| 00:02.41 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 00:26.42 | luke-jr | file: ping |
| 00:36.49 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 00:41.57 | Shadowfire_ | does anyone know if it is possible that zaptel.conf could cause eth0 not to function? |
| 00:44.05 | sniper[FOO] | Shadowfire_: could you post your zaptel.conf? |
| 00:44.16 | Shadowfire_ | yes... hold on |
| 00:48.36 | Shadowfire_ | sorry... I am on a remote session... getting it through sftp |
| 00:48.38 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 00:54.02 | Shadowfire_ | sniper: how would you like me to post it... |
| 00:55.11 | Shadowfire_ | sniper [FOO]: how would you like me to post it? |
| 00:58.20 | Teln1100A | any one here seen ss7 <-> asterisk integration? |
| 00:58.41 | [TK]D-Fender | ~pb |
| 00:58.42 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:58.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 00:59.11 | [TK]D-Fender | Teln1100A, go lookup ss7 on the WIKI for some guides & info |
| 00:59.51 | *** join/#asterisk HarryR (n=harryr@cpc2-lamb3-0-0-cust255.bmly.cable.ntl.com) |
| 01:11.50 | Yourname`` | Let me help you out then, [TK]D-Fender :D |
| 01:12.02 | Shadowfire_ | http://pastebin.ca/699079 |
| 01:12.25 | Yourname`` | What would I have to do to pick up the phone, dial some number and start talking... and all that will be heard on the speaker phone of all connected phones? |
| 01:13.10 | HarryR | Yourname``: you'd need support on the speakerphones to remain connected to a conference or something |
| 01:13.22 | Shadowfire_ | thx for the heads up on the pastebin and stuff |
| 01:13.39 | Yourname`` | HarryR: Someone did it without all that for SPA941s. |
| 01:13.42 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 01:13.42 | HarryR | Yourname``: i'm having nightmares of a boss barking orders at all his employees at once through the desktop phones |
| 01:13.48 | [TK]D-Fender | Shadowfire_, "channels=1-4" does not belong in zapata.conf |
| 01:13.59 | Shadowfire_ | eeek... oops... |
| 01:14.00 | Yourname`` | hehe HarryR, that's actually what I wanna do. |
| 01:14.04 | HarryR | oooh |
| 01:14.05 | Yourname`` | "GET BACK TO WORK!" |
| 01:14.18 | Shadowfire_ | so that may be why it's shutting down my nic??? or no |
| 01:14.19 | [TK]D-Fender | Yourname``, depends on what kind of phones you have in mind. |
| 01:14.39 | HarryR | or you could just set them up to auto-answer, but again it depends on the phone |
| 01:14.40 | Yourname`` | [TK]D-Fender: Linksys SPA941s. |
| 01:14.40 | [TK]D-Fender | Shadowfire_, No, and it has NOTHING to do with your NIC issues whatever they may be. |
| 01:14.54 | [TK]D-Fender | Yourname``, Those support auto-answer, so yes its viable. |
| 01:15.01 | [TK]D-Fender | Yourname``, "show application page" |
| 01:15.36 | Yourname`` | [TK]D-Fender: Perfect, thank you! ManxPower mentioned the same.. but he said it's 1.2 only? |
| 01:16.48 | [TK]D-Fender | Shadowfire_, Which is exactly where you left off with this hours ago. You keep thinking zaptel has something to do with this but nothing to support the theory. |
| 01:17.02 | [TK]D-Fender | Yourname``, 1.2 & 1.4 both have app_page |
| 01:17.20 | Yourname`` | Great, thanks a ton [TK]D-Fender :) |
| 01:17.21 | Shadowfire_ | D-Fender: I fixed that issue... |
| 01:17.35 | Shadowfire_ | my zaptel channels where not loading up... |
| 01:17.49 | Shadowfire_ | took care of that... with russelb help... |
| 01:18.23 | Shadowfire_ | he pointed out that I might want to look in my zaptel.config since I was recieveing a certain error |
| 01:19.42 | [TK]D-Fender | Shadowfire_, you need to have had zaptel properly compiled and the appropriate modules loaded and then have zaptel initialized before starting * |
| 01:20.18 | Shadowfire_ | well.. I would agree... I had all that in line... and it was working... but then I added to HPEC in the mix... |
| 01:20.34 | Shadowfire_ | and that is what cause me some issues... |
| 01:20.49 | Shadowfire_ | and the fact that I have a bit to learn on * |
| 01:20.50 | [TK]D-Fender | Yourname``, on my Polycom's I have both rin-answer & SILENT answer. Especially useful when I mute the incoming channel for insurreptiously spying :) |
| 01:21.07 | luke-jr | mog: ping |
| 01:21.23 | Shadowfire_ | I have not problem admitting that I have room to grow... we all start somewhere with things... |
| 01:21.54 | Yourname`` | [TK]D-Fender: As always, I had no clue that could be done, hehe.. so what's that like, just call the extension of the person you wanna spy on and it switches itself on on speaker? |
| 01:22.51 | [TK]D-Fender | Yourname``, You have to set some SIP headers based on the phone you are calling so that it knows to auto-answer on speakerphone, and the muting is done on MY phone when I initiate it (by hand of course) |
| 01:23.20 | Yourname`` | Aaah, nice. |
| 01:24.21 | Maxxed | hey have any of you guys ever seen that tivo upate thing? |
| 01:24.23 | Maxxed | update? |
| 01:24.50 | Maxxed | it s like a comercial, but its like a black n whire encoded mess |
| 01:24.56 | Maxxed | firmware update broadcasted |
| 01:26.02 | tzafrir | Shadowfire_, managed to get a working zaptel.conf ? |
| 01:26.12 | Shadowfire_ | yes... |
| 01:26.14 | tzafrir | if not, use xpp/utils/genzaptelconf in zaptel |
| 01:26.36 | Shadowfire_ | I am calling in and it picks up... |
| 01:26.41 | Shadowfire_ | then I went to ext... |
| 01:26.48 | Shadowfire_ | and it starts beeping... |
| 01:27.02 | Shadowfire_ | and then the system locks and I can not ping system |
| 01:27.24 | Shadowfire_ | the phone line starts beeping by the way |
| 01:27.44 | Shadowfire_ | one long continous beep... |
| 01:30.14 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 01:31.18 | tzafrir | Shadowfire_, do you see anything on the console? |
| 01:31.45 | Shadowfire_ | unfortunately I am on remote... I could do a asterisk -vvvvv or whatever I suppose... |
| 01:32.06 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666903.dsl.bell.ca) |
| 01:32.09 | tzafrir | Shadowfire_, if you can't even ping it, it doesn't sound like Asterisk |
| 01:32.30 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.113) |
| 01:32.35 | Shadowfire_ | Asterisk is the only thing on it |
| 01:32.45 | tzafrir | (unless you normally block pings, or Asterisk keeps the system so busy, pings hve to wait very long in the transmit queue) |
| 01:32.47 | Shadowfire_ | It's AsteriskNow |
| 01:32.52 | rob0 | aha |
| 01:32.57 | tzafrir | there's also Zaptel |
| 01:32.59 | Shadowfire_ | 1.4 |
| 01:35.01 | [TK]D-Fender | Shadowfire_, Do yourself a favour and ditch it and reinstall any standard distro and compile * yourself. |
| 01:35.35 | Yourname`` | Shadowfire_: [TK]D-Fender said that to me in April, and I listened to him. Today I'm glad I did listen to him, lol |
| 01:36.27 | Shadowfire_ | D-Fender... I almost did that.. but then I wanted a GUI... |
| 01:36.47 | Teln1100A | any progress on the skype front with Asterisk? |
| 01:37.04 | Shadowfire_ | D-Fender how does it fare with Ubuntu? |
| 01:37.41 | Shadowfire_ | Yourname: what disto are you running? |
| 01:37.41 | rob0 | He told me to take a flying leap, and I did, and now I have a broken toe. |
| 01:37.50 | [TK]D-Fender | Shadowfire_, Can work, but I'd still pick something with a STANDARD boot process and package repository. Ubuntu does mess around with a lot of stuff. |
| 01:38.24 | Shadowfire_ | D-Fender: Yeah... they are making there own packaging... |
| 01:38.28 | [TK]D-Fender | Teln1100A, No, nor should there be any expectation of any. |
| 01:39.01 | Shadowfire_ | D-Fender: What about Debian? Centos? Fedora? |
| 01:39.51 | tzafrir | Teln1100A, any progress on reverse-engeneering the Skype protocol? |
| 01:39.58 | [TK]D-Fender | Shadowfire_, All far better choices |
| 01:40.11 | Shadowfire_ | D-Fender: lol... understand |
| 01:40.41 | Teln1100A | I think skype should open it up, publish a spec |
| 01:40.56 | [TK]D-Fender | Teln1100A, its a bastardized closed protocol so don't expect any non-commerical stuff to come out for that, and from what I've heard of the ones out there, the implementation is BEYOND ugly. |
| 01:42.58 | Teln1100A | even a linux command line client perhaps would be nice? |
| 01:43.34 | sniper[FOO] | [TK]D-Fender: ever used centos? |
| 01:43.43 | [TK]D-Fender | Teln1100A, Same crack, different packaging... |
| 01:44.06 | [TK]D-Fender | sniper[FOO], I use it in the majority of my installs |
| 01:44.53 | sniper[FOO] | yay :S |
| 01:59.19 | mog | luke-jr, pang |
| 02:00.09 | luke-jr | mog: IM? |
| 02:01.46 | *** join/#asterisk De_Mon (i=de_mon@fl-71-52-101-157.dhcp.embarqhsd.net) |
| 02:05.36 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.108) |
| 02:06.42 | dijungal | is everyone sleeping? |
| 02:09.44 | dan__t | Hrm... Anyone familiar with PolyCom phones, and how to get them to dial out properly with Asterisk? I'm having a few issues here. Seems that the only way I can even SEE it connect to * and initiate a call is if I make the phone use a SIP Proxy. |
| 02:12.15 | [TK]D-Fender | dan__t, plenty of guides out there, and an especially good on on how to provision them on the WIKI in the Asterisk-at-home-handbook section. |
| 02:14.05 | dan__t | I've been through most all of them, otherwise I would not be asking like this ;) |
| 02:14.23 | [TK]D-Fender | dan__t, How are you configureing them currently? |
| 02:15.05 | dan__t | er, tftp, booting the SIP and BOOT roms with their config files. |
| 02:15.14 | dan__t | I can receive inbound calls just fine. Dialing out is what sucks. |
| 02:15.21 | [TK]D-Fender | dan__t, Well pastebin up your configs |
| 02:15.27 | dan__t | Sure, hold on. |
| 02:16.27 | dan__t | http://pastebin.ca/699120 |
| 02:17.01 | [TK]D-Fender | dan__t, host=192.168.1.3 <- should be dynamic |
| 02:17.12 | [TK]D-Fender | dan__t, dtmfmode=inband <-- should eb rfc2833 |
| 02:17.25 | [TK]D-Fender | dan__t, progressinband=no <-- just remove |
| 02:17.43 | [TK]D-Fender | dan__t, and clearly I was referring to yoru POLCYOM configs |
| 02:17.43 | dan__t | ok. |
| 02:17.59 | dan__t | Clearly, that was not clear. haha. |
| 02:18.03 | dan__t | Hold on a sec. They're quite large. |
| 02:18.23 | dan__t | I used the default ones that came with the SIP software, with a few minor changes as indicated in various guides and howtos etc etc. |
| 02:19.10 | [TK]D-Fender | dan__t, yes, thats usually where the mistakes bigin ;) |
| 02:19.13 | [TK]D-Fender | begin* |
| 02:19.36 | dan__t | I read somewhere, and with some help from some guys in here last night, that the actual PolyCOm dialplan may play a part in why it is not working properly. |
| 02:20.03 | dan__t | Give me a sec. |
| 02:20.09 | [TK]D-Fender | dan__t, entirely possible and something I can confirm rather fast once you get your butt in gear :p |
| 02:20.22 | *** join/#asterisk mishkiz (n=lincolnz@201-68-222-2.dsl.telesp.net.br) |
| 02:20.25 | dan__t | Yes yes, working on it. |
| 02:20.43 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:24.14 | *** join/#asterisk |omni| (n=rob@c-67-185-70-220.hsd1.wa.comcast.net) |
| 02:24.16 | mishkiz | hello all...Im trying to install an asterisk.1.4.11 at a Debian etch....when I do "make" I got "chan_sip.c: In function âhandle_request_referâ: chan_sip.c:14323: internal compiler error: Aborted". My gcc version is 4.1.2... |
| 02:25.02 | mishkiz | can anybody help me ? |
| 02:28.26 | dan__t | Almost there, [TK]D-Fender |
| 02:28.45 | dan__t | http://pastebin.ca/699130 |
| 02:32.09 | dan__t | I guess if it matters, when I don't set the phone to use a SIP proxy, I don't see any data pass through * on an outbound call. |
| 02:32.25 | dan__t | However, when I set a SIP proxy to that of the IP of the * server, I see activity - then run into a login issue |
| 02:35.31 | dan__t | Did I get it all there, [TK]D-Fender? |
| 02:35.31 | [TK]D-Fender | dan__t, I don't see your login or pxy setting anywhere in there |
| 02:35.48 | dan__t | Because I set it on the phone, wasn't sure I would be editing the correct directives in the confs. |
| 02:36.09 | [TK]D-Fender | dan__t, horrible mistake.... |
| 02:36.15 | dan__t | How so |
| 02:36.17 | [TK]D-Fender | :p |
| 02:36.28 | dan__t | I just don't know either way haha, I'm still really new to this. |
| 02:36.33 | [TK]D-Fender | dan__t, because you aren't showing me the very information I'm sure you've screwed up! |
| 02:36.40 | dan__t | d'oh. |
| 02:36.49 | dan__t | Good point. |
| 02:36.56 | [TK]D-Fender | dan__t, And of course defeating the entire point of provisioning... so it isn't in the PHONE |
| 02:37.11 | dan__t | haha |
| 02:37.11 | dan__t | Yes. |
| 02:37.51 | [TK]D-Fender | dan__t, thats like the joke about the Newfie who won a gold medal in the Olympics.... then had it BRONZED. |
| 02:37.59 | dan__t | haha |
| 02:38.10 | dan__t | Ah well. |
| 02:38.18 | [TK]D-Fender | dan__t, hard-flush the phone and do it all in the configs |
| 02:38.44 | dan__t | I can do that in the admin menu right |
| 02:38.58 | dan__t | reset to default... cool |
| 02:39.05 | [TK]D-Fender | dan__t, yup. "local config" and then "format filesystem" |
| 02:39.24 | dan__t | Formatting, rebooting |
| 02:39.39 | dan__t | What a good idea, though, all the configs being in XML. |
| 02:39.45 | dan__t | Bet that makes it easy to provision a large number of phones. |
| 02:40.47 | dan__t | voIpProt.server.1.address is the IP of my Asterisk server, yes? |
| 02:41.06 | [TK]D-Fender | dan__t, yes. |
| 02:41.10 | dan__t | Er, before that - which file gets edited first? |
| 02:41.35 | dan__t | Just trying to put everything into context so I get a better understanding of how it's going to work. |
| 02:41.45 | [TK]D-Fender | dan__t, do them both at the same time. put server info into sip.cfg and reg specific (user, pass, key handling) in the phone.cfg |
| 02:41.55 | dan__t | So sip.cfg is good for an entire site, whereas phoneN.cfg is for the specific phone |
| 02:41.56 | dan__t | ok, rad. |
| 02:42.46 | [TK]D-Fender | dan__t, you can actually hybridize between the two but i advise treating each as being for a certain scope. phoneXX.cfg is more like for overrides & personalizations. |
| 02:42.58 | dan__t | Sure, I understand now, thanks. |
| 02:43.19 | [TK]D-Fender | dan__t, So in sip.cfg you'd set your server ip, general microbrowser settings,dialplan, etc. |
| 02:43.40 | [TK]D-Fender | dan__t, then for individuals you'd override things in their phoneXX.cfg |
| 02:43.56 | dan__t | Ok. |
| 02:44.25 | dan__t | Does my sip.cfg look alright, minus the voIpPort.server.1.address= part which has now been set to the IP of my * machine? |
| 02:45.42 | [TK]D-Fender | dan__t, <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3|3|3|3|3|3"/> |
| 02:45.54 | [TK]D-Fender | dan__t, I'd think about that with regards to your * dialplan |
| 02:46.15 | dan__t | ok, trying to look through it, hold on a sec |
| 02:46.37 | [TK]D-Fender | dan__t, I'd personally advise : <digitmap dialplan.digitmap="x.T|*.T|#.T" dialplan.digitmap.timeOut="3|3|3|3|3|3"/> |
| 02:47.02 | [TK]D-Fender | dan__t, and above : <dialplan dialplan.impossibleMatchHandling="2" dialplan.removeEndOfDial="0" dialplan.applyToUserSend="1" dialplan.applyToUserDial="1" dialplan.applyToCallListDial="0" dialplan.applyToDirectoryDial="0"> |
| 02:49.15 | dan__t | er... can I put <digitmap> outside of a <dialplan>? |
| 02:50.00 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 02:51.41 | dan__t | I only ask by just looking at the xml context and hierarchy |
| 02:51.45 | [TK]D-Fender | dan__t, fill in the structures where they appear |
| 02:52.05 | [TK]D-Fender | dan__t, My commentary my not appear in order, jsut follow them in bits |
| 02:52.31 | dan__t | heheh |
| 02:54.06 | dan__t | ok, and that requires a reboot on the phone right |
| 02:55.53 | dan__t | The phone doesn't seem to be defaulting itself properly, because it stillc omes up with an IP which is out of the range of the DHCP scope. |
| 02:56.02 | [TK]D-Fender | dan__t, When you're finished, yeah |
| 02:56.13 | [TK]D-Fender | dan__t, check your bootrom |
| 02:56.23 | dan__t | Doing a "Reset Device Setting" now. |
| 02:56.24 | dan__t | What of it? |
| 02:56.26 | [TK]D-Fender | dan__t, that is NOT in your config files. |
| 02:56.38 | [TK]D-Fender | dan__t, For your DHCP issue |
| 02:56.57 | dan__t | Oh, no, I know - just saying that the phone kept its settings. |
| 02:57.04 | [TK]D-Fender | dan__t, TCP boot parameters are local to the phone since it can't be psychic about that stuff :) |
| 02:57.08 | dan__t | I wanted to default the phone completely. |
| 02:57.09 | dan__t | haha |
| 02:57.29 | dan__t | ok, here we go. |
| 02:58.08 | dan__t | I'll grab a nap waiting for it to boot. |
| 02:58.10 | dan__t | ... |
| 03:00.46 | [TK]D-Fender | its 2 minutes, don't have a spaz over it. |
| 03:00.52 | dan__t | haha. |
| 03:01.08 | dan__t | Now, the phone appears to be frozen, been sitting on Running App = sip.ld for a long time now |
| 03:01.39 | dan__t | And my red light is no longer blinking. |
| 03:01.47 | dan__t | But there it is :) |
| 03:02.10 | Nugget | Roxanne! Put on the red light.... Roxanne! Put on the red light.... |
| 03:02.19 | dan__t | hahaha. |
| 03:02.57 | dan__t | OK, got it back up now. |
| 03:03.03 | dan__t | Config files are reloaded, phone is back |
| 03:04.00 | [TK]D-Fender | dan__t, Solid icon for your phon reg? Because at the same time I have no idea what you filled in for your reg's for handliong, etc |
| 03:04.14 | dan__t | reg's? |
| 03:04.16 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.55 | [TK]D-Fender | dan__t, Reg info in your configs. |
| 03:05.26 | [TK]D-Fender | dan__t, Since you supposedly flushed the phones local config and are doing it in the 2 primary files I have no idea if they're sane snce you haven't shown me the new ones. |
| 03:05.45 | dan__t | I simply edited the two lines which you had suggested I do. |
| 03:06.03 | dan__t | Unfortunately right now I think I need to figure out my iax peer issue because I cannot accept inbound calls right now. |
| 03:06.50 | [TK]D-Fender | dan__t, FFS pick a problem and FINISH fixing it! |
| 03:08.21 | dan__t | HAHA. No I just noticed I can't receive inbound calls. |
| 03:08.45 | dan__t | Although 'iax2 show peers' shows that the peer is alive |
| 03:11.53 | dan__t | Cool, got that to work. |
| 03:14.06 | *** join/#asterisk coppice (n=chatzill@234.155.17.210.dyn.pacific.net.hk) |
| 03:22.01 | mishkiz | hello all...Im trying to install an asterisk.1.4.11 at a Debian etch....when I do "make" I got "chan_sip.c: In function âhandle_request_referâ: chan_sip.c:14323: internal compiler error: Aborted". My gcc version is 4.1.2... |
| 03:22.02 | mishkiz | can anybody help me ? |
| 03:26.21 | [TK]D-Fender | mishkiz, I've googled it up and they suggest trying different compiler versions. |
| 03:26.45 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:27.04 | mishkiz | I already tried it... |
| 03:29.05 | *** join/#asterisk Nombrandue (n=Satan@ip72-198-203-23.om.om.cox.net) |
| 03:29.57 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 03:32.12 | dan__t | Well, I wasn't here the last 15 mins, my inet died. |
| 03:32.14 | dan__t | Yay |
| 03:32.37 | Nombrandue | gotta love when that happens |
| 03:33.16 | dan__t | Especially when someone is taking the time to help me out, huh |
| 03:33.18 | dan__t | *sigh* |
| 03:33.29 | Nombrandue | yeah, typical |
| 03:34.40 | [TK]D-Fender | dan__t, My karma ran over your dogma |
| 03:35.28 | dan__t | heh! |
| 03:36.23 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 03:41.12 | dan__t | Ok, even though I didn't set any reg info, I should still see the phone try to contact asterisk, right [TK]D-Fender? |
| 03:41.18 | dan__t | The only thing I changed was the server address |
| 03:42.57 | [TK]D-Fender | if you flush the reg info from your phone and you didn't add the new credentials to your phoneXX.cfg then it shouldn't be trying anything at all. |
| 03:43.07 | [TK]D-Fender | dan__t, you need to set up your registration in there... |
| 03:43.19 | dan__t | Yeah, just thought it might try with no login creds |
| 03:43.27 | [TK]D-Fender | lol |
| 03:43.28 | [TK]D-Fender | NO |
| 03:43.39 | dan__t | In like "super duper dumb mode" |
| 03:43.45 | dan__t | Just set regs, I'll restart it. |
| 03:43.48 | [TK]D-Fender | dan__t, can't log in by not logging in :p |
| 03:44.15 | dan__t | logins don't always require a host/user/pass combo, sometimes just a host. That's what I was getting at. |
| 03:44.27 | dan__t | Restarting it now |
| 03:46.03 | [TK]D-Fender | dan__t, umm.. NOPE :) |
| 03:46.22 | [TK]D-Fender | dan__t, You might not always need a pass, but you ALWAYS need a user. |
| 03:46.30 | dan__t | Cool. |
| 03:46.56 | [TK]D-Fender | dan__t, you don't see someone walk up to the security desk and say "HI, I'm here, jsut let me in!" and walk past the gates now do you? |
| 03:47.24 | dan__t | No, but when I go to the bank teller I don't necessarily speak telnet, either. |
| 03:47.24 | Nugget | telnet is eeeeeeevil! |
| 03:49.19 | Nombrandue | no it isn't... it is one of the better ways to get into a computer... when you don't have a login |
| 03:49.57 | [TK]D-Fender | Nombrandue, Stop talking to the nugget-bot :p |
| 03:50.06 | sniper[FOO] | when did you do that last time? :) |
| 03:50.14 | Nombrandue | hahaha |
| 03:51.27 | Nombrandue | don't mind me and my randomness, I am a newb to the program, and just seeing what I can pick up watching the discussions |
| 03:51.57 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-833-448.area5.spcsdns.net) |
| 03:52.27 | *** join/#asterisk bmg505 (n=leon@196.209.177.59) |
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| 03:55.34 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 03:56.10 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 03:57.05 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
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| 04:04.57 | CunningPike | Nombrandue: Well, pick up this - ssh >>>>>> telnet |
| 04:05.01 | CunningPike | ;) |
| 04:05.27 | Nombrandue | Lmfao |
| 04:06.45 | *** join/#asterisk klictel (n=klictel@modemcable159.7-200-24.mc.videotron.ca) |
| 04:09.17 | *** join/#asterisk InHisName (n=Administ@c-71-225-221-149.hsd1.pa.comcast.net) |
| 04:15.55 | osiris | ok, so broadvoice wont let you force an inbound nat proxy. anyone know how to get around this ? |
| 04:17.11 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 04:17.28 | osiris | i got outbound calling working, but get the softswitch intercept on the inbound calls |
| 04:17.45 | osiris | call cannot be completed as dialed |
| 04:19.01 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195) |
| 04:24.40 | JT | Nombrandue: no-one uses telnet to log into pcs remotely anymore |
| 04:29.05 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
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| 04:32.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 04:33.11 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 04:33.14 | InHisName | <PROTECTED> |
| 04:33.49 | [TK]D-Fender | InHisName, bad auth domain / host |
| 04:39.34 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 04:46.58 | InHisName | [TK]D-Fender, in the Proxy-Authorization: Digest, username, realm, and uri are identical to my tcpdump of the ATA registration. nonce and response are harder to tell as they are always different. |
| 04:47.36 | [TK]D-Fender | InHisName, perhaps you should pastebin a complete call attempt |
| 04:47.50 | InHisName | [TK]D-Fender, OK |
| 04:48.11 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 04:50.29 | dan__t | This is cool. |
| 04:50.34 | dan__t | I can't get the phone to boot off of a bootserver. |
| 04:51.30 | *** join/#asterisk UCFmethod (n=UCFmetho@40.152.243.24.cfl.res.rr.com) |
| 04:52.17 | sniper[FOO] | now I'm puzzled |
| 04:54.16 | sniper[FOO] | could someone post a solution where BackgroundDetect actually works with a SIP channel as the B-leg? |
| 05:02.23 | InHisName | [TK]D-Fender, http://pastebin.com/m4a3ef8ad |
| 05:03.47 | [TK]D-Fender | InHisName, Trying to fake * being a locked ATA? |
| 05:05.16 | InHisName | Its possible that is what I am trying to do? The User Agent seems to be one critical issue. I suspect there might be more somewhere. |
| 05:06.43 | [TK]D-Fender | InHisName, well I sure don't see ASTERISK in there... |
| 05:06.45 | InHisName | It worked well with Sunrocket for over a year. |
| 05:07.19 | InHisName | ASTERISK as user agent or ??? |
| 05:07.47 | [TK]D-Fender | InHisName, Sorry really can't comment on this... |
| 05:08.12 | InHisName | The * capture is before the "=======" and the Innomedia capture is after the "===" |
| 05:15.19 | dan__t | Ok, well, I had to hardcode some boot and tftp options into the phone to get it to boot properly, [TK]D-Fender. |
| 05:16.29 | [TK]D-Fender | dan__t, you are supposed to send the provisioning server info in the bootrom, and tell it to use DHCP, etc. Everything else should be in your configs |
| 05:17.05 | dan__t | I have a, uh, "delicate" DHCP setup here, I do not really want to toy with it much. Giving the phone static IP information is not a problem, I would think. |
| 05:17.19 | dan__t | But the phone is snagging the config files. |
| 05:17.28 | dan__t | And now, I actually see the phone register in * |
| 05:18.51 | [TK]D-Fender | dan__t, Thats a good thing... |
| 05:18.59 | dan__t | YEah. |
| 05:19.07 | [TK]D-Fender | dan__t, Whats so delicate about your DHCP? |
| 05:19.12 | dan__t | Inbound calls are neato. But no outbound calls are to be made. |
| 05:19.44 | dan__t | I have a few clients which do some distributed computing, they all netboot CentOS5 |
| 05:19.56 | [TK]D-Fender | dan__t, so............ |
| 05:19.59 | dan__t | I don't like DHCP much, so i choose not to touch it unless absolutely necessary. |
| 05:20.11 | dan__t | Static info works fine, unless there's something really, really wrong with it right now. |
| 05:20.27 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 05:24.13 | dan__t | Yea, outbound calls make no noise in Asterisk. |
| 05:25.53 | [TK]D-Fender | "no noise"? |
| 05:29.11 | dan__t | Nothing in the console even with debug at 100 |
| 05:30.58 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 05:33.36 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 05:33.56 | dan__t | Ok, well, now I get something: http://pastebin.ca/699223 |
| 05:35.55 | ectospasm | dan__t: what version of * |
| 05:36.08 | dan__t | 1.4.11 |
| 05:36.24 | dan__t | I read this as being a bug earlier today. |
| 05:36.40 | [TK]D-Fender | lol. |
| 05:36.43 | [TK]D-Fender | No. |
| 05:37.00 | dan__t | Well. Yeah. I did read it as that heh. |
| 05:37.16 | [TK]D-Fender | tomorrow we may be able to pick this up and you might actually decide to show me your configs :) |
| 05:37.19 | dan__t | Wow, Astricon will be here in Phoenix on the 24th? |
| 05:37.33 | dan__t | What haven't I shown you? I pasted everything. |
| 05:37.55 | dan__t | I changed the bootrom files as you had suggested. For the past hour I've just been fscking with getting the phone to actually boot. |
| 05:39.15 | [TK]D-Fender | dan__t, the only time you showed me your provisioning files, they were EMPTY. |
| 05:39.26 | [TK]D-Fender | dan__t, You did not should me how you FILLED THEM IN AFTER |
| 05:40.26 | [TK]D-Fender | dan__t, You keep doing half the job required to debug this and seem never to have pastebin actual SIP debug (which I probably wouldn't even need if I'd seen your configs) |
| 05:40.48 | ectospasm | bbwis (be back when I'm sober) |
| 05:40.49 | *** part/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net) |
| 05:41.28 | dan__t | http://pastebin.ca/699229 |
| 05:44.15 | [TK]D-Fender | dan__t, and SIP debug of a failed call.... |
| 05:44.52 | dan__t | http://pastebin.ca/699223 |
| 05:45.43 | dan__t | But that's only if I hit 'dial' on a 'missed call' list. |
| 05:46.00 | dan__t | I think that's just me not following the PolyCom dialplan though, i.e. not dialing the number properly |
| 05:46.35 | [TK]D-Fender | "SoundPoint IP" <sip:SPIP@192.168.1.3>; <---- clearly not good |
| 05:47.02 | [TK]D-Fender | And that is NOT SIP debug from CLI |
| 05:47.08 | dan__t | Yep, I was trying to find it. |
| 05:47.14 | [TK]D-Fender | SPIP = who the hell is this?! |
| 05:47.30 | dan__t | wtf I don't know, I was redialing a received call from my cell phone. |
| 05:47.32 | [TK]D-Fender | ok, I've gotta get some sleep. May catch up later |
| 05:47.40 | dan__t | Right. Thanks anyway. |
| 05:48.00 | [TK]D-Fender | go check on your phone itself that you ahve indeed fully flushed the old manual settings |
| 05:48.06 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 05:48.06 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:48.09 | [TK]D-Fender | thats it for tonight, later all |
| 06:17.22 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
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| 07:30.17 | yang | Does asterisk needs to be plugged into a telephone central, or can it standalone and link to the other existing asterisk-es? |
| 07:35.18 | lowlevel | <PROTECTED> |
| 07:37.30 | fujin | the 'average' setup is asterisk with some e1/t1 interfaces, and sip phones |
| 07:37.36 | fujin | or sip/iax softphones |
| 07:38.00 | fujin | of course the only limits are your imagination, you could do a complete sip setup with softphones and an upstream sip terminator |
| 07:52.57 | _x86_ | you can also do a complete setup without SIP, MGCP, H323, SCCP, or IAX2 at all ;) |
| 07:53.02 | _x86_ | using pure zap channels |
| 08:03.07 | coppice | actually, asterisk was never designed for VoIP. that was an afterthought |
| 08:18.44 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
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| 08:55.43 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:04.08 | *** join/#asterisk colde (n=colde@pdpc/supporter/active/colde) |
| 09:04.25 | colde | Hi, i'm getting this error: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 09:04.35 | colde | I've tried updating the clock on the server with ntpdate |
| 09:04.40 | colde | its not under heavy load |
| 09:04.48 | colde | but i still don't hear any music on hold |
| 09:04.52 | colde | any ideas? |
| 09:10.45 | hi365 | using the following, i cannot spy on any calls that are allready in progress. any idea why? |
| 09:10.46 | hi365 | exten => s-spy,1,chanspy(SIP|bw) |
| 09:13.36 | duki | Quelqu'un aurait-il utilisé ekiga sans blocage intempestif? |
| 09:14.07 | duki | sorry. |
| 09:15.08 | duki | Did someone use ekiga without freezes? I tried it under several Linux, but without success. |
| 09:16.54 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 09:18.40 | *** join/#asterisk TUplink_ (n=mythtv@c-24-126-34-203.hsd1.wv.comcast.net) |
| 09:18.50 | TUplink_ | what is the difrance in 1.2 and 1.4? |
| 09:31.33 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:34.09 | tzafrir | duki, I generally prefer twinkle on Linux |
| 09:34.22 | tzafrir | But Ekiga doesn't freeze on me |
| 09:34.32 | tzafrir | TUplink_, 0.2 |
| 09:34.51 | TUplink_ | 0.2? |
| 09:35.03 | tzafrir | <TUplink_> what is the difrance in 1.2 and 1.4? |
| 09:35.06 | TUplink_ | tzafrir your a smart ass :P |
| 09:35.47 | TUplink_ | im updating..... |
| 09:36.01 | TUplink_ | my ATA reset itself... and now in asterisk i get [Sep 16 05:32:51] WARNING[12630]: chan_sip.c:8126 check_auth: username mismatch, have <20001>, digest has <> |
| 09:36.30 | tzafrir | http://svn.digium.com/svn/asterisk/branches/1.4/UPGRADE.txt |
| 09:37.33 | duki | tzafrir: thanks, I am yet using it now, It works fine, just I need some customization. |
| 09:39.53 | *** join/#asterisk cayorde (n=flexable@host233-98-dynamic.20-87-r.retail.telecomitalia.it) |
| 09:53.13 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 10:01.10 | ThoMe | hello |
| 10:01.21 | ThoMe | is it posible recieve fax with asterisk? |
| 10:08.37 | mvanbaak | yes, with app_rxfax. This is not in the default asterisk package tho |
| 10:11.02 | ThoMe | mvanbaak: have test it with hylafax |
| 10:11.39 | mvanbaak | ah |
| 10:11.46 | ThoMe | mvanbaak: but |
| 10:11.48 | mvanbaak | hylafax+iaxmodem is a good setup |
| 10:12.22 | ThoMe | but hylafax write the tif only from "landscape format" > "panel format" :-( |
| 10:12.29 | ThoMe | mvanbaak: have u a idea - why? |
| 10:13.18 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.113) |
| 10:13.33 | mvanbaak | that's a hylafax issue I guess |
| 10:13.33 | coppice | ThoMe: what exactly do you mean by that? |
| 10:14.03 | coppice | if you mean the images look squashed, your viewer is broken |
| 10:14.18 | ThoMe | coppice: if i recieve a fax in "landscape format" then is the file, tif in ""panel format" |
| 10:14.21 | ThoMe | coppice: momento |
| 10:15.58 | ThoMe | coppice: here, un example: original: http://tm.muc.de/up/archiv/SO_Business2.pdf and my fax:http://tm.muc.de/up/archiv/fax000000007.tif |
| 10:17.13 | coppice | that looks fine. your view is broken |
| 10:17.52 | mvanbaak | looks perfect here as well |
| 10:18.27 | coppice | a large percentage of viewer get the shaper wrong, or can only display the first page |
| 10:19.34 | ThoMe | perfekt?! i can display two pages but the first and the second page is not "upright format" , its "landscape format" |
| 10:19.43 | ThoMe | and the fax it only "upright format" |
| 10:20.11 | coppice | how many more times - YOUR VIEWING SOFTWARE IS BROKEN |
| 10:20.38 | ThoMe | coppice: i have also a pdf. is acrobat also broken? ;) |
| 10:20.58 | coppice | the PDF you posted looks OK |
| 10:21.06 | mvanbaak | no, your tif->pdf convertor (prolly the same one as you use to view the tiff) is broken |
| 10:21.09 | ThoMe | yes, this is the original |
| 10:21.17 | mvanbaak | I think your tiff library is borked |
| 10:22.07 | coppice | this is the absolutely most common complaint about computer faxing. i'm not stabbing in the dark for an explanation. |
| 10:22.23 | coppice | and people can be really obnoxious when you try to help them |
| 10:22.53 | coppice | it usually turns out they have never used a computer for faxing before, but they "know" their software is OK :-) |
| 10:22.55 | mvanbaak | we get the same trouble at customer sites |
| 10:23.19 | mvanbaak | most of the time we do the tiff->pdf conversion on the faxing machine |
| 10:23.28 | mvanbaak | with our tested repository of software |
| 10:23.49 | mvanbaak | that way we know the pdf is correct. |
| 10:24.52 | coppice | its hard to know what software to recommend to people. some of the bundled stuff with windows works well, but automatic updates from MS have brought a variety of bugs are times |
| 10:25.21 | mvanbaak | yup |
| 10:25.25 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 10:25.33 | mvanbaak | that's why we convert everything to pdf on the asterisk/hylafax machine |
| 10:25.42 | mvanbaak | most pdf viewers are more or less ok |
| 10:27.22 | coppice | for FAX. they do some weird stuff with graphics, though. I was tracking down a problem on a PCB's artwork with week. It turned out to be the PDF displaying the pins of a chip in the wrong order. |
| 10:29.07 | mvanbaak | meh |
| 10:29.51 | coppice | and a lot of asian PDFs come out real funky, even when using acrobat reader to look at them |
| 10:33.52 | mvanbaak | yup |
| 10:59.35 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 11:02.28 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 11:22.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:39.50 | *** join/#asterisk semur (n=semushin@83.167.96.120) |
| 11:40.17 | semur | hi all! |
| 11:48.07 | sniper[FOO] | hi |
| 11:55.07 | matt_ | hello, i peer to voipdiscount and when i phone a landline and hangup the remote phone keeps ringing |
| 11:55.17 | matt_ | does anybody know why? |
| 11:55.45 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:56.47 | semur | can you show part of extensions.conf? |
| 11:57.14 | sniper[FOO] | matt_: it's because bad answer supervision in the pstn channels |
| 11:57.38 | matt_ | sniper[FOO], is there anything i can do about it without switching services ? |
| 11:58.57 | sniper[FOO] | a full trace would be nice |
| 11:59.39 | matt_ | sniper[FOO], how do i do that ? |
| 12:00.14 | sniper[FOO] | as semur pointed out, it's impossible to tell without the appropriate part of your extensions.conf and you could issue a 'sip set debug on' and make a call and paste the output in a pastebin, too |
| 12:01.10 | matt_ | ok, my extensions.conf just contains exten => _00.,3,Dial(SIP/${EXTEN}@voipdiscount) |
| 12:01.28 | sniper[FOO] | btw, how long does it ring after you disc'ed? |
| 12:01.37 | matt_ | i'm not sure |
| 12:01.42 | sniper[FOO] | I mean does it keep ringing |
| 12:01.55 | matt_ | it use to be 30 seconds so i put ,30 on the end of the dial line |
| 12:01.58 | sniper[FOO] | go ahead and enable sip debugging |
| 12:02.04 | matt_ | which seemed to be ok aslong as i waited for the timeout |
| 12:02.13 | matt_ | but it stopped working again after a while |
| 12:03.08 | sniper[FOO] | OK, please post the detailed trace |
| 12:03.22 | sniper[FOO] | you familiar with the cli? |
| 12:03.40 | matt_ | yea, just a min |
| 12:03.47 | matt_ | my number has a enum entry lol |
| 12:03.50 | matt_ | going to have to disable it |
| 12:06.58 | jer | i've got a weird issue that just started happening out of the blue. when making a call which goes out over an IAX2 trunk, the call goes through, the other person picks up, and about 1 - 2 seconds later, the call drops, and the person who initiated the call gets a busy congested signal, and the other party hears dead air. i'm just wondering what some possible causes of this might be? (as far as i can see, the interconnect to the other asterisk server is f |
| 12:10.47 | InHisName | <PROTECTED> |
| 12:11.33 | matt_ | sniper[FOO], http://pastebin.com/m128a873e |
| 12:12.01 | matt_ | sniper[FOO], it rang for about 30 seconds after i hung up |
| 12:13.43 | semur | InHisName, what voip provider? show part of sip.conf (if your account not turned off, problem there) |
| 12:16.07 | InHisName | semur, account works with Innomedia device, trying to get it to register with *. register => 859195776986:@voiceline.net2phone.com:5060/2679661066 |
| 12:16.53 | InHisName | semur is there a way to delete that last comment ? |
| 12:17.13 | semur | no :( |
| 12:19.19 | InHisName | It gets up to the packet to register and 407 pops out as expected with auth, * responds and it gets 403 error bad username/PIN in response. Innomedia gets 200OK instead. |
| 12:20.30 | InHisName | I have tcpdumps of both the inomedia and *. Both seem the same. http://pastebin.com/m4a3ef8ad |
| 12:20.55 | semur | what about [innomedia] (approx name) section in sip.conf? |
| 12:22.32 | InHisName | * needs a section named [innomedia] ? That word does not appear anywhere in the capture or in either my sip.conf nor extensions.conf |
| 12:23.14 | InHisName | I do have a [net2phone] and a [phoneno] sections |
| 12:24.00 | semur | it was just a guess for name :) |
| 12:24.06 | semur | show them, pls |
| 12:28.29 | semur | InHisName, sometimes one of solution for yours type troubles is to change bindport from 5060 to another... |
| 12:29.49 | InHisName | the capture from the ATA(Innomedia) had 5060. |
| 12:30.16 | InHisName | breakfast, be back later |
| 12:44.08 | hi365 | (how) can i include an extensions file that is on a remote system? |
| 12:47.22 | sniper[FOO] | hi365: you can use a switch statement in the remote * (if there is one) to share your extensions.conf |
| 12:48.54 | hi365 | sniper[FOO]: ${confused}. for example? (what im trying to do would be the equivilent of: include=> http://myserver.com/extension_to_include.conf |
| 12:51.56 | InHisName | I am looking up the [phoneno] and it is commented out. |
| 12:54.06 | sniper[FOO] | hi365: http://www.asteriskguru.com/tutorials/extensions_conf.html , section 4.2 |
| 12:54.18 | sniper[FOO] | basically the same |
| 12:54.40 | sniper[FOO] | you just have to set up an iax trunk between the * boxes |
| 12:56.17 | hi365 | sniper[FOO]: so i need to be running an asterisk server to do the include? |
| 12:56.24 | InHisName | I am back semur I need to uncomment the [phonno] to see what that might do. |
| 12:57.19 | dan__t | Colin McRae died :( |
| 12:58.12 | sniper[FOO] | dan__t, did they confirm that he was on the chopper? |
| 12:58.23 | dan__t | Guess so. |
| 12:58.32 | sniper[FOO] | RIP, Colin |
| 12:58.39 | dan__t | Most definitely. |
| 12:59.10 | sniper[FOO] | played a lot with the game that beared his name |
| 12:59.58 | dan__t | It was alright. |
| 13:00.46 | sniper[FOO] | his son, too :( |
| 13:02.05 | semur | InHisName, try to change bidport for asterisk from 5060 to another |
| 13:03.28 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 13:05.34 | *** join/#asterisk roberted (n=roberted@c-76-17-219-22.hsd1.mn.comcast.net) |
| 13:06.39 | InHisName | semur, didn't do any difference. I'll put both in http://pastebin.com/m702673f9 they are in bottom. changing portno now. |
| 13:12.03 | InHisName | semur|away, I tried 8060, but capture shows 5060 and a whole bunch of Cseq: 102 REGISTER packets. |
| 13:12.29 | InHisName | leaving for Church be back in 2-3 hrs. |
| 13:13.58 | *** join/#asterisk Corydon76-home (i=twelve@pdpc/supporter/sustaining/Corydon76-home) |
| 13:13.58 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:14.05 | *** join/#asterisk Corydon76-dig (i=orange@pdpc/supporter/sustaining/Corydon76-home) |
| 13:14.05 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:15.15 | *** join/#asterisk coppice (n=chatzill@234.155.17.210.dyn.pacific.net.hk) |
| 13:17.15 | *** join/#asterisk ttt- (n=z@dD5E0C82F.access.telenet.be) |
| 13:17.28 | *** part/#asterisk ttt- (n=z@dD5E0C82F.access.telenet.be) |
| 13:37.37 | hi365 | sniper[FOO]: are you sure that switch inclludes the dialplan localy, and dosnt just execute the extension remotly? |
| 13:37.59 | *** join/#asterisk Corydon76-home (i=orange@pdpc/supporter/sustaining/Corydon76-home) |
| 13:37.59 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:38.09 | *** join/#asterisk Corydon76-dig (i=black@pdpc/supporter/sustaining/Corydon76-home) |
| 13:38.09 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:42.54 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:43.26 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 13:47.10 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 14:01.52 | *** join/#asterisk kkn088 (n=kkn088@84.4.51.15) |
| 14:05.02 | *** join/#asterisk Corydon76-home (i=ten@pdpc/supporter/sustaining/Corydon76-home) |
| 14:05.02 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:05.21 | *** join/#asterisk Corydon76-dig (i=gold@pdpc/supporter/sustaining/Corydon76-home) |
| 14:05.21 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 14:13.58 | sniper[FOO] | is there a way to get * do answer supervision on an outbound SIP call |
| 14:14.01 | sniper[FOO] | ? |
| 14:15.18 | file | that's up to the device that finally gets the call to the PSTN |
| 14:15.38 | sniper[FOO] | file: thanks for answering |
| 14:16.12 | sniper[FOO] | the point is actually that the gateway won't do that and I'm supposed to come up we |
| 14:16.29 | file | then no, it is not possible |
| 14:16.29 | sniper[FOO] | with a method using * |
| 14:16.42 | file | you would have to do analysis of the audio to determine progress |
| 14:16.46 | sniper[FOO] | indeed |
| 14:17.15 | sniper[FOO] | but it's g.711 and I'd only have to detect for a standard UK ringing tone |
| 14:17.57 | sniper[FOO] | I've seen that dsp.c includes such routines |
| 14:18.35 | sniper[FOO] | you really consider it's impossible? |
| 14:18.59 | sniper[FOO] | consider it to be impossible |
| 14:19.11 | file | it's code, it would potentially be possible if you wrote it... but how well it would work who knows |
| 14:20.02 | hi365 | does switch=> require a trunk to be setup on the remote server? |
| 14:21.04 | sniper[FOO] | hi365: yes, an IAX2 trunk is required so the 2 * boxes can interchange information |
| 14:21.22 | semur|away | InHisName, look here: http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone |
| 14:22.38 | hi365 | sniper[FOO]: does the local server also need a trunk to the remote, or is a trunk on the remote enough? (this is what i currently have on the remote server: http://pastebin.ca/699567 ) |
| 14:24.09 | sniper[FOO] | supposed to work |
| 14:24.55 | hi365 | it doesnt :( (i dont have atrunk on the local server) |
| 14:25.19 | sniper[FOO] | what does the debug say? |
| 14:26.34 | hi365 | trunk failed |
| 14:27.07 | hi365 | this is on the local server: switch => IAX2/ipconnect:1234/192.168.0.99/ipconnect |
| 14:28.48 | sniper[FOO] | and the context you wanna import is called 'ipconnect', right? |
| 14:29.31 | hi365 | right |
| 14:31.38 | sniper[FOO] | I was wrong |
| 14:31.41 | sniper[FOO] | check this out: |
| 14:31.50 | sniper[FOO] | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 14:32.06 | hi365 | saw that befor - i kinda confused me |
| 14:32.19 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 14:32.28 | hi365 | according to that, you need iax trunks on BOTH servers |
| 14:33.07 | shido6 | yes |
| 14:33.13 | shido6 | and a timing device :) |
| 14:33.20 | hi365 | hmm |
| 14:33.35 | shido6 | whats up? |
| 14:34.20 | hi365 | what im really trying to do is have a 'debug' ivr avlible on my clients systems. i want the abbility to update the file without having to update it manualy on each client... |
| 14:34.35 | hi365 | i think switch is an overkill for that |
| 14:35.01 | riddlebox | hrmm is there no newer version of asterisk in ubuntu's repos? all |
| 14:35.05 | riddlebox | I see is 1.2.16 |
| 14:35.49 | dan__t | What's wrong with rsync? |
| 14:36.06 | dan__t | rsync between the two machines every minute? heh |
| 14:36.10 | hi365 | and then force an extension reload? |
| 14:36.25 | dan__t | ssh has an -e argument used to execute a command on a remote machine. |
| 14:36.48 | dan__t | Idunno what other options * has. I'm just approaching it as a sysadmin. |
| 14:36.53 | hi365 | how would that help? |
| 14:37.09 | hi365 | oh, i get what ur saying with the e switch |
| 14:37.27 | dan__t | I don't know if * has a 'reload' CLI option though |
| 14:37.35 | dan__t | Looks like -x will do it. |
| 14:37.41 | hi365 | so use ssh to copy the file to the remote server and then asteris relaod |
| 14:38.03 | dan__t | I'd use rsync to make sure that only the latest copy is sync'd. |
| 14:38.07 | dan__t | Or lftp |
| 14:38.15 | dan__t | yea |
| 14:38.25 | dan__t | I don't think it was designed that way. |
| 14:38.39 | dan__t | Think of what hell that would raise for installations with hundreds if not thousands of extensions. |
| 14:39.12 | dan__t | There's probably a much more elegant way of doing this. |
| 14:39.12 | hi365 | true. ssh seems simple though: http://www.slug.nf.net/past/SSH/html/slide_5.html |
| 14:39.16 | dan__t | But that's what I would suggest. |
| 14:39.22 | dan__t | Yeah, ssh is kinda bad-ass that way. |
| 14:39.30 | hi365 | great. ill get started |
| 14:39.35 | dan__t | Be careful though. |
| 14:39.41 | dan__t | Use only one machine to edit configs on. |
| 14:39.55 | sniper[FOO] | hi365: dan__t's suggestion is more appropriate than mine |
| 14:39.56 | hi365 | brb |
| 14:39.58 | dan__t | If you overwrite one server that has new changes on it from a server that has old changes, that would suck. |
| 14:40.17 | hi365 | sniper[FOO]: thanks for being a really man! (admiting you mistakes) |
| 14:40.20 | hi365 | :} |
| 14:40.24 | hi365 | :-} |
| 14:40.24 | dan__t | heh! |
| 14:40.32 | hi365 | really=real |
| 14:40.46 | dan__t | example: asterisk -rx "core show channels" - Display channels on running server |
| 14:42.37 | dan__t | *sihg* Gotta get ready to head to Tucson. |
| 14:44.23 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:48.03 | hi365 | dan__t: the only shortcomming to the "push" method id that i need to have a list of all the usernames/passwords of my remote servers |
| 14:48.26 | hi365 | or the remote servers need to have my user/pass (better, but not great) |
| 14:49.01 | dan__t | Use keys. |
| 14:49.23 | dan__t | http://www.gatsby.ucl.ac.uk/~iam23/compnotes/passwordless_ssh.html |
| 14:49.31 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 14:49.49 | hi365 | but if the end user has access to their pbx, they will be able to access my server usin the key... |
| 14:50.21 | dan__t | Not true. |
| 14:50.28 | dan__t | Well, sure, they could - if you authorize it. |
| 14:51.18 | hi365 | if they have access to the pbx, can they look at get_ivr.sh and see the ssh key there? |
| 14:51.38 | dan__t | Read that article. |
| 14:51.40 | dan__t | And others like it. |
| 14:51.44 | hi365 | ok |
| 14:51.44 | dan__t | YOu'll see what I'm talking about. |
| 14:51.55 | dan__t | I need to hop in the shower. Good luck. |
| 14:53.56 | yang | I am curious if I can enable asterisk to work without the connection to PSTN telephony central ? And where could I get asterisk peers to test my connection out? |
| 14:55.10 | hypa7ia | yang: you don't need a PSTN connection at all |
| 14:55.21 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 14:55.39 | *** part/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 14:55.43 | rob0 | IAXtel, SIPphone, FWD ... or just direct *-to-* connections |
| 14:56.21 | rob0 | but, it's a whole lot more useful when you tie into PSTN somehow |
| 14:56.22 | yang | hypa7ia: ok what about the testing peers to be able to test |
| 15:00.59 | *** join/#asterisk kkn088 (n=kkn088@84.4.51.15) |
| 15:01.35 | hypa7ia | yang: what rob0 just said |
| 15:02.03 | hypa7ia | yang: you can also get cheap pay-as-you-go PSTN termination over SIP and IAX |
| 15:02.11 | hypa7ia | have a look around the voip-info wiki |
| 15:05.36 | rob0 | Softphones might also be used for testing, altho they generally suck for real use. |
| 15:11.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:16.42 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 15:27.16 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 15:27.39 | iPod-nano | My D-Link won't keep its connection to my Asterisk box. |
| 15:29.40 | iPod-nano | I'll restart it, it'll connect and I can call it/make calls with it, but after a couple minutes it can't connect. |
| 15:31.03 | hypa7ia | iPod-nano: that's very odd |
| 15:31.24 | hypa7ia | is there an IP address conflict on your network? |
| 15:36.35 | *** join/#asterisk ManxPower (n=manxpowe@241.sub-75-202-17.myvzw.com) |
| 15:41.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:41.24 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:41.41 | duki | hello, |
| 15:41.58 | duki | I am using asterisk branch 1.4, |
| 15:42.12 | duki | moh works just fine with mode=files, |
| 15:42.50 | duki | but when trying to use a streaming , there is the classic message , |
| 15:42.53 | duki | <PROTECTED> |
| 15:43.32 | duki | the stream server (icecast) is working and I use it with mplayer to listen to the stream. |
| 15:43.45 | *** join/#asterisk saftsack (n=saftsack@pD9E07B28.dip.t-dialin.net) |
| 15:43.58 | hi365 | i would like to host an extensions.conf file on the net so that my remote server could download it. can someone recommend a free service that could do this? |
| 15:44.01 | duki | I use an example from voip-info to use to the icecast server, |
| 15:45.08 | duki | I used this command line (in a script): |
| 15:45.41 | duki | /usr/bin/ogg123 -q -b 128 -p 32 -d wav -f - http://bagdad:8000/misc.ogg |sox -r 44100 -t wav - -r 8000 -c 1 -t raw - vol 0.10 |
| 15:46.07 | duki | the script is launched from musiconhold.conf with: |
| 15:46.27 | duki | application=/etc/asterisk/mohstream.sh |
| 15:47.46 | duki | the script contain just the command line above. |
| 15:48.03 | hypa7ia | hi365: give the remote server access to the machine with an ssh key, and just script it to download the script over scp |
| 15:48.30 | hi365 | hypa7ia: im a bit wary of leaving my ssh key on all my clients servers |
| 15:49.06 | sniper[FOO] | hi365: restrict that user to a specific directory containing only the extensions.conf |
| 15:50.03 | hi365 | i just feel that a remote host is: a. more secure and b. more reliable as my adsl tends to flake out every so often |
| 15:51.05 | hi365 | shame googleeeee docs has more than just text in their published docs (html + a ton of java) |
| 15:51.14 | hi365 | googleeee=google |
| 15:52.15 | hypa7ia | hi365: use jailshell to completely lock off that account |
| 15:52.20 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:53.09 | hypa7ia | hi365: if you are worried about a flaky connection, what about a cheap VM somewhere like unixshell or quantact? |
| 15:53.40 | hi365 | that would work, im just hoping for something free |
| 15:54.34 | sniper[FOO] | hi365: maybe you should try a free hosting service, upload the extensions.conf and wget it to the clients' servers at regular intervals |
| 15:54.39 | hypa7ia | hi365: are you willing to trust your customer's info with a free service? |
| 15:55.00 | hi365 | its not the customers info - its just a gineric debug ivr |
| 15:55.15 | hi365 | sniper[FOO]: thats what ide like to do - im lookig for a host. know of anyone? |
| 15:55.35 | hypa7ia | dreamhost with a coupon :p |
| 15:56.02 | hypa7ia | googlepages? |
| 15:56.03 | hypa7ia | hehe |
| 15:56.24 | hi365 | googlepages only lets u use their templaates (alot of html and graphics) |
| 15:56.31 | hi365 | got a cupon for them? |
| 15:56.49 | sniper[FOO] | hi365: pick one: http://www.google.com/Top/Computers/Internet/Web_Design_and_Development/Hosting/Free/Personal/ |
| 15:56.54 | volker__ | hi365: i germany u can get for around 7-9eur a good linux vserver u can let asterisk running in (and whatever u want, too) |
| 15:57.09 | hypa7ia | volker__: quantact is $10 usd for a vm |
| 15:57.28 | hypa7ia | hi365: 777 should still work |
| 15:57.56 | iPod-nano | hypa7ia, sorry for the delay. I have a peer-to-peer network, only three devices are on it, and they all have their own IP address. |
| 15:58.11 | ManxPower | duki: Do you have a digium card or ztdummy loaded? |
| 15:59.17 | hypa7ia | iPod-nano: what do you mean by peer-to-peer? as in ad-hoc wireless? |
| 15:59.18 | volker__ | hypa7ia: the bandwhich dont look great there. ok, here in .de we have only one provider which save ur money when u get a ddos |
| 16:00.02 | iPod-nano | hypa7ia, no. Everything is connected together via ethernet. |
| 16:00.16 | hi365 | most of the providers want to show ads :( having to strip thoes out is going to be a pita |
| 16:00.27 | iPod-nano | The adapter, the Asterisk box, and my laptop are all connected to a hub. |
| 16:00.28 | hypa7ia | volker__: hi365 needs to move a single text file. i don't think bandwidth is an issue |
| 16:00.30 | rob0 | hi365: not YOUR key, you generate a special key and dedicated account for the purpose. |
| 16:00.36 | hypa7ia | iPod-nano: hub or switch? |
| 16:00.40 | dlynes_laptop | Has anyone been able to get more than 3 lines working on the Aastra 9133's with recent firmware, boot roms, and hardware? |
| 16:00.52 | iPod-nano | Hub |
| 16:00.56 | volker__ | hypa7ia: yeah, i had it beside this fact |
| 16:01.05 | hypa7ia | iPod-nano: seriously? |
| 16:01.17 | iPod-nano | Yes. Haha. |
| 16:01.24 | iPod-nano | A ten-megabit hub, too. |
| 16:01.26 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:01.29 | hypa7ia | ok... um... try replacing that |
| 16:01.30 | sniper[FOO] | makes it easier to debug |
| 16:01.31 | iPod-nano | It has a BNC connector. |
| 16:01.36 | sniper[FOO] | LOL |
| 16:01.39 | hypa7ia | o_0 |
| 16:02.08 | sniper[FOO] | you don't have to set your NIC to promisc mode |
| 16:02.13 | iPod-nano | My laptop communicates with it just fine, though |
| 16:02.18 | volker__ | iPod-nano brings me to the question, how well does asterisk perform on some old suns (like ultra1 or ultra5), ibms rs6k etc. |
| 16:02.21 | hypa7ia | your laptop isn't doing voip |
| 16:02.31 | hypa7ia | presumably |
| 16:02.44 | iPod-nano | I have a SIP client on my laptop. |
| 16:02.48 | sniper[FOO] | some linux drivers don't really handle promiscuous mode well |
| 16:03.05 | hypa7ia | sniper[FOO]: would that explain the "working then not working" scenario? |
| 16:03.31 | hypa7ia | volker__: remember that asterisk can work on a linksys router |
| 16:03.34 | iPod-nano | volker__, I have a toy network, basically. I'm running Asterisk on Debian linux on an old, old Compaq. |
| 16:03.46 | hypa7ia | volker__: if you can get it to compile it will probably run fine |
| 16:03.49 | dlynes_laptop | volker__: i've had openpbx/callweaver (which is a fork of asterisk) running on a Netra T1 quite well |
| 16:04.11 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.113) |
| 16:04.21 | hypa7ia | iPod-nano: have you looked at the asterisk logs? |
| 16:04.24 | volker__ | hypa7ia: yeah, right. but never know how well. i mean ultra1 is just 14xMHz RISC, too |
| 16:04.40 | hypa7ia | yeah that might be pushing it |
| 16:04.46 | *** join/#asterisk ficeto (n=ficeto@mac.vdnsbg.com) |
| 16:04.46 | sniper[FOO] | nope, it's completely unrelated, a hub or a switch with port mirroring caps will just make your life easier, when, i.e. you have a NIC with checksum offloading and partially complete drivers for linux |
| 16:04.57 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 16:05.05 | ficeto | guys |
| 16:05.21 | volker__ | dlynes_laptop: yeah, netra should have enough power, but i'm not that sure about older models |
| 16:05.23 | duki | ManxPower: No, I haven't any digium card installed. Ttrying to resolve this problem , I installed the zaptel-1.4.5.1 and tried to load the ztdummy module, but it won't to load. |
| 16:05.27 | ficeto | anybody knows why i was not able to playback audio on Xeon server machine |
| 16:05.40 | sniper[FOO] | where partially complete effectively means broken and 'can be tweaked to be usable' |
| 16:05.41 | ManxPower | ficeto: do you have any zaptel drivers loaded? |
| 16:05.47 | dlynes_laptop | volker__: ah...didn't know how the power of a netra t1 compared to an ultra 5...i know it's quite a bit faster than an ultra 2 though |
| 16:05.48 | ficeto | ztdummy |
| 16:05.53 | hypa7ia | sniper[FOO]: lol |
| 16:06.00 | ManxPower | ficeto: if you unload ztdummy does audio work? |
| 16:06.01 | duki | ManxPower: here are the message error : modprobe ztdummy: |
| 16:06.07 | ficeto | nope |
| 16:06.12 | volker__ | dlynes_laptop: ultra5 is around 300mhz (lower/faster depending on the model) |
| 16:06.16 | duki | ManxPower: WARNING: Error inserting rtc (/lib/modules/2.6.22-ARCH/kernel/drivers/char/rtc.ko): Input/output error |
| 16:06.19 | duki | FATAL: Error inserting ztdummy (/lib/modules/2.6.22-ARCH/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 16:06.21 | dlynes_laptop | volker__: i've got a couple of ultra 2's, a sunblade 100 and a netra t1 |
| 16:06.27 | hypa7ia | duki: pastebin is your friend |
| 16:06.30 | volker__ | u5 could be faster as an u10 for example |
| 16:06.37 | dlynes_laptop | volker__: oh yeah...btw...one other thing....it's nice and fast on a Sunfire v250, too |
| 16:06.39 | hypa7ia | ~pastebin |
| 16:06.39 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 16:06.45 | duki | hypa7ia: Ok, sorry. |
| 16:06.47 | ManxPower | duki: request to schedule in past is generally a harmless message. |
| 16:06.51 | volker__ | isnt sunfire x86 technology? |
| 16:06.52 | hypa7ia | duki: no problemo :) |
| 16:07.00 | ManxPower | ficeto: I think there is much you are not telling us. |
| 16:07.01 | dlynes_laptop | volker__: depends on which one you're looking at |
| 16:07.14 | dlynes_laptop | volker__: the one i've got is an ultrasparc |
| 16:07.29 | ficeto | well it's a debian 4.0 r0 distro |
| 16:07.35 | ManxPower | duki: what verison of asterisk, what verison os zaptel |
| 16:07.36 | dlynes_laptop | volker__: most of them are ultrasparc...a few select models are AMD-based |
| 16:07.36 | volker__ | fire is a few years older as the ultra-line |
| 16:07.40 | ficeto | just asterisk latest |
| 16:07.48 | ficeto | compiled from source |
| 16:07.49 | ManxPower | ficeto: How is the call getting in to Asterisk? |
| 16:07.58 | volker__ | i'm more interested how low u can go with suns etc |
| 16:08.05 | ficeto | everything works fine with calls |
| 16:08.19 | ficeto | only playback audio is a bproblem |
| 16:08.30 | ManxPower | ficeto: I won't ask again. How are calls getting into the system. |
| 16:08.35 | ficeto | i even tried differen sound encodings |
| 16:08.38 | ManxPower | You can't have audio playback without calling the server. |
| 16:08.52 | ficeto | i call from internal extension |
| 16:08.57 | ficeto | or from other server |
| 16:09.06 | ficeto | does not matter the result is the same |
| 16:09.07 | ManxPower | ficeto: Using SIP, IAX, MGCP, SCCP, H323, or zaptel card? |
| 16:09.13 | ficeto | SIP |
| 16:09.23 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 16:09.24 | ManxPower | that was like pulling a tooth. |
| 16:09.30 | ficeto | sorry |
| 16:09.35 | volker__ | ok. gtg. bye |
| 16:09.39 | ficeto | i'll try better for next time |
| 16:09.45 | ManxPower | ficeto: put the CLI output of a failed call on pastebin.ca |
| 16:09.48 | ManxPower | ~pb |
| 16:09.49 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:09.54 | duki | ManxPower: all are the latest (but not svn), Installed yesterday :): asterisk-1.4.11.tar.gz |
| 16:10.01 | duki | ManxPower: asterisk-addons-1.4.2.tar.gz |
| 16:10.08 | duki | ManxPower: zaptel-1.4.5.1.tar.gz |
| 16:10.27 | ManxPower | duki: good. Just making sure you didn't have some weird version mismatch. |
| 16:10.39 | ficeto | nothing in the CLI or in log files shows problem but one thing |
| 16:10.46 | ficeto | it hangs on the playback |
| 16:11.07 | ficeto | sais playingback ss-noservice and stays there |
| 16:11.08 | ManxPower | duki: request to schedule in past is because the system is not fast enough to run some audio in real time. A Timing source will help. |
| 16:11.35 | ManxPower | ficeto: I can't help you further. |
| 16:11.47 | ficeto | thanks |
| 16:12.10 | *** join/#asterisk jtexter3 (n=jtexter3@12.159.220.114) |
| 16:12.26 | duki | ManxPower: Yes you are right. I am running asterisk, icecast and ices2, with some others services on an celeron 2Ghz. |
| 16:12.52 | duki | ManxPower: and sox to convert :( |
| 16:13.01 | ManxPower | the ice stuff will take up lots of CPU. Generally you don't want to run other services on an Asterisk box. |
| 16:13.17 | ManxPower | duki: chances you can improve things if you can get ztdummy to run. |
| 16:13.18 | jtexter3 | Anyone know how to make Polycom 301's go into headset mode on auto answer after a reboot? Seems you have to press the headset button the first time |
| 16:14.53 | duki | ManxPower: I shall try it. and an other solution, it to run all this on my laptop (amd64 x2 1,6 GHZ). but this need that I install all from zero. |
| 16:15.33 | ManxPower | what is zero? |
| 16:16.17 | duki | ManxPower: sorry for my bad english, in french this means from scratch :) |
| 16:16.51 | ManxPower | Ah. |
| 16:16.53 | ManxPower | OK. |
| 16:17.17 | duki | ManxPower: thank you very much for help :) |
| 16:17.36 | ManxPower | duki: ztdummy uses either USB or kernel RTC to schedule audio timing |
| 16:18.05 | ManxPower | you almost always want to compile everything from the source code. |
| 16:18.18 | ficeto | http://pastebin.com/m57d3a1c2 there it is |
| 16:19.02 | ManxPower | ficeto: pastebin the output of "lsmod" |
| 16:19.20 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 16:20.28 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 16:20.36 | ficeto | http://pastebin.com/d2ce0e793 done |
| 16:20.43 | duki | I am using kernel 2.6.22, so noramlly no need fo any digium hardware card, even ztdummy, and the rtc kernel module is yet loaded. My machine seems to not support the load. |
| 16:20.51 | stefmtl | with trunk zaptel, when I try to insmod, I get: Unknown symbol _GLOBAL_OFFSET_TABLE_ |
| 16:21.31 | ManxPower | ficeto: do an "rmmod ztdummy", stop and start asterisk and try the call again. |
| 16:21.58 | ManxPower | duki: only ztdummy can use the timing, not Asterisk. |
| 16:22.44 | duki | ManxPower: Ok, I understand now, I shall really try to load it in the kernel. |
| 16:23.11 | ManxPower | stefmtl: did you compile zaptel for the current kernel? |
| 16:23.24 | ManxPower | stefmtl: also, don't expect trunk to work. |
| 16:23.35 | ficeto | it worked |
| 16:23.39 | ManxPower | stefmtl: you should report it as a bug on bugs.digium.com |
| 16:23.49 | ficeto | any idea how to have timing and to work at the same time |
| 16:23.54 | ManxPower | ficeto: there is some issue with ztdummy loading, but not providing any timing. |
| 16:24.10 | ManxPower | ficeto: What is the reason for needing ztdummy? |
| 16:24.14 | stefmtl | ManxPower : compiled with 2.6.18 |
| 16:24.28 | ficeto | i thought you need it for conferences |
| 16:24.43 | ManxPower | stefmtl: that does not tell me anything unless you also tell me what the CURRENT kernel is. |
| 16:24.51 | ficeto | btw the same setup on a pentiumD machine works fine with ztdummy |
| 16:24.52 | ManxPower | ficeto: that is a good reason. |
| 16:25.11 | ficeto | same kernels and versions packets etc |
| 16:25.14 | ManxPower | ficeto: your problem is a hardware compatibility issue. |
| 16:25.25 | ManxPower | it has nothing to do with the version or your configuration |
| 16:25.31 | tzafrir | duki, you use the default Etch kernel? |
| 16:25.38 | ficeto | thanks |
| 16:25.55 | stefmtl | ManxPower : is that what you ask : Linux heberge 2.6.18-5-686 #1 SMP ? |
| 16:25.59 | ManxPower | ficeto: If you are not using the latest versions then upgrade and try it again, if you are using the current versions, then file a bug on bugs.digium.com |
| 16:26.35 | ManxPower | stefmtl: All I asked is "Was zaptel built using the same kernel that you are currently running." |
| 16:26.36 | stefmtl | ManxPower : it is the current kernel I use in my distrib |
| 16:26.37 | ficeto | ok, will do thanks alot for your help (all current versions) |
| 16:26.51 | stefmtl | ManxPower : yes, exactly the same |
| 16:27.08 | duki | tzafrir: I am under archlinux kernel 2.6.22-ARCH (without any customization). |
| 16:27.20 | ManxPower | stefmtl: make sure you are using the latest versions of everything, then file a bug on bugs.digium.com |
| 16:27.24 | duki | tzafrir: archlinux 0.8 |
| 16:27.33 | stefmtl | ManxPower : ok thanks |
| 16:27.40 | tzafrir | duki, if you have 2.6.22, get zaptel (or at least ztdummy.c) from zaptel svn |
| 16:27.44 | iPod-nano | Haha. Monkeys. |
| 16:27.45 | tzafrir | branches/1.4 |
| 16:28.02 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:28.17 | Wonka | .oO( or use mISDN... ) |
| 16:28.41 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 16:28.42 | tzafrir | misdn as a timing source for Asterisk? |
| 16:29.03 | markit | anyone here running asterisk with Soekris hardware? |
| 16:29.16 | iPod-nano | Monkeys. |
| 16:29.24 | ManxPower | duki: if you were on the mailinglists, you would have seen this issue in the past 3 days. |
| 16:29.30 | ManxPower | the report, the discussion, and the fix. |
| 16:31.01 | duki | ManxPower: I didn't know, I just installed asterisk yersteday. I shall search there for this issue and fix. |
| 16:31.53 | ManxPower | the fix is what tzafrir said |
| 16:32.51 | duki | ManxPower: ok so I shall download the ztdummy.c (because installing zaptel 1.4.5 even from source doesn't fix the problem). |
| 16:33.18 | duki | thanks again tzafrir ManxPower . |
| 16:33.25 | tzafrir | http://svn.digium.com/svn/asterisk/branches/1.4/ztdummy.c |
| 16:33.31 | tzafrir | wget it |
| 16:35.15 | duki | tzafrir: just one thing please, what to do with ztdummy.c, I suppose compile it, but where to put it? Or I replace the one existing in zaptel and compile it? |
| 16:35.55 | tzafrir | re-run 'make' / 'make install' as usual |
| 16:36.16 | duki | tzafrir: yes, correct. Thank you. |
| 16:45.49 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 16:49.29 | *** join/#asterisk Mmurdock (n=Mmurdock@9.sub-72-121-71.myvzw.com) |
| 16:51.37 | *** join/#asterisk {Sean} (n=sean@cust118.vc801.mdunetwork.com) |
| 16:51.45 | {Sean} | hey |
| 16:51.55 | {Sean} | i just upgraded from 1.2 to 1.4 |
| 16:52.01 | {Sean} | now my SIP clients get SIP/2.0 401 Unauthorized |
| 16:52.08 | {Sean} | any ideas what could cause it? |
| 16:52.30 | jwh | blindly updating without checking the huge amounts of config syntax changes :p |
| 16:52.44 | {Sean} | hahah |
| 16:52.45 | ManxPower | {Sean}: You didn't find anything that might relate to your problem when you read upgrade.txt (or whatever they call it in 1.4)? |
| 16:53.02 | {Sean} | not directly |
| 16:53.04 | ManxPower | you should also look at the 1.2 UPGRADE.txt as well, of course. |
| 16:53.10 | {Sean} | is there some sort of ACL on SIP now? |
| 16:54.31 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net) |
| 16:55.01 | ManxPower | {Sean}: not that I've run into. |
| 16:55.38 | ManxPower | {Sean}: you understand that ALL SIP dialogs start with a 401 Unauthorized, right? |
| 16:55.53 | mvanbaak | I just found a very usefull idea to use asterisk |
| 16:56.06 | mvanbaak | create a text2speech gateway for my blogtool |
| 16:56.09 | {Sean} | ah -- i didn't --its been a while since i've had to debug sip |
| 16:56.17 | mvanbaak | so visually impared ppl can still read my blog using a phone |
| 16:56.43 | ManxPower | I can't imagine why ANYONE would want to read any blog, but maybe I'm just old. |
| 16:56.51 | mvanbaak | lol ManxPower |
| 16:57.27 | WilliamK | Manx, are you as old as mr. cerf? |
| 16:57.29 | WilliamK | :) |
| 16:57.53 | ManxPower | No, not THAT old. |
| 16:58.51 | mvanbaak | well, I thought it was a good idea |
| 16:58.54 | mvanbaak | *snif* |
| 16:58.57 | WilliamK | just had to ask :) |
| 16:59.26 | *** join/#asterisk wahjava (n=wahjava@unaffiliated/wahjava) |
| 16:59.48 | wahjava | hi channel |
| 17:00.19 | WilliamK | sorry no channel drivers here... |
| 17:00.20 | wahjava | which version of asterisk to use for new installation ? 1.2 or 1.4 ? what are the differences ? |
| 17:00.21 | WilliamK | :) |
| 17:00.28 | rudholm | 1.4 |
| 17:00.39 | ectospasm | wahjava: see topic |
| 17:00.55 | wahjava | rudholm: thanks |
| 17:00.57 | wahjava | ectospasm: sorry |
| 17:01.04 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:01.19 | rudholm | look what the Cat5 dragged in |
| 17:01.39 | mvanbaak | a Strom_M |
| 17:01.43 | rudholm | yup |
| 17:02.03 | rudholm | hmm, it's not moving |
| 17:02.13 | rudholm | must have killed it before dragging it in |
| 17:03.16 | Strom_M | help help i'm being attacked by a 12" floppy disk |
| 17:03.34 | rudholm | it's ok, it's just a New Order record. |
| 17:03.38 | rudholm | it's vegan |
| 17:03.40 | Qwell | a 12" what? |
| 17:03.55 | WilliamK | must be made by micro-soft |
| 17:03.57 | WilliamK | :) |
| 17:04.05 | Strom_M | Qwell: bluemondayownersclub.com |
| 17:04.11 | Qwell | O.o |
| 17:04.23 | Qwell | ahh |
| 17:04.35 | Qwell | why? |
| 17:07.56 | *** join/#asterisk [s]Animat (n=info@d220-238-210-46.dsl.vic.optusnet.com.au) |
| 17:08.32 | [s]Animat | what about asteriskwin32 ? :( |
| 17:08.51 | [s]Animat | (referring to topic) |
| 17:08.54 | Strom_M | asteriskwin32 is a bucket of lol. |
| 17:09.08 | [s]Animat | Strom_M: Is it really? Why? |
| 17:09.17 | rudholm | wait, someone ported asterisk to Windows? |
| 17:09.32 | [s]Animat | rudholm: Yeh. |
| 17:09.40 | mvanbaak | asterisk32.dll |
| 17:09.41 | mvanbaak | ;) |
| 17:09.41 | ManxPower | I don't see anything in the /topic about AstWin32 |
| 17:09.52 | [s]Animat | ManxPower: Exactly. |
| 17:10.05 | Qwell | astwin32 != asterisk |
| 17:10.16 | mvanbaak | net stop asterisk |
| 17:10.19 | mvanbaak | net start asterisk |
| 17:10.33 | Qwell | why restart asterisk when you can reboot windows?! |
| 17:10.49 | mvanbaak | hahahaha |
| 17:10.50 | *** join/#asterisk Mmurdock (n=vnjyjta@9.sub-72-121-71.myvzw.com) |
| 17:10.52 | [s]Animat | lol... but it's so convenient. |
| 17:10.57 | Qwell | uhh...how? |
| 17:11.08 | Qwell | don't you still need cygwin? |
| 17:11.16 | [s]Animat | because I don't want to have to keep rebooting to switch OSes |
| 17:11.20 | [s]Animat | Qwell: yeh |
| 17:11.21 | Qwell | ... |
| 17:11.25 | Qwell | IT'S WINDOWS |
| 17:11.32 | Qwell | You're going to be rebooting daily anyways |
| 17:11.35 | E-bola | emulate it |
| 17:11.41 | E-bola | if u need to run windows as host os |
| 17:11.42 | mvanbaak | run vmware-server |
| 17:11.47 | E-bola | just run asterisk in vmware or similar |
| 17:11.57 | Qwell | or run windows in vmware... |
| 17:12.00 | [s]Animat | I'd like to. Don't know how though. |
| 17:12.06 | mvanbaak | or dont run windows at all |
| 17:12.06 | [s]Animat | Best google query = ? |
| 17:12.09 | E-bola | there is nothing to know |
| 17:12.10 | Qwell | mvanbaak++ |
| 17:12.20 | E-bola | its easier than installing linux on a real computer |
| 17:12.23 | Qwell | install vmware, install windows in vmware |
| 17:12.33 | mvanbaak | why would you need windows anywayz ? |
| 17:12.53 | E-bola | windows only applications? |
| 17:12.54 | [s]Animat | mvanbaak: Because most of the programs I use are native to windows |
| 17:13.02 | mvanbaak | 5 years ago my windows machine here died, and never missed it |
| 17:13.03 | E-bola | obvious answer... |
| 17:13.17 | *** part/#asterisk ficeto (n=ficeto@mac.vdnsbg.com) |
| 17:13.18 | Qwell | name one windows program where there isn't an open source alternative :) |
| 17:13.30 | Strom_M | adobe illustrator |
| 17:13.33 | E-bola | Qwell: why would u replace something that works with an alternative? |
| 17:13.33 | Qwell | gimp |
| 17:13.33 | E-bola | lol |
| 17:13.34 | Qwell | :P |
| 17:13.40 | Strom_M | no |
| 17:13.46 | mvanbaak | Strom_M: gimp, inkscape, pixel |
| 17:13.49 | Qwell | E-bola: you sure have a messed up definition of "works" |
| 17:13.50 | [s]Animat | Qwell: I'm sure there is. How long would it take to adapt ALL of my files and systems in place to the new applications? |
| 17:13.54 | E-bola | there's a billion programs that doesnt work in linux |
| 17:14.04 | E-bola | i cant think of a single reason to use linux on a desktop |
| 17:14.05 | Qwell | Strom_M: no, I was calling you a gimp |
| 17:14.12 | Strom_M | ah |
| 17:14.14 | Qwell | :D |
| 17:14.17 | Strom_M | heheh |
| 17:14.28 | mvanbaak | E-bola: less frustration :) |
| 17:14.35 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 17:14.36 | [s]Animat | mvanbaak: Inkscape is cool. |
| 17:14.38 | E-bola | windows xp is so braindead dummy simple |
| 17:14.41 | E-bola | it always works |
| 17:14.46 | Qwell | E-bola: except when it doesn't |
| 17:14.50 | Qwell | which is pretty often |
| 17:14.51 | mvanbaak | it always coredumps |
| 17:15.00 | E-bola | comparing it to any linux desktop, is like comparing a bicycle to a car |
| 17:15.12 | mvanbaak | yeah, being linux the car |
| 17:15.15 | E-bola | yes |
| 17:15.18 | [s]Animat | exactly |
| 17:15.23 | E-bola | and think about why u need a drivers license for a cvar |
| 17:15.25 | E-bola | and not a bike |
| 17:15.29 | E-bola | and we dont have to argue anymore :) |
| 17:15.29 | [s]Animat | complex. |
| 17:15.46 | Qwell | because cars get you where you need to be much faster |
| 17:15.53 | E-bola | sure |
| 17:15.56 | E-bola | but if u dont got a license |
| 17:15.58 | Qwell | and safer |
| 17:15.59 | E-bola | they dont get u anywhere |
| 17:16.00 | E-bola | :) |
| 17:16.13 | [s]Animat | Qwell: Safer, eh? |
| 17:16.17 | mvanbaak | E-bola: only icompetent ppl are not able to get the license |
| 17:16.19 | Qwell | safer than a bike? yes |
| 17:16.27 | mvanbaak | a bike is really unsafe |
| 17:16.29 | [s]Animat | sif cars are safer than bikes |
| 17:16.32 | E-bola | mvanbaak: nonesence |
| 17:16.33 | [s]Animat | statistically |
| 17:17.04 | Qwell | an accident on a bike at any speed is going to mess you up pretty good |
| 17:17.05 | [s]Animat | i'd bet my left nut that statistically, bikes are safer than cars |
| 17:17.13 | Qwell | cars have seatbelts, airbags, traction control, etc |
| 17:18.08 | E-bola | im pretty sure a car is the most insecure way to transport ur self |
| 17:18.14 | E-bola | but thats not really the point |
| 17:18.24 | E-bola | the point is the vast majority of users have no desire nor need to switch to linux |
| 17:18.39 | mvanbaak | that's because most ppl are cueless |
| 17:18.41 | E-bola | I've been running linux servers for 7 years, i couldnt dream of changing my windows laptop to running linux |
| 17:18.49 | mvanbaak | and that's not bad, but it's the truth |
| 17:19.01 | E-bola | mvanbaak: Your the clueless one, if you think linux on desktop is great atm. |
| 17:19.10 | mvanbaak | gheh |
| 17:19.59 | [s]Animat | Qwell: Cars - can easily cause your body's Momentum => 2187.5kgms^-1 and impose a force of 4375N on your body... (from 100kmh^-1 to 0kmh^1 in 0.5 secs) |
| 17:20.46 | ManxPower | [s]Animat: and amazing amount of damage if that car hits a bicycle. |
| 17:20.54 | Qwell | heh |
| 17:21.02 | [s]Animat | ManxPower: Yeah, if it was from the oposite direction |
| 17:21.11 | mvanbaak | even if from the same direction |
| 17:21.24 | [s]Animat | ManxPower: And the rider wouldn't come to a stop in 0.5 sec because they're not trapped in a vessel |
| 17:21.24 | mvanbaak | I've never seen someone go 100kmh on a bike |
| 17:21.30 | mvanbaak | not in normal conditions anyway |
| 17:21.36 | Qwell | no, they'd just fly 50 yards |
| 17:21.42 | [s]Animat | Qwell: Exactly |
| 17:21.43 | mvanbaak | indeed |
| 17:21.48 | Qwell | ...and die |
| 17:21.51 | [s]Animat | that actually decreases the force on their body |
| 17:21.53 | mvanbaak | probably |
| 17:22.02 | Qwell | yeah, trauma to your body is much worse than DEATH |
| 17:22.03 | [s]Animat | Qwell: Have you seen a motorcyclist come off? |
| 17:22.18 | [s]Animat | at speeds > 200kmh^-1 |
| 17:22.23 | E-bola | :) |
| 17:22.30 | mvanbaak | lol E-bola |
| 17:22.36 | [s]Animat | I really want my left nut |
| 17:22.45 | Qwell | we aren't talking about theoretical vehicles? |
| 17:22.57 | mvanbaak | Qwell: do you know wether asterisk will run on haiku ? |
| 17:23.04 | Qwell | never heard of it |
| 17:23.15 | E-bola | i doubt it |
| 17:23.43 | E-bola | Do anybody use a gui with Asterisk? |
| 17:23.49 | E-bola | so far ive tried asterisk-gui and freepbx |
| 17:23.58 | [s]Animat | I would if i had it installed on linux |
| 17:23.59 | mvanbaak | E-bola: yeah, vim |
| 17:24.00 | E-bola | both of them didnt allow me the same detail confs as i can do with manualy setting stuff up |
| 17:24.50 | mvanbaak | vim is the best gui for asterisk |
| 17:25.16 | duki | mvanbaak: tzafrir , all works fine here : |
| 17:25.29 | duki | 1 donload ztdummy from svn |
| 17:26.16 | duki | 2 rmmode some rtc_* modules because rtc couldn't be loaded |
| 17:26.29 | duki | 3 load rtc.ko |
| 17:26.32 | ManxPower | There are even DEDICATED CHANNELS for Asterisk GUIs |
| 17:26.40 | duki | 4 load ztdummy |
| 17:26.48 | E-bola | sure |
| 17:26.58 | E-bola | but asking about oppinions in a gui channel is bound to be biased |
| 17:27.06 | duki | and moh with streaming worked fine. |
| 17:27.11 | ManxPower | asking them here will also be biased. |
| 17:27.14 | tzafrir | duki, it should use HRtimer, and not rtc |
| 17:27.30 | tzafrir | strings zaptel.ko | grep type: |
| 17:27.41 | E-bola | Manxpower: how so and biased towards what? |
| 17:27.47 | tzafrir | hmm, well: strings zaptel.ko | grep source: |
| 17:27.50 | ManxPower | ~zeeek |
| 17:27.50 | jbot | rumour has it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 17:27.54 | [s]Animat | Is it relatively easy to set up asterisk to receive incomming calls from a SIP Proxy (Connected to PTSN) on every extention and also be able to make outgoing calls through said SIP ? |
| 17:28.01 | E-bola | Im not learning asterisk |
| 17:28.06 | E-bola | im trying to make it easier to admin |
| 17:28.07 | [s]Animat | jbot: lol... |
| 17:28.08 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 17:28.09 | ManxPower | GUIs for PBXs are horrid little beasts that should be taken out back and SHOT. |
| 17:28.18 | ManxPower | But I'm not biased. |
| 17:28.32 | [s]Animat | Why do so many linux people hate GUIs ? |
| 17:28.34 | Qwell | and logging to disk is silly - real men use line printers |
| 17:28.39 | Qwell | logging CDRs that is |
| 17:28.41 | E-bola | lol @ qwell |
| 17:28.53 | ManxPower | [s]Animat: for one thing we can't support them here because they make the config files totally unreadable by us. |
| 17:29.02 | E-bola | thats a valid point |
| 17:29.04 | tzafrir | [s]Animat, GUIs limit you to a certain flow and set of options |
| 17:29.08 | tzafrir | by design |
| 17:29.08 | E-bola | and part of why i havent used one yet |
| 17:29.18 | E-bola | its like using a wysiwyg html editor |
| 17:29.23 | [s]Animat | I wish they were in XML and more structured |
| 17:29.27 | E-bola | it makes files horrible to handedit afterwards |
| 17:29.30 | [s]Animat | (to my eyes, that is) |
| 17:29.40 | ectospasm | [s]Animat: they're simpler than XML (IMO) |
| 17:29.51 | tzafrir | [s]Animat, the asterisk config file? XML is not human-editable |
| 17:29.51 | mvanbaak | Qwell: my firewall at home is logging on an old okidata matrix printer using chainpaper |
| 17:30.01 | E-bola | But there is absolutely nothing that stands int he way of somebody making a GUI that lets u manualy edit conf files, and lets you configure ALL possible options from the GUI |
| 17:30.05 | E-bola | i just havent found one yet |
| 17:30.08 | duki | tzafrir: sudo strings zaptel.ko | grep type: |
| 17:30.14 | E-bola | So just bashing GUI's in general is cluelss |
| 17:30.16 | duki | tzafrir: nothing |
| 17:30.16 | E-bola | +e |
| 17:30.17 | [s]Animat | ectospasm: You're probably right. I am just very very accustomed to xml-like markup. |
| 17:30.22 | ManxPower | E-bola: apparently SOMETHING is standing in the way or it would have been written already. |
| 17:30.25 | tzafrir | duki, no need for sudo there, BTW |
| 17:30.29 | [s]Animat | tzafrir: How isn't it human editable? |
| 17:30.41 | tzafrir | duki, strings zaptel.ko | grep source: |
| 17:30.42 | E-bola | ManxPower: When the car wasnt invented what was standing in the way of it? |
| 17:30.52 | tzafrir | [s]Animat, too complex a structure |
| 17:30.53 | mvanbaak | knowledge |
| 17:30.54 | ManxPower | [s]Animat: because if you change the config out from under the GUI it won't work correctly. |
| 17:30.59 | duki | strings zaptel.ko | grep type |
| 17:30.59 | duki | strings: zaptel.ko: Permission denied |
| 17:31.00 | E-bola | Just because something doesnt exist, doesnt mean it requires more than some guys sitting down and making it happen |
| 17:31.04 | ectospasm | [s]Animat: XML would be too complex... yeah like tzafrir said |
| 17:31.13 | ManxPower | E-bola: there were MANY things standing in the way of the invention of the car. |
| 17:31.17 | mvanbaak | E-bola: maybe noone can be bothered |
| 17:31.25 | ManxPower | lack of an internal combustion engine was just one of them. |
| 17:31.31 | [s]Animat | I'll just have to adapt them, eh? |
| 17:31.36 | E-bola | well once they had the technology ready |
| 17:31.45 | E-bola | there where still a period where a car simply wasnt invented |
| 17:31.48 | ectospasm | [s]Animat: and the basic config file format was devised before XML became fashionable |
| 17:31.58 | E-bola | as an opensource GOOD asterisk gui havent been invented either |
| 17:32.04 | ManxPower | [s]Animat: best of luck with that. The GUIs make the asterisk config files so complicated, you'll spend all your time trying to figure how they work. |
| 17:32.05 | [s]Animat | ectospasm: Now that is one thing that i -do- know. |
| 17:32.21 | [s]Animat | ManxPower: I haven't used a GUI for asterisk. |
| 17:32.24 | duki | tzafrir: lsmod |grep rtc |
| 17:32.28 | duki | rtc 10264 1 ztdummy |
| 17:32.56 | tzafrir | duki, are you sure that this is the new one? |
| 17:32.59 | [s]Animat | ManxPower: I have, however, punched and bit things while trying to get my sip.conf and extentions.conf files to work how i want them... |
| 17:33.02 | duki | tzafrir: So I need it for my system, am I wroing? |
| 17:33.25 | ManxPower | [s]Animat: a GUI won't usually work with existing config files. |
| 17:33.29 | tzafrir | again, please give me the output of that command. Run it as root if you get a permissions error |
| 17:33.48 | [s]Animat | ManxPower: Fair enough. I don't use a GUI for asterisk., |
| 17:33.57 | duki | tzafrir: which command please. |
| 17:34.02 | E-bola | all those points a bs |
| 17:34.04 | tzafrir | duki, strings zaptel.ko | grep source: |
| 17:34.08 | E-bola | any existing GUI wont work with existing files |
| 17:34.09 | duki | ok |
| 17:34.22 | E-bola | writing one that does, is very far from impossible |
| 17:34.35 | duki | tzafrir: pwd |
| 17:34.37 | duki | /lib/modules/2.6.22-ARCH/misc |
| 17:35.24 | duki | tzafrir: (I need sudo) sudo strings zaptel.ko | grep source: |
| 17:35.32 | duki | tzafrir: nothing |
| 17:36.16 | ManxPower | duki: you prolly need a -r on the grep. |
| 17:36.32 | tzafrir | duki, so it's not a new one |
| 17:36.51 | tzafrir | its not from ztdummy of the last day |
| 17:36.56 | tzafrir | ls -l ztdummy.c* |
| 17:37.15 | duki | ManxPower: no changes with -r for grep. |
| 17:37.48 | tzafrir | duki, I meant that you run this in the zaptel source directory |
| 17:37.55 | duki | tzafrir: I did make clean && make and sudo make install |
| 17:38.11 | [s]Animat | does anybody know what it means when it outputs "Failed to expand hostname" ? |
| 17:38.12 | ManxPower | duki: well it didn't work |
| 17:41.32 | duki | tzafrir: Ok, here what I did. in the url mentionned I didn't find the ztdummy.c I ran svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel and after the download I put ztdummy.c in the yet existing zaptel-1.4.5.1. make clean && make && sudo make install. |
| 17:41.55 | Qwell | don't use zaptel trunk |
| 17:42.15 | *** join/#asterisk Mmurdock (n=vnjyjta@9.sub-72-121-71.myvzw.com) |
| 17:42.22 | tzafrir | http://svn.digium.com/svn/zaptel/branches/1.4 |
| 17:42.23 | *** join/#asterisk dalbaech (i=narf@youhackme.com) |
| 17:42.35 | tzafrir | duki, that URL is browsable |
| 17:44.27 | ManxPower | duki: did you put in in the same directory as the old ztdummy.c ? |
| 17:44.32 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 17:45.07 | duki | ManxPower: yes exactly at the same directory, |
| 17:45.15 | duki | I can repeat this operation :) |
| 17:45.38 | [s]Animat | What's the best way to check the status of the registration of an SIP Proxy? |
| 17:46.09 | duki | ~ $ find -name ztdummy.c |
| 17:46.14 | duki | ./svn/zaptel/ztdummy.c |
| 17:46.19 | duki | ./zaptel-1.4.5.1/ztdummy.c |
| 17:46.41 | ManxPower | [s]Animat: "sip show registration" to show peers Asterisk is registered too. and "sip show peers" to show who is registered to Asterisk |
| 17:46.45 | duki | and did, cp ./svn/zaptel/ztdummy.c ./zaptel-1.4.5.1/ztdummy.c |
| 17:47.03 | [s]Animat | ManxPower: Thanks :) |
| 17:47.15 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 17:47.19 | ManxPower | duki: you would have wanted to do cp ./svn/zaptel/ztdummy.c ../zaptel-1.4.5.1/ztdummy.c instead |
| 17:47.46 | ManxPower | ah, nevermind, I see what you are doing |
| 17:47.51 | [s]Animat | ManxPower: No such command :( sip show peers works though |
| 17:48.20 | ManxPower | [s]Animat: sip show <tab> |
| 17:48.49 | [s]Animat | ManxPower: asteriskwin32, remember? :P I'll just look through help. :) |
| 17:49.06 | [s]Animat | sip show registry ? |
| 17:49.15 | ManxPower | that would be it. |
| 17:49.25 | [s]Animat | oh noes, state = unregistered :( |
| 17:50.03 | ManxPower | so that means that the remote device is not registered to Asterisk |
| 17:50.13 | ManxPower | for sip show peers |
| 17:50.29 | ManxPower | and for sip show registry it means Asterisk is not registered to the remote side |
| 17:51.18 | [s]Animat | The Remote SIP Proxy is Registered on Asterisk , but Asterisk is not registered on the Remote SIP Proxy :( |
| 17:53.45 | ManxPower | [s]Animat: only one side has to register. |
| 17:54.29 | ManxPower | the entire purpose of registration is to notify what IP address is associated with a specific userid/password. It does NOTHING ELSE. |
| 17:54.56 | [s]Animat | ManxPower: Oh. Damn, I thought I understood why I wasn't receiving calls. |
| 17:55.01 | [s]Animat | Now I dont! :P |
| 17:59.50 | [s]Animat | ManxPower: Wouldn't I need to be registered on the remote side to receive calls from that SIP Proxy? |
| 18:07.17 | [s]Animat | gee whizz I've made very little progress |
| 18:07.39 | [s]Animat | Does anybody have a recommended resource for nubs ? |
| 18:16.15 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 18:17.10 | EmleyMoor | If I have an extension in my default context, is there a way I can make it Goto an extension in the calling phone's set context, or do I need to have it in all the higher contexts too? |
| 18:20.42 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 18:20.46 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 18:20.47 | ThoMe | hello |
| 18:22.10 | ThoMe | how to set "temporary not available" message in my dialplan? |
| 18:22.28 | ThoMe | u = "not available", b = busy, and "temporary not available" ?? |
| 18:22.32 | Yourname`` | Hmm, what's so good about the digium license-only codec anyway? |
| 18:30.05 | [s]Animat | catch y'all |
| 18:38.08 | *** join/#asterisk sniper[FOO] (i=Snip3r@217.27.214.111) |
| 18:38.27 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 18:39.27 | ManxPower | ThoMe: I thought it was just "temportary greeting" |
| 18:39.27 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.60) |
| 18:39.47 | ManxPower | Yourname``: you mean G729? |
| 18:40.00 | Daejeo1 | WomanxPower |
| 18:40.14 | shido6 | doesnt have the same ring |
| 18:40.26 | shido6 | unless its a machine... |
| 18:40.32 | Daejeo1 | shido6: how r u doing |
| 18:40.49 | Yourname`` | ManxPower : Yup. |
| 18:41.08 | ManxPower | Yourname``: it's the only legal way to get G729 for Asterisk. |
| 18:41.16 | ManxPower | So I would say that is something good. |
| 18:41.26 | Yourname`` | Well, there are so many other codecs, open source and free. |
| 18:41.36 | Yourname`` | To pay and get a license from Digium MUST mean something. |
| 18:41.39 | ManxPower | Yourname``: name me ones that is supported by softphones. |
| 18:41.46 | Yourname`` | Like as if it's an uber codec. What sets it apart? |
| 18:41.49 | ManxPower | Yourname``: Digium does not make any money on G729. |
| 18:42.09 | ManxPower | Yourname``: It is nothing special except it is the only compressed codec many phones support |
| 18:42.21 | Yourname`` | Ah, then who does? |
| 18:42.30 | ManxPower | Yourname``: who does what? |
| 18:42.37 | Yourname`` | Many phones support, gotcha. But isn't G711 ulaw also supported by many phones? |
| 18:42.37 | ManxPower | the patent holder makes the money |
| 18:42.48 | Yourname`` | I meant who makes the money. Whose the patent holder? |
| 18:43.01 | ManxPower | Yourname``: G729 takes 8kbps for the codec, and G711 uses 64kbps for the codec. |
| 18:43.25 | Yourname`` | There you go, that's what I wanted to know.. why is G729 so good. 8kbps answers it! :) |
| 18:43.27 | Yourname`` | Thanks, ManxPower. |
| 18:43.29 | ManxPower | Yourname``: Digium expects it to take 5 - 10 years to recover the money they paid for the G729 license. |
| 18:44.02 | ManxPower | Yourname``: there are other codecs with similar bandwidth requirements but they are not generally supported by hardphones. |
| 18:44.22 | Yourname`` | Gotcha. |
| 18:44.26 | InHisName | Are there SIP providers that cannot be set up in an asterisk ? But work with thier adapter ? |
| 18:44.36 | Yourname`` | ManxPower: What's the command in Asterisk that shows me codec information? Like that codec table? |
| 18:44.41 | ManxPower | The ONLY time I use G729 is when I must send phone calls over a WAN and one or both ends do not support GSM. |
| 18:44.51 | ManxPower | Yourname``: what codec information do you want to know? |
| 18:45.03 | Yourname`` | Nothing. Just the command, really. |
| 18:45.03 | ManxPower | ~codec |
| 18:45.11 | ManxPower | "show translations" |
| 18:45.32 | Yourname`` | Wait, got it. It's 'show codecs'. |
| 18:45.47 | Yourname`` | show translations doesn't work on 1.2. |
| 18:45.48 | ManxPower | Yourname``: show codecs does not show what codecs Asterisk supports. |
| 18:45.57 | ManxPower | Yourname``: perhaps it is "show translation" |
| 18:46.04 | Yourname`` | Yeah, JUST saw that one, lol |
| 18:48.49 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 18:48.57 | Yourname`` | Got it, thank you ManxPower. |
| 18:58.28 | InHisName | Are there SIP sevices that * cannot simulate the telephone adapter ? |
| 18:59.54 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:06.33 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 19:07.41 | ManxPower | InHisName: give us an example |
| 19:11.14 | [hC] | this is very strange.. all of a sudden just now all of my phones 160 of them) |
| 19:11.22 | [hC] | oops didnt mean to hit enter... :) |
| 19:11.29 | [hC] | just became unreachable.. even though i can ping em |
| 19:13.17 | InHisName | I have Innomedia ATA for Voiceline n2p. It works fine. Now try to setup * to register, I got as far as "invalid username/PIN" error. I AM using the correct username & PIN. |
| 19:14.58 | ManxPower | InHisName: I have never used that device. |
| 19:15.14 | ManxPower | Many providers lock their devices to only talk to their services. |
| 19:15.32 | ManxPower | and lock their services to only talk to their device. |
| 19:15.53 | ManxPower | since you are using a device and provider I am not familiar with, I really can't help you. |
| 19:16.06 | InHisName | Usually be defining a "user agent" string that is special to account. I got that from tcpdump. |
| 19:35.58 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 19:38.16 | iPod-nano | WHat are some possible reasons why I can't access my server from the internet? |
| 19:38.26 | iPod-nano | Port forwarding has been done. |
| 19:38.34 | iPod-nano | On the router, that is. |
| 19:38.47 | iPod-nano | Is my operating system preventing the traffic somehow? Is Asterisk? |
| 19:40.01 | E-bola | lol |
| 19:40.04 | E-bola | how would we know? |
| 19:40.28 | iPod-nano | Well, if it's something Asterisk is doing, I figured this would be the place to ask. |
| 19:40.37 | E-bola | well can u access it from the lan? |
| 19:40.43 | iPod-nano | If this is something Debian is doing, maybe someone in here has experience? |
| 19:40.46 | E-bola | and when you say access |
| 19:40.50 | E-bola | what on earth do you mean |
| 19:40.51 | iPod-nano | Yes, I can access it from the LAN. |
| 19:41.01 | iPod-nano | I have the X-Lite SIP client, it works fine. |
| 19:41.33 | iPod-nano | But from outsied my local network, it can't register. |
| 19:41.46 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583236.dsl.bell.ca) |
| 19:41.46 | E-bola | that can be a ton of things |
| 19:42.40 | iPod-nano | Well, I have the necessary ports forwarded from my router. |
| 19:42.46 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 19:45.06 | iPod-nano | It would help if anyone in here had experience specifically with Debian. |
| 19:45.35 | E-bola | give your debian box a wan ip |
| 19:45.39 | E-bola | and if it doesnt work |
| 19:45.44 | E-bola | its the client side thats malfunctioning |
| 19:45.56 | iPod-nano | WAN IP? |
| 19:46.38 | E-bola | yes |
| 20:06.54 | [hC] | Ok, so my polycom phones seem to freak out and go into unreachable/circuit-busy mode when their DNS server becomes unavailable, even though from what I can see, in sip.cfg, the ntp server, proxy server, are all set by IP.. anyone seen this? |
| 20:09.31 | Tili | which provider supports multiple calls on single account and has good service |
| 20:12.19 | *** join/#asterisk Shadowfire_ (n=jeff@fl-204-215-37-142.sta.embarqhsd.net) |
| 20:14.36 | ManxPower | Tili: Most of them if you don't to a monthly plan. |
| 20:15.01 | ManxPower | [hC]: and the config on the phone is also all set to IP address rather then host name? |
| 20:15.03 | Tili | ManxPower: well broadvoice has 3 way calling. i can use that to have 2 lines atleast |
| 20:15.12 | ManxPower | also you want to make sure that /etc/hosts lists the hostname and IP of the server |
| 20:15.44 | ManxPower | Tili: most providers don't care HOW many calls you have at the same time, they still charge per-min |
| 20:16.29 | Tili | yeah but with monthly plan from Broadvoice. u can endup having 2 lines if u do 3 way calling. i have to bridge 2 calls |
| 20:17.14 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 20:20.06 | iPod-nano | Monkeys. Ahahaha. |
| 20:24.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-836-010.area5.spcsdns.net) |
| 20:26.00 | [hC] | ManxPower: every server address settting in the phone config files use IP, not hostname. DNS server goes away, the phones flip out and go unreachable. bring dns server back, they all come back up. |
| 20:26.05 | *** join/#asterisk mog (i=mog@nat/digium/x-25b6eea4690198c9) |
| 20:26.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:26.42 | [hC] | ManxPower: and it does have something specifically to do with the dns server the PHONES are using, cause i changed the * server's nameserver to a working one, and the phones still had a problem. |
| 20:28.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:29.03 | *** join/#asterisk jedaustin (n=chatzill@wsip-66-210-241-251.ph.ph.cox.net) |
| 20:30.26 | *** join/#asterisk TJNII (n=TJNII@71-34-176-6.desm.qwest.net) |
| 20:37.25 | jedaustin | Anyone here using 1.4.11? |
| 20:39.29 | Shadowfire_ | hello gents and ladies... |
| 20:41.45 | ectospasm | jedaustin: I am |
| 20:41.47 | ectospasm | I think |
| 20:41.48 | ectospasm | hold on |
| 20:42.02 | ectospasm | jedaustin: yep, I am |
| 20:42.07 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 20:42.27 | jedaustin | ectospasm: having any issues? I'm running it but it was an upgrade from 1.4 (which may be my problem) |
| 20:42.53 | Shadowfire_ | Dell annoys me... saying they support linux.. and they cant even get the SATA drivers available for all distros |
| 20:43.10 | Shadowfire_ | or the most popular ones at least |
| 20:43.44 | *** join/#asterisk goupil (n=goupil@62.147.224.49) |
| 20:44.34 | ectospasm | jedaustin: not too much, but there's only one exten and one trunk on my system |
| 20:44.47 | jedaustin | ectospasm: any voip involved? |
| 20:44.59 | ectospasm | jedaustin: just IAX (-l' |
| 20:45.00 | ectospasm | ; |
| 20:45.06 | ectospasm | d'oh |
| 20:45.10 | Shadowfire_ | well... I guess they have never said it.. but it sure would be nice to clean up loading * on some of their systems |
| 20:46.21 | jedaustin | ectospasm: I'm having issues when bridging zaptel to voip (audio problems) with 1.4, not sure if it's because of asterisk 1.4 or upgrading from a wonky trixbox setup |
| 20:47.14 | ectospasm | jedaustin: my guess is the wonky trixbox setup |
| 20:47.41 | ectospasm | jedaustin: is this a dedicated machine (i.e. no X or frame buffer) |
| 20:47.54 | jedaustin | ectospasm: yes, no X |
| 20:48.06 | ectospasm | frame buffer needs to be disabled, too |
| 20:48.37 | ectospasm | what kind of Zaptel hardware do you have? |
| 20:49.05 | jedaustin | ectospasm: hmm... I don't think I've ever messed with frame buffer, how do you tell if it's on or not/ |
| 20:49.18 | jedaustin | ectospasm: tdm400 1 fxo 3 fxs |
| 20:49.23 | ectospasm | jedaustin: check dmesg for frame buffer |
| 20:49.43 | ectospasm | jedaustin: So when you bridge FXS to FXO it sounds fine? |
| 20:50.00 | jedaustin | ectospasm: yes. Loud though |
| 20:50.22 | ectospasm | jedaustin: you may need to tweak that after using ztmonitor |
| 20:50.25 | jedaustin | framebuffer not on |
| 20:51.08 | jedaustin | ectospasm: the fxs extensions are rarely used. It seems like rxgain/txgain don't have as much of an effect in 1.4 |
| 20:51.10 | ectospasm | what sort of audio problems do you get when you bridge zap to VoIP |
| 20:51.15 | *** join/#asterisk codejunky (n=jan@codejunky.org) |
| 20:51.29 | jedaustin | ectospasm: people can't understand me, distorted/etc |
| 20:51.37 | ectospasm | SIP or...? |
| 20:51.40 | jedaustin | IAx |
| 20:52.42 | ectospasm | can you hear them OK? |
| 20:52.51 | ectospasm | So it's only on outbound calls through the FXO? |
| 20:52.56 | codejunky | Hi, how does the "asterisk -r"-console communicate with asterisk? Through unix socket or tcp? And is there a protocol overview, I want to write an own client. |
| 20:52.58 | jedaustin | ectospasm: usually |
| 20:53.12 | jedaustin | ectospasm: yes PSTN -> FXO -> IAX |
| 20:53.48 | jedaustin | ectospasm: sometimes i get sidetone issues where I can hear myself 1/2 a second later |
| 20:54.15 | ectospasm | Wait, it happens on incoming calls and outgoing calls (IAX<->FXO<->PSTN) |
| 20:55.14 | jedaustin | ectospasm: inbound iax calls are rare |
| 20:56.06 | *** join/#asterisk boobsbr (n=asdg@201.18.228.167) |
| 20:56.31 | boobsbr | howdy |
| 20:57.13 | ectospasm | jedaustin: it would all depend on your dialplan |
| 20:57.18 | hypa7ia | codejunky: unix socket, i think |
| 20:57.35 | jedaustin | ectospasm: how so? |
| 20:57.38 | hypa7ia | codejunky: there are interfaces to it in several languages already, might want to look at those first |
| 20:57.58 | codejunky | hypa7ia: Ah, ok. |
| 21:00.11 | codejunky | To be precise, I want to configure asterisk with python. :) |
| 21:00.18 | boobsbr | uh, i'm trying to figure out why there's no audio on incoming calls to me but outgoing calls work perfectly? could anyone give me a hint? |
| 21:01.47 | hypa7ia | codejunky: there's already starpy |
| 21:01.50 | hypa7ia | i think it's called |
| 21:02.16 | hypa7ia | codejunky: there was a presentation on it at our local Python usergroup a few years ago :) |
| 21:02.26 | _ShrikE | boobsbr: nat? |
| 21:02.28 | codejunky | hypa7ia: Ah, cool. |
| 21:02.55 | hypa7ia | codejunky: there's py-asterisk, pyst, and starpy |
| 21:03.22 | boobsbr | <_ShrikE>, my voip provider uses astrisk and i'm using a ht386 ata behing a wrt router |
| 21:03.33 | boobsbr | *behind* |
| 21:03.44 | codejunky | hypa7ia: starpy looks like what I am searching for. Thanks. |
| 21:03.49 | _ShrikE | ~nat |
| 21:03.50 | jbot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:03.55 | hypa7ia | codejunky: you're welcome :) |
| 21:04.02 | boobsbr | i made a tcpdump at the server |
| 21:05.13 | boobsbr | i receive the invite, they receive the trying + ringing, then they receive the 200 ok |
| 21:05.35 | _ShrikE | your problem is with the rtp, not sip |
| 21:05.37 | boobsbr | i set up the port forwarding on the router |
| 21:07.43 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 21:13.16 | boobsbr | _ShrikE > i'm looking at the 200 OK packet for the outgoing call and the rtp port is 60000, and in the router i forward this port directly to the ATA... |
| 21:13.35 | Shadowfire_ | anyone compiled asterisk on Fedora 7? |
| 21:13.47 | Shadowfire_ | what was success rate? |
| 21:14.24 | TJNII | I'm trying to get asterisk working with ekiga.net and I keep getting "406 Not Acceptable" back. It is a nat problem, but I can't figure out how to fix it. |
| 21:17.09 | _ShrikE | boobsbr: Do you have a capture verifying rtp is coming through the firewall? |
| 21:17.11 | Shadowfire_ | TJNII: found this... it may help... check it out - http://bugs.digium.com/view.php?id=5824 |
| 21:19.19 | TJNII | Shadowfire_: I know its a NAT problem from playing with the ekiga client and from googling. The ekiga client uses STUN and the google results said "Not Acceptable means that you are trying to register a private IP to ekiga.net. Please register your public one." However, the results did not say how to do that. |
| 21:19.25 | Shadowfire_ | TJNII: also look at this... http://www.voip-info.org/wiki/view/Ekiga |
| 21:19.34 | boobsbr | i only have a capture at the server, i still need to install openwrt on my router so i can capture what's going on my router's firewall |
| 21:19.59 | TJNII | Shadowfire_: Yea, the second one doesn't really have a fix in it, though. |
| 21:20.00 | _ShrikE | boobsbr: I can pretty much guarantee you its not making it through |
| 21:20.20 | boobsbr | shouldn't the audio at least go 1 way, like outwards? |
| 21:20.43 | Shadowfire_ | It said this on it - UPDATE - I fixed this by commenting out externhost line in sip.conf as it was basically a DNS problem , my DNS server was resolving my dyndns domain to my internal IP address which was getting sent to ekiga.net. |
| 21:20.43 | Shadowfire_ | Thanks dsandras for the help and for the ekiga softphone. |
| 21:20.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:21.06 | TJNII | I don't have a externhost host line, though |
| 21:21.12 | Shadowfire_ | D-Fender: Howdy sir |
| 21:21.15 | Shadowfire_ | : |
| 21:21.17 | Shadowfire_ | :) |
| 21:21.20 | TJNII | So I can't fix it by commenting it out. |
| 21:22.57 | _ShrikE | boobsbr: pastebin what you have. |
| 21:23.08 | boobsbr | ok |
| 21:25.05 | Shadowfire_ | be back .. I have to restart |
| 21:25.50 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:27.56 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 21:28.18 | TJNII | Does 1.2 support STUN when operating as a SIP client? |
| 21:29.11 | boobsbr | pastebin is really slow here, uploaded the .cap file to rapidshare http://rapidshare.com/files/56203449/boobsbr.cap.html. lemme try to upload to pastebin again |
| 21:31.39 | boobsbr | ok, got pastebin working, http://pastebin.com/d5c0b9c09 |
| 21:32.43 | [TK]D-Fender | TJNII, No. * does not support STUN |
| 21:33.03 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:34.17 | [TK]D-Fender | TJNII, ---> |
| 21:34.19 | [TK]D-Fender | ~sipnat |
| 21:34.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:34.38 | fujin | hi |
| 21:36.12 | boobsbr | hi fujin |
| 21:39.17 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:41.20 | boobsbr | hmmm, i didn't know about STUN |
| 21:41.51 | boobsbr | my phase is set on FUN |
| 21:41.58 | boobsbr | *phaser* |
| 21:42.11 | boobsbr | seems to be working now |
| 21:42.20 | boobsbr | thanks a lot you guys |
| 21:45.07 | boobsbr | [TK]D-Fender, you probably need to replace the current drivers to charge those capacitors, otherwise it's gonna take forever |
| 21:45.20 | [TK]D-Fender | boobsbr, Patience! |
| 21:48.09 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 21:50.05 | boobsbr | damn |
| 21:51.59 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:53.23 | NirS | hey all |
| 21:53.29 | NirS | how is everybody doing ? |
| 21:55.21 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 21:56.29 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:59.51 | NirS | anyone has a2billing experience here ? |
| 21:59.57 | NirS | I've ran into a funky issue |
| 22:05.46 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 22:10.23 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 22:10.34 | wishes | ya know, i just noticed there are a lot of people in here |
| 22:16.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:18.30 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 22:25.19 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 22:26.13 | fujin | mm, more than usual |
| 22:28.50 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 22:29.44 | fujin | Where would I start diagnosing one-way audio dropout? |
| 22:29.49 | fujin | no NAT, canreinvite=yes |
| 22:30.07 | fujin | phone-to-phone never exhibits the same symptoms, so I'm willing to bet that it's the as5400 |
| 22:30.23 | fujin | only outbound, too |
| 22:36.27 | *** join/#asterisk xpot (n=xpot@c-71-195-241-115.hsd1.ut.comcast.net) |
| 22:51.53 | sniper[FOO] | xpot: howsgoin/ |
| 22:51.54 | sniper[FOO] | ? |
| 22:51.58 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 22:52.46 | mpruett | Does anyone know a way to see if a conference room has a lock on it? |
| 22:53.16 | mpruett | Much researching and testing has resulted in no answers for me - hoping someone here can help me |
| 22:53.44 | fujin | an asterisk MeetMe conference room? |
| 22:53.51 | mpruett | Yes |
| 22:54.01 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 22:54.02 | fujin | tried reading the documentation? |
| 22:54.35 | mpruett | I did - I know how to set and remove locks - I couldn't find a way to check for the status of a lock |
| 22:54.45 | mpruett | Is it in the doc files for Meetme? |
| 22:55.02 | mpruett | I admit I only did research on the web |
| 22:56.13 | fujin | mm, I'm not sure if meetmeadmin provides the functionality to see if a cofnerence is locked |
| 22:56.15 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-218-175.socal.res.rr.com) |
| 22:56.18 | fujin | it can lock and unlock with l,L |
| 22:56.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-836-010.area5.spcsdns.net) |
| 22:57.10 | fujin | there doesn't appear to be that functionality from what I can see |
| 22:57.24 | fujin | you could put in a request for app_meetme to have this added, although it probably wouldn't be changed until 1.6 |
| 22:57.29 | mpruett | OK - thanks anyway though! |
| 22:59.44 | fujin | just need to write some code which checks ast_conference.locked |
| 22:59.45 | fujin | unsigned int locked:1; /*!< Is the conference locked? */ |
| 23:00.12 | fujin | I'm not very familiar with the API otherwise I'd hack somethign together for ya |
| 23:03.16 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:05.27 | *** join/#asterisk nou (i=Chaton@unaffiliated/nou) |
| 23:07.00 | fujin | Anyone know how I can 'hide' the verbosity of a macro? |
| 23:07.09 | fujin | I'm using a local channel/macro for queue call delivery |
| 23:07.13 | fujin | but it spams my console/logs |
| 23:08.05 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 23:08.10 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 23:18.17 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:19.06 | [TK]D-Fender | fujin, Nope. |
| 23:19.23 | fujin | nope = not possible or nope = i don't know? |
| 23:22.31 | [TK]D-Fender | fujin, Not possible. Verbose & loggin rates are global |
| 23:22.40 | fujin | blast |
| 23:35.02 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 23:48.58 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 23:58.53 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 23:59.27 | iPod-nano | Anybody with Debian experience in here? |