| 00:00.17 | camonz | sip debug shows me the packages that i'm sending the sip proxy for my registration |
| 00:01.13 | camonz | the idea of the set up., is that a SER box will proxy some clients, wich in the end register with me running * |
| 00:01.46 | tainted- | ok |
| 00:01.53 | tainted- | where is ser box |
| 00:01.56 | tainted- | remote or on your lan |
| 00:01.58 | camonz | but the client that is registering first with SER and should be redirected to me for it's registration doesn't gets redirected |
| 00:02.02 | camonz | remote |
| 00:02.06 | camonz | public ip |
| 00:02.17 | tainted- | everything sounds good |
| 00:02.28 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-167.modem.logical.net) |
| 00:03.22 | camonz | wich brings me to another topic.., how do i handle the incoming call inside of * |
| 00:03.36 | Carp1 | extentions.conf :) |
| 00:03.38 | camonz | the docs say it will fall in the default context |
| 00:03.56 | tainted- | well |
| 00:03.58 | camonz | how do i make it fall in a different one..., by placing the register line in another extension |
| 00:04.10 | tainted- | if u don't have extensions properly defined, the call will be rejected |
| 00:04.18 | tainted- | no |
| 00:05.14 | camonz | the /1006 is the extension it will have when it falls on my default context |
| 00:05.27 | camonz | i know if that extension is not processed in extensions.conf the call will be rejected |
| 00:06.41 | tainted- | ok |
| 00:09.35 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:19.10 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 00:35.47 | *** join/#asterisk apardo (n=apardo@87.217.144.163) |
| 00:38.11 | *** join/#asterisk blitz420 (i=blitz420@12-218-222-108.client.mchsi.com) |
| 00:44.41 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 00:47.14 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:48.23 | NoName32 | hi all i am still pretty new to asterisk got a problem i cant figure out .. working on the one touch record using the featuremap with automon => *1 but it doesnt seem to be reconizing the *1 if i change it to ** it works any ideas/ sugestions i am using asterisk 1.2.6 |
| 00:49.31 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-219.modem.logical.net) |
| 00:52.44 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:57.33 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
| 01:00.37 | *** join/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 01:01.18 | kev009 | I need a device to interface with POTS on the external side |
| 01:01.49 | kev009 | just a single line, then I will have a cisco IP phone internally |
| 01:01.54 | kev009 | is there a cheap device to do this? |
| 01:02.31 | RoyK | <PROTECTED> |
| 01:02.57 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 01:14.25 | mindwarp | kev009: http://ozvoip.com/showProduct.php?device=sipura3000 <--- maybe something like this? |
| 01:14.41 | mindwarp | kev009: just a guess, i'm a newbie at this |
| 01:14.58 | Ariel_ | yes but it's going to be had to get the cisco to connect to it. |
| 01:15.53 | mindwarp | oh. i thought that that box would connect to a network and then the ip phone would connect to the same network over ethernet |
| 01:16.16 | kev009 | I just want the device to be a POTS interface, so I can use a PC to run Asterisk |
| 01:16.22 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 01:16.30 | kev009 | so I can select between POTS and VOIP |
| 01:17.10 | Ariel_ | kev009, then the sipura 3000 would work |
| 01:18.55 | *** join/#asterisk netsurfer (n=bbjunkie@dreambox.myvnc.com) |
| 01:19.04 | netsurfer | hi people |
| 01:19.11 | mindwarp | Ariel_: do you think there might be some kind of compatibility issue between the sipura and the cisco? |
| 01:19.26 | mindwarp | or were you talking about physically "connecting" the cisco to it |
| 01:19.54 | Ariel_ | mindwarp, 2nd one connecting the cisco to the sipura would be hard to do |
| 01:20.02 | mindwarp | ok, gotcha |
| 01:20.04 | Ariel_ | but if you have asterisk inbetween no problem. |
| 01:20.10 | netsurfer | i've run into a minor problem with the CDR is asterisk... it dosent log transferred calls.. does anyone know of a workaround ? |
| 01:20.41 | netsurfer | currently using 1.2.4 |
| 01:20.56 | mindwarp | Ariel_: do you know if the sipura could be used to connect to a POTS line (rather than a phone)... i guess they call that "FXO" as opposed to "FXS"? |
| 01:21.04 | mindwarp | not positive about the terminology here |
| 01:21.24 | Ariel_ | mindwarp, the sipura 3000 yes |
| 01:21.52 | Ariel_ | netsurfer, cdr entry's are really asterisk short coming. |
| 01:22.06 | netsurfer | :( |
| 01:22.17 | mindwarp | so would you just run a cable from the wall jack into the thing, then... what? or do you have to split wires and such |
| 01:22.33 | netsurfer | Ariel_ really? that is the only fault I can see with it |
| 01:22.35 | Ariel_ | mindwarp, the 3000 has one fxo port and one fxs |
| 01:22.53 | mindwarp | ok |
| 01:22.57 | Ariel_ | netsurfer, it has many others but that one they don't seem to want to do anything about |
| 01:23.17 | netsurfer | oh... great :( |
| 01:24.15 | DoktorGreg | Arn't you looking at + 200ms delay if you go phone -> 3000 -> * -> 3000 -> fxo on a wan though? |
| 01:25.13 | *** join/#asterisk Math` (n=math@modemcable120.4-81-70.mc.videotron.ca) |
| 01:25.20 | mindwarp | no idea, but sounds like something i'd want to know about |
| 01:27.05 | DoktorGreg | well as long as that spa 300? is local to asterisk you should be ok |
| 01:27.31 | DoktorGreg | but if it and * are connected by wan |
| 01:27.37 | mindwarp | i see |
| 01:28.18 | DoktorGreg | if you are already going asterisk why not one of the digium TDM400 variants? |
| 01:28.56 | DoktorGreg | gotta be comparable in price to that spa 300? |
| 01:29.04 | mindwarp | no reason, i don't have a clear idea about any of the hardware yet |
| 01:29.09 | mindwarp | just starting to look around |
| 01:29.18 | mindwarp | would you recommend a TDM card? |
| 01:29.40 | DoktorGreg | oh never mind |
| 01:29.46 | DoktorGreg | about 150 difference |
| 01:29.51 | DoktorGreg | and yes |
| 01:30.08 | DoktorGreg | TDM400p option with 2xfxo worked flawlessly for me |
| 01:30.12 | mog_work | woot |
| 01:30.17 | mindwarp | ok, but quite a bit more expensive i see |
| 01:30.22 | DoktorGreg | yup |
| 01:30.23 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
| 01:31.08 | mindwarp | still. might be worth it. i haven't even decided whether i want something to plug POTS phones in or directly a few IP phones, yet |
| 01:31.49 | DoktorGreg | http://www.voipsupply.com/product_info.php?&products_id=292 |
| 01:31.53 | DoktorGreg | thats the one i had |
| 01:31.57 | DoktorGreg | er have |
| 01:32.01 | DoktorGreg | if you wana buy it |
| 01:32.06 | DoktorGreg | ive since moved onto pri |
| 01:32.45 | mog_work | wish bri was popular in states |
| 01:33.22 | file[laptop] | can't you get a BRI in HSV? |
| 01:33.29 | Ariel_ | was humm at one time really. I have not seen any bri here at all. |
| 01:33.32 | mog_work | not for cheap |
| 01:33.37 | mog_work | if it was popular it would be |
| 01:33.40 | mog_work | or cheaper |
| 01:33.56 | mindwarp | so essentially this fxo thing is about taking a line and connecting many phones to it |
| 01:34.03 | DoktorGreg | no |
| 01:34.25 | DoktorGreg | fxo is about taking a pots phone line and connecting network or asterisk to it |
| 01:34.32 | DoktorGreg | fxs is the phone side |
| 01:34.45 | mindwarp | ok |
| 01:35.05 | mindwarp | can the other phones on that line then talk directly to asterisk? |
| 01:35.21 | {zombie} | no |
| 01:35.27 | *** join/#asterisk visio (n=visio@24.115.193.49.res-cmts.sth.ptd.net) |
| 01:35.27 | {zombie} | you'd need to plug those phones into the FXS port |
| 01:35.39 | mindwarp | ok good. makes sense |
| 01:35.42 | {zombie} | in fact you really don't want to put a spa3k on the same line as other phones |
| 01:36.00 | mindwarp | oh |
| 01:36.05 | {zombie} | but you can put it between the line and the phoens |
| 01:36.13 | {zombie} | erm, phones |
| 01:36.14 | mindwarp | through the fxs port |
| 01:36.17 | {zombie} | right |
| 01:36.20 | mindwarp | gotcha |
| 01:36.35 | {zombie} | just be aware that VoIP and analogue lines really don't mix too well |
| 01:36.40 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 01:36.43 | {zombie} | you'll get echoes |
| 01:36.48 | mindwarp | that must explain that report i read of someone frying a sipura by connecting the wrong thing to it |
| 01:36.59 | mindwarp | err must be related to rather |
| 01:37.04 | {zombie} | and the sipura's method of "dealing" with the echoes is to continuously fiddle with the txgain |
| 01:37.07 | mmlj4 | yeah, terminal echo will do that |
| 01:37.10 | {zombie} | so your volume will go up and down all the tim |
| 01:37.27 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 01:37.29 | DoktorGreg | ouch |
| 01:37.58 | kev009 | can multiple phones hang off on FXS? |
| 01:38.01 | mindwarp | alright. good thing i have a separate line to put asterisk onto then |
| 01:38.13 | Math` | kev009: that'd do the same as having 3 phones on the same line, for example |
| 01:38.25 | Math` | but you have to make sure the ring equivalent of the FXS port is high enough |
| 01:38.34 | Math` | or else your phones wont ring :P |
| 01:38.37 | kev009 | Math`: yes, it'd work fine? |
| 01:38.38 | terrapen | anybody ever looped a PRI port back to another on the same card? |
| 01:38.55 | terrapen | ie., send calls out one port and back through another |
| 01:39.13 | DoktorGreg | would need to be pri xover cable |
| 01:39.17 | terrapen | yep |
| 01:39.24 | terrapen | chan_zap.c:8970 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. |
| 01:39.31 | terrapen | I think I have something wrong in my zapata.conf |
| 01:39.38 | terrapen | somehow need to break the two ports off from each other |
| 01:39.43 | Math` | terrapen: use pri_cpe instead of pri_net |
| 01:39.54 | DoktorGreg | i know you can make a pri loopback adaptor |
| 01:39.55 | Math` | cpe is customer side, net is network side :) |
| 01:39.59 | mindwarp | that's a great error message |
| 01:40.04 | terrapen | math, same thing |
| 01:40.07 | terrapen | err: |
| 01:40.21 | {zombie} | one end needs to be pri_cpe, the other pri_net |
| 01:40.22 | terrapen | Apr 23 19:40:20 WARNING[32175]: chan_zap.c:8970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 01:40.35 | terrapen | ok, but how do I separate the "ends" in zapata.conf? |
| 01:40.39 | mmlj4 | someone should step in and explain that "cpe" stands for "customer-provided equipment" |
| 01:40.45 | terrapen | i understand that. |
| 01:40.47 | X-Rob | terrapen, remove the loopback adaptor from your PRI connection |
| 01:40.52 | terrapen | arghh |
| 01:40.56 | terrapen | I'm *trying* to loop it |
| 01:41.10 | X-Rob | you've done it. It works. Next? |
| 01:41.16 | terrapen | i want to test sending a call out over one port and back in another |
| 01:41.19 | X-Rob | you can't |
| 01:41.26 | terrapen | hrmm |
| 01:41.44 | X-Rob | you need another * box plugged into the other end of the cable |
| 01:41.52 | msw | terrapen: the best you can do is a pattern loopback test |
| 01:41.52 | X-Rob | or a telco. |
| 01:41.54 | X-Rob | or a pabx. |
| 01:41.56 | terrapen | ok |
| 01:41.57 | msw | terrapen: you can do that.... |
| 01:42.20 | terrapen | i'm playing around with this redfone fonebridge and wanted to test TDMoE |
| 01:42.30 | terrapen | i guess i'll just have to wait until i can wire a real PRI into it |
| 01:42.30 | msw | terrapen: look at patlooptest |
| 01:42.37 | terrapen | k |
| 01:42.39 | terrapen | thz |
| 01:42.41 | terrapen | err thanks |
| 01:43.08 | msw | like patlooptest /dev/zap/1 60 |
| 01:43.40 | terrapen | cool |
| 01:43.47 | terrapen | gotta make a loopback cable now :) |
| 01:43.52 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 01:44.25 | mog_work | msw!!!!!!!!!!!!!!!!!!! |
| 01:44.30 | tier_1 | so does asterisk fallow the nanpa layout for *xx and *11 exten ? |
| 01:44.38 | msw | mog!!!!1ONE |
| 01:44.43 | tier_1 | or is that a pipe dream |
| 01:44.48 | mog_work | however you program dialplan |
| 01:44.53 | mog_work | you can do any regex tier_1 |
| 01:45.24 | tier_1 | well I mapped all the nanpa code in the last 2 days |
| 01:45.27 | tier_1 | l;ol |
| 01:46.12 | tier_1 | http://pastebin.ca/50819 |
| 01:46.20 | tier_1 | just so you can see |
| 01:46.38 | tier_1 | I fallowed the nanpa layout |
| 01:46.59 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 01:47.21 | X-Rob | tier_1, you have to write the vertical service codes. |
| 01:47.28 | terrapen | weather.agi heh |
| 01:47.29 | *** join/#asterisk exten123 (n=exten1@60.49.6.190) |
| 01:47.31 | X-Rob | I think zaptel has some built in |
| 01:47.39 | terrapen | i should send you my rooftop weatherstation agi |
| 01:47.39 | tier_1 | I did |
| 01:47.53 | tier_1 | its all there |
| 01:48.16 | tier_1 | O you mean the checking stuff |
| 01:48.30 | tier_1 | thats next |
| 01:49.20 | tier_1 | this was for sip/iax lines |
| 01:49.47 | mog_work | msw whats up |
| 01:52.21 | *** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx) |
| 01:56.27 | exten123 | why there is no voice received in FXO port when I dial from softphone? |
| 01:58.25 | msw | mog_work: too much |
| 02:06.39 | ManxPower | exten123, Excellent question. Did you check the WiKi for information about one way audio |
| 02:17.13 | trelane | anyone with a Zyxel Prestige 2000V2 tell me where the fek they hid back space? |
| 02:18.45 | terrapen | heh |
| 02:19.03 | terrapen | the last decent ZyXEL product was the U-1496+ |
| 02:19.08 | terrapen | i still have one ;) |
| 02:19.10 | trelane | tend to agree |
| 02:19.16 | trelane | where was backspace on it? |
| 02:19.21 | terrapen | rofl |
| 02:19.36 | terrapen | actually, yeah. it had an LCD and keypad :) |
| 02:19.43 | {zombie} | trelane: it's the (get this) hangup button. |
| 02:19.52 | trelane | trying it |
| 02:20.01 | {zombie} | the UI on that phone blows donkey's bawlz |
| 02:20.25 | trelane | {zombie}, you are my hero |
| 02:20.42 | {zombie} | oh, and to change between upper/lower case and numbers, you don't use left/right (as the Aa1 thing implies) |
| 02:20.44 | {zombie} | but up/down |
| 02:30.36 | *** join/#asterisk jake1932 (n=Administ@pool-68-236-1-235.phil.east.verizon.net) |
| 02:31.33 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-144-118.gdrpmi.dsl-w.verizon.net) |
| 02:33.25 | *** join/#asterisk bigblockO2 (i=pplwro@ool-43504a4a.dyn.optonline.net) |
| 02:35.13 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 02:35.16 | PakiPenguin | morning |
| 02:37.52 | terrapen | howdy |
| 02:40.56 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 02:46.07 | terrapen | tier_1, that's an awesome dialplan |
| 02:46.20 | terrapen | i need a copy of that, outside of pastebin :) |
| 02:49.50 | Sedorox | isn't *70 universal for deactivate? not activate? |
| 02:50.13 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 02:50.16 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 02:51.08 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 02:52.09 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:53.33 | Supaplex | Sedorox: or 1170, depending on the $telco |
| 02:55.47 | Sedorox | I know east coast PA.. its *70... |
| 02:57.35 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 02:58.38 | ManxPower | NANPA has the standard codes for that region. |
| 02:58.56 | ManxPower | Its listed in the Wiki or a link to it is in the Wiki |
| 02:59.38 | Sedorox | ah |
| 02:59.46 | ManxPower | tier_1, For zap it follows many of them |
| 03:00.04 | ManxPower | tier_1, since for SIP it's the phones job to do those codes.... |
| 03:00.36 | op3r | is it ok to install asterisk without zaptel stuff? |
| 03:00.58 | ManxPower | op3r, as long as you don't need Zaptel timing or zaptel interfaces. |
| 03:02.25 | SpaceBass | MICS |
| 03:02.35 | SpaceBass | oops...sorry |
| 03:02.37 | ManxPower | SpaceBass, no need to be vulgar |
| 03:02.43 | SpaceBass | :) |
| 03:03.23 | op3r | <PROTECTED> |
| 03:03.29 | op3r | can anyone tell me that error? |
| 03:03.39 | jake1932 | aren't vars by default set for the call even after switching contexts? |
| 03:03.47 | ManxPower | op3r, that is not an error. |
| 03:03.47 | jake1932 | (the entire call) |
| 03:04.05 | ManxPower | jake1932, in general yes, with a few exceptions |
| 03:04.07 | op3r | ManxPower: because I was trying to barge a call using chanspy |
| 03:04.30 | ManxPower | op3r, it's still not an error. |
| 03:04.35 | terrapen | heh my boss just told me to take my poster down off the wall |
| 03:04.40 | op3r | ok |
| 03:04.45 | terrapen | pretty soon, they'll move me to Storage Room B |
| 03:05.00 | jake1932 | Set(REQ_EXTEN=${EXTEN}) should keep a copy of EXTEN in REQ_EXTEN, right? |
| 03:05.02 | ManxPower | terrapen, in the basement, right? |
| 03:05.06 | terrapen | yep. |
| 03:05.17 | terrapen | it was a cool poster, too! A K2 Telemark poster. |
| 03:05.24 | terrapen | and we're a skiing company! |
| 03:05.30 | ManxPower | jake1932, assuming you pasted and didn't retype it for the channel and fixed a typoe, yes |
| 03:05.43 | *** join/#asterisk Cardoe (n=Cardoe@gentoo/developer/Cardoe) |
| 03:05.45 | jake1932 | (no fixing :) |
| 03:05.50 | terrapen | he said that it didn't look good in our brand new expensive office space and that he would get me a framed ski photo to replace it with |
| 03:05.52 | Cardoe | So anyone have some neat at home features in their Asterisk setup? |
| 03:05.55 | ManxPower | jake1932, do you see that being run in the dialplan? |
| 03:05.59 | jake1932 | yep |
| 03:06.20 | ManxPower | jake1932, do you use a Local/ dial or anything? |
| 03:06.22 | jake1932 | looks like it goes through, but calling it is a different story |
| 03:06.32 | jake1932 | yes - i do use local |
| 03:06.55 | ManxPower | That is the exception |
| 03:07.00 | jake1932 | haha |
| 03:07.02 | jake1932 | of course |
| 03:07.22 | jake1932 | is there a workaround - i need this var |
| 03:07.26 | ManxPower | jake1932, try using /n at the end of the dial for Local/ channels, or just read "show application set" and notice the _ prefix stuff. |
| 03:07.48 | jake1932 | will do... tnx |
| 03:08.09 | ManxPower | terrapen, Framing the EXISTING poster might be cheaper. |
| 03:09.14 | terrapen | well, it's just a cheesy poster i got out of a telemark magazine |
| 03:09.32 | terrapen | not worth framing |
| 03:09.48 | CrashHD | anyone know of a system that support bridged line appearances? |
| 03:09.48 | ManxPower | make him frame it just for punishment. |
| 03:09.57 | terrapen | it just kind of pissed me off...i'm here, the only person in the office, working my ass off on a sunday night |
| 03:10.07 | terrapen | and he stops by and asks me to take a fucking poster down! |
| 03:10.13 | ManxPower | CrashHD, you mean other than "hint" |
| 03:10.22 | CrashHD | hint? |
| 03:10.25 | terrapen | oh well |
| 03:10.42 | ManxPower | CrashHD, read the wiki for "BLF" or "busy line field" |
| 03:11.00 | CrashHD | ok |
| 03:11.02 | jake1932 | ManxPower - tnx again - that worked |
| 03:11.27 | ManxPower | CrashHD, just remember that polycoms can't currently support more than 8 BLFs |
| 03:11.43 | terrapen | i wonder how much it would cost to properly port Zaptel to OpenBSD |
| 03:11.54 | CrashHD | man what phone would you recommend? |
| 03:11.55 | ManxPower | and that includes their sodecars |
| 03:11.56 | terrapen | like, production-quality |
| 03:12.00 | ManxPower | sidecars, that is. |
| 03:12.06 | ManxPower | CrashHD, I recommend not using BLF |
| 03:12.26 | CrashHD | what about bridged line appearance? |
| 03:12.28 | CrashHD | not blf? |
| 03:12.38 | ManxPower | define the difference? |
| 03:13.02 | CrashHD | one monitors status of a phone line, other monitors the phone |
| 03:13.11 | CrashHD | line keys basically |
| 03:13.12 | CrashHD | line 1 |
| 03:13.13 | CrashHD | line 2 |
| 03:13.14 | CrashHD | line 3 |
| 03:13.15 | CrashHD | line 4 |
| 03:13.16 | ManxPower | better yet, read the damn wiki or the mailing list archives. The topic seems to come up every week on the mailinglist. |
| 03:13.33 | CrashHD | lol |
| 03:13.37 | CrashHD | I've read through it |
| 03:13.49 | CrashHD | I know * doesn't support bla yet |
| 03:13.55 | CrashHD | was hoping someone knew of a system that does |
| 03:14.30 | ManxPower | Nortel does. |
| 03:14.44 | CrashHD | freeware/opensource systems |
| 03:15.00 | CrashHD | sorry |
| 03:15.01 | ManxPower | CrashHD, in your dreams. |
| 03:15.03 | CrashHD | lol |
| 03:15.06 | CrashHD | ya basically |
| 03:16.46 | jake1932 | was there a change in outbound callerid between 1.2.1 and 1.2.7 (it used to work using SIP)? |
| 03:17.04 | Math` | jake1932: no |
| 03:17.50 | jake1932 | Math` - tnx |
| 03:26.56 | *** join/#asterisk hodrige (n=Hodrige@ip68-98-172-123.dc.dc.cox.net) |
| 03:27.10 | hodrige | does anyone knows much about Sipura's SPA-3000 ? |
| 03:28.52 | jake1932 | i just did a SIP debug and it shows my RPID is being sent? anything else to check to see about the called party not getting CID? |
| 03:29.53 | jake1932 | hodrige: whats your question about it? |
| 03:31.22 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 03:31.28 | k-man | hello |
| 03:31.47 | Math` | jake1932: what's receiving the call |
| 03:31.59 | k-man | what kind of quality can i expect from voip using different codecs? is there a chart that compares them? |
| 03:32.05 | jake1932 | Math - regular POTS analog |
| 03:32.19 | Math` | whats the ata |
| 03:32.38 | jake1932 | no ATA (7960 on the asterisk side) |
| 03:33.02 | jake1932 | analog is connected to the CO |
| 03:33.09 | Math` | how are you sending calls out |
| 03:33.23 | jake1932 | VOIP terminiation (junction networks) |
| 03:33.27 | Math` | check with your provider if CID goes through... |
| 03:33.36 | jake1932 | it was - before i upgraded |
| 03:33.46 | Math` | maybe the provider changed policy without notice :P |
| 03:33.54 | jake1932 | ah - that never happens |
| 03:33.56 | jake1932 | :) |
| 03:34.10 | jake1932 | good point - i'll check tomorrow |
| 03:34.12 | Math` | in sip debug, is the From field containing the proper CID? |
| 03:34.42 | jake1932 | lemme rerun - i know RPID looked proper |
| 03:35.27 | jake1932 | hmm - that's not right - it says my user name |
| 03:35.50 | Math` | then set it properly before making the call |
| 03:36.00 | Math` | set CALLERID(number) and CALLERID(name) accordingly to what you need |
| 03:36.23 | k-man | anyone here have any experience with an NEC Xen phone system? |
| 03:36.52 | jake1932 | Math`- i'll try that - i had it that way before, but no luck, no harm in doing it again though |
| 03:37.25 | Math` | k-man: g.711u/a (ulaw/alaw) will always get you the best quality, most providers are using g729 for its less-consuming bandwidth |
| 03:37.25 | jake1932 | actually - i take that back - it does say that |
| 03:37.42 | jake1932 | exten => _NXXNXXXXXX,2,Set(CALLERID(number)=mynum) |
| 03:37.49 | Math` | oh |
| 03:38.05 | Math` | then the From: field should be set properly |
| 03:38.07 | k-man | Math`, so... whast the quality like of g729 compared to a pots line? |
| 03:38.39 | hodrige | jake1932: I cant receive calls on the SPA-3000 |
| 03:38.45 | hodrige | I can dial out |
| 03:38.55 | Math` | k-man: nothing's best than trying it |
| 03:39.16 | Math` | tho you should check on voip-info.org to see if any comparison table exists |
| 03:39.23 | jake1932 | hodrige: receive through the FXO port? |
| 03:39.32 | hodrige | FSX |
| 03:39.36 | Math` | ulaw is what POTS lines are being encoded into once at the telco |
| 03:40.08 | jake1932 | hodrige: did you check the status screen - does it show ringing? |
| 03:40.26 | hodrige | let me try ... just refresh? |
| 03:40.29 | jake1932 | yes |
| 03:42.07 | *** join/#asterisk angom_h (n=angom@red-corp-201.143.97.166.telnor.net) |
| 03:42.47 | jake1932 | hodrige: the SPA 3000 has many different configs - assuming you have it config'd properly, you should be able to see it registered with your asterisk box as an FXO and FXS (essentially two different devices) |
| 03:43.08 | *** part/#asterisk angom_h (n=angom@red-corp-201.143.97.166.telnor.net) |
| 03:45.31 | hodrige | I understand... but for troubleshooting I took the asterisk out of the equation. I am just using the LINE1 tab and uting it as an ATA with a phone connected to it |
| 03:45.53 | hodrige | I cant see a place where the status says ringing |
| 03:46.09 | *** join/#asterisk mick_linux (n=mick@adsl-10-43-125.mia.bellsouth.net) |
| 03:46.13 | jake1932 | Math` - looks like a provider issue - i tried another provider and everything works - you were right on |
| 03:46.40 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 03:47.10 | jake1932 | hodrige: it should say on hook (or something like that) on the status page until you call it |
| 03:47.18 | hodrige | it saus on hook |
| 03:47.27 | jake1932 | now call it - see if that changes |
| 03:47.33 | hodrige | or hook state on |
| 03:47.38 | hodrige | it doesnt |
| 03:47.39 | jake1932 | (you may have to refresh) |
| 03:47.41 | hodrige | I refreshed |
| 03:47.51 | jake1932 | does asterisk show it's trying the exten? |
| 03:48.16 | jake1932 | any errors in the asterisk? |
| 03:48.49 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.25) |
| 03:49.26 | hodrige | I am using PSTN to call my SIP # no asterisk is envolved |
| 03:49.47 | hodrige | I have a phone connected to the FSX port |
| 03:49.51 | Math` | jake1932: ok, call them and ask em for a month off :P |
| 03:49.56 | wunderkin | hmmmm someone is doing something nasty with my packets somewhere |
| 03:50.00 | jake1932 | rcan't hurt :) |
| 03:50.44 | hodrige | I can configure it to talk to the asterisk and see if it rings ... |
| 03:50.49 | hodrige | Brilliant! |
| 03:51.13 | jake1932 | hodrige - we're missing something - can't do PSTN -> FXS (SPA 3000) directly |
| 03:52.43 | hodrige | I am calling my SIP providers # |
| 03:53.20 | jake1932 | here i'm thinking this was an asterisk question :) |
| 03:53.22 | *** join/#asterisk bmg505 (n=leon@dsl-146-54-137.telkomadsl.co.za) |
| 03:53.36 | hodrige | It will be :) |
| 03:53.37 | *** join/#asterisk redondos_ (n=redondos@190.48.41.29) |
| 03:54.55 | joelsolanki | Hello all |
| 03:55.52 | joelsolanki | I am little confuse with buying digium product. Actually i need a pstn card which provides 4 port for pstn connectivity. This is for just testing. which card should i take from digium ?/ |
| 03:56.16 | joelsolanki | tdm400p ? |
| 03:56.34 | jake1932 | they have models bundled with FXO ports |
| 03:56.51 | jake1932 | tdm400p is not loaded with an modules |
| 03:56.54 | jake1932 | any |
| 03:57.04 | LostFrog | TDM04B |
| 03:57.11 | joelsolanki | ahh |
| 03:57.14 | joelsolanki | modules ? |
| 03:57.20 | jake1932 | The naming convention for the TDM bundles is as follows: TDM X Y B. Where "TDM" denotes that the card is TDM, "X" denotes the number of FXS modules, "Y" denotes the number of FXO modules, and "B" indicates that that this product is a bundle. |
| 03:57.36 | jake1932 | you would need 4 FXOs |
| 03:57.56 | jake1932 | TDM04B |
| 03:58.14 | joelsolanki | so i should buy TDM04B ? |
| 03:58.15 | LostFrog | Or a TDM2401B |
| 03:58.24 | LostFrog | or TDM2401E |
| 03:58.50 | jake1932 | when's that product selector coming out? |
| 03:59.20 | joelsolanki | but this cards are not in the product section of digium |
| 03:59.43 | LostFrog | You can't buy bundles directly from digium. |
| 03:59.48 | LostFrog | You have to find a reseller. |
| 03:59.56 | joelsolanki | oh. |
| 04:00.05 | jake1932 | http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?category_id=17&product_code=RTDM04B |
| 04:00.46 | *** join/#asterisk gursikh (n=m@adsl-68-92-60-60.dsl.hstntx.swbell.net) |
| 04:00.56 | LostFrog | Hmm.. The TDM04B is on the digium store now. |
| 04:01.21 | joelsolanki | ok good. |
| 04:01.23 | LostFrog | The retail version. |
| 04:01.47 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 04:01.49 | LostFrog | As is the TDM2401E |
| 04:02.07 | joelsolanki | this is just for using 2 ports. now if i want to have huge pstn lines. which product is good. |
| 04:02.25 | LostFrog | A channel bank. IMMHO |
| 04:02.36 | LostFrog | Or a PRI from the telco. |
| 04:04.16 | LostFrog | If you only need two lines, two IAXys would be cheaper. |
| 04:04.45 | joelsolanki | hmm ok. |
| 04:06.46 | LostFrog | wow.. digium charges $2000 for a fairly basic configuration. |
| 04:07.17 | Strom_C | LostFrog: for a card? or to hire someone to configure your box? |
| 04:07.38 | LostFrog | to configure a box. |
| 04:08.08 | Strom_C | well it's not exactly a huge company. there are plenty of consultants out there who will be happy to do it for less |
| 04:08.09 | joelsolanki | as per my understanding i have to buy analog interface cards and analog modules too ? |
| 04:08.55 | LostFrog | joelsolanki: If you get the bundle it includes the card and the modules. |
| 04:10.14 | joelsolanki | ok got it. |
| 04:11.03 | LostFrog | and mousepad. |
| 04:11.27 | joelsolanki | :) |
| 04:11.41 | Strom_C | I've got at least a dozen of both of those lying around my apartment :) |
| 04:11.55 | LostFrog | Biatch. :) |
| 04:11.59 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 04:12.43 | Sedorox | hmmm |
| 04:13.02 | joelsolanki | but if i create epbx system with tdm400p card. it is too costly |
| 04:13.59 | joelsolanki | bcoz tdm04B retail package costs $421 ...here we get $421 normal company pbx |
| 04:15.24 | LostFrog | A PBX or a key system? |
| 04:16.13 | joelsolanki | pbx |
| 04:16.48 | joelsolanki | i m finding other pstn card which are cheap and works with asterisk. |
| 04:17.48 | LostFrog | Hopefully not X100 clones. |
| 04:18.26 | hodrige | <PROTECTED> |
| 04:18.35 | hodrige | It is working with asrweisk |
| 04:18.54 | hodrige | I even put it outside the firewall |
| 04:19.08 | hodrige | I can dial out but not in |
| 04:22.20 | *** join/#asterisk beernuts71 (n=mattfox7@CPE-58-167-170-122.qld.bigpond.net.au) |
| 04:22.38 | hodrige | this is so weird! |
| 04:23.15 | gursikh | Hello, I was wondering if I could get some opinions on what kind of system I need. |
| 04:23.30 | beernuts71 | can anyone tell me if you can pass DNIS to an IAX2 softphone such as IDEFISK, I can see ANI no problem |
| 04:23.41 | gursikh | If anyone knowledgable about this stuff (in general sense) |
| 04:25.39 | beernuts71 | yooohooo anyone here |
| 04:26.14 | websae | howdy |
| 04:26.16 | websae | how fair you |
| 04:27.00 | *** join/#asterisk ph|ber (i=phiber@slackwaresupport.com) |
| 04:27.14 | ph|ber | anyone using the weather script ? |
| 04:28.38 | ManxPower | ph|ber, there are a billion weather scripts |
| 04:29.01 | ph|ber | hrm. ok it is the one that comes with *@home |
| 04:29.10 | NoName32 | hi all i am still pretty new to asterisk got a problem i cant figure out .. working on the one touch record using the featuremap with automon => *1 but it doesnt seem to be reconizing the *1 if i change it to ** it works any ideas/ sugestions i am using asterisk 1.2.6 |
| 04:29.11 | ManxPower | ph|ber, look at the /topic |
| 04:29.39 | ph|ber | yea, i asked in there.. |
| 04:29.45 | ph|ber | acually X-Rob is in here and there. |
| 04:30.54 | X-Rob | Flashing names. |
| 04:33.14 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:34.49 | tier_1 | ? |
| 04:35.37 | tier_1 | back |
| 04:35.58 | tier_1 | yeah but most sip phones dont all have them built in manx |
| 04:36.10 | tier_1 | so I did it just to get it done |
| 04:36.37 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:40.55 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-217-245.hsd1.il.comcast.net) |
| 04:41.14 | kuku5 | How come when an agent logs on, and then hangs up, it doesnt always free the session ? |
| 04:47.17 | tier_1 | night kids |
| 04:47.20 | *** part/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 04:53.47 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 04:56.31 | MGSsancho | ni ni |
| 05:08.11 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 05:09.58 | ManxPower | Any EuroPeople here? |
| 05:10.17 | ManxPower | it's a not a technical or asterisk question |
| 05:10.42 | CrashHD | manx just wants to know if the women really don't shave |
| 05:11.18 | CrashHD | humor him guys |
| 05:11.18 | CrashHD | :) |
| 05:11.41 | ManxPower | CrashHD, Actually in the USA I've NEVER seen a temp/water control in the shower that allows you to specify a preset temp. I saw them in Netherlands, Belgium, and Spain. I want one. |
| 05:11.56 | ManxPower | But I have NO idea what the search terms would be. |
| 05:12.22 | CrashHD | like instead of a nob you enter in a temp? |
| 05:13.19 | ManxPower | CrashHD, a nob that has a temp dial on it. You just turn it to the temp you want, wait a few moments. |
| 05:13.27 | ManxPower | no trying to adjust the hot and cold water. |
| 05:13.38 | CrashHD | ohh |
| 05:13.39 | CrashHD | sweet |
| 05:13.49 | CrashHD | like no cold water comes out |
| 05:13.50 | ManxPower | CrashHD, *nod* *nod* *nod* |
| 05:13.53 | CrashHD | waiting for the hot water |
| 05:14.18 | ManxPower | CrashHD, well, yes it does. the water is not heated in the fixture. |
| 05:14.44 | ManxPower | But no trying to adjust the hot and cold water to the temp you want. I assume there's some device that mixes the hot and cold water to create the temp you want. |
| 05:16.21 | ManxPower | Aha! http://cgi.ebay.com/THERMOSTATIC-SHOWER-MIXER-INC-SHOWER-HEAD-RAIL-HOSE_W0QQitemZ4457810047QQcategoryZ32875QQrdZ1QQcmdZViewItem |
| 05:16.34 | CrashHD | ohh ya |
| 05:16.43 | CrashHD | I just used one of those |
| 05:16.53 | CrashHD | on my cruise |
| 05:16.53 | CrashHD | about a week ago |
| 05:17.23 | hads|home | They are usually associated with an instant hot water system. e.g gas or electrically heated on the fly without a hot water tank. |
| 05:17.39 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 05:17.48 | ManxPower | hads|home, I plan on getting a tankless water heater. |
| 05:18.05 | hads|home | Cool. |
| 05:19.20 | ManxPower | I'm having a cabin built and there are a few things I REALLY want, one of the things is a tankless propane water heater and thermostatic mixer shower |
| 05:20.41 | exten123 | what the problem with my fxs port no is connected but at caller side still hear the ring tone? |
| 05:21.35 | ManxPower | exten123, that is normal |
| 05:21.37 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 05:21.45 | ManxPower | don't call the port if there's no phone connected into it. |
| 05:21.53 | ManxPower | the same happens with the PSTN analog |
| 05:22.03 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 05:22.57 | exten123 | <ManxPower>, I mean that the fxs port is pickup by some one else already. |
| 05:23.18 | ManxPower | exten123, turn off call waiting |
| 05:25.28 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 05:25.47 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-168-37.hsd1.ca.comcast.net) |
| 05:26.08 | tengulre11 | Hi,all!! |
| 05:26.41 | *** join/#asterisk lorinc (n=ang@caracas-1730.adsl.interware.hu) |
| 05:26.57 | exten123 | it like cannot bridged between caller party with receiver party |
| 05:27.51 | exten123 | although receiver party hear ringing tone and connect until my fxs port. afterward it not bridge to my softphone. |
| 05:31.57 | *** join/#asterisk shidan (i=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 05:32.10 | shidan | hi anyone here expert with vicidial? |
| 05:39.08 | ManxPower | exten123, you may want to ask on the mailing lists. |
| 05:43.11 | *** join/#asterisk NoRemorse (n=bah@203-214-92-100.dyn.iinet.net.au) |
| 05:43.37 | NoRemorse | hello, can anyone tell me if there is a SIP handset that shows trunk line number in use? a-la turnkey pabx style |
| 05:44.24 | CrashHD | is there a newsgroup type solution for the asterisk mailing lists? |
| 05:44.30 | CrashHD | rather than me getting all the email? |
| 05:45.59 | NoRemorse | pretty quiet in her |
| 05:46.01 | NoRemorse | e |
| 05:47.10 | CrashHD | 11pm PST |
| 05:47.19 | CrashHD | bed time for most |
| 05:47.32 | NoRemorse | yeah |
| 05:51.09 | exten123 | Can't detect FXS is been pickup how to solve? |
| 05:51.32 | *** part/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 05:59.04 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-60-235.dsl.irvnca.pacbell.net) |
| 06:05.47 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 06:11.38 | DoktorGreg | whats the downside to windows 64 pro other than... its windows? |
| 06:12.02 | DoktorGreg | Im finding 2 gig is not quite enough |
| 06:12.20 | DoktorGreg | and as i understand... xp pro has trouble with more than 3 gig |
| 06:12.40 | DoktorGreg | so id like to go with 6-8 gig on my desktop |
| 06:13.09 | DoktorGreg | anyone? |
| 06:15.47 | tainted- | DoktorGreg you're talking ram right |
| 06:15.54 | DoktorGreg | yah |
| 06:16.08 | tainted- | why not just install and find out |
| 06:16.47 | DoktorGreg | well it looks like i need a specialty mobo just to get more than 4 gig of ram |
| 06:17.17 | Shaun222 | why do you need 6-8GB of ram? |
| 06:17.19 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:17.21 | Shaun222 | on a desktop |
| 06:17.28 | DoktorGreg | well by the time i run everything |
| 06:17.36 | Shaun222 | everything? |
| 06:17.40 | DoktorGreg | plus i wana be able to run a couple of virtual machines |
| 06:17.45 | tainted- | Shaun222 the internet duh |
| 06:17.52 | Shaun222 | tainted-: lol |
| 06:18.22 | DoktorGreg | I duno, i have 2 gig right now |
| 06:18.23 | Shaun222 | DoktorGreg: virtual machines makes sense i guess, if you plan on spawning a bunch of devel... |
| 06:18.39 | DoktorGreg | when you get a modern dev enviornment up |
| 06:18.40 | Shaun222 | i usually run those on a seperate server those :) |
| 06:19.02 | DoktorGreg | then an adobe app or two |
| 06:19.14 | DoktorGreg | and it starts swapping all over the place |
| 06:19.30 | DoktorGreg | I guess I need to just bite bullet and buy new server |
| 06:19.33 | tainted- | why would u want to run on a separate server when u could run it all from the desktop |
| 06:20.01 | DoktorGreg | but man, i had two years uptime on this server once |
| 06:20.16 | DoktorGreg | i dont wanna mess with that |
| 06:20.19 | Shaun222 | tainted-: ha ha, [Customer Notice] - I need to restart my desktop all customer Virtual Servers will go down for a second!!!! |
| 06:20.35 | tainted- | lol |
| 06:20.52 | Shaun222 | DoktorGreg: on a windows server? |
| 06:20.55 | Shaun222 | thats pretty decient. |
| 06:20.55 | DoktorGreg | no |
| 06:20.58 | Shaun222 | haha |
| 06:21.01 | tainted- | i like to have a separate vm for each domain i host for redundancy |
| 06:21.05 | DoktorGreg | slackware |
| 06:21.11 | tainted- | that way if one domain crashes it doesn't affect the others |
| 06:21.12 | DoktorGreg | + samba |
| 06:21.24 | DoktorGreg | + few other smallish things |
| 06:21.37 | tainted- | if u can't throw more hardware at it, you're not doing it right |
| 06:22.08 | DoktorGreg | it has a 2.2 kernel on it |
| 06:22.38 | DoktorGreg | the hard drives are probably gonna go in next couple of years |
| 06:23.06 | Shaun222 | DoktorGreg: suprised it hasnt been rooted yet with that old of a kernel. |
| 06:23.33 | DoktorGreg | its behind a vpn + firewall |
| 06:23.42 | Shaun222 | ahh, ok |
| 06:23.44 | DoktorGreg | hows it gonna get rooted? |
| 06:24.00 | DoktorGreg | my email server got rooted once |
| 06:24.24 | DoktorGreg | that is a windows box |
| 06:24.33 | DoktorGreg | 2000 pro |
| 06:25.23 | DoktorGreg | i guess thats not rooted as much as |
| 06:25.28 | DoktorGreg | sploited |
| 06:26.13 | DoktorGreg | windows - firewall gets sploited in short order |
| 06:27.11 | DoktorGreg | wow, when i set up the samba server though |
| 06:27.15 | DoktorGreg | samba was hard |
| 06:27.20 | DoktorGreg | now its just... |
| 06:27.26 | DoktorGreg | apt-get install samba |
| 06:27.47 | Shaun222 | what was so hard about it before? |
| 06:28.09 | DoktorGreg | it was before they had shares and locks totally worked out |
| 06:28.28 | DoktorGreg | so i had to tinker with share and lock settings per share |
| 06:29.13 | DoktorGreg | also, i run it as a domain server |
| 06:29.27 | DoktorGreg | and it took a while for me to figure out all the hairy parts |
| 06:29.39 | DoktorGreg | of getting a computer to join the domain |
| 06:30.10 | DoktorGreg | then there were all the things that windows did automagically |
| 06:30.16 | DoktorGreg | like sync the clocks |
| 06:30.25 | DoktorGreg | printer shares |
| 06:30.30 | DoktorGreg | oh that was a big one |
| 06:30.42 | DoktorGreg | getting the printer drivers to install and connect properly |
| 06:31.30 | DoktorGreg | I had to write a script somewhere in there to get lpr to talk propery to the mopier |
| 06:33.06 | DoktorGreg | also, it was the first time i set up a raid for linux |
| 06:33.22 | DoktorGreg | and it was in the bad old days of linux + software raid |
| 06:34.27 | DoktorGreg | wow, and those 40gig hard drives were HUGE at the time |
| 06:35.06 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-13.claranet.co.uk) |
| 06:35.21 | DoktorGreg | next one will be a pair of 750's |
| 06:35.46 | DoktorGreg | That should hold me for another 5 years |
| 06:39.38 | X-Rob | DoktorGreg, it doesn't work like that |
| 06:39.51 | X-Rob | I bought 3 250's for 500G radi5 |
| 06:40.02 | X-Rob | about 18 months ago |
| 06:40.04 | X-Rob | 94% full now |
| 06:40.05 | X-Rob | 8-\ |
| 06:40.28 | DoktorGreg | roll your logs man!:P |
| 06:40.47 | X-Rob | It's bittorrent! |
| 06:41.05 | X-Rob | I should go and clean it up |
| 06:41.06 | DoktorGreg | lol |
| 06:41.13 | X-Rob | do I really need 3g of Mandrake 10.0? |
| 06:41.17 | X-Rob | I don't think so. |
| 06:41.18 | *** join/#asterisk Hali_303 (n=surfk@dsl5402AC97.pool.t-online.hu) |
| 06:41.29 | DoktorGreg | debian comes on 8 cd's for the full thing |
| 06:41.51 | *** part/#asterisk Hali_303 (n=surfk@dsl5402AC97.pool.t-online.hu) |
| 06:41.53 | DoktorGreg | full version of adobe cs2 is on 3 dvd |
| 06:42.07 | DoktorGreg | really??? 3 dvd?? |
| 06:42.25 | kamileon | uhm, why not just use an older version |
| 06:42.28 | DoktorGreg | granted some of those are tutorials |
| 06:43.16 | DoktorGreg | well, photoshop and illustrator dont seem to change much version to version anymore |
| 06:43.28 | DoktorGreg | but adobe 7 was a huge upgrade |
| 06:43.45 | DoktorGreg | if you dont have adobe 7 yet, its worth the upgrade |
| 06:44.04 | tainted- | X-Rob lol |
| 06:44.08 | DoktorGreg | go live is not mature yet so it gets better every bersion |
| 06:44.10 | tainted- | pretty pretty isos |
| 06:44.33 | DoktorGreg | see, you guys are gonna hate this |
| 06:44.36 | tainted- | debian is 8 cds? |
| 06:44.36 | tainted- | omg |
| 06:44.56 | DoktorGreg | but the RIAA has successfully scart me away from the general file sharing apps |
| 06:45.15 | DoktorGreg | tainted-, they also have a single network install cd |
| 06:45.23 | DoktorGreg | which IMO is the way to go |
| 06:45.57 | DoktorGreg | I use bittorrent though |
| 06:46.17 | DoktorGreg | but i have a multi channel dvr system running |
| 06:46.22 | kamileon | DoktorGreg : Usenet is the way to go for all of your binary download needs |
| 06:46.39 | tainted- | usenet is slow |
| 06:46.43 | kamileon | bullshit |
| 06:46.57 | DoktorGreg | comcast doesnt give us a usenet server to use for free |
| 06:46.58 | tainted- | unless u have some premium subscription someplace |
| 06:46.58 | kamileon | i get a constant 700Kb/s on my home cable modem |
| 06:47.18 | DoktorGreg | wait i take that back |
| 06:47.19 | kamileon | comcast gives you service through easynews |
| 06:47.28 | DoktorGreg | there is a 1 gig a month service they give for free |
| 06:47.29 | tainted- | is easynews free? |
| 06:47.33 | kamileon | 1gb, but you can pop the limit easily |
| 06:47.52 | tainted- | how |
| 06:48.19 | DoktorGreg | I mostly use my 1.5Mb Up 8Mb down for irc though |
| 06:48.29 | DoktorGreg | irc was just too slow before now |
| 06:48.29 | kamileon | its in the way usenet works though, you just have to queue up alot to d/l and then throttle it down as slow as possible so it never resets the connection and just keeps requesting more articles, then hope your connection doesnt go down.. |
| 06:48.49 | kamileon | but i pay 19.99/mo for unlimited downloads through usenetserver.com and its very worth it |
| 06:49.00 | tainted- | wow that sounds way more easier than irc or bittorrent |
| 06:49.09 | tainted- | and by way more easy i mean fuck that shit |
| 06:49.13 | kamileon | since its not p2p, you never send a damn thing to any other user.. so they cant get you for any type of distribution if you never upload and just leech |
| 06:49.43 | DoktorGreg | I just buy all my stuff |
| 06:50.02 | kamileon | well i only download things that i have already purchased anyways |
| 06:50.05 | DoktorGreg | it doesnt get easier than just going to new egg and keying in a cc# |
| 06:50.26 | DoktorGreg | plus the free stuff! |
| 06:50.30 | DoktorGreg | I love free stuff |
| 06:50.34 | DoktorGreg | like... |
| 06:50.36 | DoktorGreg | asterisk! |
| 06:50.39 | DoktorGreg | woo hoo |
| 06:50.41 | kamileon | DoktorGreg : im sure its great to do that, but for those of us less fortunate |
| 06:51.25 | kamileon | oh yeh asterisk rocks! |
| 06:51.27 | DoktorGreg | well one aspect is this |
| 06:51.34 | DoktorGreg | on my network of computers |
| 06:51.35 | kamileon | i have to get some of the new orange stickers |
| 06:51.46 | DoktorGreg | we process credit card numbers |
| 06:51.59 | DoktorGreg | and we have to let the cc company do quarterly audits of our systems |
| 06:52.12 | DoktorGreg | they only have come once so far |
| 06:52.18 | DoktorGreg | when we first started processing |
| 06:52.39 | kamileon | is that good or bad? |
| 06:52.52 | DoktorGreg | well, I could pass a software audit tomorrow |
| 06:52.57 | DoktorGreg | ... probably |
| 06:53.02 | DoktorGreg | but |
| 06:53.09 | DoktorGreg | they share results with BSA |
| 06:53.14 | DoktorGreg | ... |
| 06:53.23 | kamileon | ahh |
| 06:53.24 | kamileon | i see |
| 06:53.46 | DoktorGreg | I know there are several versions of winzip laying around... |
| 06:53.55 | DoktorGreg | i should go replace those with 7zip |
| 06:54.07 | DoktorGreg | 7zip works better anyhow |
| 06:54.11 | kamileon | oh? |
| 06:54.14 | kamileon | works on .rar? |
| 06:54.20 | DoktorGreg | iirc |
| 06:54.38 | DoktorGreg | http://www.7-zip.org/ |
| 06:55.27 | kamileon | thanks |
| 06:55.48 | DoktorGreg | anyhow |
| 06:55.58 | DoktorGreg | fear of audits keep me honest |
| 06:56.22 | DoktorGreg | also |
| 06:56.32 | kamileon | yeh i dont have that to worry about.. quite yet.. |
| 06:56.40 | DoktorGreg | they are a great scapegoat for when i do the bastard operator from hell routein |
| 06:56.48 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 06:56.58 | kamileon | thats always fun ;) |
| 06:57.31 | DoktorGreg | anyhow |
| 06:57.37 | kamileon | holy shit! |
| 06:57.42 | kamileon | im watching bones right |
| 06:57.47 | kamileon | theyre doing a video analysis |
| 06:57.52 | kamileon | and for the first time ever on tv |
| 06:58.16 | kamileon | one of the guys said "the less pixels there are, the more they degrade when you zoom in. I can't get a clear picture from this" |
| 06:59.17 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 06:59.17 | DoktorGreg | you could use one of the techniques detailed by richard hoagland on art bell to get more resolution right? |
| 06:59.31 | kamileon | hahaha |
| 06:59.33 | kamileon | i love art bell |
| 07:00.01 | Qwell | kamileon: Strom_C loves bell art |
| 07:00.16 | kamileon | im sure he does. |
| 07:01.19 | kamileon | oooh * for the psp! |
| 07:01.34 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 07:03.54 | Qwell | wtf? |
| 07:04.12 | Qwell | Seriously? |
| 07:07.43 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za) |
| 07:11.39 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 07:18.39 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 07:20.54 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 07:35.55 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:40.23 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:43.29 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:43.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:47.24 | *** join/#asterisk inv_arp[work] (i=junya@adsl-10-157-112.mia.bellsouth.net) |
| 07:56.33 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.25) |
| 08:04.27 | Shaun222 | by the looks of it a agent can only be logged into one phone? |
| 08:10.16 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-197-00.telkomadsl.co.za) |
| 08:18.10 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 08:18.28 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:18.36 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 08:19.25 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:23.14 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:24.07 | *** join/#asterisk acehunky (n=chat_jok@59.184.60.129) |
| 08:25.53 | acehunky | any one here from bangladesh |
| 08:28.29 | Supaplex | yea |
| 08:28.39 | Supaplex | my imagination! |
| 08:28.39 | *** join/#asterisk inv_Arp (i=junya@adsl-10-157-112.mia.bellsouth.net) |
| 08:30.55 | *** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net) |
| 08:39.14 | acehunky | :-" |
| 08:39.37 | acehunky | i m trying to find asterisk consultants in bangladesh .. is this the right place ? |
| 08:40.13 | dlynes | possibly you might be able to find someone here |
| 08:40.25 | dlynes | but you could try asking on the asterisk-biz mailing list, too |
| 08:41.15 | acehunky | umm ok |
| 08:41.19 | *** join/#asterisk Modcuts (n=bob@82.133.98.155) |
| 08:41.19 | dlynes | if you can't find anyone from bangladesh on there, you'll at least find someone in a neighboring country |
| 08:41.35 | dlynes | I've seen people on there from India, for sure |
| 08:41.47 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 08:42.10 | joelsolanki | I m here from india :) |
| 08:42.15 | dlynes | there ya go |
| 08:42.18 | acehunky | i can see the asterisk india website .. and consultant list from india .. but need someone from bangladesh .. but thanks for the info |
| 08:42.37 | acehunky | joelsolanki .. which part of india ? |
| 08:42.44 | joelsolanki | gujarat |
| 08:43.26 | acehunky | okk |
| 08:43.46 | acehunky | so u know of any consultant in bangladesh ? |
| 08:44.33 | joelsolanki | sorry dont know |
| 08:46.16 | acehunky | okk |
| 08:49.13 | wasim | will a consultant in former bangladesh do? |
| 08:53.04 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 08:53.31 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:54.32 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:56.13 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
| 08:56.16 | xbit` | hi |
| 08:58.04 | dlynes | wasim: former bangladesh is the indian province of bangladesh? |
| 08:58.18 | dlynes | heya xbit` |
| 08:58.49 | xbit` | which channel does txfax sends the tif file i'm giving? so how can i give the number to txfax to send it to? |
| 08:59.17 | dlynes | xbit`: whatever channel you answered on |
| 08:59.47 | *** join/#asterisk lorinc (n=ang@caracas-1669.adsl.interware.hu) |
| 08:59.54 | dlynes | xbit`: I don't know if it works on sip or not, but lots of people have it working on zaptel channels |
| 09:00.12 | acehunky | wasim: former bd is wat pakistan .. or india ? |
| 09:00.30 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 09:00.38 | *** join/#asterisk stoffell (n=stoffell@dD57664D6.access.telenet.be) |
| 09:00.42 | dlynes | heh...i was kinda wondering that, too |
| 09:00.59 | dlynes | pakistan used to be called west bangladesh, right? |
| 09:00.59 | xbit` | i have a fax machine connected to asterisk with an ATA, and would like to send the file to an mISDN channel |
| 09:01.11 | yxa | if a 7940 using SIP can receive incoming calls but not make outgoing calls, what are some of the possible reasons? |
| 09:01.19 | saftsack | when will digium release the b410p card? |
| 09:02.10 | xbit` | i get the file with rxfax from the machine, i guess i have to send it to the final destination. |
| 09:03.09 | dlynes | xbit`: using txfax |
| 09:03.09 | stoffell | tzafrir, you happen to know why i get "syntax error near unexpected token" on line 435 of genzaptelconf ? (am I missing something?) |
| 09:03.13 | dlynes | xbit`: or you can email it |
| 09:03.33 | yxa | anyone? |
| 09:03.43 | wasim | dlynes: :) |
| 09:03.54 | dlynes | wasim: so I'm right? |
| 09:03.58 | wasim | dlynes: sorta |
| 09:04.07 | wasim | dlynes: it was west pk / east pk |
| 09:04.13 | dlynes | wasim: oh yeah...that's what it was |
| 09:04.25 | wasim | dlynes: but in our hearts we're all one :) |
| 09:05.10 | dlynes | wasim: Well, most peeps I know from Punjab dislike the Pakistanis very much, and they consider the Punjab area of Pakistan to still be part of India :) |
| 09:05.40 | wasim | ie. if you combine the indian/pk halves |
| 09:07.03 | dlynes | Yeah...one of my friends here, is from there |
| 09:08.50 | dlynes | He's from somewhere close to karachi |
| 09:09.01 | dlynes | Can't remember the name of the city he's from, exactly |
| 09:09.40 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 09:09.45 | wasim | the only city near karachi is hyderabad |
| 09:09.51 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:10.15 | dlynes | Maybe there's a small town or village near it? |
| 09:10.19 | wasim | but thats far away from punjab, southern corner is sindh |
| 09:10.24 | dlynes | I don't think it was hyderabad, either |
| 09:11.35 | stoffell | tzafrir, damn, found it. I have to use bash instead of sh :) |
| 09:11.45 | acehunky | wow cross border stuff happening here :) |
| 09:12.02 | dlynes | ? |
| 09:12.19 | acehunky | i m trying to find a person local in bd to help me setup asterisk box .. |
| 09:13.27 | dlynes | yeah, i know |
| 09:13.40 | dlynes | wasim's in pakistan though, not bangladesh |
| 09:14.23 | wasim | yep, and not likely to be heading east anytime soon neither ... |
| 09:14.51 | dlynes | and i'll be going to china long before I ever head to bangladesh :) |
| 09:15.20 | wasim | nee hawn dlynes |
| 09:15.49 | dlynes | ni hao ma? |
| 09:17.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:17.41 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 09:18.21 | tzafrir | stoffell, what version? the one from the subversion? |
| 09:18.27 | tzafrir | ah |
| 09:18.47 | stoffell | tzafrir, i was using wrong shell ;) tnx |
| 09:19.26 | tzafrir | Maybe I should add a check for a shell to give a decent error message? |
| 09:20.03 | stoffell | tzafrir, no, it's just a habit, i always do sh "script", i could have made it +x also, then it would have been okay :) |
| 09:20.25 | tzafrir | I have that habit myself |
| 09:21.01 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 09:21.41 | puzzled | morning |
| 09:22.04 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 09:22.22 | *** join/#asterisk welemon (n=welemon@202.106.75.99) |
| 09:24.00 | welemon | hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. |
| 09:24.07 | welemon | these fellows do not need to call in asterisk , just wait for a call. |
| 09:29.03 | tainted- | welemon i've been looking for something similar |
| 09:29.11 | tainted- | welemon let me know if u find such an app |
| 09:29.49 | acehunky | hello |
| 09:30.08 | acehunky | is there any card from digium which supports SS7 ? |
| 09:30.19 | welemon | right , i can not found this app , so if it really doesn't exist, i will implet it |
| 09:31.45 | dlynes | tainted-: can't you just do that with call files? |
| 09:33.29 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 09:34.19 | yxa | if a 7940 using SIP can receive incoming calls but not make outgoing calls, what are some of the possible reasons? |
| 09:35.03 | dlynes | yxa: codec mismatch could be one of the reasons, if codec autonegotiation is involved |
| 09:35.24 | dlynes | Does it work with a non-cisco phone configured for the same extension? |
| 09:37.32 | yxa | dlynes i only have a cisco phone now |
| 09:38.23 | tainted- | dlynes drop someone into the meetme() from within meetme()? |
| 09:39.34 | tainted- | dlynes say i'm in meetme()... i would love to press '96045551234' and have that person be dropped into meetme() with me |
| 09:39.44 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 09:39.46 | kippi | hey |
| 09:40.04 | kippi | I am looking for the call reporting sofware and I have forgoten the name of it |
| 09:40.26 | kippi | can anyone point me in the right dir |
| 09:40.56 | dlynes | tainted-: yeah...why not just make a slight modification to the app_meetme.c code to generate a call file when you do something like that (assuming meetme doesn't support that already) |
| 09:41.15 | dlynes | tainted-: then when that person's connected via the call file, transfer them to the meeting room extension |
| 09:41.54 | yxa | dlynes sip clients should have no problem with nat right? |
| 09:42.12 | tainted- | yxa wrong |
| 09:42.23 | dlynes | yxa: sip has tonnes of problems with nat |
| 09:42.31 | tainted- | depends on nat type and sip stack rfc implementation |
| 09:42.47 | tainted- | dlynes maybe i'll just make a web interface |
| 09:42.57 | yxa | no i mean when a client say 7940 is behind a firewall and the sip server is a public ip |
| 09:43.16 | tainted- | yea we know what u mean |
| 09:43.19 | yxa | not the other way |
| 09:43.34 | dlynes | yxa: it doesn't matter...sip doesn't cope well with nat, period |
| 09:43.48 | yxa | any articles to read up on that? |
| 09:43.55 | dlynes | yxa: it was never designed to work with a nat in the way; that's why iax and iax2 came about |
| 09:44.14 | yxa | like that ports to open and stuff |
| 09:44.23 | dlynes | yxa: there's hacks that have come up to deal with the problem, including the nat=yes setting for asterisk to stun servers |
| 09:44.48 | tainted- | neither works 100% |
| 09:44.58 | dlynes | tainted-: like i said...hacks :) |
| 09:45.14 | dlynes | I've got one solution that works for the majority of my customers |
| 09:45.25 | dlynes | But even then, I have two customers that solution doesn't work for |
| 09:45.27 | tainted- | i've heard that ser has no issues with sip behind nat b/c of it's strong sip stack |
| 09:45.37 | tainted- | what solution is that? |
| 09:45.38 | dlynes | One of those customers I have another solution for |
| 09:45.55 | dlynes | and the other customer I had to put the sip devices on external ips |
| 09:46.07 | dlynes | nat=yes, qualify=300 |
| 09:46.13 | tainted- | i've got a stun server and nat = yes as well as qualify = 5000 but a few customers still manage to timeout |
| 09:46.14 | DoktorGreg | <PROTECTED> |
| 09:46.31 | dlynes | and then on the sipura side, you can also set your nat timeout setting too |
| 09:46.46 | *** join/#asterisk alib80 (n=chatzill@196.31.11.194) |
| 09:46.51 | tainted- | hmm |
| 09:46.55 | tainted- | it's a grandstream ata |
| 09:46.55 | dlynes | Yeah...I've found qualify=2000 doesn't even work for some routers |
| 09:47.21 | dlynes | qualify=300 works for every router I've found where that trick actually works |
| 09:47.32 | alib80 | hi all does anyone know where I can look to find info on fax and incorrect number detection with asterisk |
| 09:47.32 | tainted- | why 300? |
| 09:47.35 | dlynes | another customer I had to do port mappings for |
| 09:47.58 | dlynes | and another customer, had a piece of crap router that was broken for port mappings, so i had to put the devices on external ips |
| 09:48.28 | dlynes | tainted-: it was just an arbitrary value that I found to work |
| 09:48.37 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:48.43 | dlynes | tainted-: netgear routers are notoriously bad for timeouts at low values |
| 09:50.35 | tainted- | that's silly |
| 09:51.50 | dlynes | canreinvite should be set to no, also |
| 09:52.06 | dlynes | alib80: incorrect number detection? |
| 09:52.18 | tainted- | yea i've got all that |
| 09:52.20 | alib80 | sorry |
| 09:52.23 | dlynes | tainted-: netgear routes are silly |
| 09:52.29 | dlynes | s/routes/routers |
| 09:52.31 | alib80 | not incorrect number, but rather discontinued number |
| 09:52.54 | dlynes | alib80: ummm...it can't |
| 09:53.07 | dlynes | alib80: neither can your fax machine |
| 09:53.09 | alib80 | dlynes: basically i'm trying to find out about tone detection |
| 09:53.19 | dlynes | alib80: but what you might be able to detect is human voice |
| 09:53.33 | yxa | dlynes are you saying either way the client or the server is not public ip, SIP gonna have problems? |
| 09:54.05 | alib80 | dlynes: are there any libraries for tone detection? |
| 09:54.17 | dlynes | yxa: either way client and server are not public ips, SIP is not guaranteed to be problem-free |
| 09:54.34 | dlynes | yxa: that's not to say that you can't get SIP to work with one or the other behind nat |
| 09:55.02 | dlynes | yxa: as far as both sides being behind nat though, good luck trying to get that to work...I've never heard of anyone having success with that |
| 09:55.18 | dlynes | alib80: define tone? |
| 09:55.37 | dlynes | alib80: fax modulation tone? or busy/congestion/... tone? |
| 09:55.54 | yxa | dlynes even with stateful firewalls? |
| 09:56.30 | alib80 | dlynes: all of the above |
| 09:56.48 | alib80 | dlynes: would one be able to feed in a tone and perform a type of pattern match |
| 09:56.59 | alib80 | against the recorded tone? |
| 09:57.46 | dlynes | alib80: fax modulation tone, yes...no idea on the rest...check the asterisk wiki for the link to the spandsp project |
| 09:58.09 | alib80 | dlynes: thanks:) |
| 09:58.29 | alib80 | dlynes: would i be able to find out about fax tone detection on there as well? |
| 09:59.08 | dlynes | yxa: no idea, but i would imagine you'll have problems with that if both sides are behind stateful or stateless firewalls, but perhaps not if only one side is behind a stateful firewall, and the other side isn't behind a firewall |
| 09:59.15 | *** join/#asterisk mmmmmToop (n=chatzill@firewall.datapro.co.za) |
| 09:59.24 | dlynes | alib80: that's what that project's for....fax modulation |
| 09:59.32 | alib80 | hnaks:) |
| 09:59.32 | dlynes | alib80: g3 and mfc, specifically |
| 09:59.41 | dlynes | alib80: i dont' think g2 is supported |
| 09:59.54 | alib80 | cool:) |
| 10:01.04 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 10:02.34 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 10:03.41 | joelsolanki | anybody know wheather digium ships product to india? |
| 10:06.57 | *** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk) |
| 10:07.35 | *** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net) |
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| 10:16.26 | tainted- | does dialplan continue execution when you're in a meetme()? |
| 10:27.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
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| 10:47.50 | joelsolanki | anybody knows wheather digium ships product to india ? |
| 10:55.34 | *** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net) |
| 10:55.38 | vgster | try emailing them |
| 10:55.52 | joelsolanki | ok |
| 11:01.43 | DoktorGreg | i wouldnt want to work at voip supply |
| 11:01.47 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 11:01.55 | DoktorGreg | if you look at their personel pictures on their web site |
| 11:02.05 | DoktorGreg | no one looks happy, even mildly |
| 11:06.09 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
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| 11:23.20 | swm_ | anyone know why I get an error using asterisk 1.0.9 saying the call leg does not exist? |
| 11:23.42 | swm_ | sip -> asterisk -> asterisk -> |
| 11:25.31 | [Airwolf] | swm_, don't you think it's a good idea to upgrade. ;) |
| 11:26.33 | swm_ | Oh it's been seriously considered by the company I work for but to get a couple techie's moving is the issue heh |
| 11:28.03 | swm_ | I figure it is a issue with the protocol ... We purchased some PolyPower VoIP boxes ... dont work with SIP from 1.0.9... but they work with CVS HEAD and current stable release. Question... Whats' the differnce between 1.0.9 and Current Stable./.Head ? |
| 11:30.54 | swm_ | A2 x 3 (F3*2^3.075) / 2.2^C = (FA-Z) |
| 11:31.13 | swm_ | ~lobotomy [Airwolf] |
| 11:31.15 | jbot | ACTION pulls out a rusty saw to perform a lobotomy on [Airwolf] |
| 11:32.58 | tzafrir | swm_, there are a number of changes between 1.0.9 and current 1.2 |
| 11:33.08 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
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| 11:34.29 | [Airwolf] | swm_, sorry I can't answer that question for you |
| 11:34.35 | *** part/#asterisk Samoied (n=Samoied@201-3-227-215.fnsce7002.dsl.brasiltelecom.net.br) |
| 11:34.51 | IceManRISK | anyone here uses jiax ? |
| 11:34.59 | IceManRISK | im having problem with |
| 11:38.06 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 11:38.30 | doughecka_ | oops |
| 11:38.36 | doughecka_ | my bad |
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| 11:45.27 | Aurs | swm_: http://ftp.digium.com/pub/telephony/asterisk/old-releases/ChangeLog-1.2.7.1 |
| 11:45.47 | Aurs | that is the difference between 1.0.9 and 1.2.7.1 |
| 11:51.40 | *** part/#asterisk welemon (n=welemon@202.106.75.99) |
| 12:03.58 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 12:05.30 | *** join/#asterisk akrall (i=user@dsl-201-128-92-208.prod-infinitum.com.mx) |
| 12:05.40 | akrall | hey guys |
| 12:06.05 | tamp4x | what would be reasons the person on the other side of the calls get intermitent beeps if they have no incoming calls while on the call? |
| 12:06.30 | akrall | whats hardware would you guys recommend for a 25 sip ext. pbx wit 1 E1 and doing call recording, monitoring, etc.? |
| 12:06.53 | *** join/#asterisk markus99 (n=markus@165.154.121.219) |
| 12:07.45 | mmlj4 | akrall: there's a page on the wiki about asterisk dimensioning |
| 12:08.10 | akrall | mmlj4: I checked it out but wanted to know what you guys have used? |
| 12:08.34 | akrall | mostly the wiki says you need 1 proc, for every 4 T1 for example |
| 12:08.48 | akrall | but wanted to know what guys here have used |
| 12:10.47 | mmlj4 | hmm... 1 E1, you say? I'd use half a proc, just to be sure... |
| 12:13.11 | akrall | for the system Ive explain, would a xeon 3.0 be overkill? |
| 12:13.19 | akrall | and what do you rec. sata or ide? |
| 12:13.51 | akrall | Ive had issues with interrupts before and I noticed that if you test using hdparam on a sata server, you do get noise on the line while testing |
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| 12:25.21 | jsharp | glorp |
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| 12:42.08 | pif | hi, what kind of cable should I use to connect a telco E1 socket with a TE410P card? |
| 12:42.27 | *** part/#asterisk akrall (i=user@dsl-201-128-92-208.prod-infinitum.com.mx) |
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| 12:43.39 | tamp4x | what would be reasons the person on the other side of the calls get intermitent beeps if they have no incoming calls while on the call? |
| 12:45.33 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.224) |
| 12:45.46 | DarKnesS_WolF | Asterisk RealTime not supported by Mysql5 ? |
| 12:45.47 | Ariel_ | pif, depends on what your plugging it into. But most use a t1 crossover cable |
| 12:47.19 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:47.28 | *** join/#asterisk Hali_303 (n=surfk@dsl5402AC97.pool.t-online.hu) |
| 12:51.15 | DarKnesS_WolF | Asterisk RealTime not supported by Mysql5 ? |
| 12:51.22 | DarKnesS_WolF | anyone using realtime with mysql5 ? |
| 12:51.27 | Aurs | DarKnesS_WolF: odbc or mysql driver? |
| 12:51.33 | RoyK | DarKnesS_WolF: shouldn't be a problem |
| 12:51.35 | DarKnesS_WolF | mysql native driver |
| 12:51.46 | RoyK | DarKnesS_WolF: what's the problem? |
| 12:52.47 | DarKnesS_WolF | RoyK: Apr 24 14:47:46 WARNING[26603] loader.c: /usr/lib/libmysqlclient.so.15: version `MYSQL_5.0' not found (required by /usr/lib/asterisk/modules/res_config_mysql.so) |
| 12:53.01 | DarKnesS_WolF | Apr 24 14:47:46 WARNING[26603] loader.c: Loading module res_config_mysql.so failedbut the lib is there |
| 12:53.12 | DarKnesS_WolF | but hte lib is there |
| 12:53.34 | RoyK | dunno |
| 12:53.41 | IceManRISK | thsi lib just work for 4.0 |
| 12:54.24 | DarKnesS_WolF | IceManRISK: so what should i do ? |
| 12:54.34 | IceManRISK | use mysql 4.* |
| 12:54.55 | IceManRISK | or try to find the lib that supports mysql 5 |
| 12:55.36 | DarKnesS_WolF | IceManRISK: there is already this driver? |
| 12:57.15 | RoyK | DarKnesS_WolF: where is mysql5 installed? |
| 12:57.50 | DarKnesS_WolF | RoyK: i'm using hte deb packages so i think the default |
| 12:58.07 | RoyK | deb package?? SID? |
| 12:58.51 | RoyK | anyway |
| 12:58.58 | darkskiez | mysql is on sarge backports |
| 12:59.02 | pif | Ariel_ : thanks, I'm plugging into the telco's E1 router, |
| 12:59.05 | darkskiez | mysql5 |
| 12:59.08 | RoyK | mysqlclient 15 should be for 5.0 afaik |
| 12:59.26 | RoyK | 14 is for 4.1 and 12 is for 4.0 |
| 12:59.28 | RoyK | iirc |
| 13:00.05 | pif | Ariel_ : that might explain why I had no red leds at both end when using a straight cat5 cable? |
| 13:00.26 | pif | I mean "no green leds" |
| 13:00.33 | *** join/#asterisk shiznatix (n=shiznati@213-35-241-48-dsl.end.estpak.ee) |
| 13:01.06 | pif | RoyK : hi, to connect your sangomas to the carrier, what type of cable did you use? |
| 13:01.21 | Ariel_ | pif, hope it works. Most here in the use from the main telco's are now using smartjacks. which can use a normal Cat5 cable. But if your going to a pbx or to an router then a crossover cable is needed. |
| 13:02.25 | RoyK | pif: you mean for the PRIs? |
| 13:02.30 | pif | ack |
| 13:02.32 | RoyK | just a standard cat 5e |
| 13:03.01 | pif | and between legacy pbx and PRI card? |
| 13:03.29 | RoyK | E1 crossover |
| 13:03.51 | mut | ah shit |
| 13:03.59 | mut | my sangoma card was here friday and no one told me!!! |
| 13:04.06 | RoyK | http://www.voip-info.org/wiki/view/crossover+T1+cable |
| 13:05.57 | pif | thanks |
| 13:09.46 | DarKnesS_WolF | RoyK: sorry i'm using testing. i'll down grade to mysql 4.1 |
| 13:09.50 | *** join/#asterisk hgaillac (n=Harry@83.15.119-80.rev.gaoland.net) |
| 13:11.00 | hgaillac | Hello I need somebody for testing fax over ip (T38) |
| 13:11.22 | RoyK | hgaillac: where did you find t.38 support for asterisk? |
| 13:15.35 | hgaillac | Royk: Here http://bugs.digium.com/view.php?id=5090 |
| 13:15.53 | *** part/#asterisk Hali_303 (n=surfk@dsl5402AC97.pool.t-online.hu) |
| 13:15.57 | *** join/#asterisk zeppelin_ (n=zeppelin@201.15.175.238) |
| 13:18.24 | SpaceBass | Ariel_, what is makes something a smartjack? |
| 13:18.51 | *** join/#asterisk coppice (n=chatzill@95.162.17.210.dyn.pacific.net.hk) |
| 13:19.09 | Frogzoo | mut: what's you sangoma doing hanging out on irc? |
| 13:19.34 | Ariel_ | SpaceBass, ??? |
| 13:19.35 | mut | being a bastard |
| 13:19.45 | mut | i smacked it into submission tho |
| 13:20.25 | SpaceBass | Ariel_, saw that you mentioned smartjacks earlier....just curious what makes a smartjack different from a standard ethernet port? I know they are the prefered way to terminate a T1, etc...but I was never sure on the difference |
| 13:20.43 | RoyK | hgaillac: that's not really finished :) |
| 13:20.49 | RoyK | hgaillac: ask coppice |
| 13:21.05 | Bert- | hmm guys |
| 13:21.12 | Ariel_ | SpaceBass, smartjacks are able to detect what cable wires your using. |
| 13:21.18 | SpaceBass | gotcha |
| 13:21.27 | Bert- | does anyone knows good sites to find termination providers plz ? |
| 13:21.27 | coppice | RoyK: what is not really finished? |
| 13:21.45 | RoyK | t.38 for asterisk |
| 13:21.46 | Bert- | (maybe someone here terminate some destinations ...) |
| 13:22.01 | coppice | RoyK: works for me :-) |
| 13:22.10 | RoyK | coppice: it does? gatewaying? |
| 13:22.18 | [TK]D-Fender | Bert- : Depends where you're terminating to. Check the WIKI for a good list of places to start |
| 13:22.18 | [TK]D-Fender | ~docs |
| 13:22.19 | jbot | it has been said that 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 |
| 13:22.27 | RoyK | as in fax -> ATA -> asterisk chan_sip -> pstn? |
| 13:22.31 | coppice | RoyK: yes, but not yet within * |
| 13:22.35 | RoyK | ah |
| 13:22.35 | RoyK | ok |
| 13:23.08 | hgaillac | I patched chan_sip |
| 13:24.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:27.15 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
| 13:27.30 | Bert- | [TK]D-Fender : could you update me with the wiki url plz ? |
| 13:27.38 | Bert- | maybe I lost my eyes |
| 13:27.45 | Bert- | but I'm unable to find it on the website ... |
| 13:28.37 | [TK]D-Fender | http://www.voip-info.org/ |
| 13:29.01 | Bert- | thx |
| 13:29.21 | DarKnesS_WolF | hum i can't download i forgot i need mysql 5 on this box |
| 13:29.34 | [TK]D-Fender | Can someone who's capable of competantly updating jbot please add THEBOOK to the "~docs" script..... |
| 13:30.12 | [TK]D-Fender | And remove that archaic "Handbook Draft" link.... |
| 13:30.40 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 13:30.50 | hgaillac | bye |
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| 13:34.19 | miller7 | hello, can someone assist me with realtime extensions? I am trying the Wiki with no success. if someone has it working please reply in private |
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| 13:50.54 | tamp4x | what would be reasons the person on the other side of the calls get intermitent beeps if they have no incoming calls while on the call? |
| 13:51.33 | jake1932 | have they made any calls to known terrorist camps? |
| 13:52.41 | *** join/#asterisk stoffell (n=stoffell@dD57664D6.access.telenet.be) |
| 13:54.44 | X-Rob | they called someone who called someone who once lived on the same block as someone who is from another country? |
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| 13:56.44 | jsharp | Urgh. 2 bottles of Pepsi and I'm still dragging ass this morning. |
| 13:57.58 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:58.46 | jake1932 | jsharp: AMP |
| 13:59.58 | jsharp | No AMP support here. |
| 14:00.29 | X-Rob | Anyone asking for AMP support needs to be told to upgrade. |
| 14:00.37 | X-Rob | before you even send 'em over to us. |
| 14:01.38 | *** join/#asterisk zeppelin_ (n=zeppelin@201-34-96-165.paemt700.dsl.brasiltelecom.net.br) |
| 14:01.58 | coppice | TNVWBB - the next version will be better :-) |
| 14:02.09 | X-Rob | ooh, there you are steve |
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| 14:07.45 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:07.48 | wwalker | Can I created a "bridged" call from the asterisk console. That is, have it call me and call my voicemail and connect the calls? |
| 14:08.24 | b00mer_ | wwalker : Good question... I would like to know the answer to that as well |
| 14:08.35 | wwalker | I've done it via an AGI before, but can't figure out how to do it from the asterisk -r prompt |
| 14:08.36 | X-Rob | RTFM '.call' files |
| 14:08.50 | X-Rob | you can't do it from asterisk -r |
| 14:08.54 | X-Rob | you have to make a .call file |
| 14:09.00 | Nivex | or use the manager interface |
| 14:09.06 | wwalker | k |
| 14:09.27 | wwalker | thx, saved me an hour or two of looking for a non-existant answer |
| 14:11.09 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 14:11.51 | praet | can you connect an asterisk box to another? |
| 14:12.01 | jsharp | In many different ways. |
| 14:12.17 | ManxPower | oh god the newbies! the newbies! |
| 14:12.22 | *** part/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 14:12.40 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 14:12.45 | b00mer_ | I need a professional to do the voice prompts... any recommendations? |
| 14:12.56 | [TK]D-Fender | b00mer_ : www.thevoice.com? |
| 14:13.50 | [TK]D-Fender | http://www.digium.com/en/products/voice/ |
| 14:13.55 | b00mer_ | [TK]D-Fender : right link? they don't mention it |
| 14:14.03 | b00mer_ | ok... I try the other |
| 14:14.46 | [TK]D-Fender | Other one is a better link.. I am mistaken on the first. Allison's site is SOMEHWERE else... |
| 14:15.31 | b00mer_ | close... http://www.theivrvoice.com/ |
| 14:15.37 | b00mer_ | thanks for the recommendation |
| 14:17.52 | [TK]D-Fender | b00mer_ : Yeah, thats the one :) |
| 14:18.19 | [TK]D-Fender | b00mer_ : Just a good idea if you want your prompts to sound like they all came from the same place. |
| 14:19.25 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:19.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:21.18 | [TK]D-Fender | Hey anthm, Care to lend me a hand for something that'll take about a minute? |
| 14:21.30 | anthm | sure |
| 14:22.16 | [TK]D-Fender | anth : Could you update jbot adding "THEBOOK" to the "~docs" script and remove the reference to that archaic "handbook draft"? |
| 14:23.17 | [TK]D-Fender | I'm just incompetant when it comes to telling it what to do :) |
| 14:23.47 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 14:24.05 | anthm | you know, i actually have no clue how to work it |
| 14:24.53 | coppice | are draft handbooks anything like draft beer? :-\ |
| 14:25.10 | *** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu) |
| 14:25.52 | [TK]D-Fender | coppice : Either will leave you tipsy and confused, so my magic 8-ball says "Absolutely!" |
| 14:26.58 | Sonderblade | im installing asterisk 1.2.6 and gcc spits out like hundreds of warnings |
| 14:27.00 | Sonderblade | is that usual? |
| 14:27.35 | [TK]D-Fender | Sonderblade : Installing * how? And maybe you should try using the most current release.... |
| 14:28.43 | *** join/#asterisk azzie (n=az@azzie.net) |
| 14:28.54 | Sonderblade | [TK]D-Fender: in compiling it from source |
| 14:29.24 | b00mer_ | Sonderblade : I got the same |
| 14:29.27 | [TK]D-Fender | Sonderblade : What kind of warnings? Pastebin them. |
| 14:29.28 | [TK]D-Fender | ~pb |
| 14:29.30 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:29.45 | b00mer_ | Sonderblade : I got warnings with 1.2.7.1 also |
| 14:29.56 | Sonderblade | [TK]D-Fender: they were many, things like uninitialized variables etc.. scary warnings |
| 14:30.28 | b00mer_ | my rule... if it compiles ... its got to be good :) |
| 14:30.32 | [TK]D-Fender | Sonderblade : Grab 1.2.7.1 and start from scratch, also make sure to recompile the lastest Zaptel and any other related modules FIRST. |
| 14:31.59 | Hmmhesays | we need an 8ball in here |
| 14:32.05 | Hmmhesays | ~8ball |
| 14:32.06 | jbot | ACTION rolls the eight ball and gets: Without a doubt |
| 14:32.21 | Hmmhesays | ~8ball dead hookers? |
| 14:32.22 | jbot | Absolutely. |
| 14:32.31 | Hmmhesays | fantastic |
| 14:32.35 | jsharp | Dead hookers aren't much fun. |
| 14:32.57 | jsharp | They won't roll over. They won't play ball. |
| 14:34.36 | Pj_ | Yeah, but after a few hours they're very tight |
| 14:36.14 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 14:36.19 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) |
| 14:37.16 | *** join/#asterisk BugKham (n=BugKham@125.24.9.121) |
| 14:38.27 | Hmmhesays | LOL |
| 14:38.40 | [TK]D-Fender | jsharp : Necrophylia : The issestable urge to crack opone a cold one ;) |
| 14:38.55 | [TK]D-Fender | irresitable* |
| 14:39.01 | [TK]D-Fender | damn I can't type today... |
| 14:39.02 | jsharp | Aieeeeeeeee. |
| 14:39.56 | Hmmhesays | [TK]D-Fender: wow, i am stealing that from you |
| 14:40.22 | [TK]D-Fender | Hmmhesays : It's CDL, so don't forget the reference :) |
| 14:40.24 | tzanger | [TK]D-Fender: ooh damn that's bad |
| 14:40.48 | [TK]D-Fender | And it allows me to spot the alcoholics instantly ;) |
| 14:41.41 | BugKham | hi, how can we know which channel value is returned from Dial(Zap/g1,10)? |
| 14:42.42 | *** join/#asterisk noname32 (n=noname@38.113.5.165) |
| 14:43.52 | [TK]D-Fender | BugKham : The ",10" is a bad idea... |
| 14:44.55 | BugKham | [TK]D-Fender: okay, will change that |
| 14:44.56 | Katty | Ariel_, [TK]D-Fender, you guys ever watch dark side with wizard of oz muted? |
| 14:45.13 | [TK]D-Fender | Katty : mew. |
| 14:45.21 | [TK]D-Fender | Katty : dark side? |
| 14:45.23 | Katty | also! mew. |
| 14:45.27 | Katty | [TK]D-Fender: of the moon (= |
| 14:45.34 | BugKham | [TK]D-Fender: by the way, do you know which available channel will be selected? |
| 14:46.01 | jake1932 | i've heard it's even better while smokin |
| 14:46.03 | Ariel_ | Katty, no but I can imagine it. |
| 14:46.29 | Katty | Ariel_: i ran across an article about how it was mostly in sync...and i just watched a bit of it. the munchkin dance was /great/ and completely in sync. |
| 14:46.30 | [TK]D-Fender | BugKham : There is no such thing as "WILL be selected" by the time the next dialplan line is executed the call is already TERMINATED. |
| 14:46.37 | *** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx) |
| 14:46.43 | [TK]D-Fender | Katty : You've insaned.... |
| 14:46.44 | Ariel_ | BugKham, from lowest to higer number if you use G1 from higher to lower. |
| 14:46.53 | noname32 | anyone here use automon at all? i got some questions |
| 14:46.56 | [TK]D-Fender | Ariel_ : And you're clearly not far behind! |
| 14:46.56 | Pj_ | Nah she was already insane |
| 14:47.09 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
| 14:47.14 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:47.15 | Pj_ | she was insane years ago when I used to come, then I came back - still insane -, now I don't even notice |
| 14:47.47 | BugKham | Ariel_: which variable contains the returned ZAP channel? |
| 14:47.59 | Hmmhesays | anyone have any experience using vendor specific attributes in dhcpe? |
| 14:48.04 | Katty | [TK]D-Fender: google it (= |
| 14:48.09 | Katty | [TK]D-Fender: of course, it's really a myth.. |
| 14:48.16 | Katty | [TK]D-Fender: but the video and the album go very well together. |
| 14:48.40 | [TK]D-Fender | Katty : I'll just take your word for it :) |
| 14:48.47 | Katty | jake1932: also, i don't smoke (= |
| 14:48.51 | b00mer_ | anybody have a good example of a findme... the one on voip-info seems broken |
| 14:49.08 | jake1932 | Katty: nor do i - but for those interested |
| 14:49.15 | Katty | [TK]D-Fender: you don't have to! http://www.youtube.com/watch?v=71aJlRZARjQ |
| 14:49.15 | [TK]D-Fender | b00mer_ : TOYWY :) |
| 14:49.30 | Katty | [TK]D-Fender: go watch it (= |
| 14:49.42 | jake1932 | katty - great find |
| 14:49.44 | b00mer_ | TOYWY? not familar with that acronmy |
| 14:49.54 | Katty | jake1932: yesh. |
| 14:50.01 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:50.40 | *** join/#asterisk grem_lin (n=gremlin@2001:618:400:0:0:0:50e5:e3e7) |
| 14:50.48 | b00mer_ | [TK]D-Fender : don't follow you... googling for TOYWY doesn't help |
| 14:51.23 | [TK]D-Fender | "The One You Write Yourself" |
| 14:51.23 | jsharp | Darnit. no sound coming from youtube. |
| 14:51.34 | [TK]D-Fender | jsharp : There is.. jsut wait, its silent at first |
| 14:52.10 | jsharp | Nah. Skipped through most of the file and no sound anywhere on it. |
| 14:52.26 | [TK]D-Fender | b00mer_ : for follow me, you just need to decide what reourse to dial first, then "GotoIf" based on "DIALSTATUS" |
| 14:52.29 | b00mer_ | [TK]D-Fender : ok... I will... from expert advice... should I do it in a AGI script or something else? |
| 14:52.42 | [TK]D-Fender | jsharp : Took about 2 mins before I got sound |
| 14:52.44 | BugKham | [TK]D-Fender: I need to keep the "first available channel" in g1 in a variable for further use, is it possible? |
| 14:52.53 | [TK]D-Fender | b00mer_ : no need for AGI for most cases... |
| 14:53.04 | jsharp | I guess I'll have to play it on my Winderz box later. |
| 14:53.08 | froguz | what happened to "make config" in asterisk 1.2.7.1? i'm getting : "install: cannot `stat' over «init.asterisk»: file or directory doesn't exist" error |
| 14:53.09 | b00mer_ | [TK]D-Fender looking for something that will call people... give them a banner asking to press 1 if they want to take the caller |
| 14:53.13 | [TK]D-Fender | BugKham : not sure really... |
| 14:53.35 | [TK]D-Fender | BugKham : look up "cmd dial" on the WIKI and see if it returns anything... |
| 14:53.44 | b00mer_ | [TK]D-Fender : I'll check out dialstatus |
| 14:53.51 | Ariel_ | BugKham, you can look up on google or the wiki for an agi called dialparties.agi which is made for giving you dialstatus. |
| 14:53.57 | coppice | bloody DC. their ain't supposed to be no DC in an alaw/ulaw channel. how come that is so often not true? bloody crap equipment |
| 14:54.00 | [TK]D-Fender | b00mer_ : in "Dial" there is a cmd that will call a macro on connect.... look it up... |
| 14:55.21 | jsharp | Ow, my GUI bits! |
| 14:55.26 | [TK]D-Fender | mmmM! |
| 14:55.44 | Katty | [TK]D-Fender: i still need gui sometimes. |
| 14:55.58 | [TK]D-Fender | Katty :For configuring *? |
| 14:56.00 | Katty | [TK]D-Fender: but i don't startx most of the time. |
| 14:56.05 | Katty | [TK]D-Fender: no..i use emacs for that |
| 14:56.16 | [TK]D-Fender | Katty : Well then, I guess you get to live! |
| 14:56.28 | jsharp | Does it count if its a GUI I wrote myself? |
| 14:56.45 | [TK]D-Fender | jsharp : ... MAYBE... I'd have to pass inpection :) |
| 14:56.54 | [TK]D-Fender | it'd |
| 14:56.55 | [TK]D-Fender | ugh |
| 14:57.00 | BugKham | Ariel_: don't think DIALSTATUS will provide that information |
| 14:57.07 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 14:57.33 | [TK]D-Fender | BugKham : it won't.... go read up on Dial and see if it returns anything... |
| 14:57.54 | nettie | Hi guys, I'm having problem configuring asterisk with www.skypho.net voip carrier. Anyone has a working configuration with them please? |
| 14:58.25 | froguz | i think somebody just forgot about "make config" in the last * release.... |
| 14:58.45 | [TK]D-Fender | froguz : "make config"? |
| 14:59.10 | froguz | yeap... i allways use that option |
| 14:59.36 | froguz | on zaptel and asterisk installations |
| 14:59.42 | Ariel_ | froguz, I was under the impression that was for RH type of releases.. But your correct |
| 15:00.50 | froguz | i've allways worked with "make config" it avoid me to script init by hand |
| 15:01.45 | froguz | i've used "make config" under RH based distros (Centos) and debian (ubuntu) |
| 15:06.14 | froguz | what if i just copy the init.asterisk file from an older asterisk source (1.2.6)???? |
| 15:06.33 | froguz | into the 1.2.7.1 source and then "make config"?? |
| 15:06.35 | De_Mon | can I force a sip peer to reregister? |
| 15:07.31 | froguz | De_Mon, *CLI>sip reload? |
| 15:07.36 | De_Mon | sip show registry is always empty, even when i know there are active registrations |
| 15:07.58 | froguz | De_Mon, *CLI>sip show peers |
| 15:08.01 | De_Mon | froguz nah, that just reloaded the configs |
| 15:08.15 | De_Mon | froguz what purpose does sip show registry have then? |
| 15:09.10 | froguz | it show you the registered ITSP servises |
| 15:10.32 | De_Mon | ooh /me looks closer |
| 15:10.36 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 15:10.39 | a1fa | man |
| 15:10.41 | a1fa | this is jacked up |
| 15:10.47 | a1fa | every hour when my asterisk is supposed to re-register |
| 15:10.57 | a1fa | with a sip broker, it craps out |
| 15:11.02 | a1fa | and i cant recieve any phone calls |
| 15:11.39 | docelm0 | YAHOO!? |
| 15:11.40 | De_Mon | whats the difference between a sip.conf 'register' and a phone 'registration' |
| 15:11.41 | *** join/#asterisk lucifr (n=chatzill@ppp-71-134-54-46.dsl.irvnca.pacbell.net) |
| 15:12.08 | a1fa | i do a sip reload |
| 15:12.13 | a1fa | and it starts working again |
| 15:12.18 | a1fa | its that stupid broadvoice |
| 15:14.14 | noname32 | anyone have any ideas why automon will not work with * and a number but will work with ** or ## |
| 15:16.17 | a1fa | blah |
| 15:16.28 | mut | SELECT distinct billingnumber FROM cdr left join m33accounts.activeservices as actsvc on actsvc.voipphonenumber = cdr.billingnumber where actsvc.masteraccount is null order by cdr.billingnumber |
| 15:16.33 | mut | is there a faster way to do this? |
| 15:16.49 | mut | er |
| 15:16.50 | mut | sorry rather |
| 15:17.08 | mut | SELECT distinct billingnumber FROM cdr left join m33accounts.activeservices as actsvc on actsvc.voipphonenumber = cdr.billingnumber OR actsvc.telconumber = cdr.billingnumber where actsvc.masteraccount is null and cdr.category <> 20 order by cdr.billingnumber |
| 15:17.14 | a1fa | ok |
| 15:17.18 | a1fa | i cant recieve calls againwtf |
| 15:17.26 | mut | doing the 'OR actsvc.telconumber = cdr.billingnumber' |
| 15:17.30 | *** part/#asterisk ph|ber (i=phiber@slackwaresupport.com) |
| 15:17.33 | mut | makes it take 1000x longer |
| 15:17.48 | sivana | mut, your doing two passes of the table |
| 15:18.14 | mut | is there a way to optimise it any? |
| 15:18.21 | mut | it takes longer than 2x the time |
| 15:18.22 | sivana | do you have an index on billingumber? |
| 15:18.25 | mut | yes |
| 15:18.33 | mut | not on voip or telco number tho |
| 15:19.23 | sivana | 1 sec |
| 15:19.30 | *** join/#asterisk perlmonky (n=perlmonk@69-168-21-26.chvlva.adelphia.net) |
| 15:20.35 | De_Mon | I've never seena JOIN ... ON ... OR ... before |
| 15:21.29 | a1fa | fucking * wont pickup the call |
| 15:21.33 | a1fa | i am sick and tired of this bullshit |
| 15:21.36 | mut | well the point of the query is to find records in the cdr that don't match the records we have (voip number or telco number) |
| 15:21.56 | sivana | mut: try putting () around (actsvc.voipphonenumber = cdr.billingnumber OR actsvc.telconumber = cdr.billingnumber) |
| 15:22.10 | mut | k |
| 15:22.19 | De_Mon | what happens if its in both join expressions? |
| 15:22.47 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 15:22.53 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 15:22.56 | sivana | mut, you can try two separate queries with a UNION |
| 15:23.04 | mut | thats an idea |
| 15:23.14 | brad_mssw | a1fa: sounds like you should dump BV |
| 15:23.15 | a1fa | ok this is bullshit |
| 15:23.21 | a1fa | brad_mssw : i should |
| 15:23.21 | salviadud | freepbx is a nightmare |
| 15:23.24 | salviadud | i hate my boss |
| 15:23.35 | a1fa | omfg |
| 15:23.39 | a1fa | it wont pickup the call |
| 15:23.43 | a1fa | it registers inbound call |
| 15:23.46 | a1fa | but it wont pick it up |
| 15:23.49 | a1fa | i have to reload |
| 15:23.51 | a1fa | and then it picks it up |
| 15:24.12 | brad_mssw | what v of asterisk ? |
| 15:24.28 | a1fa | Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) |
| 15:24.32 | a1fa | 1.2.6 |
| 15:24.40 | a1fa | err. |
| 15:24.42 | a1fa | 1.2. |
| 15:24.44 | a1fa | 1.2.7 |
| 15:24.53 | a1fa | this is what pissess me off the most |
| 15:24.55 | a1fa | Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) |
| 15:25.08 | a1fa | defaultexpiry is set to 3600 |
| 15:25.16 | a1fa | but broadvoice overwrites it with 23 s |
| 15:25.19 | a1fa | or 30s |
| 15:26.15 | a1fa | mother fuckers |
| 15:26.17 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:26.46 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 15:26.48 | brad_mssw | thats pretty frequent, dunno, dumped BV long ago myself because of issues( mainly call quality and availability of service)... |
| 15:26.48 | blebleble | maybe a dumb question, is there a way to multihome did's for redundancy, like a failover point (sort of like bgp) like if our carrier went down it could be kicked over to a failsafe? |
| 15:26.57 | froguz | hey! i'd fix the "make config"... there was an mistake in the Makefile. how could i report this?? |
| 15:27.04 | froguz | how can i * |
| 15:27.10 | a1fa | brad_mssw : what you got now? |
| 15:27.32 | brad_mssw | blebleble: not unless your provider provides that functionality via redudant world-wide servers |
| 15:27.53 | blebleble | brad_mssw: so its my providers issue really and nothing i myself can implement |
| 15:28.06 | brad_mssw | a1fa: primarily junctionnetworks right now, they've been the most reliable, though somewhat pricy (we use it for business though, so reliability wins) |
| 15:28.12 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 15:28.19 | brad_mssw | blebleble: yeah, pretty much |
| 15:28.19 | a1fa | brad_mssw : i like their world plan |
| 15:28.28 | blebleble | brad_mssw: thanks for your help |
| 15:28.29 | a1fa | brad_mssw : i need free minutes across europe |
| 15:28.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:29.14 | blebleble | can anyone recommend a good/cheap/stable carrier? our current one seems to be down too much |
| 15:29.15 | a1fa | bastards |
| 15:29.16 | brad_mssw | a1fa: well, it really depends on how many minutes you use, a lot of pay-as-you-go plans are a better deal |
| 15:29.23 | a1fa | blebleble : broadvoice? |
| 15:29.38 | a1fa | brad_mssw : i use shit load of minutess across .EU |
| 15:29.42 | brad_mssw | a1fa: unless you're really using a ton of minutes (most 'unlimited' plans have caps anyway though) |
| 15:30.08 | brad_mssw | blebleble: good/cheap/stable .. don't think there is such a one |
| 15:30.24 | *** join/#asterisk imcdona (n=fff@c-24-19-80-240.hsd1.wa.comcast.net) |
| 15:30.26 | brad_mssw | blebleble: junctionnetworks isn't bad though, just not so much on the cheap side |
| 15:30.26 | blebleble | yah i know, just looking for low price did's and solid termination |
| 15:30.30 | imcdona | morning everyone |
| 15:30.47 | jake1932 | brad_mssw: u using junction now? |
| 15:30.49 | brad_mssw | blebleble: they're $2/DID per month, plus like $.02/min |
| 15:30.56 | imcdona | Anyone know of a good app to call 14k people and deliver a message? |
| 15:31.01 | brad_mssw | jake1932: yep, finally switched our 800 number off of teliax ... |
| 15:31.12 | a1fa | imcdona : phone spammer |
| 15:31.12 | brad_mssw | jake1932: teliax is total crap these days |
| 15:31.13 | jake1932 | brad_mssw: any issue with outbound callerid? |
| 15:31.14 | imcdona | AstAutodialer doesn't seem to wait for VM |
| 15:31.39 | brad_mssw | jake1932: not with outbound ...they don't support inbound callerid name though :/ but oh well |
| 15:31.53 | jake1932 | brad_mssw: nobody seems to |
| 15:31.55 | brif8 | manager.conf allows you access the current status of asterisk (right)? if so then how would you check on the status of invidivual extensions both incoming numbers and IP phones or SIP devices ? |
| 15:32.10 | jsharp | Cause it costs money to dip into the CLID name databases. |
| 15:32.18 | jake1932 | brad_mssw: within the past day i've had outbound callerid issues - gonna check with support |
| 15:32.18 | alib80 | exit |
| 15:32.28 | brad_mssw | jsharp: yeah, I'd be willing to pay extra $$ for that, personally |
| 15:32.36 | brad_mssw | jake1932: w/junction ? |
| 15:32.39 | jake1932 | yes |
| 15:32.47 | brad_mssw | hmm, odd |
| 15:32.55 | brad_mssw | may be an account setting |
| 15:33.05 | imcdona | no...not phone spammer....out customers who have a past due balance |
| 15:33.13 | jake1932 | could be |
| 15:33.23 | jsharp | That's a lot of people who owe you money. |
| 15:34.00 | *** mode/#asterisk [-o twisted[asteria]] by twisted[asteria] |
| 15:34.12 | a1fa | god damn broadvoice |
| 15:34.14 | a1fa | i hope they choke |
| 15:34.24 | a1fa | their support sucks anus |
| 15:34.37 | Hmmhesays | pig or goat |
| 15:34.41 | a1fa | both |
| 15:34.45 | a1fa | they wont pickup the call |
| 15:34.48 | Hmmhesays | raw or cooked |
| 15:34.49 | a1fa | they are not busy at all |
| 15:34.53 | a1fa | raw |
| 15:34.55 | twisted[asteria] | hahaha |
| 15:35.00 | twisted[asteria] | i love how you can assume they're not busy |
| 15:35.00 | a1fa | they pretend to be busy |
| 15:35.02 | Hmmhesays | nasty |
| 15:35.04 | twisted[asteria] | just because you don't get a busy |
| 15:35.16 | twisted[asteria] | that's just like when someone tells me i'm not doing anything when they're halfway round the world |
| 15:35.27 | a1fa | twisted[asteria] : i was on hold for 45minutes once, and I asked dude, are you guys busy today |
| 15:35.27 | twisted[asteria] | grow some patience |
| 15:35.28 | a1fa | he says |
| 15:35.49 | a1fa | we are not busy at all |
| 15:35.51 | a1fa | ok |
| 15:35.54 | a1fa | i got some1 on the call |
| 15:36.06 | a1fa | i am going to tell him to suck a pigs as |
| 15:36.08 | a1fa | aZZ! |
| 15:36.14 | twisted[asteria] | gawd... |
| 15:36.39 | a1fa | skwid! |
| 15:38.14 | sivana | haha |
| 15:39.08 | sivana | little extra latency just for you |
| 15:39.17 | *** join/#asterisk Timmerman (n=Lu@200.175.156.165.static.gvt.net.br) |
| 15:41.10 | SuperLag | I see on one of the mailing lists that you guys recommend not running X or GUI apps on an * box. Is that still the case, and what if you're running a beefy box with a lot of RAM? (in this case, the box will have a 3.73GHz P4 w/2MB cache, and 4GB of RAM. |
| 15:41.14 | Timmerman | hi folks, have a way that build a PBX with Asterisk using Skype to dial external calls? |
| 15:41.17 | SuperLag | ) |
| 15:42.04 | Hmmhesays | now there is |
| 15:42.34 | salviadud | mmmm, broadvoice employs high school students, or slackers |
| 15:42.56 | Nugget | I am a slacker. |
| 15:43.06 | sivana | me too |
| 15:43.18 | salviadud | we should all work at broadvoice |
| 15:43.26 | salviadud | and give bad tech support |
| 15:43.46 | salviadud | SuperLag, depends on how many chans you're handling |
| 15:44.07 | [TK]D-Fender | SuperLag : Depends on the load. I run EVERYTHING on mine... |
| 15:44.12 | file | Nugget is THE slacker |
| 15:44.22 | sivana | heh |
| 15:44.29 | sivana | Master Slacker |
| 15:44.35 | file | Nugget: moo |
| 15:44.35 | Nugget | I am root@slacker.com! |
| 15:44.39 | Nugget | ]:8) |
| 15:45.15 | sivana | but can he multitask at slacking? |
| 15:45.20 | brif8 | What is the CLI command to tell you if an IP phone is busy , for how long and what is the caller ID on the call ? |
| 15:45.28 | Nugget | of course. I'm in 20 irc channels. |
| 15:45.37 | sivana | multiple slacking threads? |
| 15:45.50 | sivana | heh |
| 15:45.59 | jsharp | Hyperslacking |
| 15:46.14 | Nugget | heh |
| 15:46.48 | SuperLag | salviadud: [TK]D-Fender: this would be for home use. Mainly to play with. I have recently started working for a VoIP company, and need to get familiar with Asterisk. I have a beefy machine at home that I run Gentoo on, as well as this box, but it's in a colo facility in California. |
| 15:48.43 | Nugget | no fear, I'll register something even better this weekend. |
| 15:49.15 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 15:49.45 | nettie | [TK]D-Fender, Hi, I'm gettign mad configuring skypho.net incoming calls. I'm wondering if you have experience with that carrier? reading sip debug seems they're using Cisco Sip gateway. Do you know if asterisk needs a special configuration to work properly with it please? When I call the number from my mobile phone I keep getting User is busy. any idea please? thanx in advance. |
| 15:50.13 | SuperLag | hah. masterslacker.com and .net are taken :) |
| 15:50.20 | SuperLag | how funny |
| 15:50.42 | jsharp | No .org? Definitely a master slacker. |
| 15:51.43 | [TK]D-Fender | SuperLag : Go right ahead. I use my * box as my router, file server, and HTPC sytem all in one. |
| 15:51.58 | SuperLag | interesting |
| 15:52.09 | SuperLag | HTPC? what are you running? MythTV? |
| 15:52.25 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 15:52.45 | [TK]D-Fender | SuperLag : Perhaps an overstatement : Just running X (KDE) hooked up to my 52" HDTV, and receiver :) |
| 15:53.00 | jake1932 | brad_mssw - you still around? |
| 15:53.23 | SuperLag | nice |
| 15:53.27 | SuperLag | 52"?! |
| 15:53.59 | salviadud | fender, did you fix the resolution? |
| 15:54.10 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 15:54.20 | jake1932 | anyone know why my user name (not my phone number) would appear as my sip from address? |
| 15:54.54 | [TK]D-Fender | SuperLag : Yeah, I like to think of it as "healthy" :) I don't watch "TV" per say but have a friend who downloads EVERYTHING. |
| 15:55.21 | [TK]D-Fender | salviadud : I'm actually running at 800x600, but then again considering the media resolution is just fine. |
| 15:55.43 | [TK]D-Fender | salviadud : Though I'd like to be able to use DVI for it... just need to learn some more about X which I can't be bothered with right now. |
| 15:56.07 | Hmmhesays | yeah I download all my tv |
| 15:56.32 | brad_mssw | jake1932: yep |
| 15:56.48 | jsharp | Download em from where? Reliable? |
| 15:57.13 | jake1932 | i spoke to mike at junction - i'm sending over uname@mypi in my from - don't know why - but doesn't look like it's an issue on their side |
| 15:57.22 | jake1932 | myip (no pi) |
| 15:57.57 | jake1932 | should be sending over the phonenum@myip |
| 15:58.38 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 15:58.38 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:59.12 | jake1932 | brad_mssw - what version of asterisk are you running? |
| 15:59.25 | brad_mssw | jake1932: 1.2.7.1 |
| 15:59.49 | jake1932 | same as mine - ok - tnx - i'll check some more |
| 16:00.00 | brad_mssw | jake1932: you use iax or sip ? |
| 16:00.03 | jake1932 | sip |
| 16:00.31 | brad_mssw | jake1932: oh, I use iax to junction |
| 16:00.48 | brad_mssw | (though on my fax line, I use sip) |
| 16:00.51 | jake1932 | ok - so could be soomething with chan_sip |
| 16:01.31 | brif8 | manager.conf allows you access the current status of asterisk (right)? if so then how would you check on the status of invidivual extensions both incoming numbers and IP phones or SIP devices ? Also What is the CLI command to tell you if an IP phone is busy , for how long and what is the caller ID on the call ? |
| 16:01.55 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-198-80.telkomadsl.co.za) |
| 16:02.57 | jake1932 | brif8 - #1 is on this page http://www.voip-info.org/wiki-Asterisk+manager+API |
| 16:03.14 | jake1932 | ExtensionState |
| 16:03.26 | *** join/#asterisk Seyr (n=Seyr@grant254.grantgeo.com) |
| 16:04.13 | Seyr | what is the best way to forward a number in Asterisk? I have a DID on a server and need it to forward to a landline |
| 16:04.16 | jake1932 | brif8 - looks like you can use Status |
| 16:04.53 | *** join/#asterisk stoffell (n=stoffell@211-220.244.81.adsl.skynet.be) |
| 16:05.06 | brif8 | jake1932: thanks I'll read more |
| 16:05.22 | jake1932 | Seyr - what type of phone - on mine (7960) you can do it from the keypad |
| 16:05.32 | Seyr | no phone |
| 16:05.44 | jsharp | Set the extension in Asterisk to dial back out. |
| 16:05.49 | jake1932 | in the dialplan - just use Dial |
| 16:05.53 | Seyr | ugh |
| 16:06.04 | Seyr | so there is no Forward() or anything? |
| 16:06.12 | Seyr | I'm using Dial currently |
| 16:06.21 | jake1932 | what's wrong with Dial? |
| 16:06.27 | brif8 | what load does manager put on the CPU or asterisk system to constantly check status ? |
| 16:06.32 | *** join/#asterisk skkip (n=skkip@216.160.91.91) |
| 16:06.51 | *** join/#asterisk lzhang (n=rjrae@adsl-69-152-225-92.dsl.snantx.swbell.net) |
| 16:06.54 | Seyr | thought there was a Transfer() or something |
| 16:07.21 | a1fa | something is wrong with my asterisk box |
| 16:07.23 | jake1932 | <PROTECTED> |
| 16:07.38 | jake1932 | no need for polling in that case |
| 16:07.38 | Seyr | nothing *wrong* with Dial :-) |
| 16:07.49 | a1fa | it registers, and then i cant get calls after 10 minutes |
| 16:08.01 | a1fa | it works for the first 5 minutes, i can get calls, but after that i cant get calls |
| 16:08.02 | lzhang | I'm using asterisk to group dial some 501's and some 601's, and for some reason everytime one of the 501's picks up the call the 601 keeps ringing (using asterisk 1.2.5) |
| 16:08.19 | lzhang | anybody heard of this happening before |
| 16:09.03 | brif8 | jake1932: it is a TCP connection right for manager.conf |
| 16:09.14 | jake1932 | <PROTECTED> |
| 16:09.18 | brif8 | thanks |
| 16:09.20 | jake1932 | np |
| 16:09.47 | jake1932 | lzhang - i would make sure the 601 is getting the terminate packet |
| 16:10.14 | lzhang | jake1932: so I do a sip debug and try to find the terminate packet? |
| 16:10.20 | jake1932 | yes |
| 16:12.07 | devel | anybody seeing weirdness with aastra (480i) like it registers fine on boot, but every registration attempt after that shows 'forbidden' (doing sip debug at console)? |
| 16:13.09 | *** join/#asterisk hgaillac (n=Harry@83.15.119-80.rev.gaoland.net) |
| 16:13.29 | hgaillac | Hello asterisk users |
| 16:13.47 | perlmonky | anyone doing qos on pix? |
| 16:13.54 | a1fa | wtf |
| 16:14.10 | a1fa | i can recieve calls for the first 5 minutes past my registration, then i cant make phonecalls |
| 16:14.10 | a1fa | wtf |
| 16:14.13 | hgaillac | Is there somebody in Israel for fax tests to France |
| 16:14.32 | perlmonky | a1fa: what phone? |
| 16:14.45 | perlmonky | a1fa: are you using qualify=yes? |
| 16:14.51 | devel | alfa, i'd do a 'sip debug peer' too |
| 16:14.56 | a1fa | perlmonky : no |
| 16:14.59 | a1fa | its peer |
| 16:15.02 | devel | that's helped me countless times |
| 16:15.08 | a1fa | i did sip debugip <peer ip> |
| 16:15.14 | perlmonky | doesn't matter qualify is keep alive... |
| 16:15.17 | *** join/#asterisk stoffell (n=stoffell@211-220.244.81.adsl.skynet.be) |
| 16:15.21 | perlmonky | sort of ;) |
| 16:15.30 | a1fa | perlmonky : i dont know if a peer needs a qualify |
| 16:16.09 | perlmonky | you are not using dynamic with registration correct? |
| 16:16.24 | perlmonky | host=dynamic.. |
| 16:16.34 | a1fa | ? |
| 16:16.37 | a1fa | why would i use that |
| 16:16.45 | perlmonky | just making sure you weren't... |
| 16:17.00 | perlmonky | some do host=dynamic with registration line for peering... |
| 16:17.10 | perlmonky | just another way to do it.. |
| 16:17.28 | a1fa | i didnt do that |
| 16:17.29 | a1fa | i mean |
| 16:17.33 | a1fa | it works fine for 5 minutes |
| 16:17.37 | a1fa | and this always worked |
| 16:17.41 | perlmonky | peer doesn't do anything special other than restrict incoming calling... |
| 16:17.44 | a1fa | it stop working after update to 1.2.6 |
| 16:18.15 | perlmonky | my gut would be that your registration is dropping out... |
| 16:18.17 | perlmonky | nat? |
| 16:18.24 | a1fa | no nat |
| 16:18.35 | perlmonky | both sides astersk? |
| 16:18.59 | a1fa | broadvoice |
| 16:19.19 | perlmonky | ahh.... |
| 16:19.29 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:20.34 | perlmonky | so you asterisk has public IP? |
| 16:20.43 | a1fa | yup |
| 16:20.48 | perlmonky | what do you get when you sip debug |
| 16:20.59 | perlmonky | sip debug peer broadvoice... |
| 16:21.41 | *** join/#asterisk ast_freak|Laptop (n=jesse@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 16:21.46 | perlmonky | lets back up further... after five minutes you can't receive calls? or you can't make calls... |
| 16:22.27 | a1fa | i can make calls |
| 16:22.32 | a1fa | but i cant recieve calls |
| 16:22.47 | justinu|laptop | turn on qualify or nat keepalives |
| 16:22.55 | perlmonky | ok... are you running IP tables? |
| 16:23.47 | perlmonky | and when you are attempting to receive calls do you see any sip debug? |
| 16:23.49 | justinu|laptop | if it works for a while, then stops, the problem is your NAT is closing the binding, not allowing UDP packets from the ITSP to reach you |
| 16:23.57 | a1fa | perlmonky : not afer 5minutes |
| 16:24.00 | justinu|laptop | qualify will stop that |
| 16:24.04 | a1fa | justinu: no nat |
| 16:24.05 | justinu|laptop | from happening |
| 16:24.33 | a1fa | ok |
| 16:24.34 | justinu|laptop | if there's no nat, then your registration is expiring |
| 16:24.36 | a1fa | i added qualify=yes |
| 16:24.43 | *** part/#asterisk Seyr (n=Seyr@grant254.grantgeo.com) |
| 16:24.44 | a1fa | ok |
| 16:24.47 | a1fa | it could be the case |
| 16:24.49 | perlmonky | sip reload :) |
| 16:24.52 | a1fa | that it expires every 30s |
| 16:24.56 | a1fa | but in debug |
| 16:25.01 | a1fa | it sais expriing in 3600s |
| 16:25.03 | perlmonky | iptable can cause some of the same problems as nat |
| 16:25.10 | perlmonky | if not configured properly |
| 16:25.12 | justinu|laptop | ok... then that shouldn't be the case, if expires=3600 |
| 16:25.38 | justinu|laptop | yep... disable iptables completely for troubleshooting purposes |
| 16:26.06 | a1fa | ok |
| 16:26.07 | perlmonky | make sure you allow 5060 and 10000-20000 udb |
| 16:26.08 | a1fa | 30s after |
| 16:26.11 | a1fa | it stoped |
| 16:26.16 | perlmonky | or what ever you have rtp.conf set up for.. |
| 16:26.19 | a1fa | no more sip messages from the peer |
| 16:26.53 | a1fa | i was able to recieve a call now |
| 16:26.54 | a1fa | weerd |
| 16:27.19 | a1fa | Expires: 120 |
| 16:27.20 | a1fa | lol |
| 16:27.22 | perlmonky | qualify sends a keepalive |
| 16:27.26 | a1fa | 120/60 = 2 minutes |
| 16:27.32 | perlmonky | to keep the "door" open... |
| 16:27.43 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 16:27.47 | a1fa | it is supposed to re-register every 2minutes |
| 16:27.48 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:27.50 | a1fa | thats lame |
| 16:28.03 | a1fa | Apr 24 16:27:04 NOTICE[23117]: chan_sip.c:9693 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 3443 sec (Scheduling reregistration in 3428 s) |
| 16:28.13 | a1fa | but asterisk thinks it needs to re-register in an hour |
| 16:28.15 | *** join/#asterisk oneman (n=oneman@ip68-230-208-113.rd.hr.cox.net) |
| 16:28.17 | a1fa | now that is gay |
| 16:28.22 | oneman | hello |
| 16:28.33 | a1fa | registration is expiring in 2minutes, but asterisk will re-register in an hour |
| 16:28.52 | oneman | Can anyone reccomend a provider I can get incoming sip service for asteriks with a 757 number? |
| 16:30.29 | a1fa | get shitvoice.com |
| 16:30.44 | *** part/#asterisk lzhang (n=rjrae@adsl-69-152-225-92.dsl.snantx.swbell.net) |
| 16:31.09 | oneman | ? |
| 16:31.17 | a1fa | i am kidding |
| 16:31.28 | a1fa | want cheap non reliable provider? |
| 16:31.36 | a1fa | get broadvoice |
| 16:31.45 | a1fa | their support sucks major ass |
| 16:32.19 | jake1932 | you could do a local - full time forward to toll free - that should be pretty reliable |
| 16:32.37 | jake1932 | (plenty of providers have toll free inbound) |
| 16:32.48 | a1fa | how can i find what my 911 number is binded to? |
| 16:32.54 | zaf | voicepulse is nice |
| 16:33.06 | a1fa | i want to bind my 911 number to a local 911 number |
| 16:33.15 | a1fa | is there a way to find the local emergency number? |
| 16:33.27 | Hmmhesays | call the local police department |
| 16:33.32 | a1fa | i have a different areacode on my account |
| 16:33.35 | zaf | call 911 and ask them |
| 16:33.40 | zaf | heh |
| 16:33.41 | a1fa | zaf : lol |
| 16:33.43 | jake1932 | don't call 911 for that! |
| 16:33.47 | a1fa | 911 - What is your emergency |
| 16:33.59 | a1fa | Me - Yes, i need to know what is the emergency number |
| 16:34.01 | zaf | "hello, 911? what's the number for 911?" |
| 16:34.04 | a1fa | 911 - Sir, its 911 |
| 16:34.19 | a1fa | Me - Yes I know, but what is the number number for 911 number |
| 16:34.32 | oneman | voicepulse does not offer 757 numbers.. |
| 16:34.38 | a1fa | 911 - Sir, I am calling the police now, get the fuck off the line |
| 16:34.52 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 16:35.06 | jake1932 | oneman: you want cheap or reliable? |
| 16:35.15 | SexyKen | Hey [TK]D-Fender -- you around? |
| 16:35.50 | oneman | id like to be able to compare cheap to reliable :P |
| 16:35.56 | oneman | if cheap was 4$ a month id do it |
| 16:36.02 | jake1932 | oneman: do you have an address in the 757? |
| 16:36.05 | oneman | but if reliable is 10$ ill go for it |
| 16:36.12 | oneman | yeah it must be a 757 number |
| 16:36.19 | jsharp | Nah. Most 911 operators I've dealt with are decent if you start off with "I have no emergency, I need 911 information for my office phone system." |
| 16:36.28 | *** join/#asterisk SeicherlBoB (n=noyb@dsl-93-192.utaonline.at) |
| 16:36.37 | SeicherlBoB | hi there! anyone installed asterisk@home 2.8 recently? |
| 16:36.58 | jake1932 | hehe |
| 16:37.32 | jake1932 | SeicherlBoB - you might get a better response from #freepbx |
| 16:37.35 | oneman | jake1932: what you got for me ;P |
| 16:37.52 | SeicherlBoB | jake1932: thanks. just read the hint. sorry. |
| 16:37.58 | jake1932 | oneman: i'm saying that you could get an incoming only local and forward to a toll free |
| 16:38.05 | coppice | zaf: the number for 911 is 112 :-) |
| 16:38.27 | jake1932 | coppice - perspective - the number for 112 is 911 |
| 16:38.54 | coppice | does 112 work in the US yet? |
| 16:39.09 | jake1932 | i wouldn't dial it in an emergency |
| 16:39.31 | coppice | you probably would if you use a cellphone :-) |
| 16:39.32 | a1fa | in case of emergency, RUN BITCH! RUN! |
| 16:39.42 | oneman | jake1932: I don't want the callers to have to get thru a pre-menu |
| 16:39.53 | jake1932 | oneman: a premenu? |
| 16:39.59 | a1fa | if you are getting killed press 1 |
| 16:40.07 | a1fa | if you are vitnessing a crime, press 2 |
| 16:40.07 | SexyKen | Anyone here ever develop queue status scripts using PHP and the manager API? |
| 16:40.09 | jsharp | Thank you for calling 911. If your house is on fire, press 1. If you are dying from a heart attack, press 2. If there are gangsta thugs... |
| 16:40.11 | *** part/#asterisk SeicherlBoB (n=noyb@dsl-93-192.utaonline.at) |
| 16:40.21 | a1fa | if you or someone else is in immidate emergency press 3 |
| 16:40.45 | coppice | if this is a medical emergency and you are suffering a hearing problem, press 5 |
| 16:40.51 | oneman | jake1932: I don't think a local number with verizon will be possible due to my living arangments and such |
| 16:41.27 | jsharp | All our 911 operators are currently servicing other victims at this time. Please continue to hold and your call will be answered just before you die. |
| 16:41.28 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.224) |
| 16:41.41 | DarKnesS_WolF | res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on localhost. Check debug for more info. <--- i wann try the realtime how to turn on this debug ? |
| 16:42.32 | coppice | there was a case in the US recently where a little kid phoned 911 three times about his sick mum. they wouldn't take him seriously, and his mum died. I hope the operators suffer for that |
| 16:42.56 | jake1932 | oneman: you can get one from vonage and point it somewhere else |
| 16:42.59 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 16:43.02 | jsharp | Someone's going to own that 911 center. |
| 16:43.38 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:43.40 | oneman | jake1932: would'nt this just increase latency ? Can't I just get a proper inbound service without forwarding? |
| 16:44.30 | jake1932 | oneman: http://www.junctionnetworks.com |
| 16:44.47 | jake1932 | they're pretty good |
| 16:46.31 | b00mer_ | how does one set a db variable to null? I currently have Set(DB(fwd_db/${CALLERIDNUM})="") |
| 16:46.38 | SexyKen | Hey guys, when getting Queue Status using the Manager API, how is the hold time calcuated? |
| 16:46.39 | b00mer_ | but that seems to store "" |
| 16:47.08 | b00mer_ | which is srewing with other scripts |
| 16:47.26 | b00mer_ | s/srewing/screwing |
| 16:47.52 | *** join/#asterisk kend (n=chatzill@londonderry-cuda3-68-64-252-249.lndnnh.adelphia.net) |
| 16:47.53 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 16:49.08 | marl | hi, does anyone have a macro (or is it a built in setting i just cant find?) that will ether record nativly in mp3/ogg format, or convert after a line monitor recording to mp3/ogg? |
| 16:49.24 | marl | curently using AAH2.8 |
| 16:49.40 | oneman | jake1932: junctionnetworks is pay per minute inbound ?!? |
| 16:49.55 | jake1932 | oneman - yes |
| 16:50.09 | oneman | I want flat rate inbound :D |
| 16:50.18 | a1fa | oneman : how about no inbound charges |
| 16:50.26 | oneman | a1fa: word |
| 16:50.37 | jake1932 | i haven't yet found a flat rate that has been able to provide reliable service |
| 16:51.08 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 16:51.23 | a1fa | qualify=yes fixed my inbound issue. but it worked without qualify=yes in 1.2.5.. i wonder what has changed that it must use qualify=yes now |
| 16:51.30 | nettie | hey guys anyone rember how to use wildcards with ${CALLERIDNUM} please? I would like to match all the numbers having CID starting with "3" |
| 16:51.40 | DarKnesS_WolF | res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on localhost. Check debug for more info. <--- i wann try the realtime how to turn on this debug ? |
| 16:51.41 | oneman | a1fa: what you reccomend? |
| 16:51.48 | [TK]D-Fender | nettie : Runnin * 1.0.x? |
| 16:51.53 | nettie | mnope 1.2.6 |
| 16:52.03 | znoG | OT question... if one plugs a device which is plugged into a 110v power adapter --> into a 220v socket... apart from blowing the power adapter (which is what effectively has happened), what are the chances that the device is blown too? |
| 16:52.09 | a1fa | oneman : if you want cheap, go with broadvoice.. add me as referer |
| 16:52.27 | a1fa | znoG : slim |
| 16:52.44 | coppice | znoG: I suspect from the tone of what you wrote "pretty good" :-) |
| 16:52.46 | a1fa | znoG : unless you have a dirt cheap power adapter that didnt terminate current |
| 16:52.47 | [TK]D-Fender | nettie : GotoIf($["${CALLERID(num):1}"="3"]?4) |
| 16:52.57 | b00mer_ | how does one set a db variable to null? I currently have Set(DB(fwd_db/${CALLERIDNUM})="") |
| 16:53.08 | znoG | i've had a few people tell me chances are slim that the device is blown too, so i reaaaaaaaally hope it's the case. |
| 16:53.42 | jake1932 | nettie: at one time you cound do _X/_Y where X is the ANI and Y is the number called |
| 16:53.53 | Qwell[] | jake1932: still can |
| 16:53.55 | [TK]D-Fender | b00mer_ : try Set(DB(fwd_db/${CALLERIDNUM})=) |
| 16:54.29 | oneman | broadvoice aint got 757 :P |
| 16:54.40 | a1fa | why do you need 757? |
| 16:54.47 | a1fa | where the f. is 757? |
| 16:55.12 | nettie | thanx guys, [TK]D-Fender :1 is the part I need right? is it used to have asterisk match jsut the first digit right? |
| 16:55.15 | oneman | eastern virginia |
| 16:55.39 | a1fa | damn |
| 16:55.43 | [TK]D-Fender | nettie :Oops... got that slightly wrong... |
| 16:55.47 | a1fa | good thing its not west virginia |
| 16:55.48 | [TK]D-Fender | nettie : GotoIf($["${CALLERID(num):0:1}"="3"]?4) |
| 16:55.51 | [TK]D-Fender | nettie : there |
| 16:55.54 | nettie | k |
| 16:56.01 | nettie | it didnt work ehehe |
| 16:56.08 | [TK]D-Fender | nettie : Go read up on "asterisk variables" in the WIKI |
| 16:56.16 | b00mer_ | [TK]D-Fender : thanks again! that worked perfect |
| 16:56.16 | nettie | ok |
| 16:56.17 | nettie | thanx |
| 16:56.24 | [TK]D-Fender | b00mer_ : ywc |
| 16:56.25 | a1fa | nettie : did you reload? |
| 16:56.36 | nettie | sure :p |
| 16:56.43 | nettie | now orks eheh |
| 16:56.44 | nettie | works |
| 16:57.10 | a1fa | u didnt reload, did you |
| 16:57.18 | a1fa | bad nettiem bad bad nettie |
| 16:57.35 | *** join/#asterisk ketema (n=ketema@adsl-072-156-236-193.sip.mco.bellsouth.net) |
| 16:58.32 | nettie | [TK]D-Fender do you know how do I match a number with doesnt pass the CID please? is it in the wiki as wekk? |
| 16:58.34 | nettie | wiki |
| 16:58.42 | drfoomod2 | is anyone using a cisco router and a voice t1 card as a sip endpoint? |
| 16:59.21 | drfoomod2 | i was thinking about buying a used 2621 and a VWIC-1MFT-T1 |
| 16:59.52 | jake1932 | nettie: what are you matching on? |
| 17:00.15 | nettie | I would like to match also the callers which are not sending the CID |
| 17:00.28 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 17:00.34 | kink0 | hello |
| 17:00.39 | nettie | maybe cid = "" |
| 17:00.40 | nettie | uhmm |
| 17:00.41 | [TK]D-Fender | nettie : As in you want to know if the call nas no callerid? |
| 17:00.50 | nettie | exaclty |
| 17:00.54 | jake1932 | oh |
| 17:01.05 | [TK]D-Fender | nettie : GotoIf($["${CALLERID(num)}"=""]?4) |
| 17:01.08 | nettie | ok |
| 17:01.14 | nettie | perfect as I thought eheh |
| 17:01.15 | nettie | "" |
| 17:01.16 | [TK]D-Fender | null is null.... |
| 17:01.16 | nettie | <PROTECTED> |
| 17:01.18 | nettie | thanx |
| 17:01.19 | kink0 | when my Asterisk has load there more and more S asterisk process until memory is exahust, is that normal ? |
| 17:01.30 | jake1932 | wouldn't it better to do the _X/_Y for this thing? |
| 17:03.05 | kink0 | about my last day question where Digium TE40X was detected but layer 3 was never working, the cause was PCI slot speed, then I change to 66Mhz instead Auto(133Mhz) and all goes OK, if anybody suffer for same problem. |
| 17:03.48 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-38.dsl.irvnca.pacbell.net) |
| 17:04.04 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 17:04.14 | nettie | perfect thanx guys wortlks gr8 |
| 17:04.18 | nettie | works :) |
| 17:05.20 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 17:05.26 | *** join/#asterisk hads (n=hads@203.109.245.87) |
| 17:07.48 | b00mer_ | is mpg123 still required for MOH on 1.2+ ? |
| 17:08.00 | [TK]D-Fender | b00mer_ : Nope... native MoH... |
| 17:08.24 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 17:08.34 | acehunky | hi |
| 17:09.25 | b00mer_ | my MOH starts and then immediately stops |
| 17:09.42 | jake1932 | b00mer_ - mp3? |
| 17:10.00 | b00mer_ | jake1932 : yes |
| 17:10.14 | jake1932 | b00mer_ - you might need to compile format_mp3 in the addons |
| 17:10.32 | b00mer_ | doughecka_: I forgot why... but that gave me trouble |
| 17:11.09 | *** join/#asterisk forme (i=1000@213.27.44.55) |
| 17:11.18 | b00mer_ | ugh... it was last week too... my brain is frazzeled milk toast |
| 17:14.31 | LostFrog | I still prefer mpg123. |
| 17:14.32 | mut | these sangoma a104d's can also do data, correct? |
| 17:14.32 | jake1932 | LostFrog: why? |
| 17:14.32 | LostFrog | jake1932: randomness. |
| 17:14.38 | jake1932 | LostFrog: in terms of reliability? |
| 17:14.43 | kink0 | noboy knows if is normal Asterisk forks more and more S proccess until memory is exhaust ? |
| 17:14.52 | *** part/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 17:14.59 | file | Asterisk does not fork, it is multithreaded |
| 17:15.07 | LostFrog | no.. I just don't like hearing the start of the MP3s over and over again.. I want to hear the middles too. |
| 17:16.11 | kink0 | file: well right, many threads then , over 100 and near to use 100% of RAM ( swap is not used, I don't know why ) and then asterisk goes innestable and finally crash or need to be killed |
| 17:16.32 | file | no that's not normal, get a back trace and figure out what's causing so many threads |
| 17:16.33 | jake1932 | LostFrog - that's pretty anal |
| 17:17.10 | LostFrog | That's nice.. you are calling my preferences anal.. |
| 17:17.20 | kink0 | file: is caussing where a lot of concurrents calls are arriving, and then claims about is unnable to create a new pbx |
| 17:17.23 | LostFrog | Do we all have to feel the same way about everything. |
| 17:17.41 | kink0 | file: I had increased a lot ulimit, but that appears does not fix the problem. |
| 17:17.43 | jake1932 | LostFrog - it's music on hold - for gods sake |
| 17:18.05 | file | kink0: figure out what is eating up so much memory, what are the calls doing? |
| 17:18.20 | b00mer_ | hmmm |
| 17:18.28 | b00mer_ | mOH is still not working... |
| 17:18.35 | b00mer_ | format_mp3 is compiled and installed |
| 17:18.35 | jake1932 | LostFrog - just joking with you anyways - no offense intended |
| 17:18.42 | b00mer_ | loaded the module |
| 17:18.53 | b00mer_ | still getting the start/stop |
| 17:19.02 | [TK]D-Fender | b00mer_ : pastebin your musiconhold.conf |
| 17:19.03 | jake1932 | <PROTECTED> |
| 17:19.04 | [TK]D-Fender | ~pb |
| 17:19.06 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:19.31 | kink0 | file: free Total:2000K Free:about 60k when start inestable |
| 17:19.35 | b00mer_ | [TK]D-Fender : its the default one |
| 17:19.43 | b00mer_ | [TK]D-Fender : haven't changed anything |
| 17:20.04 | [TK]D-Fender | b00mer_ : Show us what you're doing, and pastbin some CLI to back it up.. |
| 17:20.10 | kink0 | file: calls are doing mainly Congestion when that happens, due to no more channels availables to route the call ( my PRI is connected to a PRI GSM gateway ) |
| 17:20.28 | file | kink0: what technology are they coming in as? |
| 17:20.30 | LostFrog | wow, kink0, that must cost. |
| 17:20.31 | *** join/#asterisk ToTo (n=ToTo@host182-49.pool870.interbusiness.it) |
| 17:21.02 | kink0 | file: are ussing SIP and H323 , both |
| 17:21.10 | file | ugh H323 |
| 17:21.25 | kink0 | LostFrog, yes, are expansive equipments, I use 2N Stargate gateway |
| 17:21.27 | oneman | anyone used telesip? |
| 17:21.41 | file | where is this... |
| 17:21.41 | kink0 | file: hmmmmm h323... I was also suspecting about h323 |
| 17:21.58 | file | kink0: edit Makefile in the main directory, and take out the # before -include on the MALLOC_DEBUG line |
| 17:22.07 | file | kink0: it should allow you to see what is allocating memory, and not freeing it |
| 17:22.20 | [Airwolf] | Can anyone tell me if there are any free webphonebooks avalible for Asterisk ? |
| 17:22.34 | [Airwolf] | That like create a phonebook from the voicemail configuration of something ? |
| 17:22.37 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 17:22.48 | znoG | no, but that shouldn't be too hard to do |
| 17:22.49 | b00mer_ | [TK]D-Fender : http://pastebin.com/679282 |
| 17:23.18 | [Airwolf] | znoG, I'm not a web programmer. Just a network engineer. ;) |
| 17:23.26 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:23.54 | b00mer_ | [TK]D-Fender : here is my musiconhold.conf http://pastebin.com/679287 |
| 17:24.15 | kink0 | file: is uncomented on my Makefile, but show memory returns not know command in the CLI |
| 17:24.19 | znoG | [Airwolf]: i'm a network guy (not engineer as I don't have a title yet) but it's really not hard to do. |
| 17:24.22 | file | kink0: you have to recompile |
| 17:24.27 | file | kink0: and reinstall |
| 17:24.39 | file | kink0: and restart Asterisk... |
| 17:24.53 | file | if it still isn't there, then you did not do something correctly |
| 17:24.56 | kink0 | yes, but was compiled and installed with this Makefile, I have not change now, was with memory debug enabled when I compiled it last time |
| 17:25.08 | [TK]D-Fender | b00mer_ : Now I'd like to see your musiconhold.conf..... |
| 17:25.22 | b00mer_ | [TK]D-Fender : here is my musiconhold.conf http://pastebin.com/679287 |
| 17:25.28 | [TK]D-Fender | b00mer_ : umm, think I just missed that :) |
| 17:25.36 | [TK]D-Fender | b00mer_ : WRONG MODE <- |
| 17:25.42 | [Airwolf] | znoG, I know. It won't be that hard, but If somebody already did it, then why do it again. :) |
| 17:25.48 | kink0 | I will try to recompile anywise, and try again show memory .... |
| 17:25.52 | [TK]D-Fender | b00mer_ : Go read up on how to enable native MoH |
| 17:25.55 | kink0 | give me a sec to do it. |
| 17:26.05 | b00mer_ | [TK]D-Fender :( |
| 17:26.22 | ketema | \q |
| 17:26.26 | ketema | \quit |
| 17:26.32 | perlmonky | b00mer_ you are using mpg123? |
| 17:26.39 | b00mer_ | perlmonky no |
| 17:26.47 | perlmonky | than that won't work... |
| 17:26.51 | perlmonky | change mode=files |
| 17:27.03 | perlmonky | and install asterisk_addons with format_mp3.so |
| 17:27.24 | b00mer_ | ok... stupid question... I've been relying on voip-info.org and "the book"... where is the official 1.2.7.1 docs? |
| 17:27.34 | perlmonky | hahahaha |
| 17:27.46 | b00mer_ | both have soo much deprecated and pre 1.2 stuff |
| 17:27.47 | noname32 | does anyone know if it is possible to make a recorded line make a beeping noise so the parties know that it is a record line? |
| 17:27.51 | perlmonky | b00mer_ sorry... that always gets me... |
| 17:27.53 | b00mer_ | I'm pulling my hair out |
| 17:28.09 | b00mer_ | not that I have much left |
| 17:28.09 | [TK]D-Fender | perlmonky : Hey I was hoping he'd READ to figure that out once I told him what was wrong! |
| 17:28.18 | perlmonky | the best place to look is /usr/src/asterisk/docs |
| 17:28.18 | file | kink0: it's show memory allocations btw |
| 17:28.23 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
| 17:28.28 | [TK]D-Fender | b00mer_ : There is plenty of good stuff on native MoH on the wiki... |
| 17:28.30 | salviadud | official docs? |
| 17:28.34 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 17:28.38 | [TK]D-Fender | and in the SAMPLE files... |
| 17:28.38 | file | or show memory summary |
| 17:28.41 | salviadud | duuuude, do you even use asterisk? |
| 17:28.45 | file | you'll probably want show memory summary |
| 17:28.52 | [TK]D-Fender | salviadud : referring to me? |
| 17:29.00 | b00mer_ | [TK]D-Fender I got it... I am a question asking slacker... I am going to wack my own pee-pee and hide in a closet to read |
| 17:29.01 | salviadud | no, to b00mer |
| 17:29.43 | salviadud | yeah, make sure the closet is comfy |
| 17:30.45 | [TK]D-Fender | b00mer_ : I give a lot of info straight, but when I give the obvious search context and even the exact FIELD to look for I like people to try just a LITTLE you know? |
| 17:30.47 | perlmonky | [TK]D-Fender I understand the principal of Teach a man to fish... but every once in a while you gotta throw him a bone... |
| 17:31.09 | [TK]D-Fender | perlmonky : I gave him a bone... plenty in fact, jsut not the whole fish! |
| 17:31.14 | perlmonky | maybe cause I just got back from lunch and didn't catch the rest of the conversation... |
| 17:31.17 | perlmonky | :) |
| 17:32.08 | *** join/#asterisk stoffell (n=stoffell@125-40.245.81.adsl.skynet.be) |
| 17:32.25 | [TK]D-Fender | b00mer_ : Ok, fine.... but does it WORK now at least? :) |
| 17:32.31 | marcus2 | has anyone heard any status updates from nufone regarding 800 DIDs? |
| 17:32.57 | b00mer_ | I haven't tried what perlmonkey has told me... I am trying to read about it first |
| 17:34.19 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 17:34.27 | jake1932 | any people in here speak spanish also - is "cents" not used? |
| 17:34.35 | marcus2 | 3/wii jerjer |
| 17:34.36 | marcus2 | ack |
| 17:34.52 | ariel_ | jake1932, cents like pennies |
| 17:34.57 | jake1932 | yes |
| 17:34.59 | Qwell[] | cents like pesos? |
| 17:35.02 | Qwell[] | The currency? |
| 17:35.06 | jake1932 | like pennies |
| 17:35.13 | jake1932 | $10.25 |
| 17:35.13 | ariel_ | centados |
| 17:35.13 | Qwell[] | No, they use pesos :p |
| 17:35.35 | jake1932 | cause i don't see cents in a few of these sound file packs |
| 17:35.40 | jake1932 | (only dollars) |
| 17:36.17 | jake1932 | so how would you say $10.11 in spanish? |
| 17:36.26 | Qwell[] | USD? |
| 17:36.28 | jake1932 | yes |
| 17:36.39 | jake1932 | or Canadian :) |
| 17:36.48 | Qwell[] | Same way you would say 10.11 in EUD, I imagine |
| 17:36.55 | mocker | Is there a way to run a command when a call is complete through the dialplan? |
| 17:36.57 | znoG | centavos << cents in spanish |
| 17:37.04 | mocker | i.e. user hangs up then run system() command. |
| 17:37.12 | ariel_ | znoG, I said that |
| 17:37.12 | znoG | mocker: use the h extension |
| 17:37.16 | znoG | ariel_: you said centados |
| 17:37.20 | mocker | znoG: Thanks. |
| 17:37.22 | jake1932 | in english i.e - ten dollars and eleven cents |
| 17:37.28 | znoG | which means "seated", except its with an "s", not a "c" :) |
| 17:37.29 | docelm0 | dies y dies uno centavos |
| 17:37.33 | Qwell[] | jake1932: Just like that |
| 17:37.35 | ariel_ | spelling in spanish is bad for me. |
| 17:37.40 | docelm0 | same |
| 17:37.41 | ariel_ | also they say kilo's |
| 17:37.45 | znoG | diez (10) dólares con 11 (once) centavos |
| 17:38.00 | jake1932 | docelm0 - tnx - must assume "centavos" was just an oversight in these language packs |
| 17:38.24 | Qwell[] | jake1932: Is there "euros" in there? |
| 17:38.31 | jake1932 | or just a bunch of rich spanish people the don't deal in cents |
| 17:38.38 | jake1932 | qwell - nope |
| 17:38.41 | docelm0 | When my spanish speaking CSR comes back from lunch I will ask her the exact.. |
| 17:38.47 | Qwell[] | and does "dollars" say "pesos"? |
| 17:39.00 | salviadud | dollars are dolares |
| 17:39.05 | Qwell[] | If so, it's hardly an oversight |
| 17:39.05 | znoG | no, dollars are dolares |
| 17:39.06 | ariel_ | dolares |
| 17:39.08 | znoG | its a currency |
| 17:39.11 | jake1932 | qwell - it says dolars in spanish properly |
| 17:39.12 | docelm0 | Quell then you have to do a conversion from USD to MEX |
| 17:39.18 | docelm0 | err PES I think it is |
| 17:39.22 | znoG | for example, here in Argentina, 1 USD is about 3 pesos |
| 17:39.36 | jake1932 | yep - i'm talking about US Spanish people |
| 17:39.36 | salviadud | znog, che argentino |
| 17:39.39 | ariel_ | Peso is not the same as dolares |
| 17:39.47 | znoG | salviadud: como va mejicano? |
| 17:40.08 | Qwell[] | correct me if I'm wrong, but their currency has no fractional amounts, equiv cents |
| 17:40.08 | docelm0 | hay jake1932 since they are in the US.. Tell em to learn english like everyone else.. |
| 17:40.13 | salviadud | znog, muy bien jaja |
| 17:40.15 | docelm0 | I HATE that about tampa |
| 17:40.20 | jake1932 | <PROTECTED> |
| 17:40.24 | Qwell[] | docelm0: Try living in socal |
| 17:40.37 | docelm0 | 1/3 of tampa only speaks spanish.. |
| 17:40.41 | docelm0 | It SUCKS! |
| 17:40.47 | docelm0 | Qwell I feel ya.. |
| 17:40.48 | Qwell[] | 9/10 of LA speaks Spanish :P |
| 17:40.51 | docelm0 | hehe |
| 17:40.55 | salviadud | learn the freakin language |
| 17:41.01 | salviadud | you think english is hot or something? |
| 17:41.01 | docelm0 | damn illegals.. |
| 17:41.03 | a1fa | butchers |
| 17:41.10 | Qwell[] | salviadud: very hot |
| 17:41.11 | salviadud | english is the easiest language ever |
| 17:41.16 | docelm0 | salviadud hay its like this.. wanna be in the US learn our language |
| 17:41.22 | Qwell[] | salviadud: Good, so they can learn it! :P |
| 17:41.23 | docelm0 | doesnt have to be primary but still |
| 17:41.25 | salviadud | i learned english by watching nickelodeon |
| 17:41.26 | jake1932 | (what did i start here?) :) |
| 17:41.36 | docelm0 | jake1932 a can of worms.. |
| 17:41.39 | Qwell[] | salviadud: heh, nice |
| 17:41.41 | docelm0 | jake1932 where ya from? |
| 17:41.45 | brettnem | mmmm worms. |
| 17:41.47 | jake1932 | philly area |
| 17:42.00 | salviadud | lots of family double-dare in my time... |
| 17:42.06 | a1fa | haha |
| 17:42.07 | docelm0 | jake1932 really? I will be in Newark De end of next month for Labor Day |
| 17:42.11 | a1fa | i learned english watching movies |
| 17:42.13 | a1fa | john wayne |
| 17:42.14 | stoffell | tzafrir, hm, how can i reduce the logging o/t astribank (logs continuously, alot..) |
| 17:42.14 | nettie | guys, considering I would like to stop using dial prefix when I want to call a number on the pstn, how asterisk descriminate from the internal 3 digit number and the external "n" digits number? |
| 17:42.19 | LostFrog | jake1932: ever make it out to western PA? |
| 17:42.31 | docelm0 | LostFrog your in Pittsburgh right? |
| 17:42.37 | jake1932 | LostFrog: once |
| 17:42.37 | LostFrog | Yep. |
| 17:42.51 | jake1932 | LostFrog: can't remember why though |
| 17:42.54 | docelm0 | I will be there the weekend before memorial day next month |
| 17:42.56 | ariel_ | nettie, patter matching |
| 17:43.04 | LostFrog | docelm0: buy you a drink? |
| 17:43.22 | nettie | ariel_ those were the keywords I need.. thanx a lot |
| 17:43.22 | nettie | eheh |
| 17:43.29 | docelm0 | works for me.. I land around 1 or so.. Going straight to Permanni Brothers |
| 17:43.58 | LostFrog | primanti? |
| 17:44.05 | LostFrog | oops.. nm |
| 17:44.09 | docelm0 | I cant spell.. leave me alone.. :( |
| 17:44.20 | docelm0 | but yes.. Im going after a pittsburgher |
| 17:44.35 | docelm0 | havent had one in quite awhile |
| 17:44.47 | Qwell[] | docelm0: mail me one |
| 17:44.55 | Qwell[] | dryice and overnight it |
| 17:44.59 | b00mer_ | [TK]D-Fender : no haven't gotten MoH to work yet... I have compiled format_mp3, installed it. I have uncommented the native section of the musiconhold.conf file. I have put mp3s in the /var/lib/asterisk/moh-native directory and finally edited zapata.conf to say musiconhold=native |
| 17:45.38 | *** join/#asterisk brockj49464 (n=brockj49@41.105.dhcp.hope.edu) |
| 17:46.05 | docelm0 | Qwell[] you know what one is? |
| 17:46.09 | Qwell[] | no |
| 17:46.18 | Qwell[] | but I'm sure it won't suck :P |
| 17:46.22 | docelm0 | dude.. the burger is bigger that you are |
| 17:46.47 | `Kevin | i cannot get t1/pri debugging in the cli, i am troubleshooting why the d channel will not come up when connected to a shortel |
| 17:47.18 | `Kevin | are their any specific commands to enable verbose pri debugging output? |
| 17:47.26 | docelm0 | zap debug |
| 17:47.28 | docelm0 | thats bout it |
| 17:47.37 | LostFrog | Is that the cheesesteak, docelm0? |
| 17:47.46 | b00mer_ | [TK]D-Fender still the same stop start |
| 17:47.50 | b00mer_ | oops |
| 17:47.58 | b00mer_ | start stop |
| 17:48.07 | b00mer_ | ls |
| 17:48.14 | docelm0 | LostFrog, no its the 2 burgers with cheese, cabbage, and fries.. I think they may call it the cheese steak |
| 17:48.21 | jake1932 | i think there's a pri intense debug |
| 17:48.24 | docelm0 | not to be confused with a phili cheesesteak |
| 17:48.27 | Qwell[] | fries...on the burger? |
| 17:48.31 | docelm0 | yes |
| 17:48.34 | docelm0 | AND CABBAGE! |
| 17:48.39 | Qwell[] | awesome |
| 17:48.42 | Qwell[] | so, like I said... :P |
| 17:48.57 | LostFrog | Fries on burgers are a pbgh standard. |
| 17:48.59 | docelm0 | Qwell[] its a pittsburgh thing.. man we need to have an astricon there |
| 17:49.00 | LostFrog | And on subs |
| 17:49.00 | `Kevin | zap debug = unknown cm |
| 17:49.08 | docelm0 | Zap isnt loaded |
| 17:49.17 | `Kevin | their are zap cmds |
| 17:49.21 | docelm0 | You didnt compile it right or broke it |
| 17:49.27 | LostFrog | I like the unique pizza factory works. |
| 17:49.53 | docelm0 | LostFrog never been there.. I havent been in pittsburgh in about 9 years |
| 17:50.01 | docelm0 | actually 6 |
| 17:50.18 | x86 | how do you force load a peer into astdb from realtime? |
| 17:50.40 | docelm0 | x86 database put bla bla bla |
| 17:50.51 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:50.58 | x86 | err no ;) |
| 17:51.14 | x86 | there was a simple way to force it to propagate even if the realtime peer wasnt connected |
| 17:51.23 | x86 | not involving any astdb commands |
| 17:52.26 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:52.38 | x86 | ah |
| 17:52.39 | x86 | found it |
| 17:52.44 | x86 | sip show peer <num> load |
| 17:53.04 | kink0 | file: still you there ? |
| 17:53.07 | *** join/#asterisk ast_freak|Laptop (n=jesse@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 17:53.17 | file | aye |
| 17:53.42 | kink0 | file: I have now two asterisk, in differents machines, both compiled with MALLOC , but show memory on both returns not command :( |
| 17:53.51 | file | show memory summary |
| 17:53.53 | kink0 | file: version 1.2.5 |
| 17:54.20 | kink0 | file: No such command 'show memory' |
| 17:54.34 | kink0 | I type manually full phrase: show memery summary |
| 17:55.06 | file | are you sure? it's show memory summary |
| 17:55.07 | kink0 | also I edited and check astmm.h |
| 17:55.22 | file | don't edit that... you just need to edit the main Makefile, commenting out the # in front of the -include |
| 17:55.28 | file | then make clean, make, make install |
| 17:55.32 | kink0 | yes I am sure, I checked twice and that happens on two differents machines |
| 17:55.43 | kink0 | ok, will recheck on one doing a clean |
| 17:55.55 | kink0 | file : I edited just to see the .h |
| 17:56.03 | *** join/#asterisk kisu (n=daniel@2001:618:400:0:0:0:da26:a0d2) |
| 17:57.16 | kink0 | file: ahh ok, I think I saw the mistake ... the # mark was not at the begin of the line. :) |
| 17:57.27 | file | correct |
| 17:57.38 | file | thus why I said "in front of the -include" |
| 17:57.52 | opc0de | hey can anyone help me? I'm having a problem where a call from a PSTN line doesn't hang up immediately, and asterisk starts recording blank voicemail messages.. |
| 17:58.01 | opc0de | is there some setting somewhere that I can modify to change this behaviour? |
| 17:58.06 | file | opc0de: analog? |
| 17:58.25 | opc0de | file: yeah, Sangoma FXO card hooked up to PSTN line |
| 17:58.39 | file | opc0de: welcome to the wonderful world of analog |
| 17:58.43 | opc0de | so I'm screwed? |
| 17:58.48 | file | well, no |
| 17:58.52 | opc0de | heh |
| 17:59.00 | opc0de | what are my options? |
| 17:59.01 | file | you can try to get Asterisk to detect the hangup, but ymmv... |
| 17:59.17 | file | there's docs on voip-info.org and the mailing list, and tons of other places |
| 17:59.18 | praet | is there a log for this chan? |
| 17:59.23 | file | because we usually get someone like you every 2 days |
| 17:59.50 | opc0de | file: I was trying not to disappoint |
| 18:00.06 | *** join/#asterisk nitam (n=none@201.138.73.214) |
| 18:00.06 | file | it's just the fun of analog... |
| 18:00.12 | opc0de | I found the page on voip-info.org about Asterisk Disconnect Supervision, I think that's what yhou're referring to |
| 18:00.16 | file | yes |
| 18:00.30 | opc0de | I checked the mailing list, an d I found a bunch of people asking the same question, but no answers |
| 18:00.32 | file | you can sometimes get it from the telco |
| 18:01.03 | file | where they will... let me try to remember... reverse the polarity I think to indicate disconnect... or something, otherwise you have to tweak your config to try various options that tries to detect it |
| 18:02.58 | nettie | ariel_ I was able to get it working with pattern matching :) outside calls using the voip carrier are fine, local calls arent :( I need to dial an extension of a local context. any idea please? |
| 18:03.07 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 18:05.55 | kink0 | file: recompiling.... ( apparentelly required a make clean ) |
| 18:08.37 | kink0 | file: grrrr now compilation errors, I did not modify the .h but : astmm.h:49: parse error before "va_list" |
| 18:09.13 | file | grab a fresh copy and enable it... just to be sure |
| 18:13.46 | nettie | got it :) |
| 18:14.31 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 18:15.00 | kink0 | file: in the fresh copy astmm.h is the same I had, but anywise recompiling again. |
| 18:15.33 | kink0 | file:sure source for memory are ok in 1.2.5 ? |
| 18:16.05 | file | I don't use 1.2 releases, so who knows... plus 1.2.7.1 is newest... |
| 18:16.40 | praet | can you route one asterisk box to another? |
| 18:17.42 | file | praet: that's like asking can VoIP phones talk to other VoIP phones - the answer is yes, you just have to set it up... provided you meant can Asterisk boxes call eachother |
| 18:18.09 | praet | exactly.. so the sending box asts like a voip phone |
| 18:18.30 | file | of course it can... |
| 18:18.42 | praet | excellent |
| 18:20.10 | *** join/#asterisk bartpbx (n=bartpbx@p54B03111.dip0.t-ipconnect.de) |
| 18:20.10 | salviadud | can asterisk do my math homework? |
| 18:20.30 | salviadud | provided i wrote an agi |
| 18:20.47 | salviadud | i heard pulver did something similar |
| 18:20.52 | LostFrog | salviadud: depends on what math you are taking. |
| 18:20.55 | praet | are x100p cards recommended? i see generic and pro? card on ebay) |
| 18:21.12 | salviadud | simple multiplications |
| 18:21.16 | salviadud | 2 times 5 |
| 18:21.18 | Strom_C | praet: no...the TDM400P works far better than the x100p |
| 18:21.19 | salviadud | stuff like that |
| 18:21.22 | bartpbx | hi |
| 18:21.25 | LostFrog | salviadud: I don't see why not. |
| 18:21.36 | *** join/#asterisk dahunter3 (n=dahunter@64.239.166.5) |
| 18:21.53 | LostFrog | I wouldn't expect it would be easy to do calculus or linear algebra over the phone. :) |
| 18:22.12 | praet | Strom_C: do you use a fxo line? |
| 18:22.45 | acehunky | anyone has experienced callerid issue with TDM400P and Indian PSTN lines ? |
| 18:23.18 | Strom_C | praet: I have used the TDM400P with FXO lines on several installs |
| 18:23.20 | `Kevin | anyone have any idea on why calling out a pri would say congestion all lines in use, and the d channel stays down? the framing/encoding seem to be set correctly |
| 18:24.31 | kink0 | file: done ( with 1.2.5 ) : 400209 bytes allocated 2391 units total |
| 18:24.52 | file | but which file has an insanely large amount that keeps increasing? |
| 18:26.04 | praet | Strom_C: when you ad fxo it connects to the telco line, and the fxs goes to an analog phone to answer calls right? |
| 18:26.11 | Strom_C | yes |
| 18:26.27 | kink0 | file: comparing all sizes after fews minutes running |
| 18:28.54 | tainted- | i have a call file that drops a person into a meetme(), but i cannot hear or transmit audio.. any ideas? |
| 18:29.18 | tainted- | i hear join/depart sounds, but cannot hear participants |
| 18:29.35 | file | is it behind NAT? |
| 18:29.45 | file | using SIP? what protocols... technologies... |
| 18:29.45 | tainted- | no |
| 18:29.52 | file | gotta be specific |
| 18:30.00 | tainted- | let's see |
| 18:30.17 | tainted- | initial call to meetme is sip, phone behind nat |
| 18:30.32 | tainted- | subsequent call uses local channel + agi to create call file |
| 18:30.44 | tainted- | call is placed via iax2/ |
| 18:30.55 | tainted- | and dropped into context w/ meetme() |
| 18:31.30 | tainted- | i can drop callfile user into ivr context and audio will work fine |
| 18:31.34 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 18:31.42 | tainted- | only if callfile user is dropped into meetme(), then no audio |
| 18:31.54 | tainted- | using meetme() option b |
| 18:32.45 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 18:35.25 | *** join/#asterisk mitcheloc (i=user@204.8.143.106) |
| 18:36.10 | tainted- | strange |
| 18:36.44 | tainted- | i can hear background() as well as 'conf-hasleft', but not participants' audio |
| 18:37.41 | tainted- | file |
| 18:37.46 | brad_mssw | do you have a timing device? like a zaptel card, or the ztdummy module ? |
| 18:37.51 | tainted- | yea |
| 18:38.01 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 18:38.06 | tainted- | how do i loop for user dtmf in a agi script |
| 18:38.42 | tainted- | i think the culprit could be i'm background(oneSecondSilence) with no timeout |
| 18:39.13 | tainted- | is there another way to listen for dtmf w/o playing back a file? |
| 18:39.22 | file | read or waitexten |
| 18:40.10 | tainted- | during a read, would meetme() participant audio be muted? |
| 18:40.18 | tainted- | that could be the cause of all this |
| 18:41.12 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 18:42.18 | file | tainted-: you do realize that the b option doesn't work for non-zap channels? |
| 18:43.04 | tainted- | file the agi executes |
| 18:43.33 | tainted- | file so i don't know what part of non-zap channels doesn't work.. perhaps the audio? |
| 18:43.48 | *** part/#asterisk skkip (n=skkip@216.160.91.91) |
| 18:48.05 | *** part/#asterisk mitcheloc (i=user@204.8.143.106) |
| 18:48.38 | file | tainted-: you can do two things on a zap channel at once, you can't on a regular channel unless you do weird things |
| 18:50.03 | tainted- | what things |
| 18:50.13 | tainted- | maybe i can compromise those things |
| 18:50.16 | tainted- | for audio |
| 18:50.27 | tainted- | i just want to be able to dial someone into meetme |
| 18:50.40 | file | and be able to listen using an outside script for DTMF? |
| 18:50.50 | tainted- | any method |
| 18:51.10 | tainted- | the outside script was a last resort |
| 18:51.18 | tainted- | when i do dial() from the agi, it works |
| 18:51.33 | tainted- | but i am only able to add that one person into the meetme() |
| 18:51.58 | file | you're overcomplicating things... so much that I'm totally lost on what you're trying to achieve, go down to the base of how you want it to function and pastebin a little description |
| 18:52.02 | tainted- | when i use callfile to drop the person into the meetme(), then no participant audio.. but background() and even moh() works |
| 18:52.38 | tainted- | say i am in a meetme |
| 18:52.52 | FuriousGeorge | i am in a meetme |
| 18:53.01 | tainted- | i press '9-555-555-1212' |
| 18:53.17 | tainted- | and some facility dials 555-555-1212 using whatever provider |
| 18:53.28 | file | you can't. |
| 18:53.32 | tainted- | 555-555-1212 is then patched into the meetme |
| 18:53.46 | file | well, you could - it would just require mods |
| 18:54.22 | tainted- | i have it all working right now, except when the user is patched in, he can only hear background() audio cues, join/part audio, but not participant audio |
| 18:55.06 | b00mer_ | I was told I need to turn on rpid in my sip.conf... any body have the line to add? |
| 18:55.11 | file | b00mer_: sendrpid=yes |
| 18:55.21 | b00mer_ | file: thx |
| 18:55.44 | `Sauron | Yawn. |
| 18:55.53 | file | tainted-: like I said do it up in a pastebin, including how you're achieiving this now... |
| 18:56.00 | file | and give me the link so I can follow it when I have a chance |
| 18:56.04 | `Sauron | Apparently I was both in this chan, and asterisk-unregistered. |
| 18:56.08 | `Sauron | Mewf |
| 18:56.09 | file | and I will tell you why it's doing what it's doing |
| 18:58.47 | websae | file: how's life at digium? |
| 18:59.15 | file | websae: good, busy right now with something... need to get it finished up, then I'm back at the bug tracker :D |
| 18:59.26 | file | which is where I want to be since it's out of control, but this is important |
| 18:59.48 | websae | file: well good luck, have a good day |
| 18:59.56 | file | same to you! |
| 19:00.05 | websae | file: much appreciated, thank you |
| 19:01.19 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 19:06.23 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 19:08.24 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:11.24 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 19:11.38 | vader-- | hello |
| 19:11.49 | websae | hello |
| 19:11.58 | websae | vader: how are you doing? |
| 19:12.14 | vader-- | doing ok |
| 19:12.19 | vader-- | getting my asterisk system together |
| 19:13.04 | vader-- | just got done recompiling the new kernel |
| 19:13.12 | vader-- | trying to figure out what my next step is |
| 19:13.20 | websae | ahh ok |
| 19:13.24 | *** join/#asterisk brockj49464_ (n=brockj49@41.105.dhcp.hope.edu) |
| 19:13.30 | websae | compile asterisk |
| 19:13.32 | websae | :) |
| 19:13.43 | vader-- | how about zaptel? |
| 19:13.49 | vader-- | im using digium cards |
| 19:14.35 | Strom_C | zaptel, libpri, asterisk |
| 19:15.41 | blitzrage | wouldn't libpri, zaptel, asterisk make more sense? |
| 19:15.54 | [TK]D-Fender | blitzrage : SHHH! |
| 19:16.00 | blitzrage | [TK]D-Fender: :D |
| 19:16.08 | file | asterisk, zaptel, and THEN libpri - best order vah |
| 19:16.11 | file | er evah |
| 19:16.39 | Strom_C | *shrug* I always compile zaptel first |
| 19:16.41 | Strom_C | as per asterisk.org ;) |
| 19:18.00 | acehunky | <PROTECTED> |
| 19:18.04 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 19:20.40 | b00mer_ | read some stuff indicating that asterisk as a fax to email gateway is hit/miss, but it seemed to focus on network issues. If I bring a fax in on a pri and have asterisk to the fax to email magic, would that be reliable? |
| 19:20.42 | *** join/#asterisk demigod2k (n=joey@71-13-80-162.static.bycy.mi.charter.com) |
| 19:20.54 | demigod2k | yay finally got my VS1 setup and working. garbage equipment though :( |
| 19:21.34 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 19:21.47 | demigod2k | on the second unit they sent the PCI card was floating loose, not in its socket |
| 19:24.18 | drfoomod2 | is anyone using a Cisco router with a voice T1 card? |
| 19:27.30 | SexyKen | Hey guys -- I'm implementing phone stats for my users and in the cli -- it shows each time (every 10 seconds for about 8 phones) someone logs into the manager api |
| 19:27.35 | SexyKen | How can I stop this? |
| 19:27.38 | SexyKen | From showin gin the cli |
| 19:27.55 | praet | Strom_C: do you still recommend the tdm400p over the cheap x100p if i only get one fxo module? |
| 19:28.11 | Strom_C | praet: yes |
| 19:28.16 | Strom_C | you'll have less problems with echo |
| 19:28.24 | praet | ah i see |
| 19:28.27 | Strom_C | and easy room for expansion |
| 19:28.38 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 19:28.44 | praet | the drivers are the same? |
| 19:29.09 | tekati | I am eliminating my TDM card and I have already removed my X100P cards and would like to dump the zaptel drivers is this possible now or do you need those drivers for stuff like MOH etc. I remember something about timing from the zaptel cards? |
| 19:29.23 | chiardon | Hi. if I have a dead zap channel. How can I revive it without having to restart asterisk? |
| 19:29.26 | SexyKen | Anyone?? |
| 19:29.33 | praet | i mean each nodule will be regonized separately |
| 19:29.43 | Strom_C | praet: no. x100p uses wcfxo, tdm400p use wctdm |
| 19:30.03 | praet | thanks a million Strom_C |
| 19:30.18 | Strom_C | i'll bill you for two million then ;) |
| 19:30.22 | praet | haha |
| 19:30.40 | praet | where to buy now ... |
| 19:30.47 | chiardon | If I do a Zap Destroy Channel in CLI |
| 19:31.03 | chiardon | can I create the channel again somehow? |
| 19:31.27 | tekati | praet: I have a TDM with 3 fxs cards in it I will sell you. You can buy the others from where ever. |
| 19:31.43 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 19:32.07 | tekati | Actually it has 4 fxs but one of the cards needs to get warrantied from digium I just have to send it back and they will ship out a new one. |
| 19:32.21 | tekati | modules I should say. |
| 19:32.28 | praet | tekati: im looking for fxo as well |
| 19:32.45 | tainted- | tekati only for meetme & moh |
| 19:33.01 | tekati | Right you can buy fxo modules for the tdm card from digium or other places. |
| 19:33.14 | brodiem | I have an odd problem hopefully someone could give some insight on. For some reason, (very seldomly) when someone goes to make an outbound call (from SIP IP phone), they get connected to someone calling IN to us waiting to speak with someone. The extension dialing out is definitely not a member of the incoming queue. Any thoughts? |
| 19:33.16 | tekati | tainted: Will one of the x100p cards work even if it is not in use? |
| 19:33.37 | tainted- | why are u ridding tdm entirely |
| 19:33.41 | *** join/#asterisk apardo (n=apardo@87.217.144.163) |
| 19:33.49 | praet | tekati: see pm |
| 19:35.53 | Hmmhesays | so kiss me and smile for me, tell me that you'll wait for me, hold me like you'll never let me goo |
| 19:36.05 | Hmmhesays | cause i'm leaving on a jet plane, don't know when i'll be back again, oh baby, I hate to go |
| 19:36.25 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 19:36.30 | Druken | dude... you need help :) |
| 19:36.32 | tekati | tainted: I bought a couple of WAP2-NA's that I plan to use around the house wirelessly for mobility with access points. |
| 19:36.46 | Druken | no singing on irc :) |
| 19:37.03 | Strom_C | take on me |
| 19:37.04 | Netgeeks | good song tho |
| 19:37.25 | Druken | Netgeeks: agreed |
| 19:37.54 | Netgeeks | could be worse, he could be singing puff the magic dragon from the same songwriters instead |
| 19:38.12 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 19:38.20 | asterboy | hey thats a classic! |
| 19:38.22 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:38.33 | tainted- | tekati how are those working out? |
| 19:38.44 | Druken | my kid keeps asking me if i know the muffin man :) |
| 19:38.53 | tainted- | i'm trying to find a wifiphone that's reliable |
| 19:39.03 | *** join/#asterisk dsfr_ (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 19:39.03 | brettnem | Druken: love that part from Shrek |
| 19:39.21 | asterboy | the muffin man? |
| 19:39.37 | Druken | who lives on drury lane |
| 19:39.42 | file | I know the muffin man! |
| 19:39.47 | asterboy | the muffin man! |
| 19:39.55 | *** join/#asterisk esculapio__ (i=elvyn@200.88.44.66) |
| 19:39.57 | brettnem | "your a monster! Eat me!" |
| 19:39.58 | tainted- | ~the muffin man |
| 19:40.00 | jbot | PicoBot: NO |
| 19:40.08 | file | lol |
| 19:40.26 | esculapio__ | hi file |
| 19:40.31 | file | hello |
| 19:40.34 | esculapio__ | file, hi |
| 19:40.34 | brettnem | "not my gumdrop buttons!!!" |
| 19:40.53 | asterboy | lol, that was a great scean |
| 19:41.03 | asterboy | and at the end he has a candy cane |
| 19:41.35 | tainted- | lol |
| 19:41.56 | Druken | did ya notice in shrek 2 all the people running out of the starbucks, to the one across the street? |
| 19:42.00 | tekati | tainted: I have some of those and I even have some of the grandstream stuff and they all seem to work pretty darn well now. firmware keeps getting better and Asterisk support for the hardware keeps getting better as well. I am loving them. |
| 19:42.03 | praet | hilarious |
| 19:43.01 | asterboy | hey, if I make changes to zapata.conf for the txgain=, restarting * is all that is needed to pickup any changes right? Just seems to be ignoring my changes and wondering if I have to restart the computer to pickup someting with zaptel driver restart |
| 19:43.04 | Druken | i just may have to crack out shrek now... damn you people!!! |
| 19:43.14 | acehunky | can any one point me to any hardware from digium which supports SS7 ? |
| 19:43.20 | brodiem | asterboy, just a CLI reload is all |
| 19:43.29 | asterboy | ok, that is what I thought. |
| 19:43.37 | asterboy | just making sure. |
| 19:43.40 | perlmonky | asterboy : changes to zapata require that asterisk actually rstart... |
| 19:44.03 | perlmonky | brodiem : you have to reload chan_zap.so... |
| 19:44.03 | asterboy | yes, I did a "stop now" |
| 19:44.09 | perlmonky | ok... |
| 19:44.21 | brodiem | perlmonky, reload by itself would include chan_zap.so |
| 19:44.24 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 19:45.15 | asterboy | I still like to kill the process just to be sure, but reload should do it. |
| 19:45.22 | tekati | tainted or anyone: have you tried the WIP300 or WIP330 yet? I am interested to know how those work. The company I work is currently deploying city wide wifi in Anaheim, CA and thought those would be a cool way to really test the network. |
| 19:45.45 | esculapio__ | who can help me with softphone Idefisk |
| 19:45.51 | acehunky | anyone can shed light on SS7 out here ? |
| 19:45.59 | Druken | only problem with wifi is the handoffs |
| 19:46.01 | asterboy | SS7 is only good for home networks |
| 19:46.08 | asterboy | 1 or 2 lines |
| 19:46.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:46.12 | tainted- | tekati i'm in the same boat.. trying to find a good wifi handset for vancouver, ca |
| 19:46.15 | asterboy | just joking |
| 19:46.36 | asterboy | There is some good info on SS7 on the web and in the voip-info docs |
| 19:46.44 | perlmonky | brodiem : but not all configuration options are |
| 19:46.44 | perlmonky | ; re-configured during a reload. |
| 19:47.00 | esculapio__ | transf hold etc |
| 19:47.14 | asterboy | don't use it myself, since that kind of signalling is for the CLECs iirc |
| 19:47.32 | esculapio__ | who can help me with softphone Idefisk transf, hold, .... |
| 19:47.39 | tekati | Druken: Yea I agree and I really want to see how that all works. The design team claims they have a pretty good solution to that. I would like to test that theory out. To me that would rock to live in a city with WiFi coverage. I would be willing to throw my cell out the freaking window! |
| 19:48.04 | b00mer_ | am I crazy tring to patch the latest / greatest asterisk with spandsp's application patch? |
| 19:48.13 | [hC] | Ive got a WIP300, and a network that handles handoffs properly |
| 19:48.19 | [hC] | nya nya |
| 19:48.23 | praet | my state is testing out full coverage wifi |
| 19:48.48 | Druken | tekati: willing? i'd be driving over mine.... |
| 19:48.59 | brodiem | perlmonky, I set my gains 2 days ago and CLI reload took tx/rxgain changes, because I was monitoring a 1Khz tone from ztmonitor and saw it fluctuate on reload |
| 19:49.05 | Druken | course, my wifi phone would be useless outside the city limits :) |
| 19:49.46 | [hC] | anyone here use a win32 interface (manager interface i guess?) that sits in the tray and will alert you of incoming calls, answer, transfer, etc? I checked out ADM but it seems to be incomplete. |
| 19:50.04 | acehunky | asterboy: i wanted to know if there is any card thats made by digium which supports SS7 .. pardon me if this question sounds silly .. |
| 19:50.22 | asterboy | I don't know for sure, but I'm thinking no. |
| 19:50.25 | tekati | praet: What state is that? |
| 19:50.45 | acehunky | anyone from digium over here .. who can answer that question regarding ss7 ? |
| 19:51.35 | Druken | ss7 is a signalling, not a card type |
| 19:52.10 | Druken | look it up on voip-info.org |
| 19:52.10 | brettnem | acehunky: for ss7+asterisk.. go check out ss7box.org |
| 19:52.26 | brettnem | If you don't know what ss7 is.. chances are you don't need it |
| 19:52.33 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 19:52.44 | tainted- | [hC] what network is that? |
| 19:52.46 | acehunky | brettnem: ss7box.org doesnt resolve |
| 19:52.49 | praet | tekati: Rhode island |
| 19:52.57 | tainted- | [hC] that handles the handoffs.. what protocol? |
| 19:52.58 | tekati | Very cool |
| 19:53.11 | brettnem | er.. .com |
| 19:53.16 | acehunky | ok |
| 19:53.34 | brettnem | acehunky: what do you need SS7 for? |
| 19:53.41 | [hC] | tainted-: 802.11a/b/g. There's a company i buy hardware from called meru networks that has tied AP's together using a CDMA-like timing technology, and allows for seamless handoffs to independent networks. its sweet as hell. |
| 19:54.24 | tainted- | what osi layer do they implement the handoff at? |
| 19:55.11 | tainted- | handoffs w/o affecting the 802.11x protocols has been the holy grail of mesh networks |
| 19:55.43 | tainted- | there's a lot of proprietary crap out there.. but an open implementation would be amazing |
| 19:55.56 | [hC] | as far as the client goes, you need nothing special |
| 19:56.00 | esculapio__ | help my please, I have with softphone |
| 19:56.03 | [hC] | the radios all decide on an essid and channel |
| 19:56.22 | [hC] | and by using tunneling back to their controller, they allow for seamless layer2 across all AP's regardless of how they get internet |
| 19:56.47 | esculapio__ | I have problem with softphone |
| 19:56.52 | tainted- | so this is only available by using aps from meru |
| 19:57.06 | [hC] | yes, but to the client, theres nothing special required. |
| 19:57.10 | perlmonky | brodiem : you are correct... that was added in 2005... |
| 19:57.13 | [hC] | and meru's controllers and AP's are not expensive |
| 19:57.20 | perlmonky | brodiem : I admit defeat :) |
| 19:57.28 | [hC] | more expensive than say, a wrt54g.. but by no means EXPENSIVE |
| 19:57.29 | tainted- | oh that's been done |
| 19:57.46 | brodiem | perlmonky lol |
| 19:58.03 | perlmonky | brodiem : it was added in 1.0.10... |
| 19:58.32 | brodiem | wasn't trying to argue just saying that it will work both ways :) |
| 19:58.33 | demigod2k | does anybody have a suggestion for tuning gain if the CO doesn't provide a milliwatt test number? Detroit apparantly hasn't done that in years |
| 19:58.47 | acehunky | brettnem: one of our client has asked if we have any hardware which can work on asterisk and on ss7 protocol .. i m digging more on ss7 .. |
| 19:58.53 | justinu|laptop | go as low as you can go, until the calls sound too quiet |
| 19:58.55 | brodiem | demigod2k, I just googled for other telco test lines nearby |
| 19:58.58 | acehunky | www.openss7.org .. this project sounds to be dead.. |
| 19:59.14 | perlmonky | brodiem : I swear I remember having to actually restart asterisk to get htat working... which makes sense... it was 1.0.8 and 9 that I started with... I never changed... |
| 19:59.26 | docelm0 | ARGH! |
| 19:59.42 | demigod2k | brodiem, didn't find anything for Detroit... how far away can I go before the test really becomes invalid? can I use anything in the USA? |
| 20:00.55 | brodiem | demigod2k I'd be willing to bet it wouldn't throw you off much, but that's just my opinion not based on any facts :) |
| 20:01.32 | demigod2k | it's probably going to be a lossless digital transport, so my guess is the same. |
| 20:01.48 | brodiem | demigod2k, they're digital trunks separating telcos anyway |
| 20:01.55 | brodiem | yeah |
| 20:02.53 | demigod2k | any pointers where you found it? I keep running across the same stupid FAQs rather than any valid numbers. |
| 20:03.09 | demigod2k | I could probably try the "try these" numbers on random area codes and exchanges too... |
| 20:03.09 | brodiem | demigod2k, I found one in San Antonio i can give you |
| 20:03.13 | demigod2k | that would be awesome thanks |
| 20:03.38 | brodiem | 210-222-9999 |
| 20:03.41 | demigod2k | I found a bunch on the mailing lists for europe, australia, and other hard to find places |
| 20:03.52 | drfoomod2 | i was asking about SS7 the other day |
| 20:04.18 | drfoomod2 | is it beneficial for the telecom provider to use SS7 down to a customer than PRI? |
| 20:04.27 | drfoomod2 | rather than PRI |
| 20:05.48 | demigod2k | brodiem, thanks! |
| 20:06.02 | acehunky | i m actually clueless on wats SS7 .. just know that its some kind of signalling protocol ! drfoomod2 |
| 20:07.09 | acehunky | openss7.org is driving me nuts |
| 20:07.50 | acehunky | although i find some info on voip-info wiki |
| 20:08.18 | `Kevin | i do not have the option of zap debug in cli, others have used it.. i have compiled zaptel 3 times and asterisk right after |
| 20:08.49 | drfoomod2 | acehunky: http://en.wikipedia.org/wiki/SS7 |
| 20:09.15 | acehunky | drfoomod2 i m there :) |
| 20:12.08 | *** part/#asterisk esculapio__ (i=elvyn@200.88.44.66) |
| 20:12.11 | rpm | anyone here use astmanproxy and have it outputting in xml or csv? |
| 20:13.07 | jbalcomb | Is there anyway to set the number of rings on the GXP-2000? |
| 20:13.09 | brodiem | demigod2k np |
| 20:19.38 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 20:20.06 | drfoomod2 | acehunky: http://www.intel.com/network/csp/solutions/ss7/7194ovr.htm |
| 20:21.40 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:28.46 | Dr-Linux | hi |
| 20:28.48 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 20:30.37 | docelm0 | jbalcomb what do you mean number of rings? before what? |
| 20:30.53 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-232-230.telkomadsl.co.za) |
| 20:31.54 | jbalcomb | docelm0 before it stops. so maybe just two rings instead of 4. i'm thinking that based on my dialplan 'wait for answer' setting? |
| 20:33.10 | hellop | I just got 3 of those 2nd gen Ambient 100p cards. 2 were DOA |
| 20:33.12 | docelm0 | exten => 123,1,Dial(SIP/GXP2000,timeout) where time out is seconds before it stops ringing.. 23 sec is almost 4 rings |
| 20:33.19 | docelm0 | so half that.. round 11 or so |
| 20:33.39 | docelm0 | hellop and? |
| 20:33.40 | hellop | So, out of 6 cards, the last one worked. I knew I wasn't crazy!!!!! |
| 20:34.13 | hellop | docelm0, ohh just thought I'd share. They were advertised as "Digium Asterisk Genuine OEM" |
| 20:34.41 | docelm0 | The only thing genuine is something w/ the digium name or asterisk logo.. |
| 20:34.46 | docelm0 | dont accept imitations.. |
| 20:34.54 | docelm0 | even if they are cheaper |
| 20:35.02 | hellop | are they still 100 bucks? |
| 20:35.23 | docelm0 | What? |
| 20:35.43 | docelm0 | x100p? I dont even think they are made anymore |
| 20:35.49 | docelm0 | try TDM400p |
| 20:35.49 | hellop | Digium 100p cards. I saw something on ebay for $35 but it was probably a scam.. |
| 20:36.07 | docelm0 | You will find clones but thats bout it. |
| 20:36.09 | hellop | docelm0, what do you use if you just need 1 line? |
| 20:36.21 | hellop | docelm0, i.e. the home user.. |
| 20:36.22 | docelm0 | TDM401P |
| 20:36.31 | docelm0 | FXO or FXS? |
| 20:37.04 | demigod2k | they have those ethernet-based FXOs for around $100 too that seem reasonable for a home |
| 20:37.23 | docelm0 | if FXS then the one listed above is the model number |
| 20:37.27 | docelm0 | TDM401P |
| 20:37.29 | Nugget | if I had to do it over again, I'd get a sipura spa instead of the tdm400p. |
| 20:37.35 | docelm0 | I have a TDM421P |
| 20:37.51 | docelm0 | Nugget Im looking for another.. :) Wanna sell yours? |
| 20:37.52 | demigod2k | nugget, why is that |
| 20:38.06 | Dr-Linux | huh |
| 20:38.10 | hellop | docelm0, So those 400 cards can support up 2 4 modules, you're saying just buy one with one module? |
| 20:38.20 | docelm0 | hellop yes |
| 20:38.32 | Nugget | the sipura wouldn't require zaptel, which I've found to be the least reliable and most cumbersome aspect of asterisk. |
| 20:38.40 | hellop | yeah, I saw that buy it now for about $135 |
| 20:38.41 | Dr-Linux | tdm400p has a noice for me while making calls? doesn't any one encounterd same issue? |
| 20:39.35 | Dr-Linux | s/noice/noise |
| 20:39.45 | demigod2k | nugget, a reasonable explanation. hows echo cancellation with something like the sipura? that's the main worry that kept me away |
| 20:40.58 | Nugget | I can't really speak to that concern |
| 20:41.29 | Dr-Linux | demigod2k: you can't do much if you are still getting echo |
| 20:41.30 | Nugget | I have one of each and have had no echo problems with either, but clearly some people do. |
| 20:41.34 | demigod2k | ya. I may rethink as soon as we expand beyond 4 lines, I really liked the look of sipura and others |
| 20:42.08 | demigod2k | I just didnt want to deviate from the mainstream for my first effort. Bought a ready-to-go system with support |
| 20:42.28 | Dr-Linux | tdm400p incoming is crystal clear, but outgoing is echo/noise and blah blah problems |
| 20:42.47 | *** join/#asterisk clive- (n=pirch@dsl-145-18-73.telkomadsl.co.za) |
| 20:43.28 | demigod2k | ya I'm fighting my tdm eco as we speak :( |
| 20:44.16 | Dr-Linux | demigod2k: you are getting echo with incoming or outgoing? |
| 20:44.39 | demigod2k | havent tried incoming yet. outgoing to cellular and landline I hear echo on my polycom, the other end sounds fine |
| 20:45.22 | demigod2k | as a separate issue, even on the local net I get some echo but I'll address that later. the outgoing call has like 1.5 seconds echo (horrible) |
| 20:45.51 | Dr-Linux | demigod2k: what's your rx/tx gain? |
| 20:46.59 | froguz | demigod2k, you should try fxotune, it will probably fix your problem |
| 20:47.18 | demigod2k | Dr-Linux, going through that howto right now. 7.5 and climbing |
| 20:47.19 | Dr-Linux | froguz: what's fxotune? |
| 20:48.11 | Dr-Linux | demigod2k: i had the same problem, but i decrease rx gain, and it's fine now, but still noise there |
| 20:48.27 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
| 20:49.24 | demigod2k | echo noise or just line noise? |
| 20:49.44 | Dr-Linux | just line noise |
| 20:50.19 | clive- | does nayone know how to use "set variable" in an AGI script ? |
| 20:50.19 | Dr-Linux | i have different tdm400p some in USA some in Pakistan, but same problem with all of them |
| 20:50.40 | demigod2k | that fxotune is news to me, I may check that one out too. |
| 20:51.47 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 20:52.07 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 20:52.17 | qseek | good afternoon everyone |
| 20:53.09 | Dr-Linux | demigod2k: where we can check that? |
| 20:53.15 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 20:53.28 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 20:54.04 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:54.08 | *** join/#asterisk hellop1 (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 20:54.09 | demigod2k | Dr-Linux, I'm just reading about it on voip-info.org. I want to figure out what it does before I start running random commands :) |
| 20:54.23 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 20:54.54 | hellop1 | I have a problem with * that I cannot resolve. |
| 20:55.33 | Dr-Linux | demigod2k: give me the link? |
| 20:55.36 | hellop1 | I have a Mini-ITX MB, working fine, then one by one 3 generic x100p cards died. I installed * to another box, but could not get the cards to work. |
| 20:56.04 | hellop1 | So I ordered 3 new cards. On the new box, 2 cards give: ZT_CHANCONFIG failed on channel 1 |
| 20:56.27 | hellop1 | On the old box, 1 new card gives: FXO PCI Master Abort |
| 20:56.41 | [TK]D-Fender | hellop : pastebin your "cat /proc/interrupts", then your zaptel.conf and zapata.conf |
| 20:56.42 | [TK]D-Fender | ~pb |
| 20:56.43 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:56.57 | hellop1 | I have 6 cards, and 2 asterisk boxes... any suggestions? |
| 20:57.21 | hellop1 | [TK]D-Fender, k |
| 20:58.36 | *** join/#asterisk PBXtech (n=nik@70.89.247.188) |
| 20:59.34 | *** join/#asterisk lzhang (n=rjrae@adsl-69-152-225-92.dsl.snantx.swbell.net) |
| 20:59.51 | lzhang | I just tried upgrading to 1.2.7.1 from 1.2.5 and now sip peers aren't seeding |
| 21:00.05 | lzhang | not sure what I need to do to get them to register |
| 21:01.11 | qseek | lzhang i had a similar issue with my sip phones, but a reboot of the phones solved this issue |
| 21:01.28 | Dr-Linux | hellop1: what it says "/sbin/ztcfg -vvv" ? |
| 21:02.07 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 21:02.16 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 21:03.08 | hellop1 | Dr-Linux, : http://pastebin.ca/51006 |
| 21:03.24 | PBXtech | anyone know why i keep getting this error.. 2.547 Seconds. |
| 21:03.24 | PBXtech | (%) Total: 328 Opera |
| 21:03.26 | hellop1 | Dr-Linux, I'm scared to swap cards between boxes now |
| 21:03.30 | PBXtech | oops |
| 21:03.35 | PBXtech | WARNING[21587]: file.c:1032 ast_waitstream: Unexpected control subclass '-1' |
| 21:04.56 | clive- | PBXtech I also get tons of those warnings, no idea about what and why |
| 21:05.22 | [TK]D-Fender | hellop1 : You have no channel declaraion in your zapata.conf..... |
| 21:05.34 | PBXtech | hmm |
| 21:07.39 | clive- | anyone familiar with perl AGI's ? I am struggling with set_variable |
| 21:07.40 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:15.10 | *** join/#asterisk mooey (n=chris@85-210-5-108.dsl.pipex.com) |
| 21:15.38 | mooey | is it possible to install asterisk in a custom directory? im looking for the equivalent of ./configure --prefix=/home/user/asterisk |
| 21:16.40 | Dr-Linux | mooey: reason? |
| 21:16.40 | mooey | i dont have root on this machine |
| 21:16.52 | mog_work | we have configure scripts now mooey should work i think |
| 21:17.13 | mooey | in svn? i downloaded 1.2.7.1 and it didn't :( |
| 21:17.29 | mog_work | in trunk |
| 21:17.32 | mog_work | it will be in 1.4 |
| 21:17.34 | mog_work | not in 1.3 |
| 21:17.37 | mog_work | err 1.2 |
| 21:17.59 | mooey | is it likely to break? :} this needs to be a stable-ish install |
| 21:20.02 | mog_work | heh |
| 21:20.05 | mog_work | its trunk |
| 21:20.17 | russellb | you can do it in 1.2 |
| 21:20.18 | mog_work | but its not for all |
| 21:20.22 | mog_work | you can |
| 21:20.24 | russellb | edit the Makefile, and change the INSTALL_PREFIX |
| 21:20.31 | mog_work | but there isnt ./configure sexyness |
| 21:20.34 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 21:20.52 | hellop | sorry if anyone tried to help, my client locked up |
| 21:21.45 | hellop | I need help and am willing to pay up to $25. http://pastebin.ca/51012 |
| 21:22.07 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:22.25 | mooey | russellb: thats fab, thanks |
| 21:22.36 | mooey | worked a charm, thanks anyway mog_work :} |
| 21:22.48 | mog_work | awesome |
| 21:23.58 | tainted- | file did u get a chance to look at my pastebin |
| 21:24.38 | file | no, this project gets priority... |
| 21:24.40 | file | deadlines you know |
| 21:25.07 | tainted- | np |
| 21:25.17 | [TK]D-Fender | hellop : First you don't have your WCFXO in your interrupts list, and you don't have a channel declaration in your zapata.conf. |
| 21:28.06 | hellop | [TK]D-Fender, you mean channel => 1 it didn't make it to pastebin but it's there. |
| 21:28.34 | *** join/#asterisk saftsack (n=saftsack@p54A7D9A5.dip.t-dialin.net) |
| 21:28.35 | saftsack | h |
| 21:28.36 | saftsack | i |
| 21:28.47 | saftsack | is it good to use the congestion() application? |
| 21:29.04 | hellop | [TK]D-Fender, for WCFXO in proc, does that happen after ztcfg -vv? (which is broken) or does that mean I have a hardware prob? |
| 21:29.06 | jake1932 | saftsack - for what? |
| 21:29.26 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
| 21:29.31 | Hmmhesays | anyone in here use any voip-->gsm gateways? |
| 21:29.49 | saftsack | jake1932, for signalling congestion if all telephones are in use |
| 21:30.15 | Dr-Linux | saftsack: depends |
| 21:30.29 | saftsack | also i have bristuff isdn here |
| 21:30.30 | jake1932 | saftsack - can't see why not. - you could also play a message "all telephones are in use" |
| 21:30.51 | jake1932 | oh - for ISDN |
| 21:31.01 | saftsack | yes but the sense of congestion is, that asterisk doesnt answer the call |
| 21:31.08 | saftsack | so there are no costs for the caller |
| 21:31.40 | *** join/#asterisk IceManRISK (n=kart@201-40-88-106.mganm702.dsl.brasiltelecom.net.br) |
| 21:31.41 | jake1932 | but the call never goes through - isn't there a forward if both channels are in use? |
| 21:31.53 | *** join/#asterisk St1ckm4n (n=shortes9@68.178.74.166) |
| 21:32.13 | saftsack | yes this is maybe possible |
| 21:32.19 | saftsack | but for example we have one telephone here |
| 21:32.26 | *** part/#asterisk St1ckm4n (n=shortes9@68.178.74.166) |
| 21:32.26 | saftsack | and the most calls are dedicated to this telephone |
| 21:32.34 | brif8 | Is there an Open source windows client similar to FOP that just handles if an extension is busy or idle ? |
| 21:32.41 | saftsack | if this telephone is congested the number should be show congestion |
| 21:32.53 | saftsack | independend if there is a free b canal |
| 21:33.14 | hellop | All I can think to do it buy some more cards and a new MB. |
| 21:33.26 | mooey | can you specify the path that asterisk uses for configs when starting asterisk? |
| 21:33.44 | *** join/#asterisk AndyCap_ (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 21:34.02 | jake1932 | saftsack - by congested - you don't mean both b channels are used? |
| 21:34.14 | hads|home | mooey: look at asterisk.conf |
| 21:34.15 | brodiem | I have an odd problem hopefully someone could give some insight on. For some reason, (very seldomly) when someone goes to make an outbound call (from SIP IP phone), they get connected to someone calling IN to us waiting to speak with someone. The extension dialing out is definitely not a member of the incoming queue. Any thoughts? |
| 21:34.22 | froguz | mooey read asterisk.conf |
| 21:34.25 | saftsack | jake1932, no |
| 21:34.28 | hellop | $30 to whoever can get me running. I have 6 X100p cards and 2 Asterisk boxes. |
| 21:34.32 | mooey | :$ sorted now |
| 21:34.36 | Strom_C | brodiem: sounds like a classic glare problem to me |
| 21:34.39 | Mike | guys when using sip how can i return code 34? right now im returning code -1 |
| 21:34.46 | Strom_C | brodiem: what kind of trunking are you using to the outside world? |
| 21:35.11 | brodiem | Strom_C, It's a channelized T1 into a TE210P |
| 21:35.16 | hellop | brodiem, that happens on normal phones |
| 21:35.23 | jake1932 | saftsack - how are you defining congestion then? |
| 21:35.32 | jake1932 | 1 b channel in use? |
| 21:35.56 | Strom_C | brodiem: and you're hunting from 24 -> 1 on outbound and 1 -> 24 on inbound |
| 21:35.57 | Strom_C | right? |
| 21:36.02 | saftsack | also if there is an incoming call e.g. for the number 123456 |
| 21:36.14 | saftsack | 123456,1,Dial(internal telephone) |
| 21:36.20 | saftsack | 123456,102,Congestion() |
| 21:36.23 | saftsack | Jacke, this way |
| 21:37.02 | brodiem | Strom_C, there's 14 channels, but channel 1 is the primary number, so at the telco it starts at channel 1, and I would assume it's the same pattern as dialing out |
| 21:37.11 | jake1932 | saftsack - i don't think congestion will do it. I'd have to review the bristuff docs |
| 21:37.18 | brodiem | Strom_C, do they need to start at opposite ends? |
| 21:37.36 | saftsack | Jacke, mister junghanns told me to use playtones but in this case the caller has to pay |
| 21:37.38 | saftsack | and this sux |
| 21:37.54 | [TK]D-Fender | hellop : You need to modprobe for zaptel and wcfxo |
| 21:37.56 | Strom_C | brodiem: if you're on a channelized T1 I would STRONGLY RECOMMEND that your outbound calls hunt from the other end of the T1 |
| 21:38.00 | jake1932 | saftsack - right. and otherwise, it'll keep ringing |
| 21:38.13 | hellop | [TK]D-Fender, I do and receive no output... |
| 21:38.27 | [TK]D-Fender | hellop : PM |
| 21:38.31 | hellop | k |
| 21:38.45 | saftsack | jake1932, on normal tk it is handled this way: |
| 21:38.46 | brodiem | Strom_C, Hmm how would I define that? Can I still just use a single group for all 14 channels? |
| 21:38.55 | brodiem | channel => 14-1 or something? |
| 21:39.00 | saftsack | you call and then there is about 1 second nothing |
| 21:39.09 | saftsack | and then the official telecom congestion sound |
| 21:39.18 | saftsack | generated by telekom and not by the foreign tk |
| 21:39.21 | jake1932 | right |
| 21:39.36 | Strom_C | brodiem: no no...in your dial statement, if you use Dial(ZAP/G1/whatever) instead of Dial(ZAP/g1/whatever) it should start at 14 and work backwards |
| 21:39.42 | saftsack | yes and i want asterisk to do so too |
| 21:40.30 | saftsack | jake1932, do you have an idea? |
| 21:40.31 | brodiem | Strom_C, do you know of any docs that relate to this problem? just for my own peace of mind :) |
| 21:40.45 | Strom_C | brodiem: look up "glare" |
| 21:41.18 | jake1932 | <PROTECTED> |
| 21:41.38 | brodiem | Strom_C so I'm basically just lessening the chance of this happening, since if the channels meet in the middle it could still happen just as easily? |
| 21:42.09 | Strom_C | yes...that's a problem you're going to run into when using a channelized T1 |
| 21:42.14 | sevard | I'm thinking about putting an * box in liberia so I don't have to pay 0.29/minute. If I just want one line can I use a 56k voice modem? .. or do I have to go with a TDM100P |
| 21:42.35 | Strom_C | sevard: tdm400p |
| 21:42.41 | saftsack | jake1932, yes he writes the drivers |
| 21:42.45 | jake1932 | with one FXO |
| 21:42.46 | saftsack | i wrote to the official support |
| 21:42.49 | Strom_C | you should know better :) |
| 21:43.04 | sevard | Strom_C: a TDM400P is four fxs lines... This is going to be in liberia where it may or may not be bomed through civil war |
| 21:43.20 | Strom_C | sevard: um, TDM400P can be configured with a single FXO port |
| 21:43.20 | sevard | I'm thinking a 200mhz linux box in a hard case with a voice modem :P |
| 21:43.26 | brodiem | Strom_C, thanks for the help. I suppose this wouldn't be an issue on a PRI? |
| 21:43.32 | Strom_C | voice modems blow |
| 21:43.34 | sevard | Strom_C: main point being they're expensive. |
| 21:43.47 | Strom_C | brodiem: much less of an issue on PRI, yes |
| 21:43.52 | zaf | so do bombs |
| 21:44.13 | Strom_C | sevard: so if it's too expensive, don't do it |
| 21:44.13 | LostFrog | sevard: Then get a x100 and don't come back complaining of noise, echo, and caller-id problems. |
| 21:44.34 | sevard | :/ |
| 21:44.54 | jake1932 | is that card even supported today? |
| 21:45.09 | zaf | sevard: i've got 2 x100s in a * box working fine, ymmv |
| 21:45.16 | jake1932 | you could also do a Sipura FXO |
| 21:45.44 | [TK]D-Fender | zaf : I'm feeling slow today... what is "ymmv" again? |
| 21:46.08 | sevard | jake1932: hey, i never even thought of that. A sipura with a fxo port registered back in the states with my * box |
| 21:46.19 | brodiem | Strom_C maybe I'll just be lazy and ask the telco to reverse the hunt group lol |
| 21:46.26 | Strom_C | brodiem: um no |
| 21:46.30 | Strom_C | easier to do it yourself |
| 21:46.36 | sevard | jake1932: that's a much better idea. more compact, cheaper, rawrsome. |
| 21:46.38 | Strom_C | just change one character in your dial statment |
| 21:46.55 | Strom_C | sevard: and then you have to deal with IP transport from liberia |
| 21:47.10 | sevard | Strom_C: I'd have to deal with that anyway, wouldn't I? |
| 21:47.24 | jake1932 | i think you would |
| 21:47.41 | brodiem | Strom_C, I thought you said I couldn't use a group since it would go from first-to-last zap number? Wouldn't I need a string of 14 dial statements? |
| 21:47.51 | Strom_C | well if you're going to do that, why even bother with an asterisk box in the first place? just have SIP clients there and keep the asterisk box here |
| 21:47.56 | Strom_C | brodiem: no, you didnt listen to me |
| 21:47.58 | Strom_C | ill repeat |
| 21:48.13 | Strom_C | brodiem: no no...in your dial statement, if you use Dial(ZAP/G1/whatever) instead of Dial(ZAP/g1/whatever) it should start at 14 and work backwards |
| 21:48.27 | *** join/#asterisk Samoied (n=Samoied@201.2.229.138) |
| 21:48.31 | brodiem | ahh the caps |
| 21:48.53 | brodiem | I thought you were saying instead of using Dial(blah to start at 14, as in Zap/14 and work backwards down to 1 |
| 21:49.09 | Strom_C | thats what I am saying |
| 21:49.24 | Strom_C | g1 hunts 1-> 14, G1 hunts 14 -> 1 |
| 21:49.29 | brodiem | right |
| 21:49.30 | brodiem | lol |
| 21:49.32 | brodiem | this is going nowhere |
| 21:49.37 | brodiem | I understand though..thanks. |
| 21:51.00 | sevard | Strom_C: That's what i'm talking about. an ATA. that's the best idea yet. they probably don't have computers ;/ maybe my brother in law knows somebody with a cyber cafe and a landline i can use |
| 21:51.26 | Strom_C | sevard: I have no idea what you're going on about |
| 21:52.43 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 21:53.55 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 21:54.00 | *** join/#asterisk trelane_ (i=trelane@pdpc/supporter/sustaining/trelane) |
| 21:54.49 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 21:54.59 | dippo | is there a way to drop a sip session from the asterisk console? |
| 21:55.06 | dippo | i.e. something that shows up in "sip show channels" |
| 21:55.08 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 21:55.18 | Beirdo | I see JerJer is wisely hiding right now :) |
| 21:55.27 | justinu | dippo: soft hangup |
| 21:55.30 | LostFrog | soft hangup? |
| 21:55.58 | dippo | aha |
| 21:55.59 | dippo | thanks |
| 21:56.08 | jake1932 | not hiding - just overloaded with calls probably |
| 21:56.17 | Beirdo | not surprised |
| 21:56.22 | Beirdo | switch-1 is unreachable |
| 21:56.26 | Beirdo | so I can't call out |
| 21:56.32 | Beirdo | and the DIDs... welll... |
| 21:56.36 | jake1932 | haha |
| 21:56.42 | dippo | i have this one grandstream phone that keeps getting wedged |
| 21:56.44 | jake1932 | what DIDs? |
| 21:56.51 | dippo | I still see the SIP session in "sip show channels" even though the handset is hung up |
| 21:56.55 | Beirdo | precisely |
| 21:57.00 | zaf | [TK]D-Fender: your mileage may vary |
| 21:58.02 | jake1932 | i'm just hoping asterlink negoties better agreements |
| 21:58.06 | [TK]D-Fender | zaf : thanks. |
| 21:58.10 | jake1932 | negotiates |
| 22:01.03 | *** join/#asterisk dlynes (i=1000@S010600c09f9a0fc4.vc.shawcable.net) |
| 22:06.32 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 22:06.50 | DarKnesS_WolF | how to make the 1st thing is phone rings for 20 sec only ? |
| 22:07.27 | [TK]D-Fender | DarKnesS_WolF : I think you need to read up on the Dial command.... |
| 22:07.35 | linlin | anyone willing to make a test call for me? :p |
| 22:07.39 | *** join/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 22:07.48 | digime | anyone know how i can get a free london DID # |
| 22:07.50 | linlin | trying to see if my asterisk comes out as choppy to people calling it |
| 22:07.55 | DarKnesS_WolF | [TK]D-Fender: i need to know what is the application it's ring ? or rining(20) 9$? |
| 22:08.00 | Dr-Linux | DarKnesS_WolF: put 20 sec wait at the end of Dial command |
| 22:08.12 | digime | i know there was a service that offered free U.K. DID |
| 22:08.26 | DarKnesS_WolF | Dr-Linux: wait(20) will make it rings ? |
| 22:08.43 | Dr-Linux | DarKnesS_WolF: no |
| 22:08.57 | Dr-Linux | its Dial application |
| 22:09.10 | linlin | digime that could help me too... dont remember it though? :p |
| 22:09.12 | DarKnesS_WolF | Dr-Linux: i want the phone to ring 20 sec before the voicemail take the call |
| 22:09.26 | Dr-Linux | exten => 222,1,Dial(SIP/2232,20) |
| 22:10.00 | Dr-Linux | so it will ring for 20 seconds, 4 bells |
| 22:10.01 | digime | linlin: yeah i know it exists, i found it once before |
| 22:11.18 | DarKnesS_WolF | thx |
| 22:12.38 | praet | what are the downfalls of asterisk@home? |
| 22:13.34 | Dr-Linux | praet: depends on your need |
| 22:14.01 | praet | its a new intall on a p3 450 and i wanted to try it. but i could do it all custom too |
| 22:14.22 | Strom_C | praet: the downfalls of asterisk@home are: |
| 22:14.24 | Strom_C | - everything |
| 22:14.34 | praet | k what os then |
| 22:14.52 | praet | is centos not recommended i mean as a base |
| 22:14.52 | Strom_C | whatever OS you like best combined with asterisk 1.2.7.1 |
| 22:15.09 | justinu | centos is fine |
| 22:15.19 | praet | ok i thought there were os based limitations |
| 22:15.24 | Strom_C | nope |
| 22:15.28 | Strom_C | pretty much any linux |
| 22:15.35 | Strom_C | personally, I like debian |
| 22:15.52 | mindwarp | is it pretty well supported / stable on freebsd? |
| 22:15.59 | mindwarp | anybody know? |
| 22:16.06 | praet | well im comfortable on fedora bu ti know its a bit edge |
| 22:16.30 | Strom_C | then use fedora |
| 22:17.39 | Dr-Linux | i like RHEL |
| 22:17.55 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:18.01 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 22:18.14 | mooey | "chan_iax2.c:2836 auto_congest: Auto-congesting call due to slow response" -> what is the name of the setting i can use to increase the time that asterisk waits before giving up on an outbound iax call? |
| 22:19.04 | dlynes | Strom_C: So if any Linux is good, what makes Carrier Grade Linuxes better? |
| 22:19.53 | Strom_C | dlynes: if you're going for carrier-grade stuff, that's a different ballgame entirely |
| 22:20.33 | dlynes | Strom_C: I'm guessing you don't consider a simple softswitch to be a need for carrier grade then? |
| 22:20.48 | Strom_C | depends on the application |
| 22:21.08 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-93-75-171.dsl.hstntx.swbell.net) |
| 22:21.12 | dlynes | Do they have kernel tweaks then? |
| 22:21.25 | dlynes | i.e. that aren't part of the standard kernel source? |
| 22:21.29 | Strom_C | *shrug* |
| 22:21.42 | Strom_C | I have little experience with carrier-grade setups |
| 22:21.44 | Dr-Linux | what's shrug? |
| 22:21.45 | dlynes | ah |
| 22:21.47 | Dr-Linux | justinu? |
| 22:21.52 | Strom_C | dlynes: |
| 22:21.54 | Strom_C | not justinu |
| 22:22.59 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com) |
| 22:23.16 | Dr-Linux | Strom_C: justin is my friend, and he knows about my bad english, so sometime i ask words meaning from him |
| 22:23.26 | Strom_C | oh |
| 22:23.34 | Qwell | ~dict shrug |
| 22:23.37 | Strom_C | I thought you were asking if Iwas responding to him |
| 22:24.02 | Strom_C | hey qwell :) |
| 22:24.12 | Qwell | doing so right now |
| 22:24.17 | Strom_C | I'm having an ass of a time trying to get SVN Trunk to compile |
| 22:24.52 | dlynes | Strom_C: asterisk? |
| 22:24.57 | Strom_C | yes |
| 22:24.59 | *** join/#asterisk saftsack (n=saftsack@p54A7D9A5.dip.t-dialin.net) |
| 22:25.10 | Strom_C | I want to play with Qwell's SCCP stuff |
| 22:25.11 | dlynes | Strom_C: hrm...i just grabbed it yesterday or the day before and it compiled clean |
| 22:25.30 | Strom_C | dlynes: this is a fresh install of SVN Trunk |
| 22:25.40 | Strom_C | grabbed now |
| 22:25.55 | dlynes | yeah...i haven't grabbed today's |
| 22:26.04 | Strom_C | the problem is with AEL somewhere |
| 22:26.09 | tainted- | hey dlynes |
| 22:26.15 | dlynes | heya tainted |
| 22:26.18 | tainted- | remember that meetme app we were talking about |
| 22:26.21 | Qwell | Strom_C: got the latest bison and such? |
| 22:26.28 | tainted- | i implemented it, but there's one bug |
| 22:26.29 | Strom_C | Qwell: I believe so |
| 22:26.33 | dlynes | tainted-: yeah? |
| 22:26.43 | dlynes | tainted-: so you're using call files then? |
| 22:26.47 | tainted- | dlynes once the user is dropped into meetme() via the callfile, there is no audio |
| 22:26.51 | Strom_C | Qwell: I'm running debian testing...if that doesnt work I'll try upgrading to unstable |
| 22:27.02 | Qwell | So, hopefully I won't fuck up my firmware file again this time... :p |
| 22:27.09 | Strom_C | :P |
| 22:27.11 | dlynes | tainted-: it's probably a firewall issue |
| 22:27.17 | Qwell | Strom_C: I forgot to add the tar filename, so I tarred up into the firmware file...woops! |
| 22:27.17 | tainted- | dlynes yea.. the user can hear join/depart and even background() files, but not the audio of other participants |
| 22:27.25 | Strom_C | oops |
| 22:27.30 | tainted- | dlynes all public ips |
| 22:27.40 | dlynes | tainted-: oh...maybe a mismatch on codecs? |
| 22:27.49 | tainted- | 729 end to end |
| 22:28.04 | dlynes | tainted-: or maybe the other guys are using g729 and the new participant isn't, and you've run out of available g729 licenses? |
| 22:28.12 | tainted- | nope |
| 22:28.16 | tainted- | got plenty to spare |
| 22:28.37 | dlynes | Well, I'm glad to hear that you made it this far :) |
| 22:28.40 | tainted- | if it was codec issue, i don't think i'd be able to hear the conf join/departs |
| 22:28.51 | tainted- | yea i think it'd be pretty useful for others |
| 22:28.52 | dlynes | tainted-: Those would probably be ulaw->g729 |
| 22:29.14 | dlynes | tainted-: or ulaw->ulaw if the phone is using ulaw |
| 22:29.24 | Qwell | Strom_C: sent |
| 22:29.27 | dlynes | tainted-: what's the joining party's codec preference? |
| 22:29.45 | Strom_C | Qwell: awesome |
| 22:29.48 | tainted- | meetme admin ip phone is 729, joining party is pstn through iax2 provider using 729 |
| 22:29.52 | dlynes | tainted-: and have you done a sip peer debug to make sure everything's copascetic? |
| 22:30.07 | dlynes | erm iax2 debug? |
| 22:30.19 | tainted- | yea |
| 22:30.23 | Strom_C | Qwell: hopefully SVN trunk will compile on debian unstable now :) |
| 22:30.27 | tainted- | i can drop the user into an ivr no problem |
| 22:30.42 | tainted- | but once it's in meetme, it only hears join/depart |
| 22:30.51 | dlynes | tainted-: Maybe there's a problem on the other end where it's converting pstn to g729 before it gets to your box? |
| 22:30.52 | tainted- | dlynes pvt msg your did |
| 22:30.54 | tainted- | i'll show u |
| 22:31.17 | dlynes | tainted-: can't from here...crappy wireless router sucks for voip |
| 22:31.41 | tainted- | it should be 729 by the time it gets to my box |
| 22:31.55 | tainted- | let me check |
| 22:32.52 | dlynes | tainted-: Well, fwiw, I'll be updating the wiki over the next couple of days with some info for setting up asterisk like a keysystem for peeps that have phones with separate logins for each line appearance and BLF support |
| 22:33.27 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 22:33.33 | dlynes | tainted-: It'll be using Aastra 9133i's for the example |
| 22:33.57 | ctooley | Can someone give me a ballpark for a Media Gateway for SIP that can take a DS3 |
| 22:33.58 | ctooley | ? |
| 22:34.37 | tainted- | dlynes that's pretty cool |
| 22:36.05 | dlynes | Yeah...everyone kept telling me it couldn't be done |
| 22:36.10 | dlynes | I've got it deployed already |
| 22:36.26 | dlynes | With a two line system, intercom line, and six extensions |
| 22:36.42 | tainted- | hmm |
| 22:36.52 | tainted- | iax2 debug looks good |
| 22:37.11 | dlynes | You can set it up for up two seven extensions that way on an Aastra 9133i |
| 22:37.17 | dlynes | erm s/to/two |
| 22:37.22 | dlynes | erm s/two/to |
| 22:37.49 | dlynes | tainted-: nothing in your full logs about any errors? |
| 22:37.55 | dlynes | tainted-: or warnings? |
| 22:38.06 | tainted- | nothing |
| 22:38.12 | dlynes | tainted-: try setting up one log file to log everything |
| 22:38.18 | dlynes | tainted-: and then set verbose to 1000 or something |
| 22:38.23 | dlynes | tainted-: and then try it again |
| 22:39.34 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 22:39.51 | tainted- | nothing |
| 22:39.56 | tainted- | no errors, no warnings |
| 22:40.11 | tainted- | just acks, pings, pongs, hangups.. regular stuff |
| 22:40.53 | tainted- | a notice when call file is placed and when call file user hangs up |
| 22:41.33 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 22:41.38 | tainted- | if i execute a dail() from within the agi, then it works, but i'm locked to that one user... i cannot add additional users |
| 22:42.22 | dlynes | hahaha....that's hilarious |
| 22:42.31 | dlynes | Oracle just plugged 36 holes in their database :P |
| 22:42.50 | tainted- | so much for unbreakable |
| 22:43.12 | [TK]D-Fender | dlynes : 9133's? yuck.... |
| 22:43.16 | dlynes | Who said Oracle was unbreakable? |
| 22:43.21 | dlynes | [TK]D-Fender: What's wrong with 9133i's? |
| 22:43.27 | key2 | how would I have to say in my sip.conf that I want my asterisk to connect itself to a SIP server so I could make the calls go trough ? |
| 22:43.36 | [TK]D-Fender | dlynes : ugly, and not really worth their cost... |
| 22:43.39 | tainted- | dlynes it was a campagin a while back |
| 22:43.46 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 22:43.48 | [TK]D-Fender | dlynes : in terms of quality or functionality.... |
| 22:43.49 | dlynes | [TK]D-Fender: They're cheap |
| 22:43.56 | [TK]D-Fender | dlynes : how chea? |
| 22:43.57 | dlynes | [TK]D-Fender: considerably cheaper than the Polycom 501 |
| 22:44.23 | dlynes | [TK]D-Fender: besides...most offices are used to that look from the new style Nortel handsets |
| 22:44.32 | dlynes | [TK]D-Fender: About $120Cdn |
| 22:44.40 | froguz | somebody has tested openvox hardware?? |
| 22:45.25 | dlynes | [TK]D-Fender: compare it to a polycom full duplex speaker phone (501), at about $180USD(?) |
| 22:45.32 | *** join/#asterisk file (i=jcolp@216.237.114.82) |
| 22:45.39 | Strom_C | hello mr. file |
| 22:45.48 | *** join/#asterisk mitcheloc (i=user@204.8.143.106) |
| 22:45.52 | file | hola |
| 22:45.54 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 22:46.15 | cekc | how do I play a fast busy signal for 3 seconds before hanging up? |
| 22:46.26 | mitcheloc | hello |
| 22:46.31 | justinu | playtones(busy) |
| 22:46.32 | justinu | wait(3) |
| 22:46.34 | justinu | hangup |
| 22:46.37 | Strom_C | no no no |
| 22:46.40 | cekc | thanks! |
| 22:46.41 | justinu | oh, fast busy is reorder |
| 22:46.42 | Strom_C | that will play regular busy |
| 22:46.51 | justinu | playtones(reorder) |
| 22:46.52 | cekc | wait what? |
| 22:46.54 | cekc | ah |
| 22:46.59 | Strom_C | reorder |
| 22:47.04 | Strom_C | not "fast busy" |
| 22:47.19 | dlynes | playtones(congestion) is fast busy isn't it? |
| 22:47.31 | Strom_C | whoever started calling it "fast busy" deserves to be shot |
| 22:47.36 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:47.52 | tainted- | dlynes ye |
| 22:47.54 | tainted- | yea |
| 22:48.02 | tainted- | wonder if it's a 1.2.4 bug |
| 22:48.07 | tainted- | gonna try 1.2.7.1 |
| 22:48.07 | justinu | sorry, dlynes is correct |
| 22:48.09 | justinu | my mistake |
| 22:48.20 | justinu | use playtones(congestion) |
| 22:48.26 | dlynes | tainted-: I've noticed a number of problems with 1.2.4, including sip subscriptions |
| 22:48.28 | cekc | yeah, reorder didn't work |
| 22:48.31 | dlynes | tainted-: try upgrading to 1.2.7.1 |
| 22:48.45 | tainted- | god i hope it works |
| 22:48.46 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 22:48.48 | chiardon | Hi |
| 22:48.57 | dlynes | tainted-: yeah, it would be cool if it does |
| 22:49.05 | dlynes | tainted-: but, if you've gotten as far as you have |
| 22:49.09 | dlynes | tainted-: it will work |
| 22:49.12 | tainted- | the freeswitch hacks have an irc initiated conf |
| 22:49.14 | dlynes | tainted-: it's just a question of why it's not |
| 22:49.22 | chiardon | I need some VoIP gateways ... which brand would you recommend? |
| 22:49.23 | dlynes | tainted-: Yeah, I've noticed |
| 22:49.33 | cekc | is there a playtones for SIT? |
| 22:49.37 | tainted- | the freeswitch guys have a hack that initiates conf from irc |
| 22:49.38 | dlynes | sit? |
| 22:49.40 | tainted- | too tired |
| 22:49.54 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 22:49.55 | cekc | umm, that tones that tells telemarketers to fuck off |
| 22:50.04 | dlynes | cekc: Yeah, there is...one sec |
| 22:50.05 | Strom_C | cekc: why do you want to play reorder for three seconds only? typical practice is to let the reorder time out after two minutes or so |
| 22:50.33 | cekc | Strom_C: I made a phone that plays funny messages when you pick it up |
| 22:50.43 | tainted- | playback(fuckOff)? |
| 22:50.57 | Peaceful | Is there a way to change caller id display on Cisco 7960's on OUTBOUND calls? |
| 22:51.05 | cekc | heh |
| 22:51.09 | dlynes | cekc: app_zapateller |
| 22:51.20 | Peaceful | I mean, caller id for incoming is easy... |
| 22:51.40 | cekc | welcome, lines-complaining-customers, privacy-you-are-blacklisted, not-taking-your-call, goodbye |
| 22:52.01 | Strom_C | Peaceful: change the caller ID based on the line appearance used to dial out? |
| 22:52.37 | nettie | anyone how coul dbe possible that moh doesnt work anymore? I'm pretty sure I didnt touch anything important! mpg123 is loaded during asterisk startup and if I create an extension which points to it, it works. Seems that asterisk dont recognize the hold button anymore. I started the console in verbose mode and when I used to hit the hold button I remeber it used to notify me. Now the phone says the call it's in hold, asterisk doesnt notice |
| 22:53.10 | nettie | my phone is a polycom soundpoint 301 |
| 22:53.21 | dlynes | cekc: (950,0,330,0)(1400,0,330,0)(1800,0,330,0)(0,0,1000,0) <-- SIT tone |
| 22:53.33 | justinu|laptop | zapateller is easier to type :) |
| 22:53.40 | dlynes | yeah, no kidding |
| 22:53.50 | nettie | I also confirm that both ztdummy, zaptel modules are laoded |
| 22:53.52 | cekc | i got zapateller to work, which means now I know how to start applications |
| 22:54.10 | cekc | dlynes: how do I enter that, just put the tones instead of the application? |
| 22:54.19 | cekc | maybe I'll have it play a song |
| 22:54.32 | nettie | when I issue a moh reload everything looks great.. it's very stange |
| 22:54.45 | Peaceful | Strom_C: no, actually change who the phone says you're calling out to |
| 22:54.46 | dlynes | cekc: have a look at indications.conf |
| 22:55.11 | dlynes | cekc: anything in there you see you can do playtones(nameoftonesforyourlocale) |
| 22:55.16 | Peaceful | Strom_C: I have people dialed into AgentLogin(), and I want them to be able to see caller id for incoming calls |
| 22:55.45 | Peaceful | ^- not trivial, probably because cisco doesn't expect you to change what you're dialing out to |
| 22:56.06 | Peaceful | ...unless someone proves me wrong and it IS trivial. I wouldn't mind that. |
| 22:56.19 | justinu|laptop | i like to switch my indications.conf to uk |
| 22:56.22 | justinu|laptop | sounds more interesting |
| 22:56.38 | dlynes | heh...and throw your callers for a loop/ |
| 22:56.40 | Peaceful | I mean, I could execute an external app that telnets into the phone, but I don't see an option for changing what's displayed even there |
| 22:56.49 | [TK]D-Fender | dlynes : Actually $120 is pretty good for a 9133.... |
| 22:57.01 | dlynes | [TK]D-Fender: Yeah...it's a pretty good deal |
| 22:57.03 | [TK]D-Fender | dlynes : not such a bad call at that price. |
| 22:57.14 | dlynes | [TK]D-Fender: Much better than the other crappy voip phones I've used |
| 22:57.28 | dlynes | [TK]D-Fender: Grandstream 102 has sorta decent quality, but it's so damned ugly |
| 22:57.32 | [TK]D-Fender | dlynes : Yeah I suppose.... that USD or CAD? |
| 22:57.42 | dlynes | [TK]D-Fender: Like I said...Canadian |
| 22:57.45 | [TK]D-Fender | cool |
| 22:57.50 | [TK]D-Fender | yeah I'd say go for it... |
| 22:57.54 | [TK]D-Fender | how many? |
| 22:57.56 | justinu|laptop | dlynes: i just like their tone plan better for some reason |
| 22:58.01 | dlynes | [TK]D-Fender: one at a time? |
| 22:58.04 | dlynes | [TK]D-Fender: or more? |
| 22:58.15 | justinu|laptop | dlynes: maybe because i'm sick of the USA tones |
| 22:58.22 | [TK]D-Fender | dlynes : I mean how many units being ordered? |
| 22:58.25 | dlynes | [TK]D-Fender: I can get them cheaper, but the cheaper price i have to pay shipping on |
| 22:58.34 | tainted- | crossing fingers |
| 22:58.41 | dlynes | [TK]D-Fender: I can order one at a time, or fifty at a time |
| 22:58.48 | dlynes | [TK]D-Fender: That's the price for 1-9 at a time |
| 22:58.51 | justinu|laptop | i'm upgrading my production systems to 1.2.6 tonite |
| 22:58.56 | cekc | any recommendations for voice service? I've had decent experience with broadvoice. I am switching my company to IP phones because we have a $900/month phone bill |
| 22:58.59 | [TK]D-Fender | dlynes : Ok, lets try this ANOTHER way. how many WILL you be ordering? |
| 22:59.17 | tainted- | dlynes no go |
| 22:59.25 | justinu|laptop | cekc: there are people here that can hook you up |
| 22:59.35 | justinu|laptop | with global crossing service |
| 22:59.40 | justinu|laptop | good price |
| 22:59.49 | justinu|laptop | hopefully fairly reliable |
| 23:00.08 | justinu|laptop | probably at least as reliable as broadvoice |
| 23:00.15 | Strom_C | Qwell: you still here? |
| 23:00.16 | cekc | we make a lot of phone calls. and even more faxes, but I'm working on voice for now |
| 23:00.23 | Qwell | barely |
| 23:00.24 | justinu|laptop | how many minutes? |
| 23:00.28 | justinu|laptop | per month |
| 23:00.50 | justinu|laptop | cekc: i recommend you trial their service for maybe a month |
| 23:00.52 | justinu|laptop | see how it goes |
| 23:00.52 | cekc | let me check. we have a 80 page phone bill from AT&T |
| 23:01.09 | Strom_C | Qwell: do I need to make a separate conf file for the phone's MAC address like I do with the SIP images, or does the skinny.conf handle that? |
| 23:01.18 | Qwell | yeah, seperate files |
| 23:01.46 | dlynes | [TK]D-Fender: We usually order between 4 and 6 at a time |
| 23:01.55 | Strom_C | so copy SEP7960.cnf.xml to SEP[mac].cnf.xml? |
| 23:01.55 | dlynes | [TK]D-Fender: depending on our needs for the customer in question |
| 23:02.17 | dlynes | [TK]D-Fender: we don't have any order atm, but we'll be ordering 9 or so soon |
| 23:04.22 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 23:04.40 | [TK]D-Fender | dlynes : Ok, another way : how many total are you expecting to deploy? |
| 23:05.53 | Qwell | Strom_C: yeah |
| 23:06.07 | Strom_C | k |
| 23:06.21 | *** join/#asterisk jazzplyer (n=jhaar@222-153-80-251.jetstream.xtra.co.nz) |
| 23:06.31 | *** part/#asterisk jazzplyer (n=jhaar@222-153-80-251.jetstream.xtra.co.nz) |
| 23:06.49 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 23:06.51 | [TK]D-Fender | dlynes : Just reading now I'm getting the impression you are an integrator, and not an end consumer... |
| 23:07.30 | linlin | what can you guys reccomend for a cheap toll free DID provider ? |
| 23:08.47 | dlynes | [TK]D-Fender: correct...we're an interconnect |
| 23:09.07 | justinu|laptop | linlin, possibly asterlink |
| 23:09.16 | nettie | Hi again [TK]D-Fender sorry to bother you again I was wodnering if you might have an idea of what could be wrong when asterisk cant put the calls on hold anymore. I actually press the hold key on the phone but nothing happen on the server. The console used to show some messages in verbose mode when I put calls on hold, no it doesnt do it anymore. I really cant figure out what could be wrong. Do you have any idea please? thanx in adv. |
| 23:09.19 | dlynes | [TK]D-Fender: so as for a total number expecting to deploy, it would depend on over how long a period of time you are talking |
| 23:09.19 | justinu|laptop | their pricing is very reasonable |
| 23:10.16 | dlynes | [TK]D-Fender: I'm just trying to get this godforsaken billing system over and done with so we can start selling more aggressively |
| 23:10.29 | [TK]D-Fender | nettie : Hmmm, no indication... is the call staying on the phone? |
| 23:10.37 | [TK]D-Fender | billing = pain |
| 23:11.04 | dlynes | But, I found a way to make it a little less painful |
| 23:11.05 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 23:11.09 | [TK]D-Fender | damn, Slackware has fallen from distrowatch's top 10 list! |
| 23:11.17 | tmccrary | Has anyone here shopped at discountvoipoutlet.com? |
| 23:11.23 | dlynes | CGI::Application and HTML::Template are infinitely easier to work with than Java and Struts |
| 23:11.52 | nettie | [TK]D-Fender the phone actually shows that the call in on hold |
| 23:11.53 | dlynes | [TK]D-Fender: Yeah, that's a bummer, definitely |
| 23:12.07 | dlynes | [TK]D-Fender: I'm eagerly waiting for 11.0 to come out |
| 23:12.14 | nettie | [TK]D-Fender but *NOTHING* is showed by the console |
| 23:12.21 | nettie | even at the highest verbose mode |
| 23:12.30 | terrapen | SHARIF DON'T LIKE IT |
| 23:12.37 | terrapen | rock the casbah rock the casbah! |
| 23:13.27 | tmccrary | Has anyone shopped at thevoipconnection.com? |
| 23:13.59 | justinu|laptop | as soon as the shariff had cleared the square... they began to waill.... |
| 23:14.07 | justinu|laptop | tmccrary: yes, they are good people |
| 23:14.16 | justinu|laptop | tmccrary: i met them at astricon also |
| 23:14.48 | tmccrary | okay cool, because they also run discount voip outlet and the prices are really good (I was afraid they might be TOO good) |
| 23:14.51 | [TK]D-Fender | nettie : Hmm... sure your MoH is set up right? And did it work before? If so, what is the last thing changed before it stopped? |
| 23:15.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 23:15.45 | tmccrary | that's cool, they have a SIP URL in their contact us info :) |
| 23:15.51 | justinu|laptop | i heard someone wasn';t happy with the VS1 tho |
| 23:15.56 | justinu|laptop | but I would never buy something like that |
| 23:15.58 | dlynes | tmccrary: discountvoipoutlet's prices aren't that good |
| 23:16.10 | tmccrary | dlynes: do you have any suggestions? :) |
| 23:16.17 | dlynes | tmccrary: voipdepot.ca |
| 23:16.27 | dlynes | tmccrary: williamsglobal.com |
| 23:16.40 | tmccrary | I'm american |
| 23:16.45 | dlynes | tmccrary: williamsglobal.com requires you to set up an account though |
| 23:16.49 | [TK]D-Fender | tmccrary : What are you looking to buy? |
| 23:17.16 | justinu|laptop | stuff |
| 23:17.17 | justinu|laptop | and junk |
| 23:17.27 | terrapen | they've always done us right |
| 23:17.38 | nettie | [TK]D-Fender it definitely worked before, all the relative kernel timing modules are loaded, mpg123 starts with asterisk, "moh reload" looks just perfect. I dont really get what could be wrong. Just the phone-asterisk link seems to have problems because if I configure an dextension to play moh for 30 secs it works perfectly! |
| 23:17.42 | tmccrary | Nothing good (and I mean it). The jiu jitsu academy I train at is moving and I'm helping them with a phone system. Basically 4 Budgettones and a phone with more lines for the receptionist |
| 23:17.52 | dlynes | terrapen: Yeah...voipsupply's pretty good, but I've had times where it took 5 weeks to get one of their shipments across the border |
| 23:18.15 | terrapen | that's not their fault, is it? :) |
| 23:18.16 | dlynes | terrapen: it was a shipment of 100 sipura 2000's, so we had to order another shipment |
| 23:18.24 | tmccrary | voipsupply is horrible, their phone system doesn't work and they over charge (and they have that cool Best Buy-esque product replacement plan that they AUTOMATICALLY add to your bill unless you take it off) |
| 23:18.46 | dlynes | Oh yeah...their phone system is horrible...you hit '0' to get an operator on it, and it hangs up on you |
| 23:19.01 | terrapen | what? they only overcharge you if you don't ask for a discount |
| 23:19.04 | [TK]D-Fender | tmccrary : Well, when the product you;'re looking to by is such junk, why bother caring who its coming from ;) |
| 23:19.13 | terrapen | and i've never had any problem calling my rep directly |
| 23:19.17 | tmccrary | well, I want the price to be as low as possible |
| 23:19.17 | terrapen | and he's very responsive to e-mails |
| 23:19.22 | [TK]D-Fender | tmccrary : GS = SOAS |
| 23:19.23 | terrapen | so ask for a lower price |
| 23:19.27 | tmccrary | SOAS? |
| 23:19.35 | [TK]D-Fender | Shit-On-A-Stick |
| 23:19.36 | terrapen | like all purchasing, you have to hound salespeople for the best deal |
| 23:19.38 | tmccrary | I'm not under the impression that it's good |
| 23:19.39 | DoktorGreg | heya asterisk wizzes |
| 23:19.44 | tmccrary | I under the impression that it's cheap |
| 23:19.51 | tmccrary | I hate grandstream's products |
| 23:20.01 | [TK]D-Fender | tmccrary : No, Sipura is cheap... GS = SHIT :) |
| 23:20.02 | tmccrary | in fact, I've already destroyed a GXP in a fit of rage |
| 23:20.06 | DoktorGreg | Each sip phone can call other sip phone no problem |
| 23:20.18 | DoktorGreg | both ways |
| 23:20.19 | DoktorGreg | but |
| 23:20.33 | tmccrary | I had a 841 sipura phone, I hated that worse than the GXP |
| 23:20.37 | *** join/#asterisk beox (n=beos@200-161-29-5.dsl.telesp.net.br) |
| 23:20.37 | DoktorGreg | when i use the asterisk manager interface api originatecall |
| 23:20.45 | [TK]D-Fender | Everything EXCEPT the 841 :) |
| 23:20.48 | tmccrary | it wasn't mine, however, so I did not get to rip it to pieces |
| 23:20.53 | DoktorGreg | on of the phones isn't getting sound |
| 23:20.59 | DoktorGreg | on = one |
| 23:21.07 | *** part/#asterisk beox (n=beos@200-161-29-5.dsl.telesp.net.br) |
| 23:21.37 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 23:21.57 | dlynes | tmccrary: the grandstream 102's not too bad, but it's damned ugly |
| 23:22.16 | dlynes | tmccrary: it looks like a kid's toy |
| 23:22.30 | DoktorGreg | am i on my own hacking around with the asterisk manager api? |
| 23:22.40 | justinu|laptop | no |
| 23:22.41 | dlynes | DoktorGreg: probably not |
| 23:22.47 | terrapen | hah, I forgot that I had the Richard Cheese version of this song, too |
| 23:22.49 | terrapen | KICK ASS! |
| 23:22.49 | tmccrary | yeah, Grandstream's aesthetics are hilariously bad. However, they are CHEEEEAP |
| 23:22.54 | litecode | does anybody control their dialplan with an outside source instead of the standard config or external configs? |
| 23:22.55 | justinu|laptop | do I want to talk about for money? sure! |
| 23:23.10 | dlynes | lol |
| 23:23.58 | terrapen | I prefer Polycom IP501's |
| 23:24.04 | DoktorGreg | man i hate thoes |
| 23:24.12 | terrapen | the Aastra is decent for a cheapie phone |
| 23:24.14 | DoktorGreg | now its working... i didnt change anything |
| 23:24.25 | terrapen | you hate polycoms? what the hell for? |
| 23:24.43 | dlynes | terrapen: he hates the crappy phones he's using, not the polycoms |
| 23:24.47 | terrapen | ahhh |
| 23:24.56 | terrapen | well, if y'all want the name of my voipsupply rep, msg me |
| 23:25.02 | DoktorGreg | nononon |
| 23:25.03 | terrapen | ask him for a deal and you'll probably get one |
| 23:25.09 | DoktorGreg | i cant speak inteligently about polycoms |
| 23:25.22 | DoktorGreg | i hate bugs that go away without changing anything in software |
| 23:25.31 | litecode | I would rather manage asterisk with python, this would be through an AGI right? if so, which AGI is best? |
| 23:25.58 | DoktorGreg | because bugs are still there |
| 23:26.13 | DoktorGreg | i just don't know how to reproduce it |
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| 23:27.46 | *** part/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 23:28.13 | tmccrary | has anyone done autoanswer with the grandstreams? |
| 23:28.44 | tmccrary | also, does anyone know if the Budgetone 102 has one or two lines? |
| 23:28.51 | tmccrary | lines as in simultaneous calls |
| 23:29.06 | dlynes | tmccrary: one line...it's called a budgetone 102 because it has an rj45 jack on it for your pc as well |
| 23:29.25 | tainted- | dlynes i might just settle on dial() and bringing in one person |
| 23:29.27 | dlynes | thus the two....two rj45 jacks |
| 23:29.33 | tainted- | so close, but yet so far |
| 23:29.36 | hads|home | tmccrary: They are only 10meg ports to. |
| 23:29.51 | dlynes | tainted-: I'm sure if you play with it enough, you'll get there |
| 23:30.13 | dlynes | tainted-: if I wasn't so swamped with work right now, I'd offer to help |
| 23:31.06 | dlynes | hads|home: doesn't really matter unless you're doing a lot of bandwidth transfers across your lan |
| 23:31.30 | dlynes | hads|home: 10Mb is still pretty fast |
| 23:31.51 | dlynes | besides...the internet connection isn't anywhere near close to 10Mb |
| 23:35.15 | mitcheloc | hey guys, i'm working on getting music on hold to work |
| 23:35.49 | mitcheloc | it says it starts up but i don't get any sound, i'm using mpg321 though, and set application=/usr/bin/mpg321 |
| 23:36.05 | mitcheloc | there isn't any info in the log files...is there anyway i can check and see whats going on? |
| 23:37.38 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 23:37.56 | tmccrary | Has anyone here used autoanswer on the grandstream? |
| 23:38.03 | dlynes | mitcheloc: Which version of asterisk are you using? |
| 23:38.29 | mitcheloc | svn, from 12/20/05 |
| 23:38.31 | dlynes | tmccrary: I think I might have played around with it on the 102, but that was about it...that was over six months ago, too |
| 23:39.26 | dlynes | mitcheloc: try mode=quietmp3\ndirectory=... for your default class |
| 23:39.36 | dlynes | mitcheloc: erm actually...hold on a second |
| 23:40.04 | mitcheloc | dlynes: i have this http://pastebin.ca/51040 |
| 23:40.18 | mitcheloc | i tried with = or => just in case, neither seems to be the problem |
| 23:40.26 | dlynes | mitcheloc: can you pastebin your cat /proc/modules also? |
| 23:41.04 | DoktorGreg | haha nailed it! |
| 23:41.05 | mitcheloc | dlynes: http://pastebin.ca/51041 |
| 23:41.16 | DoktorGreg | nat keep alive interval |
| 23:42.18 | DoktorGreg | ok question, cant find an answer easy |
| 23:42.38 | DoktorGreg | is nat keep alive interval ms or seconds? |
| 23:42.46 | dlynes | mitcheloc: have you changed any of your mp3's? |
| 23:42.53 | dlynes | DoktorGreg: seconds |
| 23:43.22 | mitcheloc | yes, i added one, and took out the other default ones |
| 23:43.54 | dlynes | mitcheloc: Did you check to make sure it's an 8K mp3? |
| 23:44.09 | mitcheloc | dlynes: no i didn't, i'll swap out with the default ones real fast to check |
| 23:45.45 | dlynes | mitcheloc: also, why are you attempting to play it out to your speakers, instead of stdout? |
| 23:45.47 | markus99 | could anyone help with a x101p clone card not working with asterisk, the card seems to work fine when I use ztmonitor |
| 23:46.59 | Strom_C | markus99: define "not working" |
| 23:47.37 | mitcheloc | dlynes: here is the pastebin http://pastebin.ca/51042 |
| 23:47.40 | markus99 | Strom_C: i can't get asterisk to pick up the line |
| 23:47.54 | mitcheloc | dlynes: i'm not trying to play it out the speakers, i suppose that means i need some command line arguments? |
| 23:48.02 | mitcheloc | dlynes: (on the applicaton line?) |
| 23:48.08 | dlynes | mitcheloc: Yes, at a minimum you need -s |
| 23:48.16 | dlynes | mitcheloc: -s means play it to stdout |
| 23:48.43 | dlynes | mitcheloc: try this: application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
| 23:48.58 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 23:48.58 | dlynes | mitcheloc: that's the default parameters when asterisk is installed |
| 23:49.13 | mitcheloc | dlynes: right, i'm using old config files, and had to update, maybe thats it |
| 23:49.25 | mitcheloc | dlynes: but i am using mpg321, not 123 |
| 23:49.49 | dlynes | mitcheloc: didn't you say asterisk wasn't saying anything? |
| 23:49.50 | markus99 | Strom_C: i'm sure I have everything set correctly in the extensions.conf, zapata.conf and zaptel.conf but it doesn't even show that the line is ringing on the cli |
| 23:50.08 | mitcheloc | dlynes: yes |
| 23:50.40 | mitcheloc | dlynes: i meant, not saying any "errors", here is what i just got: /usr/bin/mpg321 -q -r 8000 -f 8192 -b 2048 --mono -s: No such file or directory |
| 23:50.45 | dlynes | mitcheloc: mpg123 takes the same command line parameters as mpg321 I believe |
| 23:50.48 | Strom_C | markus99: the x100p cards blow in general, and the clones are even worse |
| 23:50.50 | mitcheloc | dlynes: i suppose probably mpg321 doesn't take the same command lines |
| 23:50.53 | mitcheloc | yea |
| 23:51.10 | Micc | Does anyone know what is happening with nufone? |
| 23:51.18 | dlynes | mitcheloc: mpg321 is supposed to be a free clone of mpg123 |
| 23:51.30 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 23:51.33 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:51.37 | Micc | I have a couple 800 numbers with them. Is there any way I can move the numbers quickly or is it going to be fixed soon? |
| 23:51.42 | Jaxxan | hey guys |
| 23:51.48 | dlynes | mitcheloc: You might also try madplay with the following line: application=/usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -12 |
| 23:51.59 | mitcheloc | dlynes: right, hmm i couldn't get mpg123 to compile thats why i went with mpg321 |
| 23:52.06 | dlynes | mitcheloc: madplay doesn't have the nasty habit of leaving a session hanging like mpg123 does |
| 23:52.31 | dlynes | mitcheloc: you'll need libmad and libid3tag as well |
| 23:55.36 | *** join/#asterisk isaiah (n=test@208-187-93-4.br1.hnv.mi.frontiernet.net) |
| 23:56.28 | markus99 | Strom_C: i think the card itself is working because if I run ztmonitor i can here (pc speaker) the line ring and the audio when the line is picked up, I would think from that the card is working |
| 23:57.23 | Jaxxan | so i wanna do call recording with gsm codec to save some diskspace, but when i click on a GSM recorded file, it wont open in my player by default |
| 23:57.24 | dlynes | markus99: what does it say when you type zap show status on the cli? |
| 23:57.31 | Jaxxan | anyone have an easy solution for that ? |
| 23:57.41 | dlynes | Jaxxan: use a soundplayer that supports gsm? |
| 23:58.03 | markus99 | dlynes: i don't have that command |
| 23:58.08 | Jaxxan | it's weird |
| 23:58.09 | Jaxxan | hrm |
| 23:58.16 | dlynes | markus99: Then you forgot to load chan_zap.so in modules.conf |
| 23:58.32 | dlynes | markus99: and that's probably why you don't detect ringing in asterisk |
| 23:58.47 | Druken | hmm..... |
| 23:58.58 | markus99 | dlynes: i'll check that out |
| 23:59.18 | Druken | anyone got a better reverse lookup than 411.ca? |
| 23:59.38 | dlynes | Jaxxan: If you tell people what operating system you have, someone might have a suggestion as to which player to try for gsm files |
| 23:59.57 | dlynes | Druken: yellowpages.ca? |