| 00:00.36 | nsgn | jaytee: when i run it it lists all 8 of my channels |
| 00:00.46 | nsgn | i have two 4 port FXO modules |
| 00:01.00 | nsgn | shows them as "FXS Kewlstart" with my echo canceler and such |
| 00:01.07 | nsgn | but doesnt seem to add anything to chan_dhadi.conf |
| 00:01.10 | jaytee | ok |
| 00:01.37 | jaytee | so you need to configure chan_dahdi.conf for your 8 fxo channels then |
| 00:02.41 | nsgn | jaytee: i've had a hard time doing that. i've gotten about 50 different results on what exactly i'm supposed to do...and no concise idea of what i need to put in |
| 00:02.56 | nsgn | i'd be greatly appreciative if you could assist me with this issue |
| 00:03.22 | jaytee | greatly appreciative? you mean like as in "money"? :-) |
| 00:04.35 | nsgn | jaytee: hah, i mean like "jaytee is the best!" followed by a hug |
| 00:05.00 | nsgn | ;P i've just been stuck on this one stupid issue for a while. i've gotten the phone connected and everything on my own, somehow, despite being completely inexperienced |
| 00:05.41 | *** join/#asterisk ManxPower (n=manxpowe@193.sub-75-251-231.myvzw.com) |
| 00:08.50 | *** part/#asterisk lanning (n=lanning@nat/yahoo/x-336505e1821e8f6d) |
| 00:09.12 | nsgn | jaytee: pretty please? :) |
| 00:09.47 | jaytee | did you install by compiling or using a package? |
| 00:10.00 | nsgn | jaytee: installed via asteriskNOW |
| 00:10.13 | nsgn | detected the digium hardware no issue right away |
| 00:10.21 | nsgn | set up an extension and got a polycom phone manually connecting |
| 00:10.25 | jaytee | this is not the AsteriskNOW channel. |
| 00:10.33 | nsgn | just cant get channels |
| 00:11.08 | nsgn | jaytee: i know it's not. asteriskNOW apparently has no ability to create/modify chan_dhadi.conf. it's pretty much left up to modifying it just as you would in normal asterisk |
| 00:11.27 | nsgn | so that's why i'm here. nobody over there has much to say about it except that it's straight up asterisk |
| 00:11.37 | jaytee | http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf |
| 00:13.42 | nsgn | jaytee: i've seen that file. the examples there seem to have little to do with getting a channel out an FXO port onto POTS |
| 00:16.11 | jaytee | nsgn, actually the examples there have everything to do with getting a channel "out" of an FXO port onto POTS |
| 00:16.38 | nsgn | jaytee: then i'm painfully lost and in need of help |
| 00:17.26 | jaytee | ok, ok. settle down |
| 00:17.32 | jaytee | give me a couple minutes |
| 00:17.49 | jaytee | you have 8 channels, correct? you want them all in one trunk group or split? |
| 00:18.19 | nsgn | jaytee: forgive my tone. i do really appreciate the help. my brain is just shot after hours of diving into this head first :). |
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| 00:18.26 | nsgn | jaytee: yes, 8 channels on a digium board |
| 00:18.29 | nsgn | all FXO |
| 00:18.50 | nsgn | for outgoing calls i just need them balanced, where an outgoing call is sent over any available line |
| 00:19.02 | nsgn | incoming calls just need to all be treated as a single group, where they follow one ring pattern |
| 00:19.09 | nsgn | right now i'm focusing on outgoing |
| 00:19.11 | jaytee | ok, and do you know what the context name is for your incoming calls? |
| 00:19.42 | nsgn | i've not really set up anything for incoming at this time. it will just be something like general. we just have a few people and want all phones rung when a call comes in |
| 00:19.44 | nsgn | simple stuff |
| 00:22.03 | jaytee | http://pastebin.ca/1431584 |
| 00:22.17 | jaytee | there's your /etc/asterisk/chan_dahdi.conf |
| 00:23.06 | nsgn | thankyou! good lord i knew it had to be simple, but everything i looked at was either irrelevant or just miles long |
| 00:23.10 | nsgn | let me give it a shot |
| 00:23.29 | jaytee | you can also add echocancel=yes and echotraining=yes if you decide you need to |
| 00:24.20 | nsgn | jaytee: i've got the hardware echo canceling module, if that makes a diff |
| 00:24.32 | nsgn | i ran some config earlier that seemed to attach it in system.conf or something |
| 00:24.57 | nsgn | and once i did so the lights on it stopped dancing in their little pattern they seemed to dance in when initially powered up |
| 00:25.43 | nsgn | jaytee: rebooting, we'll have test results here in a moment |
| 00:29.21 | nsgn | jaytee: success. we hit the PSTN. however it seems to send 9 out over the PSTN with everything i dial, making it call bad numbers. wahoo |
| 00:29.33 | nsgn | when i dont dial 9, it of course tries to dial SIP and i dont get far either |
| 00:29.54 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 00:30.17 | jaytee | well, that's something probably misconfigured in the GUI |
| 00:30.42 | nsgn | yeah, i'm playing before i bother yall with it |
| 00:30.52 | nsgn | thanks for getting me on the pstn. i'm sure i'll be back later with loads more stupid questions |
| 00:30.59 | nsgn | but i'm gonna bang on it for a while before i ask blindly |
| 00:31.04 | jaytee | won't be a bother, I just won't answer since I don't run the GUI :-) |
| 00:32.34 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:34.48 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 00:36.17 | nsgn | jaytee: oh i'll save the non GUI questions for you :). thanks again. just made a call to a PSTN number and it was nice and clear |
| 00:36.41 | jaytee | cool |
| 00:37.15 | jaytee | you can add the echocancel=yes if you get echo on the lines or just add it as a default |
| 00:38.58 | jaytee | but the channel => line always has to be the last line after any settings you want to affect those channels in a group. |
| 00:39.22 | jaytee | what version of AsteriskNOW? 1.5? |
| 00:39.55 | drmessano | I wonder if someone has been punched yet for RT a tweet from someones girlfriend |
| 00:40.25 | jaytee | interesting |
| 00:40.59 | jaytee | I signed up and tried Twitter for about 2 weeks off and on. It just didn't take with me. |
| 00:41.03 | nsgn | jaytee: yes, 1.5 |
| 00:41.18 | nsgn | yeah, i dont much get twitter either |
| 00:41.22 | drmessano | I love twitter |
| 00:41.47 | jaytee | nsgn, and did you choose the FreePBX gui or the Asterisk-Gui |
| 00:41.51 | drmessano | Facebook < IRC < twitter |
| 00:42.26 | nsgn | jaytee: it defaults to freePBX |
| 00:42.28 | nsgn | so thats what i'm in |
| 00:42.40 | jaytee | drmessano, oh, c'mon!!! admit it! you're all over the web. I bet you even have a BeBu account |
| 00:43.01 | drmessano | ZOMG |
| 00:43.09 | drmessano | A BeeBuu account? |
| 00:43.27 | jaytee | or BeBo, whatever the hell it's called |
| 00:44.26 | jaytee | bebo, friendster, hi5 |
| 00:45.38 | drmessano | Friendster is for total losers |
| 00:45.43 | drmessano | Myspace rejects |
| 00:45.50 | drmessano | Which are facebook rejects |
| 00:46.18 | jaytee | but the real question is, have you ever dated an eHarmony reject? |
| 00:46.37 | drmessano | Depends on what you mean by "dated" |
| 00:47.05 | jaytee | true, that could be broadly interpreted |
| 00:50.38 | [TK]D-Fender | drmessano: "SWM seeks F for meaningful overnight relationship" |
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| 00:52.37 | drmessano | SWM seeks MWF for commandment abatement |
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| 00:54.41 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 01:31.31 | Psychobilly | hello, i have a prob with my ael code, have a look here pls http://pastebin.com/m53bb2852 |
| 01:35.13 | Psychobilly | btw i run debians stable package for asterisk 1.4.21.2 |
| 01:35.26 | Psychobilly | default lenny installation |
| 01:49.01 | nsgn | ok, i'm having a problem getting my brain around how to allow dialing outside without pressing 9 without screwing up dialing internal extensions |
| 01:49.49 | carrar | depends whats a local call and what your internal extension range is |
| 01:50.06 | carrar | make sure they don't overlap |
| 01:50.06 | nsgn | well, actually, here's what i'm hitting |
| 01:50.14 | nsgn | i can call outside just fine |
| 01:50.38 | nsgn | but somehow when i dial my three digit local extensions starting in the 100s the phone cuts off and dials just the first two digits |
| 01:50.54 | nsgn | trying to dial 103 if the phone is already off the hook and i hear a dialtone ends up with me trying to call "10" |
| 01:50.54 | carrar | check your digit maps |
| 01:51.01 | nsgn | phone or *? |
| 01:51.13 | nsgn | on the phone i'm "[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT |
| 01:51.13 | nsgn | " |
| 01:52.39 | carrar | like what |
| 01:52.51 | carrar | 111? |
| 01:53.04 | carrar | 104 |
| 01:53.13 | carrar | you have nothing there at matches |
| 01:53.27 | nsgn | yeah...but if you take the phone off-hook and press 111 |
| 01:53.33 | nsgn | the phone cuts you off after the first two and tries to call 11 |
| 01:55.20 | nsgn | if i type 111 then press DIAL it works |
| 01:55.29 | nsgn | but if i pick up and press 111 while i have a dial tone, it just dials 11 |
| 01:55.49 | jaytee | and this phone? is it a Polycom? |
| 01:56.15 | nsgn | polycom 330 |
| 01:56.20 | jaytee | yep |
| 01:56.33 | nsgn | so what gives? |
| 01:56.34 | jaytee | digitmap dialplan and timeout |
| 01:57.20 | jaytee | phone's dialplan is finding a match for two digits with no timeout |
| 01:57.35 | nsgn | i pasted the dialplan above...i dont see a match |
| 01:57.37 | nsgn | nor did carrar |
| 01:59.40 | Psychobilly | ok i found my prob, ael interpetation is rather stupid sometimes |
| 02:00.02 | jaytee | yeah, I don't see a match either. strange |
| 02:00.49 | nsgn | jaytee: i'm confused :( |
| 02:01.52 | jaytee | what's your digitmap timout look like? |
| 02:02.15 | nsgn | jaytee: on the poly or *? |
| 02:02.23 | jaytee | yeah, the polycom |
| 02:02.38 | nsgn | "3|3|3|3|3|3" |
| 02:02.51 | nsgn | could it be "Non Standard Line Seize"? |
| 02:03.48 | jaytee | what is your's set to? the default is enabled |
| 02:04.57 | nsgn | enabled |
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| 02:05.02 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 02:08.05 | nsgn | jaytee: basically everything is default on the poly. only tweak made on * is to not require 9 to dial out |
| 02:08.17 | nsgn | but this issue happened before and after that change |
| 02:14.58 | jaytee | nsgn, http://pastebin.ca/1431646 |
| 02:15.07 | carrar | probably need to read your admin guide :) |
| 02:15.37 | nsgn | jaytee: thanks |
| 02:15.42 | nsgn | carrar: my head is exploding |
| 02:15.51 | carrar | smoke some pot |
| 02:16.00 | nsgn | ;) |
| 02:16.04 | jaytee | purp by the pound |
| 02:16.05 | carrar | have a drink |
| 02:16.07 | carrar | go for a walk |
| 02:16.19 | jaytee | shave the cat |
| 02:16.24 | carrar | then read the digitmap section |
| 02:17.19 | jaytee | FTP provisioning FTW! |
| 02:17.26 | carrar | http://polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_Admin_Guide_SIP_3_1.pdf |
| 02:17.45 | jaytee | I like the whitepaper too, it has a nice breakdown in it |
| 02:22.00 | carrar | yeah it's HOT++ |
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| 02:28.46 | [TK]D-Fender | jaytee: You've been chatting with eppigy too much.... |
| 02:29.06 | rob0 | TRABAJO |
| 02:29.08 | jaytee | [TK]D-Fender, ya think? |
| 02:29.10 | jaytee | hahaha |
| 02:29.31 | [TK]D-Fender | figures napalm will be required to stop the spread of the infection... |
| 02:30.38 | [TK]D-Fender | rob0: thats MY "channel" you've hijacked! I'll have the ESP-N broadcast alliance on your ass! |
| 02:30.55 | rob0 | oops |
| 02:31.06 | [TK]D-Fender | score bonus points for all the buried references |
| 02:33.51 | carrar | WHAT |
| 02:34.32 | carrar | nsgn |
| 02:34.35 | carrar | figure it out yet |
| 02:35.42 | jaytee | I think he's abandoned VOIP and decided to pursue a career as a cooper. |
| 02:35.51 | [TK]D-Fender | mini? |
| 02:35.55 | carrar | haha |
| 02:36.01 | [TK]D-Fender | BABY YOU CAN DRIVE MY CAR! |
| 02:36.07 | carrar | Sounds like a IMPOSSIBLE MATCH for him :) |
| 02:36.45 | jaytee | no, a cooper. someone who makes barrels and casks. It's an honest trade and a dying art |
| 02:37.08 | carrar | I was making a buried regerence |
| 02:37.18 | carrar | if I could spell it right |
| 02:37.25 | carrar | heh |
| 02:39.26 | nsgn | carrar: no, was doing my ring strategy |
| 02:39.39 | carrar | fix your 1st issue |
| 02:39.48 | nsgn | got it just how i want. now i break for dinner and come back to tackle my little items like time server and internal dialing |
| 02:39.54 | nsgn | bbiab |
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| 02:41.46 | [TK]D-Fender | jaytee: Real hope for this one... |
| 02:45.52 | jaytee | my Mark 28 Veiled Sarcasm Detector just went into Red Alarm |
| 02:46.05 | hardwire | what kind of noise does that make? |
| 02:46.21 | jaytee | kind of an ah-oooop noise |
| 02:46.33 | jaytee | repeating every .5 seconds |
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| 03:37.46 | nhaynes | anyone around? |
| 03:38.01 | leifmadsen | nope |
| 03:38.16 | nhaynes | damn |
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| 03:39.34 | nhaynes | has anyone run into any problems with running an asterisk box as a vm in a production environment? I have used trixbox in a vm before and just flat out had problems but I have been playing with an install of asterisk 1.6 in a vm and so far so good. |
| 03:40.10 | nhaynes | but then again trixbox has all kinds of problems of it's own hence the switch to building the source of asterisk |
| 03:40.39 | nhaynes | i would just really like to consolidate the number of servers im running |
| 03:41.24 | lanning | I think the biggest problem would be timing sources. |
| 03:41.46 | nhaynes | what do you mean? |
| 03:42.04 | lanning | things like meetme and voicemail and voice prompts need an accurate timing source. |
| 03:42.04 | nhaynes | i've had problems in the past with ivr being really laggy |
| 03:42.27 | nhaynes | but im not really planning on using ivr anyway |
| 03:42.35 | nhaynes | im planning on offloading all of that to exchange um |
| 03:42.53 | nhaynes | i basically am just planning on using asterisk to handle my voip extensions |
| 03:42.57 | lanning | so, what is asterisk doing? pure call routing? |
| 03:43.01 | nhaynes | yea |
| 03:43.05 | nhaynes | that's all it needs to do |
| 03:43.11 | lanning | should be ok |
| 03:43.21 | nhaynes | i have a voip gateway for my phone lines |
| 03:44.41 | leifmadsen | 1.6 is getting better about being virtualized |
| 03:45.00 | nhaynes | like i said on my test virtual machine it seems to be working fine |
| 03:45.10 | nhaynes | but i haven't tried connecting it up to an exchange server yet |
| 03:45.14 | leifmadsen | yep... now it's time to load test it! |
| 03:45.38 | nhaynes | ha |
| 03:45.40 | nhaynes | not tonight |
| 03:45.58 | leifmadsen | heh |
| 03:46.18 | leifmadsen | i wanted to install ubuntu on my laptop tonight (dual-boot OSX) but I have no media |
| 03:46.49 | nhaynes | as in blank disks? |
| 03:47.17 | nhaynes | that's the beauty of virtual machines |
| 03:47.23 | nhaynes | you don't ever have to use disks |
| 03:47.27 | nhaynes | you just use the iso |
| 03:47.43 | nhaynes | i'd love to have a nice big server |
| 03:47.46 | nhaynes | and just run all vm's |
| 03:56.47 | leifmadsen | yes, I meant disks, but I want to run Linux natively since it is much faster that way :) |
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| 04:57.08 | nsgn_away | has a nice big server and runs just VMs |
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| 05:23.24 | exothermc_ | anyone know how to connect a nokia e51 to asterisk. I followed the instructions from here http://www.voip-info.org/wiki/view/Nokia but no dice |
| 05:23.41 | exothermc_ | nothing showing up in asterisk console and phone says registration failed. |
| 05:23.46 | exothermc_ | double checked server IP etc. |
| 05:24.57 | [TK]D-Fender | exothermc_: Nothing shows up on * with SIP DEBUG enabled? |
| 05:25.17 | exothermc_ | [TK]D-Fender: let me try tahat |
| 05:25.23 | [TK]D-Fender | ... |
| 05:25.26 | exothermc_ | how do enable that again? |
| 05:25.40 | [TK]D-Fender | exothermc_: You that's tantamount to looking with your eyes closed.. right? |
| 05:25.49 | [TK]D-Fender | exothermc_: "sip set debug on" |
| 05:26.01 | exothermc_ | ok got it going, let me try again. |
| 05:26.56 | exothermc_ | SIP/2.0 401 Unauthorized |
| 05:27.06 | exothermc_ | hmm odd let me check the username again. |
| 05:30.46 | exothermc_ | [TK]D-Fender: Any ideas? |
| 05:31.01 | [TK]D-Fender | exothermc_: I don't see anything and 401 tells you your auth is bad |
| 05:31.03 | exothermc_ | normal registration failures usually show up in the console. |
| 05:33.03 | exothermc_ | There is the sip trace |
| 05:33.35 | exothermc_ | not seeing anything specifically from asterisk internally as to why it send a 401 or what doesn't match |
| 05:34.10 | [TK]D-Fender | exothermc_: And I don't see configs. |
| 05:36.36 | exothermc_ | http://pastebin.com/d38a6e096 |
| 05:36.44 | exothermc_ | you looking for something other than sip.conf? |
| 05:37.43 | carrar | no user? |
| 05:38.22 | carrar | no sip port or bind ip? |
| 05:40.11 | exothermc_ | carrar: me? |
| 05:40.23 | carrar | are those in your config? |
| 05:40.36 | exothermc_ | carrar: no |
| 05:40.46 | carrar | well good luck with that then |
| 05:40.55 | exothermc_ | carrar: are the suppose to be? |
| 05:41.03 | exothermc_ | all other sip devices work fine. |
| 05:41.31 | carrar | wouldn't be the 1st time I've seen stuff not work without them |
| 05:42.00 | carrar | can you register a xlite phone with the same nokia creditals? |
| 05:42.07 | exothermc_ | carrar: let me check. |
| 05:42.09 | carrar | make sure it's not the nokia |
| 05:43.04 | exothermc_ | carrar: eyebeam works like a charm. |
| 05:43.25 | exothermc_ | there is sip trace |
| 05:44.07 | carrar | So the phone (nokia) is not on your local lan? |
| 05:44.37 | exothermc_ | carrar: right. |
| 05:44.48 | exothermc_ | carrar: kinda the plan with a mobile device. |
| 05:44.49 | carrar | so you might need to enable nat? |
| 05:44.59 | exothermc_ | carrar: it is in the sip config |
| 05:45.06 | *** part/#asterisk DarkLogik (n=darklogi@76.73.51.195) |
| 05:45.14 | carrar | ah yeah |
| 05:45.18 | exothermc_ | carrar: you can see from the sip trace that asterisk has no trouble communicating with the phone. |
| 05:45.23 | exothermc_ | it just gives a 401 |
| 05:45.30 | exothermc_ | which I think is standard |
| 05:45.43 | exothermc_ | it looks like the phone may not be getting the actually 401 |
| 05:46.16 | exothermc_ | isn't a 401 pretty standard, basically the first registration comes without a pass then 401 then proper auth? |
| 05:47.07 | carrar | sure your usename and passwords are correct |
| 05:47.35 | exothermc_ | carrar: checked them three teims |
| 05:47.38 | carrar | how about user=milesnokia |
| 05:47.43 | exothermc_ | yup |
| 05:47.52 | carrar | I don't see that |
| 05:48.42 | exothermc_ | see what? |
| 05:48.58 | carrar | user=milesnokia in [milesnokia] |
| 05:49.02 | exothermc_ | http://www.tech-invite.com/Ti-sip-CF3665.html |
| 05:49.06 | exothermc_ | that is my problem |
| 05:49.14 | exothermc_ | the nokia never responds to the 401 and it should |
| 05:49.27 | exothermc_ | looks like nat issues on the nokia |
| 05:50.10 | carrar | add: user=milesnokia |
| 05:50.15 | carrar | to sip.conf |
| 05:50.18 | carrar | in [milesnokia] |
| 05:51.01 | exothermc_ | carrar: Just look at the SIP, that isn't the issue, you are leading me down the wrong path |
| 05:51.11 | carrar | well it's obviously working for you |
| 05:51.14 | exothermc_ | I sent you a PM with the sip trace. |
| 05:51.39 | exothermc_ | You can see the nokia isn't responding to the 401 and it should. At this stage it has nothing to do with asterisk. |
| 05:52.33 | exothermc_ | looks like there is a nat fix app for nokia that I need to install. |
| 05:53.54 | exothermc_ | carrar: You see it now? |
| 06:07.31 | *** join/#asterisk Gopaul (n=Miranda@61.17.185.118) |
| 06:08.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:15.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 06:51.14 | KyleK | hrm voicemail stores files starting with msg0000 but on imap it starts at msg0001 and X-Asterisk-VM-Message-Num: 1 kinda confusing and annoying :) |
| 06:55.01 | trentcreek | how do we setup Asterisk? |
| 06:55.33 | Qwell | ~book |
| 06:55.34 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:56.09 | trentcreek | hardy har har...I just wanted Fender to get all riled out |
| 06:56.34 | KyleK | might want to see if hes online first |
| 06:57.02 | trentcreek | what else would he be doing? |
| 06:57.04 | Qwell | attempt to troll again while I'm around, and I'll show you the door. |
| 06:57.43 | trentcreek | He got all riled up before just for asking one question, just because it was too simple for him to comprehend |
| 07:01.00 | *** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld) |
| 07:01.46 | arnuld | I am using Asterisk manager API and have connected to it through 2 fds: 1 for makin calls and 2nd for reciebubg responses to calls like hangup,fail success etc. |
| 07:01.58 | *** join/#asterisk jsolis (n=Jimmy@200.106.35.142) |
| 07:02.06 | arnuld | after makign some calls, asterisk is closing the fd to which I send the calls |
| 07:02.20 | arnuld | why atserisk is cosing the fd ? |
| 07:02.24 | arnuld | closing* |
| 07:15.42 | jsolis | hey guys |
| 07:16.06 | jsolis | when the state of a dynamic agents is (busy) |
| 07:18.17 | *** part/#asterisk jsolis (n=Jimmy@200.106.35.142) |
| 07:19.29 | *** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld) |
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| 07:36.37 | *** join/#asterisk dpreacher (n=Miranda@66.236.123.130.ptr.us.xo.net) |
| 07:36.54 | dpreacher | hello |
| 07:37.13 | arnuld | Hi |
| 07:38.05 | dpreacher | Is there a way to script the action of re-registering a sip trunk on asterisknow? i've asked in their channel but there are fewer people and no one is answering. |
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| 07:38.30 | *** mode/#asterisk [+o denon] by irc.freenode.net |
| 07:39.11 | dpreacher | I'm confused between whether to use the agi-bin folder or to simply put a script in a cron job. Also I've found examples in .pl which i've not worked with before. |
| 07:39.16 | arnuld | dpreacher: well, my question is also not answered, not yet |
| 07:39.26 | arnuld | dpreacher: #asterisk is a low volume channel |
| 07:39.34 | dpreacher | I wasn't there when you asked |
| 07:39.50 | dpreacher | there are 226 users now |
| 07:40.01 | dpreacher | isnt that substantial? |
| 07:40.12 | *** join/#asterisk sfire (n=sfire@businessservers.info) |
| 07:40.32 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 07:40.32 | dpreacher | are there any good forums that are active wherein I can search for a solution on this or post my query? |
| 07:41.21 | sfire | paste your question again |
| 07:41.23 | sfire | I just joined |
| 07:42.07 | dpreacher | Is there a way to script the action of re-registering a sip trunk on asterisknow? i've asked in their channel but there are fewer people and no one is answering. sfire |
| 07:44.31 | sfire | http://forums.whirlpool.net.au/forum-replies-archive.cfm/1065622.html |
| 07:45.04 | sfire | it appeary you will have to modify the sip.conf file |
| 07:45.09 | sfire | appears* |
| 07:45.20 | dpreacher | let me read the link...thank you |
| 07:46.34 | sfire | I would do the registerattempts=0 and like 5 seconds between attempts |
| 07:47.10 | dpreacher | i see. how should i calculate the timeout between attempts? |
| 07:47.39 | sfire | I would say (just my opinion) between 2-5 seconds or so |
| 07:48.12 | sfire | or 5 as I said would probably be pretty safe |
| 07:48.23 | sfire | you want it fast enough that it doesn't stay down.. but long enough to allow it to register |
| 07:48.27 | arnuld | sfire: hey.... I have connected to Asterisk using 2 FDs, one for sending calls and other for receiving responses like hang-up, call success etc |
| 07:48.40 | arnuld | sfire: but sometimes Asterisk closes the FD automagically |
| 07:48.52 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 07:48.52 | arnuld | the FD to which I am sending callls |
| 07:49.06 | arnuld | recieving connection works without problem |
| 07:49.31 | arnuld | are there any specific situations when Asterisk closes the FD ? |
| 07:49.31 | dpreacher | there is a tiny problem. somehow the asterisk does not re-register even with registerattempts variable set and the only thing that works is a sip reload. |
| 07:49.35 | dpreacher | sfire |
| 07:49.36 | arnuld | I am using Asterisk Manager API |
| 07:49.48 | sfire | dpreacher, it hasn't re-read the configuration file |
| 07:50.07 | sfire | if all you did was modify it then you have to restart the necessary services |
| 07:50.42 | sfire | arnuld, wish I had a solution for you.. I'm a fairly new user myself |
| 07:50.57 | sfire | just figured I would lend a hand if I happened to know an answer (or could find it) |
| 07:51.37 | dpreacher | restart would mean reload itself? |
| 07:52.19 | sfire | well you could restart the entire machine.. or just restart the individual services.. |
| 07:52.40 | sfire | it reads the configuration file when it loads |
| 07:52.47 | sfire | then it doesn't look at it again until it reloads |
| 07:52.56 | dpreacher | ok...let me try |
| 07:53.37 | sfire | to test it ... if you want to make sure it will re-register... pull the ethernet cord until it unregisters then plug it back in |
| 07:53.42 | sfire | you can simulate an outtage |
| 07:53.52 | dpreacher | yes thats the plan eventually |
| 07:55.04 | sfire | granted.. I'm no asterisk wizard.. but I bet that will fix it |
| 07:56.24 | sfire | I'm setting up an asterisk server tomorrow and just wanted to get "in the mood" |
| 07:57.48 | dpreacher | all the best sfire |
| 07:58.14 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:58.56 | sfire | it'll be the first I've setup for a business (other than my own) luckly its a simple system |
| 07:59.37 | dpreacher | i'm right now on our office's first system as well |
| 08:00.40 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 08:03.40 | sfire | I'm doing an entire virtualization for them (moving 3 servers onto an ESXi box) and setting up the asterisknow server |
| 08:03.44 | sfire | should be an interesting day |
| 08:05.22 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 08:05.29 | *** part/#asterisk arnuld (n=arnuld@unaffiliated/arnuld) |
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| 08:54.29 | dpreacher | hello |
| 08:54.37 | sfire | hi |
| 08:54.45 | sfire | dpreacher, did those changes fix it? |
| 08:54.52 | dpreacher | almost there |
| 08:54.56 | dpreacher | sfire i have a query regarding the registerattempts number |
| 08:55.15 | sfire | set it to 0 0 = infinite |
| 08:55.38 | dpreacher | for testing i was keeping it to 10 attempts .... 10 seconds apart each. |
| 08:55.58 | dpreacher | but even after a time gap > 100-150 seconds. when i plugged back the ethernet |
| 08:56.06 | dpreacher | sip registered itself |
| 08:56.10 | dpreacher | which is good |
| 08:56.18 | dpreacher | but sometimes it doesn't happen |
| 08:56.23 | sfire | it probably never acknowledged the disconnect |
| 08:56.29 | dpreacher | it did |
| 08:56.38 | dpreacher | i'd done sip show registry |
| 08:56.40 | dpreacher | to verify |
| 08:56.46 | dpreacher | it'd gone unregistered |
| 08:56.51 | sfire | ok |
| 08:56.56 | dpreacher | so what am trying to determine is.. |
| 08:57.29 | dpreacher | given that i do set a custom number of attempts. its supposed to stop trying after those number of attempts right? |
| 08:57.48 | dpreacher | also one other thing...where i might be making a mistake |
| 08:57.55 | dpreacher | so let me explain that |
| 08:58.19 | sfire | yes.. if you set it to 10 attempts once it reaches 10 attempts it will stop |
| 08:58.37 | sfire | if the net is down for a long time it won't re-register |
| 08:58.46 | dpreacher | it was down for over a minute |
| 08:58.53 | dpreacher | let me explain a case here |
| 08:59.07 | dpreacher | maybe you might find the wrong i am doing |
| 09:02.16 | dpreacher | sfire the link you provided mentions sip.conf whereas in case of asterisknow/freepbx as recommended i am adding the variables to sip_general_custom.conf as this is the file where general variables are stored and its the custom file that user can add their own variables to. |
| 09:02.28 | dpreacher | I've just assumed that this be the right custom file |
| 09:03.12 | dpreacher | there also are sip_custom.conf sip_custom_post.conf sip_registrations_custom.conf...etc. |
| 09:03.51 | dpreacher | did i put the variables in the right file then? if anyone knows that this could be done from the freepbx gui then that would be even less ambiguous |
| 09:04.37 | dpreacher | sfire....any ideas |
| 09:22.02 | dpreacher | it seems like sip_general_custom.conf is the right file to configure. trying with registerattempts=0 right now |
| 09:22.36 | dpreacher | btw do i have to mention [general] keyword in sip_general_custom.conf first line |
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| 11:23.32 | linagee | is there a way to add numbers to a CID when forwarding a call? |
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| 12:04.53 | zafar_ | hey guys i have problem with my zap cart |
| 12:06.36 | zafar_ | i have installed linux and asterisk with a zap card with 8 ports |
| 12:07.19 | zafar_ | then i replaced that card with the similar card |
| 12:07.27 | zafar_ | now its not working |
| 12:08.27 | zafar_ | hey guys plz help its urgent |
| 12:08.43 | zafar_ | when i say zap show status it says not configured |
| 12:27.06 | *** join/#asterisk hfb (n=hfb@cpe-98-151-255-13.socal.res.rr.com) |
| 12:34.17 | j_kroon | is making use of an AGI the only way to generate a call-back system with asterisk or can it be done purely in the dialplan? |
| 12:44.16 | *** join/#asterisk IPGHOST (n=IPGHOST@203.175.76.14) |
| 12:45.35 | IPGHOST | hi buddeis |
| 12:45.50 | IPGHOST | any one using sangoma E1 cards??? |
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| 13:11.46 | tzafrir_laptop | doh. someone says "it's not working", ask him "what does work"? |
| 13:11.58 | tzafrir_laptop | Basic "level one support" |
| 13:12.55 | tzafrir_laptop | IPGHOST, try more specific questions. What do you want to know? Or what problems do you have? |
| 13:13.45 | IPGHOST | i ahve sangoma SMG with my binries working on a 2 yrs old gento box |
| 13:13.53 | tzafrir_laptop | (And to answer your question directly: someone probably is. That one is not me :-) |
| 13:14.30 | tzafrir_laptop | hands IPGHOST a e i o and u |
| 13:15.13 | IPGHOST | I trying to move that on centos do i need the new license? this is where confusion is any one has experaince of moving to new OS with old binries? |
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| 13:54.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:57.34 | *** join/#asterisk pzn (n=pzn@189.79.194.60) |
| 14:58.23 | pzn | Hi, I'm new to asterisk. how do I configure for the RTP always pass go from/to asterisk and clients, and never from one client to another directly? |
| 15:01.39 | [TK]D-Fender | pzn: "canreinvite=no" in every peer |
| 15:02.07 | pzn | [TK]D-Fender: thanks! easy to do... only 3 peers :-) |
| 15:03.25 | drmessano | [TK]D-Fender: Any thoughts on canreinvite=nonat ? |
| 15:04.49 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 15:05.27 | [TK]D-Fender | drmessano: Never used, so I inherently distrust :0 |
| 15:05.33 | drmessano | heh |
| 15:06.11 | drmessano | how do you see if the RTP was offloaded to the clients? |
| 15:07.58 | *** join/#asterisk Defraz (n=tim@24-117-236-174.cpe.cableone.net) |
| 15:09.49 | [TK]D-Fender | dmOnly passingly aware of "rtp debug" |
| 15:10.03 | [TK]D-Fender | drmessano: Only passingly aware of "rtp debug" |
| 15:11.43 | drmessano | Interesting.. so no way to effectively bring up a formatted list of sip channels and where the media path was last negotiated |
| 15:14.45 | *** join/#asterisk chendy_ (n=alex@119.139.82.18) |
| 15:25.36 | drmessano | Is OpenCRX the CRM that some was mentioning its close integration with Asterisk? |
| 15:26.48 | [TK]D-Fender | drmessano: SugarCRM last I heard |
| 15:28.50 | drmessano | Not so much.. the addon they have is paid and not so good.. The one I am thinking about was mentioned by some semi-regular in here, like he was part of the team.. and the name was a bit obscure.. Just ran across this OpenCRX, which looks pretty badass, and was hoping it was the same |
| 15:29.05 | drmessano | Though, the google searches are weak on OpenCRX + Asterisk |
| 15:29.21 | drmessano | and they dont have <blink>Asterisk</blink> all over the page |
| 15:30.01 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:30.43 | [TK]D-Fender | drmessano: vTiger (which IIRC is a fork of SugarCRM) is often mentioned here too |
| 15:32.18 | [TK]D-Fender | Ok, out for a while, BBIAB |
| 15:32.31 | drmessano | Yeah, and their next offering is support to have it built in the core |
| 15:32.38 | drmessano | Ok, cya later |
| 15:32.48 | drmessano | vTiger + Asterisk that is |
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| 15:54.26 | *** join/#asterisk Slavko__ (n=slavko_@the.noir.sk) |
| 15:54.39 | Slavko__ | hello all |
| 15:55.30 | Slavko__ | I would like to ask - we are receiving congestion/busy message everytime we try to make call from asterisk server, please can anyoune point me how to get this solved? |
| 15:58.12 | *** join/#asterisk Slavko__ (n=slavko_@the.noir.sk) |
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| 16:14.29 | Slavko__ | hello, is anybody here whom I can ask for help? |
| 16:19.54 | *** join/#asterisk plq (n=plq@194.27.222.252) |
| 16:22.56 | *** join/#asterisk ffreenode (n=kekoeoo@213.249.63.82) |
| 16:23.19 | ffreenode | I have an issue with SIP channels, in CLI show channels is empty, but sip show channels shows an active channel |
| 16:23.27 | ffreenode | I want to have up that channel using soft hangup |
| 16:23.44 | ffreenode | however, the only time Ive been able to do that is when it also shows up in the show channel command |
| 16:23.54 | ffreenode | is there a way of hanging up all sip channels from CLI? |
| 16:23.57 | ffreenode | if not, why not |
| 16:24.32 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
| 16:27.40 | russellb | ffreenode: "sip show channels" and "core show channels" are not a one to one relationship |
| 16:27.53 | russellb | "sip show channels" shows SIP dialogs, which may or may not be a phone call. |
| 16:28.05 | ffreenode | is there a way I can kill them? |
| 16:28.05 | russellb | For example, you will see something there for a phone's registration with the server. |
| 16:28.13 | russellb | Is there a reason you think it's wrong? |
| 16:28.21 | ffreenode | yes, because the line I call is still busy |
| 16:28.41 | ffreenode | i.e. we hang up, but it is still live / reserved |
| 16:29.06 | russellb | well if there are no calls, you can restart Asterisk, of course :-) |
| 16:29.38 | ffreenode | yeah... I dont care if there are calls |
| 16:29.41 | ffreenode | I want to kill it. |
| 16:30.02 | ffreenode | but I dont want to restart, and the calls remain connected, but zombie |
| 16:30.11 | nsgn | wow, it was actually quite easy setting up the polycoms to autoprovision |
| 16:30.15 | ffreenode | you know? I want i to hangup when I restart, but it doesnt |
| 16:30.24 | russellb | There is no command to remove an open SIP dialog. |
| 16:31.46 | ffreenode | why is that? |
| 16:32.20 | ffreenode | the call seems to die eventually, but after a long time - why cant i remove this? why am I losing control of the calls? |
| 16:36.38 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:40.40 | drmessano | ffreenode: Which version of asterisk? |
| 16:40.50 | ffreenode | 1.4 |
| 16:41.01 | drmessano | Thats not complete |
| 16:41.27 | drmessano | 1.4.______ |
| 16:42.39 | ffreenode | 1.4.21.2-2 |
| 16:43.56 | drmessano | I could be making this up, but seems like there was one recent release of 1.4 that fixed some issues like what you're having |
| 16:44.14 | drmessano | In any case, 1.4.25 is out |
| 16:44.45 | pzn | I'd like to start an echo test call from inside asterisk CLI. I'd like the phone to ring and when the person answers there is the echo test. it this possible and not too dificult to do? |
| 16:46.28 | ffreenode | drmessano, you could be making that up :-) |
| 16:46.38 | ffreenode | ok, I give up on all my issues and hope for the best, thanks |
| 16:46.51 | drmessano | Why dont you |
| 16:46.51 | drmessano | fuck off, I guess |
| 16:46.57 | drmessano | (I was thinking "update) |
| 16:47.21 | rob0 | "hope for the best" might include "update", for all we know. |
| 16:47.36 | drmessano | Doubtful.. he sounds too emo |
| 16:47.36 | rob0 | But "give up" implies not. |
| 16:50.33 | *** join/#asterisk CunningPike (n=CunningP@216.231.52.133) |
| 16:54.22 | pzn | for diagnose purposes, is it possible from inside an asterisk CLI, to run a command for ringing an extension? |
| 16:54.35 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 16:58.38 | *** join/#asterisk sambalbij (n=ipajnosn@sd5116ceb.adsl.wanadoo.nl) |
| 16:58.52 | sambalbij | hi, did anyone ever try to integrate ocs 2007 with asterisk 1.6? |
| 16:59.31 | sambalbij | the problem i have is that the incoming tcp call from ocs doesn't match the sip account, while the ip address in the sip account is static |
| 17:02.01 | nsgn | argh, anyone know why my polycom seems to be ignoring macaddress.cfg on the server? it downloads sip.dl just fine... |
| 17:02.08 | nsgn | *ld |
| 17:03.36 | nsgn | actually, better yet, i've just discovered it is getting macaddress.cfg just fine, but ignoring the contents of my custom phone and sip cfg files defining the server and extension number, etc |
| 17:08.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:08.59 | sambalbij | make sure that the custom files are loaded first |
| 17:09.34 | nsgn | sambalbij: how can i confirm they are being loaded? i know macaddress.cfg is ok cause the phone's menu will indicate to me that it is looking for my phone102.cft and local-settings.cfg, my two custom files |
| 17:10.11 | sambalbij | did you check your ftp / http log if they are retrieved? |
| 17:11.10 | *** join/#asterisk saftsack (n=saftsack@p5792479D.dip.t-dialin.net) |
| 17:11.36 | nsgn | sambalbij: GET /phone102.cfg HTTP/1.1" 200 12097 "-" "FileTransport PolycomSoundPointIP-SPIP_330-UA/3.1.3.0439" |
| 17:11.38 | nsgn | appears so |
| 17:13.53 | nsgn | sambalbij: it's grabbing my local-settings.cfg also, which is my custom sip file |
| 17:19.13 | nsgn | yeah, it's most certainly getting my config files, but seemingly ignoring them |
| 17:23.37 | pzn | any way to use dial-tones (analog ones) with other than ulaw? I'd like to use them with g729 |
| 17:30.19 | *** join/#asterisk sambalbij (n=ipajnosn@sd5116ceb.adsl.wanadoo.nl) |
| 17:31.55 | *** join/#asterisk jtodd (i=a6r0rbei@ns.fox-den.com) |
| 17:31.55 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:35.33 | *** join/#asterisk SebastianS (n=schu@adsl-dyn211.78-98-36.t-com.sk) |
| 17:37.06 | *** join/#asterisk acek__ (n=acek__@84-75-99-192.dclient.hispeed.ch) |
| 17:39.54 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 17:40.00 | ruben23 | hi |
| 17:42.08 | nsgn | hi |
| 18:05.49 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 18:12.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:12.15 | *** join/#asterisk malveo (n=malveo@79.143.115.144) |
| 18:13.36 | *** part/#asterisk malveo (n=malveo@79.143.115.144) |
| 18:15.55 | *** join/#asterisk saftsack (n=saftsack@p5792479D.dip.t-dialin.net) |
| 18:17.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:25.50 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 18:27.22 | *** join/#asterisk DarthPointer (n=no@82.218.68.216.DED-DSL.fuse.net) |
| 18:32.19 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-203-152.lns10.mel6.internode.on.net) |
| 18:46.32 | hi365 | anyone using skype for asterisk? |
| 18:46.52 | nsgn | hi365: i'm really curious what uses that has |
| 18:46.59 | nsgn | i've never used it but people ask about it a decent bit |
| 18:47.32 | hi365 | nsgn: I think the most obvious use is unlimited world calling for $9.59 p/ month |
| 18:47.34 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 18:47.53 | nsgn | hi365: arent there other, already SIP compatible providers who offer really cheap service? |
| 18:48.15 | hi365 | i doubt you can get that even from a non-reputable carier. concidering it carries the name skype makes it even better |
| 18:48.30 | hi365 | that cheap? or even close? |
| 18:48.48 | hi365 | find me one with a good reputation and ill grab it |
| 18:49.27 | nsgn | hi365: i just jump to analog lines here so i really don't know. was just curious. skype has good prices but i've always been bothered by their poor support for features like caller ID |
| 18:49.52 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 18:49.54 | hi365 | inbound or out? |
| 18:50.04 | nsgn | i dont remember now. was several years back i had played with it |
| 18:50.06 | nsgn | i think it was out |
| 18:50.16 | nsgn | either way i need it to work both ways |
| 18:51.10 | hi365 | havnt tries in, but out is semi-usable now. you can chose either a skype-in number, or a cellphone number that you verify |
| 18:52.23 | nsgn | hmm |
| 18:52.31 | nsgn | well thats better |
| 18:52.41 | nsgn | i'm curious about inbound from POTS or other non skype |
| 18:54.14 | nsgn | what are the reputable SIP services for use with skype? (i'm in the US) |
| 18:54.21 | nsgn | *asterisk |
| 18:54.23 | nsgn | not skype. argh |
| 18:56.00 | hi365 | nsgn: there are many. personaly i think teliax is pretty good. rapidvox is rock-bottom cheap, but they excpect you not to bother them - so you get what you pay for (although they might try to help you solve a problem if your nice to them) |
| 18:56.29 | hi365 | other than that, if your a home or soho user, $20-25 unlimited plans are a dime a dosen |
| 18:56.44 | nsgn | yeah |
| 18:56.45 | nsgn | ok |
| 19:02.03 | nsgn | hi365: is rapidvox pay by minute only? |
| 19:07.34 | *** join/#asterisk s14ck (n=s14ck@190-76-94-128.dyn.movilnet.com.ve) |
| 19:08.05 | carrar | If only there was a free way to make calls over the internet |
| 19:08.55 | drmessano | carrar |
| 19:09.02 | drmessano | I had this messed up idea |
| 19:09.18 | carrar | ok lets hear it |
| 19:09.25 | carrar | something to do with voip I bet |
| 19:09.29 | carrar | and free calls? |
| 19:09.30 | rob0 | There is, if you call directly to the recipient SIP-to-SIP or other such protocol. |
| 19:09.32 | drmessano | What if.. instead of using SIP and IAX to connect to an ITSP, we set up like we were an ITSP thing, like, so they could send calls to us |
| 19:09.33 | carrar | nonesense |
| 19:09.46 | carrar | woah |
| 19:09.53 | carrar | that is nutty |
| 19:09.56 | drmessano | Like SIP to some.server.com |
| 19:10.07 | drmessano | Rather than Vonage or Skype |
| 19:10.15 | drmessano | You know? |
| 19:10.16 | carrar | is that LEGAL? |
| 19:10.33 | carrar | RIAA will come after you or something |
| 19:10.41 | drmessano | I guess we would need to check.. its kinda like stealing dialtone |
| 19:10.48 | carrar | heh |
| 19:10.54 | carrar | have to put it in VPN's |
| 19:10.59 | carrar | so they can't see it |
| 19:11.29 | drmessano | But that would be so cool.. since SIP uses an Internet IP, you can just pick someone elses IP |
| 19:12.14 | carrar | woah, dial a URL |
| 19:12.18 | carrar | thats impossible |
| 19:12.33 | nsgn | i honestly dont understand why people arent just doing this in-mass and dumping stupidly price gouging POTS providers |
| 19:12.59 | carrar | because asterisk is not as simple as click on "START" |
| 19:13.17 | nsgn | well, when i say people i guess i mean administrators |
| 19:13.32 | carrar | Why aren't more people making their own hydrogen at home for energy!! |
| 19:13.57 | carrar | It's soo easy |
| 19:14.13 | nsgn | it's not cheaper than normal energy for the average consumer. IP telephony can tend to be |
| 19:15.02 | drmessano | HOLY CRAP |
| 19:15.14 | drmessano | I just dropped my water and the hydrogens fell out |
| 19:15.24 | [TK]D-Fender | blesses anoth bag of manure for drmessano |
| 19:15.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.01 | nsgn | also, darn the "Messages" button on my polycom, which does nothing no matter how i configure things. it's pretty much all i've got left not working. the entire rest of the system is ready to roll |
| 19:16.57 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 19:18.41 | nsgn | MWI works, but the actual button on the phone for MESSAGES seems inactive |
| 19:22.00 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 19:22.38 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 19:23.36 | nsgn | about 10 sources i can locate online seem to say all i need to do is set the msg.bypassInstantMessage, msg.mwi.1.callBackMode and msg.mwi.1.callBack |
| 19:23.44 | nsgn | i've got all 3 set |
| 19:24.04 | nsgn | i can press the messages softkey near the screen if i have a new message and it calls voicemail property |
| 19:24.14 | nsgn | but the physical messages button on the phone still sits dead |
| 19:24.39 | nsgn | i'd appreciate input from anyone with a polycom 330 or similar |
| 19:30.18 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:34.48 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:37.53 | carrar | nsgn |
| 19:37.58 | carrar | you only need 1 source |
| 19:38.06 | carrar | the Polycom sip admin guide |
| 19:41.11 | *** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:42.12 | nsgn | carrar: ah, i think i've found my issue. it wasnt that i'm not programming the messages functionality correctly. its that on the 330 you must do a two step process thanks to the line 2 key and the messages key sharing the same physical location. i was mistaken on how to remap it. i'm trying the fix now |
| 19:44.15 | nsgn | when you figure out their oddities, these little polycoms are nice |
| 19:44.20 | nsgn | hard to beat for the price |
| 19:46.14 | haryv | I should be outside when its nice ;) |
| 19:57.15 | nsgn | so messaging is working. current issue is that the poly takes my SNTP server from the cfg file just fine, but the gmt offset is ignored |
| 19:57.26 | nsgn | anyone hit that on a polycom? some online are saying its a glitch? |
| 20:07.18 | *** join/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 20:07.29 | PanicMan | hello |
| 20:08.52 | PanicMan | wholesale carrier Switch >>Asterisk >> SS7 PSTN Provider , but i can't see any TCP packet. i can see only UDP packet. |
| 20:08.59 | PanicMan | how can be it possible |
| 20:09.11 | PanicMan | no TCP 1720 or 5060 |
| 20:09.25 | PanicMan | seems, its working like a bridge |
| 20:09.26 | PanicMan | any idea |
| 20:12.49 | *** join/#asterisk hi365 (n=hi365@94.159.176.165) |
| 20:20.48 | *** part/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 20:23.24 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 20:27.56 | *** join/#asterisk NirS (n=NirS@77.127.209.152) |
| 20:28.01 | NirS | good evening all |
| 20:28.03 | NirS | anybody home |
| 20:28.03 | NirS | ? |
| 20:42.02 | nsgn | yes |
| 20:42.07 | nsgn | its just a lazy saturday |
| 20:46.39 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-94-rbry-bb-1.dynamic.isadsl.co.za) |
| 20:50.42 | frantic667 | <PROTECTED> |
| 20:51.34 | frantic667 | really a lazy saturday... would like to sleep, but I have to meet my girlfriend at 2 o' clock... AM... |
| 20:52.41 | drmessano | Thats a coincidence |
| 20:53.03 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.mn.warpdriveonline.com) |
| 20:53.06 | drmessano | I am supposed to meet your girlfriend at 1 o'clock... AM... |
| 20:53.38 | frantic667 | I do not think so ;-) |
| 21:00.50 | drmessano | Funny, she also told me you wouldn't have any idea |
| 21:01.58 | nsgn | --; |
| 21:05.57 | *** join/#asterisk dinhtrung (n=dinhtrun@123.24.237.44) |
| 21:06.20 | dinhtrung | hi all |
| 21:06.47 | dinhtrung | is there a way to create a trunk in realtime architecture? |
| 21:07.09 | dinhtrung | i'm trying to implement FreePBX into MySQL db |
| 21:07.24 | dinhtrung | everything works fine but trunk and outbound module |
| 21:15.02 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 21:15.41 | frantic667 | drmessano: she knows i like surprises :-) she really loves me, you see? :-D |
| 21:17.21 | carrar | frantic667, thats why I saw her and drmessano hook it up, she wanted to surprise you |
| 21:18.18 | carrar | dinhtrung= -2 points (FreePBX & MySQL) |
| 21:18.18 | frantic667 | of course, she always wants to make me happy =) |
| 21:19.08 | carrar | You must live in India |
| 21:19.47 | carrar | Where that illusion is the strongest |
| 21:19.54 | frantic667 | :-D |
| 21:20.48 | frantic667 | not really india... i live in germany, but i think the whole world is judged by illusions |
| 21:23.31 | *** join/#asterisk vvuja (n=vvuja@77-105-52-156.adsl-1.sezampro.yu) |
| 21:23.35 | vvuja | hello |
| 21:23.38 | vvuja | anyone there |
| 21:28.38 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 21:28.44 | eppigy | :D |
| 21:31.21 | nsgn | am i missing some easy way to call an IVR i've created internally? i want to hear what it's like without putting it on incoming lines quite yet |
| 21:31.27 | nsgn | make sure it dials where i want, etc |
| 21:32.13 | carrar | Your ivr a macro or extension? |
| 21:32.27 | carrar | or AGI |
| 21:33.20 | nsgn | whatever the heck the asteriskNOW module creates |
| 21:33.29 | carrar | try #asterisknow |
| 21:34.07 | carrar | or install Asterisk from source and build your system right |
| 21:36.12 | nsgn | i've asked. they're not smashingly responsive over there. their gui just generates configs for asterisk and loads them. interface aside, how does one call an IVR internally when it's not specifically bound to an extension number or feature code? |
| 21:37.16 | carrar | Macro(mymacro,8675309) |
| 21:37.52 | [TK]D-Fender | nsgn: There is no such thing as "bound to anything" |
| 21:37.54 | carrar | like I said, install Asterisk from source so you know what the hell is going on in your system |
| 21:38.13 | carrar | Bound to the Chains of AsteriskNOW |
| 21:38.22 | [TK]D-Fender | nsgn: Either a GUI writes your configs for you or they are entirely your own |
| 21:39.28 | nsgn | i'm aware. say i wanted to transfer someone to an IVR. would i have to set up a ring group to get an extension number for it? |
| 21:39.38 | edgars | hellou asterisk :) |
| 21:39.40 | carrar | 1st, install asterisk from source |
| 21:39.43 | carrar | thats step 1 |
| 21:39.58 | carrar | Let us know when you are ready for step 2 |
| 21:40.47 | nsgn | carrar: if i were interested in that, i'd have done it already. you suggested that earlier |
| 21:41.03 | carrar | then you are happy with what you have |
| 21:41.08 | carrar | move along |
| 21:41.18 | nsgn | i understand the differences and ups and downs of each. i'm attempting to keep my questions in this channel specific to asterisk concepts. GUI stuff i've been inquiring elsewhere |
| 21:41.28 | [TK]D-Fender | nsgn: I'm not sure you're following us... this is not * you are configuring, this is a **GUI** GUI's are not supported here |
| 21:41.37 | nsgn | i've gotten some good help in here on strictly asterisk concepts and appreciate those who do |
| 21:42.04 | nsgn | [TK]D-Fender: i understand. i'm not asking how to make the GUI do what i want. i'm asking how IVRs are rung if one needed to transfer a caller to one |
| 21:42.07 | nsgn | in concept |
| 21:42.17 | carrar | read the instructions |
| 21:42.24 | carrar | core show applications macro |
| 21:42.32 | lanning | AsteriskNOW did something with your configs. go figure out what AsteriskNOW did, then you will figure out what needs to be done. |
| 21:42.46 | carrar | too bad thats not a app :) |
| 21:42.51 | [TK]D-Fender | nsgn: That is the problem. that you don't understand the dialplan. Ther term "ring-group" and "transfer to IVR" is meaningless trash in *-speak. These are idaes who shape is defined by the GUI |
| 21:43.06 | carrar | core show application Macro |
| 21:43.20 | [TK]D-Fender | nsgn: This is the problem with GUI's you think that structure actually means something when in fact it doesn't |
| 21:43.23 | carrar | Everything you need is there nsgn |
| 21:43.31 | nsgn | sure. but IVRs exist in *. you can transfer calls on * too. for being my first week diving in here, i apologize for not knowing the precise terms |
| 21:43.35 | carrar | core show application ? |
| 21:43.45 | nsgn | carrar: i'll play with that, thanks |
| 21:44.57 | lanning | IVR's are created completely differently, depending on whether you hand coded it, or which GUI you used. |
| 21:45.32 | nsgn | lanning: ok, good to know. i didn't know if the GUI went about it at the same angle. what i've been learning seems to be that for some things it does, other things it just does it's own way entirely |
| 21:46.08 | lanning | IVR is a concept, not a hard fast method. |
| 21:46.09 | nsgn | the GUI has been a handy start but it sure does begin to show its limits when working up to IVRs and the like |
| 21:46.13 | nsgn | yeah |
| 21:46.14 | carrar | When you let a GUI configure your Asterisk box you are inviting the DEVIL into your house! |
| 21:46.48 | lanning | If you use a GUI, then you have to do it the GUI way. Otherwise you WILL lose. |
| 21:47.27 | nsgn | i've honestly enjoyed the GUI (for the most part, some things have already killed me) for my first time around. if i get the opportunity to play phone again in the future, knowing what i've learned now, i very well may dive into straight up * |
| 21:47.33 | carrar | That said, there's nothing that can't be fixed with some Solder and some old sound cards! |
| 21:48.09 | lanning | It's like Microsoft. good for the 80% cases it was built for, but if you need that other 20%, you have hell on your hands. |
| 21:48.15 | drmessano | Damnit, whats my trigger? |
| 21:48.23 | jaytee | don't forget those 28.8 modems in the attic. two of those will give ya 64K for ulaw when soldered to an ethernet card. |
| 21:48.32 | *** join/#asterisk cps0 (n=cps0@189-69-132-254.dial-up.telesp.net.br) |
| 21:48.36 | carrar | haha |
| 21:48.37 | lanning | it's the little lever on the bottom side of the gun... |
| 21:48.38 | carrar | or a sound card |
| 21:49.13 | drmessano | ~savemoney |
| 21:49.14 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 21:49.23 | carrar | heh |
| 21:49.24 | drmessano | YESH! |
| 21:49.28 | nsgn | after about 20 minutes playing with the polycom web gui i said heck with it and spent the past 6 hours learning remote polycom autoprovisioning |
| 21:49.43 | nsgn | i'm tickled with how flexible it is. i imagine going from asterisknow to asterisk is much the same |
| 21:49.50 | carrar | polycom web gui shoud have never been created in the 1st place |
| 21:50.01 | drmessano | nsgn: thats too bad, the next polycom firmware release uses a new XML schema |
| 21:50.05 | drmessano | We've all been learning it |
| 21:50.08 | drmessano | :( |
| 21:50.16 | nsgn | i've got the polys doing virtually everything i want right now, all the way down to a custom bitmap on screen |
| 21:50.31 | nsgn | drmessano: hah, figures |
| 21:50.54 | carrar | I wouldn't expect anything less from someone setting up polycoms |
| 21:50.55 | jaytee | drmessano, which firmware release are you referring to? |
| 21:50.58 | carrar | thats just standard |
| 21:51.32 | nsgn | sounds like carrar's a phone snob? :) |
| 21:51.53 | carrar | runs br 4.1.2rB & sip 3.1.2rB |
| 21:52.01 | carrar | snob? |
| 21:52.08 | drmessano | Wow |
| 21:52.10 | carrar | you want to use a half ass configured phone? |
| 21:52.16 | drmessano | The new scrote insulted you |
| 21:52.50 | drmessano | I need to go fix the alarm switch in my hood again |
| 21:52.54 | drmessano | :( |
| 21:53.14 | nsgn | he seems fairly down on someone diving in with affordable equipment to learn the system |
| 21:53.30 | nsgn | not everyone wants or needs the highest equipment or configuration |
| 21:53.37 | nsgn | i am curious what he uses, though. probably quite the system to see |
| 21:53.44 | drmessano | Alarm wasnt working when I got it back.. Found out from KIA that the guys that fixed my car left a rubber stopper out of the hood.. Told me without even looking at it. |
| 21:53.58 | carrar | I use Asterisk from source |
| 21:54.01 | carrar | it's free |
| 21:54.15 | nsgn | *hardware wise |
| 21:54.19 | drmessano | So it wouldnt push down the alarm switch under the hood so I could arm it |
| 21:54.22 | nsgn | i'm fairly sure most in here use * from source |
| 21:54.29 | drmessano | So .. I taped some washers over the switch |
| 21:54.40 | drmessano | 30 mins later, the tape broke and my alarm went off |
| 21:54.44 | nsgn | hah! |
| 21:54.49 | drmessano | Back to square 1 |
| 21:54.59 | nsgn | would be better if it happened while you were driving somehow |
| 21:55.09 | carrar | nsgn, since every customer likes different phones I have a few of a lot of different phones here |
| 21:55.14 | drmessano | I dont usually arm the alarm while driving |
| 21:55.29 | nsgn | carrar: cool. what devices do you tend to favor when the customer doesn't express a preference? |
| 21:55.35 | carrar | Polycom |
| 21:56.00 | nsgn | i've been pleased with my few day's experience with them, though i'm on some pretty lowly units |
| 21:56.23 | drmessano | sounds like nsgn is a phone snob |
| 21:56.27 | carrar | yeah |
| 21:56.32 | carrar | he seems fairly down on someone diving in with affordable equipment to learn the system |
| 21:56.35 | nsgn | i save my phone snobbing for my iphone ;) |
| 21:56.36 | *** join/#asterisk LakeSolon (n=blake@74-47-159-35.br1.aurr.mn.frontiernet.net) |
| 21:56.45 | drmessano | iPhone? HAH |
| 21:57.06 | nsgn | that makes me king phone snob. also makes me a pitiful slave to AT&T |
| 21:57.18 | drmessano | People actually still think they're elite with an iPhone |
| 21:57.24 | nsgn | that was high sarcasm, sir |
| 21:57.56 | nsgn | the only novelty that ever existed with iphones lasted about the first two weeks they were out |
| 21:58.18 | nsgn | then we realized we couldnt MMS or work with exchange |
| 21:58.25 | nsgn | which two years later they're now just fixing |
| 21:59.17 | nsgn | exchange support is excellent now, MMS...we'll see. speaking of exchange, what relation (if any) can * have in an environment where contacts are within exchange? |
| 21:59.43 | [TK]D-Fender | nsgn: what "contacts"? |
| 21:59.46 | carrar | all the support you can write |
| 21:59.46 | nsgn | i've not got such a situation, but hosted exchange is only getting more and more adoption in small business |
| 21:59.55 | [TK]D-Fender | nsgn: What do you expect * to do are care about them? |
| 21:59.57 | carrar | openldap |
| 22:00.17 | nsgn | [TK]D-Fender: i'm not sure. i heard ldap support was involved. i was just curious about the topic. i dont have a need for such a setup |
| 22:00.48 | [TK]D-Fender | nsgn: And you keep talking like "setup" and "integration" mean only one kind of activity |
| 22:01.07 | carrar | Just click, START, ASTERISK-SETUP, LDAP |
| 22:01.08 | [TK]D-Fender | nsgn: What is * supposed to DO with it? |
| 22:01.44 | nsgn | perhaps you tell me? i asked the question open ended. has anyone seen an asterisk installation integrated with exchange or ldap |
| 22:02.07 | nsgn | i'm curious to hear of the capability. the small taste i've gotten in two days of play have impressed me considerably |
| 22:02.07 | [TK]D-Fender | nsgn: All of your questions are open-ended it seems |
| 22:02.22 | [TK]D-Fender | nsgn: No-ons is going to have any answer for your fishing expeditions. |
| 22:02.25 | carrar | nsgn, what would you like it to do? |
| 22:02.53 | nsgn | now that the lowly goal of the system i'm configuring has essentially been met the open ended questions spring from my curiosity about a very capable open source project i've only gotten a taste of |
| 22:02.54 | [TK]D-Fender | carrar: He just said it was all "open ended". He doesn't seem to have specific goals |
| 22:03.01 | carrar | heh |
| 22:03.25 | carrar | I think setting a Asterisk box next to exchange server, they both work pretty well next to eachother |
| 22:03.36 | drmessano | [TK]D-Fender: What can asterisk do if you run the server on a network with other servers.. I am curious what the capabilities are |
| 22:03.36 | nsgn | no biting or snapping? |
| 22:03.39 | drmessano | I am taking notes |
| 22:03.39 | [TK]D-Fender | carrar: at least 3" apart for ventilation, right? |
| 22:03.46 | carrar | at least |
| 22:04.04 | carrar | 3.8752" |
| 22:04.11 | nsgn | got the formula for that? |
| 22:04.32 | drmessano | What are the capabilites of asterisk if I install it in a horizontal rack mount case? |
| 22:04.40 | [TK]D-Fender | carrar: Good. Use of arbitrary numbers has increased 43.7% in the last quarter |
| 22:04.41 | carrar | distance = (exchange users x PI) |
| 22:04.50 | nsgn | noted! |
| 22:04.51 | [TK]D-Fender | mmmmmmmm PI |
| 22:05.04 | drmessano | I asked someone earlier what the square root of water was. |
| 22:05.06 | *** join/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 22:05.19 | [TK]D-Fender | drmessano: Whats the average airspeed velocity of an unladen swallow? |
| 22:05.52 | jaytee | European or African swallow? |
| 22:05.53 | drmessano | What is the circumference of 117 volts across a circuit with 20 ohms resistance? |
| 22:05.57 | carrar | welp, time to go for a harley ride and forget I was here |
| 22:06.00 | carrar | & |
| 22:06.14 | drmessano | carrar: Do you know? |
| 22:06.17 | [TK]D-Fender | killall -9 carrar |
| 22:06.29 | [TK]D-Fender | I RELEASE THEE EVIL SPIRITS! |
| 22:06.42 | [TK]D-Fender | THE POWER OF CHRIST COMPELS THEE! |
| 22:06.55 | *** part/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 22:07.34 | jaytee | No one can withstand the power of "Christ on a Grilled Cheese Sandwich"! |
| 22:07.35 | [TK]D-Fender | drmessano: And the answer to your previous question is obvious |
| 22:08.00 | [TK]D-Fender | [18:05]<drmessano>I asked someone earlier what the square root of water was. <- Answer = The ice-cube in my scotch |
| 22:08.13 | drmessano | lol |
| 22:09.45 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 22:09.50 | [TK]D-Fender | Think I'm going to go out for indian food.... |
| 22:10.18 | *** join/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 22:10.27 | jaytee | mmmm, vindaloo |
| 22:12.59 | drmessano | SAAG PANEER |
| 22:13.01 | drmessano | SAMOSAS |
| 22:13.07 | drmessano | PAKORAS |
| 22:13.14 | *** join/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 22:13.19 | drmessano | and a big basket of naan |
| 22:13.49 | exothermc_ | [TK]D-Fender: Finally got my e51 to register, if I commented out secret. Seems that the phone doesn't want to respond to the 401 challenge. Interesting thing is that now only outbound (from the device) calls work, asterisk can't ping (via qualify) or get calls out the e51. two way sip obviously works since 18X and 200 OK makes it back to the device when it does an outbound call. |
| 22:14.58 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 22:17.04 | exothermc_ | Anyone else have any experience with the e51 or any other of the nokia devices on SIP? |
| 22:19.21 | *** join/#asterisk hi365 (n=hi365@94.159.178.104) |
| 22:19.36 | *** join/#asterisk atis_lap (n=atis_lap@193.238.213.215) |
| 22:19.55 | hi365 | is tehre an option to log unaswered calls on a per channel basis? |
| 22:21.34 | *** part/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 22:24.04 | *** part/#asterisk Errotan (n=Errotan@5403E7BF.catv.pool.telekom.hu) |
| 22:25.30 | bmoraca | why would an FXS channel hang up immediately after digits have started to be dialed? |
| 22:27.09 | bmoraca | and how the heck do i troubleshoot that? |
| 22:28.07 | [TK]D-Fender | bmoraca: bad dialplan |
| 22:28.26 | exothermc_ | hmm now that I set the realm to match on my nokia e51, and it will now register with a 401 challenge. Immediately after registration the device become unreachable, and while calls can come from the device no calls can be sent to the device. what would cause that? |
| 22:29.18 | bmoraca | [TK]D-Fender: so it hangs up immediately when it realizes digits have been dialed that don't match an extension within the context? that's strange, because sip phones that exist within the same context do not have an issue |
| 22:29.39 | [TK]D-Fender | bmoraca: SIP phones don't pass the number to * digit by digit <- |
| 22:29.43 | [TK]D-Fender | bmoraca: Apples & oranges |
| 22:29.49 | bmoraca | i suppose... |
| 22:30.37 | bmoraca | but that still doesn't quite make sense...i'm not arguing that it's possible, just saying that if I can dial 5008 within the context from a SIP phone but not from an FXS phone...doesn't make sense |
| 22:32.17 | [TK]D-Fender | bmoraca: And I'm not arguing that you aren't showing me anything useful. That is a simple fact :) |
| 22:32.18 | bmoraca | ignore me. i'm a fucking tard. stupid typos |
| 22:32.36 | bmoraca | i mistyped a letter in my context. my fault :) |
| 22:32.38 | [TK]D-Fender | bmoraca: To don't seriously think I trust your configs do you? :) |
| 22:32.44 | bmoraca | lol |
| 22:32.51 | [TK]D-Fender | you* |
| 22:33.03 | [TK]D-Fender | bmoraca: Yup.... like 99% of cases... PEBKAC |
| 22:33.28 | bmoraca | sometimes it helps to get a second head to suggest something you thought was correct just to make you double check |
| 22:44.09 | hi365 | for some reason im getting: cdr_odbc.c: cdr_odbc: Unable to connect to datasource: MySQL-cdr |
| 22:44.21 | hi365 | here are my config files: http://pastebin.ca/1432451 |
| 22:46.59 | *** join/#asterisk jo3sm1th (n=email@12.187.138.2) |
| 22:47.11 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 22:47.37 | jo3sm1th | How do you see if a SIP phone (in this case Xlite) is registered in asterisk when logged in as root in Putty... is it SHOW SIP |
| 22:49.00 | *** join/#asterisk ki4lzk (n=jjones@ip24-255-222-124.ks.ks.cox.net) |
| 22:49.05 | ki4lzk | hello all |
| 22:50.39 | ki4lzk | i am trying to setup a DID from voipvoip on my asterisk box but i don't think i am registering with there server any ideas |
| 22:51.18 | nsgn | am i just way too tired, or is it difficult to have 7 digit local dialing in a digitmap (polycom phones) if you don't want to screw up 10 digit dialing? |
| 22:53.17 | [TK]D-Fender | jo3sm1th: "sip show peer myphonesectionnamehere" in * CLI |
| 22:54.14 | [TK]D-Fender | nsgn: Your phone accepts whatever patterns you tell it to. * accepts whatever your configure it to. When they can agree, life is good. |
| 22:55.33 | nsgn | [TK]D-Fender: my issue is in the poly's digitmap. my goal would be for it to seamlessly dial both 7 and 10 digit numbers without having to press DIAL. i'm kindof wondering if the two simply conflict...since in order to dial a 10 digit number you have to dial 7 digits first. how would one prevent the dialing of a 10 digit number from being cut off at 7 digits if 7 digit dialing is configured to automatically dial? |
| 22:56.21 | nsgn | i've got the map set to work around all my internal extensions and everything...but the 7 and 10 digit dialing has me a bit stumped |
| 22:56.31 | [TK]D-Fender | nsgn: What can separate a 7 digit number you want to dial from a 10 digit on? |
| 22:57.28 | nsgn | that's what i'm trying to figure out. it's either something painfully obvious since i've been on this without a break for quite a while, or the only option may be telling people to dial slightly different (1 before 10 digit numbers or something) |
| 22:59.14 | [TK]D-Fender | nsgn: c'mon, take a guess |
| 23:00.20 | nsgn | i'm shot, man. could one not have an areacode of 512 AND a 7 digit number starting with 512? |
| 23:00.25 | nsgn | thats where i see the issue |
| 23:00.36 | [TK]D-Fender | nsgn: 1 free answer for you : TIME |
| 23:01.00 | [TK]D-Fender | nsgn: 7 digit + WAIT |
| 23:01.06 | nsgn | when it reaches 7, wait to ensure it doesnt go on to 10 |
| 23:01.25 | nsgn | ok, thats workable. thanks. i've been doing nothing but phones for nearly the past 30 hours |
| 23:01.38 | nsgn | so how do you deal with people who dial really slowly? |
| 23:01.46 | nsgn | trying to read off a small page or something |
| 23:02.02 | nsgn | you dont want the time to be too long, yet if too short and they happen to pause at 7, off you go |
| 23:02.11 | [TK]D-Fender | nsgn: Hit them. Hard. Repeat as required |
| 23:02.37 | nsgn | got ya. i'll have a bat ready. no perfect solution, but they should lear not to pause for long periods in dialing |
| 23:02.43 | nsgn | thanks :) |
| 23:03.04 | nsgn | mainly down to testing the little issues like that out of my system. i didn't find it before cause i typically dial then pick up |
| 23:03.16 | nsgn | but i know for a fact some people will never get around picking up and dialing with a buzzing phone in their hand |
| 23:04.03 | [TK]D-Fender | nsgn: Have them dial on-hook if they're slow |
| 23:04.38 | nsgn | yeah, that'd obviously be the best. the issue, as always with computers or phones or anything, you can't teach old dogs new tricks. there will always be some user who is going to do what they're going to do |
| 23:04.53 | nsgn | which means configuring the system to be as robust as possible |
| 23:04.56 | jaytee | or set the timeout for the 7 digit match to be longer than the 10 digit match |
| 23:05.08 | *** join/#asterisk rjune_ (n=rjune@38.103.117.250) |
| 23:05.20 | nsgn | jaytee: what now? |
| 23:05.30 | jaytee | exactly |
| 23:05.37 | [TK]D-Fender | jaytee: ROAD SIGN! |
| 23:05.45 | jaytee | hahahahah |
| 23:06.32 | rjune_ | I'm having trouble dialing out or in on a Digium 800 series card. |
| 23:06.38 | jaytee | nsgn, download the SIP Admin guide from Polycom if you haven't already. Read about the digitmap dialplan and the digitmap timeout. A clever guy like you might figure it out eventually |
| 23:07.08 | nsgn | oh i've had my head in it all day. i'm at the point where i'm lacking clarity because i've been going for too long. i'm about to call it quits for about 24 hours |
| 23:07.19 | nsgn | the issue is that it's honestly quite fun |
| 23:07.26 | rjune_ | asterisk 1.4, works with SIP trunks ok. |
| 23:07.29 | [TK]D-Fender | nsgn: My recommendation : x.T|*.T|#.T |
| 23:08.27 | nsgn | [TK]D-Fender: does that just pretty much leave things open, and dials after a pause whatever it is? |
| 23:08.35 | rjune_ | dahdi show status shows the card, but dahdi show channels shows no channels |
| 23:08.50 | rjune_ | Is zapata.conf no longer the location to configure these cards? |
| 23:08.51 | nsgn | rjune_: welcome to my world yesterday. you new? :) |
| 23:08.58 | rjune_ | nsgn, relatively. |
| 23:09.00 | nsgn | rjune_: i don't believe it is anymore |
| 23:09.20 | rjune_ | nsgn, any suggestions on how to make asterisk see the channels? |
| 23:09.38 | nsgn | rjune_: if one of these guys doesnt beat me to it i'll ssh in and review what i did. it wont hurt me to see it again anyhow |
| 23:10.21 | rjune_ | I would appreciate it. I got bit in the ass trying to configure this box and none of the docs match what I'm seeing |
| 23:10.30 | nsgn | rjune_: i'm fairly sure the file you're interested in is chan_dahdi.conf |
| 23:10.30 | rjune_ | looks like all the zaptel stuff went to dahdi |
| 23:10.52 | nsgn | yes, ok, my chan_dahdi.conf contains a fairly simple layout of channels from my 8xFXO setup |
| 23:11.04 | nsgn | what hardware are you using? |
| 23:11.16 | rjune_ | wildcard tdm800e |
| 23:11.24 | rjune_ | well, not 100% sure on the e part |
| 23:11.31 | rjune_ | but it's a tdm800 with two fxo modules |
| 23:12.10 | nsgn | rjune_: you have exactly what i have |
| 23:12.23 | nsgn | so you can use exactly my file, most likely |
| 23:12.42 | nsgn | what's the preferred pastebin around these parts? |
| 23:13.04 | rjune_ | I'm flexible |
| 23:13.22 | nsgn | was asking the others, mainly. some channels tend to be pretty hip on using a specific one |
| 23:13.29 | nsgn | so i've gotten used to asking |
| 23:13.45 | nsgn | standby, i'm pasting for you since nobody seems to care |
| 23:13.51 | rjune_ | ok |
| 23:14.45 | nsgn | http://pastebin.ca/1432473 |
| 23:14.58 | nsgn | rjune_: you wouldnt happen to be using asterisknow, would you? |
| 23:15.15 | rjune_ | no |
| 23:15.18 | nsgn | ok, good |
| 23:15.27 | nsgn | then you'll know what to do with the group number from there? |
| 23:15.41 | nsgn | that is a simple channel group with all 8 of the incoming lines |
| 23:15.58 | rjune_ | I think so |
| 23:16.11 | nsgn | so you can deal with group 1 from here on out |
| 23:16.17 | nsgn | let me know if it works. did the trick for me |
| 23:16.29 | *** join/#asterisk trentcreek (n=kvirc@200.94.231.94) |
| 23:16.38 | nsgn | it seems some of the older tools for autogenerating that file are pretty outdated, and not really even available anymore |
| 23:17.12 | nsgn | its such a simple config for basic operation they arent needed |
| 23:17.23 | rjune_ | your extensions.conf has OUT_1 = dahdi/g1 then for outgoing right? |
| 23:17.40 | nsgn | verifying |
| 23:18.57 | rjune_ | when I dial into a line, asterisk still doesn't see the ring either. :-/ |
| 23:19.33 | Qwell | I so don't get his quit message... |
| 23:19.39 | Qwell | I've been seeing it for how many years now? |
| 23:19.45 | nsgn | rjune_: you have to set up your inbound routes. the fact that i started on asterisknow unfortunately means i can't properly answer your past two questions |
| 23:19.54 | nsgn | because it can produce some fairly confused config files |
| 23:20.11 | nsgn | extensions.conf doesn't seem to match what even I think it should, but things are working so i'm not screwing with it at the moment |
| 23:20.21 | rjune_ | this system is freepbx, mostly has been easy and straightforward |
| 23:20.30 | nsgn | rjune_: oh...then you are basically doing what i'm doing |
| 23:20.35 | nsgn | asterisknow with freepbx |
| 23:20.40 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 23:20.51 | rjune_ | ah, ok. |
| 23:20.59 | nsgn | rjune_: if you're on freepbx and just added the channels i pasted...go to "outbound routes" and set it to ZAP/g1 |
| 23:21.08 | rjune_ | I don't know, I didn't set it up |
| 23:21.37 | nsgn | incoming is a bit odder, since you have to give it a zap channel DID before matching inbound routes, but it will work |
| 23:21.46 | nsgn | play around. dont say GUI too much in this channel anyway |
| 23:21.48 | nsgn | you'll get slapped |
| 23:21.57 | nsgn | freepbx and asterisknow both have channels |
| 23:22.16 | nsgn | freepbx's is much more active than the nearly deserted asterisknow, but they aren't as asterisk specific |
| 23:22.33 | rjune_ | ok |
| 23:22.49 | nsgn | so long as you keep it pretty strictly * related without asking GUI questions the kind ones in here will help |
| 23:23.00 | rjune_ | Added ZAP/G1 trunk |
| 23:23.05 | rjune_ | default in the outgoing dial plan |
| 23:23.13 | nsgn | however keep in mind that manual changes to conf files are frequently destroyed when you use the GUI to modify settings |
| 23:23.20 | rjune_ | right |
| 23:23.21 | nsgn | you're safe with the channels file i just gave you, though |
| 23:23.25 | rjune_ | they provide for that I thought |
| 23:23.35 | nsgn | it gets sensitive |
| 23:23.47 | nsgn | since all freepbx really does is generate conf files and feed them into * |
| 23:23.57 | nsgn | so can you call out now? |
| 23:24.22 | rjune_ | no |
| 23:24.25 | rjune_ | all circuits busy |
| 23:24.47 | nsgn | does it speak that to you or do you see it in CLI? |
| 23:24.53 | rjune_ | speak |
| 23:24.58 | rjune_ | I see it in the cli too |
| 23:25.02 | rjune_ | I have the full call trace |
| 23:26.12 | rjune_ | http://pastebin.ca/1432478 |
| 23:27.30 | nsgn | ""DAHDI/g0/" |
| 23:27.37 | nsgn | shouldnt you be going out to g1? |
| 23:28.00 | nsgn | also ensure after these changes that you're reloading |
| 23:28.22 | rjune_ | yeah, I just saw that |
| 23:28.27 | rjune_ | still no love, |
| 23:29.11 | rjune_ | http://pastebin.ca/1432481 |
| 23:29.45 | *** part/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 23:30.02 | nsgn | others may be able to drill into that further for you |
| 23:30.05 | rjune_ | ah, apparently a reload didn't help |
| 23:30.10 | rjune_ | I just restarted asterisk and it's ringing |
| 23:30.11 | nsgn | mine worked from that point, unless i'm missing something |
| 23:30.19 | nsgn | ah, very good |
| 23:30.24 | nsgn | i had done that after adding the channels |
| 23:30.40 | nsgn | was gonna say...it shouldnt be more than that |
| 23:30.49 | nsgn | going out is less complex than in, if you arent picky about which line you go over |
| 23:30.51 | ruben23 | hi can asterisk have features that can detect answering machines on call |
| 23:31.18 | nsgn | ruben23: * can do a heck of a lot, so i'd imagine so. i don't believe it's a easy peasy built in feature, though |
| 23:31.38 | nsgn | that sounds like more of an extensible functionality |
| 23:31.48 | rjune_ | nsgn, dialin is kind of working now too, thanks for the tip |
| 23:32.17 | nsgn | rjune_: no probs :). being new can be difficult. just trying to help like the few who have me |
| 23:32.28 | nsgn | * is a pretty neat world |
| 23:35.56 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 23:37.42 | [TK]D-Fender | ruben23: "core show application amd" |
| 23:38.35 | [TK]D-Fender | rjune_: Changes to Zap/DAHDI channel requires a reload of chan_zap or chan_dahdi, or a complete restart of * |
| 23:39.07 | rjune_ | [TK]D-Fender, thanks. |
| 23:39.12 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:39.29 | rjune_ | * allows for an extension to dial out to an external phone, correct? |
| 23:39.41 | nsgn | [TK]D-Fender: ook, why when using "*.T" do i try to dial *80101 and it cuts me off right away and dials *8? |
| 23:39.46 | rjune_ | that's the dahdi extension module? |
| 23:42.13 | [TK]D-Fender | nsgn: Show me the complete dialplan |
| 23:42.22 | nsgn | [TK]D-Fender: alright, one moment |
| 23:42.34 | *** join/#asterisk BrianY (n=Brian@hostshop.ru) |
| 23:42.44 | [TK]D-Fender | rjune_: "module reload chan_dahdi.so" |
| 23:42.45 | ruben23 | [TK]D-Fender:http://pastebin.com/m2ad466a9 |
| 23:42.54 | ruben23 | thats hte output |
| 23:42.59 | BrianY | Hello.I have a question about asterisk full log.Anyone available? |
| 23:43.26 | ruben23 | <PROTECTED> |
| 23:43.30 | nsgn | [TK]D-Fender: http://pastebin.ca/1432490 |
| 23:43.30 | [TK]D-Fender | ruben23: *DUH* |
| 23:43.40 | [TK]D-Fender | ruben23: Go read the INSTRUCTIONS on how to use it. |
| 23:43.58 | ruben23 | [TK]D-Fender::) sorry |
| 23:44.07 | nsgn | BrianY: just ask |
| 23:44.22 | BrianY | ok |
| 23:44.31 | nsgn | instead of waiting for someone to ask you to ask |
| 23:45.05 | BrianY | 2 hours ago i tried version 1.4.22 , of asterisk.In full log of asterisk i saw lines like > Channel SIP/w1-08200e18 was answered. |
| 23:45.30 | BrianY | I uninstalled it, because i saw a newer version and i installed 1.6.0.3-rc1 |
| 23:46.12 | BrianY | But in this version i can't see anywhere the line containing "call/channel ..was answered" |
| 23:46.14 | [TK]D-Fender | BrianY: 1.X separates major branches, and read the topic, 1.6.0.3 is old for its own branch |
| 23:46.15 | BrianY | Can you help ? |
| 23:46.29 | Qwell | newer version? 1.6.0.3-rc1? where exactly are you looking for updates? |
| 23:46.46 | BrianY | asterisk.org.probably i`m blind or smth |
| 23:47.35 | BrianY | Ok, thank you, i found Asterisk 1.6.1.0 link.I`ll install it and i`ll be back if i will be in trouble |
| 23:48.23 | nsgn | [TK]D-Fender: catch my polycom dialplan link above? |
| 23:49.38 | BrianY | BUT, if i was able to see that line on 1.4.22 , is not normal to be able to see it on 1.6.0.3-rc1 ? Sorry if this is a very dumb question, it's first time when i'm using asterisk |
| 23:52.35 | nsgn | [TK]D-Fender: found my issue through trial and error, actually. it seems what you gave me further above wont work. polycom expects x before taking . for any arbitrary number of digits |
| 23:52.37 | nsgn | odd, but it works now |
| 23:53.37 | nsgn | i cant seem to find that in a doc, but it works as "*x.T" |
| 23:58.18 | [TK]D-Fender | nsgn: Yes, I forgot the "x"'s and would have put them |
| 23:58.33 | [TK]D-Fender | nsgn: My recommendation : x.T|*x.T|#x.T |
| 23:58.55 | [TK]D-Fender | nsgn: And set your "removeendofdial" to "0" and your impossiblematch to 2 |
| 23:59.16 | nsgn | [TK]D-Fender: seems good to go now. i'm making a few other tweaks to condense what i've got. i'm probably not going entirely to your brief version, but will keep it on hand. there are a few things i want it to hit right away |
| 23:59.18 | nsgn | thanks |