| 00:01.54 | *** join/#asterisk mihinomenest (n=argh@24-180-124-128.dhcp.aldl.mi.charter.com) |
| 00:05.49 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
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| 00:09.40 | *** join/#asterisk hardhatpat (n=pat@c-24-21-226-34.hsd1.mn.comcast.net) |
| 00:10.35 | hardhatpat | is this correct: Dial(IAX2/user:pass@208.73.148.30:4569/1${EXTEN}) ? |
| 00:23.58 | iamthelostboy | hi.. i was asking the other day about how to integrate an asterisk box with an existing digital pbx.. if i want to go between the existing system and the isdn lines, what type of cards do i need? to go out to the isdn lines, i need a Digium b410p, what would i need for the panasonic to think it is just connected to standard isdn lines? |
| 00:24.50 | LemensTS | Just make the Panasonic look like the telco to the asterisk, or vice versa |
| 00:25.19 | JT | iamthelostboy: how many BRIs? |
| 00:25.23 | iamthelostboy | 4 |
| 00:25.39 | JT | well you'd need at least 8 BRI ports then |
| 00:26.11 | iamthelostboy | so the bri port will either go out to the telecom network, or the other way into the panasonic |
| 00:26.37 | iamthelostboy | so its not like FXO or FXS ports? |
| 00:26.55 | *** join/#asterisk mkelly32 (n=pioto@paludis/spork-wielder/pioto) |
| 00:27.33 | JT | right, there's TE and NT mode |
| 00:27.48 | JT | Terminal Equipment it connects to the telco |
| 00:28.08 | JT | Network Terminating it connects to an terminal equipment like the PABX |
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| 00:29.11 | *** part/#asterisk Guest9877 (n=root@189.141.98.158) |
| 00:30.02 | iamthelostboy | so would 2 b410p cards be a suitable way to do that? |
| 00:30.24 | iamthelostboy | i would hope that the connection into the existing pabx would be temporary, maybe 6 months or so |
| 00:31.16 | JT | i would suggest a Junghanns OctoBRI or a Sangoma A500 |
| 00:33.23 | iamthelostboy | so having 2 seperate pci cards would equate to worse performance? 2 cards would be nice, because i could spread the second to another server at a different location later |
| 00:34.06 | JT | zaptel often has more issues when you use multiple cards |
| 00:34.14 | JT | well |
| 00:34.22 | JT | the B410P doesn't use zaptel |
| 00:34.26 | JT | it uses mISDN |
| 00:34.30 | JT | which is just full of issues |
| 00:35.10 | iamthelostboy | so even if i just wanted 4 bri ports, id still be better off with one of the other cards? |
| 00:35.19 | Qwell | those use misdn too |
| 00:35.25 | JT | imho, yes |
| 00:35.28 | JT | err what? |
| 00:35.30 | JT | no |
| 00:35.44 | JT | the OctoBRI can use Zaptel or mISDN |
| 00:36.03 | JT | the A500 uses chan_woomera, and is now reported to work with zaptel too |
| 00:38.15 | coppice | i think the m in mISDN stands for misery |
| 00:38.35 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 00:38.36 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9d8f3c17d257240b) |
| 00:38.44 | JT | or m for "maybe"... "maybe ISDN4LINUX" |
| 00:39.20 | coppice | or megalatency |
| 00:40.36 | JT | iamthelostboy: oh btw, my number 1 recommendation with BRI on asterisk is still to get an external BRI to SIP gateway to terminate the BRI lines |
| 00:40.42 | JT | but it maybe uneconomical |
| 00:40.47 | JT | will be the most reliable though |
| 00:44.18 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 00:49.27 | iamthelostboy | thanks for your help :) |
| 00:49.47 | iamthelostboy | much appricated... answered some questions, raised many more :P |
| 00:50.08 | JT | hehe |
| 00:53.26 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 00:54.46 | hunmonk | are there any channel variables set if a realtime query tanks? i'd like to be able to test for a successfully executed query in the dialplan |
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| 01:51.59 | cplx | hi guys - anyone know of a good Cisco 1 PORT FXO WIC for a 2851? |
| 01:53.58 | *** join/#asterisk absd (n=chatzill@96.393.dsl.mel.iprimus.net.au) |
| 01:55.52 | absd | I've got an issue with a SIP trunk, seems to have silence supression enabled... I've compiled up zaptel with ztdummy and loaded the module and enabled internal_timing (I'm presuming silence supp is causing RTP to not be sent back since if I talk during the ring the audio does start working) ... Any suggestions as to where I should look next? |
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| 02:00.33 | absd | Oh, I'm using asterisk 1.4.21.2 (would upgrading to cvs or 1.6 help?) |
| 02:00.39 | mkelly32 | hi, i've been interesting in playing around with asterisk a bit. i'm wondering if it's reasonable for me to expect to find a voip provider that would just let me hook up to asterisk directly, w/o needing to get any special equipment |
| 02:01.04 | _ShrikE | mkelly32: there are plenty |
| 02:01.07 | _ShrikE | ~istp |
| 02:01.09 | _ShrikE | errr |
| 02:01.12 | _ShrikE | ~itsp |
| 02:01.12 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 02:01.17 | *** join/#asterisk chendy (n=chatzill@58.251.231.216) |
| 02:01.19 | absd | mkelly32: it's possible to make asterisk work wtih basically any voip provider that complies with SIP or IAX standards |
| 02:01.45 | absd | mkelly32: avoid providers with proprietary protocols such as skype |
| 02:01.46 | mkelly32 | ya, i figured that. i read about skype and vonage on voip-info.org |
| 02:02.30 | mkelly32 | so ruled out vonage |
| 02:02.38 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 02:03.12 | iamthelostboy | :( the decision has been made not to switch to voip just yet |
| 02:03.27 | absd | I use pennytel (though I'm in .au) (www.pennytel.com) ... if you're not choosing based on price, go for one that offer IAX trunks rather than SIP trunks :) |
| 02:03.31 | iamthelostboy | we get to try again in a couple of weeks |
| 02:04.36 | mkelly32 | ok |
| 02:04.43 | mkelly32 | well, price is a factor |
| 02:05.01 | mkelly32 | i have a cell phone already, that costs me $0 (my father works for a cellular provider :)) |
| 02:05.31 | *** part/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 02:05.33 | absd | You ain't going to match that price.... Why do you need voip? lol |
| 02:05.48 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.133) |
| 02:05.56 | mkelly32 | to play around w/ it, partly |
| 02:06.23 | mkelly32 | i'm starting to do some tasks tangentially related to our asterisk setup at work, i figure it would help me to know more |
| 02:06.25 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 02:06.29 | mkelly32 | and the besy way is to play around with it |
| 02:06.59 | absd | heaps of free ones that let you call test numbers and internally.... screw around with those and it won't cost you a cent |
| 02:07.39 | *** join/#asterisk jblack (n=jblack@pool-72-79-185-197.sctnpa.east.verizon.net) |
| 02:09.09 | absd | anyone on my internal_timing question? |
| 02:09.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:09.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:09.43 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:14.21 | mkelly32 | `/j #mysql |
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| 02:26.03 | saguti | Hello all |
| 02:26.13 | saguti | I am having trouble with manager in asterisk 1.4.18 |
| 02:26.29 | saguti | Although it appears enabled in manager.conf, it is not opening the port. |
| 02:26.56 | Putzz | did u reload? |
| 02:27.02 | saguti | Yes. |
| 02:48.22 | mchou | I'm totally confused now. If multiple sip devices are registered to the same account, they should all ring on an incoming call, no? |
| 02:48.31 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
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| 02:58.49 | Strom_C | mchou: ahahaaha, no. |
| 02:58.52 | Strom_C | doesnt work like that. |
| 03:02.04 | JT | you cannot share a registration to a single account amongst multiple endpoints |
| 03:02.10 | JT | each must have a unique account |
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| 05:08.10 | legis | I trying to test the extension s, here's my config http://pastebin.com/m5be3a97c, any ideas why is not working? |
| 05:10.59 | jameswf-home | php is sucking my ass ... |
| 05:11.44 | legis | good you are not sucking his :) |
| 05:17.14 | JT | jameswf-home: feels good? |
| 05:19.47 | carrar | legis, you have nothing matching 3045500406 |
| 05:20.52 | carrar | change 's' to 3045500406 or _NXXNXXXXXX |
| 05:21.08 | carrar | or _X. |
| 05:21.30 | drmessano | file php_suckass.dll not found |
| 05:21.45 | drmessano | :( |
| 05:22.55 | legis | carrar: so what's the point of 's'? |
| 05:23.09 | carrar | RTFM :) |
| 05:23.37 | legis | I did, don't get it heh. |
| 05:24.14 | legis | We need to explain extension s. When |
| 05:24.15 | legis | calls enter a context without a specific destination extension (for example, a ringing |
| 05:24.18 | legis | FXO line), they are passed to the s extension. (The s stands for âstart,â as this is where |
| 05:24.21 | legis | a call will start if no extension information was passed with the call.) |
| 05:24.42 | legis | Isn't my sip.conf simulating the FX0 line? |
| 05:24.47 | carrar | There you go |
| 05:24.56 | carrar | But you have a extension |
| 05:25.04 | carrar | extension 3045500406 is incoming |
| 05:25.20 | carrar | So you need to match that |
| 05:26.06 | legis | I see, so in what scenario is no extension pass to the call? |
| 05:27.47 | legis | carrar: Or better, how do I test the code that is in the extension.conf |
| 05:27.49 | legis | ? |
| 05:28.11 | carrar | scroll back |
| 05:28.41 | legis | what do you mean? |
| 05:28.49 | carrar | s/,s,/_NXXNXXXXXX/ |
| 05:28.53 | carrar | err |
| 05:28.57 | carrar | s/,s,/,_NXXNXXXXXX,/ |
| 05:29.01 | carrar | gah |
| 05:29.07 | carrar | s/s,/_NXXNXXXXXX,/ |
| 05:29.09 | carrar | there :) |
| 05:29.16 | legis | oh I mean, how do test it with extension s |
| 05:30.22 | carrar | exten => _NXXNXXXXXX,1,Goto(lesnet-incoming,s,1) |
| 05:30.38 | carrar | heh |
| 05:30.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:31.07 | legis | I know what you mean, I'm trying to follow a example from the book but don't how to test it. |
| 05:32.12 | legis | The books says, try this: |
| 05:32.13 | legis | [incoming] |
| 05:32.13 | legis | exten => s,1,Answer() |
| 05:32.13 | legis | exten => s,n,Playback(hello-world) |
| 05:32.14 | legis | exten => s,n,Hangup() |
| 05:32.53 | legis | but you have to have at least one channel configured, suppose I have one sip phone in my sip.conf, how do I test that? |
| 05:33.38 | legis | If I dial s it works but I don't think that's what i want. |
| 05:33.43 | legis | :D |
| 05:33.47 | mchou | hey, anyone here w/ experience setting asterisk to get rid of telemarketers? |
| 05:34.33 | mchou | basically filter calls that's on a personal blacklist |
| 05:35.04 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 05:35.47 | carrar | not sure how many times you can read it |
| 05:35.48 | carrar | http://lists.digium.com/pipermail/asterisk-users/2003-October/016828.html |
| 05:35.51 | carrar | (legis) |
| 05:36.58 | carrar | mchou, just answer the call saying if you are telemarker please press 2 |
| 05:37.01 | carrar | 2 hangs up |
| 05:37.13 | mchou | carrar: huh? |
| 05:37.19 | legis | carrar: thx |
| 05:37.25 | carrar | or use the zaptel |
| 05:37.26 | carrar | err |
| 05:38.06 | mchou | carrar: the whole point is not to answer telemarketer calls.... |
| 05:38.19 | carrar | Zapateller |
| 05:38.39 | mchou | examine caller ID and give them "Congestion" |
| 05:39.40 | carrar | http://www.voip-info.org/wiki-Asterisk+cmd+Zapateller |
| 05:40.24 | mchou | in any case I'm wonder if I can use asterisk to daisychain off my ITSP, so to speak, peek at caller ID's and if the caller ID is blacklisted, just hang up (using asterisk on my end) |
| 05:40.31 | mchou | wondering* |
| 05:41.14 | mchou | and if it's a "personal call," still have my SIP phone connected to the ITSP (w/o side effects) |
| 05:41.22 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:41.42 | mchou | i.e. no handset is ever connected to my asterisk box |
| 05:42.08 | carrar | run all incoming calls through a AGI script to compair caller id to whats in your postgreSQL database of dids you want to block |
| 05:42.24 | carrar | cake |
| 05:42.29 | mchou | carrar: yup, pretty much |
| 05:42.50 | carrar | pretty simple perl script |
| 05:42.52 | mchou | carrar: I'd just use grep and asterisk backticks though :) |
| 05:42.58 | carrar | no |
| 05:43.02 | *** join/#asterisk stoffell (n=stoffell@d51A4DC2A.access.telenet.be) |
| 05:43.02 | carrar | thats bad |
| 05:43.06 | mchou | why not? |
| 05:43.15 | mchou | what's wrong with it? |
| 05:43.20 | carrar | well I know there was a lot of memory leaks due to that |
| 05:43.25 | carrar | eventually crashing * |
| 05:43.26 | mchou | huh?? |
| 05:43.30 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 05:43.40 | mchou | backticks has mem leaks? |
| 05:43.46 | carrar | just write a AGI scrpit and execute it |
| 05:44.04 | mchou | AGI script sounds like a PITA |
| 05:44.06 | carrar | backticks used too |
| 05:44.12 | mchou | nothing wrong with using grep |
| 05:44.13 | carrar | not sure if it still does or not |
| 05:44.22 | carrar | but thats a crappy way to do it |
| 05:44.22 | mchou | easier to maintain |
| 05:44.26 | carrar | not really |
| 05:44.32 | carrar | agi is the simplest |
| 05:44.37 | carrar | faster |
| 05:44.39 | carrar | cleanest |
| 05:44.43 | mchou | I dont see why it's any crappier than using perl or some such |
| 05:44.53 | carrar | whatever |
| 05:44.56 | carrar | do it then |
| 05:46.32 | mchou | and there is no need to use sql, another plus |
| 05:46.38 | [TK]D-Fender | AGI is not a language. |
| 05:47.46 | mchou | so there are no issues with just using asterisk as an intelligent filtering agent with no phones connected at all? |
| 05:48.21 | [TK]D-Fender | mchou: A call is a call is a call. What part don't you get? You can use * as an alarm clock if you feel like it. |
| 05:48.24 | mchou | only a dial plan that peeks at caller ID and whether or not decide to hang up? |
| 05:48.38 | [TK]D-Fender | mchou: Yes, of course you can do this |
| 05:49.18 | mchou | I just want to make sure there are no unintended side effects |
| 05:49.55 | [TK]D-Fender | mchou: You MIGHT accidentally learn something. Not a guarantee, but its possible. |
| 05:50.01 | mchou | lol |
| 05:50.06 | [TK]D-Fender | mchou: You'll have to be careful |
| 05:50.30 | carrar | hahah |
| 05:50.30 | mchou | dont want to learn the *hard* way like it drops all my calls :) |
| 05:50.54 | carrar | When I first learned what I could do with AGI, the world changed |
| 05:51.58 | carrar | and it will for you too! |
| 05:52.02 | carrar | You can do it |
| 05:52.06 | mchou | screw AGI |
| 05:52.11 | carrar | You have the power within yourself to make that change |
| 05:52.26 | mchou | way overkill for what I want the box to do |
| 05:52.39 | carrar | Sounds like asterisk is ovekill for you |
| 05:53.14 | mchou | not when I get ~5 telemarketer calls/hr |
| 05:54.32 | [TK]D-Fender | mchou: You can do this without AGI. Everything depends on how * will inquire about whether it should accept the call or not. |
| 05:56.38 | mchou | [TK]D-Fender: I wasn't the one who proposed doing this with AGI :) |
| 05:57.18 | [TK]D-Fender | mchou: Well you also haven't sted a single idea about how you want to implemennt the good/bad list |
| 05:57.28 | mchou | sure I did |
| 05:57.36 | [TK]D-Fender | mchou: So people suggested a clear tool to let you make up your mind OUTSIDE the constraints of * |
| 05:57.37 | mchou | scroll up |
| 05:58.39 | [TK]D-Fender | mchou: I can't seem to see it. How exactly do you have in mind? |
| 05:59.02 | mchou | grep and asterisk backticks |
| 05:59.20 | mchou | and a simple blacklist of callerIDs |
| 05:59.28 | mchou | simple enough |
| 06:00.22 | [TK]D-Fender | mchou: grep is at *nix CLI. So how do you indend to do that? And then how do you intend to pass the reult on to * so it can act upon it? |
| 06:00.49 | mchou | [TK]D-Fender: which part of BACKTICKS didnt you grok? |
| 06:01.32 | [TK]D-Fender | ah, 3rd party. |
| 06:01.49 | mchou | I swear |
| 06:01.59 | mchou | fucking read b4 jumping on ppl |
| 06:02.01 | [TK]D-Fender | mchou: Because this is not even a common addon let alone part of * itself |
| 06:02.15 | [TK]D-Fender | mchou: And this wasn't jumping on you. |
| 06:02.16 | mchou | lol |
| 06:02.38 | mchou | read the conversation again and put yourself in my shoes |
| 06:03.01 | carrar | please be more demanding when asking for help |
| 06:03.08 | [TK]D-Fender | mchou: Check your persecution complex at the door along with your jacket :) |
| 06:03.12 | mchou | lol |
| 06:05.20 | [TK]D-Fender | mchou: http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks |
| 06:05.30 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 06:05.30 | [TK]D-Fender | mchou: And even better all the wiki links to it are broken. |
| 06:05.42 | [TK]D-Fender | mchou: Did you find it and get it to compile properly? |
| 06:06.07 | mchou | wiki is not end all be all |
| 06:06.23 | mchou | and it compiles properly and works just fine, thank you |
| 06:06.40 | [TK]D-Fender | mchou: No, it isn't. One might wonder how far off the map the solution is though when the common places to list it come up looking dead |
| 06:06.43 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 06:07.29 | [TK]D-Fender | mchou: Well yippy-kai-yay. Congratulations. You've figured all this out and you aren't sure if * can pick up the call and hanup on them. SMRT :) |
| 06:08.19 | mchou | [TK]D-Fender: dude, the questiion I asked was if there are side effects I should be aware of |
| 06:08.24 | [TK]D-Fender | "I just perfected my FTL space drive... now if only I could figure out this coffee-maker" |
| 06:08.52 | mchou | [TK]D-Fender: I already figured out asterisk can hang up on ppl |
| 06:08.57 | mchou | lol |
| 06:09.03 | [TK]D-Fender | mchou: Side effects of what? get call. Do Stuff. Struff says "hang up". Ain't Raw-Cat Science |
| 06:09.21 | mchou | sigh.... |
| 06:09.23 | [TK]D-Fender | mcYeah, part of answering calls usualy leads to ENDING them :p |
| 06:09.36 | creativx | side effects might be even more unsolicited calls |
| 06:09.38 | [TK]D-Fender | ~whee |
| 06:09.38 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 06:10.12 | [TK]D-Fender | mchou: What are your block/allow criteria? |
| 06:10.13 | mchou | creativx: yeah, as long as they still use the same caller ID it's a dont care :) |
| 06:10.27 | mchou | criteria?? |
| 06:10.36 | [TK]D-Fender | mchou: Fixed "bad" list? |
| 06:10.42 | mchou | why you making this overly complicated? |
| 06:10.50 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:11.05 | carrar | put your number on the do not call list |
| 06:11.09 | mchou | I know the # come from a telemarketer. I put in in my blacklist. lol |
| 06:11.16 | [TK]D-Fender | mchou: I havent made anything complicated I'm jsut threshing out the decision making process. |
| 06:11.31 | [TK]D-Fender | mchou: Ok, and for ones you don't know? |
| 06:11.31 | mchou | s/in in/it in |
| 06:11.40 | creativx | put it in the astdb |
| 06:11.56 | carrar | creativx, thats too complicated for him :) |
| 06:12.05 | creativx | i know, i want to be difficult |
| 06:12.05 | creativx | :D |
| 06:12.09 | mchou | for ones I dont know I'll just have to drin and bear it 1st time |
| 06:12.09 | [TK]D-Fender | mchou: mchou How many? You're making it sound like 1/2 numbers. |
| 06:12.20 | mchou | grin* |
| 06:12.36 | mchou | [TK]D-Fender: how many what? |
| 06:12.43 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:12.45 | [TK]D-Fender | mchou: numbers to block so far? |
| 06:13.20 | mchou | bout 7. Each generates close to 1 call/hr |
| 06:13.43 | mchou | so the blacklist is small but junk call volume large |
| 06:13.43 | carrar | 6 was ok |
| 06:13.45 | [TK]D-Fender | mchou: Then all of this is a sad waste. this is 7 lines of dialplan that coudl be copy/pasted. |
| 06:13.47 | carrar | 7 not doable |
| 06:14.03 | [TK]D-Fender | mchou: No external apps or anything |
| 06:14.11 | mchou | [TK]D-Fender: lol. I'm sure that # will grow :) |
| 06:14.22 | creativx | 7 would exceed the integral limit of astdb. |
| 06:14.24 | creativx | does not compute! |
| 06:14.36 | creativx | so mchou, how do you prefer adding numbers to this wonderful list of yours |
| 06:14.39 | [TK]D-Fender | mchou: here's an elephant gun, now go find a new ant-hill |
| 06:15.08 | mchou | creativx: what do you mean? I add it manually :) |
| 06:15.09 | creativx | if i got 5-6 junk calls an hour i sure would like to have a quick way to blacklist |
| 06:15.30 | creativx | i am thinking AMI/http and adding stuff to the astdb |
| 06:15.35 | mchou | sigh |
| 06:15.38 | carrar | vi dialplan, asterisk -rx "dialplan reload" |
| 06:15.55 | creativx | is it not complex enough mchou? |
| 06:16.01 | creativx | how about using a sqllite backend |
| 06:16.08 | mchou | who is even running a web server? you guys crack me up :) |
| 06:16.13 | creativx | asterisk is |
| 06:16.20 | creativx | natively |
| 06:16.21 | creativx | ... |
| 06:16.38 | mchou | I'm not running no stink asterisk web |
| 06:16.53 | creativx | why not |
| 06:16.57 | creativx | you should, it would solve your problems |
| 06:17.00 | mchou | this is on openwrt |
| 06:17.02 | mchou | lol |
| 06:17.03 | creativx | sending AMI commands via http |
| 06:17.31 | creativx | how about opening a tcp socket to the AMI the |
| 06:17.32 | creativx | n |
| 06:17.36 | mchou | you dudes sure are crazy :) |
| 06:17.37 | creativx | in perl thats 4 lines |
| 06:17.44 | creativx | which enviroment are you running in |
| 06:17.53 | mchou | enviroment?? |
| 06:17.54 | carrar | static entries in your dialplan is what you should use |
| 06:18.01 | carrar | end of conversation |
| 06:18.01 | mchou | hahah!! |
| 06:18.36 | creativx | yes, from which kind of workstation to you intend to update your list from |
| 06:18.38 | [TK]D-Fender | carrar: Thats what I'vwe just assessed |
| 06:18.57 | creativx | a commaseparated variable in extensions.conf |
| 06:18.58 | mchou | creativx: if you arent familiar with openwrt you can look it up on google |
| 06:18.58 | carrar | he needs to hear from everyone I think |
| 06:19.03 | carrar | hear it |
| 06:19.08 | creativx | mchou: this is #asterisk, thats what i am familiar with |
| 06:19.28 | mchou | creativx: who cares what workstaion, as long as it has ssh? |
| 06:19.30 | *** join/#asterisk Miranda (n=miranda@wikimedia/Miranda) |
| 06:19.58 | carrar | if you are on a wrt, don't bother with compiling 3rd party apps |
| 06:19.58 | creativx | you obviosly |
| 06:20.23 | creativx | i would want the ability to update the blacklist as quick and easy as it can be done |
| 06:20.52 | creativx | you could even set up a magic extension |
| 06:20.57 | creativx | where you can transfer your spammers to |
| 06:20.57 | mchou | you guys are really getting ridiculous |
| 06:21.04 | creativx | and take care of it in the extension alone |
| 06:21.04 | carrar | sure are |
| 06:21.17 | creativx | you are ridiculous sir |
| 06:21.19 | mchou | it's not like I get even 1 new telespammer a day |
| 06:21.28 | creativx | how about the day you do? |
| 06:21.29 | carrar | How do we get anything done in Asterisk you may b e asking yourself |
| 06:21.34 | creativx | exten => 666,1,ZAP TELEMARKETER |
| 06:21.35 | mchou | it's the same crooks spamming me |
| 06:21.54 | creativx | we know that |
| 06:22.02 | creativx | we've given you at least 4 or 5 ways to handle it |
| 06:22.07 | creativx | ranging from uber simple to overly complex |
| 06:22.10 | [TK]D-Fender | ok, well its be fun. s/fun/roflmaostupid/ |
| 06:22.14 | mchou | yeah |
| 06:22.23 | creativx | you fail to acknowledge that it seems |
| 06:22.27 | creativx | calling us crazy |
| 06:22.42 | mchou | I never called you guys crazy |
| 06:22.51 | creativx | potato potatoe |
| 06:22.56 | mchou | I called it ridiculous and overkill |
| 06:23.03 | carrar | exten => s/5557771212,1,hangup |
| 06:23.06 | [TK]D-Fender | [02:17]<mchou>you dudes sure are crazy :) |
| 06:23.08 | mchou | there a distinct difference |
| 06:23.09 | [TK]D-Fender | ^ |
| 06:23.12 | creativx | hah |
| 06:23.20 | [TK]D-Fender | load chan_busted.so |
| 06:23.29 | creativx | you need to stop worrying about every damn detail unrelated to solving your problem mchou |
| 06:23.30 | mchou | lol |
| 06:23.32 | creativx | are you seeking a solution |
| 06:23.38 | [TK]D-Fender | ok, I'm off, good luck, you nut-bars are going to need it. |
| 06:23.39 | creativx | or do you wish to discuss how difficult it is to resolve it |
| 06:23.44 | creativx | cya [TK]D-Fender =) |
| 06:24.10 | carrar | Oh look what you've gone and done now mchou |
| 06:24.26 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:24.40 | *** part/#asterisk Miranda (n=miranda@wikimedia/Miranda) |
| 06:25.04 | creativx | i think my idea of setting up a 666 extension was a good idea |
| 06:25.12 | creativx | that way you can pick the telemarketers number from CID, add it to astdb |
| 06:25.21 | creativx | and on all incoming DID's you check the DID for an entry |
| 06:25.25 | creativx | and drop the call if match |
| 06:25.36 | carrar | Nmap 4.68 released, I should go compile it |
| 06:26.39 | creativx | i should get back to whatever I am supposed to be doing |
| 06:43.11 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
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| 07:10.29 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 07:15.10 | SwK | http://www.youtube.com/watch?v=qOtoujYOWw0 |
| 07:18.51 | *** join/#asterisk jamaal (i=433121ef@gateway/web/ajax/mibbit.com/x-7311d4d5e19eb759) |
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| 07:41.21 | *** join/#asterisk tapic (n=tap@88.255.77.200) |
| 07:42.20 | tapic | they issued a patch for the early bridging issue(http://bugs.digium.com/view.php?id=13200). does anyone know how to install a patch for *? |
| 07:45.19 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 07:50.37 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 07:51.33 | tapic | any ideas of how a patch can be applied to asterisk? |
| 07:53.03 | gnorbert | Hi, I have an asterisk server. I connect to it's conference by linphone from two computers. The speak can be heard in one way, but in the other way nothing can be heard |
| 07:53.19 | gnorbert | Does somebody have an idea, what can be the problem? |
| 07:53.41 | gnorbert | Both computers use alsa, not oss. |
| 07:56.47 | JT | alsa/oss is irrelevant |
| 07:57.02 | JT | sounds like the RTP traffic is not making it through in one direction |
| 07:57.09 | JT | check the usual suspects: NAT and firewalls |
| 07:57.35 | gnorbert | At linphone.org it was written, that linphone can't handle well oss. |
| 07:58.16 | gnorbert | NAT? |
| 07:59.12 | JT | this isn't a linphone support channel |
| 08:00.25 | gnorbert | I know. :) |
| 08:00.51 | gnorbert | And the problem is not with linphone, that's why I wanted to say it, before somebody writes to check this. :) |
| 08:02.24 | JT | then what does oss/alsa have to do with it? :) |
| 08:03.10 | gnorbert | Nothing, that's why I wrote, to avoid people, who think that's the problem. But I guess I just made it wrong. :) |
| 08:04.24 | JT | so look into any NAT or firewalls between the 2 machines |
| 08:04.42 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:04.44 | gnorbert | Ok, Thanks I try. :) |
| 08:09.55 | gones | Hello, can you give me some advice about the asterisk IVR model ? I want to build a IVR server . |
| 08:13.31 | JT | you can setup IVRs in asterisk. |
| 08:14.25 | gones | Yeah, I want to know the performance . |
| 08:14.29 | *** join/#asterisk BeeBuu (n=beebuu@219.130.245.145) |
| 08:14.56 | BeeBuu | anyone know is callerid(rdnis) work in ISDN? |
| 08:15.06 | JT | can you try asking a better question, gones |
| 08:17.18 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 08:17.31 | gones | Sorry, My english is very weak , I can't make the question clearly . |
| 08:19.52 | BeeBuu | gones: would you tell us where are you come from? |
| 08:20.09 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:20.10 | gones | China |
| 08:20.28 | gones | And you ? |
| 08:20.39 | BeeBuu | æä¹æ¯ |
| 08:20.50 | gones | good ! |
| 08:21.07 | BeeBuu | æå¥é®é¢å¯ä»¥å°çªé®æï¼ä¹è®¸æå¯ä»¥åçãprvite chat please. |
| 08:21.13 | *** join/#asterisk torrikft (n=afj@m85-94-184-212.andorpac.ad) |
| 08:21.37 | Nugget | watashi wa baka gaijin desu. |
| 08:21.44 | gones | BeeBuu: Thank you very much . Could you show me your MSN? |
| 08:24.23 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 08:29.48 | MaliutaLap | personally I find the chinese government all together too liberal, Nth Korea is _much_ more sensible ;) |
| 08:30.32 | gnorbert | No firewall, no NAT. Any other idea? (Communication works only one way in a meetme conference, server doesn't already record it.) |
| 08:32.07 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:32.21 | gones | MaliutaLap: Haha ? |
| 08:47.16 | *** join/#asterisk tyldis (n=vidar@tyldum.com) |
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| 08:48.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:15.03 | absd | I've got an issue with a SIP trunk, seems to have silence supression enabled... I've compiled up zaptel with ztdummy and loaded the module and enabled internal_timing (I'm presuming silence supp is causing RTP to not be sent back since if I talk during the ring the audio does start working) ... Any suggestions as to where I should look next? |
| 09:23.06 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 09:47.05 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:47.25 | dominic1 | if I execute a goto am I able to see the context I were before? |
| 09:56.57 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 10:00.54 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 10:01.00 | PakiPenguin | ~centos52bug |
| 10:01.01 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 10:01.27 | *** join/#asterisk rdgr (n=rich@87-194-164-154.bethere.co.uk) |
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| 10:09.13 | *** join/#asterisk WSN (n=bart@91.179.160.28) |
| 10:09.44 | WSN | hi folks, somebody here who can help me with /dev/zap/pseudo timing issues? |
| 10:10.16 | WSN | or better, Background() / Play() choppy sound issues |
| 10:10.53 | WSN | ~centos52bug |
| 10:10.54 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 10:14.17 | *** join/#asterisk bartounet (n=bart@LMontsouris-152-61-27-191.w80-13.abo.wanadoo.fr) |
| 10:14.20 | bartounet | Bonjour |
| 10:14.38 | bartounet | y a t'il des francais qui peuvent m'aider sur la configuration d'asterisk? |
| 10:18.16 | bartounet | ? |
| 10:18.22 | bartounet | some french? |
| 10:23.15 | bminish | hello we are having an issue where calls fail to go though to agents at times, the error message that goes along with this is this: app_queue.c: The device state of this queue member, Agent/1018, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 10:23.36 | bminish | any suggestions on where to start digging? |
| 10:33.18 | bartounet | hello |
| 10:37.56 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:45.17 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
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| 10:48.40 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:58.18 | *** join/#asterisk SanityIO (n=SanityIO@77.242.106.50) |
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| 11:19.11 | WSN | can anybody help me, i'm experiencing crappy sound on Background and or Play, i think it might have to do something with timeing issues but all hardware seems to be fine |
| 11:19.28 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
| 11:23.26 | styelz | i had a weid sound issue on ubuntu |
| 11:25.26 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.121) |
| 11:30.07 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 11:33.12 | tompaw | hello |
| 11:33.34 | tompaw | how can I manually response with a SIP error (code + message)? |
| 11:34.16 | tompaw | which app does that? |
| 11:42.07 | l0verb0y | !seen gerscell |
| 11:43.11 | tompaw | there's nothing in the application list :-( |
| 11:44.56 | WSN | styelz: what was the issue that you had? (mine turns on gentoo but maybe its related) |
| 11:45.03 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:45.24 | styelz | cant remember exactly what i did now |
| 11:45.47 | styelz | its was to do with the sound files |
| 11:46.05 | styelz | i deleted the .wavs... i think .. or 1 of them |
| 11:46.11 | tompaw | I cannot believe asterisk cannot do it. |
| 11:46.17 | styelz | 1 of the formats |
| 11:46.45 | WSN | my sound files are ok... that thought accoruded to me to, but i downloaded them to an other pc, and there everything was fine |
| 11:46.53 | WSN | (i have only .gsm files btw) |
| 11:51.27 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 11:52.33 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:53.05 | styelz | WSN: yea it was the GSM files i had issue wih |
| 11:53.09 | styelz | onlu |
| 11:53.11 | styelz | only |
| 11:53.17 | styelz | damn this k/b |
| 11:53.24 | WSN | lolz |
| 11:53.45 | styelz | the sound is distorted |
| 11:53.51 | styelz | thats what it sounds like |
| 11:54.01 | styelz | very crackly |
| 11:58.29 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 11:59.10 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 12:05.54 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:06.16 | WSN | with me it's not cracky, it just like part of the file is missing/skipped |
| 12:06.23 | *** part/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 12:06.52 | WSN | like you are on a bad connection an 50% of the packets are dropped... but without gabs in the sound |
| 12:07.01 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:07.04 | WSN | don't know if the descripion is clear |
| 12:08.17 | tompaw | which app sends SIP error 400? |
| 12:08.22 | tompaw | I need to manually trigger that error. |
| 12:09.54 | WSN | 400 is that extention not found or something? most of the time its dial that will send it presume. but i don't think you can mannully trigger it |
| 12:10.12 | [TK]D-Fender | WSN: That'd be "404" |
| 12:10.22 | [TK]D-Fender | tompaw: PASTEBIN <---- |
| 12:10.22 | tompaw | 400 is Bad Request |
| 12:10.43 | tompaw | [TK]D-Fender: what would I paste? I am asking for the app name. |
| 12:10.53 | [TK]D-Fender | tompaw: and I believe * responds with that to OPTIONS packets it receivec |
| 12:11.03 | tompaw | Congestion sends 503... |
| 12:11.10 | [TK]D-Fender | tompaw: just look at what it's responding TO. |
| 12:11.17 | tompaw | ok, pastebin then. |
| 12:11.39 | [TK]D-Fender | tompaw: yes, but congestion is an OPTION. 400 is a hard response, not something an APP can trigger |
| 12:12.10 | tompaw | [TK]D-Fender: so what you're saying is - it's impossible? |
| 12:12.14 | tompaw | http://pastebin.com/m24abcec9 |
| 12:12.25 | tompaw | in the line 5 I need to send SIP/400 instead of SIP/503 |
| 12:13.41 | tompaw | please don't tell me that I have to recompile it and manually change the source :/ |
| 12:15.29 | tapic | is there way to return a value to the AGI environment from the macro which is triggered in AGI using Dial,M(testmacro) ? |
| 12:15.37 | tompaw | [TK]D-Fender: so no way? |
| 12:16.54 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:16.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:17.32 | tapic | I want evaluate the result of the Dial operation in AGI. Was it CHANUNAVAIL,BUSY,CONGESTION etc... |
| 12:18.40 | tompaw | [TK]D-Fender: if not, I'll have to set up a separate trunk with an openser doing nothing but responding with 400 ;-) |
| 12:18.48 | WSN | tapic: i thougt that was possible have youc checked voip-info.org? |
| 12:19.24 | WSN | i believe i did it somewhere, but it's a long time ago and i don't recall how anymore |
| 12:19.24 | bartounet | y a t'il des francais sur le forum? |
| 12:19.40 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.124.56) |
| 12:23.16 | bartounet | coud you help me? excsue me i am french |
| 12:23.30 | bartounet | i try to record a message |
| 12:25.14 | bartounet | astrisk answer say hello world and hang up immeditely |
| 12:29.29 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 12:31.48 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 12:35.53 | [TK]D-Fender | bartounet: www.pastebin.com . Veuiller-nous le montrer |
| 12:36.05 | [TK]D-Fender | tompaw: pastebin the CLI OUTPUT with SIP debug. |
| 12:36.38 | [TK]D-Fender | tompaw: And if you want to change what Congestion sends you need to change it in "C" and recompile |
| 12:36.57 | [TK]D-Fender | tapic: thats what the DIALSTATUS variable is for |
| 12:37.17 | [TK]D-Fender | tompaw>please don't tell me that I have to recompile it and manually change the source :/ <- YES. TFB |
| 12:37.43 | creativx | morning [TK]D-Fender =) |
| 12:39.56 | [TK]D-Fender | creativx: So did our friend move along quietly after I left? :) |
| 12:40.10 | creativx | i can't recall hehe |
| 12:40.12 | creativx | i think i went back to coding |
| 12:40.21 | creativx | quietly ignoring him |
| 12:41.45 | [TK]D-Fender | :) |
| 12:42.10 | tapic | yes what a newbie question... thanks |
| 12:42.49 | [TK]D-Fender | tapic: helps when you read your applications INSTRUCTIONS. |
| 12:43.18 | *** join/#asterisk The-Bat (n=karthik@203.199.114.33) |
| 12:43.50 | tapic | is it then possible to pass a variable from dial macro to agi? |
| 12:44.02 | tapic | different then dialstatus... |
| 12:44.45 | [TK]D-Fender | tapic: you don't "pass" variables to AGI. AGI is a way of controlling your call that is pretty much just like BEING int he dialplan and executing things in YOUR order |
| 12:45.02 | [TK]D-Fender | tapic: Of course AGI allows you access to ALL channel variables. |
| 12:45.33 | tapic | but when I dial with M then it is a different channel. right? |
| 12:45.47 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 12:46.03 | tapic | so I cant access the variables in that channel since agi is controlling the other channel. |
| 12:47.23 | [TK]D-Fender | tapic: I have no idea where you're going with this, but YES, the channel you are dialing INHERITS variables from the calling channel. Go read about it on the WIKI |
| 12:47.50 | [TK]D-Fender | tapic: and in M() I believe you are in the SAME channel you started from |
| 12:48.09 | [TK]D-Fender | tapic: remember that it is executed BEFORE you are bridged. |
| 12:48.10 | WSN | ok folks i got my sound ok again |
| 12:48.27 | tapic | oh then I can set a variable in the macro and read it back in agi.. let me try, thanks. |
| 12:48.36 | WSN | apperently zaptel timeing comes from the first device it loads |
| 12:48.58 | WSN | timing was corrupt => sound was corrupt |
| 12:49.19 | WSN | changed the priority in loading the devices... now it works |
| 12:49.26 | WSN | so happey |
| 12:49.34 | WSN | so happy |
| 12:49.35 | WSN | :p |
| 12:52.55 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:54.24 | mchou | If I'm running asterisk on my own box, and registered to my ITSP's asterisk box, what the summary of "iax2 show peers" supposed to say? mine says "0 iax2 peers [0 online, 0 offline, 0 unmonitored]" |
| 12:54.41 | mchou | can that possibly be even correct? |
| 12:55.42 | mchou | or am I just misunderstanding the command "iax2 show peers?" |
| 12:56.47 | [TK]D-Fender | mchou: And what IAX2 ITSP have you signed up with and configured? |
| 12:57.19 | mchou | [TK]D-Fender: why would that make a differnce? |
| 12:57.43 | [TK]D-Fender | mchou: I asked a simple question, can you please jsut answer it as I asked. |
| 12:57.51 | mchou | "iax2 show registry" reports all the correct things |
| 12:58.21 | [TK]D-Fender | mchou: "register" has nothing to do with wether or not you set up a PEER to send/receive calls against |
| 12:59.17 | mchou | I set my iax2 conf as a user since I'm using it only for receiving calls for blacklisting purposes |
| 12:59.33 | mchou | I meant iax.conf |
| 13:00.16 | mchou | well, at least the stanza associated with my ITSP |
| 13:00.56 | [TK]D-Fender | mchou: My statement stands. If you don't see a peer listed in "iax2 show peers", then you didn not set one up (or properly) |
| 13:01.03 | mchou | but even when type=peer it shows the same thing |
| 13:01.23 | [TK]D-Fender | mchou: Or haven't applied your changes |
| 13:01.24 | mchou | for "iax show peers" |
| 13:01.35 | [TK]D-Fender | mchou: pastebin your iax.conf masking only passwords |
| 13:02.30 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 13:04.37 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 13:05.13 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:06.52 | mchou | http://pastebin.com/m77ff969d |
| 13:07.21 | mchou | This was using type=peer |
| 13:08.29 | [TK]D-Fender | mchou: looks largely valid. |
| 13:08.40 | [TK]D-Fender | mchou: like that is should probably at least show up. |
| 13:08.56 | [TK]D-Fender | s/is/it/ |
| 13:09.47 | mchou | well, I dont understand why it shows up as "iax2 show registry" but not as "iax2 show peers" |
| 13:10.09 | [TK]D-Fender | mchou: Of course you've done far more masking than I requested and wonder if three's something I should be seeing otherwise |
| 13:10.12 | mchou | it's not like I messed up the firewall port forwarding or something |
| 13:10.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.03 | mchou | well, I know for a fact the ITSPs * box works :) |
| 13:11.39 | mchou | works in the sense I cand send/receive calls |
| 13:11.46 | mchou | can* |
| 13:12.46 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 13:13.49 | mchou | and the asterisk docs are a bit confusing regarding "Congestion" or "Busy" applications |
| 13:14.47 | mchou | I'd like to send a busy tone out-of-band (i.e. w/o actually using Answer in some form) |
| 13:14.52 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 13:15.05 | tompaw | Got SIP response 400 "BadRequest" back from 77. |
| 13:15.07 | tompaw | yeah!! |
| 13:15.08 | tompaw | ;-) |
| 13:15.09 | mchou | * docmentation recommends Playtone |
| 13:16.11 | mchou | and apparently Playtone is only in-nabd |
| 13:16.16 | mchou | band* |
| 13:16.45 | tapic | <[TK]D-Fender> : I was actually trying to measure ANSWEREDTIME variable. which was already built in! but if the caller hang ups during the call, I can not access this variable since the channel is garbaged. |
| 13:17.09 | *** join/#asterisk nikolaev (n=nikolaev@78.83.145.98) |
| 13:20.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:21.16 | *** join/#asterisk moy (n=moy@nat/ibm/x-87227b68d80dc7b3) |
| 13:22.01 | nikolaev | Hi All |
| 13:22.09 | [TK]D-Fender | mchou: Something else is wrong. If you don't see the entry for the peer and your changes have tried to be reloaded then something else has gone wrong. |
| 13:22.16 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 13:22.32 | nikolaev | does anybody know is the GUI compatible with realtime mysql scenario ? |
| 13:23.35 | mchou | [TK]D-Fender: I'm supposed to see 1 peer, right? and should it say monitored or unmonitored (if everything is working right)? |
| 13:23.54 | mchou | the 1 peer being my ITSP |
| 13:24.26 | [TK]D-Fender | mchou: if you did it properly. But then again you showed me such a "washed" entry that I can't trust that the actual entry looks right at all. |
| 13:26.34 | mchou | hmm, should have checked the logs first :) |
| 13:26.37 | mchou | WARNING[8375] chan_iax2.c: Unable to open IAX timing interface: No such file or directory |
| 13:26.43 | mchou | what's that mean? |
| 13:27.57 | [TK]D-Fender | mchou: got "trunk=yes" in there somewhere? |
| 13:28.04 | mchou | nope |
| 13:28.10 | mchou | do I need it? |
| 13:28.23 | [TK]D-Fender | mchou: Well it IS just a warkning. But IAX2 trunk mode requires zaptel for timing. |
| 13:28.54 | *** join/#asterisk emiller (n=ed@static-98-117-7-2.hrbgpa.fios.verizon.net) |
| 13:29.54 | mchou | just double checked. no trunk=yes |
| 13:30.04 | mchou | in iax.conf |
| 13:30.06 | [TK]D-Fender | mchou: Again, just a warning. |
| 13:30.19 | mchou | yeah |
| 13:30.31 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:31.17 | nikolaev | [TK]D-Fender, sorry for disturbing you but are you are of my question about the *GUI ? |
| 13:31.36 | nikolaev | *aware* |
| 13:32.30 | [TK]D-Fender | nikolaev: You should be aware of the parts of extensions.conf that the GUI generates. so SOME of your dialplan can be in there. but for users.conf, etc, no. |
| 13:33.37 | nikolaev | okay, that helps me get a decision :) |
| 13:34.07 | nikolaev | I may migrate to asterisk + postgresql + druid :) |
| 13:34.40 | nikolaev | I think that one is doing realtime changes |
| 13:35.22 | nikolaev | any points of view on this one ? |
| 13:36.06 | [TK]D-Fender | nikolaev: GUI's are not supported here. |
| 13:36.08 | bminish | anyone any ideas on this ? app_queue.c: The device state of this queue member, Agent/1018, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 13:36.22 | [TK]D-Fender | bminish: go read UPGRADE.TXT like its telling you to |
| 13:36.40 | bminish | where do I find it ? |
| 13:36.45 | nikolaev | oh okay, apologize about that |
| 13:37.38 | [TK]D-Fender | bminish: int eh docs folder of your source tarball |
| 13:43.09 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 13:43.13 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 13:43.26 | bminish | Ok found soemting about seting limitonpeer = yes in [general] is that in the sip config or in the Queue config. UPGRADE.txt does not say which |
| 13:44.25 | *** join/#asterisk Levonk (n=lk@adsl-75-62-140-223.dsl.lsan03.sbcglobal.net) |
| 13:45.45 | bminish | hmm, it's already configured (it's in sip ) as they say it should be |
| 13:45.51 | [TK]D-Fender | bminish: "The device state". which fine(s) configure DEVICES? |
| 13:45.59 | [TK]D-Fender | files* |
| 13:46.36 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 13:47.57 | bminish | these are sip devices. issue here is that the above message is accompanied by a call with no audio and is happening about 1 in three times for calls leaving the Queue |
| 13:48.49 | [TK]D-Fender | bminish: Lack of audio is a completely separate issue for which you should be describing in full detail. |
| 13:49.43 | bminish | 2 out of three calls go though, the above error message goes with the ones that don't there are free agents when this is occurring |
| 13:50.20 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 13:52.18 | bminish | I have been trying to debug this for the last 6 hours, we have full logging on an am tailing the logs, I would appreciate any suggestions you may have on how to get a better idea of what is going on version by the way is 1.4.20-1 |
| 13:53.02 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:53.05 | *** join/#asterisk phatjoe (n=brucedav@dsl-241-19-92.telkomadsl.co.za) |
| 13:53.06 | [TK]D-Fender | bminish: PASTEBIN is your friend. |
| 13:53.08 | [TK]D-Fender | ~pb |
| 13:53.09 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:53.10 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:53.54 | tapic | [TK]D-Fender : if caller hangs up, the channel is garbaged. how to get the ANSWEREDTIME variable then? |
| 13:54.05 | phatjoe | i got a very strange issue with asterisk 1.4.21.1 |
| 13:54.11 | bminish | [TK]D-Fender, I know about pasetbin. I NEED to first find out what I should be pasetbinning, I doubt you want 100's of kb of logs ;-) |
| 13:54.25 | phatjoe | every now and again all sip registrations die |
| 13:54.55 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 13:54.55 | phatjoe | asterisk also seems to hang when this happens... like it doesnt accept reload commands |
| 13:55.16 | phatjoe | it seems like this only happens when i have register commands in my sip.conf |
| 13:55.22 | *** join/#asterisk rabby (n=rabby@p4FCE9BB0.dip0.t-ipconnect.de) |
| 13:55.26 | rabby | hi |
| 13:55.28 | phatjoe | any ideas? |
| 13:55.46 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:56.20 | [TK]D-Fender | bminish: Show the CLi output including the error, and then do dumps of agents, SIP devices, etc. EVERYTHING to back it up |
| 13:56.24 | rabby | i am going to run my new freepbx in a flash with my isdn fritz card. but capiinfo tells me: capi not installed. and that's why i ask You what i have to install now... |
| 13:56.46 | [TK]D-Fender | phatjoe: First guess : DNS issues |
| 13:57.19 | phatjoe | i read that in a forum... i changed all peers to ip... issue persisted... :( |
| 13:57.34 | phatjoe | i will get some dumps |
| 13:57.45 | phatjoe | i am using realtime |
| 13:57.53 | phatjoe | incase that may be playing a part |
| 13:57.54 | [TK]D-Fender | rabby: http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+with+CAPI |
| 13:58.12 | SuPrSluG | any reason a pri channel might occasionally not go back on hook? |
| 13:58.26 | [TK]D-Fender | phatjoe: You essentially have no deatils for us. Not much we can say because of that. I would however upgrade at the very least |
| 13:59.31 | rabby | [TK]D-Fender: i will try that. thanks so far |
| 14:00.00 | phatjoe | <PROTECTED> |
| 14:00.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:00.09 | [TK]D-Fender | rabby: There is a doc on how to do this for trixbox which is built nearly the same : http://www.ivanoiu.com/avm-isdn-fritz-card-pci-on-trixbox-asterisk/ |
| 14:00.30 | [TK]D-Fender | phatjoe: what are you using for timing? |
| 14:01.14 | phatjoe | i have heard that it is good practice... even when just using SIP... |
| 14:03.40 | phatjoe | WARNING[4573]: chan_sip.c:12565 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 14:04.50 | bminish | This call failed http://pastebin.com/d2f9ce575 |
| 14:05.06 | bminish | This call worked http://pastebin.com/d29fe1779 |
| 14:05.16 | bminish | any ideas ? |
| 14:06.00 | Qwell | [TK]D-Fender: gasp |
| 14:06.37 | bminish | call in question is in both cases for NewellMaherConsultants0871267195 |
| 14:08.44 | bminish | Agents tell me that they hear a fraction of a second of audio on some of the lost calls |
| 14:15.13 | [TK]D-Fender | bminish: I don't see anything special in there. |
| 14:16.01 | bminish | nor do I :-( |
| 14:20.16 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 14:20.43 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 14:20.54 | Zeeek | morning world |
| 14:21.17 | *** join/#asterisk codefreeze (n=murf@216.166.159.235) |
| 14:21.18 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 14:23.10 | phatjoe | <PROTECTED> |
| 14:23.20 | phatjoe | <PROTECTED> |
| 14:24.09 | phatjoe | <PROTECTED> |
| 14:24.29 | phatjoe | chan_sip.c:15851 sip_poke_noanswer: Peer '*****' is now UNREACHABLE! Last qualify: 2 |
| 14:25.17 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:25.51 | *** join/#asterisk ice_croft (n=nolan__@81.26.135.117) |
| 14:26.01 | ice_croft | hi ) |
| 14:26.14 | ice_croft | do we have call-weaver channel here? ) |
| 14:26.46 | *** part/#asterisk ice_croft (n=nolan__@81.26.135.117) |
| 14:28.58 | EmleyMoor | !phones |
| 14:29.39 | keith4 | ~phones |
| 14:29.40 | jbot | hmm... phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 14:30.25 | Zeeek | I don't agree |
| 14:33.11 | [TK]D-Fender | Everyone is entitled to my opinion :) |
| 14:33.31 | EmleyMoor | Zeeek: OK - give your opinion if you wish |
| 14:33.45 | keith4 | what's with the "Ever. places like such as" at the end of that factoid? |
| 14:33.54 | Zeeek | Polycom #1. Maybe. But the rest? I don't agree. Because I've never tried most of them :) |
| 14:34.32 | Zeeek | My brother claims the Linksys sounds like crap. I've always thought it sounded decent. |
| 14:34.57 | keith4 | what's your brother's sample size? one? |
| 14:34.58 | Zeeek | I think my hearing is shot though after years in front of loud amps |
| 14:35.35 | Zeeek | EmleyMoor: my opinion is that drugs, sex and rock n roll can change your hearing as you approach old age |
| 14:36.00 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 14:36.02 | Zeeek | I too want the rest of "places like..." |
| 14:36.53 | [TK]D-Fender | Zeeek: So lets recap : You Disagree. You also have no experience with the others, and your hearing is shot. So your statement would be considered "qualified" would it? :p |
| 14:40.39 | Zeeek | Objection: badgering the witness |
| 14:41.42 | Zeeek | speaking of sample size, one. I used a TRS-80 cassette port to sample audio in 1979. That's true one bit resolution! It was to decode Morse code and RTTY |
| 14:45.36 | rwaite | i have an agi script i'd like to run after voicemailmain exits. if you exit it with #, it will run the script fine, but if you forefully end the call by hanging up, it does not |
| 14:45.52 | rwaite | is there any way to have the script run no matter how the call ends? |
| 14:46.18 | [TK]D-Fender | Objection : The defendant is slandering badgers and their noble character. |
| 14:46.20 | rwaite | i would try 'h' but that doesnt give me any way to determine what extension the call was on, which i'd need to direct the mbox's behavior |
| 14:46.22 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:46.27 | rwaite | s/mbox/script/ |
| 14:46.42 | rwaite | wow thats cool |
| 14:46.49 | [TK]D-Fender | rwaite: "h" <- read up on your "asterisk standard extensions" |
| 14:46.56 | rwaite | see above |
| 14:47.12 | [TK]D-Fender | rwaite: IIRC there is a var that hold it. |
| 14:47.19 | rwaite | i will look then |
| 14:47.20 | rwaite | thx |
| 14:47.24 | [TK]D-Fender | rwaite: and you could always try setting one before hitting VM |
| 14:47.40 | rwaite | that's true |
| 14:48.07 | rwaite | so test if a var is set, and if so, run the script |
| 14:48.19 | rwaite | otherwise exit. that would probably work |
| 14:48.31 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:48.32 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 14:48.37 | *** join/#asterisk spokra (n=spokra@host093-179-177.sea0.speakeasy.net) |
| 14:48.47 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 14:51.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:52.13 | rwaite | grr |
| 14:54.24 | Zeeek | [TK]D-Fender: why call them "extensions"? They're built right in to the dialplan? |
| 14:54.44 | Zeeek | ducks at the sound of the snare drum |
| 15:01.27 | EmleyMoor | Linksys SPA-942 - is the four-line upgrade charged for? |
| 15:02.19 | Zeeek | 941 |
| 15:02.36 | Zeeek | 942 has a hub (second etherenet socket) |
| 15:02.58 | Zeeek | But the four lines are now free |
| 15:03.55 | EmleyMoor | Ah, good |
| 15:03.58 | Zeeek | Optimum Voice won the PC Mag readers' choice for voip service. Never hoid of them |
| 15:05.26 | keith4 | as in "Optimum online"? |
| 15:05.50 | keith4 | it's the internet service side of Cablevision, in NY city / Long Island |
| 15:06.26 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 15:06.52 | EmleyMoor | The 942 is priced at less than three figures |
| 15:07.26 | EmleyMoor | (I am implementing PoE before getting any phones and getting only one phone first so that David can try it out) |
| 15:08.11 | [TK]D-Fender | EmleyMoor: SPA's are failry decent. Polycom is still a more solid choice IMO of course |
| 15:08.30 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 15:08.33 | EmleyMoor | [TK]D-Fender: Yes, but quite a heavy price! |
| 15:08.57 | [TK]D-Fender | EmleyMoor: In North America, not at all. |
| 15:09.13 | EmleyMoor | You have all the luck that side of the pond! |
| 15:09.24 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 15:09.26 | Zeeek | yeah Polycom is pretty reasonable these days with three line phones |
| 15:10.02 | Zeeek | free shipping at the moment, too. I'm actually considering buying of them new fangled 6xx jobs |
| 15:10.16 | Zeeek | with the HD Voice and toilet roll dispenser |
| 15:10.54 | keith4 | oooh, shiny |
| 15:11.15 | Zeeek | handy to clean the LCD, now with extra backlit goodness |
| 15:12.17 | Zeeek | ooVoo has rolled out a new version of their face-face (video) conferencing. (Windoze only) |
| 15:12.52 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:13.04 | Zeeek | by the way, we're going to try yet again to raise the VoIP Users Conference today in 40 minutes. http://bit.ly/voip for info |
| 15:13.54 | [TK]D-Fender | Zeeek: Colour backlit :) |
| 15:14.02 | Zeeek | color color color |
| 15:14.16 | Zeeek | I don need no stinkin' color |
| 15:14.44 | Zeeek | I wonder why it took like 6 years for them to see backlit was needed? |
| 15:15.23 | cy3o3 | do cheap single port internal fxs cards exist? |
| 15:15.53 | EmleyMoor | How cheap? |
| 15:16.10 | cy3o3 | I dunno, $20-40ish? |
| 15:16.22 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 15:16.42 | EmleyMoor | cy3o3: Not that I know |
| 15:16.49 | cy3o3 | :( Yeah |
| 15:17.01 | cy3o3 | I'm certainly not having any lucking finding anything like that |
| 15:21.57 | [TK]D-Fender | cyYes, and they suck |
| 15:22.43 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 15:28.20 | cy3o3 | bummah |
| 15:29.02 | Zeeek | I didn't know there were cheap FXS cards. FXO, yeah |
| 15:29.52 | Qwell | ugh |
| 15:29.59 | rabby | lspci does not tell me about my isdn / fritz card :-( |
| 15:30.00 | *** join/#asterisk Guest65085 (n=root@189.141.98.158) |
| 15:30.12 | Qwell | rabby: it should.. it may just be non-specific |
| 15:30.26 | [TK]D-Fender | OH, sorry, no cheap FXS cards |
| 15:30.28 | Qwell | rabby: if you'd like to pastebin the output, I'd be happy to look at the list, and try to find it |
| 15:30.57 | [TK]D-Fender | cy3o3: Go get an ATA instead |
| 15:30.58 | Zeeek | in fact I've never heard of a single channel FXS card |
| 15:31.19 | Qwell | Zeeek: well, a 4 port card with one module |
| 15:31.28 | Guest65085 | Hello guys, do you know if asterisk can send SIP INFO customized messages? |
| 15:31.59 | rabby | Qwell, sounds very nicely :-) let me restart and You get the no-paste |
| 15:33.05 | Zeeek | TDM300 with a single module, sure. That's what, $240? |
| 15:33.16 | Zeeek | oops 400 |
| 15:33.21 | Qwell | Zeeek: nowhere near |
| 15:33.22 | Zeeek | I have a couple |
| 15:33.29 | Qwell | it's maybe half that |
| 15:33.42 | Zeeek | really? When I bought mine the modules were nearly $100 |
| 15:34.04 | Zeeek | but that was in another century |
| 15:34.27 | rabby | Qwell: http://rafb.net/p/pIBiJc46.html |
| 15:34.30 | Zeeek | so long ago it was beofre FXO modules |
| 15:34.46 | malcolmd | TDM410B, the bundling of the TDM410 card and a single FXS module lists for $215.95. |
| 15:34.48 | Qwell | rabby: huh |
| 15:35.06 | Zeeek | 215 isn't exactly half 240 :) |
| 15:35.32 | Zeeek | but the quality is indisputable |
| 15:35.41 | Qwell | rabby: It's either that C-Media one (I doubt it), or...it's not there. odd |
| 15:36.13 | rabby | if i restart after removing the card, it should show me one line less, right? |
| 15:36.29 | Qwell | one would think. it looks like you're right, and it isn't there |
| 15:36.33 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-29-rrdg-esr-2.dynamic.isadsl.co.za) |
| 15:37.02 | hardhatpat | I am having trouble sending calls to one of my providers, i keep getting 'No Authority Found' |
| 15:37.17 | Zeeek | one last reminder that we'll be talking asterisk and VoIP today in 20 minutes or so: http://bit.ly/voip |
| 15:37.36 | Zeeek | see you on #voip-users-conference |
| 15:38.07 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 15:38.14 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
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| 15:44.11 | bijit | any manuals for setting tftp for astra phones? |
| 15:44.54 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
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| 15:46.58 | [TK]D-Fender | bijit: go look on the WIKI |
| 15:47.00 | [TK]D-Fender | ~wikis |
| 15:47.00 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
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| 15:49.46 | bijit | [TK]D-Fender: ty |
| 15:53.19 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 15:56.03 | nikolaev | A quick one: Is it possible to make the asterisk working with freeradius in accordance to do AAA but not just Accounting ? |
| 15:56.29 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
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| 15:59.26 | dominic1 | Can anybody tell me how I can set the faxheader in hylafax. I want if a user sends a email to fax, that the header points to his personal faxnumber. |
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| 16:00.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:00.45 | sorend | h/part |
| 16:00.49 | *** part/#asterisk sorend (i=sorend@sip-proxy.gratissip.dk) |
| 16:02.27 | zeeesh | can i configured 2 different call log statistics for the same agents.? like i have configured asterisk stats can i configured "call qast" or any other ? |
| 16:03.43 | CrashHD | hello everyone |
| 16:08.39 | *** join/#asterisk wacky_ (n=abourget@mtl.savoirfairelinux.net) |
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| 16:08.40 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:09.19 | wacky_ | In what proportions do you think the extensions conf/ael/ael2 are used ? |
| 16:09.21 | [TK]D-Fender | zeeesh: Go ask in a channel that supoprts that software |
| 16:09.45 | [TK]D-Fender | wacky_: 97,1,2 |
| 16:09.49 | wacky_ | Every system I've seen uses the standard extensions.conf exten => 123,2,Dial() methods, was AEL2 widely adopted ? |
| 16:09.59 | wacky_ | is there a reason for that ?! :P |
| 16:10.31 | [TK]D-Fender | wacky_: Sparse documentation, difficut to debug and few people to help |
| 16:10.58 | wacky_ | oh it is difficult to debug ? |
| 16:11.01 | [TK]D-Fender | wacky_: a "nice idea"..... when everything works that is |
| 16:11.30 | [TK]D-Fender | wacky_: Yes, it is. It also offer no more than is possible in extensions.conf directly, and indeed LESS |
| 16:11.52 | wacky_ | oh well :P |
| 16:13.01 | nikolaev | [TK]D-Fender any ideas about radius and AAA requests ? |
| 16:13.19 | [TK]D-Fender | nikolaev: No, and please stop targeting people with questions like that. |
| 16:13.26 | [TK]D-Fender | nikolaev: Its quite rude |
| 16:14.18 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:14.26 | wacky_ | ok thank you very much |
| 16:14.46 | nikolaev | [TK]D-Fender ok, sorry |
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| 16:22.18 | keith4 | is there a collection of free MOH files somewhere? |
| 16:25.19 | keith4 | (preferably *not* MP3s) |
| 16:26.21 | *** join/#asterisk oilinki (n=oil@ppp-124-120-19-208.revip2.asianet.co.th) |
| 16:27.50 | [TK]D-Fender | keith4: Yes, * comes with a few |
| 16:27.59 | [TK]D-Fender | keith4: And plenty of free links on the WIKI |
| 16:28.02 | keith4 | thoughts on using a streaming radio station as MOH? |
| 16:28.14 | [TK]D-Fender | keith4: AND of course you're free to convert to whatever you want. |
| 16:28.35 | keith4 | ok, i'm off to the wiki |
| 16:28.45 | [TK]D-Fender | keith4: Thought : Why make my system dependent on some external resource? |
| 16:29.12 | keith4 | true. perhaps there is some way to fall back on local audio if the stream fails? |
| 16:29.47 | *** join/#asterisk fainsys (n=fainsys@c-76-17-119-68.hsd1.ga.comcast.net) |
| 16:29.58 | keith4 | although, i guess streaming audio is almost guaranteed to be mp3 |
| 16:30.39 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 16:30.42 | vAd0r | hey |
| 16:30.50 | [TK]D-Fender | keith4: No, there is no failover |
| 16:30.57 | vAd0r | can asterisk do hot do hotdesking? |
| 16:31.05 | keith4 | vAd0r: yes |
| 16:31.08 | vAd0r | where you can walk up to a phone log in and it switches |
| 16:31.14 | keith4 | [TK]D-Fender: ok. thanks |
| 16:31.22 | vAd0r | is there a link to set it up and see how it works |
| 16:31.53 | vAd0r | is it called hotdesking or is it another name keith4? |
| 16:31.54 | [TK]D-Fender | vAd0r: No. |
| 16:32.05 | keith4 | vAd0r: try google |
| 16:32.16 | [TK]D-Fender | vAd0r: This is up to YOU to invent in your dialplan. |
| 16:34.10 | *** join/#asterisk _khan (n=shariq@124.29.194.207) |
| 16:35.10 | _khan | i have 1 mb bandwidth, how many simultaneous calls can do easily with g729 codec.... |
| 16:37.52 | [TK]D-Fender | _khan: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 16:38.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:38.06 | codefreeze | wacky_: want another viewpoint on AEL2? |
| 16:38.22 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:38.30 | [TK]D-Fender | wacky_: (from one of its developers ;)) |
| 16:39.27 | wacky_ | codefreeze: yeah go ahead :P |
| 16:40.37 | wacky_ | tell me, what state is it in.. and why do you think it's being slowly adopted like that ? |
| 16:40.38 | *** part/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 16:41.43 | codefreeze | wacky_: It's been in production about 2 years now. Bugs, when filed, are handled as quickly as possible. Several enhancement requests have been added over those two years. |
| 16:44.13 | orionr | ~books |
| 16:44.18 | orionr | ~book |
| 16:44.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:44.24 | [TK]D-Fender | wacky_: I'll clarify my use of "works" as being "when your code in AEL2 is right and you don't need to debug from CLI, the parser generally does its job". AEL2 can cover the most common things you might implement in the dialplan, but naturally some may be too "out of the box" for it to do for you. |
| 16:44.55 | *** join/#asterisk Shotygun (n=thorn@82.166.209.89) |
| 16:45.11 | wacky_ | out of the box ? |
| 16:45.36 | codefreeze | wacky_: [TK]D-Fender points out that AEL adds nothing to extensions.conf, in that you can do everything you want in extensions.conf; he keeps harping on this issue, but it's pointless. CPU assembler languages can be used to do any programming. Higher level languages cannot add anything, because they (like AEL) compile into assembler. So why use C, C++, Java, etc.? |
| 16:45.46 | *** join/#asterisk chandoo (n=chandra@ool-4353b4c7.dyn.optonline.net) |
| 16:45.56 | wacky_ | :P |
| 16:46.04 | chandoo | hi |
| 16:46.18 | wacky_ | well I've seen those for loops and stuff, which it pretty harsh in original .conf .. so I guess this is an improvement, isn't it ? |
| 16:46.20 | chandoo | what is the cheap/reliable voip provider |
| 16:46.31 | wacky_ | chandoo: where are you ? |
| 16:46.36 | chandoo | new jersey |
| 16:46.45 | chandoo | hackensack |
| 16:46.47 | wacky_ | I know of link2voip.com, les.net |
| 16:47.08 | wacky_ | flowroute.com also.. |
| 16:47.29 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 16:47.34 | codefreeze | wacky_: if all you do is simple dialplan stuff, extensions.conf is fine. But the moment you do loops, ifs, etc, you should be doing it in AEL. I'm biased, but I'm also a programmer with 30+ years experience, on projects with 500k lines and up. |
| 16:47.53 | wacky_ | chandoo: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+North+America |
| 16:48.09 | Shotygun | Doing complex conditions in extensions.conf is indeed no fun at all. |
| 16:48.39 | wacky_ | codefreeze: is it possible to create macros using AEL and use the original .conf for the rest ? |
| 16:48.50 | wacky_ | so to use both of them ? |
| 16:49.02 | chandoo | wacky_: i dont want to run voip server at home, i just need service |
| 16:49.03 | Shotygun | wacky_: I'm using them both today, only because of migration lazyass on my behalf. |
| 16:49.12 | chandoo | i can buy modem if possible |
| 16:49.20 | codefreeze | wacky_: sure! I've been working hard on the merge_and_delete stuff underneath to make that work well. |
| 16:49.33 | [TK]D-Fender | codefreeze: I'll validate that the user-end scale of difference between "assembler" and "C" is not comparable to "AEL2" vs "extensions.conf" |
| 16:49.33 | wacky_ | oh that's good.. |
| 16:49.51 | chandoo | right now i am having phonepower for $24/mo for unlimited calling |
| 16:49.52 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:50.02 | chandoo | but customer support sucks |
| 16:50.24 | [TK]D-Fender | codefreeze: Yes, it makes loops and certain branching nicer. that much however? only the jump point and "scope" is a little dirrerent |
| 16:50.28 | Guest65085 | do you know if asterisk can send SIP INFO customized messages? |
| 16:50.36 | wacky_ | chandoo: the site I gave you lists providers, like you're looking for |
| 16:51.55 | [TK]D-Fender | Guest65085 :You have the source code. Get to it. |
| 16:52.03 | chandoo | i think i am not understanding what they are selling |
| 16:52.31 | mandd | 305 => 1234,Sergey,email@domain.ca |
| 16:52.32 | [TK]D-Fender | chandoo: here : |
| 16:52.35 | [TK]D-Fender | ~itsplist-us |
| 16:52.35 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:52.36 | chandoo | all i need is make phone calls in usa unlimited, what service i have to buy |
| 16:52.37 | [TK]D-Fender | ^^^^^^^^^ |
| 16:52.45 | mandd | what else do I need to do, so that asterisk sends an email |
| 16:52.57 | mandd | once voice mail is received |
| 16:53.05 | [TK]D-Fender | mandd: have a proper sendmail compatible binary. |
| 16:53.18 | [TK]D-Fender | mandd: and "attach=yes" as a parameter <-- |
| 16:53.27 | mandd | ok, thanks! |
| 16:53.37 | [TK]D-Fender | mandd: read the SAMPLE config for the optional parameters beyond the 1st three |
| 16:53.50 | mandd | reading right now |
| 16:54.04 | codefreeze | [TK]D-Fender: we've been round and round on this. Yes, it's 'nicer' enough to justify using it. And the diff between the two, not that much to you, is enough, tho, to justify using it. If all you do is simple, stuff, then OK, leave AEL alone. But when it gets just a little complex, why shoot yourself? It's not that hard to learn! |
| 16:54.31 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:55.43 | [TK]D-Fender | codefreeze: You're right on this, and rephrased as such I agree with you :) (been there too!). My down sides are in negative situations and the fact that as you should learn extensions.conf anyways, adding a layer might not be so great for you over time. |
| 16:55.45 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 16:56.18 | *** join/#asterisk Nkoji (n=nicholas@168-215-215-195.static.twtelecom.net) |
| 16:56.28 | [TK]D-Fender | codefreeze: BIG conditional branches certainly would look a LOT nicer in AEL2. Loops are also nicer. |
| 16:57.00 | Nkoji | Where is the best site to get information on the data sent between rings when placing a call |
| 16:58.52 | codefreeze | As to docs, the wiki page on http://voip-info.org/wiki/view/Asterisk+AEL2 is what I've spent some time on, keeping it up to date as much as I can. If it's not enough, or missing something, let me know. |
| 16:59.53 | codefreeze | Someday, they'll work this stuff into Asterisk: the Future of Telephony. |
| 17:02.23 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 17:03.06 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:03.10 | codefreeze | Plus, you get all the benefits of a free-format language: easier to read, not the repetitive, tabular format of extensions.conf, using the control flow statements discourages spaghetti programming, therefore, easier/cheaper to mainain, etc. etc. |
| 17:03.25 | BCS-Satori | I am trying to register a Cisco 7921G via SCCP. Asterisk keeps reporting back "Skinny Client sent less data than expected. Expected 4 but got 0." as well as the phone stating "Registration Failed" I can't seem to find much on this error message, any ideas? |
| 17:04.13 | [TK]D-Fender | codefreeze: "includes" question. It is a section within your "context" section. And you have multiple "include" sections so you have INCLUDES > other code > MORE INCLUDES in order to enforce prioritization(sorting)? |
| 17:05.09 | [TK]D-Fender | codefreeze: I also agree on the de-pastafying benefits. |
| 17:05.20 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 17:05.31 | *** part/#asterisk roe_ (n=roe___@216-164-160-45.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 17:06.53 | [TK]D-Fender | Nkoji: Rings on what? |
| 17:07.29 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 17:07.37 | CrashHD | I'm dealing with a massive number of dropped calls |
| 17:07.49 | CrashHD | what should I be looking to trouble shoot? |
| 17:07.53 | CrashHD | connectivity is good |
| 17:08.02 | CrashHD | pings show no packet loss |
| 17:08.07 | CrashHD | servers are not loaded at all |
| 17:09.42 | wacky_ | codefreeze: see.. the only thing I can see why people don't adopt it.. is that those exten => s,1,Stuff() lines are like concrete |
| 17:10.36 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 17:10.48 | Nkoji | [TK]D-Fender: When you place a call information is passed back and forth like caller ID |
| 17:10.56 | C4away | anyone know if it is possible and/or simple to set up a BLF for a Queue on 1.4 ? |
| 17:11.00 | codefreeze | [TK]D-Fender: Usually, when I wrote the compiler, we were thinking of just one includes construct in a context. If multiple includes get mucked up, file a bug, and I'll make it work right. |
| 17:11.22 | Nkoji | i was wondering if there was a list of all the possible information sent way |
| 17:11.26 | C4away | well, I should say notification / hint ... the BLF would be determined by the phone |
| 17:11.27 | Nkoji | sent that way** |
| 17:11.42 | codefreeze | wacky_: yeah, ... my philosophy is, if it works, don't fix it. re-write when the need arises |
| 17:11.44 | [TK]D-Fender | codefreeze: Just saying that (yes, in freak) case may be you want a few includes, more direct extens, followed by more include. |
| 17:12.18 | [TK]D-Fender | codefreeze: this is a "suggestion", as is : http://pastebin.com/m34a3b650 <- is this already possible? If not its the sort of thing I have seen asked before |
| 17:12.37 | [TK]D-Fender | Nkoji: at this point I'm guessing you're referring to an ANALOG LINE? |
| 17:12.45 | Nkoji | Yes |
| 17:12.51 | codefreeze | [TK]D-Fender: I don't have anything against it, but underneath, the position of the includes in a context mean nothing. They are gathered into a single list down deep. Only the order matters. |
| 17:13.09 | [TK]D-Fender | Nkoji: there are plenty of telecom references for this. Give this a scan first, then get google-ing |
| 17:13.11 | [TK]D-Fender | ~101 |
| 17:13.12 | jbot | 101 is probably Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 17:13.27 | Nkoji | I just want to review the information passed by phones when the call is placed because i am currently waiting ona replacement card from digium |
| 17:13.40 | Nkoji | Thank you |
| 17:13.45 | [TK]D-Fender | codefreeze: So extensions.conf groups them as well? Indeed that would render my idea completely moot |
| 17:13.52 | BCS-Satori | I am trying to register a Cisco 7921G via Skinny. Asterisk keeps reporting back during the registration process "Skinny Client sent less data than expected. Expected 4 but got 0." followed by "Skinny Session returned: Success" and "Rejecting Device SEP001F6C7A0BBF: Device not found" |
| 17:14.31 | [TK]D-Fender | BCS-Satori: We heard you 10 minutes ago, and few people use skinny, and even fewer that phone model. |
| 17:15.19 | [TK]D-Fender | C4away: You want a HINT is a queue has a call in it or not? |
| 17:15.27 | madison | for a large scale deployment, would one consider asterisknow? |
| 17:15.49 | [TK]D-Fender | madison: Everything depends on complexity |
| 17:16.10 | BCS-Satori | [TK]D-Fender: thats fine, i just don't know what that error means, when i search it out, all i get back is resposnes from 2006 with an asterisk 1.4.0(beta) bug |
| 17:16.13 | [TK]D-Fender | madison: its a distro. It GUI's stuff up. If you can customize it enough to do what you want it'll work as well as anything else. |
| 17:17.50 | madison | [TK]D-Fender, i just do not want to re-invent the wheel or backtrack. we are using asterisk now on opensuse and i am not happy with it |
| 17:18.11 | [TK]D-Fender | madison: Then multiply that by the words "large scale". |
| 17:18.14 | madison | err |
| 17:18.26 | madison | we are current using asterisk i mean |
| 17:18.34 | [TK]D-Fender | madison: And "not happy with it" doesn't say much at all |
| 17:19.08 | madison | i do not know opensuse and i do not feel that it is suitable for a commercial product |
| 17:19.10 | codefreeze | [TK]D-Fender: yes, the includes, switches, and ignore patterns are gathered into ordered lists and attached to the context underneath. Where they were in the input dialplan is lost. When we generate AEL or extensions.conf, they are usually put in at the top. Try 'dialplan show' to see |
| 17:19.19 | madison | the astersik book suggests centos |
| 17:19.31 | [TK]D-Fender | madison: It'll work on any distro you can manage |
| 17:19.42 | [TK]D-Fender | madison: and you haven't clarified your problems. |
| 17:20.23 | madison | myproblems are that i am not able to manage opensuse and thus, things like festival, etc will not work right |
| 17:20.49 | [TK]D-Fender | codefreeze: Thanks for the quick breakdown there. What do you think on my "semi-global" context variable idea? Perhaps it could be treated as a "substitution" at parse-time instead of even being a "variable". |
| 17:21.27 | [TK]D-Fender | madison: Any reason we should blame your distro for this instead of your management of it? regardless, CentOS is a more popular base. |
| 17:22.42 | madison | let me ask this then |
| 17:22.55 | codefreeze | [TK]D-Fender: as to the variables, there are global vars, and channel vars. They are stored in linked lists. No per-context or per-extension vars. But that's an idea! You might run it by Corydon76-dig; It's not clear to me at the moment, tho, how we might search easily for them. Maybe make them globals, and underneath, prepend the context/exten names they are associated with. AEL would have to keep track of them and prepend the names for you underneath |
| 17:23.20 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 17:23.46 | madison | what distro would you use if you going to deploy say 1000 residential voip accounts |
| 17:24.06 | mvanbaak | madison: whatever you know best |
| 17:24.15 | madison | i know gentoo |
| 17:24.22 | mvanbaak | so use gentoo |
| 17:24.30 | madison | but it is not carrier grade |
| 17:24.38 | mvanbaak | no distro is |
| 17:25.02 | codefreeze | doubts even carrier-grade s/w is carrier-grade |
| 17:25.50 | mvanbaak | dont think so |
| 17:31.31 | *** join/#asterisk s0lid (n=s0lid@122.53.102.162) |
| 17:32.29 | shido6 | heh |
| 17:33.14 | keith4 | which is more likely to have problems for you... a "carrier-grade" OS that you don't know how to use, or a "non carrier-grade" distro that you're familiar with? |
| 17:34.07 | [TK]D-Fender | codefreeze: as I wrote it this could be a pure text substitution at the parser level. |
| 17:35.08 | [TK]D-Fender | madison: You are missing the point. Distro doesn't matter. Your setup either works, or it doesn't. 5 user or 500 on the same setup shouldn't make a difference in comparing deployments on 2 different distros |
| 17:35.36 | madison | when you have 1000 customers, you cannot have an unstable distro |
| 17:36.57 | keith4 | use debian then, it's stable |
| 17:37.41 | [TK]D-Fender | madison: You don't seem to have much of a grasp of Linux at this point. Its not the distro thats unstable, its the combination of the pieces you choose. |
| 17:38.59 | [TK]D-Fender | madison: So if someone says "I'm running XYZ on my distro jsut fine", odds are you can too |
| 17:39.23 | [TK]D-Fender | madison: The only slight against SUSE I've heard is the difficulty of compiling kernel modules. |
| 17:40.10 | keith4 | alright, i'm clearly an idiot. where can I find the FreePlay music? |
| 17:40.28 | keith4 | what file extension should they have? |
| 17:41.09 | madison | what will scale better? centos or opensuse? |
| 17:41.44 | *** join/#asterisk Segnale007 (n=Segnale0@host188-251-dynamic.26-79-r.retail.telecomitalia.it) |
| 17:41.54 | madison | and what are the pros and cons of asterisk vs asterisk now |
| 17:42.12 | [TK]D-Fender | madison: Ok, you're just not getting. NO SUCH THING! |
| 17:42.22 | tzanger | madison: one is more impatient |
| 17:42.31 | [TK]D-Fender | madison: TAnd your last comparative quesion is like comparing plastic vs a CAR. |
| 17:42.44 | madison | sigh |
| 17:43.18 | [TK]D-Fender | madison: Asterisk is a piece of SOFTWARE. AsteriskNOW is a DISTRO that just happens to come bundled with Asterisk, and the GUI |
| 17:43.21 | keith4 | he's talking about scalability, and considering *now? |
| 17:43.37 | [TK]D-Fender | madison: Whats better, Apache, or MacOS? |
| 17:43.38 | madison | i am not a he thanks |
| 17:43.50 | keith4 | she's talking about scalability, and considering *now? |
| 17:43.59 | madison | i did not know it was a distro sorry |
| 17:46.18 | codefreeze | [TK]D-Fender: true; could be done fairly straightforwardly in either conf or AEL. |
| 17:46.27 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 17:51.55 | [TK]D-Fender | codefreeze: Yes, when you think of it, it could.... in that case it be better to do it in extensions.conf and have an equivalent syntax in AEL2 that will parse back to that standard |
| 17:53.20 | [TK]D-Fender | madison: Now for a final sense of things, drop the word "scale" right out the windo with regards to distro's & GUI's, or anything else. Scaling is not a software issue, its a HARDWARE one based on how many calls, what kind of load, etc |
| 17:54.06 | chandoo | how much you guyz pay for monthly usage |
| 17:55.26 | Nugget | monthly usage of what? |
| 17:56.28 | [TK]D-Fender | Nugget: Red-light district services? |
| 17:56.47 | madison | ok, i have decided to go with what i know best. FreeBSD and Asterisk |
| 17:57.18 | [TK]D-Fender | madison: Good, now maybe you can move forward. |
| 17:58.30 | *** join/#asterisk kfife (n=mIRC@home.chicagoventure.com) |
| 18:00.05 | chandoo | Nugget: phone service |
| 18:00.15 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:00.15 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:00.27 | *** part/#asterisk Guest65085 (n=root@189.141.98.158) |
| 18:00.37 | chandoo | right now i pay 24.99 for unlimited calls for my provider |
| 18:00.44 | chandoo | some thing like that |
| 18:01.26 | [TK]D-Fender | chandoo: All of us have different needs and the prices vary. That's like asking how much gas costs to a scooter owner vs a Hummer owner. |
| 18:01.37 | CrashHD | how can I make sure asterisk is using the ztdummy for timing? |
| 18:01.53 | [TK]D-Fender | chandoo: You've been given links to a variety of providers. Go shop for yourself and see whose price and service best match your needs |
| 18:02.18 | chandoo | [TK]D-Fender: i am looking at them , but not quiet following them |
| 18:02.20 | CrashHD | I hear a tire kicker! |
| 18:02.21 | [TK]D-Fender | CrashHD: USE IT. (MeetMe might be an idea) |
| 18:02.44 | chandoo | so far i have only used phone service from sun rocket and phone power |
| 18:02.49 | CrashHD | tk, any more definitive way? I've never used meetme, don't know what I would be looking for as a problem |
| 18:02.58 | [TK]D-Fender | chandoo: If you can't read, and can't call up a CSR to ask questions, then I don't think anyone here wan't to be a substitute brain for you. |
| 18:02.59 | chandoo | so i dont know much about the thing what i want from them |
| 18:03.03 | CrashHD | maybe a "show my damn timing source" command? |
| 18:03.57 | [TK]D-Fender | chandoo: If you don't know what you want NOBODY can help you. |
| 18:03.57 | [TK]D-Fender | CrashHD: "zap show status" |
| 18:03.57 | chandoo | i am looking for swtiching my phone service to some one with some cheap and reliable |
| 18:03.57 | [TK]D-Fender | CrashHD: Should show it being there. But "ready" is in the proof |
| 18:03.57 | CrashHD | perfect! |
| 18:04.05 | CrashHD | what does ready mean? |
| 18:04.20 | [TK]D-Fender | CrashHD: is the timing reliable, etc. go TEST it. |
| 18:04.42 | [TK]D-Fender | CrashHD: Seriously jsut go set up a MeetMe room. |
| 18:04.46 | CrashHD | ahh yes of course |
| 18:04.54 | CrashHD | definitely testing it |
| 18:05.04 | CrashHD | but its nice to not have to make assumptions |
| 18:05.08 | [TK]D-Fender | ~assume |
| 18:05.09 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 18:05.14 | CrashHD | and I'll setup a meet me room right now |
| 18:05.44 | chandoo | i know what i want(i want same like what i have unlimited calls every month), but dont know how to get them from the link i am browsing it |
| 18:05.52 | chandoo | i am purely residential user |
| 18:06.18 | chandoo | i make calls some days and i dont at all some days |
| 18:08.23 | *** join/#asterisk tobias (n=tobias@66.152.121.39) |
| 18:08.28 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 18:09.24 | kfife | Hi folks. Here's a stumper: I'm taking my sipphone/gizmo5 calls and conditionally forwarding them out to the PSTN & the TDMA cell network. I notice that the call 'volume'/amplitude is somewhat lower/attenuated. Essentially it's just subjectively quieter. How is this possible since Asterisk is just just forwardign the ulaw frames on to the next network?? It reminds me of the call degredation that occurred when I used to bridge alalog trunks wit |
| 18:10.07 | [TK]D-Fender | chandoo: Go get off your ass and READ and see who offers "unlimted" and keep in mind that "unlimited" might cost you MORE depending on your actual needs |
| 18:10.47 | [TK]D-Fender | kfife: If its all just codec data passing through, its the fault of your terminators |
| 18:11.38 | kfife | I see. So you're saying for example that the cell network may be incidentally doing some da/ad conversion? |
| 18:12.20 | [TK]D-Fender | kfife: They could be passing it through string and a pair of tin-cans for all you know |
| 18:12.32 | kfife | True. Actually that sounds like a perfect explanation since the cell network is clearly doing some transcoding, and I only notice it when forwarding to the cell network. |
| 18:12.40 | [TK]D-Fender | kfife: And ocnsidering cell antenna varience etc, there is nothing to consider "normalized" |
| 18:12.40 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.144) |
| 18:12.42 | CrashHD | tk would there be any timing sources other than the ones that show up in zap show status? or does asterisk solely rely on zaptel for timing? |
| 18:12.50 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:12.56 | [TK]D-Fender | CrashHD: timing for WHAT? |
| 18:13.04 | CrashHD | timing in general |
| 18:13.09 | CrashHD | for rtp |
| 18:13.14 | CrashHD | sip calls |
| 18:13.31 | [TK]D-Fender | CrashHD: You are talking about specific things like ZTDUMMY with no context as to what you expect it to DO for you exactly |
| 18:13.45 | CrashHD | my apologies |
| 18:13.47 | [TK]D-Fender | CrashHD: You don't generally need Zaptel at all just for SIP |
| 18:13.52 | kfife | Fender: Is there a way to open these packets and rewrite them 'ein bissien' amplified? |
| 18:14.13 | CrashHD | I'm troubleshooting a massive dropped call issue |
| 18:14.22 | CrashHD | and one way audio half way in between calls |
| 18:14.37 | CrashHD | timing is what I'm assuming is the problem |
| 18:14.39 | CrashHD | at this point |
| 18:14.49 | CrashHD | and trying to rule that out so I can look at other items |
| 18:15.49 | CrashHD | I was thinking maybe the rtp stream from XO my sip provider was shit |
| 18:15.56 | CrashHD | and causing asterisk to pickup bad timing from the stream |
| 18:16.08 | CrashHD | from what I understand it's reliant on rtp source for timing |
| 18:16.19 | CrashHD | so I thought I would install ztdummy |
| 18:16.34 | CrashHD | and use internal timing = yes |
| 18:17.12 | CrashHD | but upon doing so (using RTC with a modified clock, 2048) it made the problem worse and additional complaints about robotic sounding voices started happening |
| 18:17.23 | CrashHD | this is with centos 5.2 and 2.6.18.XXXXX kernel |
| 18:17.39 | CrashHD | so I compiled a 2.6.26 kernel |
| 18:17.44 | CrashHD | with high resolution timing |
| 18:17.53 | CrashHD | and some preemption settings |
| 18:18.04 | CrashHD | and recompiled ztdummy to use the HRT |
| 18:18.04 | [TK]D-Fender | CrashHD: I recommend "stock" <- it wors |
| 18:18.09 | CrashHD | "stock"? |
| 18:18.21 | [TK]D-Fender | CrashHD: Stock kernal no funny business |
| 18:18.24 | coppice | chicken stock |
| 18:18.33 | CrashHD | 2.6.18 doesn't have HRT support |
| 18:18.35 | CrashHD | only RTC |
| 18:18.54 | [TK]D-Fender | kfife: I think 1.6 opens up possibilities, but gain mods aren't really viable in 1.4 and below |
| 18:19.05 | CrashHD | centos doesn't have a "stock" kernel past 2.6.18 |
| 18:19.08 | CrashHD | gain mods? |
| 18:19.14 | [TK]D-Fender | CrashHD: I have penty of people working fine off ZTDUMMY and CentOS stock installs |
| 18:19.21 | kfife | Fender: Thanks! |
| 18:19.27 | CrashHD | understandable |
| 18:19.35 | CrashHD | I'm grasping at straws |
| 18:19.53 | CrashHD | I was getting 99.97's and 99.96's in zttest's |
| 18:20.00 | CrashHD | so I figured I'd go for the HRT support |
| 18:20.03 | CrashHD | see if that helped |
| 18:20.14 | CrashHD | I'm getting 99.99's now |
| 18:20.19 | CrashHD | *shrugs* |
| 18:20.27 | CrashHD | not that any of that should matter |
| 18:20.36 | CrashHD | like I said, totally grasping at straws at this point |
| 18:22.25 | C4away | is there a module that would monitor a relay contact? maybe from the serial port? |
| 18:22.53 | *** join/#asterisk sucituanbo (n=blah@c-24-21-121-148.hsd1.wa.comcast.net) |
| 18:23.02 | C4away | the alarm receiver would do it, but only if the relay contact was connected to an alarm panel |
| 18:23.49 | C4away | app_rpt might has the ability to close a relay contact for PTT functionality on a radio repeater |
| 18:24.01 | C4away | -might |
| 18:24.24 | C4away | I don't know if there is something that would monitor two pins for closure / opening |
| 18:24.46 | CrashHD | any thoughts on the dropped calls and the one way audio half way through a call tk? |
| 18:25.01 | CrashHD | swear this issue is gonna drive me in to a mad house |
| 18:25.03 | CrashHD | lol |
| 18:25.13 | C4away | do you have canreinvite=no ? |
| 18:25.20 | CrashHD | yes |
| 18:25.21 | n3hxs | I don't know if this would help but: http://www.smarthomeusa.com/ShopByManufacturer/ACNC/Item/DP-28C/ |
| 18:25.31 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:25.44 | C4away | I was looking at power failure dialers |
| 18:26.02 | C4away | hmm |
| 18:26.23 | C4away | most power failure dialers are about $300, that's only 100 |
| 18:26.39 | [TK]D-Fender | CrashHD: Half way? no idea beyond maybe bandwidth. |
| 18:27.26 | C4away | exceeding sip timers maybe? |
| 18:27.31 | CrashHD | sip timers? |
| 18:28.02 | C4away | t1 t2 etc timers in sip.conf |
| 18:28.11 | C4away | I don't know what they do ... but I know when they are exceeded things go funny |
| 18:28.21 | CrashHD | bandwidth is 100mbit link, pushing 512k ping packets to carrier source no problem and phone end point is over a t1 with 10% utilization |
| 18:33.03 | jameswf-home | still hates php arrays |
| 18:36.45 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:44.30 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 18:46.12 | neurosys | :) |
| 18:46.31 | *** join/#asterisk BZBW (n=wlwzhang@67.110.250.132.ptr.us.xo.net) |
| 18:48.30 | *** join/#asterisk arooni (n=arooni__@c-67-168-26-198.hsd1.wa.comcast.net) |
| 18:48.59 | *** join/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 18:49.06 | nny_1 | any vitelity users here? |
| 18:49.12 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:50.55 | fogo | nny_1: I have them as an all else fails link; they're alright |
| 18:52.52 | nny_1 | fogo: yeah having dtmf issues righ tnow |
| 18:52.54 | *** part/#asterisk wacky_ (n=abourget@mtl.savoirfairelinux.net) |
| 18:52.56 | nny_1 | right* |
| 18:53.10 | nny_1 | have dtmf=inband but the last two test calls, no dtmf got passed |
| 18:53.51 | nny_1 | have a couple of clients getting ready to use an ITSP for long distance, trouble shooting DTMF problem is not something I want to do |
| 18:55.15 | fogo | nny_1: never used dtmf on them, so I have no idea |
| 18:55.46 | nny_1 | may be time to try another itsp :\ |
| 18:56.14 | fogo | nny_1: have you tried calling/emailing them for assistance? |
| 18:57.05 | nny_1 | been on hold for 10 minutes so far |
| 18:57.24 | nny_1 | not seeing a lot of positive responses with a quick google search for vitelity and dtmf :\ |
| 19:00.29 | CrashHD | C4away: I've never heard of the t1 t2 stuff |
| 19:00.35 | CrashHD | where can I find the docs on it? |
| 19:01.46 | C4away | hmm |
| 19:02.32 | *** join/#asterisk n9urk (n=IceChat7@rrcs-70-62-74-122.midsouth.biz.rr.com) |
| 19:02.52 | kfife | <PROTECTED> |
| 19:02.57 | C4away | check the rtptimeout value in sip.conf for starters |
| 19:03.05 | C4away | CrashHD ^^ |
| 19:03.11 | fogo | winks |
| 19:03.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.24 | C4away | the sip timer values may be compiled into chan_sip.so on asterisk |
| 19:03.38 | kfife | I understand that with Analog DID trunks, the CPE provides the battery voltage to the CO |
| 19:03.39 | C4away | on many devices, such as SIP desk phones, the timer values can be set |
| 19:04.13 | kfife | so who winks at whom? |
| 19:04.33 | kfife | Wouldn't you have to be the one providing the battery voltage to wink? |
| 19:04.46 | kfife | yet I keep hearing about how the CO winks at the CPE |
| 19:05.17 | kfife | It would make more sense that the CPE detects 'off hook' and then sends a wink back to the CO to say 'ready for DTMF' |
| 19:05.32 | kfife | Can anybody here confirm my alternate scenario? |
| 19:06.47 | kfife | I think analog DID trunks are pretty obscure in the softswitch world, but Rhino FXS cards are supposed to be able to act as analog DID trunks |
| 19:07.32 | [TK]D-Fender | nny_1: http://archives.free.net.ph/message/20080722.194402.bdd3de1e.fi.html |
| 19:07.51 | [TK]D-Fender | nny_1: Claims you should be using rfc2833 as of a few days ago |
| 19:08.31 | madison | [TK]D-Fender thank you for your feedback and thoughts |
| 19:08.38 | [TK]D-Fender | kfife: Either side could potentially "wink" |
| 19:09.11 | [TK]D-Fender | madison: You're welcome, and I hope you find yourself a manageable solution |
| 19:09.32 | madison | as do i |
| 19:09.57 | kfife | Fender: my understanding is that a 'wink' is a battery reversal. momentary (ca 200ms )T-R and R-T. If the CPE is providing battery, wouldn't the CPE be the only one in a position to wink? |
| 19:10.41 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 19:10.42 | _MrSeb_ | Hi to all |
| 19:11.10 | [TK]D-Fender | kfife: Could be taken as a circuit cut. |
| 19:11.41 | [TK]D-Fender | kfife: and CPE = phone, != telco therefor doesn't provide battery |
| 19:12.23 | CrashHD | ok |
| 19:12.24 | CrashHD | thanks |
| 19:12.40 | coppice | kfife: I've never seen an analogue DID line use wink |
| 19:12.41 | nny_1 | [TK]D-Fender: ooh |
| 19:12.47 | n9urk | CrashHD: what kind of connection do you have to the CO? |
| 19:13.06 | nny_1 | [TK]D-Fender: wow good eye, I'll try that, altough i dunno why they wouldn't notify me of the change, as I have 4 or 5 accounts with them :\ |
| 19:13.14 | [TK]D-Fender | coppice: I've heard of it once in here myself |
| 19:13.29 | n9urk | CrashHD Sorry disreguard |
| 19:13.41 | [TK]D-Fender | nny_1: Customer (Dis)Service.... what won't we do for you today?! |
| 19:13.46 | n9urk | kfife: what kind of connection do you have to CO? |
| 19:13.50 | *** join/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 19:14.37 | nny_1 | [TK]D-Fender: yup |
| 19:14.46 | *** join/#asterisk moy (n=moy@nat/ibm/x-91a2b605a430dfcd) |
| 19:15.02 | nny_1 | [TK]D-Fender: i have 2 large clients wanting to use an ITSP as the main long distance provider, this doesn't sit well with me.. |
| 19:15.35 | kfife | n9urk: they are analog DID trunks. If you disconnect the CPE (a Legacy Nortel switch), there is no battery on the terminals. |
| 19:15.38 | coppice | think of any weird funky combination of signalling, and someone somwhere will be using it :-) |
| 19:16.09 | n9urk | coppice :) I like how you put that. You are so right :) |
| 19:16.16 | kfife | n9urk: that makes perfect sense since the CPE is providing the battery. |
| 19:17.11 | n9urk | kfife: If you unhook the Nortel switch and plug a POTS phone into the analog line, will the phone work? |
| 19:17.28 | *** join/#asterisk sorend (i=sorend@sip-proxy.gratissip.dk) |
| 19:17.31 | nny_1 | [TK]D-Fender: yup that did it |
| 19:17.34 | nny_1 | :\ |
| 19:17.42 | kfife | The CO recieves an incoming call for one of our DID's, selects the first available analog DID trunk, and after some exchange, presumably involving a wink, it sends teh last 4 digits (DNIS) of the DID to the CPE, which routes it to the appropriate DN on the nortel |
| 19:18.00 | kfife | n9urk: No. |
| 19:18.23 | kfife | The DID trunk offered by the CO is the REVERSE of a POTS line |
| 19:18.46 | sorend | hi, i'm trying to use asterisk in a linux setup with ip aliases. my asterisk has to run sip on one of the aliases, using bindaddr in sip.conf it binds correctly, but when it sends out messages, they go over the "main" ip, not the aliased one. |
| 19:18.54 | kfife | The FXO is at the CO, and the FXS-like port is in the CPE |
| 19:18.59 | n9urk | kfife: Sorry, i am off my rocker |
| 19:19.02 | sorend | does anyone have experience with a similiar setup with ip alises ? |
| 19:19.09 | coppice | kfife: that was the normal DID line until the 80s. they didn't usually use wink, though |
| 19:19.27 | n9urk | kfife: I was thinking in the inverse when I was asking the last question. |
| 19:19.35 | kfife | to simplify: the CO gets a 'dial tone' provided by the CPE and 'calls' the DID number that belongs to that subscriber |
| 19:20.30 | kfife | coppice: so after the 80's what did it change to? |
| 19:20.43 | kfife | coppice: we got these DID trunks in the mid 90's |
| 19:20.52 | coppice | T1s, except for very small setups |
| 19:21.14 | n9urk | kfife: so you have to wink back to the CO to signal on-hook? |
| 19:22.41 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 19:23.45 | kfife | I think the scenario is: call comes in to the CO, CO switch uses a 'finder' algorithm to pick the first available trunk, goes off hook. CPE detects off-hook, and does a 200ms battery reversal to the CO (a wink). CO senses the wink and sends DTMF tones carrying the DNIS of the DID. CPE reads DNIS digits, looks them up on the routing table (that I wrote) and sends the call to the appropriate extension (DN). |
| 19:23.48 | smth | hi, question about srvlookup? is there any way to have asterisk without being block when wan interface missing/down or dns server down? |
| 19:24.09 | coppice | kfife: are you sure they wink? I've seen them require reversal to start dialing, and then a further reversal indicates answer |
| 19:25.31 | kfife | coppice: not sure. What I am sure of is that we're using the most 'normal', 'basic', 'traditional' form of analog DID trunking that there is. |
| 19:26.21 | kfife | coppice: when you say 'wink' to start dialing, do you mean the CO dialing the DNIS digits, or do you mean the CPE dialing out to the PSTN? |
| 19:26.41 | coppice | the most normal and traditional is a stroger emulation, that doesn't wink at all |
| 19:27.13 | coppice | if its a DID line , the CPE doesn't make outgoing calls on it |
| 19:27.36 | kfife | coppice: Strowger ala 'step by step' pulse dialing switch circa 1930's?? |
| 19:28.14 | coppice | wel, there are some people still making strowger kit :-) |
| 19:28.35 | kfife | coppice: Exactly. I was asking the latter to confirm that we were on the same page. Many people don't understand ADID :-) |
| 19:29.46 | kfife | coppice: Bingo. In fact I read a thread recently about strowger switch collectors using IAX and asterisk to emulate tie-lines to make a global network of legacy strowger step-by-step switches that have a coordinated pulse-dialing namespace |
| 19:30.42 | kfife | That's got to be the coolest, geekyest, most impressive, most anachronistic waste of time in the world. |
| 19:31.47 | n9urk | kfife: do you have a link to that? |
| 19:32.02 | kfife | n9urk: Let me try to find it. |
| 19:32.10 | n9urk | kfife: thanks :) |
| 19:32.27 | coppice | I've seen strowger modules from ITI (Indian Telephone Industries) that were date stamped in the late 90s. |
| 19:32.59 | n9urk | coppice: India is still running steam trains as well, right? |
| 19:33.28 | kfife | coppice: Can you tell me what you think the exchange between CO and CPE looks like on an analog trunk? |
| 19:34.36 | coppice | the CPE provides battery. the CO applies loop and dials with DTMF after a short pause, the CPE reverses to indicate answer |
| 19:35.21 | n9urk | kfife: was this what you read? http://mysite.verizon.net/dalderdi/phones/sxs.htm |
| 19:36.12 | kfife | n9urk: Different but similar. |
| 19:36.58 | coppice | I think someone should make a SIP phone with a dial |
| 19:37.00 | n9urk | That is awesome. thanks and sorry I couldn't have been of better help earlier. I had an idea in mind, but then realized I was thinking of something entirely different |
| 19:37.02 | kfife | If anyone here knows of someone selling an old step-by-step switch, I would be interested in buying it. |
| 19:37.41 | kfife | I use a 1940's western electric 302 to make sip calls all the time. |
| 19:38.35 | kfife | There's a device that you can install 'passively' in the phone that translates rotary pulses into DTMF tones. |
| 19:38.39 | coppice | ah, but that's not integrated. a dial on the front and an RJ45 on the back is the right level of silliness |
| 19:38.55 | n9urk | kfife have you thought about setting up ebay alerts for one? I had heard there was one on there a couple years ago |
| 19:39.40 | kfife | doesn't modify the phone at all so as not to desecrate a piece of history. Uses a innovative idea of holding the numeral at the fingersto for an extra moment to do fancy stuff like #, * and speed dials. Very cool |
| 19:40.20 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:40.21 | coppice | real men tap out their phone numbers on the hook switch |
| 19:40.41 | kfife | Coppice: LOL |
| 19:40.53 | kfife | I used to do that on a phone that had a mechanical rotary lock on it. |
| 19:41.16 | coppice | many people did. much easier than going to get the key |
| 19:41.21 | kfife | i became quite proficient at it. I could dial 7 digit and 10 digit numbers usually witout error. |
| 19:41.31 | kfife | :-) |
| 19:41.52 | coppice | its harder these days. modern hook switches are less responsive |
| 19:42.01 | kfife | By the way that device I mentioned is called a rotatone. |
| 19:42.09 | kfife | Here's a link to the owner's manual http://www.wedidit.ca/Product%20Images/Oldphones/rotatone%20operation%20manual.pdf |
| 19:42.30 | kfife | Cost is about $100 |
| 19:42.32 | coppice | most ports are still happy to accept pulse dialing |
| 19:42.55 | kfife | indeed you are right. Unfortunately too many IVR's are not :-) |
| 19:43.19 | [hC] | what protocol do avaya phones speak, for VOIP? |
| 19:44.54 | anonymouz666 | tzafrir_laptop? |
| 19:44.58 | kfife | coppice: from earlier: your description of the ADID makes sense. So you're saying the CPE winks to indicate ANSWER and not to indicate READY FOR DNIS |
| 19:45.08 | kfife | I think I understand now |
| 19:45.34 | tzafrir_laptop | anonymouz666, here |
| 19:45.53 | kfife | so in fact it's the CPE that winks, and not the CO. That's what I suspected, although thanks to you I now understand where it is in the exchange |
| 19:46.28 | kfife | Coppice: Have you ever heard of the CO sending CLID information in addition to the DNIS digits? |
| 19:46.47 | anonymouz666 | tzafrir_laptop: If I have a TDM800P with QUAD FXO let's say from port 5 to 8 and my zaptel.conf is configured to 1-4, running ztcfg -vvv shouldn't alert anything? |
| 19:47.08 | coppice | yes. in a similar way to a T1, as *<ani>*<dnis>* |
| 19:47.27 | coppice | or are ani and dnis the other way around? :-\ |
| 19:47.29 | kfife | Coppice: It seems there'd be no technical reason why it couldn't do it, although naturally it would have to use FSK to send CNAM |
| 19:48.11 | coppice | i very much doubt anyone would use the FSK method, except in japan |
| 19:48.30 | kfife | coppice: Interesting. |
| 19:48.43 | tzafrir_laptop | anonymouz666, of course it should give an error |
| 19:49.19 | kfife | coppice: so what does taht exchange look like? First DNIS/ANI second CLID? |
| 19:50.02 | kfife | is sending CLID over ADID the exception or the rule? |
| 19:50.05 | keith4 | is the asterisk cvs repository web-viewable anywhere? i see references to cvsweb.digium.com in google results, but it seems to not exist anymore |
| 19:50.42 | tzafrir_laptop | keith4, try svn.digium.com |
| 19:50.58 | anonymouz666 | tzafrir_laptop: it does not with zaptel 1.4.10. the quad module was installed from port 5 to 8 in the TDM800 card and guy configured the zaptel from 1 to 4. so... of course nothing was expected to work no matter where him plugged the lines since the zaptel was configured in the wrong ports... |
| 19:51.03 | keith4 | thanks tzafrir_laptop |
| 19:51.50 | [TK]D-Fender | anonymouz666: SMRT |
| 19:51.58 | kfife | Has anyone used a Rhino FXS card to do ADID? I know they say they're the only ones who support it? |
| 19:52.07 | anonymouz666 | [TK]D-Fender: ? |
| 19:52.52 | coppice | kfife: any FXS port provides the hardware features to do DID. you just need to avoid applying ring volts |
| 19:52.56 | tzafrir_laptop | anyway, good night |
| 19:53.59 | anonymouz666 | [TK]D-Fender: what do you mean? |
| 19:54.27 | keith4 | i am so smart... S-M-R-T |
| 19:54.38 | kfife | coppice: I see. So far only RHINO says that they can do it, even if it's only a matter of providing the correct driver capability. |
| 19:54.45 | anonymouz666 | heh. |
| 19:55.09 | anonymouz666 | indeed. I was requested to help but I think ztcfg should produce an error on this... |
| 19:55.21 | [TK]D-Fender | kfife: the card being capable is one thing... is ZAPTEL capable of being configured to use this however <- |
| 19:55.29 | kfife | coppice: are you saying that even the Digiuym fxs cards currently will work properly on a DID trunk? |
| 19:55.46 | *** join/#asterisk dwelsh (n=dave@ottawa-hs-69-20-226-218.s-ip.magma.ca) |
| 19:56.01 | kfife | fender: Your point was my suspicion. That's why I was looking at Rhino. They say they can do it. |
| 19:56.26 | kfife | but I don't want to be beta testing. |
| 19:56.27 | dwelsh | Hi everyone. I just installed AsteriskNow. Does the CDR data in the CSV file also get stored in a database? |
| 19:56.46 | keith4 | ~asterisknow |
| 19:56.46 | jbot | from memory, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 19:56.57 | [TK]D-Fender | dwelsh: Go see if a databse is running |
| 19:57.10 | kfife | The ADID question may become irrelavent if we can get the customer to buy a T1 PRI. |
| 19:57.54 | dwelsh | ps -Af | grep sql returns nothing :( |
| 19:58.21 | [TK]D-Fender | dwelsh: good odds on "NO" then. |
| 19:58.36 | [TK]D-Fender | dwelsh: How to use a Db for CRD is well documented in THE BOOK |
| 19:58.38 | [TK]D-Fender | ~book |
| 19:58.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:58.42 | dwelsh | Oh ok. #asterisknow seems dead. |
| 19:58.53 | keith4 | that should tell you something |
| 19:59.02 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 19:59.23 | kfife | kfife: LOL |
| 19:59.52 | nny_1 | keith4: heh |
| 20:00.05 | kfife | That book is worth its weight in gold to anyone starting out. |
| 20:00.27 | kfife | and a great command reference to anyone. |
| 20:00.34 | dwelsh | Thanks. I'll check that out |
| 20:01.23 | kfife | dwelsh: Good luck! Another good book is: |
| 20:01.52 | *** join/#asterisk TheCompWiz (n=tmealey@wsip-68-109-200-102.mc.at.cox.net) |
| 20:02.04 | kfife | switching to VOIP |
| 20:02.27 | TheCompWiz | can someone help me understand the effects of using user/peer vs friend? |
| 20:02.31 | kfife | 978-0596008680 |
| 20:02.35 | kfife | ISBN: 978-0596008680 |
| 20:02.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:02.59 | kfife | Switchin to VOIP is less asterisk-centric |
| 20:03.16 | kfife | but easily worht the $26 bucks |
| 20:03.33 | TheCompWiz | anyone? |
| 20:03.57 | [TK]D-Fender | TheCompWiz: In most cases the only relevent type is "peer" since 1.4 |
| 20:04.33 | TheCompWiz | [TK]D-Fender: that I do understand... but I'm still trying to grasp the difference between user & peer... because... wouldn't every *user* need to be a peer? |
| 20:04.57 | [TK]D-Fender | TheCompWiz: You are thinking of "user" as a PERSON. |
| 20:05.24 | TheCompWiz | true... and by the way you said that... my thinking is wrong. so... what is a "user"? |
| 20:05.38 | [TK]D-Fender | TheCompWiz: In the old way : user = INCOMING account. peer = used to PLACE calls. friend = both |
| 20:06.30 | [TK]D-Fender | TheCompWiz: But 1.4 did away with that methodology for the most aprt and peer is more like "friend" at this point. |
| 20:06.59 | TheCompWiz | I'm still not following what you mean by "incoming account" |
| 20:07.04 | [TK]D-Fender | TheCompWiz: "user" still gets prioritized for authing incoming calls in cases where the auth for in/vs out is different even from the same place |
| 20:07.48 | [TK]D-Fender | TheCompWiz: pre 1.4. a PEER could ounly be used to PLACE a call. It would not be used to match an INCOMING call. the revers for USER |
| 20:07.52 | [TK]D-Fender | revers |
| 20:07.55 | [TK]D-Fender | reverse |
| 20:08.50 | TheCompWiz | incomming... meaning... from the sip-context used as the last-resort if unmatched? |
| 20:10.15 | [TK]D-Fender | TheCompWiz: these are ALL sections of sip.conf |
| 20:10.45 | kfife | Here's a DISA() question: I find that the DISA dialing timeout is too short. If a user pauses while dialing in the DISA() application will give you a 'reorder' tone. |
| 20:10.54 | TheCompWiz | yeah... but if that's correct... you can have multiple peers that answer to 1 "user" for an incomming call... correct? |
| 20:10.58 | kfife | It makes it so too many users have to retry to dial a simple ten digit number. |
| 20:11.27 | [TK]D-Fender | TheCompWiz: You are mixing terminology in a really bad way there.. |
| 20:11.57 | kfife | core show application DISA, does not show any parameters for setting the timeout. Is this value configured elsewhere? |
| 20:12.02 | TheCompWiz | probably... which is why I am having troubles grasping the concept. |
| 20:12.14 | [TK]D-Fender | kfife: Don't use DISA, it is not necessary. Fake it out with an IVR |
| 20:12.28 | TheCompWiz | or rather... what the difference is actually... or what situations you would use one or the other. |
| 20:12.34 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:12.49 | [TK]D-Fender | TheCompWiz: your ppeers don't answer to a USER. |
| 20:12.59 | kfife | I see! Write my own disa context, capture the digits in whatever fashion I like. Great idea! |
| 20:13.18 | kfife | Fender: Thanks |
| 20:13.27 | kfife | Fender: Have you done this for yourself? |
| 20:13.48 | [TK]D-Fender | TheCompWiz: Picture this : you use an ITSP. You'd have to fill out a PEER for them as well as a USER. the PEER section will hold the auth needed for you to SEND them a call. The USER section would have the auth required to match THEIR attempt to send YOU a call. |
| 20:14.06 | [TK]D-Fender | kfife: Never needed to, but a really simple thing to do. |
| 20:14.20 | TheCompWiz | ah. that makes more sense now. |
| 20:14.24 | keith4 | how does hold time for queues get estimated? |
| 20:14.50 | TheCompWiz | [TK]D-Fender: so... there is not a situation where you'd use a "handset" as a user/peer.... |
| 20:15.07 | TheCompWiz | only as a friend. |
| 20:15.22 | kfife | I suppose you'd do something like background(dial-tone), and wait for the appropriate DTMF input? |
| 20:15.27 | TheCompWiz | unless it was some funkey-ass phone |
| 20:15.41 | [TK]D-Fender | TheCompWiz: ... in 1.4 EVERYTHING may as well be "peer" because since then peers get matched for incoming calls. "user" just gets a higher PRIORITY |
| 20:15.54 | [TK]D-Fender | kfife: Exactly. |
| 20:16.18 | [TK]D-Fender | kfife: At which point you could set super-relaxed timeouts collecting 1 digit at a time. |
| 20:16.30 | kfife | fender: is tere a dial-tone recording provided? I'd actually prefer taht Alison Smith do a 'fake' dial tone with her voice. |
| 20:16.50 | [TK]D-Fender | kfife: I have done IVRS that literall collected digits the long way to mke an equivalent of READ that is "*" terminated instead of # termintated |
| 20:16.51 | kfife | fender: when you say one digit at a time, you're suggesting a subroutine? |
| 20:17.13 | *** part/#asterisk sorend (i=sorend@sip-proxy.gratissip.dk) |
| 20:17.16 | kfife | fender: such as a macro, building a string digit-by-digit? |
| 20:17.19 | [TK]D-Fender | kfife: a generic term, but not a "bad" one |
| 20:17.27 | [TK]D-Fender | kfife: IVR, not "macro" |
| 20:18.44 | kfife | Fender: so you'd build the whole thing out of contexts, rather than using something lke [macro-getdigit] so taht execution returns to the original context |
| 20:19.35 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 20:19.36 | TheCompWiz | [TK]D-Fender: ... ok... I think I'm grasping the concept now. it makes such little sense now to keep 'em separate (excluding in situations of using a sip-based itsp).... I guess I can understand why they merged tem. |
| 20:19.38 | TheCompWiz | *them. |
| 20:19.50 | kfife | Fender: What's your idea of 'best practices' for something iterative in the asterisk dialplan language. |
| 20:20.00 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 20:20.33 | [TK]D-Fender | kfife: Not sure how to answer that... its all so very basic and common-sense |
| 20:22.02 | kfife | Fender: I understand. I guess my question was, do you mean to suggest that you don't use dialplan macros, instead favoring dialplan goto's, labels etc? |
| 20:22.09 | *** join/#asterisk LND (n=Lee@89.192.134.32) |
| 20:23.16 | kfife | Coming from a traditional programming background I have always found the dialplan language to be a very clumsy to way to implement basic logic. |
| 20:23.50 | [TK]D-Fender | kfife: Yes I use macro's all the time. THIS is a case of playing games with TIMOUTS which is something I leave up to IVR's. Also allows you test the accumlated entry as it builds. |
| 20:24.00 | nny_1 | anyone have a clever way of making two SIP devices act like "one" where a call on one line appears on the other etc. I know it seems pointless but some of our older clients seem to think thats how a system is supposed to work >< |
| 20:24.18 | [TK]D-Fender | kfife: extensions.conf IS clumsy period. You can get the job done in it once you get past that |
| 20:24.35 | [TK]D-Fender | nny_1: |
| 20:24.45 | [TK]D-Fender | nny_1: "appears"? |
| 20:24.53 | [TK]D-Fender | nny_1: Please try to reword that. |
| 20:24.55 | kfife | fender: LOL. |
| 20:26.09 | nny_1 | [TK]D-Fender: well.. i think this is a lost cause, but a way for two sip devices to have pass a call back and forth without forwarding or parking.. I am thinking follow me might be a creative way to do it, but I haven't tried *'s implementation of follow me yet |
| 20:26.27 | kfife | Fender: have you played around with AEL2 as an alternative to Extension.conf? I spoke with Steve Murphy at Astricon last year about it somewhat. It seems to be more my speed, but it appears that very few people use it, and I'm already steeped in extensions.conf, so I've never made the plunge. |
| 20:26.43 | [TK]D-Fender | nny_1: 1.4 does have that sad fake-out SLA wanna-be. Feel free to read up on it. |
| 20:26.50 | kfife | Anyone here have experience with AEL2 ? |
| 20:27.12 | kfife | Steve Murphy is the guy at Digium behind it. |
| 20:27.18 | kfife | (fyi) |
| 20:27.24 | [TK]D-Fender | kfife: We have tiffs over it now and again, and again this morning. Codefreeze and I toss it around a bit |
| 20:27.39 | seanbright | has exp. with it |
| 20:27.50 | nny_1 | [TK]D-Fender: i'll take your high praise of the feature as a sign of it's endearing success |
| 20:27.54 | [TK]D-Fender | kfife: Its just compiles down to extensions.conf. I just makes som of the mroe common programming structures less cumbersome to implement. |
| 20:28.06 | [TK]D-Fender | nny_1: its crapTASTIC! |
| 20:28.12 | seanbright | yikes |
| 20:28.15 | nny_1 | [TK]D-Fender: I shall go absorb more internets on the buject |
| 20:28.18 | nny_1 | subject too |
| 20:28.32 | kfife | Fender: so you usually do most of yoru heavy lifing done with extension.conf? |
| 20:29.00 | [TK]D-Fender | kfife: * is 99% dialplan and 1% channel driver config |
| 20:29.23 | TheCompWiz | LOL... ain't that the truith. |
| 20:29.30 | [TK]D-Fender | kfife: And I have never had a dialplan of a complexity that I would consider implementing in AEL |
| 20:29.43 | kfife | fender: I mean extensions.con dialplan language vs AEL2 syntax to implement yoru dialplan? |
| 20:29.54 | kfife | I see. Well said. |
| 20:30.16 | [TK]D-Fender | I love the schmucks who think "hey I set up 2 SIP phones, why can't the call each other?" and those saying "now how do I make an outbound route". |
| 20:30.33 | [TK]D-Fender | kfife: I use extensions.conf direct only. |
| 20:32.22 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 20:32.22 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- Related channels: #asterisknow, #asterisk-gui, #switchvox, #freepbx, #asterisk-commits, #asterisk-bugs, #asterisk-dev |
| 20:32.40 | nny_1 | so asterisk does have support for SLA? interesting |
| 20:33.36 | kfife | Fender: I would be very intersted in seeing some dialplans that you've written to see some of the ways you get things done. Everyone seems to have their own techniques, methods etc, but you seem very thoughtful, and I would estimate that I could learn something by studying a dialplan of yours. I found that my coding efficiency jumped dramatically as a beginner when I studied John Todd's dialplan. |
| 20:35.17 | hardwire | how do you guys differentiate between internal extension caller id and outbound did matching to internal extensions, when setting caller id for outbound calls |
| 20:35.25 | hardwire | sorry if that was incomprehensible. |
| 20:35.41 | dwelsh | I have a slightly different question now: does anyone know if the AA50 Asterisk Appliance comes with a database preinstalled? |
| 20:36.12 | kfife | hardwire: It can be set in a number of places. your endpoint device can set it, Asterisk can pre-empt it, and yoru ITSP can ignore it. |
| 20:36.54 | kfife | hardwire: It's a little bit fiddly. A bit like ring generation in sip trunks |
| 20:37.53 | kfife | I have my endpoints set to specify their internal DN's (extension numbers). Then if calling to the PSTN, I pre-empt the CLID info and tell my ITSP what E.164 number I want to assign to the call. |
| 20:38.09 | kfife | I have pre-arranged with my ITSP to NOT set my callerID digits. |
| 20:38.24 | hardwire | PRI here. |
| 20:38.28 | hardwire | it passes everything |
| 20:38.39 | TheCompWiz | PRI here too... but my telco sux & won't let me send anything. |
| 20:38.45 | hardwire | including 4 digit local numbers as caller id |
| 20:38.57 | TheCompWiz | hardwire: who's the telco? |
| 20:39.01 | kfife | hardwire: does that mean that you can set your CLID to a DID that you do not own? |
| 20:39.08 | hardwire | TheCompWiz: Alaska Communication Systems |
| 20:39.13 | kfife | Or one that is outside of your rate area? |
| 20:39.14 | TheCompWiz | grrr. |
| 20:39.16 | hardwire | kfife: absolutely. |
| 20:39.26 | kfife | That's quite permissive. |
| 20:39.31 | hardwire | hah |
| 20:39.33 | kfife | I've only see that kind of permissiveness from ITSP's |
| 20:39.40 | hardwire | this is the worst state to give somebody the means to do that. |
| 20:39.43 | kfife | not from the ILECS |
| 20:40.05 | hardwire | we have $0.10/minute intra-state calling |
| 20:40.10 | kfife | ouch |
| 20:40.14 | hardwire | so yeh |
| 20:40.22 | hardwire | letting somebody hack the hell out of the phone systems is just asking for issues. |
| 20:40.33 | hardwire | they should really restrict it to only DID's I have routed to me. |
| 20:40.51 | hardwire | with a "default" one if not specified and not blocked. |
| 20:41.24 | hardwire | lots of people get in deep crap up here for just assuming since it works, it's legit. |
| 20:41.33 | kfife | I agree. I'm afraid that there are too many abusers, and some non-techical legislator is goign to pass some sweeping law that's goign to ruin it for the rest of us. |
| 20:41.43 | kfife | us, that use it properly and responsibly. |
| 20:41.47 | hardwire | kfife: that's EXACTLY what I'm afraid of. |
| 20:42.00 | hardwire | and it keeps happening |
| 20:42.34 | hardwire | I always want some scientific organization to be present at any decision making process. |
| 20:43.21 | hardwire | I'm pretty sure the entire world is noticing how fun everything is nowadays thanks to bad decision making with our laws. |
| 20:43.59 | hardwire | "All the 60 year old people in this room agree, we don't like it.. no sir." |
| 20:44.01 | kfife | hardwire: I've heard of people setting their clid to an NPA/NXX that gives them a cheaper call termination rate. For example intra-state versus inter-state can have a quite different rate because of some stupid tarrifs that the ILEC holds in their rate area. These may not apply if calling from another LATA. |
| 20:44.55 | hardwire | kfife: only if it goes off-net first. |
| 20:45.00 | hardwire | most telcos are state wide. |
| 20:45.14 | hardwire | you'd think their systems would laugh at attempts like that. |
| 20:45.21 | kfife | indeed |
| 20:45.36 | kfife | Fender: are you still there? |
| 20:45.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:46.56 | hardwire | riddlebox sounds like a fun asterisk telemarketer torture plan. |
| 20:47.17 | riddlebox | ? |
| 20:47.27 | hardwire | "Please say 'See sells sea shells by the sea shore' ten times fast after the tone." |
| 20:47.44 | hardwire | riddlebox: you could make app_riddlebox :) |
| 20:47.54 | hardwire | she |
| 20:47.55 | hardwire | blah. |
| 20:47.59 | hardwire | coffee is wearing off. |
| 20:48.03 | TheCompWiz | it's *She sells sea shells by the sea shore...* :P |
| 20:48.11 | hardwire | TheCompWiz: II KNOW. |
| 20:48.13 | hardwire | it was a test. |
| 20:48.17 | TheCompWiz | heh. |
| 20:48.22 | kfife | hardwire: w/r/t/ coffee: LOL |
| 20:48.24 | hardwire | so |
| 20:48.33 | riddlebox | why am I in this conversation? |
| 20:48.41 | Alan_Hicks | I always thought it was "She sells cshs by the sea shore." |
| 20:48.50 | Shotygun | I am wondering, do telemarketers actually play along with the telemarketer tortune dialplans? I'm personally not suffering from telemarketing in order to test it.. |
| 20:48.56 | hardwire | I'm looking at this jabra usb/telephone wireless headset thing |
| 20:49.02 | hardwire | and watching USB HID events hit my linux box. |
| 20:49.09 | hardwire | I have to know what the protocol is for this thing. |
| 20:49.13 | _MrSeb_ | hi... if my asterisk server is behind a nat and all client are behind nat too, in [general] section is correct nat=no? |
| 20:49.22 | hardwire | I would love to stop buying phones and expensive lifters and just use the headset. |
| 20:49.30 | riddlebox | I transfer any telemarketers to *99, which plays screaming-monkeys |
| 20:49.40 | hardwire | tt-monkeys ftw |
| 20:49.45 | hardwire | I use that to test my queues. |
| 20:49.51 | Shotygun | _MrSeb_: better do it nat=yes, but having the two sides in nat is not exactly the ideal thing.. |
| 20:49.56 | hardwire | it bothers my co-workers immensely. |
| 20:50.08 | hardwire | riddlebox: you're in this conversation because I made it so. |
| 20:50.10 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:50.24 | riddlebox | shhh hes here |
| 20:50.29 | Shotygun | hardwire: Years ago I used to ssh to coworkers' boxes, sudo, and cat /dev/urandom >/dev/dsp |
| 20:50.34 | Shotygun | During their mp3'ing |
| 20:50.39 | Shotygun | That really freaked them out.. |
| 20:50.39 | _MrSeb_ | Shotygun: the internal network is the same, asterisk and client are 192.168.0.0/24 |
| 20:50.47 | hardwire | Shotygun: you guys had expensive sound cards. |
| 20:50.53 | Shotygun | hardwire: sb live |
| 20:50.59 | hardwire | sexy |
| 20:51.00 | Shotygun | The onboards weren't popular as they are today |
| 20:51.14 | hardwire | I really hate most modern audio chipsets |
| 20:51.14 | Shotygun | I am talking about 2000 or so |
| 20:51.18 | Shotygun | indeed |
| 20:51.22 | hardwire | like.. why exactly did they dumb it down? |
| 20:51.29 | Shotygun | _MrSeb_: If your asterisk and client are both on the same network then no need for nat |
| 20:51.55 | hardwire | Shotygun: I got auto-pickup paging working internally. |
| 20:52.00 | hardwire | I use that to annoy people now |
| 20:52.05 | Shotygun | hardwire: The worst thing is, that unless you buy SB Live Value, everything lower than this is software hw mixing |
| 20:52.10 | hardwire | write a call file to the asterisk spool.. have them call the local strip clubs. |
| 20:52.17 | riddlebox | hardwire, I need to test that with my new dialplan, I used to have it working |
| 20:52.24 | Shotygun | SB 128bit for instance is software |
| 20:52.45 | hardwire | riddlebox: forward your boss to the local republican party office while he's in a meeting with a bunch of democrats. |
| 20:52.47 | Shotygun | I'm using SB Live in callcenter agents' workstations for amplifying crappy headsets |
| 20:52.56 | riddlebox | lol |
| 20:53.13 | hardwire | Shotygun: that's genius |
| 20:53.29 | hardwire | saves a lot of processing time on the pbx |
| 20:54.28 | Shotygun | hardwire: The CPU on my pbx is minimal, I do most of the stuff on client side |
| 20:54.40 | hardwire | Shotygun: yer a call center? |
| 20:54.46 | Shotygun | hardwire: yeah |
| 20:54.52 | Shotygun | customer service |
| 20:54.59 | Shotygun | small one though |
| 20:55.23 | *** join/#asterisk Mikeonline (i=Mike@p57A7F55E.dip.t-dialin.net) |
| 20:55.23 | hardwire | I'm working with one right now. |
| 20:55.27 | Mikeonline | hi |
| 20:55.31 | hardwire | Hi Mike! |
| 20:55.37 | hardwire | Welcome to Asterisk IRC! |
| 20:55.45 | hardwire | Mamosa? |
| 20:55.47 | n9urk | Shotygun a friend of mine has his * setup to say, "To Talk to me please press any number on your keypad" and he never got any telemarketers through ti |
| 20:56.04 | hardwire | n9urk: I've been tempted to do the same |
| 20:56.13 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 20:56.16 | hardwire | "Please play 'Mary had a little lamb' on your keypad" |
| 20:57.36 | Mikeonline | hm i cant use DB(a/b) in voicemail.conf like it is possible in extension.conf. is there any trick? |
| 20:58.06 | n9urk | hardwire: then if they do that then ask "please play Blue Danub on your key pad followed by the pound sign" |
| 20:58.13 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 21:03.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.40 | hardwire | n9urk: whistle the Strange Brew hockey tune. |
| 21:04.11 | hardwire | n9urk: "What is the konami code" |
| 21:04.12 | hardwire | hah |
| 21:04.27 | hardwire | I think voice rec could deal with that easily |
| 21:06.31 | *** part/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 21:08.06 | *** part/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 21:08.43 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 21:16.23 | kfife | QUESTION: Sometimes when I type CLI>extensions reload, the server sits idle for a 30 seconds to a minute. What's going on here? Server may or may not be heavily loaded. Just now it happened with just one call in progress. |
| 21:16.38 | kfife | ...sits idle before actually doing the reload. |
| 21:17.00 | Mikeonline | if $callerid(num) is the callers number, whats the variable for the number he called/incoming number? |
| 21:17.09 | Mikeonline | maybe he waits to finish active calls? |
| 21:17.28 | kfife | ...with this version 1.4.21.1, at least it will still process calls in this state, whereas it used to be 'frozen' |
| 21:17.53 | kfife | mikeonline: ${EXTEN} |
| 21:18.07 | kfife | ...try also DNIS |
| 21:18.40 | kfife | DNIS is more proper, but EXTEN will work unless you send the call to another context/priority |
| 21:18.42 | Mikeonline | thx kfife |
| 21:18.50 | kfife | My pleasure! |
| 21:19.03 | Mikeonline | i use it for one "script" for all incoming isdn(digital line) numbers |
| 21:19.24 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583145.dsl.bell.ca) |
| 21:19.31 | kfife | without looking at my code it may be CALLERID(dnis) |
| 21:19.43 | hardwire | kfife: you're 1.4.21.1? |
| 21:20.26 | kfife | correct |
| 21:20.36 | hardwire | do an strace -t -o debug.strace before you run asterisk |
| 21:20.46 | hardwire | then match up the times and see if it's doing some sort of awful loopage |
| 21:21.11 | hardwire | not that it will matter.. I dunno if it's doing any I/O during that time above and beyond just reading the extensions.conf again |
| 21:21.24 | kfife | you mean before I start the asterisk service, or before I do the reload? |
| 21:21.26 | hardwire | but I'm no gdb whiz |
| 21:21.56 | hardwire | strace can attach itself to an existing process ID.. but if you kill it it kills the PID |
| 21:22.18 | hardwire | oh.. and you should run it with -f to capture fork information as well |
| 21:22.20 | hardwire | anyways. |
| 21:22.29 | hardwire | #> strace ... asterisk ... |
| 21:22.32 | hardwire | same line |
| 21:22.42 | kfife | what should I be looking for in the debug file |
| 21:22.46 | kfife | ? |
| 21:23.03 | hardwire | soemthing happening at the same physical time you do a reload |
| 21:23.17 | hardwire | kfife: are you using dundi or IAX switch? |
| 21:23.51 | kfife | no, just SIP. Not even ZAP/DAHDI |
| 21:25.13 | hardwire | there needs to be a LAHDI driver for DAHDI |
| 21:25.17 | hardwire | srsly folks. |
| 21:27.16 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:27.47 | kfife | QUESTION: CDR Puzzle: Call comes in from SIP ITSP for a given DID, I set the CallerID and call another number. Call goes through perfectly. BUT BUT BUT, the CDR has no record of the original call, just my outbound call, and all instances of CLID show the CLID that I set the call to. How do I do this forward, while preserving the CLID of the ORIGINAL CALLER in the CDR? |
| 21:28.24 | Strom_C | kfife: use a Local channel so that Asterisk accounts for the calls separately |
| 21:30.01 | kfife | Strom_C: Thanks! That sounds like the right solution for sure. Question: is that the only way to make asterisk account for each leg of the call separately? |
| 21:30.45 | kfife | ...in other words is there a simple dial() switch that might also work as a quick and dirty workaround for testing purposes |
| 21:31.15 | Strom_C | you know, now that I think about it, I'm not 100% sure that the Local channel will work that way |
| 21:32.23 | kfife | Strom_C: You are right, it will work. Local channels are a gift from god. I only learned about them recently. if anyone here feels confident about their creative usage of lcoal channels, I'd love to study your dialplans. |
| 21:32.24 | Strom_C | why are you resetting the caller ID? |
| 21:33.07 | C4away | to make his dialplan more "creative" |
| 21:33.21 | kfife | Strom: Long story, but one of the endpoints for the call is a cell, and right now, the callee [sic] needs to know that the call is of a certian type. |
| 21:34.06 | Strom_C | kfife: repeat after me: "called party" |
| 21:34.16 | Strom_C | "mobile phone" |
| 21:36.50 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:38.15 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 21:38.43 | Defraz | exten => 8006583853,1,Dial(SIP/8006583853@wc-pbx.fuzecore.com) |
| 21:38.43 | Defraz | <PROTECTED> |
| 21:39.02 | Defraz | of course that would be two lines but would that send the call to wc2 if wc was offline |
| 21:39.04 | kfife | Strom: :-) |
| 21:39.10 | kfife | away for a moment |
| 21:39.51 | Strom_C | Defraz: it will, but I recommend you actually do some error testing and branch based on DIALSTATUS instead of just mindlessly calling again |
| 21:39.57 | kfife | indeed those are the proper terms. |
| 21:40.27 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 21:40.45 | Defraz | Okay are there some examples out there, I would agree that would be better, kinda like what SS7 did for TDM |
| 21:40.59 | Strom_C | Defraz: um |
| 21:41.02 | kfife | Anyone know of some dialplans that are example-worhy to help a "pretty good" dial plan writer become a "really good" dial plan writer? |
| 21:41.03 | Strom_C | this is completely unlike SS7 |
| 21:41.15 | Strom_C | kfife: just read the documentation and experiment |
| 21:41.25 | Defraz | Well, SS7 would see if the call could be completed before the call was setup, unlike before ss7 |
| 21:41.37 | Strom_C | Defraz: not quite |
| 21:41.55 | Defraz | Hmm, thought that was basicly how it worked. |
| 21:42.01 | Strom_C | no, not really |
| 21:42.18 | Defraz | Well you learn something different everyday. |
| 21:42.48 | Strom_C | Defraz: look at the GotoIf() application and the DIALSTATUS variable |
| 21:42.59 | Defraz | okay |
| 21:43.49 | Mikeonline | hm does asterisk use yacc for evaluating extensions.conf? |
| 21:45.03 | Strom_C | Mikeonline: not as far as I'm aware... |
| 21:45.09 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 21:45.39 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:46.21 | kfife | Strom: That's of course what everyone should do for incremental improvements in their abilities, but for example I found I learned a lot by carefully studying john todd's dialplan when I was beginning. Using some of those 'big picture' ideas, I was able to learn much more quickly. |
| 21:46.59 | Strom_C | kfife: yeah, but I already gave you the tool you need. If you understand what you're doing, you'll do fine. |
| 21:47.52 | kfife | ...So I'm hoping somebody who is particularly proud of their creative architecutre would willing to let me (or others) study their dialplan |
| 21:49.01 | Strom_C | you know, you'll be a lot more convincing if you just admit you learn best by example |
| 21:49.47 | florz | Mikeonline: nope - asterisk has a huge load of manually implemented parsers that generally don't do what you'd expect from them without reading the source due to the lazy implementation |
| 21:50.08 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 21:51.00 | kfife | Bingo. THat's a very good way to say it. As with any language, c, python, GERMAN, I learn much more quicly by example, particularly when it comes to big picture things, such as sentence/grammar structure. |
| 21:51.19 | eXistenZ | kfife, kannst du deutsch |
| 21:51.44 | kfife | doch |
| 21:51.56 | florz | .o( not quite ;-) |
| 21:51.58 | kfife | Ich bin ein Jahr Austauschschueler gewesen |
| 21:52.12 | eXistenZ | schön |
| 21:52.28 | kfife | Sind sie Deutscher/in |
| 21:52.39 | Strom_C | kfife: example is a good way to learn, but you need to learn to think for yourself too :) |
| 21:53.54 | kfife | I don't think anyone with half a brain would try to make a case against that. However believe it is a foolish person who does not look to others to stand on their shoulders rather than stand beside them. |
| 21:54.15 | Strom_C | yeah, blah blah blah, you can quote aphorisms at me all day |
| 21:54.27 | Strom_C | won't make any difference |
| 21:54.30 | kfife | In other words, if you've invented the wheel, should I not try to invent the car, instead of re-inventing the wheel? |
| 21:54.57 | kfife | Strom: so put your money where your mouth is: can I study YOUR dialplan? |
| 21:54.59 | florz | kfife: plus you should be aware that most dialplan code you happen to see on the web is broken in some way or another - I'd say that asterisk dialplan is more difficult to get right than C, even though it might seem just as simple as C seems at first glance ... |
| 21:55.23 | kfife | flroz: I agree with you. |
| 21:55.33 | *** part/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:56.06 | florz | kfife: really, reading the source gets you a lot further than reading other people's dialplans, I'd assume |
| 21:56.13 | Strom_C | kfife: my personal dialplan is an absolute disaster, because I'm constantly experimenting and toying around with it. All the clean dialplans I've written are the confidential property of my clients. |
| 21:58.00 | kfife | strom: :-) Mine too. I think I'd learn more in a day of carefully studying ten people's inventions to a given problem than spending the day inventing my own. |
| 21:58.23 | Strom_C | that's just because you have no confidence in your own ability to invent |
| 21:58.50 | kfife | I'd take the second half of the day to STOP studying ten dial plans and cherry-pick the best ideas, and put them together in a way that none of the ten had done |
| 22:02.43 | _MrSeb_ | someone can explain to me possible cause because my asterisk server is able to do outgoing call, but not incoming call? I've tryed it without nat and all work good (asterisk server is setted as dmz from router) |
| 22:02.51 | kfife | Strom: Ever read the book "Design Patterns"? I'm 'confident' that you'd be a better 'inventor' after reading it. ISBN-13: 978-0201633610 |
| 22:02.53 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 22:03.41 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 22:04.11 | *** join/#asterisk xpot (n=jim@75.149.224.186) |
| 22:04.19 | jasonwoot | hey, can someone pastebin an example of their agents.conf for me please? |
| 22:04.20 | kfife | MrSeb: Sounds like a NAT issue. You can do a sip debug from the console and see if the invites are reaching your server. My guess is that they're not. |
| 22:06.36 | _MrSeb_ | kfife: registration is working and I see packet with (NAT) flag |
| 22:07.36 | kfife | successful registration does not mean anything. Your NAT router has to be friendly to SIP. Many are not. |
| 22:08.06 | EmleyMoor | Are there any good small unmanaged PoE switches other than Netgear's FS108P? |
| 22:08.40 | kfife | You should try to put asterisk on the internet without DMZ/NAT in front. If that works, you know you have a NAT issues. If not, look to your ITSP to ensure they're sending the invites for inbound calls. Mine did not at fist. |
| 22:08.43 | kfife | ...first. |
| 22:09.30 | kfife | emleymoor: Look at the Linksys SRW208p. |
| 22:10.06 | kfife | ...I was going to suggest the FS116p as a joke :-) |
| 22:10.13 | _MrSeb_ | kfife: yes, but if I redirect all traffic to asterisk server teorically it must work |
| 22:11.11 | kfife | MrSeb: INCORRECT. Even 1:1 NAT is not the same as a native IP address even thought there is no port address translation. |
| 22:11.24 | EmleyMoor | Twice the price |
| 22:11.29 | kfife | ...ensure that your externip setting in sip.conf is set to your external IP address |
| 22:12.03 | Nkoji | [TK]D-Fender: I have read much about the information passed during a call but still havent found anything about information passed when a user calls out. Essentially i am calling numbers to determine wether or not they are PSTN lines or VOIP |
| 22:12.15 | EmleyMoor | More futureproof, I grant you |
| 22:12.20 | _MrSeb_ | kfife: yes, I've checked it too, I use something like dyndns |
| 22:13.13 | kfife | mrseb: try it without yoru nat/dmz. You need to knwo what your problem is before you try to solve it. |
| 22:14.10 | _MrSeb_ | kfife: this server was working before nat, now I can't use it without nat because the only system to go online is with router |
| 22:14.25 | _MrSeb_ | I need to find what settings change |
| 22:14.48 | EmleyMoor | It looks like I will be paying about 60 just to deploy one phone - but a second one in the same room will only be another 90 or so |
| 22:14.58 | EmleyMoor | 160 |
| 22:15.14 | kfife | emleymoor: Fedex LITERALLY just dropped the SRW208P two minutes ago. I bought an FX116p on ebay for $150, that gave me 8poe's. I just bought this one from e4strategies for just over $200. |
| 22:15.33 | EmleyMoor | I'm mostly Zap and softphones now - plus my N95 |
| 22:15.52 | kfife | ...It's crazy but 8 ports cost you 200, 24 ports costs you 400 |
| 22:15.56 | EmleyMoor | kfife: They seem to be cheaper that side of the pond |
| 22:16.07 | Qwell | kfife: file a claim if they LITERALLY just dropped it. |
| 22:16.25 | kfife | There's just no damned inexpensive solution for a small number of POE's |
| 22:16.34 | kfife | Quell: LOL |
| 22:16.58 | EmleyMoor | kfife: I just hope I'll manage with whatever I go for for a long time |
| 22:17.29 | kfife | mrseb: if you're using dynamic DNS you need to use the externhost (not externIP) parameter. ALso, if your IP changes, you have to restart the asterisk service. |
| 22:17.34 | EmleyMoor | (two PoE will do here - can see me using four - maybe more but I could always consider upgrading then |
| 22:18.18 | kfife | emlymoor: I bought a FS108p for $99 delivered. |
| 22:18.24 | kfife | ...no sales tax |
| 22:18.51 | kfife | ...a good investment because when you need to upgrade to 8 POE, you'll be able to get $80 for it on ebay |
| 22:19.22 | kfife | I don't think you can get 4 POE injectors for that cheap. |
| 22:19.32 | EmleyMoor | They're a minimum of 75 quid here! |
| 22:19.51 | EmleyMoor | (94 from my provider) |
| 22:20.50 | jasonwoot | anyone here using cisco asa w/voip? |
| 22:21.08 | kfife | I have just resigned myself to POE. It's just so much better than screwing around with the alternatives. |
| 22:23.16 | drmessano | <kfife> mrseb: if you're using dynamic DNS you need to use the externhost (not externIP) parameter. ALso, if your IP changes, you have to restart the asterisk service. <-- WHAT? |
| 22:24.07 | _MrSeb_ | kfife: yes, I've setted it correctly, packet I've seen in debug seems ok |
| 22:24.16 | *** join/#asterisk eaglexl (n=egsda@bzq-82-81-105-46.red.bezeqint.net) |
| 22:24.42 | eaglexl | Hey guys. if NAT is enabled on my router, but the asterisk server is in the DMZ section, should I still indicate nat=yes in sip.conf? |
| 22:24.56 | drmessano | Don't mix NAT and DMZ |
| 22:25.22 | EmleyMoor | kfife: I am short of power outlets here so PoE is a definite plus |
| 22:25.30 | eaglexl | I know its 2 different things, but I got mixed answers for this question and nothing has worked |
| 22:25.51 | drmessano | NAT it, forward the correct ports, and set the config up PROPERLY |
| 22:25.59 | drmessano | Forget DMZ |
| 22:26.05 | *** part/#asterisk JackEStorm (n=no@ip70-171-83-146.no.no.cox.net) |
| 22:26.19 | eaglexl | I forwarded 5060 and 10000-20000 according to rtp, didnt work. |
| 22:26.35 | drmessano | Then you didnt have asterisk configged properly |
| 22:26.54 | eaglexl | I can connect with no problems from my network, just not outside connections |
| 22:27.11 | drmessano | Again, not configged properly |
| 22:27.50 | _MrSeb_ | kfife: how can I try to debug the problem? outgoing call are ok |
| 22:27.54 | eaglexl | Can you help me with that? I have followed 1000 info pages and manuals, googled it too much times, but nothing is working. |
| 22:28.43 | kfife | mrseb: I might suggest doing an etherial capture to see exactly what the sip conversation looks like. I found NAT traversal and DYNDNS to be too unpredictable for my tastes even on my non-production test server at home. I bought a static IP and do 1:1 nat. |
| 22:29.03 | drmessano | externhost/externip, localnet |
| 22:29.05 | kfife | through a Soekris router running PFSense |
| 22:29.17 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-130-247.telkomadsl.co.za) |
| 22:29.19 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-29-rrdg-esr-2.dynamic.isadsl.co.za) |
| 22:29.45 | eaglexl | drm, allready configured - externhost = eaglex.dyndns.org localnet=10.0.0.2/255.255.255.0 |
| 22:29.48 | kfife | mrseb: drmessano's suggestion is correct. Also check localnet |
| 22:29.59 | drmessano | Localnet is not an IP |
| 22:30.00 | EmleyMoor | I am fortunate that I have a public /28 |
| 22:30.02 | eaglexl | oh, it wasnt for me :X |
| 22:30.05 | drmessano | 10.0.0.0 |
| 22:30.10 | drmessano | Not 10.0.0.2 |
| 22:30.39 | xacatecas | hi guys, i was hoping somehere can inform me wrt to some SIP headers. In the one direction I'm seeing "a=rtpmap:18 g729/8000/1" and in the other direction I see "a=rtpmap:18 G729/8000" |
| 22:30.46 | drmessano | localnet is your local network, not a specific IP |
| 22:30.52 | EmleyMoor | (I might be able to manage with a /29 once I've redesigned) |
| 22:30.53 | xacatecas | what's the difference with and without the /1 ? |
| 22:30.55 | drmessano | 10.0.0.0/255.255.255.0 |
| 22:30.57 | _MrSeb_ | kfife and drmessano: yes, local net is configured, 192.168.0.0/255.255.0.0 and 10.0.0.0/255.0.0.0 |
| 22:31.02 | eaglexl | thanks changed it |
| 22:31.23 | bkw_ | eaglexl: 10.0.0.2/255.255.255.0 shoudl still eval to 10.0.0.0/24 |
| 22:31.31 | drmessano | I found it does not |
| 22:31.33 | kfife | mrseb: AND? |
| 22:31.36 | bkw_ | you can also just put 10.0.0.0/24 |
| 22:31.50 | kfife | is your asterisk server multi-homed? |
| 22:31.50 | bkw_ | drmessano: I would consider it a bug if it didn't |
| 22:32.08 | _MrSeb_ | kfife: yes, I've two line, and two class, one real and one vpn |
| 22:32.27 | Mikeonline | yeah yeah yeah ya yeehaa i feel hardcore :) |
| 22:32.50 | eaglexl | drmessano, its stil not workin |
| 22:32.50 | eaglexl | g |
| 22:32.59 | drmessano | Did you take it out of the DMZ? |
| 22:33.00 | Mikeonline | so enough for tonight |
| 22:33.03 | Mikeonline | good night everyone |
| 22:33.19 | eaglexl | yep |
| 22:33.22 | kfife | Mrseb: divide and conquer: kill the VPN? Try it to see if ONE of them is causing a problem. |
| 22:33.27 | drmessano | and the ports are UDP? |
| 22:33.44 | eaglexl | yep |
| 22:33.47 | kfife | mrseb: that's just a suggestion. I've never tried asterisk with a multi-homed asterisk server |
| 22:33.50 | *** part/#asterisk Mikeonline (i=Mike@p57A7F55E.dip.t-dialin.net) |
| 22:34.21 | drmessano | are phones registering? |
| 22:34.30 | _MrSeb_ | kfife: ok, but for now the trial is only done with real network |
| 22:34.46 | eaglexl | mine is registering, from the local network |
| 22:34.56 | _MrSeb_ | drmessano: yes, I've two sip provider registered |
| 22:35.51 | drmessano | If nothing is registering from the outside, eaglexl, then you need to work on the router.. Asterisk NAT settings come into play with audio issues.. If you can register, you have a very basic port forwarding or router issue |
| 22:35.59 | drmessano | cant* |
| 22:36.19 | eaglexl | ok, I will check that agin |
| 22:36.22 | eaglexl | again |
| 22:37.15 | _MrSeb_ | drmessano: asterisk server and client are on the same network, so I don't need nat for server, but nat is only for sip provider |
| 22:38.17 | [TK]D-Fender | .... |
| 22:38.26 | Nkoji | is there any information passed back to the person that is placing the call? IE: caller Id, type of phone line |
| 22:38.27 | drmessano | _MrSeb_: I've pretty much been addressing eaglexl.. You don't need NAT considerations in Asterisk for connection to an ITSP |
| 22:39.07 | drmessano | You can put Asterisk behind a firewall with no ports forwarded and make a successful connection to an ITSP |
| 22:39.36 | [TK]D-Fender | drmessano: Sure... as long as you aren't expecting to RECEIVE calls..... |
| 22:40.25 | [TK]D-Fender | Nkoji: No. |
| 22:40.49 | EmleyMoor | When does referring to Zap as such rather than as DAHDI become deprecated? |
| 22:40.57 | Nkoji | so you cant find out any information about the number you are dialing? |
| 22:41.00 | drmessano | [TK]D-Fender: No different than a fully NAT'ed ATA connected to whatever |
| 22:41.14 | drmessano | [TK]D-Fender: You dont need open ports to recieve calls |
| 22:41.30 | drmessano | Well, forwarded |
| 22:41.33 | [TK]D-Fender | drmessano: * won't do anything to force a NAT keep-alive so that a router keeps ports forwarded |
| 22:41.38 | drmessano | s/open/forwarded |
| 22:41.46 | [TK]D-Fender | drmessano: And HELL YES you need those forwarded for * |
| 22:42.04 | [TK]D-Fender | drotherwise the UDP mappings will close up behind it |
| 22:42.54 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 22:43.00 | drmessano | So an ATA or a phone does a better job connecting to an ITSP than Asterisk? |
| 22:43.43 | _MrSeb_ | my problem is that if I do a call I see packet that go to sip provider, if I do an incoming call no packet are received from server |
| 22:43.46 | [TK]D-Fender | drmessano: Yes, they DO send keep-alives of thier own |
| 22:44.04 | [TK]D-Fender | _MrSeb_: PASTEBIN your sip.conf masking only password |
| 22:44.04 | Nkoji | [TK]D-Fender: No information is passed back to the user that places the call only to the user that is receiving the call? |
| 22:45.44 | [TK]D-Fender | Nkoji: What on earth are you talking about? When I phone someone I don't magically get information about who I'm calling. when THEY receive my call they typically get to see MY CID, etc because thats what the telco sends to them. |
| 22:46.32 | [TK]D-Fender | drmessano: ATA have neutral keep-alives, qualify, and high reg frequency to keep the mappings open. |
| 22:46.35 | _MrSeb_ | [TK]D-Fender: ok, I go to extract configuration |
| 22:48.11 | Nkoji | [TK]D-Fender: I am wondering if i can set up an autodialer that places a call to a phone and determine what system(pots, voip) that phone is on. I was just wondering if there was any information that is passed back to the user that places the call that can determine what type of phone system the receiver is using |
| 22:48.33 | eaglexl | drmessano, I double checked everything, conf files, router ports, nat, dmz is off |
| 22:48.40 | eaglexl | and.. well. cant connect from the outside |
| 22:48.41 | [TK]D-Fender | Nkoji: No, |
| 22:48.50 | eaglexl | externhost=eaglex.dyndns.org |
| 22:48.56 | eaglexl | any other setting I should put there? |
| 22:49.07 | [TK]D-Fender | eaglexl: read THE GUIDE : |
| 22:49.09 | [TK]D-Fender | ~sipnat |
| 22:49.10 | 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 |
| 22:49.11 | [TK]D-Fender | ^^^^^^^^^ |
| 22:49.21 | eaglexl | Fender |
| 22:49.26 | eaglexl | you linked me to this guide few days ago |
| 22:49.32 | eaglexl | and you tried helpimh me |
| 22:49.35 | eaglexl | it didnt help :| |
| 22:49.37 | Nkoji | [TK]D-Fender: Thank you |
| 22:50.08 | [TK]D-Fender | easpastebin your sip.conf masking only passwords |
| 22:51.51 | jameswf-home | okay arays are not the enemy misplaced "{" are |
| 22:53.22 | _MrSeb_ | [TK]D-Fender: sip.con is at http://rafb.net/p/KCsqbx19.html |
| 22:53.27 | eaglexl | http://pastebin.com/m4a23aaa1 |
| 22:55.14 | [TK]D-Fender | _MrSeb_: [general] nat=yes <---- [Eutelia] should be nat=no and HOLY SHIT canreinvite=yes <- NEVER DO THIS |
| 22:55.30 | *** join/#asterisk igorw (n=igorw@24.85.224.244) |
| 22:56.52 | [TK]D-Fender | eaglexl: You forgot to put "qualify=yes" under each or your remote peers |
| 22:56.54 | _MrSeb_ | [TK]D-Fender: I don't connect to server from internet, the client are all under the same router |
| 22:57.01 | [TK]D-Fender | eaglexl: What have you forwarded to your * box? |
| 22:57.36 | [TK]D-Fender | _MrSeb_: host=voip.eutelia.it <- whats this then? |
| 22:58.23 | eaglexl | Fender - added. what do u mean what did I forward - the ports? |
| 22:58.49 | _MrSeb_ | [TK]D-Fender: yes, but only sip providers are out of my network, in the relative section nat are setted to yes |
| 22:58.53 | [TK]D-Fender | eaglexl: Yes, what PORTS have you forwarded? |
| 22:59.05 | eaglexl | 5060, and 10000-20000 according to rtp |
| 23:00.35 | [TK]D-Fender | eaglexl: what protocols? |
| 23:00.41 | _MrSeb_ | [TK]D-Fender: checking debug info, when I set nat=yes in general, my client try to connect to the server using nat, and this is not correct |
| 23:00.53 | eaglexl | 5060 I forwarded both tcp and udp, the rtp ones udp |
| 23:01.06 | [TK]D-Fender | _MrSeb_: pastebin the SIP DEBUG for the call ateempt from CLI |
| 23:01.32 | [TK]D-Fender | eaglexl: enable SIP debug an look at call attempts |
| 23:01.44 | [TK]D-Fender | eaglexl: You are also missing EXTERNREFRESH <- |
| 23:02.02 | _MrSeb_ | [TK]D-Fender: with this configuration or with nat=yes in sip.conf? |
| 23:02.06 | eaglexl | Fender, I put externrefresh before, but it didnt really help |
| 23:03.02 | _MrSeb_ | [TK]D-Fender: the call from CLI is working, outgoing call are ok, the problem is only for incoming call |
| 23:03.42 | [TK]D-Fender | PASTEBIN <------ |
| 23:04.51 | eaglexl | Fender, There are tons of info when I make a call, but I dont see anything unusuall |
| 23:05.07 | [TK]D-Fender | PASTEBIN <------ |
| 23:06.12 | _MrSeb_ | [TK]D-Fender: outgoing call... http://rafb.net/p/Sq7AC845.html |
| 23:07.46 | eaglexl | http://pastebin.com/m2dde05f2 |
| 23:09.19 | [TK]D-Fender | _MrSeb_: your current sip.conf please. |
| 23:10.42 | [TK]D-Fender | eaglexl: And what am I supposed to be seeing in there, looks like you did a local call to an exho test. And this is looking like I've helped you on this before... |
| 23:11.30 | eaglexl | You shouldnt be seeing anything... because I CANT make or RECIVE external calls.. and yes you helped me, but yet its not workin |
| 23:12.29 | [TK]D-Fender | eaglexl: And why are you showing me some useless internal echo test? |
| 23:12.52 | eaglexl | what do u want me to show you? I cant make any other calls |
| 23:13.45 | [TK]D-Fender | eaglexl: try to call out. |
| 23:14.11 | eaglexl | but I cant call out, because no one else is able to conect to my asterix server |
| 23:14.44 | [TK]D-Fender | eaglexl: what model of router are you using? |
| 23:15.04 | eaglexl | siemens sl2-141, not something common, got it from my isp |
| 23:15.33 | [TK]D-Fender | eaglexl: have them reset their phones and watch for traffic. |
| 23:15.36 | _MrSeb_ | [TK]D-Fender: config http://rafb.net/p/YiSWjM18.html and debug info for outgoing call http://rafb.net/p/8sOrwg32.html |
| 23:15.58 | [TK]D-Fender | eaglexl: If you've got none I'm betting your host doesn't resolve properly or their config is screwed up |
| 23:16.52 | [TK]D-Fender | _MrSeb_: "REGISTER"'s have to come AFTER everything else in [general] |
| 23:17.15 | [TK]D-Fender | _MrSeb_: and I told you that for [Eutelia] you should be putting "nat=no" |
| 23:18.27 | _MrSeb_ | ok, I modify and retry |
| 23:21.44 | _MrSeb_ | [TK]D-Fender: http://rafb.net/p/xwDG1C75.html and http://rafb.net/p/Dqg8VZ47.html |
| 23:23.06 | [TK]D-Fender | _MrSeb_: [Messagenet] also looks like an ITSP for which you should put "nat=no" |
| 23:24.01 | _MrSeb_ | [TK]D-Fender: yes, but I don't use it for now, I change too it |
| 23:25.03 | [TK]D-Fender | _MrSeb_: it looks like its going OK till your sjphone cancels |
| 23:26.09 | _MrSeb_ | [TK]D-Fender: yes, I've stopped call, outgoing call is working with the previous setup too, my problem is for incoming call |
| 23:26.23 | [TK]D-Fender | _MrSeb_: show me the incoming call then |
| 23:28.02 | lmadsen | Yourname``: ping! |
| 23:30.11 | _MrSeb_ | [TK]D-Fender: is like no traffic was received for the call, but only registration packet... http://rafb.net/p/RHmpeW29.html |
| 23:32.04 | [TK]D-Fender | _MrSeb_: Call your ITSP to have them watch and have them tell you what they see |
| 23:33.11 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:34.28 | _MrSeb_ | [TK]D-Fender: the two ITSP are different... now I try with the other |
| 23:35.41 | _MrSeb_ | [TK]D-Fender: the second ITSP is ok, sjphone rings |
| 23:36.18 | _MrSeb_ | [TK]D-Fender: for the first is like the incoming call goes in timeout |
| 23:36.49 | [TK]D-Fender | _MrSeb_: Call your ITSP <_ |
| 23:37.43 | _MrSeb_ | [TK]D-Fender: ok, very thanks for the help |
| 23:43.52 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 23:44.47 | *** part/#asterisk eaglexl (n=egsda@bzq-82-81-105-46.red.bezeqint.net) |
| 23:50.34 | _MrSeb_ | I go... 'night to all |
| 23:50.39 | CrashHD | tk looks like problem is some faulty network stuff |
| 23:50.45 | *** part/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 23:50.48 | CrashHD | bad cabling maybe |
| 23:50.55 | CrashHD | but because it is a bonded link |
| 23:51.02 | CrashHD | had some weird affects |
| 23:51.12 | CrashHD | also the network switch we are using |
| 23:51.18 | CrashHD | dell pc 6248 |
| 23:51.22 | CrashHD | has some firmware issues |
| 23:51.28 | CrashHD | for dropped packets with flow control on |
| 23:51.35 | CrashHD | and dropped packets across a stacked link |
| 23:51.52 | CrashHD | not that you cared, but I thought I would share |
| 23:55.48 | [TK]D-Fender | CrashHD: Glad you found it.. |
| 23:56.03 | *** join/#asterisk `paul (n=aldee@125.252.68.126) |
| 23:56.45 | `paul | how do i set the monitor file with the agent extension, the callerid and the number dialled? |
| 23:56.52 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:57.44 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 23:59.40 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 23:59.49 | *** join/#asterisk BBHoss (n=hoss@65.4.31.168) |