxs_ks signaling? |
| 03:57.18 | kc5cqm | on the 101p that is |
| 03:57.21 | iCEBrkr | fxsks=1 |
| 03:57.34 | elvisthedj | i take it that's a no.. |
| 03:58.15 | kc5cqm | same |
| 03:58.22 | [av]bani | o noes |
| 03:58.38 | rob0 | mine is fxsks=1 too |
| 03:58.54 | masked | same and it did work earlier |
| 03:58.56 | kc5cqm | now in zapata.conf I have signalling=fxs_ks |
| 03:59.15 | Jabroni | questin.. is it posible to send video from sip clients to another * box via iax ? |
| 03:59.18 | *** join/#asterisk j4m3s_ (n=j4m3s@24.96.145.117) |
| 03:59.46 | ManxPower | I wasn't aware IAX2 supported Video |
| 04:00.03 | Qwell | ManxPower: I don't think it does |
| 04:00.15 | Supaplex | don't go installing asterisk on your tivo now |
| 04:00.15 | kc5cqm | oh holy crap...got it working |
| 04:00.16 | kc5cqm | ;-) |
| 04:00.33 | masked | kc5cqm what did u do? |
| 04:00.48 | masked | kc5cqm and what errors did u receive when it didn't work? |
| 04:01.12 | Jabroni | Supaplex its not about tivo.. i was thinkin in having soft phones connected to each * on each office, and use iax since thats the way I have interconnected all my * |
| 04:01.30 | kc5cqm | masked: I had channels=>1 |
| 04:01.32 | kc5cqm | not channel => 1 |
| 04:01.37 | masked | odd |
| 04:01.41 | masked | i have channels atm |
| 04:01.53 | kc5cqm | I was only getting "could not create channel of type Zap (0 - unknown) |
| 04:01.54 | masked | well |
| 04:01.57 | iCEBrkr | kc5cqm: Fucking eh.. I'm sick of Asterisk being all Python like... |
| 04:02.02 | iCEBrkr | ....white space sensitive |
| 04:02.14 | Qwell | iCEBrkr: whitespace and extra chars |
| 04:02.14 | rob0 | I think it was the "s" |
| 04:02.17 | kc5cqm | and I saw documentation both ways |
| 04:02.21 | *** join/#asterisk Prival (i=user69@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 04:02.22 | Qwell | yeah, damn asterisk for not knowing about the s |
| 04:02.23 | kc5cqm | got some nasty echo |
| 04:02.28 | iCEBrkr | Oh, I missed that part |
| 04:02.28 | kc5cqm | but I had the echo shit commented out |
| 04:03.35 | masked | hrmm |
| 04:03.44 | masked | well thats not the prob im gettin' then |
| 04:03.47 | Prival | Hi all, anyone can give me hints on troubleshooting echo? I have a customer with a PRI and on some call it's all ok, on some other call he gets echo for 15-20s and some calls no echo at the beginning, but a lot of echo after 20-30seconds... |
| 04:04.36 | kc5cqm | ok, now gotta get inbound wokring hehehe |
| 04:04.43 | kc5cqm | this should be simple |
| 04:05.48 | elg | my x101p has a hard time detecting remote disconnect too |
| 04:06.06 | bweschke | Prival: what echo canceler with zaptel are you using? |
| 04:06.50 | Supaplex | cotton swabs in a cardboard tube |
| 04:07.00 | bweschke | lol |
| 04:07.07 | Supaplex | s/swabs/balls/ |
| 04:07.28 | kc5cqm | I found some X100P cards on ebay for $15 that claim to not be clones... |
| 04:07.51 | Supaplex | the flooded clone marked has driven the price down |
| 04:08.02 | Supaplex | CYA, ask the seller for the full scoup |
| 04:08.27 | rob0 | That's a lie, Digium isn't selling them now! |
| 04:08.35 | Supaplex | they could be used |
| 04:08.36 | elg | how can you tell if it's a clone, if you already have one? |
| 04:08.44 | elg | I imagine mine is, but I'm curious |
| 04:08.52 | Supaplex | something in proc iirc |
| 04:09.22 | kc5cqm | rob0, they could be old stock |
| 04:09.39 | rob0 | <== still thinks it's a lie :) |
| 04:09.43 | masked | mine is supposed to be genuine |
| 04:09.51 | masked | who cares tho its only a winmodem |
| 04:11.12 | *** join/#asterisk wasan (i=wer@ip70-178-95-216.ma.dl.cox.net) |
| 04:11.20 | wasan | what is aterisk? |
| 04:11.24 | masked | lol |
| 04:11.25 | Supaplex | this * |
| 04:11.39 | Supaplex | what is wasan? |
| 04:12.02 | iCEBrkr | wasan: www.asterisk.org |
| 04:12.03 | masked | wasan: latest version fixed some pixel display problems with all fonts so the asterisk displays properly, this is version 1.2.4 |
| 04:12.15 | masked | * see looks neat hey. |
| 04:12.37 | iCEBrkr | haha |
| 04:13.40 | rob0 | ~asterisk |
| 04:13.41 | jbot | i guess asterisk is the best free PBX in the world |
| 04:14.23 | wasan | how can I use a pbx at my house? |
| 04:14.31 | Supaplex | jbot: no, asterisk is <reply> asterisk is the best free PBX in the world. |
| 04:14.32 | jbot | okay, Supaplex |
| 04:14.38 | Supaplex | ~asterisk |
| 04:14.39 | jbot | asterisk is the best free PBX in the world. |
| 04:14.39 | *** join/#asterisk jasonwolfe0u812 (n=jasonwol@adsl-072-151-106-082.sip.asm.bellsouth.net) |
| 04:14.49 | mogorman | heh |
| 04:14.53 | Supaplex | no abiguity on that now :) |
| 04:15.02 | jasonwolfe0u812 | is there a way to play audio to only one channel of two that are natively bridged? |
| 04:15.17 | wasan | asterisk make me horny |
| 04:15.57 | Supaplex | tmi |
| 04:16.01 | tronix | ohm! ohm! (watt? watt?!) |
| 04:16.12 | masked | omgwtfbbq |
| 04:16.29 | wasan | I just saved up enought to buy a license to register domain names |
| 04:16.50 | wasan | which one should I register first? |
| 04:17.02 | Supaplex | as a registrar, a reseller, or an end user? |
| 04:17.19 | tronix | omgwftbbq.com :-) |
| 04:17.27 | rt | pretty consistently, the audio that goes to console/dsp is completely garbled. Not sure what's going on. Any ideas? |
| 04:17.38 | Supaplex | asterisk-makes-wasan-horny.com |
| 04:18.16 | wasan | end user |
| 04:18.24 | masked | wasan how much for .au tlds? |
| 04:18.42 | jasonwolfe0u812 | anyone here know anything about how to emulate a 'whisper' in voip? |
| 04:18.54 | Qwell | jasonwolfe0u812: just talk quietly |
| 04:19.02 | jasonwolfe0u812 | nice |
| 04:19.36 | Supaplex | %-) |
| 04:19.36 | ManxPower | jasonwolfe0u812, you mean like has been discussed on the mailing lists over the past week or two? |
| 04:19.38 | wasan | I dont know |
| 04:19.51 | jasonwolfe0u812 | ahhh... I follow the list but didn't see anything like that... asterisk-users? |
| 04:19.59 | jasonwolfe0u812 | how can I search the list? |
| 04:20.28 | wasan | I need to masturbate |
| 04:20.31 | Prival | bweschke> Sorry, I was away... Let me look at the source |
| 04:20.42 | ManxPower | jasonwolfe0u812, heck if I know. It's not something I care about, but I vaguely recall the issue was developement related and the end result is that it's not easy to impliment, and there is no such feature in asterisk currently. |
| 04:20.57 | ManxPower | ~mailinglist |
| 04:20.59 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 04:21.29 | jasonwolfe0u812 | Thanks! |
| 04:22.17 | ManxPower | google is always behind, the asteriskguru link should have current messages. |
| 04:22.19 | *** join/#asterisk dalabera (n=dalabera@adsl-9-131-236.mia.bellsouth.net) |
| 04:22.31 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 04:23.05 | ManxPower | This whole FAP thing with DirecTV internet sure is a pain in the ass. |
| 04:23.19 | ManxPower | brb |
| 04:24.51 | jasonwolfe0u812 | so maybe in searching, i'll find this answer too, but... can someone clarify... can't two channels that are bridged be broken apart and then put back together... then you could put them in different extensions and use playback, then bridge |
| 04:25.05 | wasan | i need to masturbate |
| 04:25.16 | jasonwolfe0u812 | or am I going around the world to cross the street... or missing the point altogether |
| 04:25.28 | ManxPower | jasonwolfe0u812, the answer is no |
| 04:25.51 | Prival | bweschke> Isn't that supposed to be in the Makefile of zaptel? |
| 04:25.54 | wasan | one handed fuck |
| 04:26.14 | bweschke | Prival: zconfig.h |
| 04:26.27 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 04:26.37 | jasonwolfe0u812 | doesn't -t in dial stop native bridging so that users can transfer... even those can't be kept seperate? |
| 04:27.29 | *** join/#asterisk ManxPowe (n=ewieling@dpc6745150107.direcpc.com) |
| 04:28.10 | Prival | bweschke> Ok, found it I have ECHO_CAN_MARK2 |
| 04:28.17 | *** join/#asterisk ManxPowe (n=ewieling@dpc6745150107.direcpc.com) |
| 04:28.39 | *** join/#asterisk wassabi (i=identd@sol.eyz.us) |
| 04:28.39 | rt | okay, another stupid question: when you are executing background, does it just read a single digit extension to transfer to, or any extension defined in the current context? |
| 04:28.53 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 04:28.54 | bweschke | Prival: try EHCO_CAN_MG2 |
| 04:29.00 | Prival | bweschke> I don't have AGGRESSIVE_SUPPRESSOR defined |
| 04:29.08 | bweschke | Prival: you probably don't need it |
| 04:29.35 | bweschke | Aggressive is basically gonna make your conversations like a half duplex speakerphone to try and cut echo |
| 04:29.36 | Prival | bweschke> That must be in 1.2.x, because I don't see that in mine (1.0.9). We can't make the switch to 1.2.x right now. |
| 04:29.42 | bweschke | oh |
| 04:29.43 | bweschke | ya |
| 04:29.47 | bweschke | hmmm.. |
| 04:29.52 | bweschke | that's gonna be a problem |
| 04:29.54 | jasonwolfe0u812 | rt, if there is an extension with more than one digit that you could possibly mean, it will wait until you have entered more digits I think |
| 04:30.01 | bweschke | EHCO_CAN_MG2 is real effective |
| 04:30.09 | bweschke | we've had real good success with it |
| 04:30.24 | Prival | bweschke> Do you think it could be backported to 1.0.9? |
| 04:30.50 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 04:30.52 | wassabi | anyone have a recommendation for a single or double port FSO supporting IAX? is echo cancellation needed on FSOs as well? anyone seen the x100p.com FSO, and is it junk? |
| 04:31.00 | wassabi | err. IAX2 |
| 04:31.01 | Qwell | Prival: I'd say that is very unlikely |
| 04:31.07 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
| 04:31.08 | bweschke | Prival: have you tried running Zaptel 1.2 w/asterisk 1.0.9? I don't know if this will work, but you might have better luck with that than you will backporting MG2 |
| 04:31.22 | *** part/#asterisk jasonwolfe0u812 (n=jasonwol@adsl-072-151-106-082.sip.asm.bellsouth.net) |
| 04:31.40 | bweschke | wassabi: FSO ? |
| 04:31.41 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 04:31.50 | Prival | bweschke> That is worth a try... |
| 04:32.06 | wassabi | sorry. FXS, I know bad first impression |
| 04:32.22 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 04:33.07 | dalabera | !list |
| 04:34.45 | ManxPowe | wasan, are you looking for an FXO or an FXS? |
| 04:34.49 | ManxPowe | ~fxofxs |
| 04:34.51 | jbot | hmm... fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 04:35.11 | ManxPowe | ~fxsfxo |
| 04:35.13 | jbot | fxsfxo is probably An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 04:35.38 | wassabi | yeah, I know. sorry.. I meant FXS.. I'd like to buy an external FXS that supports IAX2, like the iaxy's and I'd like a recommendation |
| 04:35.51 | ManxPowe | wasan, the IAXy is the only one |
| 04:36.04 | ManxPowe | the X100P is an FXO and it's a card |
| 04:36.15 | wassabi | I know.. check this out.. lemme know if this looks like a scam |
| 04:36.15 | wassabi | http://www.x100p.com/products_2.htm |
| 04:36.17 | wunderkin | no im pretty sure someone mentioned an iax ata but it must not be very popular |
| 04:36.31 | wassabi | yeah.. an ata.. |
| 04:36.56 | rob0 | That's a single port, isn't IAXy a dual port? |
| 04:37.18 | wassabi | no, I don't think so |
| 04:37.50 | ManxPowe | the IAXy is 1 port |
| 04:38.09 | ManxPowe | That device looks like one of the AT186 or whatever that chip is. |
| 04:38.12 | *** part/#asterisk dalabera (n=dalabera@adsl-9-131-236.mia.bellsouth.net) |
| 04:38.25 | ManxPowe | wassabi, basically you won't get help for such a device here since nobody here has them |
| 04:38.51 | wassabi | gotcha |
| 04:38.52 | ManxPowe | I'll stick with SIPura, thankyouverymuch |
| 04:39.10 | wassabi | :) |
| 04:39.25 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 04:39.52 | wassabi | any that support IAX2 directly? I suppose I could take a chance on this company, but I'm sure as heck not buying two until I know if they're junk |
| 04:40.28 | wassabi | the reason I'm asking for IAX support is because I'm worried about SIP through NAT and all those complications |
| 04:41.19 | wassabi | the IAX protocol doesn't seem to have any problems with NAT from what I've read |
| 04:43.38 | wassabi | I'm going to idle for a bit.. if I don't hear anything I'll go with the iaxy.. thanks for the comments all |
| 04:44.04 | slan | Err: _Sam-- Are you here? Got the GS phone all updated with new firmware. |
| 04:44.13 | *** join/#asterisk schuylerdigium (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 04:44.58 | *** join/#asterisk jgomata (n=jgomata@red-corp-201.143.78.76.telnor.net) |
| 04:45.56 | justinu | sip works fine with nat |
| 04:46.26 | wasan | bitch |
| 04:46.29 | wasan | shudup |
| 04:46.35 | wasan | i dont ewant to know |
| 04:46.36 | wasan | your name |
| 04:46.37 | wasan | i want |
| 04:46.39 | wasan | bang bang bnag |
| 04:47.30 | *** join/#asterisk B_Rando (n=dotirc@c-24-0-180-132.hsd1.tx.comcast.net) |
| 04:49.54 | B_Rando | Does anybody know how to incomming calls via broadvoice to work? I can dial out via broadvoice perfectly. |
| 04:52.18 | kc5cqm | hey, with a zap device, is it possible to leave it in a ringing state (unanswered) until an extension is answered? |
| 04:52.30 | droops | B_Rando |
| 04:52.36 | droops | in your sip.conf |
| 04:52.38 | ManxPowe | kc5cqm, of course, that's the default. |
| 04:52.47 | droops | when you register with broatvoice |
| 04:52.53 | droops | the end is something liek this |
| 04:52.55 | droops | @sip.broadvoice.com/201 |
| 04:53.05 | kc5cqm | well, I have that context started with an s,1,Answer ...so it doesn't |
| 04:53.08 | droops | that 201 is the extension in extensions.cong that will handle the call |
| 04:53.11 | B_Rando | I don't have the extention on the end, is that necessary? |
| 04:53.27 | droops | if you want it to go somewhere |
| 04:53.31 | iCEBrkr | B_Rando: Huh, if you want to get calls |
| 04:53.53 | droops | i had that problem 2, B_Rando, so dont feel bad |
| 04:54.00 | droops | hence why i know the answer |
| 04:54.05 | wassabi | :) |
| 04:54.37 | B_Rando | droops: Okay, I'll try the extension specification.... |
| 04:54.58 | kc5cqm | ManxPowe, how are you handling incoming calls on your zap to do that? |
| 04:55.44 | wassabi | I tried Broadvoice.. took forever to find the right configuration.. worked great with Asterisk, but the voice quality was horrible.. perhaps it was just my area |
| 04:56.48 | B_Rando | droops: I added the extension to the end of the registration line, but it didn't fix it. |
| 04:57.18 | iCEBrkr | B_Rando: You need a inbound context and an extension assignment for it to work |
| 04:58.51 | B_Rando | I have a context [from_broadvoice]... do I need an exten => rule to point to my extension? |
| 04:59.04 | iCEBrkr | Yeah |
| 04:59.09 | iCEBrkr | How else are you going to ring your phone? |
| 04:59.39 | iCEBrkr | ...and take voicemail, etc |
| 04:59.42 | *** part/#asterisk schuylerdigium (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 04:59.52 | wassabi | are there any books on asterisk you guys recommend? or just use the web sites? |
| 05:00.01 | *** join/#asterisk rajiv|wo1k (n=rajiv@gentoo/developer/rajiv) |
| 05:00.11 | iCEBrkr | Use the wiki ... it's almost always updated.. kinda.. |
| 05:00.32 | wassabi | k |
| 05:02.19 | droops | B_Rando |
| 05:02.25 | droops | in your sip.conf |
| 05:02.38 | droops | when you do the whole [sip.broadvoice.com] thing |
| 05:02.47 | droops | you need to have a context=something |
| 05:02.55 | droops | mine is context=incoming |
| 05:03.03 | droops | then in my extensions.conf |
| 05:03.11 | droops | i ahve a [incoming] |
| 05:03.42 | B_Rando | yes, I have [from_broadvoice] |
| 05:03.47 | droops | exten => 100,1,goto(201,1) |
| 05:03.47 | droops | exten => 201,1,Answer() |
| 05:03.50 | droops | errr |
| 05:03.54 | droops | just this |
| 05:03.56 | droops | exten => 201,1,Answer() |
| 05:04.19 | dogtanian | context = from-pstn is normally a good way to play incoming calls |
| 05:04.21 | droops | that tells all calls from broadvoice to go to [incoming] and extension 201 |
| 05:05.00 | droops | i need to be more descriptive in my dialplan, so when it gets bigger, ill know whats going on |
| 05:05.16 | B_Rando | Is the "Answer()" necessary? Can't I just pick up the phone/extension? |
| 05:08.49 | *** join/#asterisk Mark5 (n=mar@201.144.181.242) |
| 05:09.15 | Mark5 | hello to everybody, I am from Mexico |
| 05:09.54 | Mark5 | ¿se puede hablar español? |
| 05:10.14 | wassabi | Answer() apparently tells Asterisk to pick up the line.. I would think that as long as you're signed into your extension it would ring through as long as you didn't call Answer() |
| 05:10.16 | wassabi | just a guess though |
| 05:10.25 | wassabi | ^ http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x518.html |
| 05:11.25 | Mark5 | there are any super user of voip? who talks spanish to give a update about this chatroom? |
| 05:11.25 | *** join/#asterisk denon (i=denon@209.161.207.62) |
| 05:11.25 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:12.09 | droops | B_Rando |
| 05:12.14 | droops | i use the answer() |
| 05:12.26 | droops | so that asterisk pics it up, and then goes into my ivr |
| 05:12.47 | kc5cqm | how can you not use answer and ring other extensions...keeping the calling line 'ringing' |
| 05:13.17 | iCEBrkr | Don't think you can |
| 05:13.32 | *** join/#asterisk Pegger (n=peg@pool-68-163-180-64.bos.east.verizon.net) |
| 05:13.50 | iCEBrkr | ..and I'm stil not sure why people want to do that |
| 05:14.20 | kc5cqm | iCEBrkr, so I can keep my voicemail on packet8 |
| 05:14.21 | kc5cqm | ;-) |
| 05:14.27 | Pegger | do most voip componies connect to t1 or just but termination from the big guys like level3 and sprint |
| 05:14.48 | Pegger | *do most voip componies connect to t1 for termination or just buy termination from the big guys like level3 and sprint |
| 05:14.51 | Mark5 | hola alguien que hable español? |
| 05:15.14 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 05:15.47 | Supaplex | does asterisk still keep the cvs date w/the version number in the makefile? |
| 05:16.30 | Mark5 | well i see, anybody who wants to help me to install a asterisk in my server? |
| 05:18.20 | Pegger | Supaplex, they use subversion now |
| 05:18.24 | Pegger | Mark5, what help do you need |
| 05:18.58 | flashnet | Mark5 en esta sala no se habla español |
| 05:19.34 | Mark5 | yes pegger |
| 05:19.38 | Supaplex | still using a makefile right? |
| 05:19.55 | Supaplex | humm.. I guess I'll dig around |
| 05:20.01 | Mark5 | okey i not gonna speak espanish |
| 05:20.05 | Pegger | Supaplex, yup |
| 05:20.15 | Pegger | Mark5, what kind of help do you need |
| 05:20.39 | Mark5 | well i am tryin to setup voip channel in my server |
| 05:20.42 | Pegger | <PROTECTED> |
| 05:20.59 | Mark5 | using of course asterisk, it´s just for my internal use |
| 05:21.18 | Pegger | Mark5, channel? you mean a meet me, or voice bridge? |
| 05:21.32 | Mark5 | yes like voice bridge |
| 05:21.59 | Mark5 | flashnet i can give you my msn address? to keep in touch with you? |
| 05:22.23 | flashnet | claro |
| 05:22.44 | *** join/#asterisk fugitivo (n=ajf@201.255.179.144) |
| 05:22.48 | Mark5 | it´s cursormx@hotmail.com |
| 05:23.01 | flashnet | ahora te agrego |
| 05:23.39 | Mark5 | thxs u |
| 05:23.48 | flashnet | not at all |
| 05:25.08 | Pegger | Mark5, compile in the timer to the kernel |
| 05:25.12 | *** join/#asterisk alk (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 05:25.37 | wassabi | droops, on your example "exten => 201,1,Answer()".. after that line, it would just ring through to whoever is signed into extension 201? |
| 05:26.19 | wassabi | so Answer() is needed to patch the call through? |
| 05:26.27 | Pegger | i have a intersting perdicitment i am getting this error Feb 1 00:25:17 NOTICE[29099]: chan_iax2.c:6776 socket_read: Rejected connect attempt from 69.25.143.141, who was trying to reach '1617xxxxxxx but i have the extension in extensions.conf anyone know what the error means |
| 05:26.28 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 05:27.36 | droops | wassabi |
| 05:27.40 | *** join/#asterisk lithi (n=interp3@HSE-Quebec-City-ppp128693.qc.sympatico.ca) |
| 05:27.42 | droops | after that 201,1 |
| 05:27.45 | droops | i have exten => 201,n,wait(2) |
| 05:27.45 | droops | exten => 201,n,Background(greeting) |
| 05:27.45 | droops | exten => 201,n,Background(sounds/silence/10) |
| 05:27.45 | droops | exten => 201,n,Background(hurryup) |
| 05:27.46 | droops | exten => 201,n,Hangup() |
| 05:27.52 | droops | that plays my greeting |
| 05:27.58 | droops | and waits 20 seconds |
| 05:28.08 | droops | so that people have time to dial the extension they want |
| 05:28.18 | droops | if they dont dial an extension then they get the hangup |
| 05:28.42 | kc5cqm | I just ring all my extensions at once for all incoming contexts |
| 05:28.43 | wassabi | gotcha |
| 05:28.44 | kc5cqm | hehe |
| 05:28.50 | kc5cqm | but then again...this is at my house |
| 05:28.52 | wassabi | I'm a total newbie, spoiled by AMP :) |
| 05:28.57 | wassabi | and asterisk@home :) |
| 05:28.58 | droops | its all good |
| 05:28.59 | kc5cqm | never touched amp |
| 05:29.08 | droops | i took the astercon training |
| 05:29.15 | wassabi | got pretty far considering I don't know crap about dialplans :) |
| 05:29.16 | droops | and im getting better all the time |
| 05:29.33 | droops | for the phone on my desk, i have this |
| 05:29.36 | droops | exten => 0,1,dial(SIP/phone1,20) |
| 05:29.36 | droops | exten => 0,n,congestion() |
| 05:29.37 | droops | exten => 0,n,busy() |
| 05:29.46 | wassabi | awesome.. so what's the code to direct to the start once they pick an extension off 201? |
| 05:29.55 | wassabi | ,20 ? |
| 05:29.58 | wassabi | wassat? |
| 05:30.05 | wassabi | and what's ,n ? |
| 05:30.06 | droops | so durring the greeting and 20 secnds afterwards, if they press 0 they get my sip phone1 |
| 05:30.22 | wassabi | cause of 0, right? |
| 05:30.27 | droops | n replaces the 1,2,3,4,5,6,7 |
| 05:30.30 | droops | and yes because of 0 |
| 05:30.47 | droops | if you use n, then if you add something, you dont have to go back through and change all the numbers |
| 05:30.49 | wassabi | is it literally n in the code or are you summarizing it? |
| 05:30.54 | droops | its really n |
| 05:30.58 | wassabi | its a catchall, I take it? |
| 05:31.07 | droops | its just goes to the next one |
| 05:31.21 | lithi | wassabi: n = next in order it is in in extensions.conf |
| 05:31.51 | wassabi | could it be written out in another form? |
| 05:32.00 | wassabi | trying to wrap my mind around that. |
| 05:32.08 | droops | i could have used |
| 05:32.14 | kc5cqm | g'nite y'all |
| 05:32.15 | *** part/#asterisk kc5cqm (n=kc5cqm_@cpe-68-206-116-214.stx.res.rr.com) |
| 05:32.27 | wassabi | is it like 0,2,congestion() 0,2,busy() and 0,3,congestion() 0,3,busy() ? |
| 05:32.42 | wassabi | .. etc? |
| 05:32.50 | wassabi | like, shortform for that? |
| 05:33.11 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 05:33.39 | Pegger | does anyone know what this error means Feb 1 00:25:17 NOTICE[29099]: chan_iax2.c:6776 socket_read: Rejected connect attempt from 69.25.143.141, who was trying to reach '1617xxxxxxx but i have the extension in extensions.conf anyone know what the error means |
| 05:33.58 | wassabi | what does the 1617 refer to? hmm. |
| 05:34.02 | BugKham | can anyone care to answer newbie questions? |
| 05:34.11 | wassabi | droops sure is ;) |
| 05:34.12 | droops | exten => 0,1,dial(SIP/phone1,20) |
| 05:34.12 | droops | exten => 0,2,congestion() |
| 05:34.12 | droops | exten => 0,3,busy() |
| 05:34.12 | droops | but if i added something above congestion, i would have to change the 2 to a 3, and the 3 to a 4 |
| 05:34.12 | droops | and this is 2006, changing line numbers is something we did in basic |
| 05:34.12 | droops | as i was told |
| 05:34.12 | droops | =o) |
| 05:34.46 | wassabi | oop.. I'm sorry.. I mean 2,1,congestion() 2,2,busy() and 3,1,congestion() 3,2,busy() |
| 05:34.51 | wassabi | reading your code drooops... |
| 05:35.08 | BugKham | I am about to get an E100P but there are some things I'm not sure about |
| 05:35.15 | Pegger | wassabi, that is the area code of the 10 digit number that is trying to ring me |
| 05:35.22 | wassabi | ahh. |
| 05:35.39 | wassabi | ok, gotcha now totally droops |
| 05:36.11 | BugKham | ISDN PRI is capable of handling fxo signalling but what are the applications? |
| 05:36.14 | wassabi | Pegger, I had that same error I believe when I was trying to get a call in from Broadvoice.. I think it was context. |
| 05:36.27 | wassabi | I think I had to point it to the right context and it went through, but not sure. |
| 05:37.19 | BugKham | for TDM400P, fxs ports are used to connect analog phones but what about on ISDN PRI? |
| 05:37.47 | *** join/#asterisk many_ (i=many@krikkit.ukeer.de) |
| 05:37.52 | wassabi | aren't those for connecting asterisk together via ISDN? |
| 05:38.11 | wassabi | or for taking ISDN calls? |
| 05:38.35 | Pegger | wassabi, the weird thing is that it used to work, and I am not sure what changed to make it not work |
| 05:39.03 | BugKham | wassabi: for taking ISDN calls I need fxs signalling, right? |
| 05:39.03 | wassabi | where is the incoming call coming from? do you have to somehow associate your asterisk server with that? |
| 05:39.19 | wassabi | do you have a specific card model? |
| 05:39.23 | wassabi | it is on digium? |
| 05:39.25 | BugKham | wassabi: E100P |
| 05:39.29 | wassabi | oh, yeah |
| 05:39.39 | BugKham | wassabi: I am about to get one |
| 05:39.40 | wassabi | lookin |
| 05:39.42 | droops | np wassabi |
| 05:40.14 | BugKham | wassabi: but I saw in the sample config some channels can be set to fxs |
| 05:40.38 | wassabi | umm.. BugKham.. I think that's for connecting asterisk to asterisk.. not for taking incoming ISDN calls.. oh really? that's cool.. ISDN incoming would be nice |
| 05:40.57 | wassabi | maybe FXS ports on it were unintended but supported somehow? |
| 05:41.22 | wassabi | do you have the URL? maybe someone here can take a look |
| 05:41.40 | wassabi | ^ regarding those sample configs. |
| 05:41.56 | *** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo) |
| 05:42.57 | wassabi | Pegger, found this URL maybe it may help?? http://www.voipuser.org/forum_topic_394.html |
| 05:43.24 | BugKham | wassabi: sure, it's http://www.voip-info.org/wiki/view/Asterisk+PRI |
| 05:44.01 | wassabi | Pegger, someone said "Rejected connection is normally because the context/extension doesn't exist. Try typing: iax2 debug" |
| 05:44.23 | wassabi | Pegger, sure its not a context issue? or maybe need to mark as peer or something? |
| 05:45.25 | wassabi | BugKham, your DSL in Europe is definately more interesting than most in the US.. we just get a few channels.. you guys get a ton. |
| 05:45.30 | wassabi | reading your URL. |
| 05:46.58 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:47.57 | *** join/#asterisk wunderkin- (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 05:47.58 | wassabi | BigKham, yeah.. kinda looks like you may be able to assign a zap channel to each ISDN channel, hmm |
| 05:48.32 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 05:48.36 | BugKham | wassabi: more specifically it's in http://www.digium.com/downloads/configuring_zaptel.pdf |
| 05:49.24 | wassabi | it would make sense that each would be individually addressable.. I don't know if FXO and FXS really apply to ISDN though. |
| 05:49.26 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:49.33 | BugKham | =#:E1@4 |
| 05:49.37 | BugKham | sorry |
| 05:49.39 | wassabi | np. |
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| 05:52.24 | wassabi | oop.. BugKham I mean.. I just wonder if the ISDN protocol used for voice is the same used for each ISDN channel.. if it is it seems like it could potentially work |
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| 05:52.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:52.57 | masked | who is familiar with x100p's? |
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| 05:53.19 | masked | can someone tell me if /dev/zap/1 works or not without a pots line plugged in? |
| 05:53.21 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) [NETSPLIT VICTIM] |
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| 05:53.29 | masked | asterisk thinks there is no such device |
| 05:53.40 | masked | oh yeah i forgot there was a netsplit |
| 05:53.40 | masked | haha |
| 05:55.05 | masked | X-Rob r u there? save me :P |
| 05:55.43 | wassabi | BugKham, "The T100P is a single span T-1 (24-channel) card. This card supports both voice and data modes on its single-T span. The T100P supports standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC and Frame Relay modes. The T100P can also be connected to channel banks for use with Asterisk." |
| 05:55.58 | *** join/#asterisk SibRphrek (i=SibrPhre@user-12lccke.cable.mindspring.com) [NETSPLIT VICTIM] |
| 05:55.59 | wassabi | "The E100P version is essentially the same card supporting the E-1 European standard." |
| 05:56.04 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) [NETSPLIT VICTIM] |
| 05:56.15 | wassabi | sounds promising |
| 05:56.28 | Qwell | t100p is old. get the te100p |
| 05:56.30 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) [NETSPLIT VICTIM] |
| 05:56.57 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 05:57.08 | wassabi | trying to find out if the E100P can be used with multiple ISDN voice lines |
| 05:58.53 | BugKham | wassabi: I will need E100P with all channels configured for making calls |
| 05:59.16 | BugKham | wassabi: where can I find the configuration? |
| 06:00.09 | wassabi | not finding it on google and I don't know where else to look.. tried searching for the last 10 minutes for you.. that zaptel configuration you found looked promising |
| 06:00.10 | masked | wasim that card (t100p) will work with upto 24 voice channels |
| 06:00.20 | masked | the e100p likely 32? |
| 06:00.22 | Qwell | that card is old... |
| 06:00.23 | masked | i forget. |
| 06:00.25 | Qwell | get the te100p |
| 06:00.40 | Qwell | You likely won't even be able to find those anymore...don't waste your time trying |
| 06:00.53 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 06:01.16 | BugKham | Qwell: the new one is still costly |
| 06:01.32 | BugKham | Qwell: I might get started with E100P first |
| 06:02.20 | BugKham | wassabi: thanks, I think I will just follow the example in the wiki |
| 06:02.53 | wassabi | cool.. I'm new here anyway.. I'll probably hang out and try to learn more |
| 06:03.36 | BugKham | wassabi: =-) so am I |
| 06:04.01 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 06:04.06 | wassabi | its not like there's a lot of good documentation, but its definately gaining momentum |
| 06:04.12 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
| 06:04.17 | *** join/#asterisk sack (n=sack@244.Red-81-38-35.dynamicIP.rima-tde.net) [NETSPLIT VICTIM] |
| 06:04.18 | wassabi | I'm interested in the aspect of hosting Asterisk somewhere and using it as a hosted PBX. |
| 06:04.30 | wassabi | don't know much about hardware and stuff.. |
| 06:04.34 | wassabi | in terms of these cards.. |
| 06:04.47 | masked | wassabi so am i |
| 06:04.48 | BugKham | wassabi: what type of hardware r u using/ |
| 06:05.06 | Gordo | Can anyone tell me why my telco tells me to "please check teh number and try again then I am using exten => _9.,1,Dial(SIP/${EXTEN:1}@pstn,60,tr) to dial from * thru a spa-3k |
| 06:05.36 | BugKham | wassabi: I have TDM stuff running quite perfectly here |
| 06:05.40 | masked | have the two of u seen asteriskdocs.org? |
| 06:06.05 | wassabi | server = dual processor opterons running Linux.. no hardware at my place yet.. I want an IAX2 capable ATA to test with |
| 06:06.36 | wassabi | I'm looking at either an iaxy or this: http://www.x100p.com/products_2.htm |
| 06:06.38 | BugKham | masked: me? |
| 06:06.40 | *** join/#asterisk lorinc (n=ang@caracas-0209.adsl.interware.hu) [NETSPLIT VICTIM] |
| 06:06.48 | wassabi | masked, yeah.. I just found that site, luckily |
| 06:06.58 | masked | BugKham either of u |
| 06:07.37 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com) [NETSPLIT VICTIM] |
| 06:07.51 | BugKham | masked: I normally go to www.voip-info.org |
| 06:07.56 | wassabi | I'm a bit leery of buying one of the ATAs from x100p.com because they don't have a high rep on ebay where they apparently do quite a bit of business.. but their rating is like 98.5% |
| 06:08.03 | wassabi | yeah.. that's where I found this ATA I'm considering |
| 06:08.07 | masked | only reason i suggest it is for the oriellys book |
| 06:08.13 | wassabi | ahh |
| 06:08.19 | wassabi | yeah, I saw that.. didn't check that book out yet though |
| 06:08.21 | wassabi | will soon. |
| 06:08.47 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) [NETSPLIT VICTIM] |
| 06:09.32 | wassabi | I want to get two ATAs, one for me and one for my business partner, so we can pass our calls through a hosted Asterisk server |
| 06:09.42 | wassabi | just to start off.. |
| 06:10.17 | wassabi | I have root access on the box so I can configure whatever.. |
| 06:10.32 | wassabi | .. and plenty of bandwidth.. |
| 06:10.44 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 06:14.18 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 06:14.42 | *** join/#asterisk pb__ (n=pb@cpc3-cmbg6-5-1-cust33.cmbg.cable.ntl.com) [NETSPLIT VICTIM] |
| 06:15.52 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 06:16.22 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 06:20.51 | *** join/#asterisk Psykick (n=anon@203-167-215-33.dsl.clear.net.nz) |
| 06:22.15 | *** join/#asterisk svenna_ (n=svenna@p548D23B9.dip0.t-ipconnect.de) |
| 06:22.20 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 06:23.35 | Gordo | can anyone tell me what teh "TR" means at the end of exten => _9.,1,Dial(SIP/${EXTEN:1}@pstn,60,tr) |
| 06:24.30 | shido6 | show application Dial |
| 06:24.32 | shido6 | and look for T |
| 06:24.34 | shido6 | and t |
| 06:24.35 | shido6 | and r |
| 06:24.52 | shido6 | do a show application Dial at the CLI |
| 06:25.27 | Gordo | okso it wont have any effect on the actaul number that is dialed..? |
| 06:26.23 | *** join/#asterisk ThaZZa_Work (n=me@CPE-221-121-158-1-DSL.hypermax.net.au) |
| 06:31.11 | *** join/#asterisk wassabi (i=identd@sol.eyz.us) |
| 06:32.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 06:32.13 | *** join/#asterisk coyote10 (i=jamil@unaffiliated/coyote10) |
| 06:33.04 | coyote10 | algun canal de asterisk en spanish? |
| 06:33.38 | *** part/#asterisk coyote10 (i=jamil@unaffiliated/coyote10) |
| 06:39.17 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 06:39.32 | *** join/#asterisk j4m3s_ (n=j4m3s@24.96.145.117) |
| 06:39.57 | *** join/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net) |
| 06:48.51 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 06:49.29 | trixter | hey benjk |
| 06:49.39 | benjk | hi |
| 06:50.29 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 06:51.59 | trixter | how much longer are you here for? |
| 06:52.18 | benjk | about a week |
| 06:52.28 | trixter | wow you are making a holiday out of this :P |
| 06:52.37 | benjk | not really |
| 06:52.43 | benjk | work holiday if you want |
| 06:52.48 | trixter | heh |
| 06:52.56 | trixter | I wouldnt want to stay in that area for that long |
| 06:53.23 | trixter | but then I dont like that area generally |
| 06:53.25 | benjk | well I had a couple of meetings ligned up and now I will visit two customers |
| 06:53.38 | benjk | do you mean that riff-raff on the streets of SF |
| 06:53.53 | trixter | no .,.. well yeah all the communist hippies that live there |
| 06:54.02 | trixter | I just dont like the people there generally |
| 06:54.07 | benjk | they can be happy I don't live here |
| 06:54.26 | benjk | I'd probably go into politics and not rest before the streets are clean and sterile |
| 06:55.26 | trixter | yeah good luck with that |
| 06:55.28 | benjk | In London they still have a very old law in place that makes it an offence not to have any money on you |
| 06:55.39 | trixter | more dead people would vote against you than any live ones would vote for you |
| 06:55.47 | trixter | you do know that SF has the highest count of dead voters right? |
| 06:55.53 | benjk | using this, the police can remove anyone from the city limits if they don't have any money |
| 06:56.03 | trixter | SF has a vagrant law, a lot of cities do but it cant be enforced in america |
| 06:56.10 | trixter | they just never took it off the books |
| 06:56.24 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 06:56.32 | trixter | WAY back when they could enforce it but no more |
| 06:56.33 | benjk | well, I say its about time somebody cleaned up there |
| 06:56.37 | trixter | heh |
| 06:56.50 | benjk | after all those folks are bad for business and bad for tourism especially |
| 06:56.50 | trixter | but all those dead people would vote against you! |
| 06:57.14 | trixter | to quote one mayor of san francisco "vote early vote often" |
| 06:57.21 | benjk | come on, don't pretend that there is such a thing as free elections in the US |
| 06:57.26 | trixter | there is a LOT of voter fraud there, its known but no one does anything about it |
| 06:57.51 | benjk | well, then that's where the cleaning would have to start |
| 06:58.16 | trixter | yeah good luck, there are actually designs built in to enable voter fraud in california elections |
| 06:58.19 | trixter | go figure |
| 06:58.55 | trixter | I was talking with someone from the local polling office on one of the recent holidays and they were describing all the things that the law requires be done that aids in fraud |
| 06:59.10 | trixter | and the solution is quite simple, but no one wants to change the system |
| 06:59.39 | benjk | on another note, I went to Oakland city centre and it's a ghost town |
| 06:59.39 | trixter | for example, any officer at a polling place knows who voted and who didnt, as such they know how many ballots they can cast themselves, since they arent signed its trivial to sneak a stack in the box |
| 06:59.45 | trixter | ha |
| 06:59.46 | benjk | totally dead |
| 06:59.54 | benjk | nice buildings and all that |
| 06:59.55 | trixter | were you interested in radio stuff? HRO in oakland isnt that bad |
| 06:59.56 | benjk | but dead |
| 07:00.01 | trixter | and yeah oakland has been that way for a while |
| 07:00.08 | trixter | biggest reason is not enough work |
| 07:00.13 | benjk | reminded me of Indianapolis |
| 07:00.18 | trixter | walmart went there 400 jobs available, 11,000 people applied |
| 07:00.32 | trixter | that gives you a little idea of how many people in that immediate area want work or better work |
| 07:00.56 | trixter | the financial district in SF is *totally* vacant on weekends, went there to work on a banks systems one sunday and it was weird |
| 07:01.09 | trixter | cause you see stuff like bus stops and big cross walks and stuff, but absolutly no one around |
| 07:01.18 | benjk | well, that's normal |
| 07:01.20 | trixter | not even cars on the roads, nothing is open, no one is there |
| 07:01.25 | benjk | financial centres are like that |
| 07:01.34 | trixter | not totally before that I worked in NYC and wall street always had someone nearby |
| 07:02.00 | benjk | The city of London is dead on weekends too |
| 07:02.01 | trixter | although most of the banks are a couple blocks north of the trade centers rather than south (wall street is about 1-2 blocks south depending on how you count) |
| 07:02.13 | benjk | and so it Otemachi in Tokyo |
| 07:02.18 | benjk | s/it/is |
| 07:02.24 | trixter | perhaps |
| 07:02.32 | trixter | edinburgh is dead at 3am I know that :P |
| 07:02.40 | trixter | but then the pubs were all closed so ... hehe |
| 07:02.57 | benjk | yeah but I was in Oakland at 2pm on a weekday |
| 07:03.28 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:04.20 | trixter | ahh, well its all blue collar jobs |
| 07:04.29 | trixter | like dock workers and stuff, and some parts are always like that |
| 07:04.34 | trixter | cauyse there isnt any work in some parts of oakland |
| 07:04.48 | benjk | but this was right in the city centre |
| 07:04.50 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:04.53 | trixter | so there are no supporting businesses like restaurants and other stuff, thus no reason to go there, so no work, ... |
| 07:04.58 | trixter | lather, rinse, repeat |
| 07:05.10 | trixter | ahh I havent really hung out in oakland so I dont know |
| 07:05.23 | trixter | I only went to ham radio outlet :P |
| 07:05.36 | BugKham | anyone know the cause of the "Failed to authenticate on INVITE to" error? |
| 07:05.38 | benjk | it looks like a nice place judging by the buildings and the architecture and all that |
| 07:05.47 | trixter | BugKham: wrong password? |
| 07:06.11 | trixter | benjk: any cars on the street? were they newer? was there a lot of damaged cars that werent fixed? |
| 07:06.20 | trixter | those are signs in america of a bad place to be |
| 07:06.33 | benjk | nothing of that |
| 07:06.38 | benjk | a few cabs |
| 07:06.41 | benjk | a few buses |
| 07:06.53 | benjk | didn't look trashy at all |
| 07:07.04 | BugKham | trixter: hmm, it happened when I try to call another * using Dial(SIP/user@domain.com) |
| 07:07.10 | trixter | you gotta watch though, oakland does have some problem areas |
| 07:07.33 | benjk | sure, but I went right to the city centre |
| 07:07.37 | trixter | like daly city just outside SF is a problem area (although much of the west side of SF can be problematic if you look like a target) |
| 07:08.17 | trixter | BugKham: sounds like its set up to require auth for the call and you didnt specify it. they need to do insecure=very for 1.0 or insecure=port,invite for 1.2 |
| 07:08.30 | benjk | BugKham, it looks that your counterpart doesn't accept unauthenticated incoming calls |
| 07:09.00 | BugKham | trixter: on which box? the recipient box? |
| 07:09.01 | trixter | personally the only place I was ever concerned was driving in jersey city new jersey.. it was dark and I felt the need to lock my doors... any other place even by myself I have never felt there would be a problem |
| 07:09.17 | trixter | yeah |
| 07:09.28 | trixter | thus 'they' and benjks 'counterpart' comment :) |
| 07:09.40 | trixter | that way you can connect without authentication |
| 07:09.44 | benjk | well, I can tell you I have been in some very dangerous places all over the world |
| 07:10.02 | trixter | but be careful if people can connect without authentication you gotta wa5tch contexts to avoid them from dialing anything on the box |
| 07:10.08 | chapeaurouge | hi guys.. i have a very simple sip setup here... no firewall/nat involved.. but i have no sound coming thru. sip debug shows the calls connect, however. |
| 07:10.12 | chapeaurouge | what could i be missing? |
| 07:10.16 | Qwell | chapeaurouge: upgrade |
| 07:10.19 | Qwell | 1.2.2 is b0rked |
| 07:10.19 | chapeaurouge | i did |
| 07:10.23 | chapeaurouge | 1.2.4 |
| 07:10.46 | trixter | benjk: well I have too, and done some rather stupid things.. when people were killing each other for shoes in NY I went in wearing the shoes they were killing each other for, drank way too much (new years) and ran around town all night |
| 07:10.48 | chapeaurouge | 1.2.2 worked better :P (at home anyway) |
| 07:10.50 | trixter | in some questionable areas |
| 07:10.55 | trixter | stuff like that |
| 07:11.12 | trixter | part of it is how you carry yourself, if you look like an easy mark people are more likely to try |
| 07:11.18 | benjk | I used to live in Bogota, Colombia |
| 07:11.41 | trixter | ok you win |
| 07:11.42 | BugKham | trixter: thanks man |
| 07:11.51 | trixter | BugKham: did that work? |
| 07:12.39 | BugKham | trixter: haven't tried it but from the wiki it's like I need |
| 07:12.51 | trixter | there was a guy from bogata that wanted me to build some survielance gear for him, at least walk him through the theory, but I had to part and nevergot back to him.. he described it as an interesting place |
| 07:13.26 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:13.39 | trixter | basically this guy wanted to be able to monitor computer systems without physically attaching anything to them, which the basics arent that hard, the implementation can be tricky and often requires some DSP work |
| 07:13.52 | benjk | frankly, I'd prefer roaming through the shabbiest neighbourhoods of Cairo anytime at night to walking around in SF city centre after the shops close |
| 07:14.14 | Qwell | downtown SF is fun at night |
| 07:14.27 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 07:14.37 | Qwell | we were there last week just before midnight |
| 07:14.55 | benjk | I was there just an hour ago |
| 07:14.55 | Qwell | bunch of weirdos out |
| 07:15.08 | trixter | yeah like hollywood blvd in LA |
| 07:15.16 | trixter | they normally leave you alone though |
| 07:15.29 | Qwell | we all got hit up for change probably 6 times in a row |
| 07:15.32 | benjk | saw a guy crush another guys head by smashing him against a wall in Powell St station |
| 07:16.05 | trixter | heh |
| 07:16.11 | benjk | they had an argument over a stolen laptop computer |
| 07:16.27 | trixter | that happens |
| 07:16.35 | trixter | ex dot com CEOs |
| 07:16.36 | trixter | :P |
| 07:16.55 | benjk | haha |
| 07:18.51 | *** join/#asterisk moreece (n=m@196.46.142.23) |
| 07:18.57 | chapeaurouge | would not having the same broadcast domain make sip/rtp not work? |
| 07:19.04 | chapeaurouge | i have no idea why i have no sound... |
| 07:19.09 | moreece | what is a good GSM sound recorder, I want to make my own voicemenu for our office |
| 07:19.13 | moreece | ;) |
| 07:19.19 | Qwell | moreece: asterisk |
| 07:19.21 | Qwell | app_record |
| 07:19.26 | trixter | moreece: the asterisk record application |
| 07:19.33 | trixter | chapeaurouge: um what? |
| 07:19.37 | chapeaurouge | well |
| 07:19.46 | chapeaurouge | sound is not coming thru |
| 07:19.56 | chapeaurouge | nothnig between * and my machine... it's the next hop. |
| 07:20.00 | moreece | app_record, is it built into asterisk? |
| 07:20.03 | trixter | I dont understand "broadcast domain" but it sounds like a firewall and/or nat issue |
| 07:20.13 | Qwell | moreece: yes, show application record |
| 07:20.16 | chapeaurouge | trixter, cant be.. no firewall. |
| 07:20.20 | moreece | ah thanks |
| 07:20.20 | chapeaurouge | next hop kinda thing. |
| 07:20.22 | chapeaurouge | odd.. |
| 07:20.27 | trixter | what about nat? is it localnet or what? |
| 07:20.28 | benjk | chapeaurouge: what's the SIP client? |
| 07:20.40 | chapeaurouge | linphone |
| 07:20.51 | chapeaurouge | it works at home. not here (work) |
| 07:20.52 | benjk | does it try to be smarter than its user? |
| 07:20.56 | benjk | like X-Lite? |
| 07:21.01 | Qwell | I couldn't get sound from linphone either |
| 07:21.06 | chapeaurouge | i can, at home. |
| 07:21.08 | Qwell | Are you sure your soundcard isn't locked? |
| 07:21.09 | chapeaurouge | same app. |
| 07:21.26 | moreece | ah so with the record u simply specify a channel, phone in and record the "conversation" as a .gsm file |
| 07:21.27 | Qwell | try a different softphone |
| 07:21.28 | moreece | ? |
| 07:21.35 | Qwell | moreece: pretty much |
| 07:21.41 | moreece | ah |
| 07:21.43 | moreece | thanx |
| 07:21.50 | chapeaurouge | Qwell, sound card is well. |
| 07:21.51 | moreece | now ... all I need is a mic |
| 07:21.52 | moreece | lol |
| 07:22.05 | Qwell | moreece: what, no IP phone? |
| 07:22.33 | benjk | last time I looked there were those thingies called telephones and they seemed to have mics built in |
| 07:23.38 | benjk | then again, more recently it appears that items called telephones are more and more what my generation would have called a camera, so I wouldn't bet on it |
| 07:24.10 | wasim | masked: ok |
| 07:25.35 | moreece | naah I working on the acutally linux box running Asterisk, our clients have SIP phones |
| 07:26.18 | moreece | its all working with a nice dialplan and automated menu for forwarding calls to different departments |
| 07:26.40 | moreece | buts its using lots of generic voice gsm files, we want to customise it for the company |
| 07:27.12 | moreece | have some sexy sales ladies be like "for pure business pleasure pretty 1 now" |
| 07:27.14 | moreece | lol |
| 07:28.16 | *** join/#asterisk EriSan (n=erisan@151.8.109.74) |
| 07:28.36 | argos73 | anyone know if a euro eicon bri card will work in the US? |
| 07:30.31 | wasim | argos73: it should BRI is BRI |
| 07:30.39 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net) |
| 07:30.42 | argos73 | wasim: cool - tnx |
| 07:33.03 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.230.177.telnor.net) |
| 07:37.54 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:38.59 | chapeaurouge | a tcpdump trace shows the sip packets going straight from one host to the other. yet no sound :\ |
| 07:39.39 | trixter | does anyone know if the license dispute has finally officially been resolved? the FSF seemed to say that you only had to do gpl compatible licenses for modules, digium said gpl only no gpl compatible licenses allowed, there was a promise of a public statement on this and since I have been away for a little bit I didnt know if I missed it |
| 07:40.20 | trixter | that promise iirc was 2 weeks ago with a qualifier 'soon' |
| 07:44.37 | *** join/#asterisk hmodes (i=hmodes@71.224.116.132) |
| 07:44.52 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:45.40 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:45.46 | trixter | so I take it no one knows if the promised public statement on the licensing issue about modules has been made? hrm I will google around to see if I see anything ... |
| 07:47.12 | *** join/#asterisk Frawg (n=Frawg@benc.instant.net.au) |
| 07:48.16 | benjk | I had wanted to ask Mark about this at Etel, but I didn't get a chance to speak to him |
| 07:49.01 | benjk | in any event, the FSF's statement was as clear as it gets: "LGPL is a suitable license for Asterisk modules" |
| 07:49.47 | eyz | good. |
| 07:50.07 | eyz | I hope that's the final result too |
| 07:50.14 | benjk | I guess Digium will simply keep quiet about it |
| 07:50.23 | BugKham | trixter: tried the insecure=port,invite but still can't get it working |
| 07:50.24 | *** join/#asterisk oatis (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net) |
| 07:50.34 | *** join/#asterisk Babayfax (n=Babayfax@dccom04.cafe.tg) |
| 07:50.54 | eyz | it will only help if they open up a bit I would think |
| 07:50.56 | benjk | eyz: just release a module under LGPL or BSD |
| 07:51.12 | oatis | is it possible to run multiple instances of asterisk? running differnt sip.conf / extensions.conf etc? or is it best to do it with a vps type situation? |
| 07:51.21 | eyz | xen. :) |
| 07:51.34 | eyz | if all else fails :) |
| 07:51.39 | trixter | BugKham: what version of asterisk is the remote box? |
| 07:52.00 | trixter | I have a template BSD module on my site if anyone wants it :P |
| 07:52.26 | trixter | oatis: its possible, but you have to ask yourself why |
| 07:52.27 | Babayfax | i want have information about txfax |
| 07:52.35 | benjk | and I have released a headerfile that invalidates Digium's DRM scheme |
| 07:52.36 | eyz | if you're not modifying their code, pff.. they should be happy to have the extra code contributed.. |
| 07:52.41 | trixter | odds are you can work around stuff with one instance instead of running multiple |
| 07:52.43 | oatis | eyz, what should I give as far as system resources go to a asterisk vps? |
| 07:52.48 | trixter | granted virtual hosting isnt perfect but there are work arounds |
| 07:52.59 | eyz | what type of vps? |
| 07:53.20 | oatis | eyz, I was thinking wmware running debian |
| 07:53.26 | eyz | there are a wide range of virtual server solutions out there |
| 07:53.35 | trixter | benjk: the 6th circuit court of appeals citing cases from toher circuits and the supreme court invalidated the DRM scheme (which is almost identical to lexmarks, what the case was over, yeah lexmark used SHA1 digium uses MD5 but come on) at least in america :P |
| 07:53.52 | benjk | yeah, well bad wording on my part |
| 07:53.58 | eyz | so you're going to run multiple wmware instances.. hmm.. well, better than something that can be oversold :) |
| 07:53.58 | trixter | hehe |
| 07:54.08 | trixter | I dont recommend asterisk in vmware |
| 07:54.19 | trixter | timing can drift a bunch and it can sound horrible |
| 07:54.25 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 07:54.30 | eyz | anyone tried it under Xen? |
| 07:54.32 | jebba | oatis, i'm running asterisk within a vserver ok. |
| 07:54.36 | trixter | plus vmware is quite piggy about cpu and memory |
| 07:54.44 | trixter | eyz: yes and that has worked |
| 07:54.47 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-220.claranet.co.uk) |
| 07:54.48 | trixter | although I dont know at what load |
| 07:54.50 | eyz | sweet. |
| 07:55.00 | trixter | the more layers you add the lower the capacity the system can handle |
| 07:55.06 | eyz | that's what I want to get set up on my unixshell.com account |
| 07:55.19 | eyz | xen hosting.. |
| 07:55.19 | trixter | so if you can its best to run it with the fewest layers possible to get the most bang for your buck |
| 07:55.27 | trixter | um |
| 07:55.32 | trixter | unixshell.com sucks |
| 07:55.34 | Babayfax | Can we use asterisk to fax to ordinar machine fax |
| 07:55.37 | eyz | who do you recommend then? |
| 07:55.43 | trixter | my friend has one and his network speeds vary GREATLY |
| 07:55.56 | eyz | well, that's just a test environment really. |
| 07:56.00 | oatis | jebba, ive never used vserver, got a url? |
| 07:56.07 | trixter | havent tested cpu load but he will download at 7kBps then 100 then 7 ... |
| 07:56.10 | Babayfax | Can we use asterisk to fax to ordinar machine fax? |
| 07:56.20 | trixter | Babayfax: depends |
| 07:56.31 | eyz | I have access to a co-lo at a meet-me type of place so I'm not too worried |
| 07:56.39 | jebba | oatis, http://linux-vserver.org/ practically no overhead (especially compared to things like vmware....) |
| 07:56.47 | trixter | if you try to do it voip over the internet good luck, if its voip over a managed controlled network then most likely, if its pstn and not voip then generally yes |
| 07:57.01 | Babayfax | svp can you give me more information |
| 07:57.15 | Babayfax | <PROTECTED> |
| 07:57.18 | eyz | I'm not looking for perfection, but yeah |
| 07:57.31 | benjk | you're not? shame on you |
| 07:57.34 | eyz | hehe. |
| 07:57.40 | eyz | not initially :) |
| 07:58.10 | benjk | ah, you're subscribed to the worse is better philosophy then |
| 07:58.33 | oatis | jebba, so it runs pretty smooth? what kind of system resources do you dedicate to it? |
| 07:59.04 | jebba | oatis, ya, it runs fine. But what do you mean by system resources? What kind of box? |
| 07:59.10 | *** part/#asterisk litage (n=nick@203.220.55.70) |
| 07:59.13 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 07:59.45 | eyz | if I can get a few concurrent VOIP sessions to a hosted Asterisk install with a bit of loss here and there I'd be content for now |
| 08:00.04 | oatis | jebba, oh well with vmware you tell it how much disk space, ram, etc to use for the vps |
| 08:00.31 | eyz | just not comfortable enough with the config files to get it running without AMP yet just.. reading the books online |
| 08:00.37 | jebba | oatis, vserver isn't quite like that. It's more like running in a chroot. |
| 08:00.39 | eyz | err. just yet.. fuck, tired I guess |
| 08:00.44 | *** part/#asterisk eyz (i=identd@sol.eyz.us) |
| 08:01.09 | oatis | jebba, i think i see.. im reading now.. thanks |
| 08:01.23 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:01.45 | iDunno | vserver isn't quite the same as a chroot. |
| 08:01.49 | jebba | oatis, for example, i have a fedora core 3 (blag) vserver inside a debian host. |
| 08:01.58 | iDunno | you only get your processes in ps, for a start ;) |
| 08:02.07 | jebba | iDunno, ya. It isn't the same. But it's much more similar to a chroot than it is to vmware... |
| 08:02.11 | iDunno | and it's running in a different kernel context ;) |
| 08:02.38 | iDunno | it's more similar than, say, uml or vmware or bochs, yes :) |
| 08:03.47 | trixter | eyz: I tried vmware on a crappy old box with WAY too little ram (granted those are problems in themselves) and asterisk was VERY choppy in vmware.. now the vmware player isnt bad |
| 08:04.06 | trixter | did that on a pIII 1GHz bridging a channel and it ran fine |
| 08:04.09 | trixter | 2 sip clients |
| 08:04.20 | trixter | that was the astlinux demo that I did for the sacaug.org stuff |
| 08:04.21 | *** join/#asterisk zamsler_ (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 08:05.07 | chapeaurouge | it works |
| 08:05.09 | trixter | so it really depends you can do it but you do take a performance hit to do it |
| 08:05.17 | chapeaurouge | i had to explicitly specify nat=no |
| 08:05.19 | chapeaurouge | :\ |
| 08:05.25 | chapeaurouge | i thought it was the default |
| 08:05.36 | trixter | which means less bang for your buck, but unixshell.com or whatever is $7/mo or something so its not bad as a playground if you dont have your own system |
| 08:05.53 | trixter | chapeaurouge: never trust defaults :P |
| 08:05.57 | argos73 | normal to get a bunch of "B-channel 0/1 successfully restarted on span 2" messages every so often? (fairly routine) |
| 08:06.14 | argos73 | (span 2 is a pri0 |
| 08:07.22 | *** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:10.17 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:10.22 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:10.34 | argos73 | nvr mind - found the answer |
| 08:12.52 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 08:13.56 | wasim | 3 down ... 7 to go :) |
| 08:14.02 | wasim | oops |
| 08:14.12 | wasim | 4 down ... and tendulkar gone :) |
| 08:16.04 | *** join/#asterisk wassabi (i=identd@sol.eyz.us) |
| 08:17.09 | BugKham | trixter: it's 1.2.4 and I added insecure=port,invite to the ast_config table, reloaded |
| 08:17.50 | trixter | that should be to the user entry that you are trying to connect to |
| 08:17.58 | BugKham | trixter: the remote box doesn't complain tho |
| 08:18.01 | trixter | although iirc 1.2 has a allowguest in sip.conf |
| 08:18.04 | trixter | you can try that instead |
| 08:18.23 | BugKham | trixter: k, will try again |
| 08:18.32 | trixter | its in the sip.conf.sample |
| 08:20.36 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:21.37 | chapeaurouge | is the too_many_tries something in *? or something I made myself ? :P bc i can't find any other reference to it in my dialplan |
| 08:21.51 | chapeaurouge | ah got it. |
| 08:21.52 | chapeaurouge | :P: |
| 08:25.15 | *** join/#asterisk j4m3s_ (n=j4m3s@user-24-214-119-188.knology.net) |
| 08:25.41 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 08:26.34 | cjk | hi, i have many problems that iax stops responding from time to time. sip works great. the only solution is to restart aterisk. anyone an idea?i am useing 1.2.4 |
| 08:26.57 | Babayfax | i have installed Asterisk |
| 08:27.49 | Babayfax | so i want to know more informations about "txfax" and "rxfax"* |
| 08:28.10 | Babayfax | i need help |
| 08:29.00 | Babayfax | If you want you can write meon anatolorg@yahoo.fr |
| 08:29.09 | tronix | what exactly would you like to know about it? |
| 08:29.48 | tronix | http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax |
| 08:29.51 | RoyK | Does anyone know what on earth there is on two linux boxes here that generates vast amounts of NMIs? |
| 08:29.52 | tronix | perhaps that will help? |
| 08:30.22 | tronix | real time clock? |
| 08:31.11 | RoyK | NMIs? |
| 08:31.20 | RoyK | nothing should use NMI |
| 08:31.23 | RoyK | imho |
| 08:32.12 | Babayfax | i want to know if we can do fax to ordinary machine fax via asterisk |
| 08:33.16 | wassabi | like a bridge between offices? |
| 08:35.24 | Babayfax | svp i want to know if we can do fax to ordinary machine fax via asterisk |
| 08:35.24 | BugKham | trixter: thanks, I really appreciated your call |
| 08:35.34 | trixter | np |
| 08:35.41 | trixter | fix your echo problem :P |
| 08:35.46 | trixter | is that a soft phone you answered with? |
| 08:35.52 | trixter | maybe it was your mic picking up your speakers |
| 08:36.05 | BugKham | trixter: yeah, it's the Xten eyebeam |
| 08:37.30 | BugKham | trixter: could be, I am using a headset with my laptop |
| 08:37.47 | *** part/#asterisk BugKham (n=lamer@202.8.86.170) |
| 08:37.53 | *** join/#asterisk BugKham (n=lamer@202.8.86.170) |
| 08:38.13 | trixter | ahh laptop sound cards are often very cheap and bleed over |
| 08:38.14 | trixter | not all but many |
| 08:38.15 | Babayfax | i want to know if we can do fax to ordinary machine fax via asterisk |
| 08:38.26 | trixter | and the headset mic may pick up your ear peice if the volume is way up |
| 08:38.36 | trixter | Babayfax: croll up I answered that earlier |
| 08:39.01 | trixter | trixter if you try to do it voip over the internet good luck, if its voip over a managed controlled network then most likely, if its pstn and not voip then generally yes |
| 08:39.12 | chapeaurouge | what's the name of the web interface for voicemail? |
| 08:39.20 | trixter | firefox? |
| 08:39.22 | trixter | :P |
| 08:39.24 | chapeaurouge | lol |
| 08:39.39 | benjk | lynx |
| 08:39.43 | chapeaurouge | come on... |
| 08:39.46 | wassabi | hehe |
| 08:39.52 | benjk | what? |
| 08:39.53 | wassabi | links :) |
| 08:39.59 | benjk | don't like lynx? |
| 08:40.04 | benjk | lynx rocks |
| 08:40.04 | BugKham | trixter: yeap, you are absolutely right. I do not have thi echo problem when using a separated mic |
| 08:40.05 | chapeaurouge | i like elinks better :P |
| 08:40.18 | chapeaurouge | so, there is a web interface for voicemail stuff right? |
| 08:40.26 | chapeaurouge | the one you can open in lynx |
| 08:40.27 | chapeaurouge | :P |
| 08:40.44 | *** join/#asterisk tzafrir (n=tzafrir@82.166.242.248) |
| 08:40.49 | chapeaurouge | i think i read it somewhere |
| 08:41.36 | wassabi | referring to vmail.cgi perhaps? |
| 08:41.40 | knight_ | ARI |
| 08:41.40 | chapeaurouge | maybe... |
| 08:41.46 | knight_ | Asterisk Recordings Interface |
| 08:41.53 | chapeaurouge | no |
| 08:41.56 | chapeaurouge | not an spi |
| 08:41.58 | chapeaurouge | api* |
| 08:42.05 | tzafrir | hi all. This is my shiny new amd64. Finally have a desktop at work again, and thus "tzafrir" will lurk here and elsewhere... |
| 08:42.07 | knight_ | Let's you login to a mailbox, listen to voicemail, and also monitor's. |
| 08:42.09 | benjk | ari == Japanese for ant |
| 08:42.10 | knight_ | It's not an API. |
| 08:42.12 | chapeaurouge | yea.. vmail.cgi i think it is |
| 08:42.25 | knight_ | vmail.cgi is old and nasty. |
| 08:42.29 | chapeaurouge | ah |
| 08:42.42 | knight_ | http://www.littlejohnconsulting.com/ari |
| 08:42.47 | *** join/#asterisk [chico] (n=chico@p54916591.dip.t-dialin.net) |
| 08:42.50 | knight_ | Enjoy... |
| 08:43.51 | chapeaurouge | thanks for the link. |
| 08:43.54 | chapeaurouge | i will try it |
| 08:44.13 | benjk | the Japanese word for "thank you" (arigatoh) can be misrepresented as "there are 10 ants" |
| 08:44.24 | chapeaurouge | ... harakiri... |
| 08:44.54 | chapeaurouge | knight_, stable or dev? |
| 08:45.04 | knight_ | pretty stable |
| 08:45.10 | chapeaurouge | dev is pretty stable? |
| 08:45.11 | knight_ | as far as my experience with it has been anyway |
| 08:45.13 | chapeaurouge | ok |
| 08:45.21 | chapeaurouge | will give that a shot. thanks for the input |
| 08:45.24 | knight_ | np |
| 08:45.45 | knight_ | tzafrir, you are tzanger? |
| 08:45.58 | tzafrir | Is there any asterisk webmail interface that uses some well-defined interface to check passwords? |
| 08:46.14 | knight_ | not that i know of |
| 08:46.35 | tzafrir | knight_, no, I'm not. I'm only here to be confused for tzanger in case of auto-completion |
| 08:46.57 | knight_ | i wrote a webmail interface called phpvoipmail (and subsequently a gui app that connects into it using XML), but it uses/used the voicemail config files |
| 08:47.02 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 08:47.16 | wassabi | and that java app to connect to it? |
| 08:47.16 | knight_ | tzafrir, I think we've been over that a million times and I keep forgetting |
| 08:47.31 | knight_ | wassabi: http://www.kevinelliott.net/asterisk/AVC/ |
| 08:47.49 | wassabi | hehe.. cool, I was just looking at that page a few minutes back |
| 08:48.07 | knight_ | I'm still trying to hunt down the most recent phpvoipmail sources, since asteriskdocs.org lost everything |
| 08:48.08 | wassabi | I was about to mention it earlier |
| 08:48.26 | knight_ | So I've been digging in my backups for like 2 months |
| 08:48.38 | wassabi | ack.. :) |
| 08:48.41 | knight_ | ;) |
| 08:49.03 | wassabi | ack, dead link? http://projects.asteriskdocs.org/modules/xfmod/project/?phpvoipmail |
| 08:49.07 | knight_ | yep |
| 08:49.10 | wassabi | k |
| 08:49.11 | knight_ | that's what i was saying |
| 08:49.18 | wassabi | I see |
| 08:49.25 | knight_ | asteriskdocs.org had a crash or something, and lost all the sources |
| 08:49.47 | chapeaurouge | knight_, so i need to configure * with mysql ? |
| 08:50.08 | knight_ | chapeurouge, I'm pretty sure that Asterisk Realtime is required. |
| 08:50.21 | chapeaurouge | hmmk |
| 08:50.23 | wassabi | is there anything "out of the box" besides call info logging associated with MySQL? |
| 08:50.34 | RoyK | knight_: realtime required for what? |
| 08:50.43 | knight_ | RoyK, ARI |
| 08:50.51 | RoyK | ~ari |
| 08:50.57 | RoyK | wtf is ari? |
| 08:51.06 | knight_ | http://www.littlejohnconsulting.com/ari |
| 08:51.07 | chapeaurouge | http://www.littlejohnconsulting.com/ari |
| 08:51.25 | wassabi | like for instance, configuration running live out of MySQL or something? |
| 08:51.34 | wassabi | like Postfix, Courier, Apache, etc.. |
| 08:52.16 | knight_ | something like that |
| 08:54.18 | benjk | there cannot be any out of the box anything with MySQL unless you get a commercial license for MySQL |
| 08:54.28 | wassabi | oh, true |
| 08:54.52 | benjk | why even bother with MySQL |
| 08:55.03 | chapeaurouge | ~amp |
| 08:55.05 | jbot | amp is probably NOT supported here! people using it should join #amportal |
| 08:55.24 | chapeaurouge | does ARI need AMP??? |
| 08:55.26 | chapeaurouge | wtf. |
| 08:55.33 | knight_ | no |
| 08:55.38 | knight_ | ARI doesnt need it. |
| 08:55.41 | chapeaurouge | hmm |
| 08:55.42 | chapeaurouge | ok |
| 08:55.46 | knight_ | Although it is integrated with it |
| 08:56.23 | knight_ | I just wish I could find the latest copy of the phpvoipmail code so I could update it and release it again |
| 08:56.42 | knight_ | AVC is a useful app to run on the desktop |
| 08:56.54 | wassabi | yeah, looked nice |
| 08:57.18 | knight_ | I even started writing skinning code so people could customize the look of it |
| 08:57.25 | knight_ | But I stopped once the phpvoipmail code got lost |
| 09:00.19 | wassabi | benjk, I agree that MySQL licensing is troublesome.. do you recommend PostgreSQL or something like that for those with 0 budget? |
| 09:00.50 | benjk | Postgres would be my choice yes, but at the very least you should use ODBC |
| 09:01.00 | benjk | just to be on the safe side |
| 09:01.29 | wassabi | what is the benefit of using ODBC for safety? |
| 09:01.40 | benjk | license wise I mean |
| 09:01.57 | trixter | and if you really want to be safe have asterisk write everythying to a flat text file, then have another process pick up that text file, send it across a tcp socket to another box, then a 3rd process on that box pick it up and via odbc stick it in the DB |
| 09:02.03 | trixter | but odds are that level of safety isnt required :P |
| 09:02.04 | wassabi | as a layer in-between regarding licensing? |
| 09:02.21 | trixter | yeah odbc removes the 'dynamic linking' argument for the most part |
| 09:02.27 | trixter | so gpl stuff isnt an issue |
| 09:02.36 | chapeaurouge | hmm... mysql licensing changed? (sorry i wasn't aware) |
| 09:02.37 | benjk | if you put ODBC between your application layer and MySQL, then you isolate your code sufficiently so as to not be MySQL specific and consequently they cannot claim you forced anyone to download and install MySQL |
| 09:02.51 | wassabi | I saw this on the MySQL site, just in case anyone is wondering "Free use for those who never copy, modify or distribute. As long as you never distribute the MySQL Software in any way, you are free to use it for powering your application, irrespective of whether your application is under GPL license or not." |
| 09:02.56 | trixter | mysql states on their page that if you use any commercial app you must buy a commercial license but they dont mean it |
| 09:03.02 | trixter | not with the gpl being what it is anyway |
| 09:03.20 | benjk | the trouble is though that MySQL are very litigious |
| 09:03.20 | wassabi | I take it that installation of MySQL is not copying, but that's pretty vague. |
| 09:03.30 | trixter | but they are very unclear on their page about that issue, there is one sentence that makes it seem like if a unrelated program is bundled on the same CD but never uses mysql you have to get a mysql license |
| 09:03.33 | trixter | which is um yeah |
| 09:03.39 | wassabi | is PostgreSQL stable enough for production use? |
| 09:03.54 | benjk | and they may simply cause you lots of nightmares by haveing their lawyers come after you, whether their claims are actually supported by the GPL or not |
| 09:03.55 | wassabi | it's supposed to be totally free to use, from what I've seen anyway |
| 09:04.15 | wassabi | I see.. how are web hosting providers getting away with providing it? |
| 09:04.19 | benjk | this alone is reason enough to avoid them |
| 09:04.24 | benjk | in my book at least |
| 09:04.26 | wassabi | yeah |
| 09:04.43 | wassabi | trixter, I've seen that line.. forget where it was. |
| 09:04.59 | trixter | I will get it |
| 09:05.02 | trixter | I had to hunt it last time |
| 09:05.17 | wassabi | regarding the client library and something along the lines of "basically, if your software is written to only use MySQL, you need to pay us" |
| 09:05.17 | [av]bani | you can use mysql freely, its when you sell it bundled in a commercial package that you have issues |
| 09:05.18 | benjk | companies who play with the sword of damocles hanging over their customers' heads should be boycotted |
| 09:05.22 | wassabi | is that what you're referring to? |
| 09:05.38 | [av]bani | the whining about mysql amuses me greatly |
| 09:05.39 | wassabi | what if the bundle is a service instead of something that is downloaded? |
| 09:05.43 | wassabi | yeah.. |
| 09:05.47 | *** join/#asterisk matteo (n=matteo@81.208.84.216) |
| 09:06.08 | [av]bani | services are clear, mysql has made that very crystal clear many times (but the detractors avoid mentioning it) |
| 09:06.14 | benjk | see the fact that you have to worry about this nonsense at all is reason enough to avoid them |
| 09:06.23 | wassabi | benjk, haha |
| 09:06.26 | [av]bani | services on top of mysql is fine, its distribution of the software bundled as product to end users that the problem exists |
| 09:06.29 | benjk | use a product that is clear as clear can be |
| 09:06.32 | wassabi | [av]bani, see, that's what I thought too |
| 09:06.36 | wassabi | ok |
| 09:06.46 | wassabi | I'd like to see that spelled out somewhere though |
| 09:06.47 | [av]bani | benjk: linux sucks, use openbsd instead |
| 09:06.49 | benjk | Postgres doesn't give you any such issues to worry about, none whatsoever |
| 09:07.01 | wassabi | what if you're selling an application as a service? |
| 09:07.03 | benjk | that |
| 09:07.15 | benjk | 's a different story though |
| 09:07.17 | wassabi | I imagine that would be clear too, but its kinda weird there. |
| 09:07.19 | [av]bani | database elitism is as funny as the 12 year olds who argued their atari was better than their amiga |
| 09:07.26 | benjk | we're talking about licensing issues and claims |
| 09:07.44 | benjk | not about how good or bad we perceive a given product to be |
| 09:07.57 | trixter | If you distribute a proprietary application in any way, and you are not licensing and distributing your source code under GPL, you need to purchase a commercial license of MySQL - http://www.mysql.com/company/legal/licensing/index.html |
| 09:07.59 | benjk | Linux is very clear on the license side |
| 09:08.00 | [av]bani | its just fashionable and leet to bash mysql, its trendy, like openbsd militants bash linux |
| 09:08.12 | wassabi | MySQL somehow got itself in a lot of opensource software, and so its tricky now. |
| 09:08.25 | benjk | so whatever you think about Linux in technical terms, the licensing is not a source of worries |
| 09:08.40 | [av]bani | benjk: but linux SUCKS!@#!@$*& use OPENBSD because its better)*!(# |
| 09:08.40 | wassabi | hmm.. time to look up the word "distribute" |
| 09:08.42 | wassabi | heheh |
| 09:08.45 | wassabi | hahaha |
| 09:08.57 | trixter | So if you use MySQL with GPL-licensed software (or a license that is GPL-compatible) we encourage you to use the GPL license. For all other users of MySQL, we recommend that you purchase a MySQL commercial license |
| 09:08.57 | [av]bani | wassabi: no worries about mysql |
| 09:08.59 | trixter | http://www.mysql.com/company/legal/licensing/faq.html |
| 09:09.02 | benjk | avbani: that's a different discussion |
| 09:09.06 | trixter | they dont make a odbc exemption which they should |
| 09:09.17 | [av]bani | benjk: and openbsd has a better license, linux gpl is too restrictive |
| 09:09.24 | [av]bani | if you want TRUE FREEDOM DONT USE LINUX |
| 09:09.27 | [av]bani | use openbsd! |
| 09:09.39 | [av]bani | or freebsd |
| 09:09.40 | [av]bani | or netbsd |
| 09:09.42 | benjk | sure there will be many situations where BSD has preferable licensing terms |
| 09:09.45 | wassabi | or dragonfly, or.. |
| 09:09.51 | [av]bani | bsd is always preferable to GPL |
| 09:09.51 | benjk | still, Linux licensing is clear |
| 09:09.53 | [av]bani | GPL restricts you |
| 09:10.01 | benjk | not like that FUD from MySQL |
| 09:10.02 | wassabi | GPL seems to favor services. |
| 09:10.20 | wassabi | "because if you're using it for your own company.." that line |
| 09:10.27 | wassabi | its a mess. |
| 09:10.32 | [av]bani | wassabi: unless you're selling mysql, you dont have anything to worry about |
| 09:10.39 | benjk | as long as you know what the license permits you to do and what it doesn't it is ok |
| 09:10.45 | trixter | netbsd is still the 4 clause license |
| 09:10.48 | [av]bani | benjk: no! gpl is restrictive! |
| 09:10.51 | trixter | so its not the same as open or free bsd licenses |
| 09:10.53 | wassabi | ok.. its not like they even need to know its MySQL or anything else on the backend anyway, with a service |
| 09:11.04 | benjk | but MySQL have created a zone of uncertainty with claims that are unlikely to be supported by the GPL |
| 09:11.18 | benjk | and that is a total knock-out criteria |
| 09:11.21 | [av]bani | wassabi: or you could use sqlite |
| 09:11.29 | wassabi | true, that. |
| 09:11.31 | [av]bani | benjk: talk about fud |
| 09:11.38 | wassabi | I hear sqlite is very clean. |
| 09:11.43 | benjk | for that FUD MySQL should never be used under any circumstances |
| 09:11.53 | benjk | unless you want to pay for a commercial license that is |
| 09:12.02 | trixter | I think t hat you can get something in writing over that though |
| 09:12.07 | trixter | if you really want mysql in that regard |
| 09:12.07 | [av]bani | wassabi: ignore benjk fud, use sqlite instead of postgresql |
| 09:12.08 | benjk | but it should never be considered for GPL use |
| 09:12.29 | [av]bani | sqlite is better |
| 09:12.36 | trixter | the reality is that they know they cant enforce that because the gpl as stated in section 0 DOES NOT COVER RUNTIME |
| 09:12.37 | trixter | :P |
| 09:12.43 | trixter | but it might cost $2M to fight it |
| 09:12.44 | benjk | shoot the messenger is your thing |
| 09:12.52 | benjk | I don't make the fud, MySQL does |
| 09:12.54 | [av]bani | benjk: you sure are whiny :D |
| 09:12.57 | wassabi | "does not cover runtime?" explain please? |
| 09:13.11 | benjk | no, I am a practical person |
| 09:13.23 | [av]bani | you're a cheerleader |
| 09:13.31 | benjk | software has legal question marks attached => don't use it |
| 09:13.38 | benjk | very practical approach |
| 09:13.45 | *** join/#asterisk Babayfax (n=Babayfax@dccom04.cafe.tg) |
| 09:13.47 | trixter | wassabi: read the gpl in there it clearly states that there are NO LIMITATIONS ON RUNNING A GPL PROGRAM and that RUNNING A GPL PROGRAM IS OUTSIDE THE SCOPE OF THE GPL |
| 09:13.50 | [av]bani | linux has legal question marks attached (sco, patents) => don't use it |
| 09:13.57 | trixter | which invalidates the FSF dynamic linking argument but meh |
| 09:13.58 | [av]bani | use free/open/netbsd instead of linux |
| 09:14.10 | trixter | that is a different conversation both of those things are very clear in th4e gpl itself |
| 09:14.37 | wassabi | its like we need to guess what the spirit of it is and run with that.. too bad its a legal document now |
| 09:14.46 | *** join/#asterisk Tribastian (n=tribasti@62-2-138-202.business.cablecom.ch) |
| 09:14.49 | [av]bani | benjk: the whole reason behind the 'signing off' of git now is that there are legal question marks surrounding linux. |
| 09:14.55 | benjk | that was true in the beginning of the SCO/IBM case |
| 09:14.56 | tzafrir | [av]bani, HUH? do you actually claim SCO's case has any merit? |
| 09:15.05 | wassabi | oh geez.. SCO. |
| 09:15.09 | wassabi | topic over!!! :) |
| 09:15.11 | wassabi | j/k |
| 09:15.11 | [av]bani | tzafrir: if there wasn't, linus wouldnt have people signing off on code |
| 09:15.12 | benjk | but the signals the judge has given are clearly not in favour of SCO |
| 09:15.23 | trixter | hey at least its not lexmark with the sha1 of some code to ensure its not unlicensed |
| 09:15.30 | [av]bani | the fact that linus implemented that indicates he takes teh threat seriously, or at least saw some merit in it |
| 09:15.31 | trixter | lexmark and their stupid toner cartridges |
| 09:15.39 | [av]bani | otherwise... why bother? |
| 09:15.47 | benjk | sure, but by now this case is irrelevant |
| 09:15.54 | tzafrir | [av]bani, I assume that you bought some of their stocks, right |
| 09:15.56 | benjk | SCO are going to lose it |
| 09:15.59 | [av]bani | hardly, it's only irrelevant when dismissed |
| 09:15.59 | *** join/#asterisk Modcuts (n=sam@ppwood.gotadsl.co.uk) |
| 09:16.05 | Tribastian | hello, erverybody! small question: is it possible to insert into the [generl] section of sip.conf, two different ports and how do the client react to that? |
| 09:16.18 | [av]bani | tzafrir: no, but that's a great red herring |
| 09:16.25 | benjk | SCO will lose it, the judge has already picked the winner |
| 09:16.34 | benjk | and it aint SCO |
| 09:16.39 | [av]bani | if thats the case, why is it going to trial in feb 2007? |
| 09:16.50 | benjk | all the actions of the judge are aimed at helping IBM to win the case |
| 09:16.59 | [av]bani | the judge doesnt pick, the jury does... |
| 09:17.05 | benjk | dreamer |
| 09:17.23 | benjk | go read up on it at groklaw |
| 09:17.27 | tzafrir | [av]bani, the fact that they were not thrown off the court completely does not mean that their case actually has any merits |
| 09:17.35 | [av]bani | not that i expect sco to win, but you have a very bizarre idea of how the system works |
| 09:17.43 | trixter | the judge does give instructions to the jury |
| 09:17.43 | benjk | read the professional commentary by professional lawyers |
| 09:17.48 | [av]bani | tzafrir: it means it has enough merit to not be thrown out :) |
| 09:17.52 | trixter | and the judge can overturn a jurys decision |
| 09:17.57 | benjk | no it doesn't mean that at all |
| 09:18.04 | benjk | it means the exact opposite |
| 09:18.04 | trixter | however in a civil case in america it only takes 2/3 majority not all |
| 09:18.22 | benjk | it means the judge wants SCO to crash down very hard |
| 09:18.27 | [av]bani | if sco can snow the jury... well, anything is possible |
| 09:18.33 | benjk | rather than allowing them a soft landing |
| 09:18.36 | [av]bani | i mean look at oj simpson |
| 09:18.43 | trixter | sco has to prove some things to a jury |
| 09:18.50 | [av]bani | its what lawyers do, lie |
| 09:18.52 | trixter | and a tech case is largely not an easy thing to prove |
| 09:19.02 | trixter | juries are really STUPID |
| 09:19.04 | trixter | trust me on this |
| 09:19.04 | [av]bani | its not a tech case.. its a contractual breach case |
| 09:19.14 | wassabi | those new IBM Linux commercials with the kid are starting to make sense to me now from a political standpoint |
| 09:19.19 | trixter | it gets muddled with tech though |
| 09:19.21 | tzafrir | [av]bani, sweet dreams. Anyway, someone asked here a Q about an off-topic issue such as sip.conf |
| 09:19.27 | tzafrir | anybody? |
| 09:19.28 | wassabi | heheh |
| 09:19.31 | wassabi | sockets. |
| 09:19.44 | Tribastian | sorry to disturb you.... |
| 09:19.46 | [av]bani | wassabi: the commercials are.. bizarre |
| 09:20.05 | wassabi | indeed. |
| 09:20.47 | wassabi | well, that was fun :) I'll try not to stir everyone up with political talk again :) |
| 09:20.51 | [av]bani | benjk: a judge _cant_ want either side to win, a judge is supposed to be unbiased. if either side believes a judge is biased, they object |
| 09:21.11 | [av]bani | benjk: a judge is supposed to judge, not be prejudiced |
| 09:21.19 | benjk | I prefer BSD over Linux myself, however, from a licensing point of view Linux cannot be compared to the situation with MySQL |
| 09:21.22 | Tribastian | sorry i have to ask that now, this is the asterisk-server, or did i land somewhere else? :-) |
| 09:21.30 | wassabi | <Tribastian> hello, erverybody! small question: is it possible to insert into the [generl] section of sip.conf, two different ports and how do the client react to that? |
| 09:21.34 | [av]bani | (idealism i know, but if sco thought the judge was biased against them, they would have objected by now) |
| 09:21.36 | benjk | dream on |
| 09:21.46 | tzafrir | Tribastian, you are in the right channel. |
| 09:21.57 | wassabi | Tribastian, my fault.. I riled everyone up. |
| 09:22.11 | benjk | as I said before, go to groklaw and read up before you make nonsense statements here about things you so obviously know absolutely nothing about |
| 09:22.14 | Tribastian | well i forgive you... :-) |
| 09:22.25 | wassabi | first time in an open source channel.. ;) |
| 09:22.39 | Tribastian | me? |
| 09:22.43 | wassabi | no, me. |
| 09:22.52 | benjk | that was for avbani |
| 09:24.35 | chapeaurouge | Feb 1 10:22:52 NOTICE[744]: manager.c:574 authenticate: 127.0.0.1 tried to authenticate with nonexistent user '' |
| 09:24.35 | chapeaurouge | <PROTECTED> |
| 09:24.37 | chapeaurouge | :\ |
| 09:24.54 | wassabi | nonexistent user NULL? |
| 09:25.05 | chapeaurouge | well |
| 09:25.11 | chapeaurouge | i dont see when i could pass a user |
| 09:25.22 | wassabi | weird |
| 09:25.24 | trixter | chapeaurouge: portscan? |
| 09:25.35 | chapeaurouge | on localhost? |
| 09:25.39 | wassabi | would it pop up that message from the socket being opened? |
| 09:25.54 | wassabi | without anything else happening? |
| 09:26.01 | chapeaurouge | im trying to use that ARI stuff... this is what appears in the * console |
| 09:26.25 | chapeaurouge | ah |
| 09:26.27 | chapeaurouge | hold |
| 09:26.50 | chapeaurouge | ok... another error, but better |
| 09:29.03 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 09:29.03 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 09:30.10 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 09:30.32 | chapeaurouge | ARI seems to be missing the DB.php file |
| 09:30.35 | chapeaurouge | i dont have it |
| 09:31.31 | wassabi | google had no hits on this "Unable to open pseudo channel for timing... Sou |
| 09:31.31 | wassabi | nd may be choppy. |
| 09:31.34 | wassabi | ack.. sorry |
| 09:31.40 | *** join/#asterisk Speeder (n=psilva@est-213-228-152-121.netvisao.pt) |
| 09:31.43 | Tribastian | hello, erverybody! small question: is it possible to insert into the [generl] section of sip.conf, two different ports and how do the client react to that? |
| 09:32.18 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 09:32.22 | jozsab1 | hy all |
| 09:32.30 | Tribastian | by |
| 09:32.59 | jozsab1 | Can anybody tell me what does this mean (got it from CLI) : |
| 09:33.00 | jozsab1 | m=audio 16410 RTP/AVP 0 100 101 |
| 09:33.00 | jozsab1 | a=rtpmap:0 PCMU/8000 |
| 09:33.00 | jozsab1 | a=rtpmap:100 NSE/8000 |
| 09:33.00 | jozsab1 | a=rtpmap:101 telephone-event/8000 |
| 09:33.00 | jozsab1 | a=fmtp:101 0-15 |
| 09:33.23 | jozsab1 | Where can i set these, get info about them ? |
| 09:36.11 | *** join/#asterisk svenna_ (n=svenna@p548D3B7A.dip0.t-ipconnect.de) |
| 09:36.25 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
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| 09:38.08 | Krill | anyone has experience working with the avaya units with asterisk? |
| 09:38.28 | *** join/#asterisk mike240se (i=mike240s@ool-45730608.dyn.optonline.net) |
| 09:40.11 | mike240se | hey, I have a verizon line connected to my zap card, i want to only use the zap card for outbound calls, so i call forwarded the verizon line to my voip line, well every time a call is forwarded, verizon rings the line once which makes asterisk try to answer it and it ofcourse causes problems, i cant find anywhere how to make zaptel ignore the incoming ring... I just want it to ignore anything incoming |
| 09:40.45 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:40.56 | areski | good morning |
| 09:43.15 | benjk | OK, I found the article at groklaw now ... |
| 09:43.22 | benjk | go to www.groklaw.net |
| 09:43.36 | wassabi | k |
| 09:43.45 | benjk | click on search and search for the exact phrase "SCO, you are not going to win your case in my courtroom. I've chosen the winner and it isn't you" |
| 09:44.28 | benjk | the commentary of this article should give you an insight how the court system works behind the curtains |
| 09:44.56 | wassabi | oh, that wasn't my conversation.. whoops.. |
| 09:46.05 | wassabi | that's someone's interpretation according to the article |
| 09:46.06 | wassabi | btw |
| 09:46.35 | wassabi | interesting though |
| 09:46.50 | benjk | of course it is, but the folks who comment on that side are people with a clue in respect of legal issues |
| 09:47.07 | benjk | s/side/site |
| 09:47.18 | trixter | not a guarantee |
| 09:47.22 | trixter | and even lawyers are idiots |
| 09:47.26 | trixter | most of the ones I have met are |
| 09:47.33 | trixter | only the ones I keep in contact with arent :P |
| 09:47.41 | trixter | like jennifer granick who is an amazing lawyer |
| 09:47.50 | trixter | and cute |
| 09:48.18 | benjk | there is no chance SCO will win this case |
| 09:48.26 | trixter | there is a slim chance |
| 09:48.36 | benjk | they have failed to deliver a shred of evidence |
| 09:48.48 | trixter | according to some theories of parallel universes they have to win in some alternate universe :) |
| 09:48.56 | [av]bani | of copyright infringement yes, of contractual breach, that is still up in the air |
| 09:48.58 | trixter | because everything that can happen wil happen all at the same time |
| 09:48.59 | wassabi | mohah |
| 09:49.07 | benjk | the judge noted that in his findings |
| 09:49.09 | [av]bani | its funny benjk has a bug up his butt still about it :) |
| 09:49.26 | wassabi | had to find the quote.. can't blame him |
| 09:49.57 | benjk | there is one universal and timeless golden rule of law |
| 09:50.05 | benjk | never ever piss off the judge |
| 09:50.28 | benjk | for example by failing to deliver any evidence |
| 09:50.59 | benjk | if you make a court feel you wasted their valuable time, you'll get punished |
| 09:51.16 | wassabi | btw, I think its phpMyAdmin that is keeping MySQL afloat. :) it is for me, anyway |
| 09:51.33 | wassabi | looking into alternatives :) |
| 09:51.48 | benjk | how much of an effort to adapt that to Postgres? |
| 09:52.09 | wassabi | I've seen one for Postgres, it sucked. |
| 09:52.35 | [av]bani | O RLY |
| 09:52.46 | benjk | ok, then I guess the question would be how much of an effort to fix it |
| 09:52.57 | wassabi | yeah |
| 09:53.50 | [av]bani | what, no whinge about the evils of php? |
| 09:53.55 | [av]bani | i'm disappointed. |
| 09:54.29 | benjk | what would that be? |
| 09:54.41 | wassabi | heh.. they want to sell their accelerators and caching stuff |
| 09:54.48 | benjk | so? |
| 09:54.59 | wassabi | I guess that's all :) |
| 09:55.19 | wassabi | different thing entirely I suppose |
| 09:55.39 | benjk | as long as they don't make dubious claims and threaten to send their lawyers after you it's just a matter of choices |
| 09:56.11 | benjk | again, I don't particularly like PHP, but using it doesn't put you in limbo |
| 09:56.27 | [av]bani | php isn't leet, its insecure, and has apache license (which atm isn't clear if it is gpl compatible or not) |
| 09:56.41 | trixter | its not if its apache 1 license |
| 09:56.48 | trixter | apache 2 is compatible becuase its like the bsd 3 clause |
| 09:56.52 | benjk | according to the FSF's website it is very clear |
| 09:57.01 | trixter | and how exactly is php insecure specifically? |
| 09:57.34 | [av]bani | trixter: you've never programmed php? |
| 09:57.45 | [av]bani | oterhwise you wouldnt ask that :) |
| 09:57.59 | trixter | I want to know how the language is insecure |
| 09:58.07 | trixter | rather than poor coders who write insecure code |
| 09:58.09 | wassabi | its easy to get sloppy passing this here and that there |
| 09:58.09 | benjk | yeah, well, PHP is like the modern equivalent of 1960s BASIC |
| 09:58.25 | trixter | ahh so its poor coders that are insecure |
| 09:58.34 | *** join/#asterisk silly (n=silly@cpe-24-174-162-34.satx.res.rr.com) |
| 09:58.38 | litage | where can i find out information about how to deal with CDRs using the Manager interface/API? |
| 09:58.39 | [av]bani | no, php had a lot of engine-level exploits |
| 09:58.40 | trixter | if that is your argument http://www.trxtel.com/crashterisk.c proves that C is insecure and shouldnt be used |
| 09:58.41 | wassabi | I'd agree with that. |
| 09:58.52 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 09:58.55 | trixter | had or has? |
| 09:58.57 | benjk | that's almost like splitting hairs, because it is always the coders who product insecure code |
| 09:59.00 | [av]bani | a lot of sloppy php applications sure, but a lot of language internal exploits |
| 09:59.00 | wassabi | [av]bani, do you happen to know where I could find a list of those? |
| 09:59.04 | trixter | and are those default configuration options loke globals? |
| 09:59.07 | [av]bani | trixter: had, very recent too |
| 09:59.10 | [av]bani | nope |
| 09:59.24 | [av]bani | not default, engine level like multibyte exploits, function exploits |
| 09:59.32 | [av]bani | causing internal errors |
| 09:59.46 | [av]bani | you havent been paying attention :) |
| 09:59.46 | wassabi | python? |
| 09:59.47 | trixter | ok, so that is seperate from the language |
| 09:59.50 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:59.50 | dudes | I've never heard of perfectly secure code |
| 10:00.00 | trixter | so the language itself doesnt have the problems you described |
| 10:00.03 | trixter | I just wanted to be clear on this |
| 10:00.18 | wassabi | its the interpreter? |
| 10:00.21 | benjk | it'd be the runtime system |
| 10:00.22 | [av]bani | no, it does. it's had a long history of language exploits -- both internal and functional |
| 10:00.32 | wassabi | hmm |
| 10:00.36 | [av]bani | early php had no way to turn register globals off for example |
| 10:00.48 | wassabi | well, sucks to be someone coding PHP with MySQL I guess |
| 10:00.48 | [av]bani | register_globals switch was added to fix that language design flaw |
| 10:00.49 | trixter | again is it the language or the poor coding ability of those that wrote some add on like pear stuff |
| 10:01.04 | trixter | ok, so early php had something that doesnt mean its currently that way |
| 10:01.06 | [av]bani | trixter: poor design of php, which had no way to turn it off |
| 10:01.13 | *** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net) |
| 10:01.13 | benjk | I guess what trixter is trying to say is that insecurity is probably not inherent to the language itself but to its implementation |
| 10:01.27 | trixter | you still havent given me a current language example, instead you are mixing stuff together as one big ball and I want to identify specifically what you are saying |
| 10:01.32 | benjk | in other words you could reimplement the language and avoid the problems |
| 10:01.39 | wassabi | there must be ways of patching most of the security holes |
| 10:01.45 | [av]bani | benjk: it is inherent to the language, and php is making big architectural changes to fix the design flaws in the next release |
| 10:01.46 | trixter | exactly |
| 10:01.51 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:01.55 | dudes | There will always be exploits regardless of language or implementor |
| 10:01.56 | wassabi | sweet |
| 10:02.00 | [av]bani | safe_mode continues to be a huge problem |
| 10:02.01 | trixter | well there are a lot of design flaws in a lot of things |
| 10:02.08 | benjk | I have a bit of a problem accepting this statement |
| 10:02.10 | trixter | and sometimes it takes a total redesign to fix the problems |
| 10:02.15 | [av]bani | its a fundamental design flaw in current php |
| 10:02.16 | JamesDotCom | LIKE ASTERISK |
| 10:02.17 | JamesDotCom | huhuhu |
| 10:02.18 | trixter | those that refuse to take this step will be overrun by others that do |
| 10:02.24 | trixter | I DIDNT SAY THAT |
| 10:02.25 | trixter | :P |
| 10:02.28 | trixter | I only implied it |
| 10:02.32 | JamesDotCom | haha |
| 10:02.43 | benjk | I think it would be very difficult to prove that a language is inherently insecure and cannot be implemented in a secure fashion |
| 10:02.44 | wassabi | hacksterisk |
| 10:02.47 | [av]bani | php also makes it easy to make mistakes :( writing secure c code is hard, writing secure php is nigh impossible :) |
| 10:03.13 | trixter | I agree that php lets you do stuff that is inherently bad.. $$blah for example |
| 10:03.15 | benjk | language as in grammar of the language in question |
| 10:03.17 | [av]bani | but that tends to be the case with HLL's anyway, php just did a worse job thaan most |
| 10:03.26 | [av]bani | perl had the same problem for at least a decade |
| 10:03.39 | wassabi | python? |
| 10:03.41 | dudes | What is to say Perl still doesn't? |
| 10:03.52 | wassabi | I wonder when PERL 6 will come out. |
| 10:03.53 | [av]bani | perl is better at preventing you from making dumb mistakes though, with tainting strings |
| 10:03.56 | benjk | and yes, of course some language encourage writing sloppy code |
| 10:04.14 | [av]bani | php doesnt have that... |
| 10:04.30 | [av]bani | so its easy to get things past php scripts in unchecked input paths |
| 10:04.39 | [av]bani | maybe next php will fix that... |
| 10:04.39 | wassabi | you have to code up with coding practices to remind yourself when a variable is potentially tainted |
| 10:04.39 | trixter | C allows you to easily smash the stack, why in 1997 I wrote a paper on smashing the stack (although 2 others done the same year are better, aleph ones smashing the stack for fun and profit and the l0phts all 3 were released about the same time) |
| 10:04.44 | wassabi | err.. come up with. |
| 10:04.46 | benjk | but is that specified in the grammar? |
| 10:04.50 | dudes | Well looking at C++, for example, one can almost write C and it'll compile. But that, IMO, is a design flaw not writing true C++ and instead C. |
| 10:05.04 | trixter | so there are a lot of things that exist, but I dont blame the language for that |
| 10:05.05 | benjk | I bet you could reimplement the same language differently and avoid this particular flaw |
| 10:05.07 | [av]bani | benjk: in perl it's sorta in the grammar. it's functional not syntactic |
| 10:05.14 | trixter | I blame the developers who dont know enough to write good code |
| 10:05.34 | wassabi | javascript! |
| 10:05.41 | dudes | Most developers don't understand security enought to look at it clearly |
| 10:06.00 | trixter | I have seen people write php that will do stuff like enumerate the $_POST array and do $$key=value without checking to see if anything is getting overwritten, yes php allows that and yes that is generally bad to do, but there are a lot of things bad to do |
| 10:06.05 | [av]bani | a proper language shouldnt let you make dumb mistakes (ADA :) |
| 10:06.19 | chapeaurouge | ~amp |
| 10:06.21 | jbot | methinks amp is NOT supported here! people using it should join #amportal |
| 10:06.22 | benjk | but consider Murphy's law |
| 10:06.25 | [av]bani | trixter: worse, it makes it so easy and almost desirable to do it that way |
| 10:06.36 | benjk | if you make something absolutely foolproof only a fool will want to use it |
| 10:06.37 | trixter | yeah but I see that as a choice |
| 10:06.50 | [av]bani | trixter: it should make it hard to make bad choices, like perl does |
| 10:06.51 | trixter | I would rather have a choice than to be so restricted in what I can do that I cnat get what I need done |
| 10:06.56 | trixter | nah |
| 10:07.05 | trixter | I believe in freedom why I dont support the gpl |
| 10:07.10 | [av]bani | perl lets you shoot yourself in the foot, but it will whinge before letting you do it blindly |
| 10:07.12 | wassabi | personal use mode versus production mode |
| 10:07.12 | dudes | [av]bani - that's like saying there should be only 1 right way todo one task when in reality there is 100 different ways to do the same thing. |
| 10:07.27 | [av]bani | dudes: perl doesnt _prevent_ you, it just tells you that it's dangerous |
| 10:07.37 | [av]bani | php doesnt even do that, it happily eats it all blindly |
| 10:07.53 | dudes | I haven't used Perl since 98', and even then I didn't much care for it. |
| 10:07.54 | [av]bani | so you get jillions of php exploits, and relatively few perl ones |
| 10:08.01 | benjk | well, in that case, what would stop somebody from adding such a warning mechanism to PHP? |
| 10:08.13 | [av]bani | benjk: we've been asking the php devs that for 5 years now |
| 10:08.13 | trixter | so you are complaining that php requires knowledge and skill to use? citing that as reasons its bad? |
| 10:08.15 | wassabi | storing variables differently, adding tainting |
| 10:08.21 | wassabi | new classes, etc etc |
| 10:08.23 | trixter | I dunno I think that doesnt make a language bad or insecure per se |
| 10:08.49 | wassabi | it comes back to a matter of personal comfort with the uncertainty |
| 10:08.51 | dudes | [av]bani - Perhaps draft a interreptur that does just that? |
| 10:08.53 | wassabi | same with MySQL and all |
| 10:08.56 | [av]bani | benjk: they havent taken security seriously till very recently. php has taken a real bad rap because of the legions of problems |
| 10:08.57 | benjk | time to fork it and add it yourself I guess |
| 10:09.00 | trixter | but its late I need sleep so I will be back tomorrow or something |
| 10:09.03 | wassabi | :) |
| 10:09.08 | [av]bani | dudes: the next php will solve a lot of the problems (hopefully) |
| 10:09.08 | dudes | err, interpreter |
| 10:09.11 | chapeaurouge | <PROTECTED> |
| 10:09.22 | dudes | YES |
| 10:09.23 | trixter | depends on what and how you modify stuff |
| 10:09.27 | chapeaurouge | =] |
| 10:09.30 | JamesDotCom | [av]bani: i've heard that many times before |
| 10:09.31 | trixter | I have done it many times and never had a problem |
| 10:09.33 | dudes | AMP will overwrite it so fast you're f'n head will spin |
| 10:09.34 | wassabi | if you don't intend to use AMP anymore it should be fine ;) |
| 10:09.41 | [av]bani | JamesDotCom: so have i, which is why i say "we'll see" |
| 10:09.42 | chapeaurouge | lol |
| 10:09.43 | chapeaurouge | ok |
| 10:09.44 | chapeaurouge | thx |
| 10:09.52 | trixter | and of course amp didnt overwrite my hand modifications |
| 10:10.01 | trixter | but I guess that is the difference in how you modify stuff |
| 10:10.07 | [av]bani | python is cute, but applciations written in it tend to be very fragile |
| 10:10.13 | wassabi | yeah? |
| 10:10.20 | wassabi | hmm |
| 10:10.31 | wassabi | java.? |
| 10:10.38 | trixter | amp generally doesnt modify sip.conf it modifies sip_additional.conf for example |
| 10:10.39 | dudes | I pretty much don't like AMP, and having to work with people that use it ... oh, hate it |
| 10:10.40 | benjk | python is very nice but it appears to be growing out of proportion |
| 10:10.42 | [av]bani | write once, run nowhere |
| 10:10.45 | [av]bani | = java |
| 10:10.51 | trixter | it will overwrite the additional file so changes in sip.conf can be untouched |
| 10:11.04 | [av]bani | good luck writing code which is compatible with all the demented JVMs out ther |
| 10:11.14 | trixter | I want to see a web based UI that looks like benjks stuff for astmasters.net |
| 10:11.26 | trixter | that would resolve a ton of questions about how to configure stuff |
| 10:11.33 | [av]bani | python also tends to be slow... but its preferable to perl in most cases |
| 10:11.40 | wassabi | anyone know of a bare-bones example of what I need to connect via IAX with a soft phone? and perhaps just echo back? |
| 10:11.43 | dudes | I've never liked anything that does something for me, ie, webmin, AMP, ect. I think, doing the configs myself make things work better in the end. |
| 10:11.47 | [av]bani | php makes things so easy though that i use it for quick shell script level hacks |
| 10:12.07 | wassabi | m0n0wall uses it for most everything it seems |
| 10:12.27 | trixter | astlinux comes with a phpconfig style |
| 10:12.40 | trixter | its the same thing as editing the config files just using a webbrowser instead of a text editor |
| 10:12.42 | [av]bani | its possible to write secure php, but you have to be very very careful |
| 10:12.59 | benjk | yeah, always use a condom |
| 10:13.12 | [av]bani | starting out with good habits helps, but sadly almost all php tutorials and guides tell you the wrong way |
| 10:13.13 | wassabi | takes proper discipline |
| 10:13.14 | trixter | ok sleep for real |
| 10:13.23 | dudes | bag the old wiesel |
| 10:13.30 | [av]bani | 'this is easy, do it this way' -- they dont tell you its wildly insecure too |
| 10:13.40 | benjk | dudes: how come you haven |
| 10:13.50 | benjk | t mentioned any booze yet today? |
| 10:13.56 | dudes | haven? |
| 10:14.04 | benjk | haven't |
| 10:14.10 | dudes | haven't mentioned booze today? |
| 10:14.14 | benjk | yeah |
| 10:14.19 | JamesDotCom | 17:56 < [av]bani> php isn't leet, its insecure, and has apache license (which atm isn't clear if it is gpl compatible or not) |
| 10:14.22 | JamesDotCom | 18:02 < [av]bani> php also makes it easy to make mistakes :( writing secure c code is hard, writing secure php is nigh impossible :) |
| 10:14.25 | dudes | Should I have mentioned booze today? |
| 10:14.25 | JamesDotCom | 18:12 < [av]bani> its possible to write secure php, but you have to be very very careful |
| 10:14.28 | JamesDotCom | hahah |
| 10:14.35 | *** join/#asterisk litage_ (n=nick@203.220.55.70) |
| 10:14.37 | benjk | eveytime I met you here you said you were running out of booze |
| 10:14.50 | dudes | I still got half a liter, heh |
| 10:14.58 | benjk | ah that's why :) |
| 10:15.15 | dudes | I've only had a two drinks tonight |
| 10:15.23 | dudes | I've cut back a lot the last month. |
| 10:15.31 | benjk | well, you're almost dry then |
| 10:15.59 | dudes | I normally get hammered with some friends but they pass out before me. |
| 10:16.25 | benjk | I bet they do |
| 10:16.40 | dudes | So I've been having a few drink before bed which seems to aid sleeping better. |
| 10:16.52 | wassabi | all night, or just the first half? |
| 10:17.05 | benjk | I brought half a suitcase full of sake from Japan |
| 10:17.09 | *** join/#asterisk fulgas (n=fulgas@209.8.233.252) |
| 10:17.21 | Damin | Ahhh.. Booze.. |
| 10:17.31 | Damin | I've forsworn the pleasures of the evil drink... |
| 10:17.32 | benjk | distributed it equally amongst friends |
| 10:18.26 | dudes | I normally get a half gallon and no one wants my whisky |
| 10:18.46 | [av]bani | http://phpsec.org/projects/vulnerabilities/securityfocus.html |
| 10:18.54 | dudes | I don't see how anyone would like Vodka more than whisky |
| 10:19.10 | dudes | Or even Bacardi 151 |
| 10:19.55 | benjk | I guess if you're Russian though it would be the exact opposite |
| 10:20.32 | dudes | then water and pure alcohol would make sence |
| 10:20.42 | wassabi | [av]bani, point taken |
| 10:20.43 | benjk | haha |
| 10:20.57 | niZon | can anyone point me to some example configs for the new musiconhold |
| 10:20.59 | niZon | ? |
| 10:21.07 | niZon | I'm stuck with this right now: |
| 10:21.08 | niZon | -- Started music on hold, class 'default', on channel 'SIP/300-20b8' |
| 10:21.08 | niZon | -- Stopped music on hold on SIP/300-20b8 |
| 10:21.17 | dudes | there is a musiconhold.conf.sample included with * |
| 10:21.33 | niZon | it doesn't like me much |
| 10:21.38 | chapeaurouge | well, fuck.. i cant make that ari stuff working. |
| 10:22.13 | *** join/#asterisk urmelZaus (n=urmel@u16-13.dsl.vianetworks.de) |
| 10:24.08 | [av]bani | heh, debian has problems with the php license |
| 10:24.24 | urmelZaus | hello, I have a problem with DID and chan_capi. Is this the right place for questions? |
| 10:26.07 | dudes | benjk - what do you do? |
| 10:27.06 | benjk | right now? |
| 10:27.18 | dudes | I mean, what do you do? |
| 10:27.37 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 10:27.47 | benjk | I guess you did have a lot of booze before because I told you already :) |
| 10:28.16 | dudes | I'm sure I have some blanks from my nights in here, heh |
| 10:28.21 | benjk | hehe |
| 10:28.58 | chapeaurouge | what's the MOH status? what should we use on 1.2.4 ? |
| 10:29.08 | [av]bani | wassabi: you are reading your pm's :) |
| 10:29.12 | benjk | I do consulting in Japan which is where I live and I also look after Asterisk on OSX/Darwin |
| 10:29.26 | cypromis | chapeaurouge: try ALAW moh files with native music on hold |
| 10:29.28 | cypromis | quite nice |
| 10:29.35 | dudes | Japan would be a nice place to visit |
| 10:29.40 | chapeaurouge | ok thanks cypromis ;) |
| 10:33.47 | benjk | yeah lots of sake and whiskey |
| 10:33.54 | benjk | ;) |
| 10:34.16 | dudes | nah, I don't much care to get hammered anymore |
| 10:34.35 | dudes | I'd be more looking at the tail |
| 10:34.48 | benjk | that sounds like you made serious progress |
| 10:34.58 | dudes | After new years |
| 10:35.21 | benjk | cause I was worried there a bit |
| 10:35.50 | dudes | I got so drunk that I don't remember anything. Let alone why I had blood all over my bathroom. |
| 10:36.07 | benjk | that's scary |
| 10:36.34 | dudes | I don't much care to hear about picking fights with people and not remembering |
| 10:36.42 | benjk | well, you weren't always very gentlemen like in here too |
| 10:37.03 | dudes | I'm not very gentelement like sober. |
| 10:37.18 | Damin | oej: It's going to be cool.. |
| 10:37.20 | benjk | but you did apologise for any typos |
| 10:37.24 | benjk | :) |
| 10:37.43 | dudes | Which I did make a lot of |
| 10:38.16 | dudes | I actually want to be a English teacher when I get older. |
| 10:38.23 | niZon | ok this is strange |
| 10:38.25 | benjk | considering your alcohol to blood ratio that wasn't all to suprising |
| 10:38.37 | niZon | I update asterisk and now I can't dial out to my IAX providers :\ |
| 10:39.00 | dudes | niZon - What did you upgrade too? |
| 10:39.19 | benjk | English teaching is the sort of thing that people do to fund an extended stay in Japan |
| 10:39.42 | niZon | 1.2.4 |
| 10:39.56 | dudes | So I could teach English in Japan w/o a college education? |
| 10:40.27 | dudes | that'd be sweet.. |
| 10:40.29 | benjk | I think they prefer folks who have done this TOEFL thing or whatever it is called |
| 10:40.38 | wassabi | ESL? |
| 10:41.06 | wassabi | well, I connected to my server and go out to Digium |
| 10:41.08 | wassabi | so that's good |
| 10:41.26 | benjk | its not that well paid, but as I said, many folks do it to fund an extended stay in Japan |
| 10:41.42 | dudes | I wouldn't be concerned with the money, heh |
| 10:42.00 | benjk | well, you'll have to pay for food and shelter |
| 10:42.18 | benjk | so a little income obviously helps |
| 10:42.41 | dudes | That's a given |
| 10:42.52 | dudes | What's the internet like in Japan? |
| 10:43.19 | wassabi | its fast like bullet trains |
| 10:43.26 | benjk | all ADSL or FTTH, some CATV |
| 10:44.12 | dudes | I'm jamming to Barry Manilow - Mandy |
| 10:44.19 | dudes | Such a good jam |
| 10:44.20 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 10:44.30 | benjk | the bullet trains are smart, not so much because of the speed, but the way they are run |
| 10:44.50 | trixter | benjk: pm |
| 10:44.53 | dudes | run/ran |
| 10:45.01 | benjk | Neither the French nor the Germans run their high speed trains as smart as the Japanese |
| 10:45.40 | dudes | anyway, I'm going to bed. |
| 10:45.57 | benjk | they have two tracks in each direction |
| 10:46.15 | benjk | it works like a highway |
| 10:46.31 | benjk | one track is for the trains that stop at each station |
| 10:46.46 | benjk | the other for the express trains that stop only every 200-500 kms |
| 10:47.53 | benjk | all the seats are numbered and the trains will stop exactly in position so that people waiting on the platform know where the door for their car will come to stop |
| 10:48.18 | Skumling | what's the best practice for dialing multiple VoIP providers with priority? the goal is just to have backup-providers to route the call through in case of breakdown on the primary one. will it be fine to just have multiple Dial() commands under eachother? |
| 10:49.14 | benjk | if you go from a place where the express train doesn't stop, you'll get off at the next express station, stay there for five minutes and board the same car, get into the same seat on the express train |
| 10:49.38 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 10:50.29 | benjk | this way, you'll typically maintain an average speed of about 200 kph from any point to any other point even though the trains that stop every 50kms or so won't maintain that average |
| 10:50.35 | benjk | very clever |
| 10:51.54 | *** join/#asterisk grey (n=grey@host54-106.bol.co.tz) |
| 10:51.59 | grey | hi all |
| 10:53.03 | grey | i keep getting login timeout when trying loging into asterisk with xlite |
| 10:53.08 | grey | can anyone help ! |
| 10:54.10 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
| 10:54.52 | Gordo | Can anyne tell me how to configure extensions.conf to play hold music to an call whilst trying to connect it to the first availabe extenstion in a group...? |
| 10:55.53 | *** join/#asterisk redax (n=redax@r6.hu) |
| 10:55.56 | redax | hi, |
| 10:56.25 | Gordo | <PROTECTED> |
| 10:56.49 | grey | hello ? |
| 10:57.43 | redax | does normal ISDN phones displays CallerName if I do reverse lookup for the callerid via reverse.agi ? |
| 10:58.20 | {zombie} | Gordo: look at the m flag to Dial |
| 10:58.50 | {zombie} | redax: if your phone supports calleridname then sure |
| 10:59.15 | grey | can anyone help me with a timeout problem when xlite tries to login |
| 11:00.13 | Gordo | zombie: Where would I set the m flag...? At teh end of a Dial string next to the t? |
| 11:00.51 | redax | {zombie}: if I call the ISDN extension from a SIP extension (xten lite) it displays my sip name |
| 11:01.02 | redax | like "redax_sip" |
| 11:01.38 | redax | where redax_sip is my sip login name |
| 11:02.07 | {zombie} | Gordo: that's right. now the next question is, do you really need t? |
| 11:02.21 | {zombie} | many people seem to add it to their dialplans because it's in all the examples but most people don't need it |
| 11:02.53 | Gordo | I have only included the t as I have seen it in examples... I am guilty of including an option that I have no idea what is does... :( |
| 11:03.03 | Gordo | exactely |
| 11:03.18 | Gordo | could you enlighten me as to its purpose..? |
| 11:03.21 | jozsab1 | anybody knows where can i download a working version of g729 codec for asterisk ? (also free version) |
| 11:03.51 | {zombie} | Gordo: it allows the extension that answered to transfer the call by using the # key |
| 11:04.01 | {zombie} | for phones that don't have a transfer or hold button |
| 11:04.04 | Gordo | ahh, really |
| 11:04.17 | Gordo | ok, so the then the m option does what... |
| 11:04.24 | *** join/#asterisk _grey_ (n=grey@host54-106.bol.co.tz) |
| 11:04.31 | {zombie} | Gordo: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
| 11:04.32 | {zombie} | have a read :) |
| 11:04.40 | {zombie} | there's loooots of funky things you can do with Dial |
| 11:04.47 | Gordo | will do.... thanks mate |
| 11:05.06 | _grey_ | hi all |
| 11:05.27 | benjk | jozsab1: g729 is patent encumbered => there can be no free version until the patent runs out |
| 11:05.32 | _grey_ | can someone help me with getting xlite to work with asterisk, i keep getting login timeout |
| 11:06.40 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 11:07.53 | {zombie} | _grey_: you're going to have to give more details about your network, rather than repeating the same question over and ove |
| 11:08.43 | Modcuts | good morning. |
| 11:08.46 | *** join/#asterisk TallAndy (i=TallAndy@83.104.196.72) |
| 11:09.50 | *** part/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
| 11:12.35 | _grey_ | zombie : sorry! The connection the xlite is comming from is bad average 2000ms delay and 20% packet loss |
| 11:13.29 | _grey_ | what is the channel for AMP help ? |
| 11:13.49 | wassabi | #amportal ? |
| 11:13.49 | wasim | #electricity |
| 11:14.00 | wassabi | ack. |
| 11:14.01 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
| 11:14.08 | eyz | damn name collision :) |
| 11:15.32 | _grey_ | thanks wassabi |
| 11:15.43 | eyz | np |
| 11:16.24 | _grey_ | don't want to repeat myself but can anyone help with my timeout problem |
| 11:18.03 | eyz | what's up with 484 address incomplete in xlite? |
| 11:19.38 | af_ | how could I print someway the vars in an extensions? |
| 11:23.48 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 11:23.48 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 11:23.49 | wasim | af_: NoOp() |
| 11:24.11 | urmelZaus | I have PtP with chan_capi. The DID exten are 6 (main) or 7 digits. If I dial in with single digits (no redial), asterisk only receives the first 6 digits, the 7th will be lost. How can I resolve this? |
| 11:31.43 | *** join/#asterisk cpm (n=Chip@border1.avitecture.net) |
| 11:34.24 | CaViCcHi | Sorry do you have a nice and serious ael.vim ? |
| 11:34.30 | _grey_ | can someone help me with getting xlite to work with asterisk, i keep getting login timeout |
| 11:34.38 | _grey_ | The connection the xlite is comming from is bad average 2000ms delay and 20% packet loss |
| 11:36.43 | *** join/#asterisk tzafrir (n=tzafrir@85-64-21-208.barak-online.net) |
| 11:37.43 | wasim | the fat lady sings! |
| 11:38.59 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 11:44.09 | Damin | Sounds like your problem is a crappy network, not Asterisk.. |
| 11:45.11 | eyz | anyone on FWD? |
| 11:48.07 | benjk | _grey_ I have seen such environments and worse in Africa and the Middle East |
| 11:48.29 | benjk | in my experience SIP doesn't work under those conditions |
| 11:48.47 | *** join/#asterisk Mod-cuts (n=sam@ppwood.gotadsl.co.uk) |
| 11:49.18 | benjk | IAX works well enough although not all of the time |
| 11:49.29 | *** join/#asterisk gmanev (n=gmanev@213.91.216.51) |
| 11:49.56 | benjk | depending on how desperate you are and how much or how little control you have over the network, you may want to try your luck with IAX |
| 11:50.15 | eyz | I'm currently connecting through a crappy DSL connection with occasional packet loss to asterisk on a shared server with probably about 64 virtual servers running on it, and I'm getting through to Free World Dialup test numbers without much loss of voice |
| 11:50.35 | benjk | the first choice is of course to get that network connection fixed if that is a possibility |
| 11:50.39 | eyz | IAX between Asterisk and FWD |
| 11:52.18 | *** join/#asterisk pengyong (n=lala@218.93.155.56) |
| 11:52.21 | _grey_ | benjk: is IAX possible between softphone and asterisk ? |
| 11:52.33 | eyz | yes |
| 11:52.34 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 11:55.06 | _grey_ | really stupid question, but what exatly is FWD ? |
| 11:55.15 | eyz | Free World Dialup |
| 11:55.17 | eyz | one sec |
| 11:55.47 | eyz | they can connect you out to other VOIP networks |
| 11:56.01 | eyz | and you can register your Asterisk server there for incoming calls via that network as well |
| 11:56.20 | eyz | its like a free way to connect your server or softphone to other existing VOIP networks |
| 11:56.28 | eyz | and test, I guess |
| 11:57.21 | _grey_ | and where can i find details on how to setup xlite to use aix ? |
| 11:57.48 | eyz | it only supports sip as far as I know |
| 11:58.02 | eyz | there are other soft phones that support IAX |
| 11:58.21 | eyz | unfortunately I just found out that Firefly isn't supporting 3rd party IAX, only their own proprietary network |
| 11:58.29 | eyz | and that's what I was about to recommend. |
| 11:58.31 | eyz | there are others though |
| 11:58.50 | benjk | X-Lite doesn't speak IAX |
| 11:58.58 | benjk | you have to use an IAX softphone |
| 11:59.06 | benjk | there are a bunch of those now |
| 11:59.14 | benjk | most recent one is Idefisk |
| 11:59.20 | benjk | which is multi-platform |
| 11:59.42 | eyz | have you tried it? |
| 12:00.03 | eyz | the GUI looks pretty decent |
| 12:00.03 | benjk | yeah, but I got to try it before it was released, so it wasnt' usable yet |
| 12:00.14 | benjk | but it made a good overall impression nevertheless |
| 12:00.25 | eyz | nice.. looks like they're in beta now |
| 12:00.59 | eyz | newest version released 1/4/06 |
| 12:01.11 | benjk | is it April already? |
| 12:01.21 | cpm | I sure hope not. |
| 12:01.28 | eyz | err.. I'm in the US |
| 12:01.29 | benjk | so do I |
| 12:01.40 | benjk | no excuse |
| 12:01.42 | benjk | :) |
| 12:01.43 | eyz | January 4th, 2006 |
| 12:01.57 | eyz | April Fools! |
| 12:01.58 | eyz | j/k |
| 12:02.18 | eyz | well, it looks like the license is roughly, " |
| 12:02.19 | eyz | If you find a bug in this program, please file a bugreport on support@asteriskguru.com |
| 12:02.31 | eyz | and the important 2nd part, " All other feedback or suggestions are also appreciated." |
| 12:02.42 | benjk | beerware license |
| 12:02.55 | eyz | that's how I take it |
| 12:03.07 | benjk | if you like this software and you meet the author by chance one day in a pub, you may buy him a beer |
| 12:04.09 | eyz | oop.. sorry.. most recent is January 20, 2006 |
| 12:04.17 | eyz | his page must not have been updated or something, or I'm blind.. either |
| 12:04.35 | benjk | yeah that's what I was going to say, there was another release last week or so |
| 12:04.46 | _grey_ | thanks, you have been very helpful am off to do some research on softphones :) |
| 12:04.54 | eyz | :) |
| 12:06.08 | cpm | eyz, yeah, it's neat looking software, but no license, no joy. But I'd still buy the fellow a beer |
| 12:06.10 | benjk | remember, the best thing to do is fix your network connection - if you can |
| 12:06.47 | benjk | you'd have to buy five beers then |
| 12:06.48 | eyz | so no license means its not free to use? |
| 12:07.12 | benjk | its zoa the project manager and four programmers |
| 12:07.30 | _grey_ | actually I am in the process of setting up a company to offer VoIP for cheap international calls, so I am exploring "limits" |
| 12:07.40 | benjk | I guess they were just too busy to get their beta out for testing and bug fixing |
| 12:07.49 | benjk | they'll probably think of a license by the time they do their proper release |
| 12:08.29 | benjk | well, you won't get good quality out of 2 seconds latency, that;s for sure |
| 12:08.42 | eyz | over. |
| 12:08.45 | eyz | mohah |
| 12:09.09 | benjk | most people will rather pay a few cents per minute extra to avoid having to put up with that kind of a lag |
| 12:10.11 | benjk | if it's only sporadically and you get well under 1 second on average then it should be bearable |
| 12:10.15 | Damin | oej: Yep.. |
| 12:10.23 | eyz | what would the "type" be for a phone like that? |
| 12:10.25 | eyz | trying to set it up |
| 12:10.53 | *** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
| 12:10.58 | benjk | usually you use type=friend for phones |
| 12:11.26 | eyz | hmm.. failing registration |
| 12:11.26 | _grey_ | am i correct in assuming that the best way for a soft/hard phone to talk to an asterisk server is iax ? |
| 12:11.37 | benjk | depends |
| 12:12.24 | benjk | if you want to use hardphones and you want high quality phones, then there ins't much choice, you probably have to use SIP |
| 12:12.58 | benjk | there are IAX hardphones but they are few and at the lowest end of the spectrum |
| 12:14.27 | eyz | hmm.. getting "peer (foo) is not dynamic" |
| 12:14.30 | _grey_ | and with regard to softphones is best quality achieved with iax ? |
| 12:15.48 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 12:15.48 | eyz | w00t.. got in |
| 12:15.48 | eyz | type=friend |
| 12:15.48 | eyz | host=dynamic |
| 12:16.24 | cypromis | you could use MGCP |
| 12:16.25 | cypromis | :) |
| 12:16.33 | eyz | MGCP? |
| 12:16.38 | benjk | iax has a number of advantages |
| 12:17.11 | benjk | mgcp is for telephony in a cemetery |
| 12:18.39 | Tribastian | hello, erverybody! small question: is it possible to insert into the [generl] section of sip.conf, two different ports and how do the client react to that? |
| 12:19.34 | eyz | I still don't think anyone quite understands what you're trying to do |
| 12:19.39 | eyz | its been what? like 4 hours? |
| 12:19.51 | eyz | hmm |
| 12:20.09 | benjk | 4 hours is the keyword |
| 12:20.24 | eyz | yeah.. just about sun-up over here |
| 12:20.38 | benjk | past 4 am, time to get some sleep |
| 12:20.45 | eyz | here too |
| 12:21.16 | I-MOD | you guys in cali? |
| 12:21.21 | eyz | I'm in AZ |
| 12:21.30 | benjk | SF |
| 12:22.21 | benjk | anyway, good night |
| 12:22.48 | eyz | nite |
| 12:23.13 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:24.58 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 12:26.20 | RoyK | wtf does canreinvite default to yes? |
| 12:30.02 | *** join/#asterisk coppice (n=chatzill@168.166.17.210.dyn.pacific.net.hk) |
| 12:39.29 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:41.14 | skeffling | We have a TE410P, and have a problem on incoming audio. What we here is the normal speech, then every so often, a short beep, silence for a few seconds, a short beep, and then back to the speech. The other party does not hear any thing unusual. We think it may be to do with DTMF detection in zaptel, can this be disabled? |
| 12:41.43 | skeffling | the silence can be 'heard' in the Monitor recordings too |
| 12:43.29 | I-MOD | skeffling: when you load the zaptel kernel module, you can specify zpmdtmfsupport=0 or dtmfsupport=0 |
| 12:43.46 | I-MOD | *vpmdtmfsupport |
| 12:44.06 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 12:44.08 | skeffling | thanks I-MOD I'l try that |
| 12:46.44 | *** join/#asterisk kletter-1atze (n=kletter-@dslb-084-056-214-010.pools.arcor-ip.net) |
| 12:47.26 | *** join/#asterisk _grey_ (n=grey@host54-106.bol.co.tz) |
| 12:48.06 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 12:48.47 | *** join/#asterisk Marz (n=marzjema@mar-san.demon.nl) |
| 12:48.49 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-214-010.pools.arcor-ip.net) |
| 12:48.53 | Marz | hi |
| 12:48.59 | Marz | would like to try asterisk PBX |
| 12:49.08 | Marz | what type of hardware should I make use of? |
| 12:49.30 | Marz | would like to connect for testing to one ISDN2 phoneline |
| 12:50.31 | Marz | hello? |
| 12:51.18 | *** join/#asterisk rigid (n=The@port-212-202-73-202.dynamic.qsc.de) |
| 12:51.46 | rigid | what ways are there in asterisk/sip to transmit the CID? |
| 12:52.05 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 12:53.10 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 12:53.27 | rigid | I transfer (e.g.) 0123-45678 as my real number (which works)... Is it possible to append a digit to the number so the called person sees (e.g) 0123-456789... where do i have to make the change?? |
| 12:53.30 | rigid | -= |
| 12:53.32 | rigid | -? |
| 12:54.09 | markit | hi :) trying to register in FWD with iax, but seems I'm not registered, and when calling I get " Auto-congesting call due to slow response", any clue? (I subscribed long time ago, and yes, I've turned on the iax check in my FWD profile) |
| 12:54.31 | dpryo | rigid: something like Set(CALLERID(number))=${CALLERID(number)}9) perhaps? |
| 12:55.43 | rigid | dpryo, do you know what i have to search for to get documentation about that? |
| 12:55.57 | dpryo | rigid: Callerid, i guess :) |
| 12:56.59 | rigid | dpryo, hmm... didn't find anything useful on voip.org... i'll have another look |
| 12:57.02 | rigid | dpryo, tnx |
| 12:57.16 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 12:57.46 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 12:57.47 | asteriskmonkey | morning |
| 12:58.07 | Marz | would like to try Asterisk PBX, but where to start? |
| 12:58.10 | Marz | do I need hardware? |
| 12:58.16 | cypromis | a pc would help |
| 12:58.17 | asteriskmonkey | Marz: no |
| 12:58.32 | asteriskmonkey | if you dont know anything grab a pc and asterisk@home |
| 12:58.35 | dpryo | Marz: You need a linux-computer. |
| 12:58.54 | cpm | Marz, what asteriskmonkey said, go here: http://asteriskathome.sourceforge.net/ |
| 12:59.16 | Marz | yes I know that, but with only linux and the software installed I can make internal phone calls? |
| 12:59.20 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 12:59.24 | asteriskmonkey | hey does anyone know how i can limit the time of call? |
| 12:59.26 | Marz | through software of hardware IP phones? |
| 12:59.41 | asteriskmonkey | Marz: yes you can make internal calls |
| 13:00.01 | asteriskmonkey | Marz: providing you have sip phones or ata's or softphones |
| 13:00.17 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:01.07 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:04.48 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:10.42 | *** join/#asterisk kio (n=kio@195-11.customer.cloud9.net) |
| 13:14.48 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 13:16.16 | *** join/#asterisk coppice (n=chatzill@103.199.17.210.dyn.pacific.net.hk) |
| 13:20.49 | jozsab1 | what does this mean : "SIP responses, class 4: Request failures : 488 Not Acceptable Here " ? |
| 13:21.22 | *** join/#asterisk cuco (n=diego@85-64-16-56.barak-online.net) |
| 13:22.23 | _grey_ | i get "Error inserting zaptel (/lib/modules/2.6.13-1.1532_FC4/misc/zaptel.ko): Invalid module format" when trying to install ztdummy |
| 13:22.43 | _grey_ | and "zaptel: version magic '2.6.13-1.1532_FC4 586 REGPARM 4KSTACKS gcc-4.0' should be '2.6.13-1.1532_FC4 686 REGPARM 4KSTACKS gcc-4.0'" in messages log |
| 13:23.37 | _grey_ | does this mean the kernel is compiled as 686 or the module ! :) ? |
| 13:25.25 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:26.46 | *** part/#asterisk urmelZaus (n=urmel@u16-13.dsl.vianetworks.de) |
| 13:27.16 | *** join/#asterisk razu_ (n=razu@ip59.cab62.mus.starman.ee) |
| 13:27.17 | *** join/#asterisk Flyboy-SR22 (i=rsears@gateway.adnc.com) |
| 13:28.27 | *** join/#asterisk robbie2 (n=rob@CPE-144-137-188-224.qld.bigpond.net.au) |
| 13:28.34 | robbie2 | anyone awake ? |
| 13:30.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-134-171.buff.east.verizon.net) |
| 13:30.14 | _grey_ | i get "Error inserting zaptel (/lib/modules/2.6.13-1.1532_FC4/misc/zaptel.ko): Invalid module format" when trying to install ztdummy |
| 13:30.15 | _grey_ | and "zaptel: version magic '2.6.13-1.1532_FC4 586 REGPARM 4KSTACKS gcc-4.0' should be '2.6.13-1.1532_FC4 686 REGPARM 4KSTACKS gcc-4.0'" in messages log |
| 13:30.15 | _grey_ | does this mean the kernel is compiled as 686 or the module ! :) ? |
| 13:30.17 | Flyboy-SR22 | good morning |
| 13:30.48 | SuPrSluG | hello |
| 13:30.57 | *** join/#asterisk QbY (n=Kelvin@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 13:31.05 | jozsab1 | godd morning 2 u. it's 3 o'clock here :) |
| 13:31.33 | *** join/#asterisk Skarmeth (n=Skarmeth@201009024115.user.veloxzone.com.br) |
| 13:31.38 | *** join/#asterisk bon (i=bon@localhost.sk) |
| 13:31.44 | Skarmeth | hi all |
| 13:32.08 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
| 13:32.11 | QbY | does anyone know why the default music on hold won't load up, but other music on hold that i have defined will?? |
| 13:32.11 | [TK]D-Fender | jozsab1 : Thats a codec negociation failure. What are you trying to use? |
| 13:32.11 | Flyboy-SR22 | ah..5:30 here |
| 13:32.12 | Flyboy-SR22 | :-) |
| 13:33.10 | Skarmeth | I was searching for docs about the cost vs benefict for using IP Phones vs Analog Phones on a Asterisk solution... someone know a guide or any other help docs that show the cost difference? |
| 13:33.46 | Skarmeth | I'll have a E1 link (10 active channels for voice) and about 30 extensions |
| 13:34.19 | robbie2 | anyone here successfulyl configured a TE110P |
| 13:34.38 | robbie2 | im not sure if i have been given the right card |
| 13:34.43 | SuPrSluG | I keep getting echo on my sipura 2100 ata. Doesn't happen with the Polycom. I am pots free also. Anyone else have this problem? |
| 13:34.52 | robbie2 | /proc/pci shows |
| 13:34.55 | robbie2 | Network controller: Tiger Jet Network Inc. Model 300 128k |
| 13:35.06 | _grey_ | how can i make zaptel compile for 686 not 586 |
| 13:36.07 | Skarmeth | I know that I will need three E1 ports to link with channel banks (about two) and telco company if I want to use POTS (actual telephony infraestructure) |
| 13:36.23 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 13:37.23 | Skarmeth | and if I use IP phones and softphones, I'll just need the telco E1 port |
| 13:37.53 | coppice | why does everyone have a strange echo problem? what happened to all the normal boring echo problems? :-) |
| 13:38.51 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:41.08 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:41.15 | *** part/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:41.33 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:41.36 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:42.46 | _grey_ | do u have to specify in asterisk if you want clients to be able to connect using iax ? |
| 13:42.55 | asteriskmonkey | yes |
| 13:42.58 | asteriskmonkey | in iax.conf |
| 13:43.56 | asteriskmonkey | you have to create a user in iax.conf and set them up in extension.conf ie. exten=>yourext,1,dial(IAX/username) |
| 13:44.38 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 13:45.46 | _grey_ | i had the user setup and was using SIP now want to test IAX |
| 13:46.40 | tzanger | http://www.mixdown.ca/~andrew/dump/dirty.jpg |
| 13:49.30 | [TK]D-Fender | tzanger : LOL |
| 13:53.30 | *** join/#asterisk trym_ (n=trym@c213-158-252-242.sdsl.no) |
| 13:53.47 | sivana | tzanger: haha |
| 13:55.24 | Katty | mew. |
| 13:55.30 | bigjb | anyone know why when using phonerlite to connect externally to asterisk that it creates the incoming channel ok, but the outgoing channel doesnt connect? |
| 13:55.53 | Katty | iDunno: allo. |
| 13:56.03 | iDunno | :) |
| 13:56.07 | [TK]D-Fender | Katty: Mew. |
| 13:56.29 | Katty | [TK]D-Fender: hiya. |
| 13:56.40 | Katty | surgery in 16 days. |
| 13:57.37 | [TK]D-Fender | :/ |
| 13:58.10 | *** join/#asterisk basta (n=basta@213-156-52-98.fastres.net) |
| 13:58.41 | iDunno | Katty: :( what's the surgery for? |
| 13:59.15 | Katty | iDunno: doc says my impacted wisdom teeth must go |
| 14:00.35 | iDunno | Katty: ohh, pulling stuff out and fixing things then - hope it all goes well |
| 14:01.09 | Katty | iDunno: thanks :) |
| 14:01.11 | Hmmhesays | nickelback rocked last night |
| 14:01.20 | Katty | iDunno: i'm trying not to worry myself sick about it, but not having much luck there. |
| 14:01.28 | Katty | maybe Hmmhesays will come hold my hand through the operation ;) |
| 14:01.36 | Hmmhesays | when is it? |
| 14:01.45 | Katty | read up. |
| 14:01.46 | iDunno | Katty: *hugs* - it'll all go well, and you'll feel better for it afterwards :) |
| 14:02.06 | Katty | iDunno: yeah i know, i'm just not too keen about taking out another loan, having a surgeon slice me open, rip bones out, etc. |
| 14:02.42 | iDunno | Katty: the first bit I can understand, loans are evil... the rest is fairly much par for the course, though :) |
| 14:04.45 | robbie2 | how do i specify TRUNK for a TE110 ? |
| 14:04.48 | Katty | iDunno: indeed...i've talked to many people, read many websites... |
| 14:04.55 | robbie2 | i keep getting channel unavailable |
| 14:04.59 | Katty | iDunno: and the wisdom teeth aren't hurting too bad just yet.. |
| 14:05.07 | robbie2 | been trying Zap/1 |
| 14:05.08 | Katty | iDunno: the dentist even said i have gorgeous teeth! |
| 14:05.17 | robbie2 | my wife had a cesarian |
| 14:05.19 | iDunno | Katty: \o/ |
| 14:05.26 | robbie2 | the surgeon said she could win miss uterus |
| 14:05.34 | robbie2 | he had a sick sense of humor |
| 14:06.06 | Katty | iDunno: i just hate having to wait almost 2 weeks to get it over with... |
| 14:06.43 | Katty | iDunno: in my opinion, that's way too long to worry. |
| 14:06.59 | _grey_ | I had a "text book case" of an impackted wisdom tooth and it took them over 3 hours to get it out |
| 14:07.02 | iDunno | Katty: just over 2 weeks... yeah - I can understand that, it is *far* too long, the solution is to not worry about it though :) |
| 14:07.35 | Katty | iDunno: yeah, well that i /can't/ do...worry about everything. |
| 14:07.46 | iDunno | Katty: don't worry, be happy :) |
| 14:07.52 | markit | any one with FWD, iax, and registration problems (solved)? seems that can't register, but can't understand why |
| 14:07.56 | Katty | iDunno: that's impossible :) |
| 14:08.10 | Katty | iDunno: i find that the food network on cable is a good distraction though. |
| 14:08.37 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 14:09.11 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 14:10.53 | [Atlas] | is 16khz sampling in the works? |
| 14:11.35 | iDunno | Katty: hmmm - do they cook stuff? :) |
| 14:11.56 | iDunno | Katty: you could just cook lots of interesting stuff, or avoid thinking about it, or something :) |
| 14:14.49 | Katty | iDunno: nah, i'd rather watch them cook. |
| 14:14.53 | Katty | iDunno: this one guy gets really into it. |
| 14:14.58 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 14:14.59 | Katty | iDunno: tis funny. |
| 14:15.06 | axscode | Calling ID is Not Registered |
| 14:15.14 | axscode | anyone happen to know how to fix that? |
| 14:15.24 | Katty | iDunno: so you had this surgery? |
| 14:16.14 | iDunno | Katty: nah - I've got well behaved teeth |
| 14:16.16 | *** join/#asterisk coppice (n=chatzill@196.162.17.210.dyn.pacific.net.hk) |
| 14:16.23 | Katty | iDunno: excellent :) |
| 14:16.39 | markit | solved, stupid me :( |
| 14:16.44 | iDunno | Katty: very good at avoiding worrying about things though :) |
| 14:16.53 | Katty | iDunno: lucky bastard :< |
| 14:17.39 | _grey_ | I get: |
| 14:17.41 | _grey_ | <PROTECTED> |
| 14:17.43 | _grey_ | <PROTECTED> |
| 14:17.45 | _grey_ | <PROTECTED> |
| 14:17.47 | _grey_ | <PROTECTED> |
| 14:17.49 | _grey_ | <PROTECTED> |
| 14:17.51 | _grey_ | how can I make asterisk use gsm as requested by the client ? |
| 14:17.54 | iDunno | Katty: nah - means that I end up doing things like working stupidly long hours to avoid thinking about the things that worry me, then I can't sleep because things nag at me to pay them attention :) |
| 14:19.50 | Katty | iDunno: still better than worrying. |
| 14:19.53 | *** join/#asterisk Abbas (n=Abbas@203.81.196.140) |
| 14:20.22 | iDunno | Katty: well, there's no point in worrying about the things that you can't fix - and the things you can fix, you'll fix, so why worry :) |
| 14:22.00 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-244.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 14:22.09 | *** join/#asterisk nix_ (n=nix@193.198.162.13) |
| 14:22.39 | Katty | iDunno: because i can't stop. |
| 14:22.43 | Babayfax | #asterisk |
| 14:22.49 | iDunno | Katty: why not? :) |
| 14:23.02 | Katty | iDunno: because my brain just isn't setup that way. |
| 14:23.17 | axscode | hi guyz.. im trying to make an outside call to a trunk .... |
| 14:23.18 | axscode | <PROTECTED> |
| 14:23.26 | iDunno | Katty: just needs some reconfiguring... hang on :) |
| 14:23.29 | axscode | and I got that response fromt he cli console... please help.. |
| 14:23.44 | Katty | iDunno: i've tried reconfiguring it many a time ;) |
| 14:24.21 | iDunno | Katty: damn, maybe it's just missing some wires? :) |
| 14:24.41 | Katty | iDunno: nope, just the way i'm wired. |
| 14:24.57 | iDunno | Katty: see! just missing some in the right places :) |
| 14:29.30 | Katty | thanks :) |
| 14:35.08 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 14:35.21 | *** join/#asterisk TallAndy (i=TallAndy@83.104.196.72) |
| 14:37.25 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 14:37.49 | mover | hi all |
| 14:38.01 | mover | are any dialplanguru here? |
| 14:38.10 | _grey_ | is it possible to have an extension allow different types of connection protocol, (i.e both SIP and IAX) |
| 14:38.44 | srt | sure |
| 14:38.59 | mover | i need to figure a "parallel call" for incoming calls on an asterisk |
| 14:39.00 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:39.24 | [TK]D-Fender | mover : pLEASE ELABORATE |
| 14:41.01 | mover | [TK]D-Fender : i need to spread an incoming call to multiple Dial's with multiple handlings |
| 14:41.19 | mover | [TK]D-Fender first to pickup the phone win |
| 14:41.54 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 14:42.25 | mover | [TK]D-Fender Dial SIP&SIP&SIP work but i cant handle the SIP Peers sepatate |
| 14:43.16 | mover | [TK]D-Fender as an info can i mix tech in Dial? |
| 14:44.02 | *** join/#asterisk bidatz (n=ircap8@6.Red-83-32-234.dynamicIP.rima-tde.net) |
| 14:44.14 | bidatz | epaaa |
| 14:44.42 | *** join/#asterisk Peste (i=Peste@195.230.162.134) |
| 14:44.46 | Peste | hello! |
| 14:45.20 | Peste | is an ISDN/PRI-profi online? |
| 14:45.33 | hackeron | I bought a Wildcard TDM400P REV I card and I followed the configuration procedure "Asterisk+Zaptel+Installation" on the wiki, but when an incoming call comes in, I see the following http://rafb.net/paste/results/dsLpIm16.html - any ideas? |
| 14:45.38 | *** part/#asterisk bidatz (n=ircap8@6.Red-83-32-234.dynamicIP.rima-tde.net) |
| 14:45.58 | mover | [TK]D-Fender ok i see i can mix tech |
| 14:46.40 | hackeron | I can see the channels when I run zap show channels and all |
| 14:47.50 | watchy | i think your extensions are screwed up |
| 14:47.52 | watchy | i think |
| 14:47.59 | watchy | i by far am no asterisk god hackron |
| 14:48.12 | mover | hackeron paste your extensions.conf to pastebin.ca |
| 14:48.18 | _grey_ | is it possible to have an extension allow different types of connection protocol, (i.e both SIP and IAX) |
| 14:48.28 | watchy | grey: someone said year earlier i think |
| 14:48.53 | _grey_ | watchy: what ? |
| 14:49.12 | watchy | yea i mean |
| 14:49.13 | watchy | not year |
| 14:49.14 | mover | _grey_ yes pint sip to context=incoming ant iax context=incoming and so you have two tech to incoming |
| 14:49.32 | asteriskmonkey | anyone knoe if i could set each item in an extion conext to have a limit then go to next? |
| 14:49.32 | mover | pint=point |
| 14:49.40 | hackeron | mover: watchy: hmm, the extensions.conf works with IAX and SIP - is there anything special I need for it to work with zapata? |
| 14:49.51 | mover | asteriskmonkey what limit? |
| 14:50.04 | watchy | hackeron: i just think somethings incorrect, post your extensions.conf |
| 14:50.08 | _grey_ | mover: umm sorry don't understand |
| 14:50.14 | asteriskmonkey | like a time limit of 1 minute then rolls the next pritorty in ext.. |
| 14:50.19 | mover | hackeron mybe, depent on yout extensions |
| 14:50.20 | axscode | anyone tried using astbill? |
| 14:50.32 | hackeron | watchy: mover: hmm, ok, sec - its quite big :) |
| 14:50.50 | mover | _grey_ open sip.con and context unter general ist the default context |
| 14:50.57 | mover | same with iax |
| 14:51.11 | asteriskmonkey | move: i want do be able to do something like exten=>xxxx,1,something,for so long and so on |
| 14:52.00 | mover | asteriskmonkey what is something? an agi? |
| 14:52.30 | asteriskmonkey | mover: i want to be able to do it in extensions.conf |
| 14:52.34 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 14:52.37 | *** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net) |
| 14:53.13 | *** join/#asterisk sd-tux (i=sd@2001:4ca0:0:fe00:0:0:a96:3f18) |
| 14:53.54 | mover | asteriskmonkey what you wnat to do depent on how long it takes. if you want to do something a while you need a flow control in asterisk or in an agi |
| 14:55.08 | asteriskmonkey | mover: so setting absoulte timeoust dosnt work anymore :) |
| 14:55.20 | hackeron | mover: watchy: here you go - http://rafb.net/paste/results/2ayvBl67.html -- just replaced the phone numbers and userid |
| 14:55.53 | _grey_ | mover: maybe i phrased my questin wrong, I want to have a single extention say 1040, but allow a softphone to connect using IAX, but also occosionally a hardphone using SIP |
| 14:56.12 | asteriskmonkey | mover : its built in :) exten => x,p,absolutetimeout,limit |
| 14:56.25 | asteriskmonkey | well was in 2003 dont know about present :P |
| 14:56.46 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 14:58.41 | *** join/#asterisk sthw45ywyw5 (n=sthw45yw@38.136.42.2) |
| 14:59.36 | axscode | <PROTECTED> |
| 15:00.17 | *** join/#asterisk Cresl1n (n=matt@146.229.182.227) |
| 15:01.24 | _grey_ | I asume ulaw requires more bandwidth than gsm ? |
| 15:01.32 | dudes | you would be corrent |
| 15:01.36 | dudes | err correct* |
| 15:02.07 | dudes | http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 15:02.09 | mover | hackeron what is the context in zapata.com |
| 15:02.15 | _grey_ | how can I make asterisk use gsm I keep getting > requested format = gsm, |
| 15:02.20 | hackeron | mover: incoming |
| 15:02.24 | _grey_ | <PROTECTED> |
| 15:02.24 | _grey_ | <PROTECTED> |
| 15:02.24 | _grey_ | <PROTECTED> |
| 15:02.24 | _grey_ | <PROTECTED> |
| 15:02.27 | hackeron | mover: just like in sip.conf and iax.conf |
| 15:02.36 | asteriskmonkey | _grey_ : 80k for ulaw with over head vs 18ish for gsm |
| 15:02.38 | *** join/#asterisk hnr (n=hnr@213-156-52-98.fastres.net) |
| 15:02.53 | mover | hackeron what version you use? |
| 15:03.06 | hnr | what can i use in c for programming an agi ? |
| 15:03.11 | iDunno | hmm - is it home time yet? |
| 15:03.29 | hackeron | mover: 1.2.1 |
| 15:03.29 | sthw45ywyw5 | Can someon help me with this zap problem I have posted the details to the asterisk user's mailing list but no one seems to know. here is the post: http://lists.digium.com/pipermail/asterisk-users/2006-January/144974.html |
| 15:04.22 | mover | hackeron add a s,1,noop(mainmenu) at first in context mainmenu |
| 15:04.36 | _grey_ | what is the least bandwidth hungry FREE codec? |
| 15:04.38 | asteriskmonkey | yea you are using a psnt instead of a pstn |
| 15:04.54 | hackeron | mover: what does that do? |
| 15:05.03 | mover | hackeron i guess the goto dont throw the call into mainmenu |
| 15:05.05 | Ahrimanes | _grey_: hm gsm og ilbc? |
| 15:05.08 | sthw45ywyw5 | bad typist |
| 15:05.40 | mover | it only print the word "mainmenu" in the asteisk console if the dialplan reach this extens. line |
| 15:06.10 | asteriskmonkey | sthw45ywyw5: youve probably got the bugged version of asterisk, upgrade it youll be fine |
| 15:06.19 | mover | _grey_ speex or g726-8 :-) |
| 15:06.25 | Peste | is an ISDN/PRI-profi online? |
| 15:06.29 | sthw45ywyw5 | are you kidding me?!?!? |
| 15:06.48 | *** join/#asterisk Cresl1n (n=matt@146.229.182.227) |
| 15:07.30 | Peste | how can i switch between "explicit" and "implicit" channel identifier? |
| 15:07.47 | sthw45ywyw5 | is asterisk really that unstable. This seems like a very basic function. How could it not work? |
| 15:08.04 | coppice | I think G.726-8 would sounds quite interesting :-) |
| 15:08.13 | asteriskmonkey | sthw45ywyw5: easy they broke sip in one of the newer verions it was fixed right away though |
| 15:08.30 | sthw45ywyw5 | i will try. Thanks |
| 15:08.56 | Ahrimanes | sthw45ywyw5: dont run 1.2.2 |
| 15:08.59 | asteriskmonkey | sthw45ywyw5: youll be fine :) fixed 4 other people with that problem.. just make sure you grab your svn src from branches |
| 15:09.11 | *** join/#asterisk tzafrir (n=tzafrir@85-64-243-145.barak-online.net) |
| 15:09.23 | sthw45ywyw5 | I am going to download http://ftp.digium.com/pub/asterisk/asterisk-1.2.4.tar.gz |
| 15:09.32 | Ahrimanes | me too |
| 15:10.25 | _grey_ | does anyone know of good tools for testing network latency other than ping ? |
| 15:11.16 | mover | hi coppice ! |
| 15:11.27 | coppice | hi |
| 15:11.28 | Err | udping will give you more accurate numbers, usually, as ICMP Echo is often lowest-priority |
| 15:11.30 | mover | long time not reade! |
| 15:11.45 | mover | :-) |
| 15:11.49 | Err | (it'll also work on hosts that filter ICMP Echo) |
| 15:12.01 | mover | whois coppice |
| 15:12.04 | mover | =) |
| 15:12.27 | mover | coppice can you help me with my problem? |
| 15:12.44 | mover | i guess you are a dialplan guru too |
| 15:12.52 | hackeron | mover: no effect |
| 15:13.07 | mover | hackeron no message appear? |
| 15:13.09 | coppice | nope. not a diallan expert at all |
| 15:13.36 | mover | coppice you have a guru status for me |
| 15:13.44 | hackeron | mover: same error: http://rafb.net/paste/results/dsLpIm16.html |
| 15:14.05 | coppice | mover: dial plans are too dull to know about |
| 15:14.43 | mover | coppice mom i elaborate it |
| 15:14.57 | [TK]D-Fender | hackeron : That error appears to be pretty blatently obvious... |
| 15:15.04 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 15:15.05 | mover | hackeron my guess is rith if it is the output with the noop |
| 15:15.06 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 15:15.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:15.07 | Peste | can nobody help me :( |
| 15:15.34 | hackeron | [TK]D-Fender: ? |
| 15:15.41 | mover | the goto dont throw the call into meinmenu context |
| 15:15.50 | [TK]D-Fender | You are missing the context being targeted by Zapata |
| 15:16.39 | hackeron | [TK]D-Fender: oh, it doesnt detect the number, I see now |
| 15:16.52 | mover | coppice i need to simulate "paralel call" with one incoming call to diffrent tech and diffrent cdr settings |
| 15:18.18 | mover | coppice is this possible? |
| 15:18.47 | mover | i need to spread it to many dials :-) |
| 15:18.50 | hackeron | yep, I had exten => <the-number>,1,Goto(mainmenu,s,1) instead of exten => s,1,Goto(mainmenu,s,1) -- thanks |
| 15:19.01 | hackeron | now to fix the crackly sound quality |
| 15:19.04 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 15:21.13 | *** part/#asterisk hnr (n=hnr@213-156-52-98.fastres.net) |
| 15:22.24 | *** join/#asterisk harryk (n=me@195.245.80.178) |
| 15:22.27 | harryk | hi |
| 15:22.45 | hackeron | [TK]D-Fender: wow, that configuration was relatively painless :) -- asterisk rules! |
| 15:23.01 | harryk | i have a dumb question about asterisk and cisco 5350 |
| 15:23.50 | infobox | hello harryk |
| 15:24.05 | harryk | is this able to originate call from one atsrisk server to another through cisco? |
| 15:24.06 | mover | coppice ? |
| 15:24.08 | harryk | using SIP |
| 15:25.21 | *** join/#asterisk rmorris (n=rmorris@d221-85-117.commercial.cgocable.net) |
| 15:25.50 | _Sam-- | hey brad_mssw: you are still using junction networks instead of teliax? |
| 15:26.10 | brad_mssw | _Sam--: for termination, yes |
| 15:26.17 | _Sam-- | what about origination? |
| 15:26.28 | brad_mssw | _Sam--: origination is still teliax right now |
| 15:26.36 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 15:26.36 | brad_mssw | _Sam--: until I find something better :/ |
| 15:26.45 | tzafrir | harryk, I believe that this is basically a matter of the cisco's dialplan/calls routeing |
| 15:26.46 | _Sam-- | what phones do you use with teliax? |
| 15:27.20 | brad_mssw | _Sam--: what do you mean? got asterisk then a mixture of sip/iax atas, zap fxs, and sip phones |
| 15:27.28 | *** join/#asterisk leopardus (n=leopardu@217.22.179.15) |
| 15:27.31 | harryk | tzafrir: i can't do this... i'm getting 404 sip msg |
| 15:27.55 | harryk | tzafrir: do you have examples for inbound voip dialpeers? |
| 15:28.08 | _Sam-- | i was just curious what hardware phone(s) you were using |
| 15:28.12 | tzafrir | harryk, I don't know that cisco. But give mor details on what you try to do. There are several possible points of failure |
| 15:28.40 | *** join/#asterisk RoyK (n=roy@242.80-203-45.nextgentel.com) |
| 15:28.44 | leopardus | hello : how do dial the console ?? ;) |
| 15:29.00 | _Sam-- | leopardus: press 0 |
| 15:29.03 | _Sam-- | <just kidding> |
| 15:29.09 | tzafrir | leopardus, two options: 1. load chan_oss / chan_alsa |
| 15:29.29 | rmorris | anyone able to help with a dialplan problem? |
| 15:29.31 | brad_mssw | _Sam--: for sip phones, got the linksys 941 |
| 15:29.33 | tzafrir | leopardus, 2. write a script to generate a call file and run it with ! |
| 15:30.42 | rmorris | I am having a hard time getting unavailable voice mail, but busy works fine |
| 15:30.53 | tzafrir | There should be a wait to produce an "originate" through some external module (from the CLI, I mean), but I forgot which |
| 15:30.53 | harryk | tzafrir: i need two dialpeers - inbound voip and outbound voip. outbound voip dialpeer is working, bcoz i have tested it by calling from POTS |
| 15:31.14 | _Sam-- | rmorris: do you use the std-extension macro? |
| 15:31.17 | tzafrir | harryk, Asterisk can provide both peers |
| 15:31.18 | [Atlas] | has anyone tested the Aastra 9112i phone ,, any good? |
| 15:31.18 | harryk | (using inbound pots dialpeer) |
| 15:31.36 | leopardus | _Sam-- : so dialing 0 from a sip phone should get the console ringing? |
| 15:31.37 | harryk | tzafrir: asterisk is working normally ;-) |
| 15:31.38 | rmorris | can I paste here? |
| 15:31.43 | _Sam-- | leopardus: NO. |
| 15:31.44 | tzafrir | ~pb |
| 15:31.46 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:31.46 | _Sam-- | that was just getting |
| 15:31.50 | _Sam-- | er just kidding. |
| 15:32.29 | leopardus | _Sam-- : so how can one call the console from an internal sip phone? |
| 15:32.58 | _Sam-- | im not sure....i dont know why you would want to call the console though? you could call the computer that is running asterisk |
| 15:33.04 | _Sam-- | but im no expert on that by any means |
| 15:33.42 | Hmmhesays | argh why won't auth_radius work? |
| 15:33.56 | leopardus | _Sam-- : that's exactly what I'm asking, do I need a sip phone on the computer running asterisk, or is there a way to call asterisk directly? |
| 15:34.24 | rmorris | here is the link http://pastebin.ca/39486 |
| 15:34.27 | CaViCcHi | why: CallerID feed failed: Unknown error: 0 |
| 15:34.28 | leopardus | _Sam-- : I think asterisk has its own 'sip phone' |
| 15:34.30 | CaViCcHi | ? |
| 15:34.35 | _Sam-- | im not positive, but i dont think you can use the console as a softphone |
| 15:34.35 | p0g0__ | Hi, I've two Sipura SPA-2002's and Asterisk. I can ring an extension on the second SPA-2002 from the first SPA-2002. However, no voice is transmitted, and after about 15 seconds, I get a busy signal. Any suggestions? |
| 15:34.40 | rmorris | Busy works, but unavail does not |
| 15:34.46 | _Sam-- | maybe there is a way but i just dont know |
| 15:35.12 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 15:35.31 | Hmmhesays | anyone in here got auth_radius to work with SER? |
| 15:36.01 | *** join/#asterisk rene- (i=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 15:36.14 | ctooley | What kind of column is the cdr field "accountcode" supposed to be? The wiki shows that it's a varchar but I can't get anything except ints to go into it? |
| 15:37.18 | leopardus | _Sam-- : thanks anyway, I'm going to have a look at the docs. There's an alsa.conf, so ... |
| 15:38.01 | sevard | baweeted |
| 15:38.10 | rmorris | _Sam--, did you see the paste? |
| 15:39.05 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 15:39.16 | synthetiq | is there any call center software that works with asterisk? |
| 15:39.44 | _Sam-- | i did just now..... |
| 15:39.53 | _Sam-- | rmorris check that pastebin again |
| 15:39.58 | rene- | synth: try gnudialer |
| 15:40.01 | _Sam-- | ;exten => s,1,Dial(${ARG2},30,t) ; Ring the i$ |
| 15:40.01 | _Sam-- | exten => s,1,Dial(${ARG2},30,t) ; Ring the in$ |
| 15:40.01 | _Sam-- | exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on$ |
| 15:40.01 | _Sam-- | exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send $ |
| 15:40.01 | _Sam-- | exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, retu$ |
| 15:40.02 | _Sam-- | exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voic$ |
| 15:40.04 | _Sam-- | exten => s-BUSY,2,Goto(default,s,1) ; If they press$ |
| 15:40.08 | Hmmhesays | pastebin.ca |
| 15:40.08 | _Sam-- | ah shit |
| 15:40.10 | _Sam-- | sorry |
| 15:40.14 | _Sam-- | its on pastebin too |
| 15:40.17 | _Sam-- | sorry bout that |
| 15:40.19 | rene- | _Sam: flooding bad |
| 15:40.30 | _Sam-- | what can i do, it was unintentional |
| 15:40.33 | _Sam-- | s happens, sorry |
| 15:40.36 | Hmmhesays | bow down |
| 15:40.44 | rene- | hehehe |
| 15:40.45 | Hmmhesays | you must sacrifice a virgin to me |
| 15:40.51 | _Sam-- | lol |
| 15:40.54 | Hmmhesays | a hot one |
| 15:41.00 | Hmmhesays | female |
| 15:41.08 | JunK-Y | Hmmhesays: ur sister? |
| 15:41.10 | rene- | fuck no, dont sacrifice a hot one |
| 15:41.10 | _Sam-- | (i thought i was pasting a pastebin url) |
| 15:41.11 | JunK-Y | :P |
| 15:41.14 | sevard | it's hot to put knives in women. |
| 15:41.16 | _Sam-- | but apaprently i didnt copy it |
| 15:41.21 | sevard | does that turn you on? |
| 15:41.22 | sevard | mmmmmm |
| 15:41.24 | Hmmhesays | stabby stabby |
| 15:41.25 | _Sam-- | rmorris: http://pastebin.ca/39487 |
| 15:41.44 | Hmmhesays | you know, the whole dying for your cause and getting the 70 virgins |
| 15:41.54 | ctooley | Is there a way to set the accountcode from a call file or the Manager's Originate |
| 15:41.55 | ctooley | ? |
| 15:42.07 | Hmmhesays | you never hear that they are hot female virgins, they could be middle aged balding fat male virgins |
| 15:42.23 | watchy | haha |
| 15:42.46 | Hmmhesays | "woohoo, i car bombed some people, bring on the virgins........<opens door> WTF" |
| 15:43.12 | Hmmhesays | JunK-Y: my sister is 15 |
| 15:43.17 | rene- | that might be just what they are up to |
| 15:43.45 | rmorris | _Sam--, thanks, seems a lot different! I will have to go back and read the docs again!!! |
| 15:44.00 | _Sam-- | rmorris: use the stdextension macro |
| 15:44.08 | _Sam-- | it will do what you want, for many extensions, very easily |
| 15:44.28 | *** join/#asterisk lorinc (n=ang@caracas-3168.adsl.interware.hu) |
| 15:44.39 | _Sam-- | example: exten => 100,1,Macro(stdexten,100,IAX2/sam) |
| 15:44.41 | Hmmhesays | seriously though, has anyone gotten auth_radius to work in SER? |
| 15:45.10 | rmorris | Thanks _Sam-- I will read up on it and see what I can work out. Cheers |
| 15:45.20 | _Sam-- | sure thing, you will get it |
| 15:47.26 | *** join/#asterisk Prival (i=user65@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 15:47.27 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfjsd.dialup.mindspring.com) |
| 15:47.58 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:48.12 | Prival | Hi, what could make all phone stop registering to the asterisk server and the only cure is to reboot the server and reboot the phones (aastra 9133i)? |
| 15:51.31 | Prival | Just notive that I had full => notice,warning,error, debug,verbose and that the full log file is over 500Mb... Would that cause a problem... |
| 15:51.49 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:52.50 | Err | is your logging disk full? |
| 15:52.55 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:53.35 | jbalcomb | Prival does asterisk provide any error message that might be relevant to your problem? |
| 15:54.01 | Prival | Err: Nope, 148Gb left |
| 15:54.19 | Prival | jbalcom, looking at the log right now. 500Mb file is long to search... |
| 15:54.36 | Err | Prival: then it seems unlikely that the log files would be the problem - most loggers don't actually re-read what they've written |
| 15:54.45 | Err | Prival: the 'tail' command will come in handy to see what's at the end of the file |
| 15:55.46 | Prival | Yup, but the error occured 1 hour ago. Since then I was busy getting this call center back on it's feet so no calls were lost... |
| 15:56.09 | Err | are there timestamps in the log? |
| 15:56.23 | *** join/#asterisk coppice (n=chatzill@7.197.17.210.dyn.pacific.net.hk) |
| 15:56.35 | Prival | Err: Yes, I'm extracting the last few hours right now... |
| 15:56.58 | Mod-cuts | afternoon, i have asterisk fully up and running in our office and i'm trying to unregister the sip phones that are not in use, even when unplugged ,asterisk still looks at them as registered? |
| 15:57.40 | *** join/#asterisk sack (n=sack@208.Red-81-32-160.dynamicIP.rima-tde.net) |
| 15:59.22 | MikeJ__ | untill timeout |
| 16:00.02 | *** join/#asterisk Speeder (n=psilva@est-213-228-152-121.netvisao.pt) |
| 16:00.07 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 16:00.14 | ctooley | I get "Feb 1 15:59:01 ERROR[31901]: pbx.c:1408 ast_func_write: Function CDR not registered" now that I"ve got "Account" in my call files |
| 16:00.21 | *** join/#asterisk spunz (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:00.32 | *** part/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
| 16:00.37 | ctooley | CDR's are being writting but using Set(CDR(accountcode)) seems to fail. |
| 16:01.20 | ManxPowe | One might wonder if it should be Set(CDR(accountcode)=blah) |
| 16:01.37 | Speeder | hi peopple.I'm would like authenticate another asterisk to my 'central' asterisk pbx. I have run astgenjey in central asterisk and copy the pub key to the other asterisk. but i can't get connected. |
| 16:01.56 | Speeder | i would like to use iax rsa auth |
| 16:02.22 | ctooley | ManxPowe, Well, the pbx_spool is calling the Set, not my diaplan. I assumed it was correct in the spool handler. |
| 16:02.36 | ManxPowe | ctooley, Ah. |
| 16:02.40 | Prival | I see a big loop where a customer tried to get in a queue and no agent were logged-in... |
| 16:02.41 | ctooley | var = ast_variable_new("CDR(accountcode|r)", c); |
| 16:03.51 | Speeder | how do i use the outkeys/inkeys statement |
| 16:04.10 | ManxPowe | So you have something like this in your .call file? Set: CDR(accountcode|r)=blort |
| 16:04.28 | ctooley | No. I have "Account: blort" |
| 16:04.29 | ManxPowe | AND are running 1.2.x, I assume |
| 16:04.35 | ctooley | 1.2.x, yes |
| 16:04.43 | ManxPowe | ctooley, then you are not following what is in sample.call |
| 16:05.03 | ctooley | if (!strcasecmp(buf, "account")) { |
| 16:05.04 | ctooley | <PROTECTED> |
| 16:05.20 | ManxPowe | All I can do is go by what is in the sample.call included with 1.2.x |
| 16:05.28 | ctooley | According to pbx_spool.c it's looking for Account: blah |
| 16:06.27 | ManxPowe | ctooley, Well one of them has to be wrong. |
| 16:06.43 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 16:07.46 | ctooley | ManxPowe, not really |
| 16:08.04 | *** join/#asterisk j4m3s_ (n=j4m3s@user-24-214-119-188.knology.net) |
| 16:08.13 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:08.31 | ManxPowe | yes, really. sample.call indicated it should be done one way and that part of the source code indicated it should be done a different way. |
| 16:09.33 | ctooley | No, sample.call shows that it _can_ be done one way and the code shows that it _can_ be done another. It doesn't mean that either is the exclusive way... even if one if them is the more _right_ way |
| 16:11.29 | jbalcomb | Is this syntax correct? Dial(LOCAL/854041) Our CDR reporting broke because we are getting a second count for duration. |
| 16:11.46 | jbalcomb | s/are/are not/ |
| 16:11.48 | RoyK | show application resetcdr |
| 16:11.54 | *** join/#asterisk krokodilerian (n=vasil@pirus.securax.be) |
| 16:11.57 | ManxPowe | jbalcomb, you should have a context |
| 16:12.01 | rene- | what does LOCAL refers to? |
| 16:12.07 | RoyK | ~docs |
| 16:12.09 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:12.25 | ManxPowe | rene-, Dial(Local/extension@context), it's sort of like a goto. |
| 16:12.33 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:13.02 | rene- | ill check that out, thanks |
| 16:13.17 | RoyK | ManxPowe: can you detail the difference? dial(local...) creates a new call, right? or channel? what about the rtp/iax2 data stream? |
| 16:13.34 | malverian[work] | Hmm.. I never really understood what the "Local" channel was. |
| 16:13.57 | ManxPowe | RoyK, I dunno, it just works. |
| 16:14.46 | fa_back | I want connect my Siemens HiPath PBX with ZapHFC card running on asterisk box, which signalling type shuld I set in zapata.conf? |
| 16:14.54 | fa_back | and switchtape |
| 16:15.15 | rene- | Royk: you already know where the docs are |
| 16:15.36 | *** join/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch) |
| 16:15.38 | arkanis | ji |
| 16:15.39 | arkanis | hi¨ |
| 16:16.03 | arkanis | does somebody know I have to configure asterisk to work with asternic? |
| 16:16.06 | sevard | DAMNIT! why can't i figure this out |
| 16:17.00 | arkanis | My problem is, that I cannot transfer calls |
| 16:17.00 | sevard | i was probably beaten too much when I was a child |
| 16:17.00 | sevard | that's why I'm so dumb. |
| 16:17.15 | ManxPowe | arkanis, There are at least 6 different ways to transfer a call. Which one are you using? |
| 16:17.29 | *** join/#asterisk zoa (n=zoa@87.215.18.236) |
| 16:17.38 | zoa | hey hoooo |
| 16:17.49 | arkanis | erm |
| 16:17.57 | ManxPowe | fa_back, If you /msg me agan I will put you on /ignore for the rest of your natural lifetime. |
| 16:18.00 | arkanis | I don't really know |
| 16:18.21 | ManxPowe | arkanis, WHAT are you actually doing to transfer the call? |
| 16:18.27 | sevard | can somebody help me here |
| 16:18.28 | sevard | Warning: fopen(/var/spool/asterisk/voicemail/default/140/INBOX/msg0001.wav) |
| 16:18.28 | sevard | [function.fopen]: failed to open stream: Permission denied in |
| 16:18.32 | ManxPowe | flashnet[BNC], *, or TRANSFER keys? |
| 16:18.33 | sevard | <PROTECTED> |
| 16:18.38 | zoa | sevard: then change the permission |
| 16:18.47 | sevard | zoa: It's already asterisk:asterisk |
| 16:18.56 | sevard | -rwx------ |
| 16:19.03 | ManxPowe | sevard, you are running asterisk as non-root and that user does not have permission to open that file. |
| 16:19.08 | paryl | i switched my company to asterisk about 2 months ago, and things have generally been fine, but i have noticed that since the switch, our fax server fails on a lot more faxes that it used to. it appears to be noise on the lines, but we have a million incoming/outgoing faxes and modem connections that are totally fine. the faxes are connected directly to a rhino channel bank which is... |
| 16:19.10 | paryl | ...connected to a TE205P |
| 16:19.29 | sevard | ManxPowe: |
| 16:19.31 | sevard | ManxPowe: root 8912 0.0 1.3 16572 3464 ? S Jan31 0:00 asterisk |
| 16:19.32 | arkanis | @ManxPowe I have configured asterisk that I can hit # to redirect a call |
| 16:19.38 | ManxPowe | paryl, We had something similar happen. The fix was easy enough. |
| 16:20.01 | ManxPowe | arkanis, so you are using DTMF transfers. Any reason you are not using the transfer function of the phone? |
| 16:20.31 | paryl | manxpowe: and the fix was.... ;) |
| 16:20.42 | ManxPowe | paryl, our fix was to get an analog line direct from the telco and not use Asterisk for faxes. |
| 16:20.44 | arkanis | well, that works too |
| 16:20.53 | sevard | ManxPowe: it's clearly running as root, what else? |
| 16:21.02 | paryl | hrmm |
| 16:21.19 | ManxPowe | paryl, that has fixed it for at least three sites. These days we don't even try to run faxes thru Asterisk. They never work well for us. |
| 16:22.08 | ManxPowe | sevard, well audio.php does not have permission to open that .wav file. |
| 16:22.37 | malverian[work] | paryl, You might want to make sure you have your timing set correctly. |
| 16:22.43 | malverian[work] | paryl, Are you using a PRI? |
| 16:23.08 | malverian[work] | paryl, When I set up the asterisk box here, at first I had time sync from line turned off.. that caused a LOT of frame slips, and thus a lot of failed faxes. |
| 16:23.27 | malverian[work] | paryl, After turning it back on, almost every fax comes through fine. |
| 16:23.49 | paryl | malverian: it is a BRI yes, but if the sync was off i shouldn't be having successful conenctions... should i? |
| 16:23.51 | arkanis | @ManxPowe: So, when I call someone I see the phone ringing in the asternic-panel, I try to drag the ringing telephone to another user, but it doesnt work |
| 16:24.04 | malverian[work] | paryl, We had successful connections, successful calls, some successful faxes.. |
| 16:24.18 | malverian[work] | paryl, Even dialing up with AOL from the port would work occasionally. |
| 16:24.50 | malverian[work] | paryl, Paste your span= from zaptel.conf |
| 16:26.22 | paryl | malverian: span = 1,1,0,esf,b8zs |
| 16:26.28 | paryl | that's for the BRI |
| 16:26.30 | malverian[work] | paryl, Hmm.. okay, nevermind ;) |
| 16:26.34 | paryl | :) |
| 16:26.59 | malverian[work] | paryl, To make sure it's not a problem with your channel bank configuration, you could set up spandsp and use rxfax to test receiving a fax. |
| 16:27.14 | malverian[work] | That would let you know if the noise/whatever was on the BRI and not somewhere else. |
| 16:27.31 | paryl | malverian: the thing is, i can't test it because 90% work just fine |
| 16:27.38 | malverian[work] | paryl, Ah.. |
| 16:28.04 | malverian[work] | paryl, Are the faxes coming through at all? Handshake failing, etc? |
| 16:28.07 | *** join/#asterisk los415 (i=los415@los.race.com) |
| 16:28.35 | paryl | malverian: it's only happening with outgoing connections. it's normally a corrupted confirmation that makes it error |
| 16:28.43 | *** join/#asterisk j4m3s_ (n=j4m3s@gateway.digium.com) |
| 16:28.50 | Hmmhesays | argh this is driving me nuts, ser is not giving me any information about why it fails loading auth_radius |
| 16:28.53 | infobox | hello |
| 16:28.55 | malverian[work] | paryl, These days we do faxes digitally.. the fax is received and stored as a tif file, and it's passed off to CUPS for printing and stored in a place where they can be retrieved easily. |
| 16:29.05 | *** join/#asterisk crich1999 (n=crich@p54BF99C6.dip0.t-ipconnect.de) |
| 16:29.05 | ManxPowe | arkanis, so you are not having a problem with Asterisk, you are having a problem with whatever bizzare GUI you are using. |
| 16:29.39 | fa_back | How can i connect my siemens hipath carrier with asterisk via zaphfc card. |
| 16:30.04 | ManxPowe | fa_back, No idea. Not a lot of people use BRIs here. |
| 16:30.26 | malverian[work] | paryl, Not sure. If you can cause it to be reproducable, then you can start making changes to see if it goes away. |
| 16:30.33 | malverian[work] | paryl, Until then you're just shooting in the dark. |
| 16:31.17 | ctooley | Can someone tell me what provides the the "CDR" function that set is looking for? |
| 16:31.40 | fa_back | ManxPowe Can you look at this http://www.pro-linux.de/work/asterisk/asterisk-1.html Is it what i need? |
| 16:31.56 | ManxPowe | fa_back, no. |
| 16:32.00 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:32.12 | infobox | does anyboyd have experiencie with TE110P and TDM2400 in the same CPU? |
| 16:32.21 | ManxPowe | I can't look at it because I have never used BRI with Asterisk |
| 16:32.28 | Mod-cuts | MikeJ__: until timeout stated where do i use that in the sip.conf for each account? |
| 16:32.29 | arkanis | @ManxPowe maybe, but I thougt somebody knows about asterix together with asternic |
| 16:32.39 | ManxPowe | ~amp |
| 16:32.41 | jbot | it has been said that amp is NOT supported here! people using it should join #amportal |
| 16:32.52 | ManxPowe | arkanis, try #amportal |
| 16:33.38 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-214-010.pools.arcor-ip.net) |
| 16:34.40 | CaViCcHi | sorry... Zap/3 is a valid FXS channel connected to a phone, how can i dial him?... i use Dial(Zap/3,10); for timeout... it says is ringing.... but nothing happens... |
| 16:34.46 | arkanis | ampportal? |
| 16:35.03 | arkanis | isn't amp just one specific gui? |
| 16:35.52 | *** join/#asterisk netdur (n=adel@adsl196-111-60-217-196.adsl196-10.iam.net.ma) |
| 16:36.26 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 16:36.29 | _Sam-- | amp is cdr reporting, config interface, plus fop |
| 16:36.58 | arkanis | fop => flash operator panel? |
| 16:37.08 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 16:37.17 | _Sam-- | fraternal order of phones |
| 16:37.32 | _Sam-- | jk...flash operator panel |
| 16:37.35 | *** join/#asterisk tronix (n=dsf@mappy.catbert.org) |
| 16:37.59 | Katty | what's a nice terminal based ripper? |
| 16:38.04 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 16:38.13 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 16:38.55 | *** join/#asterisk amir (n=amir@shield.guindehi.ch) |
| 16:38.58 | Hmmhesays | lame |
| 16:39.05 | Hmmhesays | oops thats an encoder |
| 16:39.07 | iDunno | Katty: for CDs? jack :) |
| 16:39.21 | iDunno | Katty: it's curses based and rocks |
| 16:40.34 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 16:41.12 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 16:41.39 | Hmmhesays | time to completely fark this debian install |
| 16:41.41 | Hmmhesays | woot |
| 16:42.36 | CaViCcHi | Sorry... asking for the dumbest question... |
| 16:42.47 | CaViCcHi | how can i dial an FXS channel? |
| 16:42.53 | tronix | I was asking couple days ago how to see if g729 codec loaded. figured out issue... 'show g729' from * console didn't work because codec didn't load. |
| 16:42.57 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 16:43.05 | CaViCcHi | Dial(Zap/${FXSCHANNEL}) |
| 16:43.14 | CaViCcHi | correct? |
| 16:43.18 | tronix | codec didn't load because the path to licenses dir was owned root:root w/out world read perms, and * was running as asterisk:asterisk |
| 16:43.30 | tronix | so after fixing that, codec loaded and 'show g729' confirmed. |
| 16:44.01 | tronix | (used strace to point out the EACCESS error for accessing the dir.) |
| 16:45.35 | jarrod | hey if smp_affinty on the digium irq is all f's, why does /proc/interrupts show the te410p only sending interrupts to CPU3 (out of 0-3)? |
| 16:45.37 | Dandan | I am looking for someone who has a first hand experience with Voicetronix boards... |
| 16:45.45 | Dr-Linux | _Sam--: hi :) |
| 16:46.03 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 16:46.38 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:47.10 | Dandan | i need to know how to cancel the echo (timer issue?) on voicetronix board... |
| 16:47.19 | Dandan | tronix: put it in the wiki |
| 16:47.27 | tronix | Dandan: ah, good idea |
| 16:47.45 | Hmmhesays | ugh i hate using binary packages |
| 16:47.59 | Hmmhesays | i feel so dirty |
| 16:48.22 | Dandan | not again... |
| 16:48.27 | Dandan | i just compiled 1.2.3... |
| 16:48.30 | tronix | :) |
| 16:48.51 | _Sam-- | hey Doc |
| 16:48.51 | tronix | anybody here know how I'd find a freenode admin? need to sort out nick registration |
| 16:48.57 | _Sam-- | how goes it today? |
| 16:48.58 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 16:49.14 | Dandan | tronix: /admin <se.rv.er> |
| 16:49.19 | tronix | ahh, thanks |
| 16:49.23 | Hmmhesays | bounce baby out the door i ain't gonna take this no more |
| 16:49.27 | iCEBrkr | :( |
| 16:49.35 | fa_back | ManxPowe so what i need? |
| 16:50.17 | Modcuts | Can until timeout be set on a sip account bias? |
| 16:50.48 | _Sam-- | tronix: /stats O |
| 16:50.54 | _Sam-- | er |
| 16:50.58 | _Sam-- | nm |
| 16:51.04 | tronix | :) (and thanks. all good) |
| 16:51.43 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 16:51.48 | Dandan | I am looking for someone who has a first hand experience with Voicetronix boards... |
| 16:51.59 | iCEBrkr | What about second hand? |
| 16:52.01 | *** join/#asterisk signaleleven (n=evan@lion.ragga-jungle.com) |
| 16:52.04 | iCEBrkr | or even third? |
| 16:52.06 | iCEBrkr | :D |
| 16:52.09 | Dandan | :P |
| 16:52.19 | Dandan | need to talk to someone who used those cards! |
| 16:52.24 | Dandan | even with their toes :) |
| 16:52.49 | signaleleven | does anyone know if there's a way to get channel status via a unique id (from the console)? |
| 16:53.24 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 16:53.36 | signaleleven | without having to do a get channel status on each channel |
| 16:55.14 | *** join/#asterisk doofoo (n=irc@68.238.167.98) |
| 16:55.14 | Dandan | no idea... |
| 16:57.14 | Dr-Linux | _Sam--: i figured out that the problem is with script :) |
| 16:57.48 | _Sam-- | good, makes me feel better |
| 16:59.18 | *** join/#asterisk doofoo (n=irc@68.238.167.98) |
| 16:59.31 | *** join/#asterisk FastJack (i=fastjack@p5091DD66.dip.t-dialin.net) |
| 16:59.37 | Dr-Linux | _Sam--: you know how did i check |
| 16:59.48 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
| 17:00.04 | Dr-Linux | _Sam--: i found 2 ways to check the AGI problems |
| 17:00.47 | *** part/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 17:03.14 | iCEBrkr | Dr-Linux: Ever get Sphinx working :) |
| 17:03.26 | fiber0pti | I'm using polycom 500's and I've got just about everything working except the transferring. When someone transfers a call to another extension without talking to the person the transfer just rings indefinately. How do I get the voicemail to pickup after so many seconds after a transfer? |
| 17:03.35 | Dr-Linux | iCEBrkr: never |
| 17:03.53 | iCEBrkr | :( |
| 17:04.21 | Dr-Linux | iCEBrkr: Phinx is working fine with me indivisually, but i'm not sure how it work with AGI/Asterisk :S |
| 17:04.27 | iCEBrkr | fiber0pti: You don't have your extensions setup to take voicemail |
| 17:04.42 | Dr-Linux | i'm not sure how it communicate with asterisk |
| 17:04.42 | iCEBrkr | Dr-Linux: Yea, that's about as far as I got |
| 17:04.50 | _Sam-- | fiber0pti: use macro stdexten |
| 17:04.55 | Dr-Linux | iCEBrkr: does it work for you? |
| 17:05.07 | iCEBrkr | Dr-Linux: Sphinx works, but the sphinx_agi.c thing doesn't |
| 17:05.37 | Dr-Linux | iCEBrkr: you mean it doesn't work for anyone ? :S |
| 17:05.46 | iCEBrkr | I really have to get this trunking setup correctly.. So my extension here at home can ring my extension at my desk at the office all via Asterisk |
| 17:06.00 | [TK]D-Fender | fiber0pti : Thats a dial-plan problem, not a Polycom one... |
| 17:06.04 | jarrod | should "ff" in /proc/irq/XX/smp_affinty load balance interrupts across all CPUs? |
| 17:06.06 | Dr-Linux | iCEBrkr: i tried much, but i didn't find any appropirate docs help for asterisk/sphinx |
| 17:06.07 | iCEBrkr | Dr-Linux: I don't know anyone who has it working, except that webpage with all the junky perl code |
| 17:06.43 | [TK]D-Fender | iCEBrkr : I've "borrowed" a DID from work for just that reason :) |
| 17:06.52 | iCEBrkr | [TK]D-Fender: :) |
| 17:06.59 | fiber0pti | I'm just wondering why it's different when someone transfers versus dialing the extension and waiting. I'm masking the extensions as _XXXX and voicemail picks up fine if someone dials it and the person doesn't pick up. But the transfer is different, why? |
| 17:07.02 | CaViCcHi | zt_call: Unable to reset default ring on 'Zap/4-1' ??? |
| 17:07.37 | [TK]D-Fender | fiber0pti : A transfer is no different from a normal call except the audio gets passed off afterwards. |
| 17:07.51 | Dr-Linux | iCEBrkr: i tried to use a bunch of AGI example scripts but no seems to work, all of them have errors |
| 17:08.14 | [TK]D-Fender | fiber0pti : I'm betting the context creating the incoming call isn't built like the one you're using to do the transfer... |
| 17:08.36 | Dr-Linux | iCEBrkr: asterisk is on one side, but i can't even execute them like >> ./script.agi |
| 17:08.42 | fiber0pti | D-Fender: That's what I'm not sure about. What context is used for the transfer button on a polycom 500? |
| 17:08.43 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 17:08.48 | CaViCcHi | no one can help me? |
| 17:08.49 | fiber0pti | the softkey, that is |
| 17:08.50 | iCEBrkr | Dr-Linux: That's not really how AGI's work. |
| 17:08.52 | Hmmhesays | chmod +X filename |
| 17:09.06 | [TK]D-Fender | fiber0pti : AGain, it has NOTHING to do with your phone, and everything to do with your SIP and extensions setup |
| 17:09.28 | Dr-Linux | iCEBrkr: i know, but atleasy it shows if there is something wrong in script. |
| 17:09.42 | iCEBrkr | Dr-Linux: Oh yeah. yeah, I've done that before |
| 17:10.02 | Dr-Linux | iCEBrkr: i figured out an other thing with AGI |
| 17:10.05 | iCEBrkr | Dr-Linux: Supposedly Asteriks provides a file descripter #3.. Which is supposed to be the audio stream... |
| 17:10.33 | iCEBrkr | The AGI is supposed to fopen() that descriptor and snag the audio and pass it through Sphinx. |
| 17:10.44 | Dr-Linux | iCEBrkr: if the agi script is totally wrong, or empy file, i still get "returning 0" at the CLI :S |
| 17:10.49 | iCEBrkr | I understand exactly how it's supposed to work, but getting it to work is another story :) |
| 17:11.07 | iCEBrkr | I have a good use for Sphinx + Asterisk. |
| 17:11.34 | Dr-Linux | iCEBrkr: does it work for you? :S |
| 17:11.53 | Modcuts | if you add varibles to the global in extensions.conf, should it for any reason delete them/ |
| 17:11.54 | Modcuts | ? |
| 17:11.56 | iCEBrkr | I thought about using Record() and a clever AGI.. But I figured that'd be too damn slow |
| 17:12.27 | exonic2 | Call transfering with asterisk is freaking impossible to account for |
| 17:12.35 | exonic2 | asterisk needs a entirely rebuilt CDR system.! |
| 17:14.02 | Dr-Linux | iCEBrkr: actually we have many expectations from AGI in future, we hve to do to many things, thats all can be done by AGI so for :S |
| 17:14.14 | Netgeeks | you mean a simple transfer to an outside line doesn't get logged/ |
| 17:14.22 | Netgeeks | ?? |
| 17:14.30 | ctooley | BTW the CDR function that wasn't registered is provided by pbx_function.so |
| 17:15.30 | iCEBrkr | Netgeeks: Transfers in general don't get logged. |
| 17:15.32 | exonic2 | Netgeeks, It does |
| 17:15.47 | Netgeeks | you can use the local channel to make sure all calls are logged regardless whether they were part of a transfer or not |
| 17:15.52 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 17:16.07 | exonic2 | Netgeeks, but it logs the CDR src and dst when the call is hungup, A => B, A transfers to C, CDR src, dst is A,C respectively |
| 17:16.09 | iCEBrkr | Netgeeks: and if you're Monitor()'ing a channel, the channels get split and your wave files are all jacked up and confused :P |
| 17:16.12 | exonic2 | What the hell happened to B! |
| 17:17.13 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 17:17.37 | ManxPowe | exonic2, is the transfer happening across an IAX2 link? If so, try notransfer=yes in iax.conf |
| 17:17.46 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 17:17.50 | _grey_ | could you use a modem on asterisk to connect to the PTSN ? |
| 17:18.07 | exonic2 | ManxPowe, I want to be able to transfer, They're SIP channels. The only thing I have to go by is the 'channel' field in the CDR. I might be able to get the job done |
| 17:18.25 | ManxPowe | exonic2, Ah, SIP channels should work. |
| 17:18.30 | *** join/#asterisk mkrufky-away (n=mk@68.160.103.77) |
| 17:18.41 | ManxPowe | exonic2, nontransfer=yes means "keep asterisk in the loop when transfering" |
| 17:19.17 | ManxPowe | but notransfer=yes is an iax.conf specific option |
| 17:19.36 | exonic2 | ManxPowe, that's set to Yes, but the problem is this: SIP User A calls B, A transfers to C, they all Hangup. CDR states src as A's dialed number, dst as C's # |
| 17:20.22 | exonic2 | I might have it taken care of, just have to add some state to my CDR application |
| 17:20.43 | exonic2 | ManxPowe, shouldnt 'asterisk generate a CDR for both calls? |
| 17:20.48 | exonic2 | because it's not. |
| 17:21.04 | ManxPowe | exonic2, I have no idea. I don't bill for calls. |
| 17:21.42 | *** join/#asterisk tronix (n=dsf@mappy.catbert.org) |
| 17:23.18 | Hmmhesays | rockstar by nickelback is kickass |
| 17:23.20 | *** join/#asterisk ronn (n=ronn@62-249-247-240.no-dns-yet.enta.net) |
| 17:23.31 | ManxPower | <PROTECTED> |
| 17:23.36 | ManxPower | I like to torture my Asterisk |
| 17:23.57 | exonic2 | ManxPower, yeah, crazy. |
| 17:24.56 | Dr-Linux | iCEBrkr: do you know any good user interface program? like any opensource call center program that works with asterisk i.e agents etc ? |
| 17:25.59 | iCEBrkr | Dr-Linux: Not really.. I have pretty high standards, so anything that IS out there, is junk to me. |
| 17:26.31 | _Sam-- | dr-linux you need zoa |
| 17:26.34 | _Sam-- | zoa you there? |
| 17:26.46 | *** join/#asterisk roulduke_ (i=2sl5q1k4@p508D1086.dip0.t-ipconnect.de) |
| 17:26.53 | _Sam-- | zoa has a custom call center app for * |
| 17:26.54 | Dr-Linux | _Sam--: zoa is a program? :S |
| 17:26.58 | _Sam-- | but i dont think its open source |
| 17:27.11 | _Sam-- | zoa is a mere program that i wrote, but he hangs out here |
| 17:27.21 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:27.31 | Dr-Linux | _Sam--: i have installed QueueMetrics thats very good, but not free |
| 17:27.45 | _Sam-- | zoa is the king of all queues |
| 17:27.45 | Dr-Linux | zoa: around? |
| 17:27.51 | jarrod | does the digium te410p have hardware DSPs onboard? |
| 17:28.08 | _Sam-- | he's probably busy pushing IDEfisk on people |
| 17:28.09 | Dr-Linux | :S |
| 17:28.18 | ManxPower | jarrod, only if you buy the addon card. |
| 17:28.21 | Dr-Linux | some one packet zoa |
| 17:28.34 | ManxPower | and even then I think the addon card is only echocan and DTMF, not a real DSP. |
| 17:29.28 | CaViCcHi | Sorry can you help me? |
| 17:29.30 | _Sam-- | i dont zoa's call center stuff is free, so it may not matter |
| 17:29.58 | CaViCcHi | sorry... Zap/3 is a valid FXS channel connected to a phone, how can i dial him?... i use Dial(Zap/3,10); for timeout... it says is ringing.... it maybe rings but if I answer... nothing happens |
| 17:30.28 | ManxPower | CaViCcHi, try Dial(Zap/3) |
| 17:30.42 | CaViCcHi | it dials without timeout |
| 17:30.43 | Dr-Linux | _Sam--: i asked many places, but there is no one Perl institute in my country, and no one knows how it works |
| 17:30.59 | CaViCcHi | but happens the same... :( |
| 17:31.03 | _Sam-- | Dr-Linux: what do you want it to do? |
| 17:31.03 | ManxPower | CaViCcHi, then increase the timeout to 30 |
| 17:31.17 | ManxPower | CaViCcHi, then you have some OTHER problem. |
| 17:31.29 | CaViCcHi | ManXpower Like? |
| 17:32.30 | Dr-Linux | i wanna learn any language, so i thought for perl |
| 17:32.55 | _Sam-- | i am biased, but i personally like php better |
| 17:33.12 | _Sam-- | might be better to know php for interacting with asterisk...but that may be a personal opinion |
| 17:33.17 | Err | perl is a better general-purpose language, IMO (but I don't like perl, either) |
| 17:33.46 | _Sam-- | we use php for so many web apps and stuff |
| 17:33.58 | _Sam-- | i couldnt imagine doing it in perl, but only because i wouldnt know how |
| 17:34.16 | Dr-Linux | my basic field is network, but my company wants me to move to software Dept |
| 17:34.31 | Err | oh, if you're writing web applications, PHP makes sense |
| 17:34.51 | CaViCcHi | zt_call: Unable to reset default ring on 'Zap/3-1' |
| 17:35.03 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 17:35.55 | Dr-Linux | Err: but how can i learn PHP we have no institute for such things |
| 17:36.14 | fugitivo | read |
| 17:36.35 | jarrod | does sangoma pri cards have hardware dsps? |
| 17:36.44 | fugitivo | why people believe that they need a person to teach them? |
| 17:36.55 | Err | Dr-Linux: there are probably some good books on PHP, and the PHP homepage has some decent tutorials |
| 17:36.59 | justinu | it's how society works |
| 17:37.07 | *** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 17:37.11 | JunK-Y | Dr-Linux: where do u live? |
| 17:37.14 | Dr-Linux | Err: all i know is thats from google or from irc .. i never saw any institute, even i just saw computer 2 years ago for the first time |
| 17:37.22 | justinu | Dr-Linux: what's up there, friend? |
| 17:37.31 | Dr-Linux | Pakistan (Tribals) |
| 17:37.53 | JunK-Y | jarrod: yes. |
| 17:37.54 | Dr-Linux | justinu: heyyyyyyyyyyyyyyyyyyyy where're you seksy man ;) |
| 17:38.05 | justinu | with my wife :P |
| 17:38.24 | fugitivo | Dr-Linux: you don't need an institute to learn programming |
| 17:38.35 | justinu | that's true, i learned from books |
| 17:38.39 | justinu | and friends |
| 17:38.39 | Dr-Linux | justinu: Njoy! .. but don't worry i'll have one too :P |
| 17:38.48 | justinu | and reading other people's code |
| 17:38.54 | _Sam-- | most people learn from staying up til 4am pulling their hair out |
| 17:39.02 | _Sam-- | and other people's code |
| 17:39.03 | Dr-Linux | oo yeah |
| 17:39.04 | fugitivo | books, internet, code, and coding :) |
| 17:39.17 | fugitivo | you don't learn to code from an institute |
| 17:39.21 | Dr-Linux | justinu: you know i learn to many good things from the book that you told me once |
| 17:39.27 | fugitivo | they can teach you basic programming only |
| 17:39.35 | justinu | Dr-Linux: yeah, that's why I told you about it! |
| 17:40.18 | Dr-Linux | i can struggle much, but i dont know what to do .. |
| 17:40.31 | justinu | experiment |
| 17:40.34 | tronix | Dandan: ok, well, just updated the wiki for the g729 issue, per your suggestion. one less thing to forget about now. :-) |
| 17:40.35 | justinu | try different things |
| 17:40.49 | Dr-Linux | hhm.. |
| 17:40.51 | _Sam-- | or if you cant get it, sometimes you have to give up and have someone else do it |
| 17:40.56 | _Sam-- | and then learn from what they did |
| 17:41.02 | Dr-Linux | justinu: whats dialplan? scripting? or what? |
| 17:41.13 | justinu | you might call it a script |
| 17:41.35 | justinu | a pretty funky scripting language tho :P |
| 17:41.43 | Dr-Linux | funky? :S |
| 17:41.47 | Dr-Linux | bad or good ? |
| 17:41.56 | justinu | funky == odd, bizzare, weird |
| 17:42.07 | _Sam-- | james brown |
| 17:42.08 | Dr-Linux | what is funky ?? is it fun key ? |
| 17:42.16 | justinu | yes, said like funkey |
| 17:42.35 | _Sam-- | like monkey |
| 17:42.43 | Dr-Linux | ooo ic |
| 17:42.45 | _Sam-- | that funky monkey |
| 17:42.55 | justinu | brass monkey |
| 17:43.34 | Dr-Linux | justinu: actually i'm starting work with my java developers for AGI stuff, so they asked me what scrip/language dialplan use, |
| 17:43.38 | Dr-Linux | i said i don't know :S |
| 17:43.41 | _Sam-- | oops, sorry to all i /ver'd |
| 17:43.54 | justinu | it's proprietary to asterisk |
| 17:44.06 | Dr-Linux | _Sam--: i'm using mirc? you checked my version? |
| 17:44.10 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:172:18:0:41) |
| 17:44.18 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 17:44.25 | _Sam-- | i type /ver....it checked everyone's version on the whole channel...i meant to check mine. |
| 17:44.33 | Dr-Linux | /ver |
| 17:44.43 | _Sam-- | BitchX-1.0c19+ by panasync - Linux 2.4.29 |
| 17:44.51 | Dr-Linux | :P |
| 17:44.55 | *** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc) |
| 17:44.58 | iCEBrkr | Irssi 0.8.9 (20031210) - http://irssi.org/ |
| 17:45.10 | tronix | go, irssi! :) |
| 17:45.15 | Dr-Linux | _Sam--: load an ircd and simble.c exploit |
| 17:45.40 | Dr-Linux | i wish i can use Linux on my desktop but can't |
| 17:46.37 | tronix | Dr-Linux: you could also use Linux through qemu, vmware, bochs, etc. |
| 17:46.46 | af_ | gxp2000: if I don't register but use host=ip speeds buttons won't work: any idea how to solve it? |
| 17:47.05 | mogorman | you can use that new thing that runs linux kernel as a windows proccess |
| 17:47.09 | mogorman | whats it called again |
| 17:47.12 | Makenshi | colinux |
| 17:47.22 | Dr-Linux | i don't wanna use like that |
| 17:47.33 | Dr-Linux | i have to do to many stuff, |
| 17:47.40 | Dr-Linux | i have some other problems with my home PC |
| 17:48.11 | *** join/#asterisk bjames (n=bjames@67-102-228-17.adsl.lbdsl.net) |
| 17:48.17 | bjames | hi there |
| 17:48.33 | Dr-Linux | what language is hot and much relevant with Voip/asterisk |
| 17:48.36 | Makenshi | at home i run xchat on my xp desktop using a fedora vm and cygwin x11 |
| 17:48.47 | Dr-Linux | and not difficult, i'll start to learn that! |
| 17:48.56 | mogorman | asterisk is written in C Dr-Linux |
| 17:49.01 | Makenshi | Dr-Linux, for scripting the dial plan? |
| 17:49.02 | Dr-Linux | yeah, i know |
| 17:49.03 | bjames | I just got five Grandstream GXP 2000's |
| 17:49.13 | malverian[work] | Dr-Linux, Perl is fun. It's a good first language. |
| 17:49.21 | Makenshi | perl and python are good to start with |
| 17:49.22 | bjames | these are nice phones for $84! |
| 17:49.22 | Dr-Linux | no for otherthings as well, like AGI stuff |
| 17:49.23 | malverian[work] | Dr-Linux, However, it's probably also good to learn a statically typed language.. |
| 17:49.32 | malverian[work] | Dr-Linux, Eg, C or Java |
| 17:49.56 | Makenshi | malverian[work], does asterisk support c# on mono? i cant remember |
| 17:49.59 | Dr-Linux | malverian[work]: no, we have buch of Java guru over here |
| 17:50.17 | *** join/#asterisk infobox (n=dpizarro@libra.infostar.com.pe) |
| 17:50.35 | infobox | hi |
| 17:50.39 | Dr-Linux | i wanna learn something like perl, php, python, these names are unknown in my country .. but php is a bit |
| 17:51.13 | Dr-Linux | if someone know perl here he can get double money they java coder |
| 17:51.16 | Hmmhesays | Katty you around? |
| 17:51.17 | tronix | Dr-Linux: you can write AGI scripts in perl, C, python, etc. pick whichever one you'd like to learn. any is fine. |
| 17:51.25 | Hmmhesays | ~seen Katty |
| 17:51.31 | jbot | katty is currently on #asterisk. Has said a total of 105 messages. Is idling for 1h 13m 32s, last said: 'what's a nice terminal based ripper?'. |
| 17:51.31 | infobox | please,does the Asterisk/zaptel have any problem of compatibility with a INtel 865 GVSL motherboard? |
| 17:51.31 | *** join/#asterisk ToTo (n=ToTo@host46-49.pool870.interbusiness.it) |
| 17:51.45 | tronix | Dr-Linux: it's easy to do AGI scripts in perl. there are examples at voip-info.org |
| 17:51.49 | Katty | Hmmhesays: yeah hun? |
| 17:51.51 | Dr-Linux | tronix: and hows java for agi? |
| 17:52.04 | Hmmhesays | What firmware are you running on your 501's? |
| 17:52.12 | tronix | Dr-Linux: not sure, never done it with that, but if there's an AGI module for Java, should work. |
| 17:52.27 | Katty | uhh |
| 17:52.32 | Dr-Linux | tronix: yeah i tried all, but i don't know anything about perl or any language, |
| 17:53.00 | Katty | Hmmhesays: bootrom? |
| 17:53.17 | Hmmhesays | maybe |
| 17:53.18 | Hmmhesays | hold on |
| 17:53.34 | sevard | I'm having some problems. Can anyone help me, this just started happening. |
| 17:53.34 | Katty | Bootmrom: 3.0.1.0023 |
| 17:53.40 | sevard | [res_config_mysql.so]Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! |
| 17:53.41 | Dr-Linux | how can i make changes in voicemail stuff ? :S |
| 17:53.43 | sevard | Ouch ... error while writing audio data: : Broken pipe |
| 17:53.54 | sevard | wtf! |
| 17:54.00 | Hmmhesays | what else you got in there? |
| 17:54.16 | Katty | bootblock, ip, sn, model, assembly.. |
| 17:54.17 | Dr-Linux | sevard: try >> pkill -9 mpg123 |
| 17:54.24 | Katty | Hmmhesays: that's under 'phone' |
| 17:54.29 | Katty | Hmmhesays: under 'application'... |
| 17:54.53 | Netgeeks | yuo thinking what version of the sip app? sip.ld? |
| 17:54.54 | Katty | Hmmhesays: main: label, version, p/n, file; components: label, version, p/n |
| 17:55.12 | Dr-Linux | sevard: kill all mpg123 process then try to start asterisk |
| 17:55.16 | Hmmhesays | hmm ok |
| 17:55.27 | Katty | let me look at the ftp server |
| 17:55.32 | *** join/#asterisk Falle (i=falstaf@voip-forum.se) |
| 17:55.33 | Netgeeks | I want one of those new keyboards that knows what you mean to type and accounts for sleepiness and other things |
| 17:56.17 | Katty | Hmmhesays: <!-- $Revision: 1.71.4.4.2.1 $ $Date: 2005/03/10 21:12:00 $ --> |
| 17:56.21 | justinu | a telepathic keyboard? |
| 17:56.21 | Katty | Hmmhesays: that's on sip.cfg |
| 17:56.45 | Hmmhesays | K |
| 17:56.48 | Katty | Hmmhesays: if you can tell me which file to look at the on ftp server, i can tell you |
| 17:56.51 | malverian[work] | Dr-Linux, Perl is still my favorite language. |
| 17:56.53 | Hmmhesays | theres not a mainpage that just says "firmware"? |
| 17:57.00 | malverian[work] | Dr-Linux, It's just _fun_ to program.. most languages aren't very fun :) |
| 17:57.02 | Katty | not that i saw. |
| 17:57.16 | malverian[work] | Dr-Linux, But be careful, Perl (like many languages) lets you write TERRIBLE code ;) |
| 17:57.38 | malverian[work] | Dr-Linux, use strict; and be mindful of regular expressions (use multiline regexp with comments when they're insane) |
| 17:57.45 | Makenshi | i like perl, but a lot of the perl coders i know have defected to python, so i'm picking that up now |
| 17:58.00 | malverian[work] | Makenshi, Python is good too, but it's not as fun as Perl :-P |
| 17:58.10 | malverian[work] | Makenshi, There's a distinction.. Perl is fun, Python is _good_ ;) |
| 17:58.19 | Makenshi | malverian[work], indeed :o) |
| 17:58.20 | Netgeeks | Katty, in the tftpboot dir do you have a file called sip.ver? |
| 17:58.32 | malverian[work] | Python is also much more IDE friendly... since Perl doesn't declare arguments in the function definition. |
| 17:59.02 | Makenshi | i find the way python does code blocks with indention to be a bit weird |
| 17:59.14 | Makenshi | i guess it makes sense though |
| 17:59.31 | malverian[work] | Basically, the more strict a language is, the better it will fit in with an IDE. In order of strictness, it goes about like this: Perl, Python/Ruby, Java/C++/C, Ada |
| 17:59.48 | Netgeeks | if you are allowing the polycoms to write their logs back to the tftp server you should have a file that looks like <mac addr>.app-log and you would look for a line like the following |
| 17:59.49 | Netgeeks | 0201134949|so |*|00|Application, main: Label=SIP, Version=1.6.3.0067 21-Sep-05 13:56 |
| 17:59.52 | malverian[work] | Makenshi, I thought it was weird too, but since I always use proper whitespace, it started to become second nature. |
| 18:00.08 | *** part/#asterisk CaViCcHi (n=matteo@81.208.84.216) |
| 18:00.14 | *** join/#asterisk YARICK (n=spiderma@pool-71-255-198-81.bltmmd.east.verizon.net) |
| 18:00.26 | malverian[work] | Makenshi, In otherwords, it basically allowed me to remove two lines from each block :-P |
| 18:00.40 | Makenshi | malverian[work], it's good, but kind of annoying when you need to quickly make a new code block with a bunch of code.. |
| 18:00.48 | Makenshi | malverian[work], but i guess that depends on your ide |
| 18:01.02 | Katty | Hmmhesays: 1.5.2.0054 |
| 18:01.15 | malverian[work] | Makenshi, If you use any decent editor, vim/emacs/eclipse/gedit.. basically anything besides nano and notepad, it's easy to tab a group of lines. |
| 18:01.16 | Katty | Netgeeks: thanks for heads up (= |
| 18:01.28 | Makenshi | malverian[work], what's the vim syntax? |
| 18:02.12 | malverian[work] | Makenshi, To indent a group of lines? |
| 18:02.19 | Makenshi | malverian[work], yeah |
| 18:02.22 | Netgeeks | any time |
| 18:02.49 | malverian[work] | Makenshi, Select the group and use < and > |
| 18:03.02 | Makenshi | malverian[work], aha, thanks |
| 18:03.49 | malverian[work] | http://vimdoc.sourceforge.net/cgi-bin/vimfaq2html3.pl#14.8 |
| 18:03.51 | [Atlas] | Anyone Looked at or tested the Thomson st2030? |
| 18:03.55 | JunK-Y | :12,19s/^/\t\1/g |
| 18:04.30 | Beirdo | heh, he said vim, not sed :) |
| 18:04.42 | sevard | Can anyone help me with this? I just started getting this error, it might have been because I updated MySQL, I also installed festival. [res_config_mysql.so]Junk at the beginning 49443303 |
| 18:04.45 | sevard | Warning, flexibel rate not heavily tested! |
| 18:04.46 | sevard | Ouch ... error while writing audio data: : Broken pipe |
| 18:04.58 | Makenshi | sed is good too, though i wouldnt try using it as an editor :o) |
| 18:05.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:05.37 | Dr-Linux | sevard: did you try what i said before? |
| 18:05.37 | Beirdo | much of the vi : commands are thinly veiled sed |
| 18:05.41 | sevard | Dr-Linux: I did |
| 18:05.50 | [Atlas] | Beirdo: that will work in vim ;p |
| 18:05.53 | sevard | I made sure there was no mpg123 running and started it, got the same error |
| 18:05.55 | Beirdo | aye |
| 18:06.01 | Dr-Linux | sevard: ps -ef | grep mpg123 |
| 18:06.20 | sevard | all i get back is the grep process |
| 18:06.59 | Dr-Linux | sevard: what version you are using? |
| 18:07.11 | sevard | Asterisk 1.2.1 |
| 18:07.54 | Netgeeks | I doubt the problem is mpg123.. sounds liek asterisk is dying probably due to a bad module or such after it has started mpg123... you are seeing the mpg123 error because asterisk has kaput |
| 18:08.27 | Netgeeks | damn this evil keyboard!!! type what I mean! not what my fingers tell you to! |
| 18:08.31 | ManxPower | sevard, Asterisk will start mpg123 if you have a musiconhold.conf |
| 18:08.52 | *** join/#asterisk sthw45ywyw5 (n=sthw45yw@38.136.42.2) |
| 18:09.07 | sevard | ManxPower: I do have a musiconhold.conf and it worked great 10 minutes ago |
| 18:09.21 | Netgeeks | did you downgrade asterisk versions, sevard? |
| 18:09.25 | sevard | I did not |
| 18:09.28 | malverian[work] | sevard, Were you running as a different user before? |
| 18:09.30 | ManxPower | sevard, well rename it to something else, then try starting asterisk |
| 18:09.30 | Netgeeks | or do a upgrade? |
| 18:09.39 | *** join/#asterisk jaiger (n=jaiger@c-67-165-4-34.hsd1.ct.comcast.net) |
| 18:09.43 | sevard | malverian[work]: I am currently running it as root |
| 18:09.52 | *** join/#asterisk Cresl1n (n=matt@146.229.184.0) |
| 18:10.09 | mogorman | malverian[work]!!! |
| 18:10.12 | mogorman | sphinx? |
| 18:10.12 | malverian[work] | sevard, Hmm.. one of your other modules is failing. |
| 18:10.17 | ManxPower | then when you start "asterisk -cvvv" you should see the REAL error message. |
| 18:10.20 | malverian[work] | mogorman, I've been so busy man, you have no idea :-P |
| 18:10.29 | mogorman | i belive ya |
| 18:10.30 | malverian[work] | mogorman, New job soon. Interview today. |
| 18:10.30 | Dr-Linux | sphinx :P |
| 18:10.44 | malverian[work] | mogorman, It kills me as much as it kills you for me not to have time to work in it :) |
| 18:10.53 | sevard | ManxPower: I'm running it with cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 18:11.01 | Netgeeks | note however, ManxPower said, "you should see..." unfortunately 90% of the time, what you see is the last good module that loaded, and you have to guess at the real failure |
| 18:11.05 | ManxPower | sevard, so what is the error message now? |
| 18:11.10 | malverian[work] | sevard, You might want to rebuild the mysql module after updating mysql. |
| 18:11.16 | sevard | <PROTECTED> |
| 18:11.16 | sevard | Warning, flexibel rate not heavily tested! |
| 18:11.16 | sevard | Ouch ... error while writing audio data: : Broken pipe |
| 18:11.24 | sevard | malverian[work]: how do I do that? |
| 18:11.29 | Netgeeks | what are the last few lines BEFORE the 'Warning, flexible...' |
| 18:11.32 | malverian[work] | sevard, ldd /usr/lib/asterisk/modules/res_config_mysql.so |
| 18:11.35 | ManxPower | sevard, you didn't rename musiconhold yet. |
| 18:11.37 | sevard | <PROTECTED> |
| 18:11.37 | sevard | <PROTECTED> |
| 18:11.40 | mogorman | yeah i hear you malverian[work] , i am anxious |
| 18:11.44 | mogorman | to the extreme |
| 18:11.47 | malverian[work] | sevard, See if there are any missing so files. |
| 18:11.56 | sevard | ManxPower: when I rename musiconhold.conf to musiconhold.bak my last line is [res_config_mysql.so]= |
| 18:11.58 | mutilator | heh |
| 18:12.09 | malverian[work] | sevard, Yeah, it's failing ot load the module.. do what I said :-P |
| 18:12.10 | mutilator | this lady didn';t want to buy an ATA from us but she wanted our voip |
| 18:12.21 | p0g0__ | Hi, I've two Sipura SPA-2002's and Asterisk. I can ring an extension on the second SPA-2002 from the first SPA-2002. However, no voice is transmitted. Any suggestions. |
| 18:12.22 | mutilator | so she went out and bought a polycom 601 |
| 18:12.23 | ManxPower | Netgeeks, prolly so, but I almost never have problems with modules not indicating that they didn't load. |
| 18:12.30 | mutilator | O_O |
| 18:12.31 | malverian[work] | sevard, Likely the MySQL abi changed (it does every release nearly), so it's having trouble linking with mysql. |
| 18:12.32 | sevard | malverian[work]: sorry, i missed it. running. |
| 18:12.47 | Weezey | Do I need to do something special to make $AGI->wait_for_digit('5000') work? Right now, it's not waiting for anything. |
| 18:12.50 | ManxPower | p0g0, start by including all information, like the fact that you have NAT involved. |
| 18:12.54 | sevard | malverian[work]: same error |
| 18:12.55 | Netgeeks | Manx: aye, I just always seem to have to make educated guesses |
| 18:13.01 | malverian[work] | sevard, Same error? |
| 18:13.09 | sevard | <PROTECTED> |
| 18:13.09 | sevard | Warning, flexibel rate not heavily tested! |
| 18:13.09 | sevard | Ouch ... error while writing audio data: : Broken pipe |
| 18:13.14 | p0g0__ | ManxPower: no nat, all on the same net, running *1.2.4 |
| 18:13.19 | malverian[work] | sevard, You ran ldd on it? |
| 18:13.27 | ManxPower | p0g0, then remove allow=all from sip.conf. |
| 18:13.27 | malverian[work] | sevard, You need to show me the output of the ldd command. |
| 18:13.28 | mutilator | do those things support remote admin/webpage? |
| 18:13.29 | Netgeeks | hrm, junk at the beginning... never seen that |
| 18:13.36 | sevard | malverian[work]: sorry, i misunderstood |
| 18:13.40 | sevard | <PROTECTED> |
| 18:13.40 | sevard | <PROTECTED> |
| 18:13.40 | sevard | <PROTECTED> |
| 18:13.40 | sevard | <PROTECTED> |
| 18:13.44 | sthw45ywyw5 | I am usin Polycom 301 IP phones with asterisk I would like to get one of the polycom sounstation phones. Do they work with asterisk? What model do you recommend? |
| 18:13.50 | Beirdo | ~pastebin |
| 18:13.52 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 18:13.54 | ManxPower | disallow=all then allow=onlythecodecyouwant but do NOT use the G726 codec with SIPura unless you patch the source code for Asterisk |
| 18:14.13 | ManxPower | sevard, well now you know what your problem is. |
| 18:14.14 | Hmmhesays | anyone had problems with polycomm 600's still ringing after asterisk has signaled answer? |
| 18:14.27 | ManxPower | Netgeeks, junk at beginning is also an mpg123 message |
| 18:14.28 | Netgeeks | Hmm, yes |
| 18:14.30 | sevard | ManxPower: for some reason it's not linking libmysqlclient.so.15 |
| 18:14.38 | malverian[work] | sevard libmysqlclient.so.15 => not found |
| 18:14.45 | malverian[work] | Recompile the module. |
| 18:14.53 | p0g0__ | ManxPower: the only allow in sip.conf are ulaw and alaw |
| 18:15.03 | ManxPower | sthw45ywyw5, I think the soundstations are analog only |
| 18:15.09 | Netgeeks | Hmm, all polycoms and some sipuras have had that behavior for me at different times |
| 18:15.10 | p0g0__ | ManxPower: there is a disallow=all |
| 18:15.13 | Hmmhesays | was that directed at me Netgeeks |
| 18:15.21 | sevard | I have 2 of those modules, /opt/lampp/lib/mysql/libmysqlclient.so.15 and /usr/local/lib/mysql/libmysqlclient.so.15 |
| 18:15.35 | Netgeeks | Hmm, yes, it was, sorry |
| 18:15.38 | ManxPower | p0g0, Your problem is usually caused by 1) NAT or 2) allow=all. I guess it could be caused by allowing BOTH alaw and ulaw. |
| 18:15.43 | Hmmhesays | what was your fix? |
| 18:15.50 | malverian[work] | sevard, I see, rebuild the res_config_mysql module in your asterisk source. |
| 18:15.57 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 18:16.08 | MstlyHrmls | sthw45ywyw5: the only soundstation that runs SIP is the IP 4000 |
| 18:16.08 | *** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com) |
| 18:16.14 | ManxPower | sevard, your problem is not with asterisk. It's with ldconfig |
| 18:16.15 | p0g0__ | ManxPower: ulaw only then? |
| 18:16.19 | sevard | malverian[work]: I wouldn't know how to go about that besides make clean ; make ; make install _all_ of asterisk, how would I recompile one module? |
| 18:16.26 | ManxPower | p0g0, no reason to allow both ulaw and alaw |
| 18:16.35 | malverian[work] | sevard, make res/res_config_mysql.so |
| 18:16.40 | malverian[work] | At least, I assume it would work. |
| 18:16.48 | sthw45ywyw5 | So the soundstation 4000 will work with asterisk? |
| 18:16.52 | malverian[work] | But there is no damage from recompiling all of it.. |
| 18:16.57 | p0g0__ | ManxPower: 'k (if ignorance is a reason, then I have one). |
| 18:16.58 | Netgeeks | at the time I shotgunned... so I don't know if it was an upgrade in the asterisk code or a upgrade in the polycom. I'm running 1.2.3 now and sip version (Application, main: Label=SIP, Version=1.6.3.0067 21-Sep-05 13:56) and it's gone |
| 18:17.04 | ManxPower | unless he has the path for the library in /etc/ld.so.conf it's not going to work |
| 18:17.15 | MstlyHrmls | sthw45ywyw5: I haven't tested it personally, but AFAIK it should |
| 18:17.28 | *** join/#asterisk Cresl1n (n=matt@146.229.184.0) |
| 18:18.05 | sthw45ywyw5 | can anyone tell me the difference between the polycom 301 and the 501 |
| 18:18.26 | sevard | recompiling module. |
| 18:18.26 | Netgeeks | Hmmhesays, I was in a hurry and didn't do any real investigation into the reason |
| 18:18.43 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 18:19.53 | ManxPower | sthw45ywyw5, 30x has a line based display and no microphone and two call appearences. the 501 has a full pixel based display, 3 lines, and a microphone |
| 18:20.03 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 18:20.04 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 18:20.06 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 18:20.06 | sevard | Asterisk has been loaded. : - ) |
| 18:20.09 | p0g0__ | ManxPower: set only allow=ulaw in sip.conf, killed * & restarted, same- can ring the extension, but no voice |
| 18:20.22 | ManxPower | p0g0, you should see a message on the console. |
| 18:20.50 | ManxPower | p0g0, does your asterisk server have more than 1 IP address |
| 18:21.19 | ManxPower | p0g0, unless you have a disallow=all before the allow=ulaw it's not going to work as expected. |
| 18:21.22 | p0g0__ | ManxPower: no, only 1 IP |
| 18:21.44 | SplasPood | ztdummy on this 2.6.14 box keeps giving me this, and I think it's causing bad lag in my calls: |
| 18:21.45 | SplasPood | rtc: lost some interrupts at 1024Hz. |
| 18:21.48 | p0g0__ | ManxPower: disallow precedes allow |
| 18:21.49 | SplasPood | any thoughts? |
| 18:22.03 | ManxPower | p0g0, paste the single Dial line from your extensions.conf |
| 18:22.38 | ManxPower | SplasPood, I thougt we needed RTC to be 1000Hz |
| 18:22.46 | sevard | awesome!!!! festival works! asterisk works! it sung mary had a little lamb! |
| 18:22.50 | sevard | no way! |
| 18:22.54 | SplasPood | Manx; Well yea, I assume thats the problem.. but why is it not.. |
| 18:23.13 | ManxPower | SplasPood, no idea. check the mailing list archives? |
| 18:23.13 | sevard | that's the coolest shit since coffee! |
| 18:23.22 | SplasPood | google isn't turning up much |
| 18:23.24 | Netgeeks | coffee is normally hot |
| 18:23.29 | SplasPood | I'd assume they get indexed? |
| 18:23.35 | sevard | i like waiting until it's cooled so i can chug it |
| 18:24.12 | ManxPower | SplasPood, google is behind indexing, but unless it's a NEW issue, it should be inthe archives. |
| 18:24.13 | ManxPower | ~mailinglist |
| 18:24.14 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 18:24.27 | ManxPower | the asteriskguru link has current messages indexed I think |
| 18:25.54 | SplasPood | url in jbot is wrong |
| 18:26.00 | jaiger | sthw45ywyw5, the 501 has much nicer features |
| 18:26.01 | SplasPood | for asteriskguru |
| 18:26.19 | ManxPower | try .net and .org |
| 18:26.25 | SplasPood | no domain is right, just a 404 |
| 18:26.29 | SplasPood | I can figure out where to go |
| 18:26.34 | ManxPower | ah |
| 18:26.34 | SplasPood | just think someone should update jbot :) |
| 18:26.48 | ManxPower | tell zoa, it's his link, I think |
| 18:26.49 | SplasPood | http://www.asteriskguru.com/archives/ |
| 18:26.52 | SplasPood | thats what it should be |
| 18:26.55 | *** join/#asterisk loick__ (n=loick@APuteaux-151-1-54-147.w82-120.abo.wanadoo.fr) |
| 18:26.57 | *** join/#asterisk ComPuTeR (n=DeIi-Mav@85.102.154.44) |
| 18:27.14 | iCEBrkr | SplasPood: Whut'up fool |
| 18:27.14 | tronix | sevard: best part is, near my home town, is a monument to Mary Sawyer and her lamb, and nearby is site of the schoolhouse |
| 18:27.31 | p0g0__ | ManxPower: exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) (I think this is the line you want, this is generated by AMP afaict) |
| 18:27.33 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 18:27.43 | SplasPood | iCE: not my asterisk at the moment :( |
| 18:27.56 | iCEBrkr | SplasPood: Shit dude, what's wrong? |
| 18:28.20 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 18:28.25 | ManxPower | p0g0, that does me NO good at all. |
| 18:28.29 | ManxPower | paste it from the console output. |
| 18:28.40 | SplasPood | iCE: somethin /w ztdummy/rtc/?? |
| 18:29.09 | ManxPower | p0g0, and you really should be on #amportal for amp stuff |
| 18:29.14 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
| 18:29.23 | iCEBrkr | SplasPood: What'd ya do this time :) |
| 18:29.37 | SplasPood | iCE: nothin, its always been doing that since I switched to this machine |
| 18:29.43 | SplasPood | dunno if its the kernel or what.. |
| 18:29.50 | iCEBrkr | ahh |
| 18:30.01 | g4m | does anyone here have any experience with meetme rooms that sounds really bad (i'm using ztdummy on linux2.6 and asterisk 1.2.1)? |
| 18:30.03 | *** join/#asterisk arosen (n=arosen@modemcable166.132-82-70.mc.videotron.ca) |
| 18:30.12 | SplasPood | getting this, and after a while calls start to lag horribly |
| 18:30.13 | SplasPood | rtc: lost some interrupts at 1024Hz. |
| 18:30.13 | iCEBrkr | I ran some tests with my new asterisk box.. only able to launch 16 simultanious calls-- but it all worked. |
| 18:30.39 | _Sam-- | ice did you make it solid state |
| 18:30.56 | iCEBrkr | _Sam--: lol.. That'll be a personal project of mine, so I won't get around to that for a while |
| 18:31.17 | _Sam-- | once you start its addicting |
| 18:31.57 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 18:32.00 | _Sam-- | i just want to build solid state devices that i have no use for |
| 18:32.06 | iCEBrkr | _Sam--: I wanna know the results of that USB->PSTN gadget Nugget bought |
| 18:32.15 | SplasPood | g4m: any errors in your kernel output? (dmesg) |
| 18:32.19 | Delvar2 | quick question: is there any problems with using a remote mannager connection that conencts/runs a couple commands/disconnects every 30 seconds or so... iv been told that it can case problems after a while.... |
| 18:32.21 | iCEBrkr | It's some sort of Skype device |
| 18:32.33 | _Sam-- | i didnt look at the device, but i saw the discussion here yesterday |
| 18:32.43 | _Sam-- | what would that allow me to do that i cant do now? |
| 18:32.44 | iCEBrkr | If it works out, you'll be able to bridge Skype <=> Asterisk :P |
| 18:32.56 | *** join/#asterisk sac|h0p|werk (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) |
| 18:33.00 | _Sam-- | i dont have a big need for that |
| 18:33.11 | _Sam-- | in fact, hate to admit it, but ive never not once even used or installed anything skype related |
| 18:33.14 | ManxPower | g4m, only when some callers are using an RTP packet size that is not 20ma |
| 18:33.15 | iCEBrkr | _Sam--: Me wither, but I'd make a use for it :) |
| 18:33.34 | _Sam-- | iCEBrkr: what would be the advantage? so you could call people who use skype? |
| 18:33.41 | _Sam-- | if i wanted to call them, i would just call their cell phone :) |
| 18:33.48 | iCEBrkr | _Sam--: Yeah |
| 18:34.37 | iCEBrkr | _Sam--: Come'on man, you know useless shit is cool :) |
| 18:34.39 | iCEBrkr | Just to say you can do it |
| 18:34.52 | ManxPower | Feb 1 12:33:45 NOTICE[31460]: channel.c:1903 ast_read: Dropping incompatible voice frame on Local/99661320@toll-access-b0f6,2 of format ulaw since our native format has changed to slin |
| 18:34.52 | ManxPower | weird |
| 18:34.54 | iCEBrkr | Like, I have a FWD account I never use, but my Asterisk box still registers there. |
| 18:35.12 | g4m | SplasPood: Something like this? : zaptel: no version for "struct_module" found: kernel tainted. |
| 18:35.12 | g4m | Zapata Telephony Interface Registered on major 19 |
| 18:35.18 | g4m | pardon the double line |
| 18:35.32 | _Sam-- | i guess if you had a lot of friends and family using skype i could see the benefits |
| 18:35.36 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 18:35.39 | _Sam-- | but i dont even know anyone on skype |
| 18:35.47 | iCEBrkr | _Sam--: Me either |
| 18:35.49 | iCEBrkr | lol |
| 18:35.50 | ManxPower | g4m, sounds like a zaptel/asterisk/kernel mismatch |
| 18:35.55 | _Sam-- | lol! |
| 18:36.02 | iCEBrkr | _Sam--: But depending on Nuggets results, I'm gonna get one anyhow |
| 18:36.11 | _Sam-- | cool, i'll call ya |
| 18:36.13 | _Sam-- | :) |
| 18:36.26 | Weezey | I can't seem to get perl AGI to do read_data or wait_for_digit is there a trick to it? |
| 18:36.28 | iCEBrkr | _Sam--: Skype supposedly is gonna have a PSTN interconnect. ( or maybe they already do ) |
| 18:36.29 | g4m | ManxPower: i'm running the debian apt install of asterisk, but i built the zaptel package myself, should i build my own Asterisk as well? |
| 18:36.38 | _Sam-- | they already do |
| 18:36.40 | ManxPower | g4m, always |
| 18:36.41 | _Sam-- | skypeout |
| 18:36.47 | g4m | ManxPower: Thanks |
| 18:36.54 | g4m | SplasPood: thanks |
| 18:37.07 | iCEBrkr | _Sam--: Yeah |
| 18:37.16 | iCEBrkr | _Sam--: So it's just another option for LCR |
| 18:37.18 | SplasPood | g4m: heh, I was hoping you were having a problem similar to mine :) |
| 18:37.41 | _Sam-- | it seems like , to me, that skype is turning into nothing more than a softclient that charges for calls |
| 18:37.49 | jaiger | g4m they have a zaptel driver source that matches the packaged asterisk. I've built/used that without problems |
| 18:37.50 | [Atlas] | sweet Asterisk@Xbox works! |
| 18:37.57 | [Atlas] | err kinda |
| 18:37.58 | iCEBrkr | _Sam--: You mean like Dialpad? |
| 18:37.59 | [Atlas] | hmm |
| 18:38.08 | ManxPower | [Atlas], you pervert |
| 18:38.18 | [Atlas] | LOLOL |
| 18:38.35 | _Sam-- | if dialpad was a softphone, yes. |
| 18:38.42 | [Atlas] | Sorry I had to rescue the perfectly good hardware that was xbox |
| 18:39.01 | iCEBrkr | _Sam--: Dialpad was a webbased internet phone |
| 18:39.07 | Weezey | 700MHz is perfectly good? |
| 18:39.19 | ManxPower | Feb 1 12:37:33 NOTICE[31593]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15 |
| 18:39.20 | ManxPower | weird |
| 18:39.28 | SplasPood | Manx: you think my interrupt issue might be SMP related? |
| 18:39.31 | Weezey | Atlas: build a MythTV box. |
| 18:39.48 | ManxPower | SplasPood, I have no opinion on the issue. |
| 18:39.53 | jaiger | Weezey, depends on your needs. I've used * on a pentium 200 for 1 channel |
| 18:40.06 | iCEBrkr | _Sam--: But DialPad had some serious delay and echo problems. |
| 18:40.07 | _Sam-- | there are some myth/asterisk hybrids out there |
| 18:40.20 | _Sam-- | i think there is a mythsterisk distribution |
| 18:41.11 | iCEBrkr | WTF? Myth + Asterisk? What for? |
| 18:41.12 | SplasPood | Opened pseudo zap interface, measuring accuracy... |
| 18:41.12 | SplasPood | 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.975586% |
| 18:41.24 | iCEBrkr | SplasPood: That looks good actually.. Your box idle? |
| 18:41.31 | SplasPood | ice: mostly.. |
| 18:41.35 | _Sam-- | so you can have a set top box that does many things and shows you caller id on your tv |
| 18:41.47 | iCEBrkr | Hi |
| 18:41.48 | iCEBrkr | err |
| 18:41.52 | SplasPood | you can show your CID /wo runnin asstricks on the settop |
| 18:41.53 | iCEBrkr | Wrong window |
| 18:41.59 | iCEBrkr | CID OSX |
| 18:42.00 | iCEBrkr | lol |
| 18:42.01 | *** join/#asterisk nibbler_ (n=nibbler@some.host.name) |
| 18:42.02 | steve___ | use your phone as a remote >:-) |
| 18:42.06 | _Sam-- | just think about when video sip gets bigger |
| 18:42.10 | Netgeeks | I want 60 inches of CallerID! |
| 18:42.14 | sevard | what cool shiz does everyone do with Festival besides reading weather from noaa |
| 18:42.19 | _Sam-- | you have your setup asterisk box, myth, and do videoconference on the tv? |
| 18:42.22 | iCEBrkr | Netgeeks: I think you want 60 inches of man meat :P |
| 18:42.34 | SplasPood | sevard: I run a successful phone sex business based around asterisk and festival |
| 18:42.49 | sevard | SplasPood: I hope you're making a profit. |
| 18:42.54 | VxJasonxV | hahahaha |
| 18:42.56 | Netgeeks | for some reason, I should have seen that comming |
| 18:42.57 | _Sam-- | he gets paid in barter |
| 18:42.59 | SplasPood | successful.. |
| 18:43.08 | iCEBrkr | Netgeeks: Yes, yes you should have lol |
| 18:43.28 | sevard | bah, it was just a curious question |
| 18:43.39 | SplasPood | that'd be good for a laugh |
| 18:43.41 | _Sam-- | people STILL use traditional phone sex? |
| 18:43.46 | _Sam-- | instead of video stuff online? |
| 18:43.50 | SplasPood | who knows |
| 18:43.50 | Netgeeks | however having callerID display on the TV is a great thing... don't have to hit pause and hunt down a phone to see who is calling |
| 18:43.52 | iCEBrkr | _Sam--: LOL |
| 18:43.56 | SplasPood | I dunno people who use that shit |
| 18:44.03 | iCEBrkr | MSN::NetSex w00t |
| 18:44.05 | _Sam-- | you dont know who your customers are? |
| 18:44.10 | iCEBrkr | _Sam--: hahaha |
| 18:44.25 | _Sam-- | lol |
| 18:44.34 | kend | Polycom 501 -- any way to have it dial (on-hook) when numbers are simply entered on keypad, w/o hitting speaker or dial buttons? |
| 18:44.58 | jaiger | SplasPood, I'd pay for festival phone sex.... I mean yeah, good idea |
| 18:45.24 | SplasPood | kend: there's a dialplan spec in the config files |
| 18:45.31 | jaiger | I mean, my friend would |
| 18:45.37 | SplasPood | kend: you define the digitmap so it dials upon match |
| 18:45.54 | kend | SplasPod: including for it being on-hook? Any idea what it might be called? |
| 18:46.03 | g4m | jaiger: what driver source did you build? |
| 18:46.21 | jaiger | g4m, IIRC apt-get install zaptel-source |
| 18:46.29 | SplasPood | kend: you mean make a call /wo pressing new call or lefting the handset or throwing it on speaker? |
| 18:46.37 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 18:46.48 | SplasPood | lifting, even |
| 18:47.05 | g4m | jaiger: hrm, i guess that was the one that i used |
| 18:47.07 | kend | SplasPod: Exactly. Our old phone system (people hate change) would go to speakerphone when you started dialing with it on-hook. |
| 18:47.09 | [Atlas] | Weezey Nice idea bro :) |
| 18:47.35 | Weezey | [Atlas]: My buddy's got one on a myth box, works pretty well. |
| 18:47.38 | jaiger | g4m, it has worked for me in the past |
| 18:47.53 | Weezey | [Atlas]: on=as |
| 18:48.05 | kend | SplasPod: Which makes sense, really. I mean, it's not like you'd be hitting the keypad to get excercise. ;-) But if they "have" to make that extra keystroke and hit "dial" or the speakerphone button, they'll live. |
| 18:48.06 | [Atlas] | sweet |
| 18:48.11 | g4m | jaiger: i assume that your running a source build now? |
| 18:48.31 | g4m | jaiger: everything works fine, with the exception of meetme rooms, which sound like your under water. |
| 18:48.33 | SplasPood | kend: hrm.. ok yea, I dunno bout that, sorry |
| 18:48.47 | jaiger | g4m, no. actually I use the debian packaged asterisk if at all possible |
| 18:48.50 | kend | SplasPood: NP. Thanks, anyway... |
| 18:48.51 | SplasPood | kend: Although if you do figure something out I'd be interested in knowing how.. In case I get any similar customers |
| 18:49.15 | jaiger | g4m, ahh. I've never used meetme so I can't vouch for htat feature |
| 18:49.34 | g4m | jaiger: ahh, ok well then i guess its a kernal thing |
| 18:49.55 | *** join/#asterisk TallAndy (i=TallAndy@83.104.196.72) |
| 18:51.11 | *** join/#asterisk loick__ (n=loick@APuteaux-151-1-54-147.w82-120.abo.wanadoo.fr) |
| 18:51.43 | *** join/#asterisk usleopard (n=leopardu@217.22.179.15) |
| 18:52.35 | *** join/#asterisk loick__ (n=loick@APuteaux-151-1-54-147.w82-120.abo.wanadoo.fr) |
| 18:52.54 | usleopard | hello : how can one 'invite' the console? |
| 18:53.26 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 18:56.21 | p0g0__ | ManxPower: fwiw- the ring but not voice was a silent firewall rule. |
| 18:57.15 | ManxPower | p0g0, that would do it |
| 18:58.33 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:01.30 | *** join/#asterisk nahirean (n=nahirean@c-68-36-161-8.hsd1.nj.comcast.net) |
| 19:03.34 | jbalcomb | [av]bani is that you requesting seperate volume controls for the grandstream phones? |
| 19:04.06 | [TK]D-Fender | jbalcomb : So, how goes the war? |
| 19:04.46 | sthw45ywyw5 | What does the 501 have that the 301 does not |
| 19:05.06 | jbalcomb | [TK]D-Fender we have established a cease-fire. new server is coming and problems are within 'acceptable limits' |
| 19:05.18 | jaiger | sthw45ywyw5, speaker phone (!), more lines, better display, higher price |
| 19:05.39 | jbalcomb | [TK]D-Fender turning down the gains seems to have made a big difference and the new firmware is also much improved. |
| 19:05.59 | bigjb | can anyone give me some help with a nat/sip issue? |
| 19:06.03 | jbalcomb | [TK]D-Fender only thing now is feedback on handset calls because the volume has to be so high for speaker phone calls. |
| 19:06.26 | jbalcomb | haha.. bigjb.. why didn't i pick that nick? ;) |
| 19:06.41 | bigjb | :P |
| 19:06.55 | bigjb | tis mine! |
| 19:07.02 | mutilator | anyone have a util to convert/add from a cdr-csv to cdr-mysql |
| 19:07.09 | mutilator | so i can import old records to mysql |
| 19:07.13 | jbalcomb | but, bigjb, /my/ initals are JB damnit! |
| 19:07.18 | sthw45ywyw5 | is it possible on the 301 (or the 501) to tell asterisk that if aanyon dials my extention, to forward it to my cell phone? |
| 19:07.26 | bigjb | so are mine =oP |
| 19:07.39 | jbalcomb | sthw45ywyw5 that is possible using asterisk regardless of your phones. |
| 19:07.46 | jaiger | sthw45ywyw5, I would think that is a function of asterisk and not the phone you use |
| 19:08.18 | sthw45ywyw5 | can the user set that from the phone or does the pbx admin have to set it. |
| 19:08.48 | jyukes | hi -- whats like a good round-trip VoIP latency? I'm doing the Asterisk echo test with Level3 and getting 380ms |
| 19:09.00 | jaiger | sthw45ywyw5, dunno but you can probably program it either way you want |
| 19:10.08 | jbalcomb | sthw45ywyw5 you can setup and 'application' that tells asterisk to set that. |
| 19:10.23 | Netgeeks | mutilator: do you have a windows system available? |
| 19:10.49 | jbalcomb | sthw45ywyw5 ie. dial *61XXXXXXXXXX and asterisk can put the info in the DB and know to forward calls to that number. |
| 19:10.52 | sthw45ywyw5 | So it can not be done from the 301/501 phone? Wht is the fwd function on the 301 phone do. |
| 19:11.09 | jbalcomb | sthw45ywyw5 it shouldnt be done from the phone. |
| 19:11.22 | jaiger | sthw45ywyw5, it allows you to forward to anotehr extension/phone - that's how I've used it before |
| 19:11.31 | jbalcomb | sthw45ywyw5 unless the phones function actually lets asterisk know what is being done |
| 19:11.53 | jaiger | sthw45ywyw5, if you had an extension setup for your cell phone then I guess you could use the fwd feature |
| 19:12.02 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 19:12.28 | MattH | Hi... I'm getting this error when an IAX peer tries to sync up.. any ideas why? |
| 19:12.28 | MattH | <PROTECTED> |
| 19:12.40 | [TK]D-Fender | sthw45ywyw5 : yes you can forward every which way you want ON the Polycom's direct. |
| 19:12.50 | Netgeeks | you don't have host=dynamic |
| 19:12.53 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 19:12.58 | sthw45ywyw5 | <[TK]D-Fender>: how? |
| 19:13.13 | }btorch{ | any knows how asterisk works on a 64bit plataform ? |
| 19:13.14 | [TK]D-Fender | sthw45ywyw5 : Try using the FORWARD button right on it..... |
| 19:13.17 | Netgeeks | you prolly have something like host=1.2.3.4 where 1.2.3.4 is some ip address you expect the host to be at |
| 19:13.18 | }btorch{ | does it work ? |
| 19:13.22 | jbalcomb | [TK]D-Fender wouldn't it be bad form to have the phone doing something without asterisks knowledge? |
| 19:13.40 | [TK]D-Fender | jbalcomb : DETAILS! Remember in SIP "phone is king". |
| 19:13.41 | *** join/#asterisk fulgas (n=fulgas@209.8.233.252) |
| 19:13.52 | jbalcomb | }btorch{ it uses numbers that are twice as long as on a 32bit platform |
| 19:14.05 | }btorch{ | :-) |
| 19:14.14 | }btorch{ | thanks for making that clearn |
| 19:14.15 | Netgeeks | you can only accept registers to entries with host=dynamic... if you have host=<ip address> then a register is superfulous |
| 19:14.16 | jbalcomb | [TK]D-Fender mmmm.. i love DETAILS... |
| 19:14.47 | }btorch{ | serious does it compile and work ok on a 64bit suse install ? |
| 19:14.55 | jbalcomb | }btorch{ yes, its handy for dialing international using calling cards cause the numbers are too long for 32bit systems. =) |
| 19:15.21 | bigjb | soooo anyone know why "asterisk <==> nat router <== internet ==> nat router <==> sip client" is causing me to only have an incoming channel on the sip phone? |
| 19:15.32 | bigjb | 5060 is forwarded to the relevant pc on both routers |
| 19:15.59 | *** join/#asterisk Eraserhead (n=Miranda@c-67-164-201-80.hsd1.ut.comcast.net) |
| 19:16.13 | sevard | Is there anyway to cron a dialplan? I have to move my car every 2 hours or I get a ticket. I was trying to think of way to have asterisk call an extension every 1 hour and 15 minutes with some GETOFFYOURASS greeting via festival. I just got a $10 ticket :'( |
| 19:16.26 | sevard | should have asked that question yesterday |
| 19:16.41 | jbalcomb | }btorch{ hrmm.. cant imagine why it wouldnt. i doubt they had to port all the linux apps to 64bit. it simply wont take advantage of it. |
| 19:17.02 | jbalcomb | }btorch{ there may be something about 32bit compatibility in the kernel setup but its prolly default |
| 19:18.03 | jbalcomb | sevard yes, make the cron job build a call file and it in the /var/spool/asterisk <-? directory and asterisk will make the call and you can play any audio file your record |
| 19:18.25 | jbalcomb | sevard iCEBrkr has details on building call files if you cant get it all right from the wiki |
| 19:18.28 | bigjb | $10!!! we get minimum of £30 in most places in uk |
| 19:18.32 | bigjb | tis about $65 |
| 19:18.35 | sevard | wow |
| 19:18.41 | sevard | it's $25 if i don't pay it in 2 days |
| 19:18.47 | sevard | 75 if i don't pay it in 5 |
| 19:19.02 | jbalcomb | i post my parking tickets on my cork board and pay them exactly one year later |
| 19:19.06 | sevard | jbalcomb: I didn't see anything on the wiki but perhaps i wasn't searching for the right thing |
| 19:19.23 | jbalcomb | given the time value of money this makes the tickets cheaper and annoys the lady at the desk |
| 19:19.43 | Beirdo | the nuisance-factor is almost worth it |
| 19:19.49 | sevard | it's retarded to have an office in town without an employee parking lot |
| 19:20.07 | *** join/#asterisk j4m3s_ (n=j4m3s@gateway.digium.com) |
| 19:20.14 | *** join/#asterisk Dorphalsig (n=9dfd0e3b@yossman.net) |
| 19:20.29 | usleopard | --- : how can I call the console? |
| 19:20.37 | sevard | asterisk -r |
| 19:20.53 | jbalcomb | yes'm. the lady told me that i shouldn't do it again because they really do enforce 'these'. A year later I handed her the ticket and payment from the one I got the day I saw her before. =) |
| 19:21.01 | usleopard | sevard : no I mean call the console from a sip phone |
| 19:21.03 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 19:21.08 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 19:21.08 | Weezey | OSS/dsp |
| 19:21.08 | Dorphalsig | can I connect 5 analog lines to a TE100P |
| 19:21.11 | sevard | usleopard: iirc that's not possible. |
| 19:21.21 | Dorphalsig | I mean like with some funny adapter or something? |
| 19:21.26 | sevard | usleopard: unless you have a SIP phone on the same server |
| 19:21.31 | jbalcomb | Dorphalsig are there enough ports to plug five lines into? |
| 19:21.39 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 19:21.41 | usleopard | sevard : but it can dial |
| 19:21.42 | jaiger | Dorphalsig, you need a channel bank |
| 19:21.48 | Netgeeks | Dorphalsig: funny adapter = channel bank |
| 19:22.00 | sevard | usleopard: really? how? i'm an asterisk nublet |
| 19:22.08 | Dorphalsig | I'm speaking of 5 analog lines |
| 19:22.13 | Dorphalsig | I have: |
| 19:22.20 | Dorphalsig | 1 E1 |
| 19:22.30 | Netgeeks | yes, channel bank will convert from T1/E1 to analog fxs/fxo ports |
| 19:22.47 | Dorphalsig | So I connect to CB then to the * card |
| 19:22.51 | jaiger | Dorphalsig, you need a channel bank that takes an E1 and gives you 5 analog (fxs/fxo) lines |
| 19:22.58 | cypromis | 0/w 14 |
| 19:22.59 | Weezey | usleopard: looks like it can't receive calls |
| 19:23.01 | usleopard | sevard : when asteriks calls a sip phone 'asterisk' is displayed |
| 19:23.19 | jaiger | usleopard, that's the default callerid |
| 19:23.24 | Weezey | usleopard: it uses channel OSS/dsp to send the call, but you can't call that. |
| 19:23.30 | sevard | usleopard: i'm curious, what command are you issuing to call a sip phone from the asterisk cli |
| 19:23.45 | Dorphalsig | no no no |
| 19:23.45 | Weezey | sevard: Dial exten@context |
| 19:23.47 | Dorphalsig | you dont understand |
| 19:23.57 | Dorphalsig | I have the e1, my internal extensions |
| 19:24.03 | sevard | No such command 'Dial' (type 'help' for help) |
| 19:24.06 | Dorphalsig | and additionaly I need the 5 lines for an ouitbound campaing |
| 19:24.08 | usleopard | sevard : dial 'ext' |
| 19:24.14 | Weezey | sevard: running svn? |
| 19:24.21 | [TK]D-Fender | jbalcomb : A pity you buy a new server to sompensate crappy phones :/ |
| 19:24.29 | sevard | No such command 'dial' (type 'help' for help) |
| 19:25.03 | Weezey | sevard: they took it out of the source for a while, but it's back now |
| 19:25.07 | jaiger | sevard, it's an "extra" feature |
| 19:25.09 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 19:25.11 | Weezey | I also think you need a sound card for it to work. |
| 19:25.26 | sevard | i have a sound card, does it need to be compiled or is it a module? |
| 19:25.30 | jbalcomb | [TK]D-Fender really, its a backup and test server more than that. over 50% of our revenue is made through the phone system so I'm ok with a zero downtime solution like this. |
| 19:25.36 | jaiger | Dorphalsig, so you need to get your 5 asterisk lines to talk to your existing E1? |
| 19:26.04 | jaiger | Dorphalsig, and that existing E1 already has a channel bank with extensions on it? |
| 19:26.09 | jbalcomb | [TK]D-Fender I have two cisco 7940Gs coming and a SPA-2002. We are using them for testing and then will decide what to do about the phone from there. |
| 19:26.42 | [TK]D-Fender | 7940G = sadly overpriced, and SPA is too terribly cheap and harder to deploy. |
| 19:26.51 | Dorphalsig | yes |
| 19:26.54 | Dorphalsig | basically |
| 19:26.54 | jbalcomb | [TK]D-Fender im looking hard at the SNOM 360 Business as our phone of coice |
| 19:27.16 | Dorphalsig | I need my internal extensions to be able to dial out the 5 analog lines |
| 19:27.34 | Dorphalsig | they already have access to the E1 |
| 19:27.38 | jbalcomb | [TK]D-Fender overpriced certainly but quality. it allows us to testing for phone specific issues. the SPA-2002 is for our fax machines. |
| 19:27.39 | [TK]D-Fender | jbalcomb : SNOM360 is an interesting choice, but I wouldn't pick it over a Polycom IP501 for general use.... only for a receptionist... |
| 19:27.40 | *** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc) |
| 19:27.57 | jbalcomb | [TK]D-Fender polycoms are not backlit and dont have PoE |
| 19:28.03 | iCEBrkr | jbalcomb: Whut'up foo' |
| 19:28.06 | justinu | the 601 has PoE |
| 19:28.22 | jbalcomb | iCEBrkr sup sucka. hows your little IVR project going? |
| 19:28.33 | jbalcomb | justinu how much is the 601? |
| 19:28.37 | [TK]D-Fender | jbalcomb : Ummm yeah polycom's are PoE, and backlight is very rarte. Its a plus for SNO & GS, but GS= suck, and snom is an iffy topic.... |
| 19:28.38 | *** join/#asterisk zoa (n=zoa@87.215.18.236) |
| 19:28.39 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 19:28.47 | [TK]D-Fender | justinu : all Polys can be PoE |
| 19:28.53 | jbalcomb | 'can be' |
| 19:29.05 | jbalcomb | +++$$$ |
| 19:29.09 | [TK]D-Fender | jbalcomb : Just takes a small adapter cable |
| 19:29.17 | jaiger | Dorphalsig, I don't think I understand what you're trying to do well enough to help. I would guess you need fxo/fxs ports for your asterisk box as well as some way to integrate asterisk with your existing E1+CB |
| 19:29.18 | justinu | yeah, but the 601 is true PoE with no extra money. |
| 19:29.27 | justinu | no special cables |
| 19:29.31 | [TK]D-Fender | well when you're done adding it to the price, its pretty identical to the SNOM. |
| 19:29.34 | sevard | adapters are cheaper than a all in one solution, in my research |
| 19:30.05 | jbalcomb | especially if you make the cables yourself. |
| 19:30.06 | Dorphalsig | jaiger: --> I just need to use 5 analog lines with my asterisk installation, I have two free ports in my TE400P |
| 19:30.29 | pifiu | hey everyone |
| 19:30.29 | jaiger | Dorphalsig, then you would need another CB with fxs/fxo for your 5 ports |
| 19:30.31 | Dorphalsig | so I wonder: Should I just grab the lines, make a small adapter for them to fit an RJ45 jack |
| 19:30.37 | iCEBrkr | jbalcomb: Slow I guess.. I really need to stress test this thing |
| 19:30.44 | iCEBrkr | jbalcomb: and 'predictive' dialing is a bitch |
| 19:30.54 | jaiger | Dorphalsig, no you cant put analog phone lines on a digital card (TE400P) |
| 19:31.08 | jbalcomb | [TK]D-Fender justinu SNOM 630 = $199; polycom IP 601 = $249 |
| 19:31.18 | Dorphalsig | jaiger: --> I have 5 free channels in my channelbank ... so I would just need to get the lines into the cb and define those channels with the appropiate signalling |
| 19:31.32 | justinu | snom360 at $199 isn't a bad deal |
| 19:31.43 | jbalcomb | agreed |
| 19:31.44 | justinu | but you can get poly501's with the poe cable for less than that. |
| 19:32.03 | justinu | depends on what you want to do... i'd be inclined to recommend the 501s for most people over the 360 |
| 19:32.08 | jbalcomb | IP 501 + PoE cable = $209 |
| 19:32.17 | jbalcomb | plus no backlight |
| 19:32.17 | justinu | from where? |
| 19:32.21 | jbalcomb | atacomm |
| 19:32.25 | justinu | hmm |
| 19:32.41 | jaiger | Dorphalsig, yeah I guess so. I have a T100P hooked into a CB with 4 FXO to the PSTN that I use here |
| 19:33.05 | jbalcomb | more importantly, i definitely appreciate the advice but it needs to come with logic, reason, and evidence of the preference |
| 19:33.19 | jbalcomb | so why the IP 501 over the 360B? |
| 19:33.22 | justinu | well, i own both phones |
| 19:33.32 | justinu | i've used them both fairly extensively |
| 19:33.41 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 19:33.53 | justinu | the 501 sounds better, and people seem to like it more |
| 19:33.53 | jaiger | is your existing CB hooked into your TE400P? |
| 19:33.57 | jbalcomb | just the FAQs ma'am, just the FAQs. |
| 19:34.01 | justinu | most people I know aren't happy with the aesthetics of the Snom |
| 19:34.06 | justinu | but can't give me a real reason why |
| 19:34.09 | cypromis | and the sound |
| 19:34.28 | justinu | the 501 is universally accepted, imo |
| 19:34.40 | cypromis | yah |
| 19:34.46 | jbalcomb | what about the sound? less static, less audio cut outs, less echo, less feedback, etc? |
| 19:35.02 | cypromis | they implemented into the 500's and 600's the stuff from their large conference phones |
| 19:35.03 | justinu | yeah - feel like the snom doesn't have a very good jitter buffer (or any?) or PLC |
| 19:35.04 | sevard | How does one change/designate language sets for festival in the dialplan? |
| 19:35.05 | rajiv|work | justinu: you ever compare a 501 to a spa-942 ? |
| 19:35.07 | cypromis | so it sounds perfect |
| 19:35.13 | justinu | rajiv|work: sorry no |
| 19:35.19 | justinu | i have an 841 tho |
| 19:35.45 | jbalcomb | iCEBrkr what is 'predictive' dialing? |
| 19:35.57 | justinu | i've used g711 on the 501 over some really really terrible IP links, and was floored by the audio quality. |
| 19:35.58 | [av]bani | rajiv|work: 942 would never come even close to a 501 |
| 19:35.58 | Weezey | I wanna try one of those 942s. |
| 19:36.00 | iCEBrkr | jbalcomb: It's gotta throttle the number of calls |
| 19:36.12 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 19:36.18 | [av]bani | Weezey: 942 is too little, too late. |
| 19:36.23 | iCEBrkr | jbalcomb: I get 5000 numbers to dial in a 13.5hr period.. It's supposed to spread them out over 13.5hrs |
| 19:36.28 | justinu | avbani: agreed |
| 19:36.30 | Weezey | how's the sound quality? |
| 19:36.35 | [av]bani | it's just a 941 with two 10mb ethernets. not even 100mb. no new features. |
| 19:36.41 | Weezey | 10mb! |
| 19:36.43 | Weezey | useless |
| 19:36.45 | justinu | lol |
| 19:36.47 | jbalcomb | iCEBrkr ah, yes, i remember talking about that. |
| 19:36.51 | justinu | even the gxp2000 has 100mbit! |
| 19:36.56 | justinu | wtf? |
| 19:36.58 | [av]bani | for passthrough, it's important. unless you want your desktop pc to be 10mb ether |
| 19:37.01 | Netgeeks | a predicitve dialer uses historical data to predict the number of calls it needs to make to fully load an available bank of agents, including accounting for no answers, fax/voicemail, etc. |
| 19:37.18 | [TK]D-Fender | jbalcomb : Well comparatively the 601 beats the SNOM in pretty much everything except presence support right now, and for the price, I have found the 601 for $240. Then again, if you don't need the extra line appearances, the 501 w/ poe saves some cash. |
| 19:37.19 | iCEBrkr | Netgeeks: Yea, and it's a pain in the ass to code |
| 19:37.20 | justinu | [av]bani: a lot of my clients are pissed about the fact that there's no GbE passthru on these phones. |
| 19:37.21 | Weezey | [av]bani: exactly. I'm lovin' the 79xx |
| 19:37.38 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 19:37.44 | [av]bani | passthrough 10mb is pretty useless. |
| 19:37.47 | Netgeeks | it's only predictive if it has the ability to predict the number of calls needed to be made to fully load but not overload the available agent base, else it's an autodialer |
| 19:37.53 | Netgeeks | i know technicalities |
| 19:38.14 | iCEBrkr | Netgeeks: Well, my deal doesn't have any agents to deal with. |
| 19:38.16 | Netgeeks | i know,... technicalities... is the way that should be read |
| 19:38.25 | [av]bani | for $179 for the spa-942, you'd be better off buying the CHEAPER polycom 501 which has two 10/100 and far better sound quality |
| 19:38.31 | iCEBrkr | Netgeeks: I just have to keep the lines full and spread the numbers over a period of time |
| 19:38.42 | jbalcomb | [TK]D-Fender 'pretty much everthing' is vague and ambiguous. i dont know how things work for everyone else but in my world that is weak and useless. |
| 19:38.53 | *** join/#asterisk loick_ (n=loick@APuteaux-151-1-54-147.w82-120.abo.wanadoo.fr) |
| 19:38.56 | Netgeeks | ah, i didn't see the lead-in ice, just saw the question |
| 19:38.59 | jbalcomb | [TK]D-Fender the 501 w/PoE is $10 more at atacomm |
| 19:39.00 | [av]bani | spa-942: $179. polycom 501. $169. |
| 19:39.18 | Netgeeks | i'm only halfway here... looking over now and then and catching snippets of conversations |
| 19:39.26 | iCEBrkr | Netgeeks: Most of us are :P |
| 19:39.30 | jbalcomb | [av]bani if you need PoE its +$40 |
| 19:39.46 | [av]bani | [TK]D-Fender: the 601 beats the snom in everything? not imo. and i have a snom 360 and a polycom 601. you only have the 601 so you have no real life basis for comparison :) |
| 19:40.03 | [TK]D-Fender | [av]bani : 942? Shudder... crappy appearance and call control support..... Not worth the money... |
| 19:40.09 | Netgeeks | the 360 looks nice.... |
| 19:40.09 | justinu | pretty much everything |
| 19:40.14 | justinu | i'd have to agree with him |
| 19:40.20 | justinu | snom has a nice web interface tho |
| 19:40.22 | Netgeeks | i really ought to get a 360 and sidecar to play with |
| 19:40.24 | justinu | if that's important for you |
| 19:40.24 | [av]bani | i can tell you the warts of the snom 360, fender can't because he doesnt have one :) |
| 19:40.37 | [av]bani | :D |
| 19:40.39 | justinu | snom boots up pretty fast |
| 19:40.44 | [av]bani | yes it does |
| 19:40.49 | hardwire | anybody used sql vuiews for static configs? |
| 19:40.51 | _Sam-- | get a gxp :P |
| 19:40.53 | hardwire | views :) |
| 19:41.00 | justinu | snom has lots of programmable buttons |
| 19:41.04 | jbalcomb | I have no comprehension of how this discussion is continuing and not one person has stated actually specific features or performance that is the basis for the decision. |
| 19:41.19 | Netgeeks | polycom definately has the title for slowest booting phone i've ever seen |
| 19:41.22 | [av]bani | jbalcomb: welcome to #asterisk. enjoy your stay. |
| 19:41.31 | hardwire | Netgeeks: I have a 360 + Sidecar |
| 19:41.37 | hardwire | its not that fun |
| 19:41.53 | hardwire | Netgeeks: and I just pought an ip4000 polycom |
| 19:41.53 | Netgeeks | hardwire, can i borrow it for a week?!? :) |
| 19:41.59 | hardwire | it takes like 5 years to boot |
| 19:42.00 | [TK]D-Fender | Netgeeks : If you have to reboots your phones often enough to care, then AMYBE its not a feature so much as a NECESSITY :) |
| 19:42.04 | hardwire | Netgeeks: aren;t you .nl? |
| 19:42.12 | Netgeeks | nope, Oregon |
| 19:42.21 | Netgeeks | but according to fed ex, might as well be .nl |
| 19:42.22 | hardwire | walk on up to alaska and snag one |
| 19:42.33 | [av]bani | [TK]D-Fender: it's a polycom wart. like the firmware issue and polycom non-support. |
| 19:42.49 | justinu | snom MWI integration was a little funky |
| 19:42.55 | justinu | but do-able |
| 19:42.57 | [av]bani | justinu: and blf |
| 19:43.00 | Netgeeks | fed ex, ups, and USPS all won't offer overnight shipping to here... bah |
| 19:43.05 | jbalcomb | [av]bani BTW, I like your hard sip phones page. |
| 19:43.07 | justinu | blf seemed easy to me |
| 19:43.07 | _Sam-- | [av]bani: y ou use blf on the gxp? |
| 19:43.14 | hardwire | justinu: how do you mean its funky? |
| 19:43.16 | [av]bani | _Sam--: not yet, but i plan to |
| 19:43.20 | _Sam-- | i was going to try it today |
| 19:43.23 | [TK]D-Fender | [av]bani : Yeah, I suppose they could be a bit nicer about it, but it still gets the job done and any decent reseller will give it to you pronto, like Atacom, and mine have.... |
| 19:43.24 | jbalcomb | [av]bani did you request the GS feature for different volume control for handset and speaker? |
| 19:43.25 | hardwire | you mean the vm exten it uses? |
| 19:43.33 | justinu | hardwire: i can't remember the exact details, just that it took me longer to get it going than polycom |
| 19:43.40 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 19:43.41 | rajiv|work | [av]bani: 10mb on the 942? that is useless. where did you see that |
| 19:43.41 | hardwire | justinu: its really really easy |
| 19:43.44 | hardwire | you set the mailbox. |
| 19:43.47 | hardwire | and you are done |
| 19:43.51 | [av]bani | [TK]D-Fender: the problem is, that makes "correct vendor choice" part of the polycom purchase. which is a wild variable. |
| 19:44.12 | [av]bani | [TK]D-Fender: some people won't like that risk. your polycom support depends on the vendor not being shit. |
| 19:44.14 | Netgeeks | I'm still quite tempted to buy a 360 and play with it... |
| 19:44.28 | [av]bani | and not going under after you buy the phone... it also makes it nigh impossible to get support for an ebay phone. |
| 19:44.40 | [av]bani | Netgeeks: what are your parameters? |
| 19:44.40 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 19:44.41 | [TK]D-Fender | [av]bani : Then again you can always just "sign on" with another reseller just for support. But it COULD be a PITA. Then again so could buying a "commercial" PBX. |
| 19:44.58 | justinu | heh |
| 19:44.58 | [av]bani | [TK]D-Fender: or you could come to #asterisk and beg for warez |
| 19:44.59 | Netgeeks | bani: curiosity |
| 19:45.03 | [TK]D-Fender | [av]bani : Keep in mind you cans till get a revision behind publicly. |
| 19:45.03 | Netgeeks | nothing more |
| 19:45.08 | [av]bani | Netgeeks: ah. then go right ahead :) |
| 19:45.14 | [av]bani | [TK]D-Fender: still sucks :) |
| 19:45.26 | rajiv|work | everyone here seems to talk about the 501 a lot. i'll look at it... |
| 19:45.38 | [TK]D-Fender | [av]bani : I still wouldn't base an entire decision on it. Add up the bits and pieces. |
| 19:45.45 | [av]bani | rajiv|work: if you have a choice, the 601 is far better. nicer lcd and xml. |
| 19:45.45 | EriSan | does anyone have an updated manual for a HandyTone 386? Alot of settings are not explained on the Grandstream website |
| 19:45.49 | _Sam-- | bani how do you get your calls to PSTN? |
| 19:45.52 | [TK]D-Fender | [av]bani : depends on the overall image I guess. |
| 19:46.23 | [TK]D-Fender | Netgeeks : I too have been looking at the 360 funny.... just waiting till I open my company and can write it off :) |
| 19:46.24 | rajiv|work | [av]bani: whats the diff on the lcd ? i think in the end it will come down to price |
| 19:46.44 | Netgeeks | For my personal phone, I have pretyt much one overriding parameter: it must support a high quiality headset with active noise cancellation |
| 19:47.02 | [av]bani | rajiv|work: the 601 has double the rez of the 501, and the 601 supports xml microbrowser, which is a huge feature. for some braindamaged reason polycom won't add xml to the 501. |
| 19:47.18 | nibbler_ | Netgeeks: does the HBH-300 count? |
| 19:47.27 | Netgeeks | I currently have a 7960 and a TuffSet 100, but my 7960 has been getting sick lately and I can't figure out why.. may be time for a replacement |
| 19:47.31 | [av]bani | _Sam--: spa-3000, but i'm disappoitned enough with the echo canceller i'm looking into other options |
| 19:47.31 | MstlyHrmls | [av]bani: have they said they won't, or have they just not gotten around to it yet? |
| 19:47.33 | [TK]D-Fender | Netgeeks : Thats a real problem with most. For call-centers you're pretty much guaranteed to want a seperate amplified headset solution like Plantronics.... |
| 19:47.38 | [av]bani | MstlyHrmls: they have said they won't. |
| 19:47.53 | _Sam-- | [av]bani: let me know what you find |
| 19:48.01 | _Sam-- | i think i am going to need 8 fxo |
| 19:48.05 | [av]bani | _Sam--: what do you use for pstn? |
| 19:48.20 | _Sam-- | right now all of my clients and my company use remote gateways |
| 19:48.21 | [av]bani | hmm, then you might like this new page :) http://bani.anime.net/gateways/ |
| 19:48.22 | _Sam-- | i had a PRI here |
| 19:48.23 | [av]bani | <3 |
| 19:48.23 | [TK]D-Fender | _Sam-- : A200 w/ HWEC :D |
| 19:48.48 | [av]bani | [TK]D-Fender: nobody has one yet. no guarantees it actually works for shit... |
| 19:48.51 | Netgeeks | ewww, no, HBH-300 not a good headset for me |
| 19:49.04 | [av]bani | i'll wait till someone on the ML posts a review |
| 19:49.10 | g4m | has anyone had problems with ztdummy and asterisk on a 64 system? |
| 19:49.20 | *** join/#asterisk HeyEveryBody (n=Aces1Up@ip70-189-157-31.lv.lv.cox.net) |
| 19:49.30 | MstlyHrmls | [av]bani: Interesting, I wonder why. with 4 Megs they've got the room... |
| 19:49.38 | Netgeeks | pretty much has to be a wired headset... some days I'm on it for 10 hours straight |
| 19:49.55 | [av]bani | MstlyHrmls: i'm guessing they don't want to undercut their 601 sales. still sucks though. |
| 19:50.26 | [av]bani | for instance, aastra supports xml across their entire product line, and they only have dinky character LCDs. |
| 19:50.41 | jaiger | [av]bani, what do you use the xml for? I haven't found a use yet although I'm interested to play with it |
| 19:50.46 | [av]bani | they support xml even on phones where it doesnt really make sense :) |
| 19:51.00 | [TK]D-Fender | Netgeeks : I just picked up some Plantronics H101 headsets and M12 amplifiers for my call-center here... they are very happy with them... very noisy envinment and the binaural headset is a huge plus. |
| 19:51.10 | [av]bani | jaiger: planning to use it to monitor queue status, let supervisors override lines, etc. |
| 19:51.22 | _Sam-- | [TK]D-Fender: do you have a URL where i can read about the A200 |
| 19:51.25 | Netgeeks | So far I have to say I love my TuffSet 100.. it's got a quick disconnect, so you can step away without taking the headset off, and it works in the 7960's headset jack.. you can also get a USB quick-disconnect adapter for it and use it with a softphone, so I can travel with it as well |
| 19:51.34 | HeyEveryBody | what is a binaural headset? |
| 19:51.34 | jaiger | [av]bani, that's probably the only use I've come up with - watching queues & status |
| 19:51.37 | MstlyHrmls | [av]bani: maybe... Who did you hear this from, if you don't mind my asking? |
| 19:51.39 | [TK]D-Fender | _Sam-- : www.sangoma.com or for sale at www.voipsupply.com |
| 19:51.42 | Netgeeks | binaural = covers both ears |
| 19:51.43 | [av]bani | jaiger: browsing pr0n? |
| 19:51.45 | _Sam-- | ty |
| 19:51.55 | jaiger | [av]bani, it does graphics? |
| 19:51.59 | HeyEveryBody | ahh, thanks. |
| 19:51.59 | [av]bani | yes |
| 19:52.08 | jaiger | didn't know that |
| 19:52.35 | [av]bani | [TK]D-Fender has some xml apps he uses on his 601's, but the screenshots he gave me were incomprehensible |
| 19:52.39 | [TK]D-Fender | jaiger : Thats what I do with mine.... I provide full-company presence support, Queue /VM stats, and announcements over Polycom IP60x XML. |
| 19:53.04 | _Sam-- | its the A2000 you are talking about, not A200? |
| 19:53.13 | [TK]D-Fender | [av]bani : Its my Cell-camera, what did you expect? :) |
| 19:53.35 | [TK]D-Fender | _Sam-- : A200 series (different suffixes depending on type and density) |
| 19:53.38 | [av]bani | i think you were trying to mislead me :) |
| 19:53.45 | _Sam-- | they are linked together like daughter cards? |
| 19:53.47 | [av]bani | give me blurry photos to cover up the real purpose |
| 19:53.48 | _Sam-- | just one PCI? |
| 19:53.54 | [TK]D-Fender | [av]bani : Lemme see if I can get a better camera here... |
| 19:54.07 | [TK]D-Fender | _Sam-- : Yup |
| 19:54.11 | _Sam-- | i guess i need to open my eyes, it says that right in front of me |
| 19:54.43 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 19:54.47 | [av]bani | [TK]D-Fender: i expect better pics! |
| 19:54.55 | _Sam-- | what is the advantage, or why do you like that better, than say a digium card with ec |
| 19:55.06 | [av]bani | _Sam--: the software ec is poo |
| 19:55.12 | _Sam-- | i see |
| 19:55.30 | [av]bani | no way they can come close to a real hw ec |
| 19:55.39 | _Sam-- | so the tdm2400 doesnt use hw ec? |
| 19:55.51 | sevard | sed magic! |
| 19:55.52 | justinu | it's optional, iirc |
| 19:56.02 | [av]bani | it has ec option, but the reviews i've read have not been favorable |
| 19:56.35 | [TK]D-Fender | [av]bani : Working on it but its hard to filter out reflections on the LCD.... |
| 19:56.46 | jaiger | I've found the software EC to be junk too. Been using ATAs & gateways instead of zaptel |
| 19:57.02 | [TK]D-Fender | _Sam-- : TDM2400 and A200 can both have EC, so just compare the specs. |
| 19:57.05 | [av]bani | to be fair, ec is _hard_ |
| 19:57.26 | _Sam-- | is ec relatively new, or why is it so un-developed |
| 19:57.35 | [av]bani | i wish digium would just license a g.168 ec and be done with it, like they license g729 |
| 19:57.52 | justinu | i had a $150,000 ditech triple DS3 echo cancellor |
| 19:57.57 | justinu | and it still sucked :P |
| 19:58.12 | [av]bani | tellabs! |
| 19:58.16 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 19:58.50 | _Sam-- | [TK]D-Fender: does A200 use something different from zaptel modules? |
| 19:58.57 | _Sam-- | or zaptel runs that too |
| 19:59.02 | justinu | patched zaptel, iirc |
| 19:59.17 | jaiger | I have some ebay tellabs EC that work OK but still leave some echo on the line |
| 19:59.24 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 20:00.05 | [av]bani | _Sam--: the a200 uses the same EC they use on their PRI cards |
| 20:00.24 | [TK]D-Fender | _Sam-- : And lets say that echo is a word that doesn't exist here :) |
| 20:01.10 | _Sam-- | its all configured in zapata.conf and zaptel.conf? |
| 20:01.22 | [TK]D-Fender | _Sam-- : All Sangoma cards integrate just like Zaptel with a few extra little steps |
| 20:01.23 | _Sam-- | <aside from extension related stuff obviously> |
| 20:01.36 | [TK]D-Fender | Yes, std Zaptel & zapata. |
| 20:01.52 | _Sam-- | just making sure, i dont want to get in over my head with stuff ive never worked with...sounds doable. |
| 20:02.03 | [TK]D-Fender | _Sam-- : I've used both kinds.... |
| 20:02.08 | *** join/#asterisk tris- (i=tristan@camel.ethereal.net) |
| 20:02.22 | justinu | the sangoma wanpipe drivers are super easy to install |
| 20:02.25 | justinu | very nice setup |
| 20:02.48 | rajiv|work | [av]bani: could you add the digium tdm400p to that page and maybe put it all on the wiki? |
| 20:03.39 | [TK]D-Fender | rajiv : It doesn't count as a Gateway.... its a CARD. |
| 20:03.49 | _Sam-- | lol |
| 20:03.57 | _Sam-- | the iaxy |
| 20:04.02 | _Sam-- | that could get on there |
| 20:04.53 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 20:05.12 | mogorman | a card can be a gateway... ^_^ |
| 20:05.31 | _Sam-- | the page title should be EXTERNAL gateways |
| 20:05.35 | [TK]D-Fender | _Sam-- : Not by the definition he's working by (needs FXO) |
| 20:05.36 | [av]bani | yay |
| 20:05.48 | *** join/#asterisk [dc] (n=dc@24-205-223-175.dhcp.slto.ca.charter.com) |
| 20:05.54 | _Sam-- | oo i c |
| 20:06.12 | [TK]D-Fender | [av]bani : And you need to fix the price column for that D-LInk |
| 20:06.17 | [av]bani | ? |
| 20:06.57 | [dc] | anybody use aah to connect to FWD and have trouble dialing toll free #'s ? |
| 20:07.22 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 20:07.35 | Nivex | [dc]: FWD IAX has been down for over a week, so I haven't been placing any calls. |
| 20:07.49 | [TK]D-Fender | [av]bani : And you should add Clipcomm's 4port FXO gateway to your list @ $399.95 |
| 20:07.59 | [dc] | ahhh... what is the proper dial pattern to reach toll free #'s via FWD (assuming it were back up?) |
| 20:08.07 | [dc] | my route pattern is set to 393|X. |
| 20:08.17 | [dc] | so 393xxxxx conns me to whatever FWD # i call |
| 20:08.20 | [av]bani | [TK]D-Fender: have a url? |
| 20:08.27 | [dc] | but if i do 39318005551212 for example i get circuits busy |
| 20:08.38 | [dc] | and if i do 393*18005551212 i get 404 address incomplete |
| 20:08.51 | [dc] | (tho fwd's docs say you need to inject the * in there to get access to their toll free gateways) |
| 20:09.04 | justinu | try ** |
| 20:09.39 | [dc] | call failed 404 address incomplete :( |
| 20:09.49 | justinu | i had similar issues with FWD |
| 20:09.58 | justinu | it worked for a while, then stopped |
| 20:10.47 | [dc] | werd |
| 20:11.01 | jaiger | [av]bani, http://www.voipsupply.com/product_info.php?cPath=286_120&products_id=241&osCsid=faa35dead7aeffe5db3776cd292ae05a |
| 20:11.22 | [av]bani | imean the vendor's site |
| 20:11.32 | justinu | :google: |
| 20:11.36 | justinu | clipcomm.com? :P |
| 20:11.45 | [av]bani | clipcomm.com <- domain squatters |
| 20:11.49 | justinu | hmm |
| 20:11.53 | justinu | suckas |
| 20:11.56 | [av]bani | :) |
| 20:11.57 | *** join/#asterisk blop (i=blop@openbeer.be) |
| 20:12.10 | jaiger | [TK]D-Fender, have you used the clipcomm gw? I've been looking for a 4port gw for a while |
| 20:12.13 | Flyboy-SR22 | http://www.clipcomm.co.kr/ |
| 20:12.24 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 20:12.29 | justinu | korean, eh? |
| 20:12.31 | Netgeeks | domain squatters are scum |
| 20:12.38 | [av]bani | jaiger: afaik the best fxo gw used with * so far is the mediatrix 1204. that is, it has a very good EC |
| 20:12.48 | [av]bani | jaiger: but the management, like the polycom 601, is ass |
| 20:12.49 | Beirdo | they should all be slapped silly |
| 20:12.57 | [av]bani | Netgeeks: criminals are everywhere |
| 20:13.01 | Beirdo | domain squatters su-diddly-uck |
| 20:13.05 | Netgeeks | criminals are scum1 |
| 20:13.11 | Netgeeks | scum! |
| 20:13.31 | Beirdo | and some become politicians, which is a special type of scum :) |
| 20:13.32 | jaiger | [av]bani, I bought a 1204 and couldn't configure it to a customer's needs. I haven't had a chance to revisit it |
| 20:13.54 | jaiger | as you point out the config is ass |
| 20:14.01 | [av]bani | jaiger: it's a pita afaict, but from what i understand it can be beaten into submission |
| 20:14.15 | [av]bani | and it works |
| 20:14.39 | *** join/#asterisk ToTo (n=ToTo@host46-49.pool870.interbusiness.it) |
| 20:14.40 | *** join/#asterisk CloseCall (n=borgirc-@dongma.xs4all.nl) |
| 20:15.18 | *** join/#asterisk Pinston (n=ejo@87.252.72.16) |
| 20:15.23 | *** join/#asterisk sigmounte__ (n=sigmount@www.sighq.net) |
| 20:15.31 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
| 20:15.32 | jaiger | [av]bani, I had an electrical engineer look at the board and he says it's a standard off-the-shelf design so I would expect it to work IF you can configure it |
| 20:15.57 | [av]bani | there, clipcomm added |
| 20:15.58 | mdave | ok.. im pounding my head on the wall here.. |
| 20:16.07 | mdave | i have a bv account, and inbound calling *was* working to my * box |
| 20:17.09 | _Sam-- | mdave: sip show registry |
| 20:17.09 | CloseCall | hi |
| 20:17.09 | mdave | but now, it wont complete calls.. I see the sip messages coming in |
| 20:17.09 | _Sam-- | make sure you are registered to broadvoice |
| 20:17.09 | mdave | and one of the things im seeing coming from my end is '404 not found' |
| 20:17.10 | *** join/#asterisk sancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 20:17.10 | CloseCall | how can i check if asterisk is running ? |
| 20:17.10 | Flyboy-SR22 | asterisk -rvvvvv |
| 20:17.10 | jaiger | CloseCall, ps auxw | grep ast |
| 20:17.13 | mdave | outbound calls work fine |
| 20:17.21 | mdave | and i am registered |
| 20:17.25 | Flyboy-SR22 | or /etc/inin.d/asterisk status |
| 20:17.30 | Flyboy-SR22 | opps |
| 20:17.35 | mdave | its like bv is trying to send the call to *, but * isnt able to accept it, and I cant see why |
| 20:17.38 | Flyboy-SR22 | or /etc/init.d/asterisk status |
| 20:17.44 | _Sam-- | *yawn* |
| 20:17.54 | CloseCall | i dont have a init for asterisk ? |
| 20:18.08 | CloseCall | and when i run asterisk -rvvv i get: |
| 20:18.17 | CloseCall | [root@Shana ~]# asterisk -rvvvv |
| 20:18.17 | CloseCall | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 20:18.17 | justinu | mdave: 404 indicates a problem with your contexts/dialplan |
| 20:18.22 | CloseCall | [root@Shana ~]# asterisk -rvvvv |
| 20:18.22 | CloseCall | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 20:18.30 | Flyboy-SR22 | [av]bani, on the mediatrix, I could have one of those a 1000 miles from my * box and still route calls to it and out the pots line attacehed to it correct..? |
| 20:18.40 | mdave | justinu, the dp is the dead simplest, same as it was |
| 20:18.46 | mdave | i had made some changes, but reverted them |
| 20:18.57 | [av]bani | Flyboy-SR22: yes, thats the point :) |
| 20:19.04 | mdave | [frombv] |
| 20:19.04 | mdave | exten => s,1,Dial(SIP/phone1,25,Ttrw) |
| 20:19.04 | mdave | exten => s,2,Hangup |
| 20:19.08 | Flyboy-SR22 | I am looking for a 911 solution, so I want to be able to grab a 911 call and redirect it back to a FXO gateway and out a local pots line |
| 20:19.09 | mdave | phone1 is a valid extension |
| 20:19.15 | Flyboy-SR22 | soulds like it will work for that.. |
| 20:19.28 | mdave | er.. should I remove the hangup |
| 20:19.34 | mdave | hrm |
| 20:19.53 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 20:20.25 | *** join/#asterisk chops (n=moise@146-115-127-60.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
| 20:20.33 | mdave | ok removing the hangup didnt change anything |
| 20:20.38 | *** join/#asterisk lookatthis (i=lookatth@209.250.133.54) |
| 20:21.08 | _Sam-- | great...no incoming or outgoing service from teliax. |
| 20:21.31 | mdave | what is it that its looking for thats 'not found' ? |
| 20:21.34 | justinu | mdave: then the context might be wrong in sip.conf |
| 20:21.40 | rajiv|work | _Sam--: ya see that is what i was concerned about yesterday |
| 20:21.41 | justinu | the dialed digits |
| 20:21.46 | mdave | nope.. context is 'frombv' |
| 20:21.51 | mdave | and that is the name of the context |
| 20:21.51 | _Sam-- | darwin_35: get of your ass and fix some shit |
| 20:21.59 | mdave | and no digits are dialed |
| 20:21.59 | mdave | I have |
| 20:22.38 | mdave | register => mynumber@@sip.broadvoice.com:mypassword:mynumber@@sip.broadvoice.com/s |
| 20:22.46 | mdave | er |
| 20:22.51 | mdave | the double @ is a typo pasting |
| 20:22.56 | mdave | it has one in the config |
| 20:23.15 | *** join/#asterisk _blop (i=blop@openbeer.be) |
| 20:23.24 | mdave | so how can I enable some sort of debugging that will show me what its doing? |
| 20:23.29 | mdave | the sip debug isnt telling me much |
| 20:23.50 | *** join/#asterisk Trazzz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 20:24.19 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 20:24.38 | MattH | Hi.... I have an ATA that I have setup (As far as I can tell) correctly but all Asterisk shows me is: "chan_sip.c: Auto destroying call" any thoughts? |
| 20:24.41 | justinu | paste the sip debug, and i'll point out the line you need to pay attention to. |
| 20:24.41 | lookatthis | these absolutely have to go today. 2 alienware area51-m 5700 laptops $550 each includes shipping, case, wireless router and 1 alienware area51 7500 desktop $700 includes monitor, keyboard, mouse,. speakers. message me if you wnat to buy any of these items at mcsltd@telusmail.net, aim at ogd443 or yahoo at mcsltd2 thanks and have a good day. |
| 20:24.55 | CloseCall | can anyone tell me why i dont have a /etc/init.d/asterisk ? |
| 20:25.00 | justinu | hey, everyone... spam aim ogd443! |
| 20:25.27 | *** join/#asterisk ocnarfid9 (n=ocnarfid@207.34.36.50) |
| 20:25.52 | Flyboy-SR22 | CloseCall - depends on what version or vendor of linux your running |
| 20:26.49 | chops | Hi! I'm another newbie trying to set up SIP with asterisk. It's, er, not working. Here's what it does: |
| 20:27.23 | *** join/#asterisk Cresl1n (n=matt@146.229.184.109) |
| 20:27.28 | Flyboy-SR22 | CloseCall - I run Gentoo so my init scripts are located in /etc/init.d |
| 20:27.28 | mdave | CloseCall, becuase you didnt put on there? |
| 20:27.48 | Flyboy-SR22 | CloseCall - RedHad puts them in /etc/rc.d/init.d as I recall |
| 20:28.08 | }btorch{ | does the TE110P card work on a 64 bit slot ? |
| 20:28.09 | Flyboy-SR22 | And of course, like mdave points out - you may have to create them and put them there :-) |
| 20:28.11 | chops | I dial my softphone number (using vonage). ethereal shows an incoming sip INVITE, but my asterisk returns "Status: 404 Not Found" to the invite. I go to vonage voicemail. |
| 20:28.29 | chops | 'asterisk -vvvvdddd' says only "REGISTER attempt 1 to 1xxxxxxxxxx@sphone.vopr.vonage.net" as this is going on; it doesn't seem to give me any information about the incoming call failing. |
| 20:28.32 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 20:28.33 | }btorch{ | I have one installed on a desktop 32bit 5v slot |
| 20:28.41 | chops | Does anyone have any advice for me? |
| 20:28.51 | g4m | correct me if i'm wrong but i dont think you can get asterisk to work with vonage |
| 20:28.56 | g4m | broadvoice is easy though |
| 20:29.15 | Assid | i think you can use vonage.. IF you register for the softphone version of vonage.. |
| 20:29.28 | chops | g4m: http://www.voip-info.org/wiki/view/Asterisk+and+Vonage claims that it can work (not with the default account setup, but with an optional extra feature of vonage). |
| 20:29.28 | Assid | but oyu need to be an existing customer.. first |
| 20:29.31 | sevard | I have a quick question, when I text2wave a file it is a nice and soft voice, But in my dialplan when I Festival(some stuff here) it screams "SOME STUFF HERE!!!!" |
| 20:29.53 | mdave | http://jupiter.microwave.com/sipdebug.txt |
| 20:30.26 | justinu | Looking for mynumber in frombv (domain my.ip.address) |
| 20:30.36 | *** join/#asterisk Abbas__ (n=Abbas@203.81.196.140) |
| 20:30.36 | justinu | you're saying "mynumber" exists in frombv? |
| 20:30.44 | [av]bani | anyone hacked packet8 to work with * ? |
| 20:30.45 | mdave | justinu, before it was matching 's' ? |
| 20:30.51 | mdave | why would it not do that now? |
| 20:31.06 | justinu | mdave: not sure, but i had issues with matching s also |
| 20:31.19 | mdave | well lemme add it there and see if that changes anything |
| 20:31.23 | mdave | it *was* working with s before |
| 20:32.23 | *** join/#asterisk sthw45ywyw5 (n=sthw45yw@38.136.42.2) |
| 20:33.07 | mdave | fscking bv.. now its not even trying to pass the call to me |
| 20:33.35 | _Sam-- | i have some bv testing accounts, working fine here. |
| 20:33.42 | justinu | To: "My Name"<sip:s@my.ip.address;user=phone> |
| 20:33.44 | mdave | either that or bv isnt accepting it from pstn |
| 20:33.45 | justinu | that looks suspect |
| 20:33.48 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 20:33.54 | mdave | thats sedified |
| 20:34.12 | mdave | and the 's' should be matching the 's' shouldnt it? |
| 20:34.56 | mdave | im gonna go turn bv's voicemail back on so I can see if the call is even getting to bv |
| 20:35.05 | badboyz | has anyone here developed a solution to monitoring the health of multiple * servers? |
| 20:35.16 | justinu | mdave: the problem is this: |
| 20:35.21 | justinu | INVITE sip:mynumber@my.ip.address:5060 SIP/2.0 |
| 20:35.24 | justinu | To: "My Name"<sip:s@my.ip.address;user=phone> |
| 20:35.35 | _Sam-- | rajiv: just for reporting purposes...i shot the teliax guy an instant message via AOL IM and my service was fixed within 1 minute. |
| 20:35.48 | [av]bani | _Sam--: the teliax guy isnt answering my IMs :( |
| 20:35.51 | _Sam-- | but it was definitely broken |
| 20:35.57 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
| 20:36.02 | mdave | justinu, ok, but what in the *config* is the problem? |
| 20:36.02 | _Sam-- | im talking to him now, want me to mention anything? |
| 20:36.09 | [av]bani | yeah, answer his IMs |
| 20:36.14 | mdave | i can sanitize that and paste it, if it would help |
| 20:36.22 | justinu | mdave: i think it's an issue on bv's side, actually. |
| 20:36.23 | _Sam-- | i dont think he counted on the public being able to IM him at will :) |
| 20:36.29 | [av]bani | :) |
| 20:36.37 | _Sam-- | what was the main issue? |
| 20:36.46 | _Sam-- | i will tell him my friend had an issue too, and he may message |
| 20:36.47 | [av]bani | _Sam--: teliax doesn't want my money, though i want to give it to them |
| 20:36.53 | [av]bani | and choppy audio |
| 20:37.08 | _Sam-- | that is what my client is complaining about...why i am asking about 8 port fxo |
| 20:37.11 | mdave | justinu, well at the moment they arent even accepting from the pstn now |
| 20:37.15 | mdave | i just get a reorder tone |
| 20:37.17 | mdave | sigh |
| 20:37.18 | [av]bani | i get 'error 23' on billing, but my cc is 100% ok (junction networks has no problem billing it) |
| 20:37.18 | mdave | fscking bv |
| 20:37.19 | badboyz | i recommend telasip.com personally |
| 20:37.32 | *** part/#asterisk sevard (n=kynan@198.174.233.25) |
| 20:37.38 | mdave | well finally |
| 20:37.41 | mdave | they fixed whatever it was |
| 20:37.43 | [av]bani | badboyz: it's hard to find an ITSP who has local DIDs. so far teliax is the only one |
| 20:37.48 | _Sam-- | my next provider to test is going to be asterlink |
| 20:38.02 | [av]bani | _Sam--: you have choppy audio with teliax too? |
| 20:38.03 | _Sam-- | maybe keep teliax for origination only |
| 20:38.15 | mdave | id like to find one offering a cheap did only, so I can get outbound seperately |
| 20:38.17 | _Sam-- | [av]bani: i dont at 2 locations, but 1 location does, even though the routes are fine, and no packet loss |
| 20:38.21 | mdave | cheap per month, no per-minute |
| 20:38.30 | badboyz | [av]bani: i got local DID's to St Louis from telasip |
| 20:38.36 | [av]bani | yeah, no packet loss, and very low ping. i hear remote callers fine but they say i am choppy |
| 20:38.42 | [av]bani | badboyz: i'm not in a major metro area. |
| 20:38.45 | justinu | mdave: the fact that they're sending you that sip uri s@x.x.x.x |
| 20:38.49 | justinu | indicates they're using asterisk :P |
| 20:38.53 | badboyz | [av]bani: doesnt hurt to email and ask the guy |
| 20:38.55 | justinu | and that it's not working out so well for them |
| 20:38.56 | *** join/#asterisk ast[away] (i=sdsd@85.206.68.100) |
| 20:39.05 | *** part/#asterisk ast[away] (i=sdsd@85.206.68.100) |
| 20:39.18 | rajiv|work | i'm using gizmoproject (sipphone) for origination and they seem to be working okay |
| 20:39.26 | mdave | justinu, maybe * was mapping that? |
| 20:39.31 | mdave | I do have "/s" at the end of the register |
| 20:40.09 | justinu | oh, could be. |
| 20:40.23 | justinu | i'd register with your actual number, and forget about s. |
| 20:40.31 | justinu | that's how i'm doing it with bv, and it works ok |
| 20:40.33 | _Sam-- | i pasted your messages to him |
| 20:40.34 | mdave | yeah, thats what their example had tho |
| 20:40.36 | mdave | i just changed it |
| 20:40.42 | _Sam-- | whether he will respond to me, or you...that remains to be seen :) |
| 20:40.43 | mdave | tried again, but now im getting the reorder again |
| 20:41.08 | _Sam-- | [av]bani: you use co3? voip-co3.teliax.com ? |
| 20:41.10 | mdave | ok.. it went thru again |
| 20:41.47 | [av]bani | _Sam--: yes, they set me for co3 |
| 20:42.05 | _Sam-- | my client with the problem is on co3 also...i never have that problem, and im on co2 |
| 20:42.09 | _Sam-- | coincidence? maybe |
| 20:42.10 | [av]bani | i dont have problems with junction networks, though they are 3x farther |
| 20:42.23 | [av]bani | teliax is 25ms, junction is 90 |
| 20:42.23 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 20:43.27 | [av]bani | i hear remote caller fine, but remote caller says i am choppy |
| 20:43.42 | [av]bani | 0 packet loss |
| 20:44.00 | _Sam-- | i pasted all that to him |
| 20:44.36 | sthw45ywyw5 | how many registrations can I have on a polycom 301? It seems to only let me have 2. Is the # of registrations equal to how many lines the phone can handle? (301 has two lines) |
| 20:44.42 | [av]bani | did you post the error 23 billing problem too? |
| 20:44.46 | _Sam-- | bani, do you want to try co2? |
| 20:44.54 | [av]bani | sthw45ywyw5: yes, polycom 301 only supports 2 |
| 20:44.55 | MstlyHrmls | sthw45ywyw5: yes, that is correct. |
| 20:45.03 | [av]bani | _Sam--: i'll try, if they think it will fix the issue |
| 20:45.09 | joe | anyone running asterisk on CentOS 4.* ? who has a source for clean rpms? |
| 20:45.36 | [TK]D-Fender | sthw45ywyw5 : 301 = 2 reg's with up to 24 calls per line key, 501 =3 regs, 601 = 6 regs |
| 20:45.43 | _Sam-- | maybe shoot him an email from the live chat / support thinger: [15:44] davidcaldworth: if you would like to try them on co2 that can easily be arranged |
| 20:46.16 | BlueDevi1 | joe: what do you meen with "clean rpms" |
| 20:46.35 | [TK]D-Fender | rarely do you end up needing multiple registrations for a given phone. Proper PBX design can reduce the need for these situations enormously. |
| 20:47.03 | [av]bani | [TK]D-Fender: snom 360: 12 regs+ (more with sidecar) :) |
| 20:47.03 | joe | BlueDevi1: that are well made and maintainable :) |
| 20:47.10 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 20:47.26 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 20:47.57 | [TK]D-Fender | [av]bani : Ok... and who needs that many regs? :) |
| 20:48.02 | *** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com) |
| 20:48.12 | joe | BlueDevi1: ie also the src.rpms so I can look and rebuild them... |
| 20:48.15 | [av]bani | [TK]D-Fender: best not to dictate to customer what they can or can't do if it can be done in software |
| 20:48.26 | _Sam-- | [av]bani: |
| 20:48.27 | _Sam-- | [15:47] davidcaldworth: are the audio artifacts on inbound or outbound calls? |
| 20:48.31 | [av]bani | _Sam--: i have an open ticket on the billing |
| 20:48.52 | [TK]D-Fender | [av]bani : You take "purist" to "practically irrelevent" proportions :) Kinds like me, only LESS sane :) |
| 20:48.59 | [av]bani | _Sam--: seems both. i hear remote fine, but remote says i sound choppy. |
| 20:49.01 | sthw45ywyw5 | [TK]D-Fender: What does it mean to have 24 calls per line key. What is a line key |
| 20:49.15 | [av]bani | _Sam--: i can have someone call me and check though. |
| 20:49.28 | Dr-Linux | uffff hard to configure cisco 7940 phone remotely ... if the phone handy person non tech.. |
| 20:49.45 | BlueDevi1 | joe: i have build my own packages...but you can look at http://atrpms.net/dist/el4/asterisk/ |
| 20:49.46 | _Sam-- | i switched my client over to co2 |
| 20:50.26 | [TK]D-Fender | sthw45ywyw5 : The 301 has 2 line keys (buttons on the side). These can both be assiciated to a single registration and have it so that if you're on a call on 1 key it'll ring on the next on an incoming call. You can also have it so it queue's up call-waiting on a single key so you have have up to 24 calls going on assiciated with that only "line-key". and use the other for a different registration altogether. |
| 20:50.35 | _Sam-- | it certainly wont be any worse, in my opinion at least |
| 20:50.48 | _Sam-- | and its 1 hop closer! |
| 20:50.49 | Dr-Linux | [TK]D-Fender: i had configure 9 cisco phones with sip firmware |
| 20:51.11 | [TK]D-Fender | Dr-Linux : And? All went well? |
| 20:51.20 | *** join/#asterisk dumb---me (n=dwayne@64-42-204-117.mb.skyweb.ca) |
| 20:51.24 | Dr-Linux | yeah, all of them are working fine |
| 20:51.36 | [av]bani | _Sam--: checking if origination from teliax has the same issue |
| 20:51.37 | [TK]D-Fender | Dr-Linux : good to hear. |
| 20:51.39 | dumb---me | hi |
| 20:51.42 | Dr-Linux | but this one is messing |
| 20:51.42 | [av]bani | i expect it will though |
| 20:51.55 | Dr-Linux | the girl on the phone is non technical |
| 20:52.05 | dumb---me | does anyone have time to help me out? |
| 20:52.25 | [TK]D-Fender | Dr-Linux : you shouldn't need to ask the user to do too much... at most to reboot the phone to take remotely configured changes.... |
| 20:52.39 | jbalcomb | can any confirm that once the grandstream GXP-2000 is flashed to 1.0.2.3 that there is no going back? |
| 20:52.47 | [TK]D-Fender | dumb---me : Just ask your question and see who'll help you. |
| 20:53.07 | [TK]D-Fender | jbalcomb : Uh oh! |
| 20:53.20 | Dr-Linux | [TK]D-Fender: what about unlock the phone and assgin it ip and tftp stuff? :) |
| 20:53.24 | dumb---me | i'm building a timer for my pbx and would like the user to have the option of entering a psswrd and overiding it |
| 20:53.38 | [TK]D-Fender | Dr-Linux : Another reason I don't pick Cisco :) |
| 20:53.52 | [TK]D-Fender | dumb---me : Ok.... and? |
| 20:53.57 | Dr-Linux | [TK]D-Fender: i know its very easy to configure but .. |
| 20:54.09 | dumb---me | i'm using the absolute timeout function |
| 20:54.19 | s34n | If I dial into my asterisk box from an outside pstn line and check vm, it doesn't ask for a password. |
| 20:54.34 | dumb---me | if someone enters, say one, is there any way to cancel the hangup |
| 20:54.40 | s34n | If I dial into my asterisk box from an inside sip line and check vm, it does ask for a password, but won't accept the correct one. |
| 20:54.50 | Dr-Linux | [TK]D-Fender: if the phone has private IP from dhcp and tftp server has public ip assign to it, should it work? |
| 20:55.12 | Dr-Linux | in this case phone is not recognizing the tftp server |
| 20:55.17 | [TK]D-Fender | Dr-Linux : Dunno... never had to mess with a Cisco. I should soon though.... SNOM as well |
| 20:55.33 | s34n | why is vm password behavior acting wierd? |
| 20:55.35 | [av]bani | fender is getting a snom? |
| 20:56.11 | [av]bani | <delldude>dude you're gettin a snom!</delldude> |
| 20:56.42 | s34n | ok. both are asking for a password now, but * won't accept password from sip phone. |
| 20:56.43 | sthw45ywyw5 | [TK]D-Fender: What is I had 5 people that need to share a phone |
| 20:57.09 | badboyz | would a utility like this be of use to anyone here? http://www.invalidrequest.com/monitor/index.html |
| 20:57.12 | dumb---me | fender-did u get that? |
| 20:57.57 | sthw45ywyw5 | [TK]D-Fender: Correction: What if I had 5 people that need to share a phone |
| 20:58.13 | Hmmhesays | ok so has anyone else run into polycomms still ringing after asterisk signal's an answer? |
| 20:58.13 | _Sam-- | [av]bani: does your teliax route stay on one backbone, or do you have some interconnects someplace? |
| 20:58.23 | [TK]D-Fender | [av]bani : I'm not closed-minded about all of this you need to realize. While I love Polycom for most applications, its not to say that other products aren't good as well, just not fitting the kind of profiles I suggest |
| 20:58.31 | *** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 20:58.38 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 20:58.43 | *** part/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 20:58.50 | [TK]D-Fender | sthw45ywyw5 : Ok, how would you envision managing this? |
| 20:59.33 | [av]bani | _Sam--: qwest->savvis->rockynet |
| 20:59.34 | sthw45ywyw5 | I don't know. I just assumed that the purpose of registrations was to have multiple people on one phone. Is that a wrong assumption |
| 20:59.47 | [TK]D-Fender | sthw45ywyw5 : I would much sooner suggest you have them share an IP 601. much easier than most other options. |
| 20:59.54 | dumb---me | is there any way to cancel the absolute timeout by pressing a button |
| 20:59.56 | jbalcomb | badboyz: I should think so. I would be particularly interested because we are setting up Asterisk servers at remote office to handle LCR or ARS. |
| 21:00.09 | Hmmhesays | i'm having trouble with the polycomm ip600 |
| 21:00.23 | [TK]D-Fender | sthw45ywyw5 : No, if you want multiple people to SHARE a phone, then multiple-registrations is the best way in most cases. |
| 21:00.59 | dumb---me | Does anyone have any experience with timing calls? |
| 21:01.11 | _Sam-- | the teliax guy thinks some call problems are related to crappy interconnects/peering...which im not 100% buying because if there is no packet loss and minimum jitter, how could that be it? |
| 21:01.14 | [TK]D-Fender | sthw45ywyw5 : So you'd use 1 link key/reg, with multiple calls on a given line-key (to handle conferencing, call waiting, etc) |
| 21:01.42 | [TK]D-Fender | Hmmhesays : What kind of problems? |
| 21:03.19 | badboyz | jbalcomb: check your /msg |
| 21:03.33 | sthw45ywyw5 | Let me just clarify: If there was only one phone in an office with three people, I use three registrations, correct? Or is it better to have one phone per person? |
| 21:04.13 | _Sam-- | [av]bani: do you think teliax terminates the calls right there or they hand off to another gateway? |
| 21:05.14 | [TK]D-Fender | sthw45ywyw5 : Of course its better to get a phone for each person.... who wants to wait in line? |
| 21:06.03 | badboyz | jbalcomb: are you able to msg? |
| 21:06.10 | dumb---me | has anyone use the absolute timeout function? |
| 21:07.31 | [av]bani | _Sam--: dunno |
| 21:07.39 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 21:07.53 | [av]bani | _Sam--: if it were interconnects, why is junction networks fine? that would imply rockynet is having problems |
| 21:08.03 | _Sam-- | what is your route to junction? |
| 21:08.10 | [av]bani | qwest->savvis->rockynet |
| 21:08.15 | _Sam-- | that is to teliax |
| 21:08.19 | _Sam-- | what about to junction |
| 21:08.27 | bigjb | grrrr |
| 21:08.44 | [av]bani | qwest->wcg->jn |
| 21:08.59 | _Sam-- | so it could possibly be interconnect |
| 21:09.02 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 21:09.06 | _Sam-- | because you use a different internconnect to get to teliax |
| 21:09.09 | *** join/#asterisk p0g0__ (n=pogo@mrtc-dsl-610149.mis.net) |
| 21:09.11 | _Sam-- | but im not believing it |
| 21:09.13 | [av]bani | well, i have 2 paths to teliax |
| 21:09.15 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 21:09.27 | [av]bani | also charter->att->pnap->rockynet |
| 21:09.28 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 21:09.30 | [av]bani | and that has the same issue |
| 21:09.41 | _Sam-- | but i do think it may be something to it...because from my host, teliax is further away ms wise, and has 0 packet loss than the host that is having problems |
| 21:09.51 | _Sam-- | and the host that is having problems uses a different route to teliax |
| 21:10.02 | [av]bani | so basically pnap->rockynet and savvis->rockynet have the exact same issue |
| 21:10.10 | [av]bani | 2 paths, both same |
| 21:10.59 | _Sam-- | seeing what route my problem host takes |
| 21:11.00 | [av]bani | in factdoesnt seem to be any better or worse either path, its about same |
| 21:11.18 | _Sam-- | you are an expert...what do you theorize is the problem? |
| 21:11.25 | [av]bani | i think teliax may not be handling jitter well |
| 21:11.42 | _Sam-- | my problem host is att-->pnap--> rockynet |
| 21:11.56 | [av]bani | well i get it savvis->rockeynet also |
| 21:12.15 | [av]bani | but i hear the remote end perfectly |
| 21:12.29 | [av]bani | i am going to test teliax origination in a bit |
| 21:13.56 | _Sam-- | if jitter were the problem, you couldnt fix that on your end with jitterbuffer? |
| 21:14.32 | [av]bani | no, that only affects receiving end |
| 21:14.35 | [av]bani | i hear them fine |
| 21:14.41 | _Sam-- | i see |
| 21:14.44 | [av]bani | remote end complains about stuttering |
| 21:14.55 | [av]bani | which means teliax may not be handling jitter well |
| 21:14.56 | *** join/#asterisk homebrew-hsv (n=homebrew@mail.kancharla.com) |
| 21:15.00 | _Sam-- | is intermittent or consistent? |
| 21:15.03 | [av]bani | consistent |
| 21:15.10 | *** join/#asterisk [Outcast] (n=bill@222-153-151-243.jetstream.xtra.co.nz) |
| 21:15.43 | dumb---me | can anyone help me with a timer? |
| 21:15.55 | _Sam-- | if you want i will ask him to switch your account to voip-co2 |
| 21:16.07 | [av]bani | i'm gonna test originatin in a bit to double check |
| 21:16.08 | _Sam-- | but i doubt he will, since he doesnt know if i do or dont have authority from you to do that |
| 21:16.33 | [Outcast] | does asterisk need to be compiled with IEEE-compliance so the software kick in when numbers denormals on the ia64 processors |
| 21:16.40 | [av]bani | if he wants i can call him via teliax :) |
| 21:16.58 | _Sam-- | he hasnt spoken for a while |
| 21:17.12 | homebrew-hsv | Hi all |
| 21:17.26 | homebrew-hsv | Anyone know how to remove a +1 prefix from the From field and Contact field in an outgoing sip call? |
| 21:18.37 | *** join/#asterisk clint__ (n=clint@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 21:19.16 | *** part/#asterisk homebrew-hsv (n=homebrew@mail.kancharla.com) |
| 21:19.20 | _Sam-- | he's got a point |
| 21:19.24 | _Sam-- | [16:18] davidcaldworth: if i mess with jitter settings then what happens when your quality goes to shit? |
| 21:19.29 | Katty | pointy |
| 21:20.14 | Beirdo | heya, Katty |
| 21:20.36 | *** join/#asterisk sevardd (n=kynan@198.174.233.25) |
| 21:20.58 | [av]bani | hmm weird, now the stutter is gone |
| 21:21.05 | _Sam-- | interesting |
| 21:21.45 | sthw45ywyw5 | [TK]D-Fender: It doesnt seem to make sense to have one-phone per person if most of us rarley use the phone. In addition we do not have enough ethernet jacks. So can you verify that The purpouse of multiple registrations is so multiple people can be "registered" to the same phone. |
| 21:21.49 | Katty | hiya, Beirdo |
| 21:21.54 | _Sam-- | i think you just lucky on the call |
| 21:22.18 | Beirdo | I'm so happy... but I wish my fiancee were STILL here |
| 21:22.19 | Beirdo | heh |
| 21:22.28 | [Outcast] | is kevin around? |
| 21:22.35 | badboyz | mitnick? |
| 21:22.37 | *** join/#asterisk Lebowski (n=fl@87.252.72.16) |
| 21:22.43 | Beirdo | distance relationships can be a serious PITA |
| 21:22.44 | [Outcast] | ah no, flemming |
| 21:22.46 | sevardd | he's out getting freepizza4lieflolz |
| 21:23.52 | [TK]D-Fender | sthw45ywyw5 : well there are lost of alternatives depending on what your needs are. For 5 people you could get them each an analog phone (5 x $10) on an ATA (3 x $70) = $260 and they'd each have their own phone...... |
| 21:24.03 | *** join/#asterisk dumb---me (n=dwayne@64-42-204-117.mb.skyweb.ca) |
| 21:24.07 | dumb---me | hullo |
| 21:24.12 | dumb---me | i have a time q |
| 21:24.13 | dumb---me | ? |
| 21:24.16 | [TK]D-Fender | And add a 20$ 5 port switch on. |
| 21:24.29 | sthw45ywyw5 | It is not a matter of cost |
| 21:24.44 | _Sam-- | stkn: for the same price they could buy gxp2000s :) |
| 21:24.56 | _Sam-- | er [tk]: for the same price they could buy gxp2000s |
| 21:24.58 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 21:24.59 | sthw45ywyw5 | I would rather have the cool ip phones |
| 21:25.00 | [TK]D-Fender | sthw45ywyw5 : Well if you really only want 1 phone, then I'd suggest a Polycom IP 601 for them then. |
| 21:26.30 | sthw45ywyw5 | Does nayone know why the display on my 301 polcom shows the last 4 digits of my name preceded by three periods? |
| 21:26.54 | badboyz | because the 301 has a limited display on it, the ... means its been abbreviated |
| 21:27.29 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 21:27.42 | [TK]D-Fender | ok, back later..... |
| 21:27.44 | MstlyHrmls | sthw45ywyw5: what version of s/w are you using? |
| 21:28.02 | sthw45ywyw5 | What is s/w? |
| 21:28.50 | CloseCall | hi |
| 21:28.52 | CloseCall | im back |
| 21:29.02 | MstlyHrmls | sthw45ywyw5: software |
| 21:29.17 | sthw45ywyw5 | On the polycom: App verion = 1.4.1.0040 is that what you want |
| 21:29.17 | CloseCall | ok this is my latest status: init script is now placed in /etc/init.d/ |
| 21:29.36 | CloseCall | but when i run and then do status i get this: |
| 21:29.36 | CloseCall | Starting asterisk: [ OK ] |
| 21:29.36 | CloseCall | [root@Shana log]# /etc/init.d/asterisk status |
| 21:29.36 | CloseCall | asterisk dead but pid file exists |
| 21:29.36 | CloseCall | [root@Shana log]# |
| 21:30.10 | CloseCall | nothing to be found in /var/log/message nor in /var/log/asterisk/event_log |
| 21:30.20 | sthw45ywyw5 | bootrom=2.6.1.0003 |
| 21:30.55 | *** join/#asterisk VJ (n=vijay@203.123.32.80) |
| 21:31.15 | sevardd | When a phone is disconnected from the network Voicemail tells me they're on the phone, why? |
| 21:31.52 | [av]bani | because a disconnected phone is 'busy' |
| 21:32.16 | sevardd | is there any way to change that function? |
| 21:32.20 | *** join/#asterisk MrMagic (n=bleem@dynamic-62-56-40-54.park-s46b.dslaccess.co.uk) |
| 21:32.26 | [av]bani | do chanisavail before dial |
| 21:32.27 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 21:32.42 | [av]bani | and give them a different message if the channel isnt available |
| 21:32.43 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 21:32.50 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 21:32.57 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.110.201.Dial1.Chicago1.Level3.net) |
| 21:33.17 | sevardd | [av]bani: in extensions.conf ? |
| 21:33.18 | [av]bani | yes |
| 21:33.43 | [av]bani | http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail |
| 21:33.53 | sevardd | thank you. |
| 21:33.56 | *** kick/#asterisk [ComPuTeR!i=denon@sassinak.net] by denon (denon) |
| 21:33.56 | *** mode/#asterisk [+b *!*@85.102.154.44] by denon |
| 21:33.58 | sevardd | reading |
| 21:34.10 | *** mode/#asterisk [+b Computer!*@*] by denon |
| 21:34.13 | denon | (message spammer) |
| 21:35.14 | *** join/#asterisk Coccyx (n=clint@typhoon.org) |
| 21:35.38 | sevardd | [av]bani: wow, i really don't know where it would go |
| 21:35.40 | VJ | hwo can we do a conference in asterisk |
| 21:36.01 | *** join/#asterisk Defraz_ (i=t0tal@tim.mychoice.cc) |
| 21:36.34 | VJ | any idea |
| 21:36.43 | VJ | how can we do a conference in asterisk |
| 21:37.20 | _Sam-- | someone would tell you but they are allin conference calls |
| 21:37.28 | MrMagic | hehe |
| 21:37.33 | MrMagic | meetme perhaps |
| 21:37.49 | }btorch{ | anyone knows a cool IAX2 softphone for linux similar to cubix ? |
| 21:38.17 | MrMagic | http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe |
| 21:38.45 | brad_mssw | }btorch{: 'twinkle' if i remember correctly was cool |
| 21:38.49 | brad_mssw | }btorch{: just a bad name |
| 21:39.22 | *** join/#asterisk nemy (n=piout@narya.piout.net) |
| 21:39.37 | nemy | hi |
| 21:39.57 | *** join/#asterisk somedood (n=somedood@63.225.225.227) |
| 21:40.41 | somedood | Has anyone had problems with Asterisk At Home, where there is one-way audio when using IAX + g729, but when using IAX + ulaw or gsm it works properly> |
| 21:40.52 | brad_mssw | actually, looks like it's only SIP, not iax ... hmm |
| 21:41.02 | *** join/#asterisk yiddoX (n=yiddoX@host-84-9-43-72.bulldogdsl.com) |
| 21:41.07 | somedood | This is 2 asterisk machines talking to each other, one is asterisk at home, the other isn't |
| 21:41.11 | somedood | they're both version 1.2.x |
| 21:41.14 | brad_mssw | somedood: i assume you've purchased the g729 licenses |
| 21:41.21 | somedood | yes, on both ends |
| 21:41.28 | brad_mssw | how many? |
| 21:41.30 | tronix | brad_mssw: heh yup... (re: twinkle) -- I kept typing 'twinkie' in the config files :P |
| 21:41.36 | somedood | and from the asterisk console, show g729 shows the available channels |
| 21:41.38 | somedood | we have 22 |
| 21:41.39 | somedood | they have 2 |
| 21:41.55 | somedood | we are using g729 to connect sip calls with others |
| 21:42.07 | *** join/#asterisk _cleric_ (n=dacleric@87.193.28.105) |
| 21:42.08 | somedood | and those calls work properly |
| 21:42.10 | brad_mssw | somedood: hmm, dunno, that was my only thought, has worked fine here |
| 21:42.33 | somedood | http://forums.whirlpool.net.au/forum-replies-archive.cfm/442714.html |
| 21:42.41 | somedood | that is identical to what is happening with us |
| 21:42.53 | nemy | are there any french speaking guy here ? |
| 21:43.10 | dumb---me | i would like to set up a specail timer, can anyone help? |
| 21:43.23 | nemy | i'm in charge of traducing Asterisk : the future of telephony for O'reilly |
| 21:43.49 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 21:43.57 | dumb---me | i want the user to be able to enter an ext and override the time |
| 21:43.57 | dumb---me | timer |
| 21:44.00 | nemy | and i need some help for a few terms |
| 21:44.10 | dumb---me | anyone |
| 21:44.11 | dumb---me | ? |
| 21:44.38 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 21:45.47 | dumb---me | anyone out there know how to keep a function from executing? |
| 21:45.59 | dumb---me | by dialing a number |
| 21:46.36 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 21:47.24 | dumb---me | HELLLOOOOO |
| 21:48.09 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 21:48.09 | VJ | i am speakign with a customer on trunk line |
| 21:48.38 | VJ | now i want one another client of mine to be in conference, and he cannot call me, i have to call him |
| 21:48.49 | VJ | and take the other client into conference |
| 21:48.51 | VJ | how to do it |
| 21:48.57 | *** join/#asterisk sevard (n=kynan@198.174.233.25) |
| 21:49.01 | *** part/#asterisk sevard (n=kynan@198.174.233.25) |
| 21:49.03 | [av]bani | o_O |
| 21:49.25 | sevardd | o-0 |
| 21:49.28 | *** join/#asterisk trek (n=rsi@lns-bzn-35-82-250-207-194.adsl.proxad.net) |
| 21:50.34 | trek | Anyone as a running asterisk 1.2.4 and bri_stuff working ? |
| 21:51.23 | VJ | any idea how can i take an external party into conference |
| 21:53.18 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:54.26 | *** part/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 21:54.49 | Hmmhesays | transfer them |
| 21:55.31 | *** join/#asterisk arcy (n=arcanum@ppp43-adsl-17.ath.forthnet.gr) |
| 21:55.36 | g4m | VJ: check out features.conf |
| 21:58.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net) |
| 21:59.04 | VJ | ok |
| 22:00.22 | VJ | it gives the information about parkin the call |
| 22:00.40 | sevardd | How do I control the volume of Festival() |
| 22:04.05 | jaiger | festival(yell) or festival(whisper) |
| 22:04.36 | sevardd | Won't that just speak "yell" or "whisper" if so then it's the worst joke i've heard all day. |
| 22:05.09 | sevardd | If I text2wave a file it speaks normly but if I Festival(something) it SCREAMS 'something' |
| 22:05.10 | jaiger | it was a joke |
| 22:05.15 | jaiger | I've never used festival |
| 22:05.24 | sevardd | good one, dude. |
| 22:07.56 | EriSan | can i take a call that rings on extention A on extention B? |
| 22:08.00 | somedood | does anyone here use asterisk at home? |
| 22:08.09 | somedood | I think the problem may be associated more with that than anytthing |
| 22:08.22 | *** join/#asterisk Ceki (n=cekicvel@hsiproxy.astra-net.com) |
| 22:08.41 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:08.46 | *** join/#asterisk nestar (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 22:09.09 | nestar | long time, no chat peeople. |
| 22:09.34 | Ceki | when i run my * i get following error "floating point exception" |
| 22:09.44 | Ceki | any help would bee nice |
| 22:09.54 | MrMagic | nasty error ceki |
| 22:10.09 | MrMagic | and errors when compiling asterisk? |
| 22:10.31 | MrMagic | any rather |
| 22:10.39 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:10.46 | dumb---me | can anyone help me with timing a call? |
| 22:11.31 | MrMagic | I know your not meant to share it but if anyone has the Cisco 7960 firmware id greatly appreciate it |
| 22:11.42 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 22:12.00 | *** join/#asterisk ZeroOn797 (n=ZeroOne@81.189.57.6) |
| 22:12.59 | Ceki | i successfully compiled asterisk |
| 22:13.11 | Ceki | and i made my config files |
| 22:13.27 | Ceki | and it worked some time without any errors |
| 22:13.28 | MrMagic | did u try running asterisk before making any custom configs ? |
| 22:13.41 | Ceki | yes |
| 22:13.53 | MrMagic | and it ran or still with the floating point error ? |
| 22:14.12 | Ceki | i tested it for about one month with small changes |
| 22:14.28 | MrMagic | its suddenly started doing it ? |
| 22:14.36 | Ceki | i tested it for about one month with small changes then one day i simply get that error |
| 22:14.43 | nestar | Did polycom ever change their phones so that you could disable call waiting? |
| 22:15.02 | MrMagic | mm any changes on the system? automatic updates? up2date/yum/emerge etc? |
| 22:15.18 | MstlyHrmls | nestar: you could try setting the number of call appearences to 1 |
| 22:15.30 | Ceki | i only changed version of zaptel from 1..9 to 1.2 |
| 22:15.35 | MstlyHrmls | nestar: or did you just want to disable the sounds? |
| 22:15.43 | *** join/#asterisk santiago (n=santiago@63.245.86.155) |
| 22:16.08 | Ceki | yes i did update |
| 22:16.16 | Ceki | i use suse 10 |
| 22:16.59 | MrMagic | mm possibly an update could of broke it although prolly not |
| 22:17.46 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 22:19.00 | *** part/#asterisk santiago (n=santiago@63.245.86.155) |
| 22:19.23 | Ceki | well do you suggest any fix or the best solution is to recompile * |
| 22:20.11 | nestar | MstlyHrmls: yeah, trying to have more than one call rejected, for the purpose of call Queues |
| 22:21.21 | *** join/#asterisk HumanSky (n=info@h-67-101-227-106.phlapafg.covad.net) |
| 22:21.55 | HumanSky | if I sign up to a VoIP provider like BroadVoice, can I keep my landline as well? |
| 22:22.07 | HumanSky | keep the same phone number |
| 22:22.25 | HumanSky | I tried to look up that question on their FAQ but it didn't mention it |
| 22:23.04 | MstlyHrmls | nestar: you could try setting "reg.x.callsPerLineKey" to 1 for each registration, or set it globally in "call.callsPerLineKey" |
| 22:23.41 | *** join/#asterisk svenna_ (n=svenna@p548D3B7A.dip0.t-ipconnect.de) |
| 22:23.56 | *** join/#asterisk Ceki (n=cekicvel@hsiproxy.astra-net.com) |
| 22:24.09 | nestar | MstlyHrmls: ok, thanks. |
| 22:24.11 | Ceki | again any suggestions |
| 22:26.45 | *** join/#asterisk Ceki (n=cekicvel@hsiproxy.astra-net.com) |
| 22:27.41 | SwK[Work] | short of restarting asterisk... anyone know how to hangup a channel stuck in this state? Feb 1 16:26:46 WARNING[1254]: chan_zap.c:4328 __zt_exception: We're Zap/136-1, not |
| 22:29.48 | dumb---me | anyone out there experienced with call timers? |
| 22:31.58 | dumb---me | can anyone read this? |
| 22:32.05 | sevardd | i can't. |
| 22:32.15 | somedood | read what? |
| 22:32.18 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
| 22:32.22 | dumb---me | i |
| 22:32.45 | dumb---me | 'm wondering if there's anything wrong here and my messages don't come through |
| 22:32.54 | dumb---me | looks like they are |
| 22:33.08 | dumb---me | soooo no one's every used the absolute timeout function? |
| 22:33.14 | somedood | I never have, no |
| 22:33.15 | sevardd | I can't see your messages clearly. If you would please press ALT and F4 at the same time to clear them up. |
| 22:33.36 | dumb---me | what's u're point |
| 22:33.47 | sevardd | Please clear up your messages. |
| 22:34.06 | dumb---me | r u always this stupid or are u just pretending? |
| 22:34.27 | dumb---me | can u read that |
| 22:34.27 | dumb---me | ? |
| 22:34.34 | sevardd | dumb---me: hello? |
| 22:34.35 | *** join/#asterisk highwymn (n=highwymn@0-1pool138-244.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:34.48 | dumb---me | ? |
| 22:34.53 | somedood | hehehhe |
| 22:35.02 | somedood | I should come here more often |
| 22:35.13 | dumb---me | i mean is it wrong to ask a question here? |
| 22:35.20 | *** join/#asterisk sigmounte__ (n=sigmount@www.sighq.net) |
| 22:36.37 | dumb---me | is there someone here with experience writing scripts? |
| 22:36.45 | highwymn | Anyone able to point me in the right direction for documentation on using an asterisk server for the following: |
| 22:37.22 | *** join/#asterisk Cool_One (n=bclinton@h84.65.255.206.cable.htsp.cablelynx.com) |
| 22:37.26 | highwymn | I need to use my cell phone to dial into work, and then use asterisk to pick an extension to dial from out to another phone number replacing my caller id |
| 22:37.46 | highwymn | where do I need to start reading? |
| 22:37.51 | jbalcomb | badboyz: you have a link where I can d/l that app? |
| 22:38.27 | jbalcomb | highwymn I think thats called DISA (Direct Inward System Access) |
| 22:39.07 | highwymn | Is that covered in the Asterisk handbook or on the wiki somewhere? |
| 22:39.48 | *** join/#asterisk pb__ (n=pb@2002:5246:d929:1:20e:2eff:fe2d:60bf) |
| 22:40.05 | jbalcomb | highwymn you should be able to just tell it to take ext. XXXX and execute an application to take the number XXXXXXXXXX and Dial(Zap/XXXXXXXXXX) |
| 22:40.33 | jbalcomb | highwymn I'm not sure. Check on the Dial application on the wiki. |
| 22:40.36 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 22:41.05 | highwymn | ok. looking now |
| 22:41.21 | Cool_One | does anyone in here run asterisk@home |
| 22:41.34 | De_Mon | what does the Queue option n(no retries on the timeout) mean exactly? The queue times out by default it just goes back into the queue again? |
| 22:42.31 | highwymn | I dink around with asterisk@home in my home |
| 22:43.08 | Cool_One | do you use a modem or a service for calls |
| 22:43.35 | highwymn | I use a telecom card through my PTSN line |
| 22:43.45 | Cool_One | cool |
| 22:43.49 | Cool_One | that is what I am trying to do |
| 22:43.51 | *** join/#asterisk test34 (n=test34@102.174.204.68.cfl.res.rr.com) |
| 22:44.08 | Cool_One | but can't seem to get my routes right I guess |
| 22:44.21 | *** join/#asterisk dumb---me (n=dwayne@64-42-204-117.mb.skyweb.ca) |
| 22:44.25 | highwymn | since it is just for dinking around at home I got a cheap diguim clone card from digitnetworks and toy around with it |
| 22:44.36 | Cool_One | all documentation I find online use broadcom or some other service |
| 22:44.40 | highwymn | which direction are you trying to call and with what |
| 22:44.46 | Cool_One | I got the same cards |
| 22:44.59 | highwymn | do you already have them configured |
| 22:44.59 | Cool_One | I have a grandstream phone |
| 22:45.09 | Cool_One | and I can call extension to extension fine |
| 22:45.21 | highwymn | you are trying to get out then? |
| 22:45.27 | Cool_One | correct |
| 22:45.30 | Cool_One | wil not work |
| 22:45.37 | Cool_One | <PROTECTED> |
| 22:45.38 | [av]bani | De_Mon: one way to find out, set a short timeout and see what happens :) |
| 22:45.45 | highwymn | you already tried dialing a 9 and then the number |
| 22:46.12 | *** join/#asterisk sch19 (n=sch19@adsl-8-228-216.mia.bellsouth.net) |
| 22:46.28 | sch19 | howdy folks |
| 22:46.36 | dumb---me | hey dudes |
| 22:46.42 | highwymn | asterisk@home was pretty much already set up with minimal dialplan and once my telecom card was recongnized and ip phones were up, only had to dial 9 to get out |
| 22:46.55 | Cool_One | ok |
| 22:46.56 | dumb---me | need to hang up a call after a certain amount of time, anyone help? |
| 22:47.05 | Cool_One | I am a *$($*# didn't know about the 9 |
| 22:47.09 | Cool_One | but it just dialed out |
| 22:47.13 | Cool_One | man... |
| 22:47.18 | Cool_One | I feel smart now |
| 22:47.37 | highwymn | good. then you learned something. no worries. I do stuff like that all the time too |
| 22:47.43 | *** join/#asterisk MatsK (n=mk@84-217-5-20.tn.glocalnet.net) |
| 22:47.43 | Cool_One | hah |
| 22:48.02 | Cool_One | I installed gentoo and asterisk from hand the other day and got it to work |
| 22:48.05 | sch19 | .. I'm pretty asterisk newb myself, I'm hoping to idle a bit and absorb some info :P |
| 22:48.28 | Cool_One | but I wanted to install AMP and I kept getting errors so I found asterisk@home and it had all the packages made together |
| 22:48.42 | Cool_One | I installed it and have been playing around with it for a couple of hours |
| 22:48.44 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 22:49.03 | sch19 | agreed, a@h seems nice, at first glance. but it dummifies the process so much, it's hard to learn much :P |
| 22:49.08 | highwymn | I like asterisk@home, but still am very nostalgic about compiling and configuring myself |
| 22:49.08 | Cool_One | how many lines and extensions do you think the @home package will support |
| 22:49.35 | sch19 | as many as you need |
| 22:49.38 | highwymn | since it is a full asterisk server, really, it all depends on your hardware |
| 22:49.59 | Cool_One | I read as much but kind was shakey on @ home thing |
| 22:50.00 | sch19 | sorry for butting in, trying to help |
| 22:50.30 | Cool_One | you said that it came default with some dialplans |
| 22:50.39 | Cool_One | what do you have to add |
| 22:50.42 | highwymn | it all depends on the hardware... asterisk@home is just a cute name. It is a full blown system |
| 22:50.54 | Cool_One | sweet |
| 22:51.19 | highwymn | what you add is what you want... like automated menus, music on hold, etc |
| 22:52.35 | SkramX | Hi All. |
| 22:52.42 | highwymn | you can tinker all you want, but it really depends on what your end goal is what you have to add |
| 22:52.57 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 22:53.16 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 22:53.18 | Cool_One | that is just to neat |
| 22:53.50 | Bakermd | Anyone able to help on a Cisco Voice q? - getting Requested circuit/channel not available |
| 22:54.04 | highwymn | so first, you really need to draw out what you want your setup to be, then you evaluate what it already has, then set up the rest |
| 22:55.00 | SkramX | Hi All. |
| 22:56.14 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 22:58.49 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 23:00.58 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 23:01.34 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 23:01.56 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 23:04.08 | darwin_35 | anyone here running fbsd and a t100p card |
| 23:04.15 | darwin_35 | are the drivers workign |
| 23:04.17 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 23:04.47 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 23:06.36 | highwymn | If I use DISA, is it possible to monitor the calls, ie customer service quality control? |
| 23:07.08 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 23:07.18 | HumanSky | anyone recommend a VoIP provider that will let me transfer my Verizon landline number, also time is not a factor, so whenever the xfer takes place will be fine |
| 23:09.02 | tuxinator_linux | HumanSky, almost all of them will allow that |
| 23:10.52 | *** join/#asterisk trek (n=rsi@lns-bzn-35-82-250-207-194.adsl.proxad.net) |
| 23:16.02 | *** join/#asterisk sevard (n=kynan@198.174.233.25) |
| 23:18.46 | dumb---me | i'm trying to set up a call timer, anyone? |
| 23:19.51 | hardwire | dumb---me: what would it do? |
| 23:20.42 | *** join/#asterisk konrads (i=rats@84.237.135.166) |
| 23:21.01 | konrads | Hello. What is the status of asterisk on 64bit machines? Do modules work as expected? |
| 23:21.08 | tronix | don't know, but could have uses in a prepaid application (re: what could a timer do) |
| 23:21.08 | konrads | for hfc e.g. |
| 23:21.16 | dumb---me | it would warn the person after a time and if password would not be entered it would cut them off |
| 23:21.41 | jyukes | hi |
| 23:21.56 | jyukes | how quickly does the asterisk echo test hairpin audio back? |
| 23:21.58 | tronix | dumb---me: sounds like you might be looking at rolling an AGI script or something |
| 23:22.43 | dumb---me | tronix: thanks for replying, i have set up an absolute timeout, could I reset it if someone pressed the right keys? |
| 23:24.01 | dumb---me | tronix: is there any literature on AGI scripts? |
| 23:24.50 | Ahrimanes | lots |
| 23:25.39 | *** part/#asterisk knight_ (n=knight@blackhole.phunc.com) |
| 23:25.45 | dumb---me | could you point me to a goodone? |
| 23:26.40 | konrads | Common hardware was 5v or 3v? |
| 23:26.47 | *** join/#asterisk trek (n=rsi@lns-bzn-35-82-250-207-194.adsl.proxad.net) |
| 23:29.01 | trek | help seek for hfc card support and asterisk 1.2.4. Anyone ? |
| 23:29.09 | _Sam-- | i know this is the wrong channel..but there are some smart folks...does anyone know how to limit/throttle apache requests for hosts that have downloaded too much MB |
| 23:29.24 | Nugget | what does that have to do with asterisk? |
| 23:29.31 | _Sam-- | #apache is clueless |
| 23:29.34 | _Sam-- | so i figured id at least ask |
| 23:29.34 | Nugget | heh |
| 23:29.46 | [av]bani | _Sam--: you need to use a custom mod_* |
| 23:29.51 | _Sam-- | it doesnt exist |
| 23:29.53 | [av]bani | mod_throttle |
| 23:29.56 | [av]bani | mod_bw |
| 23:29.56 | *** join/#asterisk dumb---me (n=dwayne@64-42-204-117.mb.skyweb.ca) |
| 23:30.02 | rajiv|work | mod_bandwidth or mod_throttle or something liek that |
| 23:30.03 | _Sam-- | they are all based on reqeuests per period |
| 23:30.09 | _Sam-- | not on amount of throughput |
| 23:30.11 | _Sam-- | and mb |
| 23:30.12 | [av]bani | no, one of them can throttle throughput |
| 23:30.15 | [av]bani | i used it |
| 23:30.20 | _Sam-- | i just read mod_bw |
| 23:30.23 | rajiv|work | bbl |
| 23:30.28 | *** join/#asterisk acehunky (n=chat_jok@59.184.4.145) |
| 23:30.53 | [av]bani | http://www.cohprog.com/v3/bandwidth/doc-en.html |
| 23:30.56 | [av]bani | i think thats the one we used |
| 23:31.05 | _Sam-- | ty |
| 23:31.39 | [av]bani | you could always put a server on its own ip and use GTS on cisco or rate limiting on linux |
| 23:32.06 | _Sam-- | its on its own ip |
| 23:32.15 | dumb---me | can anyone point me to some good agi guides, |
| 23:32.32 | _Sam-- | i have people that crawl our site and just try to wget the whole thing |
| 23:33.37 | [av]bani | so block wget :) |
| 23:33.54 | _Sam-- | we have a whole list of blocked clients |
| 23:33.57 | _Sam-- | including wget |
| 23:33.58 | acehunky | this one sounds good to me : http://home.cogeco.ca/~camstuff/agi.html |
| 23:34.02 | _Sam-- | but its easy to fake a client name |
| 23:34.06 | acehunky | dumb---me: http://home.cogeco.ca/~camstuff/agi.html |
| 23:34.10 | arcy | is it possible to do the following? when an incoming call comes in , Ring the extensions i want, and _if_ someone picks up, _then_ answer() the incoming and forward to the extension that answered |
| 23:34.28 | arcy | because otherwise, callers are charged while waiting for someone to pick up |
| 23:34.36 | Nugget | instead of trying to block or throttle the crawlers, start sending them wrong information. ;) |
| 23:34.55 | [av]bani | arcy: dial() instead of answer() |
| 23:35.00 | _Sam-- | Nugget: its easier said than done...especially if you dont know which are the good or bad clients. |
| 23:35.07 | arcy | thank you [av]bani |
| 23:35.08 | _Sam-- | the only way to know is based on how much they are downloading |
| 23:36.02 | _Sam-- | hmmm bani, the last version of that mod_bandwidth is from 2003...did you use apache 2.0? |
| 23:36.06 | znoG | should a POS (Point Of Sale) unit plugged into a Digium FXS port be able to dial out and establish its connection, etc? |
| 23:36.07 | [av]bani | yes |
| 23:36.15 | znoG | i heard that they've fixed the wctdm drivers to do this |
| 23:36.15 | *** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com) |
| 23:36.40 | znoG | ie. send faxes with fax machines plugged into TDM FXS ports, should be the same as what the POS is trying to do |
| 23:36.48 | znoG | (handshake and send some info over a telephone line) |
| 23:36.56 | _Sam-- | [av]bani: as always, thanks again for your advice. |
| 23:37.16 | [av]bani | \o/ |
| 23:37.22 | tronix | znoG: you probably also want to set /etc/zaptel.conf re: faxdetection setting while at it |
| 23:37.39 | tronix | (assuming this is a digium board or a zaptel compatible hw) |
| 23:38.06 | znoG | its a digium TDM board, yes |
| 23:38.09 | znoG | but this is for outbound only |
| 23:38.13 | tronix | errr zapata.conf i meant. my bad |
| 23:38.19 | *** join/#asterisk andio (n=andio@port-195-158-165-243.dynamic.qsc.de) |
| 23:38.23 | znoG | just need a POS machine to establish its link via a FXS port on the TDM card |
| 23:38.35 | znoG | (and out the FXO port) |
| 23:38.35 | tronix | yeah, you can set faxdetect=outgoing in /etc/asterisk/zapata.conf |
| 23:38.46 | znoG | even if its not a fax? |
| 23:38.49 | tronix | hmm |
| 23:38.57 | tronix | i guess that code listens for the tones |
| 23:39.12 | znoG | are you taking a wild guess or have you tried something similar? |
| 23:40.41 | tronix | well, let's see, unless this is something really special hw on the POS side... |
| 23:40.50 | andio | hi. upgraded to zaptel 1.2.3 today with a TE110P on a E1 line, and now there a lot of "HDLC Bad FCS" errors scrolling through the screen. but there are no problems with older T100P cards and 1.0.7 and 1.0.9. is there any information or solution to that problem? |
| 23:40.52 | tronix | I can't see why standard setup on * side shouldn't work for fax stuff |
| 23:41.08 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 23:41.18 | tronix | If POS = point of sale, then no, I haven't hooked up a POS |
| 23:41.24 | tronix | but I've played with fax stuff |
| 23:42.04 | tronix | both the pstn version and voip version |
| 23:42.26 | znoG | its a physical fax machine though, not fax stuff in asterisk |
| 23:42.29 | *** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net) |
| 23:42.46 | ahattar | hi all |
| 23:42.50 | znoG | i heard that the zaptel code had some problem with fax machines and the likes connecting to it and establishing a connection |
| 23:42.55 | *** join/#asterisk bigb (n=bigb@static-70-21-248-201.nwrk.east.verizon.net) |
| 23:42.58 | znoG | but it was supposed to be fixed, so I thought |
| 23:43.02 | tronix | znoG: does work. |
| 23:43.08 | bigb | Question for you guys |
| 23:43.24 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com) |
| 23:43.27 | bigb | Any reason why audio would be dropped when speaking? |
| 23:43.34 | znoG | tronix: i also tried doing this through a sipura ATA but didn't have much success |
| 23:43.35 | ahattar | quick question, does asterisk 1.2.4 support H323 phone? |
| 23:44.46 | BlueDevi1 | ahattar: yes |
| 23:44.46 | bigb | Hardware involved : MP108(FXO)x2, ~45 Grandstream gxp2000 phones |
| 23:45.13 | tronix | znoG: haven't played with the sipura (understand it's good) but in general, some ATAs can be a little quirky about these things. i do have the cisco ata-186 but my fax machine isn't here at moment or i'd test again |
| 23:46.31 | ahattar | bluedev: i have the phone in the discovering state I ran add station in my * box, wut should I do next? |
| 23:46.44 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:47.03 | pifiu | what is the asterisk echo test? |
| 23:47.04 | litage | what's asterisk-1.2.4-netsec? |
| 23:47.07 | pifiu | what number? |
| 23:47.14 | Qwell[] | litage: For the ranch networks devices |
| 23:47.42 | litage | Qwell[]: ranch? |
| 23:47.52 | Qwell[] | ranch networks |
| 23:47.57 | pifiu | like ranch sauce |
| 23:48.00 | litage | i'm guessing that a company? |
| 23:48.14 | Qwell[] | http://www.ranchnetworks.com/asterisk/asterisk_main.htm |
| 23:48.15 | litage | s/that/that's/ |
| 23:48.23 | litage | thanks Qwell[] |
| 23:49.46 | *** join/#asterisk Hunter_SC (n=Junior@201-25-249-237.fnsce703.dsl.brasiltelecom.net.br) |
| 23:51.01 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:51.05 | generalhan | hey everyone ! |
| 23:51.35 | pifiu | wtf is this ranch networks thing |
| 23:51.41 | pifiu | sounds like ranch dressing |
| 23:51.43 | Qwell[] | dynamic firewalls, basically |
| 23:51.43 | pifiu | so yummy |
| 23:52.09 | sevard | yum lickous. |
| 23:52.20 | Qwell[] | would be cool to see that support in iptables in the near future |
| 23:52.22 | [av]bani | bigb: aggressive echo cancellation might mute audio while speaking |
| 23:53.16 | generalhan | quick question: i want to put a "forward" on all calls coming into one specific number, so in the number definitions i want to do something like "exten => $VoIP_DID,1,Dial($My_Cell_Number)" how does the syntax work with that? or can i even do that ? lol. |
| 23:53.45 | Hunter_SC | hey everyone, beauty? Who has Asterisk installed here in the Slack? Type I have. Plus My question he is the following one. Necessary to install the Complete Slack? Why Yesterday I was to install in another machine more so with some package and gave error "ASTERISK_VERSION" when I was to compile. They know to say me why of the this? |
| 23:54.16 | sevard | awesome. |
| 23:54.17 | Qwell[] | Hunter_SC: A new translator need you do |
| 23:54.26 | generalhan | Qwell[]: LOL |
| 23:54.46 | [TK]D-Fender | Grammar rangers.....ATTACK!!!!!!! |
| 23:54.47 | generalhan | ... so says Yoda |
| 23:54.51 | Hunter_SC | Qwell[] ?? |
| 23:54.56 | Qwell[] | Hunter_SC: Exactly |
| 23:54.57 | pifiu | isnt there an echo test number for asterisk? |
| 23:55.11 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 23:55.12 | Qwell[] | pifiu: You could make one. 5555,1,Echo() |
| 23:55.20 | sevard | dude, don't respond to him. you'll be sucked into a foriegn language triad that will suck your soul |
| 23:55.23 | SibRphrek | i'm bored |
| 23:55.23 | pifiu | gotcha |
| 23:55.25 | SibRphrek | i got asterisk to work |
| 23:55.28 | SibRphrek | i got moh to work |
| 23:55.33 | znoG | tronix: yea, same, but fortunately I have a FXS module in the TDM that should work better than the ATA |
| 23:55.41 | SibRphrek | i got CDR -> mysql and even made a Filemaker pro front end for the mysql to work |
| 23:55.44 | SibRphrek | i dunno what else to do |
| 23:55.58 | Hunter_SC | Qwell[]: Exactly what? |
| 23:56.01 | generalhan | Anyone have any ideas about my forward issue ? i really want to set this up before i go home, so if someone, My Boss, calls me they think im here ! lol |
| 23:56.01 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 23:56.01 | pifiu | how does that work sibrpherk? |
| 23:56.14 | SibRphrek | pifiu: how does what work? |
| 23:56.19 | pifiu | this file maker pro thing |
| 23:56.24 | pifiu | i dont understand what you did |
| 23:56.27 | SibRphrek | oh |
| 23:56.27 | Qwell[] | generalhan: Sure, that'll mostly work. You can't use variables in the pattern, but... |
| 23:56.38 | Hunter_SC | Qwell[]: You know why of the o error "ASTERISK_VERSION" in cli.c? |
| 23:56.42 | Qwell[] | exten => 5551212,1,Dial(${MYCELL}) |
| 23:56.46 | SibRphrek | pifiu: it works like this - you have asterisk output it's CDR into MySQL (that part is easy) |
| 23:56.47 | generalhan | Qwell[]: thats what i need to know though is the syntax to make that work |
| 23:56.58 | Qwell[] | Hunter_SC: Get a better translator. Nobody can understand you. |
| 23:57.04 | generalhan | ohh i dont need like a DialIAX2/phonenumber or anything like that ? |
| 23:57.05 | SibRphrek | pifiu: then you have FileMaker use a ODBC extention to see the mySQL and import the data on load |
| 23:57.21 | Qwell[] | generalhan: you do, but that could be contained in MYCELL |
| 23:57.22 | pifiu | and what do you do with it in filemaker? |
| 23:57.26 | cypromis | .w 4 |
| 23:57.36 | De_Mon | Asterisk 1.2.1 -- I've got some agents and queues setup. queue timeout=10, but never the dialplan never gets to the next step. So I tried Queue(name|tT|||10) queue continues to call agents instead of timing out. |
| 23:57.45 | SibRphrek | pifiu: wahtever i want, i haven't done much - but i can manipulate the data so i can make bill records and such |
| 23:57.54 | generalhan | ok ill see what i can find out ! |
| 23:57.55 | generalhan | thanks ! |
| 23:58.00 | pifiu | interesting |
| 23:58.08 | pifiu | cant bill records be created dynamically? |
| 23:58.14 | pifiu | im sure someone has a billing app |
| 23:58.21 | De_Mon | Here's the kicker, if I hangup between the CLI saying 'nobody picked in 10000ms' and the queue looping, asterisk crashes |
| 23:58.43 | *** join/#asterisk _upsite (n=upsite@wls.swh.uni-halle.de) |
| 23:58.58 | SibRphrek | pifiu: yeah but you have to pay for them |
| 23:59.35 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:59.37 | *** join/#asterisk Zodiacal (n=hehe@bdsl.66.14.242.199.gte.net) |
| 23:59.39 | Ariel_ | hello everyone |
| 23:59.44 | Hunter_SC | PO Alguem FAla Portugues Entao hehehe |
| 23:59.46 | Zodiacal | anyone know why i can't seem to get ground start to work. i keep getting the following error when trying to run ztcfg: "Changing signalling on channel 1 from FXS Loopstart to FXS Groundstart ZT_CHANCONFIG failed on channel 1: Invalid argument (22)" here is my pastebin of my conf files: http://pastebin.com/534674 |
