| 00:28.37 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 00:29.27 | nny | getting Error: Dahdi device failed to come up Possible Cause: UDEV not installed! after updating to 2.4.0 dahdi on a system here. What could be the possible causes? I have the modules loaded |
| 00:30.21 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 00:33.23 | nny | i have checked, zaptel is not loaded or running |
| 00:37.09 | xSmurf | is there a quick way for a DISA to use the caller's voicemail password to authenticate? |
| 00:39.27 | *** join/#asterisk jamko (~chatzilla@174-146-5-240.pools.spcsdns.net) |
| 00:42.58 | nny | ok |
| 00:43.00 | nny | Unable to open master device '/dev/dahdi/ctl' |
| 00:43.04 | nny | when running dahdi_cfg |
| 00:43.22 | nny | i am using a sangoma card, but just trying to get dahdi loaded without it right now, using ztdummy |
| 00:43.31 | nny | i have commented out any modules not used by the system |
| 00:43.37 | nny | in /etc/dahdi/modules.conf |
| 00:47.30 | zleslie | How do I check voicemail? |
| 01:08.25 | *** join/#asterisk Evet (~Evet@unaffiliated/evet) |
| 01:10.05 | zleslie | got it |
| 01:12.54 | *** join/#asterisk sourcode (~code@ppp-58-8-233-78.revip2.asianet.co.th) |
| 01:34.57 | nny | line 0: Unable to open master device '/dev/dahdi/ctl' dammit |
| 01:35.06 | nny | can't get dahdi to load no matter what I do |
| 01:36.51 | WIMPy | So what happens if you 'modprobe dahdi'? |
| 01:41.54 | nny | WIMPy: it loads dahdi_dummy, dahdi and crc_ccitt |
| 01:42.36 | nny | WIMPy: but using dahdi_cfg -vvvv or /etc/init.d/dahdi start both fail for the same reason |
| 01:42.58 | nny | WIMPy: and no dev nodes are created when I load the modules manually |
| 01:43.01 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 01:43.49 | *** join/#asterisk Texou (~Texou@unaffiliated/texou) |
| 01:44.37 | Texou | hi |
| 01:45.07 | Texou | I'm sorry for my questions, but really reading doc didn't help me answering about my needs with asterisk |
| 01:45.26 | nny | WIMPy: think i figured it out |
| 01:45.28 | Texou | 2 questions |
| 01:45.38 | nny | WIMPy: older system, doesn't use udev it seems |
| 01:45.54 | WIMPy | nny: You need the modules for the channels loaded before you can dahdi_cfg. |
| 01:47.07 | nny | WIMPy: i have them loaded, however there are no /dev nodes. Some research shows this base OS (it's 3 years old) uses /dev/ instead of udev |
| 01:49.03 | nny | WIMPy: fyi (in case this sad issue comes up again) ./build_tools/make_static_devs in dahdi dir fixed it |
| 01:49.29 | Texou | 1st tell me: is asterisk a client or only a server that a client must connect to in order to phone? |
| 01:51.03 | WIMPy | It is a back-to-back User Agent. |
| 01:51.06 | p3nguin | texou: Asterisk is a Back-to-Back user agent. |
| 01:51.21 | p3nguin | B2BUA |
| 01:51.43 | p3nguin | texou: What that means to you: Asterisk is both a client and a server. |
| 01:51.55 | Texou | ok |
| 01:53.11 | Texou | p3nguin: and can I use it as tool to call people from my computer even if my buddies don't have asterisk installed on their computer? |
| 01:54.56 | p3nguin | texou: Absolutely. |
| 01:56.11 | p3nguin | texou: But it is far more powerful than just that. |
| 01:56.22 | Texou | I imagine |
| 01:56.35 | Texou | it's I try understanding but I have to ask as reading doesn't help me |
| 01:56.48 | Texou | p3nguin: ok. and what does my buddy mhave to have installed? what kind of tool (on Windows for instance since my buddies are often windows users)? |
| 01:57.09 | p3nguin | texou: You want to call him on his computer or on his telephone? |
| 01:57.19 | Texou | computer |
| 01:58.06 | p3nguin | texou: You really don't even need Asterisk if you only want to call your friend from your computer to his. Asterisk is intended to use for a PBX. |
| 01:58.25 | Texou | ah |
| 01:58.41 | p3nguin | texou: In either case, you can use zoiper, x-lite, ekiga, or any other software SIP or IAX2 phone. |
| 01:59.02 | Texou | p3nguin: my idea was to workaround skype or solutions which don't work on Linux: MSN, Netmeeting... |
| 01:59.47 | p3nguin | texou: Yeah. You could both have soft phones on your computers and call each other that way. You could install twinkle and he could install zoiper, for example. |
| 01:59.49 | WIMPy | Netmeeting works with anything that uses H323, like e.g. ekiga or Asterisk. |
| 02:00.30 | Texou | I didn't know them. I can search what they do |
| 02:01.08 | Texou | WIMPy: That's why I ask fkr asterisk because ohphone I didn't understand it, and I'd like a single tool for various tasks |
| 02:02.03 | WIMPy | Well, Asterisk speaks many proticols. |
| 02:02.19 | p3nguin | Most people think of asterisk as kind of a centralized "server" because of typical placement and integration, but technically it's both a server and a client. |
| 02:02.57 | p3nguin | If you only want to voice chat with your friends, it is probably not the right tool for the job. |
| 02:03.14 | Texou | p3nguin: well now I'll try explaining French situation: in France, we have some "boxes" which provide phone, TV, Internet. We have a phone number and our "relay" is the box. The computer is connected to the box, the phone too. The box works as a router. Is it possible to redirect calls which arrive on my telephone via my phone number, on my PC via asterisk? I mean: instead of ringing my phone, it goes on |
| 02:03.16 | Texou | <PROTECTED> |
| 02:04.06 | p3nguin | The short answer: yes it is possible. |
| 02:04.18 | WIMPy | That depends on your provider and if they are willig to give you the data you need to use other hardware. |
| 02:04.39 | Texou | WIMPy: ah |
| 02:04.39 | WIMPy | The short answer ist in theory yes, in practice probably not. |
| 02:04.57 | p3nguin | The long answer: yes it is possible, but in addition to Asterisk configured to answer your phone line, you'll need to configure it to then send the call to your software phone on your computer. |
| 02:05.04 | Texou | well I can connect any telephone on the box, so I guess it's not so secrete :) |
| 02:05.27 | WIMPy | Most providers won't allow you to use any other hardware than the box you got from them and some go to extremes to make it (nearly) impossible. |
| 02:05.32 | Texou | hmm |
| 02:05.54 | p3nguin | The box provides the phone jack. There's not much else they control. |
| 02:06.01 | Texou | ah because I've to use another hardware than the box? |
| 02:06.30 | p3nguin | You'd connect your phone line from that box to some other piece of hardware that allows it to interface with Asterisk. |
| 02:06.35 | WIMPy | Ye, you could connect via telephony hardware to the IAD, but that requires hardware ind goves you the worst of both worlds. |
| 02:06.59 | p3nguin | You'll either use an ATA or an FXO card in the computer that runs Asterisk. |
| 02:07.32 | Texou | ah ok |
| 02:07.39 | WIMPy | You'd best try to google for your provider and the type of IAD you've got and see if you find a howto on getting your phone account out of it. |
| 02:08.39 | WIMPy | Off course it is possible to connect Asterisk to a SIP account via two gateways, but it is not that clever. |
| 02:08.55 | Texou | hmm so a simple PC then a box are not enough. Additional hardware seems needed |
| 02:09.06 | p3nguin | In that type of system, there isn't much use trying to replace your "box" that they provide. That box is specialized hardware that both combines and separates the services you buy from the company. |
| 02:09.07 | *** join/#asterisk Dovid (~Dovid@pool-71-172-225-209.nwrknj.east.verizon.net) |
| 02:09.34 | WIMPy | I actually do so, as my provider is one of those having spent quite some effort to kep the account details secret, but it's a shitty solution. |
| 02:09.54 | Texou | ok |
| 02:10.25 | Texou | p3nguin: I want to keep my box, but using computer instead of telephone to take my calls :) |
| 02:10.47 | WIMPy | Google or buy hardware. |
| 02:10.53 | p3nguin | That's no problem. Just connect a specific piece of hardware to the box via phone cord. |
| 02:10.57 | Texou | I understand |
| 02:10.59 | p3nguin | SPA-3102 |
| 02:11.25 | WIMPy | I hope your IAD at least has an S0 port? |
| 02:11.25 | Texou | ah ok I need a specific piece of hardware between box and PC |
| 02:11.43 | p3nguin | Phone cord hooks from the box's jack to the SPA-3102 line jack. |
| 02:11.44 | WIMPy | Yes |
| 02:12.10 | WIMPy | Or you find the account data your box uses and put that into Astersks config directely. |
| 02:12.15 | p3nguin | If all you want to do is send the call to a soft phone, you don't even need asterisk. |
| 02:12.48 | p3nguin | There probably isn't "account data." It's probably standard phone service that just happens to be bundled in this fancy box. |
| 02:12.49 | Texou | ah ok I see |
| 02:13.06 | p3nguin | The ones I am familiar with send the services over a standard copper pair. |
| 02:13.10 | WIMPy | p3nguin: It's most definitely a SIP account. |
| 02:13.19 | p3nguin | I doubt it. |
| 02:13.53 | p3nguin | It's no more a SIP account than if you have both phone service and DSL service coming out of the same wall jack. |
| 02:13.56 | WIMPy | That's the standard solution today. |
| 02:14.18 | WIMPy | It's become pretty hard to get a real phone line over here. |
| 02:14.40 | WIMPy | In mos cases you get an IAD, i.e. a DSL router with integrated ATA. |
| 02:17.19 | Texou | is investigating :) |
| 02:18.38 | WIMPy | Google for your provider, the model of your IAD and "Asterisk". I'd bet you will find something. |
| 02:19.05 | Texou | I can redirect coming in calls to the SIP account or the box. :) |
| 02:19.43 | WIMPy | What redirect? |
| 02:19.49 | WIMPy | Or rather where? |
| 02:21.43 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 02:23.14 | Texou | WIMPy: it's what I try understanding :) My provider gives me login and passwords: seems I can redirect calls I receive on the box or on any SIP account |
| 02:23.29 | WIMPy | Great! |
| 02:24.02 | WIMPy | Looks like you have found a customer friendly provider. |
| 02:24.06 | Texou | yes |
| 02:24.20 | Texou | WIMPy: yes it's the most cool in France |
| 02:24.39 | WIMPy | So not extra hardware and not lossy double conversion. |
| 02:24.46 | Texou | but now I search what I can to with such info :) (really, I know nothing in this matter) |
| 02:24.57 | Texou | oh great |
| 02:25.13 | WIMPy | As far as Asterisk goes you will put that data in to your sip.conf. |
| 02:25.36 | Texou | ok |
| 02:25.44 | WIMPy | So Asterisk will talk directly to your Telco over IP. |
| 02:26.07 | Texou | then I need a phone software installed on the PC so right? |
| 02:26.47 | p3nguin | If you guys get SIP accounts for home phone service, that's pretty friggin' awesome. |
| 02:26.49 | WIMPy | You can use that data to put it in to a softphone directely as well, without Asterisk inbetween. |
| 02:27.23 | WIMPy | No. That the downfall of telephony. |
| 02:27.24 | Texou | ah |
| 02:28.31 | WIMPy | If they'd at least use H323 or something (like originally specified for NGN) it would be half as bad, but SIP must be the worst possible choice. |
| 02:28.47 | Texou | hmmm |
| 02:29.10 | Texou | ah |
| 02:29.20 | Texou | I should avoid SIP so? |
| 02:29.51 | WIMPy | If your provider uses it, which they probably do, that's what you have to use. |
| 02:30.03 | Texou | ok |
| 02:30.39 | WIMPy | Somehow SIP stacks seem to cost a cent less than H323 stacks or so. No idea why they all use SIP today. |
| 02:30.45 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 02:30.54 | WIMPy | On the backbomes it's still H323 AFAIK. |
| 02:31.03 | WIMPy | s/mes/nes/ |
| 02:31.19 | Texou | so I configure sip conf on asterisk, then enable my provider's service, then my calls will arrive on asterisk right? |
| 02:31.24 | p3nguin | How do they deliver this service to your home? |
| 02:31.42 | WIMPy | ADSL |
| 02:31.50 | p3nguin | over a single copper pair? |
| 02:31.53 | Texou | yes adsl |
| 02:31.59 | WIMPy | yes |
| 02:32.20 | p3nguin | To me, this sounds very flexible. |
| 02:32.30 | WIMPy | Some split things at IP level, others set up several ATM PVCs. |
| 02:33.15 | WIMPy | Some allocate fixed bandwith for (user) internet and phone service, others do it dynamically. |
| 02:33.33 | Texou | but once calls arrive on asterisk, how do I take them? does asterisk provide a tool for this or I need a software? |
| 02:34.02 | WIMPy | You can use Asterisk as a phone, but it's very simple. |
| 02:34.15 | WIMPy | So you'd connect an IP phone to it. |
| 02:34.27 | WIMPy | Either a hardware one or a softphone on your PC. |
| 02:35.22 | Texou | WIMPy: why is asterisk simple? |
| 02:35.54 | WIMPy | It isn't |
| 02:36.03 | WIMPy | Just as a phone. |
| 02:36.22 | Texou | "it's very simple" you said |
| 02:36.59 | WIMPy | You can connect Asterisk to quite a number of very different devices and protocols. |
| 02:37.15 | Texou | ok |
| 02:37.23 | WIMPy | Your sound card is one of them, but not something that looks like a phone. |
| 02:38.04 | Texou | ah ok so I need a softphone to use my PC right? |
| 02:38.08 | WIMPy | You'd have avery simple command line interface to it. |
| 02:38.27 | Texou | ah |
| 02:38.35 | Texou | I need commandline interface :) |
| 02:38.39 | WIMPy | You don't _need_ one, but you most probably _want_ one. |
| 02:38.49 | Texou | I prefer it's more accessible than GUI for me |
| 02:39.13 | WIMPy | Well, ok, you can use that if you like. |
| 02:39.36 | Texou | WIMPy: well... I want one if I can't without it compose a phone number or take a call :) |
| 02:39.54 | Texou | if asterisk allows this it's perfect |
| 02:39.56 | WIMPy | No, that will work. |
| 02:40.05 | Texou | cool |
| 02:40.50 | Texou | what's its name? (the interface)? |
| 02:41.16 | WIMPy | console, ALSA or OSS |
| 02:41.36 | Texou | ah :) |
| 02:41.58 | WIMPy | Console uses some library to acces your sound system, the other two are pretty obvious. |
| 02:42.54 | Texou | WIMPy: wrll ok if I understand you mean that if I don't use softphone, my soundcard is a phone itself and I can use it via alsatools (for example). If I prefer more friendly interface, I need a softphone. Did I understand? |
| 02:43.25 | WIMPy | Correct |
| 02:43.30 | Texou | ok |
| 02:43.49 | WIMPy | You can access the sound card via those three interfaces from the Asterisk CLI. |
| 02:43.50 | Texou | well so harder :) because don't know alsatools to compose or take call :) |
| 02:44.20 | Texou | and not for pulseaudio too |
| 02:44.23 | WIMPy | No you do it via commands on the Asterisk shell. |
| 02:44.33 | Texou | aah ok |
| 02:44.40 | Texou | I prefer this so :) |
| 02:44.59 | WIMPy | For volume settings you have to use some standard mixer application. |
| 02:45.22 | Texou | no problem for this |
| 02:46.41 | Texou | WIMPy: where can I find asterisk's commands? |
| 02:47.07 | WIMPy | ~newbook |
| 02:47.07 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/ |
| 02:47.19 | WIMPy | That sould give you lots of information. |
| 02:47.30 | Texou | ah :) |
| 02:47.46 | Texou | if I understand them :) |
| 02:47.53 | Texou | it's a paper book? |
| 02:48.06 | WIMPy | You can read it online. |
| 02:48.15 | Texou | ok cool |
| 02:48.20 | WIMPy | That version isn't printed, yet. For paper |
| 02:48.25 | WIMPy | ~book |
| 02:48.25 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 02:49.08 | Texou | I note |
| 02:53.41 | Texou | I think I'll try, it's very interesting even if complex for me :) |
| 02:54.02 | WIMPy | Yes, there is some learning curve. |
| 02:54.20 | WIMPy | But it looks worse than it is. |
| 02:54.24 | Texou | yes and in a matter I don't know at all: telephony |
| 02:54.44 | WIMPy | You will probably only need a fraction of the functionality and don't have to care about the rest. |
| 02:55.26 | Texou | hard for me to sort info; that's I ask online |
| 03:02.47 | Texou | WIMPy: and so asterisk isn't useful to access to protocol used for instance by buddies which use MSN on Windows to speak or skype right? |
| 03:03.31 | WIMPy | MSN uses H323. That's included in Asterisk. |
| 03:03.43 | Texou | ah cool |
| 03:03.49 | WIMPy | Skype is a commercial option, I think. |
| 03:05.01 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 03:05.07 | Texou | ah ok |
| 03:05.20 | shmaltz | hi every1 |
| 03:08.04 | Texou | WIMPy: and do you know by experience if MSN dialogs have a quite quality? |
| 03:09.33 | Texou | hi |
| 03:10.20 | WIMPy | No current one. When it was netmeeting it was fully configurable. |
| 03:10.38 | Texou | ok |
| 03:11.51 | Texou | WIMPy: and does asterisk allow to cumulate calls? to do conferences I mean |
| 03:12.03 | WIMPy | Yes |
| 03:12.54 | Texou | even commandline tool? |
| 03:13.23 | WIMPy | That does not matter. That would just be like any other channel on the conference. |
| 03:13.54 | Texou | ok |
| 03:14.02 | WIMPy | However I personally experience ever increasing delays it I add my sound card to a conference. |
| 03:14.08 | WIMPy | if |
| 03:14.36 | Texou | ok |
| 03:15.01 | Texou | WIMPy: and does it work properly even via Internet? (I especially hear of asterisk on local networks) |
| 03:15.37 | WIMPy | Well internet is generally not good for realtime, but usually it works very well in practice. |
| 03:16.15 | Texou | ok |
| 03:16.36 | Texou | I'll find how to configure this protocol too :) |
| 03:17.04 | WIMPy | There are many to choose from. |
| 03:18.10 | Texou | WIMPy: yes, beyond my telephone purpose, I also try finding to communicate with person on Windows with common popular clients, hence my MSN question or Skype question |
| 03:18.17 | p3nguin | Your payment for US $0.16 has been processed by PayPal. You will receive an email receipt shortly. |
| 03:19.53 | Texou | hm? lol |
| 03:20.21 | p3nguin | That's probably the cheapest ebay purchase I have ever made. |
| 03:20.34 | Texou | :) |
| 03:20.43 | WIMPy | What was it? |
| 03:21.22 | Nivex | I made a similarly small purchase once. It was a friend doing a joke eBay auction. I think I paid like $0.25 or something. |
| 03:21.28 | p3nguin | It was a wrist strap for a digital camera. |
| 03:21.47 | p3nguin | 16 cents, free shipping |
| 03:25.49 | Texou | WIMPy: do you know is book is translated? |
| 03:26.18 | WIMPy | The current one isn't even finished ;-) |
| 03:26.50 | Texou | WIMPy: 3rd edition? |
| 03:26.54 | WIMPy | I could vove you a german one. No idea about any french texts. |
| 03:26.59 | WIMPy | yes |
| 03:27.55 | Texou | ok |
| 03:28.05 | Texou | WIMPy: and second one only in German so? |
| 03:28.42 | WIMPy | It's a different book available in german and a less current version in english. |
| 03:29.03 | Texou | ok |
| 03:29.15 | Texou | less current? |
| 03:29.23 | WIMPy | Of the german version. |
| 03:29.25 | Texou | WIMPy: English older ? |
| 03:29.59 | WIMPy | Yes, the current german one hasn't been translated. (yet?) |
| 03:30.40 | Texou | hmmmm...... |
| 03:30.51 | Texou | WIMPy: what's the original release? :) what language? |
| 03:31.06 | WIMPy | German. |
| 03:31.20 | WIMPy | The two I gave you in the beginning are english only. |
| 03:32.07 | Texou | yes and older than German so? |
| 03:33.04 | WIMPy | The german one is in between the english ones. |
| 03:33.17 | Texou | ok |
| 03:33.31 | WIMPy | They are for Asterisk 1.4, 1.6 and 1.8 respectively. |
| 03:34.06 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 03:34.42 | Texou | so 1.8=English, 1.6=German, 1.4=English? |
| 03:34.56 | WIMPy | yes |
| 03:35.27 | Texou | ok so English is the newest :) |
| 03:35.37 | WIMPy | yes |
| 03:36.08 | Texou | and is there a source code of it? a translation team or anyelse? |
| 03:36.35 | WIMPy | They are by different people. |
| 03:36.59 | WIMPy | The new book for 1.8 is done byt the guys who did the book for 1.4 as well. |
| 03:37.00 | Texou | hmm I'm nearly sure there's a French team no? |
| 03:37.20 | WIMPy | I think there has been a 1.4 version of the german book as well. |
| 03:37.44 | WIMPy | Might be that the english version of the german book is also only 1.4. It is definitely older. |
| 03:38.09 | Texou | ok |
| 03:38.23 | Texou | well the urls you gave point at 1.8, English? |
| 03:38.40 | WIMPy | 1.8 and 1.4 |
| 03:38.45 | Texou | ok |
| 03:38.58 | Texou | I could help French team if I find them and source of doc |
| 03:39.56 | WIMPy | I think a good understanding of Asterisk would help a lot in translating such a book. |
| 03:40.35 | Texou | right. I don't know it at all lol |
| 03:42.11 | WIMPy | Ok, I really need some sleep. |
| 03:43.21 | Texou | WIMPy: goodnight so and thanks very much :) |
| 03:43.28 | WIMPy | nn |
| 03:44.52 | xSmurf | so we have one of those conference phones, is it possible to make it so that when someone enters a conference room, that phone is dialed in automatically?? |
| 03:49.59 | Texou | WIMPy: only to guide me, can you give me the name of one command to use telephone feature (soft phone) of asterisk? |
| 03:53.39 | Texou | or someone else? |
| 04:54.35 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 05:12.34 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 05:29.53 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.250.92.adsl.dynamic.totbb.net) |
| 05:30.56 | BugKhaM | I am using this confguration for my clients and found that the dtmf digits recieved are sometimes double, Mobile Phone Clients -> SIP Gsm GW [rfc2833]-> Asterisk Server1 -> IAX2 -> Asterisk Server 2 [IVR] |
| 05:31.19 | BugKhaM | will inband work better than rfc2833? |
| 05:32.14 | *** join/#asterisk coppice (~chatzilla@210.17.219.137) |
| 05:35.10 | BugKhaM | and is the parameter dtmfmode working under IAX2 protocol? |
| 05:35.19 | p3nguin | nope |
| 05:38.36 | BugKhaM | p3nguin: OK, so how does the dtmf behave when it's goining through IAX2? that's probably the problem I'm having |
| 05:39.18 | p3nguin | I have no idea. I just know that dtmfmode is not a configurable parameter in IAX2. |
| 05:40.48 | p3nguin | I will say that I use IAX2 to my provider and I rarely have any problem with DTMF. |
| 05:50.15 | BugKhaM | p3nguin: I don't have any problem if those clients are directly sent to my IVR server either |
| 05:51.40 | BugKhaM | p3nguin: but redirecting through another server sometimes causes problems, probably due to latency |
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| 09:46.19 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 09:47.01 | DND | hi guys. how can i check if the sip account is disabled? |
| 09:47.38 | DND | x-lite is telling m: account failed to enable. |
| 09:47.40 | DND | *me |
| 09:53.03 | ectospasm | DND: you can see if it's in "sip show peers" |
| 09:55.05 | DND | ectospasm lemme check |
| 09:55.50 | DND | will it show Disabled or Unknown? |
| 09:55.59 | DND | also how can i enable/disable account? |
| 09:56.00 | ectospasm | Unknown, I think |
| 09:56.16 | ectospasm | DND: observe the CLI as the client tries to register |
| 09:56.45 | ectospasm | If you have verbose ("core set verbose 10") and debug ("core set debug 10"), you should see some traffic that gives you clues. |
| 09:57.13 | ectospasm | ...the debug isn't absolutely necessary, and may actually provide more output than you need to see. |
| 09:57.19 | DND | hmm maybe i'll tail to the "full" log instead. i feel dizzy tring to keep up with cli scrolling :D |
| 09:57.23 | ChannelZ | There is no 'enabled' or 'disabled' in asterisk... this sounds more like an x-liteism |
| 09:58.06 | DND | yeah seems x-lite is giving some misleading error |
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| 09:58.17 | ChannelZ | Don't turn on debug.. turn on verbose at about 3 and that should be sufficient |
| 09:58.19 | ectospasm | DND: use screen (or your terminal client) to scroll up |
| 09:59.15 | DND | ok thanks. will do that. also by the way can the registration to asterisk be denied because of too much lag? |
| 09:59.41 | DND | i mean if asterisk detects 2000ms lag, will it disconnect the offending client? |
| 09:59.55 | DND | *2000ms lag from the sip client |
| 09:59.58 | ectospasm | DND: if Asterisk doesn't get the proper set of responses to a registration in a timely fashion, the registration will fail |
| 10:00.05 | ectospasm | yeah, if you |
| 10:00.37 | ectospasm | 've got qualify=yes, the sip peer must respond within 2000ms (2sec), or else it will be considered UNREACHABLE |
| 10:00.45 | DND | hmm maybe i got the answer. one user is trying to connect to our server via vpn |
| 10:01.05 | DND | i will check again if its a lag issue |
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| 10:49.10 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:49.14 | AliRezaTaleghani | hi all |
| 10:50.07 | AliRezaTaleghani | i am looking for a some thing like a popup or an applet for MS-windows Client witch linked to their Extension ... |
| 10:50.10 | AliRezaTaleghani | oom |
| 10:50.19 | AliRezaTaleghani | what a hard thingto explain.. |
| 10:50.25 | AliRezaTaleghani | in a clear way.. |
| 10:51.11 | AliRezaTaleghani | i want to Send some URL in Dial application, and open that URLs on Agents PC (we use Softphone) |
| 10:51.48 | AliRezaTaleghani | what should i search for? or any other way or idear |
| 10:51.53 | AliRezaTaleghani | idea* |
| 10:51.57 | AliRezaTaleghani | plz |
| 10:53.31 | ChannelZ | well some softphones support incoming URLs... like Zoiper Biz |
| 10:54.36 | AliRezaTaleghani | ChannelZ: hi man, :) how are u... |
| 10:55.08 | AliRezaTaleghani | ChannelZ: yep, but i am going to find an independent way |
| 10:56.19 | AliRezaTaleghani | ChannelZ: first of all, ;) should tel you i had been replace my Elastix with a Asterisk1.8 (compiler on CentOS) So ! all my new quesion will just depent on astersik :P |
| 10:57.32 | AliRezaTaleghani | by the way... i think or hear, some about Applets (maybe via AMI) that let the Agent Login/out our see Callers INFO not just an Soft/Hard Phone.. |
| 10:58.01 | AliRezaTaleghani | i need this to be able to do some integration with our CRM |
| 10:59.07 | ChannelZ | well yes AMI can show you a lot of things as they happen |
| 11:00.12 | AliRezaTaleghani | umm, i am so newbie on AMI! |
| 11:01.47 | ChannelZ | It's a relatively simple socket-based interface to Asterisk. Google 'asterisk ami' for info |
| 11:02.33 | ChannelZ | But basically you can set it up to tell you about events that occur and then act accoringly |
| 11:03.26 | ChannelZ | Otherwise you could run an AGI or otherwise execute a little binary prior to doing a Dial() which would do the pop-up on the agent's computer by whatever means/protocol you invent |
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| 11:21.18 | [sr] | howdy friends |
| 11:25.25 | ChannelZ | howdee |
| 11:28.48 | AliRezaTaleghani | ChannelZ: huuumm, should read more on AMI and the the ways that i can handel event, or how should i send the events to my Applet :( |
| 11:28.58 | AliRezaTaleghani | by the way, tnx for your help... |
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| 11:29.12 | AliRezaTaleghani | like ever, your are kind to me.. tnx |
| 11:29.23 | AliRezaTaleghani | @};- be happy :) |
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| 16:03.40 | Texou | hi |
| 16:07.14 | Texou | well I've gone ahead a bit but... very hard for me telephony. :) |
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| 16:11.26 | Texou | well. could you tell me what tools offers asterisk for softphone? I can then search how use them, but I cannot find their name :x |
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| 16:28.16 | DelphiWorld | hi |
| 16:28.23 | DelphiWorld | what's the libpri svn trunk please? |
| 16:31.49 | tzafrir | DelphiWorld, branhces/1.4 |
| 16:32.32 | tzafrir | http://svn.asterisk.org/svn/libpri/branches/1.4/ |
| 16:32.57 | DelphiWorld | tzafrir: dear, trunk please |
| 16:33.31 | tzafrir | That's the development branch. There's no branch called "trunk" |
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| 16:36.12 | DelphiWorld | tzafrir: ah lol ok |
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| 16:41.40 | DelphiWorld | slaps trumee around a bit with a large trout |
| 16:42.09 | trumee | DelphiWorld: having fun with openwrt |
| 16:42.18 | DelphiWorld | trumee: that's sure |
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| 16:45.32 | x86 | FYI -- the problem with Gtalk no longer working with 1.8.0 was not fixed by 1.8.1 directly. Someone here directed me to a patch on Asterisk's bug tracking system that made it work perfectly again! |
| 16:46.00 | pabelanger | x86: correct, the fix will be in 1.8.2 |
| 16:46.33 | x86 | ETA on 1.8.2? |
| 16:46.56 | x86 | pabelanger: also, what was the underlying issue? |
| 16:47.14 | pabelanger | x86: dot releases are usually 1 month apart |
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| 16:47.44 | pabelanger | x86: Google made a change to the protocol, so we had to do the same |
| 16:48.08 | x86 | those punks... I told them not to do that without first running it by me and making sure I was cool with it! |
| 16:54.08 | x86 | :p |
| 16:54.25 | x86 | shakes fist at the changing of the protocol |
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| 16:57.12 | DelphiWorld | tzafrir: can you give it to me back please? |
| 16:57.44 | DelphiWorld | tzafrir: lost :P |
| 16:58.23 | tzafrir | http://svn.asterisk.org/svn/libpri/branches/1.4/ |
| 16:58.44 | DelphiWorld | thanks tzafrir |
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| 17:28.41 | Texou | hmm I still don't understand while sip.conf and extensions.conf are configured how they link to the softphone |
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| 18:17.58 | greezmunkey | good morning! |
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| 18:49.45 | Texou | well, thanks [TK]D-Fender for useful help. I really needed explanations now I know better things. So thanks :) |
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| 19:05.34 | greezmunkey | lo |
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| 23:13.43 | FlashDeluxe | hi @all can somebody tell me where i can alter the modules zaptel is loading? i want zaptel to load only qozap, but it is loading other modules, too :( |
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