| 00:01.21 | freezey | any real reason in particular that asterisk has to be started before the dahdi module is loaded? |
| 00:03.04 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:04.04 | *** join/#asterisk bobisa (~bobisa@pppoe.66.234.20.171.dslqz.com) |
| 00:04.53 | bobisa | hi, i need to have some help about asterisk |
| 00:05.08 | bobisa | someone speak french ? |
| 00:05.26 | Shazaum | say |
| 00:06.18 | bobisa | ok, im new in ipbx, i want to change my exsting pbx, and i want to know if asterisk is the best solution for me |
| 00:06.46 | Shazaum | =/ |
| 00:06.52 | Shazaum | maybe |
| 00:07.18 | bobisa | what do yo mean by maybe? |
| 00:07.54 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:08.12 | Shazaum | bobisa: Asterisk is a great solution, but it depends on its structure |
| 00:09.57 | *** join/#asterisk ZeXr0 (~ZeXr0@70.82.80.251) |
| 00:10.39 | bobisa | ok if i tell you my structure an you help me a little? |
| 00:11.48 | bobisa | i have about 15 phone, 6 phone line, and i want to integrate skype alse |
| 00:11.51 | bobisa | srry also |
| 00:14.59 | Shazaum | bobisa: wow |
| 00:15.17 | Shazaum | yes, asterisk is the best solution for you |
| 00:16.07 | bobisa | so that the answer i wanted to ear :) |
| 00:16.38 | bobisa | so where do i start, i have download a book about asterisk,but i am a little confusing. |
| 00:17.08 | bobisa | i have read that i need card to plug my phone line into my server, which one i choose? |
| 00:18.25 | Shazaum | line analog? |
| 00:18.33 | bobisa | yes |
| 00:19.28 | Shazaum | how many lines? |
| 00:20.16 | bobisa | 6 |
| 00:21.52 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:23.21 | Shazaum | bobisa: http://www.digium.com/en/products/analog/tdm800p.php |
| 00:25.00 | bobisa | thk |
| 00:25.34 | bobisa | it is easy to configure everything? |
| 00:27.15 | *** join/#asterisk aceio (~c2cbd7fe@gateway/web/freenode/x-kljjsibuuaiywuud) |
| 00:27.49 | aceio | hi all |
| 00:27.58 | bobisa | hi aceio |
| 00:28.10 | aceio | iam new too asterisk |
| 00:28.31 | aceio | i am look to install debian on 1650 dell server |
| 00:28.32 | bobisa | me two |
| 00:28.48 | aceio | cool |
| 00:29.03 | bobisa | not so cool, idont know where to start :) |
| 00:30.01 | aceio | well maybe i can help |
| 00:30.44 | aceio | what do you have problem understanding |
| 00:34.23 | NightMonkey | bobisa: Perhaps try PBX-in-a-Flash? http://pbxinaflash.net/ |
| 00:35.16 | bobisa | i like that Installation Tips for Everyman... and Woman hahahaha |
| 00:39.07 | ZeXr0 | :/ where is the sip.conf file ? |
| 00:40.47 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 00:45.15 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 00:53.50 | ZeXr0 | Is it normal if I installed AsteriskNow that within asterisk -r , I don't have the sip command ? |
| 00:55.20 | ZeXr0 | I'd like to try and connect via callcentric |
| 01:08.10 | *** join/#asterisk aandrade (~aandrade@189.34.124.123) |
| 01:15.26 | *** join/#asterisk pawz (~pawz@ppp118-208-92-171.lns20.bne4.internode.on.net) |
| 01:18.28 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-145-34-126.mia.bellsouth.net) |
| 01:19.56 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 01:25.59 | NightMonkey | Oh, the one change from 1.2 to 1.6... [flag]EXTEN@CONTXT no workie no more. Now it's EXTEN@CONTXT[,[flag]]. |
| 01:26.09 | NightMonkey | That's got my greetings back. Whew. |
| 01:26.52 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 01:27.23 | NightMonkey | Found the answer here: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail . Love that voip-info.org. :) |
| 01:33.11 | p3nguin | nightmonkey: Just so you know, the greet subdirectory that you mentioned isn't for an extension, but rather for each mailbox. |
| 01:39.41 | NightMonkey | p3nguin: Thanks. What I lack in knowledge of Asterisk's internal config organization, I make up for in boldness. ;) |
| 01:40.15 | NightMonkey | p3nguin: But is it just a semaphore, or are there supposed to be files within greet? |
| 01:43.56 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 01:44.00 | p3nguin | Extensions, devices/phones, mailboxes... all _can_ be numbered or lettered correspondingly, but the terms should never be used interchangeably when that happens to be the case. |
| 01:44.37 | p3nguin | I have nothing in greet, currently. |
| 01:47.13 | *** join/#asterisk Dibri (~gavit@200.2.163.95) |
| 01:47.36 | NightMonkey | p3nguin: Good advice. |
| 01:47.38 | *** join/#asterisk pawz (~pawz@ppp118-208-92-171.lns20.bne4.internode.on.net) |
| 01:48.04 | *** join/#asterisk Deeewayne (~dwayne@pato.shelbycountyalabama.com) |
| 01:48.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:53.35 | *** join/#asterisk nightrid3r (kvirc@41.214.128.56) |
| 01:57.30 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 01:59.22 | *** join/#asterisk pawz (~pawz@ppp118-208-92-171.lns20.bne4.internode.on.net) |
| 02:20.27 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 02:28.45 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 02:30.22 | QbY | i'm going to upgrade the os and asterisk on a box that hasn't been touched in 5 years.. only thing that i'm worried about is the zaptel card inside.. are there different versions of zaptel devices? |
| 02:32.16 | *** join/#asterisk chendy (~chatzilla@58.250.9.161) |
| 02:38.01 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 02:41.33 | manxpower | Many |
| 02:42.06 | manxpower | If you upgrade Asterisk you want to read the UPGRADE*.txt files that come with the new version of Asterisk |
| 02:55.15 | ChannelZ | reading manuals is for sissies |
| 02:55.31 | doneir | o_O |
| 03:09.54 | *** join/#asterisk thereminbr (~theremin@187.36.14.195) |
| 03:11.59 | thereminbr | Hello, I have to build an asterisk pbx which will do certain things and I wonder if anyone could point me on where to look at. I have experience with Trixbox, but as I need a customized solution I decided to setup asterisk from scratch. |
| 03:14.31 | NightMonkey | QbY: What version of Asterisk is on the box? |
| 03:14.35 | QbY | 1.2 |
| 03:14.48 | thereminbr | Basically I need two things: when an incoming call comes with callerID, I must hangup the call and store the number somewhere. If no callerID available I play an IVR which will ask for the user phone number, store somewhere and hangup. |
| 03:15.05 | NightMonkey | QbY: Funny, I just did that upgrade today, from 1.2 to 1.6. However, I have a relatively simple home-office PBX setup. |
| 03:15.26 | QbY | NightMonkey: the only thing i'm concerned about is the wildcard interface; never used one before.. I'm completely blowing out the dialplan--its a disaster |
| 03:15.46 | NightMonkey | QbY: I got bit by this, but it was a minor irratant: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail |
| 03:16.29 | manxpower | ~answers |
| 03:16.30 | infobot | well, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 03:16.35 | manxpower | QbY: read The Book |
| 03:16.48 | QbY | i'm reading now |
| 03:16.58 | QbY | just didn't know if there were any "gotchas" |
| 03:16.58 | NightMonkey | Anyone know of a method to "speak" (with festival) the CallerID during mailbox message retrieval? |
| 03:17.23 | QbY | for example, it appears that in 1.2 there was zaptel modules, now there's DAHDI |
| 03:17.35 | manxpower | NightMonkey: 1.6 has MiniVM, which lets you build your own VM in the dialplan |
| 03:17.52 | manxpower | Using the stock Asterisk Voicemail, no you can't unless you modify the source code. |
| 03:18.31 | NightMonkey | manxpower: Jeez. Enough with the freedom and flexibility, Asterisk! :) |
| 03:18.49 | NightMonkey | manxpower: Thanks, I'll look into that. |
| 03:21.07 | NightMonkey | manxpower: Wow, not a lot of docs on voip-info.org for MiniVM. Do you have suggested guides to seek? |
| 03:21.54 | [TK]D-Fender | NightMonkey: there is already a VM option for "Envelope" that reads back the callerid |
| 03:22.15 | NightMonkey | [TK]D-Fender: Ah, thanks, I'll look into that. |
| 03:23.54 | NightMonkey | [TK]D-Fender: Hrm. Will this "speak" alphanumeric text, or just the phone number? |
| 03:24.31 | [TK]D-Fender | NightMonkey: just the number AFAIK. Go check... it may offer more |
| 03:25.04 | NightMonkey | [TK]D-Fender: What I'm looking to have is something that reads "...first message, sent <date> at <time>, from ACME INC" |
| 03:25.33 | [TK]D-Fender | NightMonkey: Looks like you'll bel looking to make your own VM then |
| 03:25.59 | NightMonkey | [TK]D-Fender: Signs point to "no". Just date, and number. |
| 03:26.21 | NightMonkey | So, is the source code the best docs for MiniVM ATM? |
| 03:27.04 | manxpower | NightMonkey: I imagine other places would be best. |
| 03:27.10 | *** part/#asterisk thereminbr (~theremin@187.36.14.195) |
| 03:27.13 | manxpower | like "core show applications like voice" |
| 03:28.01 | NightMonkey | Ah, there is an extensions_minivm.conf example file in 1.6, whee. |
| 03:47.15 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:54.29 | LemensTS | I had a 1.4 asterisk server that quit letting calls go in/out. I tried to open xlite when i noticed this, and it would not register. I could log on the cli and view sip peers and channels, but had to restart now to resolve. My question is, in this instance is thier a way I could know about this? I can write bash scripts that run on cron to do asterisk cmds at the cli, but im unsure asterisk cmd would be best |
| 03:56.35 | *** join/#asterisk Terminus (~justin@124.107.174.131) |
| 03:56.44 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 04:02.46 | *** join/#asterisk iam8up (~jluthman@rrcs-24-123-230-47.central.biz.rr.com) |
| 04:03.43 | iam8up | can anyone suggest a good place to read about the cisco 7900 headset port? i want to use this port for basic analog use but it seems by simply bringing up the headset on i get no audio with a regular phone, but i do a headset - is there any kind of oddball wiring maybe? |
| 04:03.53 | *** join/#asterisk chendy (~chatzilla@58.250.9.161) |
| 04:04.39 | manxpower | iam8up: perhaps cisco.com |
| 04:05.55 | *** join/#asterisk joobie (~joobie@CPE-121-214-5-126.lnse3.win.bigpond.net.au) |
| 04:06.41 | [TK]D-Fender | Iamit is nothing like an analog phon |
| 04:06.43 | [TK]D-Fender | e |
| 04:06.45 | joobie | hey guys.. i have a script that im using to monitor if a sip peer is available.. currently i do 'sip show peer <peername>' and look at the status... problem is, ive seen the status say OK but when a call is made it then returns FORBIDDEN.. is there a better way i can check for sip peer status? short of making a test call??? |
| 04:07.19 | [TK]D-Fender | joobie: What defines a peer as "available"? They can reject you at any time |
| 04:07.48 | [TK]D-Fender | joobie: if they registered to you then you can go by there being an IP assigned |
| 04:07.59 | [TK]D-Fender | joobie: If you have qualify enabled, you can go on that. |
| 04:08.09 | [TK]D-Fender | joobie: but at any time they can refuse calls |
| 04:08.40 | iam8up | manxpower, nothing useful/relevant to my application (at least from my searching |
| 04:08.41 | joobie | if i have qualify enabled, would that have presumably changed the status on next register? |
| 04:08.52 | joobie | im just trying to see if calls can route through a sip peer |
| 04:09.05 | joobie | that one instance, i was in asterisk seeing them drop.. our monitoring said htey could route cos of the status |
| 04:09.13 | joobie | had to reload the sip module to force re-register which fixed it |
| 04:09.19 | joobie | only caught that one by chance |
| 04:09.26 | joobie | need a good way to check calls can definitely route |
| 04:09.35 | [TK]D-Fender | joobie: How can you tell when they are going to start refusing calls? |
| 04:09.47 | manxpower | joobie: Qualify sends a SIP OPTIONS packet to the peer and if it doesn't respond or takes too long to respond Asterisk will consider the peer down. |
| 04:09.50 | [TK]D-Fender | joobie: you'd have to flag things at the end of call attempts |
| 04:09.59 | iam8up | [TK]D-Fender, any idea where to get more details on this? |
| 04:10.34 | [TK]D-Fender | iam8up: What are you actually trying to accomplish? |
| 04:10.46 | manxpower | However, I don't use qualify unless it's for a peer with a crappy internet connection |
| 04:11.07 | manxpower | [TK]D-Fender: he's plugging a handset into the headset port. Dawg knows why. |
| 04:11.25 | iam8up | [TK]D-Fender, the radio station wants to use the cisco ip phones for their callers, and the easiest way to connect them to the sound board is an analog input (as they normally do this with a PSTN line) |
| 04:11.54 | [TK]D-Fender | iam8up: NO |
| 04:12.06 | iam8up | no? |
| 04:12.20 | [TK]D-Fender | iam8up: Cisco's aren't analog phones. that is not a place to plug a LINE into. |
| 04:12.36 | iam8up | no i'm not plugging the PSTN line into the headset |
| 04:12.40 | iam8up | i'm plugging the headset into the soundboard |
| 04:12.41 | [TK]D-Fender | iam8up: Do you pour gas in your windshield washer tank? |
| 04:12.48 | p3nguin | The headset is an analog device, though. |
| 04:12.56 | iam8up | p3nguin, exactly why i'm confused |
| 04:13.04 | [TK]D-Fender | iam8up: very different spc. |
| 04:13.16 | p3nguin | You just need a pin-out of the jack and/or the headset/cable. |
| 04:13.24 | iam8up | p3nguin, yes! |
| 04:13.29 | p3nguin | I'm not willing to do it right now, though. |
| 04:13.29 | iam8up | [TK]D-Fender, spc? |
| 04:13.35 | [TK]D-Fender | spec |
| 04:13.43 | iam8up | so it isn't analog out? |
| 04:14.06 | p3nguin | I'm sure there's voltage on it which is not square wave. |
| 04:14.11 | iam8up | the reason i thought it was is the cheap plantronic headsets work just fine, and are nothing (to my eye) then a speaker and mic wired into the 4p4c connector |
| 04:14.16 | [TK]D-Fender | iam8up: 4-wire <- |
| 04:14.37 | iam8up | yes so? |
| 04:14.41 | [TK]D-Fender | iam8up: not a phone. |
| 04:14.43 | iam8up | inside pair and outside pair |
| 04:14.54 | p3nguin | I suspect two wires are for the mic and two for the speaker, but I never bothered to try to figure it out. |
| 04:15.40 | iam8up | HAH! http://www.voip-info.org/wiki/view/Cisco+Phone+Headsets |
| 04:16.18 | iam8up | exactly what i was after |
| 04:16.37 | p3nguin | I don't see a pin-out on that page. |
| 04:16.53 | p3nguin | Oh, it's in the link. |
| 04:17.58 | p3nguin | That's actually a pretty nice adapter. |
| 04:18.32 | *** join/#asterisk brookshire (mbrooks@hijacked.us) |
| 04:18.40 | dlynes | man...supporting voip is way too much stress |
| 04:18.41 | p3nguin | I'd rather just use my Plantronics, though. |
| 04:19.21 | iam8up | p3nguin, connecting to a radio station soundboard...not a plantronics headset |
| 04:24.56 | joobie | [TK]D-Fender, thanks |
| 04:26.36 | *** part/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 04:27.52 | manxpower | My Anti-Drug Is Alcohol: http://www.theonion.com/content/node/33360 |
| 04:33.18 | manxpower | This one was on my TiVo when I got home tonight. Totally hilarious: http://www.theonion.com/content/video/breaking_news_some_bullshit |
| 04:49.02 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 04:51.19 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-yujnowwafhcggvge) |
| 04:58.18 | vandebo | feel free to point me to a more appropriate channel - Any recommendations for a per minute outbound provider (usa)? I was using voicepulse, but they've instituted a monthly minimum. |
| 04:59.00 | [TK]D-Fender | vandebo: voip.ms has a decent rep so far |
| 04:59.53 | p3nguin | I use voip.ms, but I have some friends using flowroute without complaints. |
| 05:00.46 | p3nguin | VoIP.ms offers termination to USA numbers as low at 1.05 cents per minute, and Flowroute as low as 0.98 cents per minute. |
| 05:01.36 | vandebo | hmm, the .ms domain seems a bit sketchy |
| 05:02.17 | p3nguin | I don't even know what that's all about, really. |
| 05:03.29 | vandebo | though they have a pop in my city so lower latency |
| 05:04.29 | p3nguin | I'm glad I chose VoIP.ms, actually. |
| 05:04.51 | p3nguin | It is certainly better than a lot of the alternatives. |
| 05:05.00 | vandebo | any particular reason, or just a good experience so far? |
| 05:06.21 | p3nguin | I haven't had any problems other than at least two of their proxies think I want to run comfort noise, which throws a notice on my console. |
| 05:06.27 | manxpower | ~itsp |
| 05:06.28 | infobot | [~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 |
| 05:07.27 | vandebo | manxpower: thanks |
| 05:07.55 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 05:08.10 | vandebo | anyone know why voicepulse went a monthly minimum? just trying to stay profitable? |
| 05:19.53 | *** join/#asterisk roe (~roe___@216-164-214-182.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) |
| 05:24.54 | manxpower | I use Vitelity at the moment. |
| 05:26.35 | *** join/#asterisk chendy_ (~chatzilla@58.250.9.245) |
| 05:28.29 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 05:35.07 | p3nguin | VoIP.ms uses Vitelity, too. |
| 05:54.49 | tkrn | when a voip provider offers "additional incomming channels" that basically means i can have 2 calls come in instead of 1 basically |
| 05:57.26 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:03.31 | p3nguin | tkrn: They usually give you at least 2 incoming channels already, so "additional" would be 3 or more. |
| 06:04.28 | *** join/#asterisk keith4__ (~keith@unaffiliated/keith4) |
| 06:05.07 | tkrn | thats what i thought, thanks for the heads up |
| 06:05.32 | tkrn | this damn iaxmodem wont register! |
| 06:06.37 | p3nguin | What seems to be the problem? |
| 06:07.01 | keith4__ | i have a strange problem. using realtime SIP friends, the CLI gets spammed with qualify notifications from chan_sip at 10 lines/second |
| 06:07.14 | keith4__ | like "[Mar 16 02:02:17] NOTICE[24970] chan_sip.c: Peer '606' is now Reachable. (34ms / 400ms)" |
| 06:09.47 | ChannelZ | The same one over and over? |
| 06:10.05 | keith4__ | all of them |
| 06:10.08 | tkrn | well i am working with Elastix, and the iaxmodem registration times out registering... ports are right, iax extension is there |
| 06:10.23 | keith4__ | ChannelZ: i turned off all phones except 1, right now |
| 06:10.43 | ChannelZ | keith4_: but the same one (606) is spewing? |
| 06:10.53 | keith4__ | ChannelZ: all of them do it |
| 06:11.09 | ChannelZ | ok so don't answer the question |
| 06:11.31 | ChannelZ | your device is fucked, there's your answer |
| 06:11.51 | keith4__ | no |
| 06:12.15 | keith4__ | ChannelZ: if I stop and start asterisk... it's quiet until it loads the sip user from the DB. like, 'sip notify polycom-check-conf 606' causes the NOTICE messages to start |
| 06:13.26 | ChannelZ | I'll ask one more time. Any 1 single peer is spewing registrations at "10 lines/second"? |
| 06:14.25 | keith4__ | I'll answer one more time, then: yes. each and every peer does it. if I turn on 3 phones, 3 different peers are interspersed, spewing at 10 to 15 lines/second |
| 06:16.02 | ChannelZ | What I was trying to clarify is if all of them together were making ~10lps total or if every single one each was doing ~10lps individually |
| 06:16.50 | keith4__ | e.g.: http://pastebin.com/RveuXgCx |
| 06:16.58 | ChannelZ | Do you get corresponding 'peer XXX is unreachable'? |
| 06:17.25 | ChannelZ | nevermind |
| 06:18.06 | p3nguin | Is your qualify time set to 400? |
| 06:18.07 | ChannelZ | Can you turn qualify off for the peers? |
| 06:18.33 | keith4__ | p3nguin: yes |
| 06:18.36 | keith4__ | ChannelZ: sure |
| 06:18.58 | ChannelZ | And do they stop? |
| 06:19.03 | keith4__ | let me try |
| 06:19.05 | p3nguin | If you turn it off, no OPTIONS should be sent, so ther won't be anything to respond to. |
| 06:19.33 | keith4__ | can I do that globally? or do I have to do each one individually? |
| 06:19.52 | p3nguin | Did you enable it for every one individually?? |
| 06:19.54 | ChannelZ | What I'm wondering is whether or not it's * sending shitloads of qualifies for some reason, or if the device is registering over and over |
| 06:20.54 | ChannelZ | although I guess you'd see additional register messages.. hmm |
| 06:21.44 | *** join/#asterisk smooth_penguin (~smoove@59.95.56.60) |
| 06:23.22 | keith4__ | turned off qualify |
| 06:23.29 | keith4__ | let's see how this goes |
| 06:25.54 | keith4__ | what would cause asterisk to send qualifies over and over? |
| 06:26.15 | ChannelZ | ?? The timeouts are set in seconds. Is the clock on your * server going ape-shit crazy? |
| 06:26.16 | p3nguin | qualify being set to anything other than no. |
| 06:26.30 | keith4__ | qualify is ms |
| 06:26.51 | ChannelZ | well their responses are but I think in sip.conf it's all seconds |
| 06:27.05 | keith4__ | Channelz: clock's fine. using ntp, too |
| 06:27.06 | p3nguin | If qualify is set to a number or yes, it sends OPTIONS to the devices. |
| 06:27.44 | vandebo | ChannelZ: ms http://www.voip-info.org/wiki/view/Asterisk+sip+qualify |
| 06:28.06 | p3nguin | Yes, qualify is in ms. |
| 06:28.27 | tkrn | any ideas on the iaxmodem, Registration Timed Out? |
| 06:28.46 | p3nguin | Reconfigure it correctly. ? |
| 06:29.11 | keith4__ | okay. I just booted 20 phones, and they all registered correctly. |
| 06:29.15 | ChannelZ | vandebo: but that's the duration to consider whether it's considered down or not |
| 06:29.25 | keith4__ | no qualify spamming. yay! |
| 06:29.39 | keith4__ | what are the implications of not using 'qualify'? |
| 06:29.51 | p3nguin | No NAT keepalives. |
| 06:29.56 | ChannelZ | * won't know if a peer is truly available until it tries to send a call to it |
| 06:30.01 | p3nguin | Other than that, nothing. |
| 06:30.09 | keith4__ | okay, I can live with that |
| 06:30.37 | p3nguin | If the phones are not behind NAT, you probably don't need to qualify them anyway. |
| 06:30.51 | ChannelZ | something is whack with your system though. What version are you running and on what? |
| 06:30.52 | tkrn | yup tried a bunch of configs.. just reg is timing out... :-/ |
| 06:31.11 | keith4__ | ChannelZ: 1.4.21 on debian |
| 06:34.35 | ChannelZ | hmm |
| 06:36.55 | ChannelZ | https://issues.asterisk.org/view.php?id=15470 |
| 06:40.06 | *** join/#asterisk ecolitan (~aaron@li57-124.members.linode.com) |
| 06:40.49 | ChannelZ | keith4__: are you using hints perchance? |
| 06:41.02 | *** join/#asterisk gardo (~gardo@125.212.88.237) |
| 06:41.18 | keith4__ | ChannelZ: nope, not yet |
| 06:41.41 | keith4__ | but good find. this seems to be my issue |
| 06:42.12 | ChannelZ | well that one I pasted implied some incorrect network setup causing some issues |
| 06:42.32 | ChannelZ | but not quite the same because yours seems to be sending packets to the right place and getting a response |
| 06:43.28 | keith4__ | ah, right |
| 06:44.14 | ChannelZ | but I found another relating to dialplan hints |
| 06:44.22 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 06:44.34 | ChannelZ | with some other possible interactions.. but around your version. https://issues.asterisk.org/view.php?id=15950 |
| 06:44.56 | ChannelZ | Perhaps you are just running into something else bizarre, who knowz |
| 06:44.58 | ChannelZ | err knows |
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| 06:48.21 | keith4__ | ChannelZ: interesting. thanks for your help |
| 06:48.24 | keith4__ | p3nguin, too |
| 06:58.29 | vandebo | FYI, voip.ms's TOS are pretty sketchy |
| 06:59.08 | vandebo | definitely not written by a lawyer |
| 06:59.29 | ChannelZ | we call that "wiggle room" |
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| 07:13.27 | ChannelZ | whoa holy crap how did it get to be 1am |
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| 07:16.33 | sawgood | Just a question ... but does anyone run Asterisk 1.6 with no FreePBX or other GUI type front end? |
| 07:16.54 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:17.13 | sawgood | I was thinking of trying this tomorrow on a LAB box |
| 07:18.04 | sawgood | I was trying to 'put together' in my mind what is that FreePBX does for Asterisk which cannot be done through various scripts, etc. |
| 07:19.07 | kaldemar | sawgood: most people who run asterisk, do. |
| 07:19.38 | kaldemar | freepbx limits asterisk's features. |
| 07:19.42 | sawgood | kaldemar: I figured so ... but I wanted to learn both sides of things ... on two different systems (one with a GUI and one with out) |
| 07:20.02 | ChannelZ | sawgood: I do |
| 07:20.06 | sawgood | kaldemar: I've never heard that before .. amazing ... |
| 07:20.34 | sawgood | ChannelZ: everything you do is from the command line or maybe X forwarding ? |
| 07:21.04 | ChannelZ | command line. FreePBX is a web app GUI anyways |
| 07:21.04 | kaldemar | sawgood: you can only do certain pre-defined things with GUIs, but config file editing gives you much more. |
| 07:21.07 | ChannelZ | so no X needed |
| 07:22.12 | sawgood | So, if one was going to start with a hard drive of say only CentOS 5.4 .... what would be the min. RPM or yum packages I would have to add on top of this to have an Asterisk only box? |
| 07:22.31 | sawgood | I don't feel like working from src just yet ... rpm is fine for me |
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| 07:22.43 | sawgood | I like the thought of having updates take care of themselves |
| 07:23.39 | ChannelZ | then put on the * rpm and it will select any other crap it needs |
| 07:24.18 | sawgood | oh ... really ... cool .. just add say aserisk16 and asterisk16-core to my CentOS build ... and all depend. will come through? |
| 07:24.28 | sawgood | sorry to ask so basic of questions |
| 07:24.35 | ChannelZ | That's the idea anyway |
| 07:24.36 | kaldemar | sawgood: http://www.asterisk.org/downloads/yum |
| 07:24.45 | ChannelZ | I build from source so I can't tell you definitively it all works |
| 07:25.01 | kaldemar | you can use that repository to get more up-to-date packages. |
| 07:25.15 | sawgood | ChannelZ: nice work doing it all from src ... do you worry about manual udating? |
| 07:25.22 | ChannelZ | no |
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| 07:26.09 | sawgood | kaldemar: thank you for the URL ... explained everything! |
| 07:26.19 | ChannelZ | not for asterisk, I keep on top of any security patches but my asterisk box is 1. behind a firewall machine and 2. not accepting SIP/IAX from the outside world so it's fairly safe |
| 07:26.48 | sawgood | cool ... |
| 07:27.05 | sawgood | I was thinking of trying this out on a $200 dollar Atom 330 PC ... |
| 07:27.17 | sawgood | what do you think ... it does have 2GB of RAM |
| 07:30.02 | ChannelZ | shrugs |
| 07:30.16 | ChannelZ | As a test box with not much traffic it's probably fine |
| 07:30.48 | sawgood | cool ... thanks! |
| 07:33.12 | ChannelZ | sure good luck |
| 07:34.08 | sawgood | ChannelZ: do you use CentOS? |
| 07:35.37 | ChannelZ | no, Ubuntu mostly |
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| 07:55.17 | sawgood | usually ... what is the 'time' it takes for Digium to make yum updates work to upgrade Asterisk 1.6.0 to Asterisk 1.6.1, etc. |
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| 09:03.06 | Terminus | hello. question, is there a reference anywhere on what a database table should look like when using realtime configuration? the most i can find are column names in the docs, nothing about the data types that should be used. |
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| 09:13.50 | pisg | hi, i set up redirection, http://pastebin.com/zSfZFtqN and when someone call to number astersik redirect to my GSM number, but in my GSM phone i see numer asterisk, not numer caller |
| 09:16.56 | kaldemar | pisg: your telco decides what numbers you can use as caller id, and in this case it looks like they don't allow anything but what is associated with your line. |
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| 09:21.31 | pisg | someone have numer 123 and call to my asterisk numer 666, asterisk redirect to my GSM, and in my gsm i see 666 (asterisk number), can i see number who calls etc 123 |
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| 09:22.16 | Heretic | lo all |
| 09:22.43 | kaldemar | pisg: no |
| 09:23.12 | TommyBotten | pisg: That really depends on your provider. I have that "feature", allowing me to spoof callerids |
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| 09:26.59 | pisg | ok thx |
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| 10:25.08 | Karlitoo | hi all |
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| 11:02.02 | spenguin[work] | is there a skype for asterisk free trial |
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| 11:12.02 | *** join/#asterisk ik_5 (~ik@85.64.245.182.dynamic.barak-online.net) |
| 11:12.14 | ik_5 | hello |
| 11:13.05 | ik_5 | i'm having problems setting callerid on a dialplan it constantly forcing the original callerid rather accepting my settings |
| 11:14.05 | TommyBotten | Are you using some kind of framework/GUI on top of asterisk? |
| 11:15.01 | ik_5 | http://gist.github.com/333850 |
| 11:15.06 | ik_5 | I'm using freepbx |
| 11:15.12 | ik_5 | but my own dialplan |
| 11:16.32 | tzafrir | ik_5, that's an oxymoron |
| 11:16.48 | ik_5 | tzafrir, what do you mean ? |
| 11:16.57 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 11:17.06 | tzafrir | "FreePBX but my own dialplan" |
| 11:17.22 | TommyBotten | FreePBX contains the dialplan, and would overwrite your changes. |
| 11:17.33 | tzafrir | and yeah, there's a :-) there somewhere |
| 11:17.52 | tzafrir | TommyBotten, ik_5 knows FreePBX well enough |
| 11:18.06 | ik_5 | i'm adding it to my extensions_custom.conf |
| 11:19.50 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 11:20.42 | ik_5 | and another wierd issue is that logger.conf is set to give me on all outputs notice,warning,error,debug,verbose,dtmf but it does not provide me any debug information, only notice, warnings, errors and 'simple" output |
| 11:20.56 | ik_5 | on all outputs (full, debug, messages and console) |
| 11:21.17 | ik_5 | asterisk 1.4.30 |
| 11:23.01 | TommyBotten | Is the variable set in the FreePBX agi's ? ... |
| 11:23.14 | TommyBotten | Might be asking stupid questions here, as I haven't used freepbx in a very long time |
| 11:24.46 | ik_5 | TommyBotten, no, it's set on my dialplan |
| 11:25.09 | TommyBotten | well, yeah... but does the AGI set it as well? |
| 11:25.21 | TommyBotten | Meaning, it overwrites |
| 11:26.48 | ik_5 | TommyBotten, no, my code works after the freepbx's agi finish working and calling me |
| 11:27.45 | kaldemar | ik_5: you're setting it wrong |
| 11:28.03 | TommyBotten | ok |
| 11:28.05 | ik_5 | kaldemar how should i set it ? |
| 11:28.16 | kaldemar | Set(${CALLERID(all)}=1234567) won't work. Set(CALLERID(all)=1234567) will. |
| 11:30.26 | ik_5 | kaldemar, testing it, thanks |
| 11:32.58 | ik_5 | kaldemar, thanks, you are right |
| 11:33.02 | ik_5 | lammer |
| 11:35.00 | *** join/#asterisk aceio (~5d60a88a@gateway/web/freenode/x-ayopwgpebpaqrqyl) |
| 11:39.47 | ik_5 | now i need to figure out why the logger does not provide full information |
| 11:43.50 | kaldemar | ik_5: what does "logger show channels" say? |
| 11:45.14 | ik_5 | kaldemar, http://gist.github.com/333864 |
| 11:46.19 | ik_5 | but it does not provide me any information about function that executed beside normal dialplan display |
| 11:47.13 | kaldemar | be more specific. what is missing? |
| 11:47.56 | ik_5 | for example if i'm executing a function then to see what it returns and what actually set (for CALLERID for example) |
| 11:48.18 | ik_5 | it's a debug information as far as i know |
| 11:48.38 | leifmadsen | ik_5: define "full information" |
| 11:48.48 | kaldemar | ik_5: executing how? |
| 11:49.09 | leifmadsen | if you want to see what is set (for CALLERID() for example) output with Verbose() |
| 11:49.10 | ik_5 | a sec i'll give an example of a different machine with what i'm looking for |
| 11:49.14 | kaldemar | functions are not executed by themselves. |
| 11:49.18 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 11:49.27 | leifmadsen | Verbose(2,Value of CALLERID(num) is -- ${CALLERID(num)}) |
| 11:51.46 | ik_5 | http://gist.github.com/333867 |
| 11:52.04 | ik_5 | on the not working machine, i see only "verbose" but no "debug" |
| 11:54.14 | leifmadsen | ik_5: core set debug 5 |
| 11:54.26 | leifmadsen | ik_5: you need to enable it from the CLI |
| 11:54.28 | *** join/#asterisk niekvlessert (~niek@92.70.112.34) |
| 11:54.30 | niekvlessert | hello! |
| 11:54.40 | niekvlessert | i want to call pickup a ringing extension |
| 11:54.51 | niekvlessert | but this extension is not connected to a phone |
| 11:54.57 | niekvlessert | how would i do this? |
| 11:55.16 | ik_5 | leifmadsen, it does not apper also on full and debug logs |
| 11:55.17 | niekvlessert | or how would i search this on google? :) |
| 11:55.25 | leifmadsen | ik_5: then you have something incorrectly configured |
| 11:55.38 | leifmadsen | niekvlessert: what do you mean "not connected to a phone" ? |
| 11:55.51 | leifmadsen | I doubt asterisk cares about that -- call pickup a ringing extension should work the same regardless |
| 11:56.02 | leifmadsen | and that is my time... rebooting into documentation mode |
| 11:57.36 | ik_5 | my logger.conf settings are: http://gist.github.com/333868 |
| 12:03.44 | *** join/#asterisk aandrade (~aandrade@187.58.239.123) |
| 12:04.25 | pisg | how i can stop flood in CLI this : [Mar 16 12:58:19] NOTICE[1949]: chan_sip.c:18223 handle_response_peerpoke: Peer '402' is now Reachable. (81ms / 2000ms) ? |
| 12:04.48 | LemensTS | pisg: /etc/asterisk/logger.con |
| 12:04.55 | LemensTS | turn off notice in console in that file |
| 12:05.12 | pisg | this is only options ? |
| 12:06.31 | LemensTS | pisg: 402 is apparently having network problems or the server is. |
| 12:07.19 | TommyBotten | Or there are two clients with the same credentials registering |
| 12:07.25 | pisg | 402 is in local network 192.168.1.0/24 |
| 12:07.54 | LemensTS | pisg: is 402 the only one doing that? I was assuming you had lots of phones doing it |
| 12:08.07 | pisg | all phones doing that |
| 12:08.18 | pisg | and i have next flood : [Mar 16 13:03:05] WARNING[1949]: chan_sip.c:23861 build_peer: Invalid peer port configuration at line 0 : 0, |
| 12:08.28 | pisg | hmm i have peers in realtime |
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| 12:20.02 | J4zen | Hi guys, i'm looking for a proper VoIP DECT phone other than the SNOM M3/M9 systems. Any suggestions? |
| 12:20.20 | Chainsaw | J4zen: Siemens has a whole range of them. |
| 12:20.33 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 12:20.43 | J4zen | you're refering to the Gigaset range? |
| 12:20.58 | Chainsaw | J4zen: To phones like the C460 IP, yes. |
| 12:21.25 | J4zen | Have you tried any? |
| 12:21.43 | Chainsaw | J4zen: Yes, we use them for staff. (Working from home, etc) |
| 12:22.37 | *** join/#asterisk shader (~user@janustw.tavve.com) |
| 12:22.48 | J4zen | How is the docking station for those phones, is it stable? My client is complaining that the M3/M9 docking stations are quite unstable. |
| 12:22.57 | andrebarbosa | Anyone has the problem with attendend transfer using BRIA pro softphone? |
| 12:23.02 | Chainsaw | J4zen: Electrically stable? Mechanically stable? Please be specific. |
| 12:23.55 | J4zen | Chainsaw: Sorry, i was refering to the actual stability of the phone when placed in the docking station. ; Will it fall over when you bump into the desk? |
| 12:24.27 | Chainsaw | J4zen: It seems okay here. |
| 12:24.30 | J4zen | Chainsaw: I don't know if you've tried the M3/M9's from SNOM, but they tend to fall over very easely if you bump it a little. |
| 12:24.44 | Chainsaw | J4zen: We have them on metal shelves that the engineers move to place their laptop on. |
| 12:24.49 | Chainsaw | J4zen: That action does not make the phone fall off. |
| 12:25.02 | Chainsaw | (It's a sliding shelf) |
| 12:25.10 | J4zen | Chainsaw: Alright, i'll order a sample and try it out. Thank you Chainsaw. |
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| 12:37.40 | Gurizim | bom dia |
| 12:39.38 | Gurizim | alguem sabe como configurar uma operadora automatico no freepbx? |
| 12:40.17 | ManxPower-work | ~freePBX |
| 12:40.17 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:53.47 | *** join/#asterisk arossouw (~arossouw@dsl-146-50-11.telkomadsl.co.za) |
| 12:54.19 | arossouw | hi, how can i configure asterisk, to only pickup inbound calls, currently it also picks up outbound calls |
| 12:56.04 | ManxPower-work | arossouw, Asterisk does not pick up any calls unless you configure it to do so. |
| 12:56.31 | arossouw | i configured it to pick up calls, with *8 |
| 12:56.33 | ManxPower-work | arossouw, you are not using a GUI like FreePBX, are you? |
| 12:56.40 | arossouw | nope, file based asterisk |
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| 12:56.50 | *** part/#asterisk Gurizim (~IceChat7@200-232-166-122.dsl.telesp.net.br) |
| 12:57.01 | ManxPower-work | arossouw, Hopefully you configured *8 and the callgroup= and pickupgroup= items as well |
| 12:57.05 | arossouw | enabled *8 in /etc/asterisk/features.conf |
| 12:57.13 | arossouw | ManxPower-work: yes in sip.conf |
| 12:57.27 | arossouw | and zapata.conf |
| 12:57.47 | ManxPower-work | remove it from zapata.conf |
| 12:58.07 | *** join/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 12:58.11 | arossouw | ah, that easy, ok |
| 12:58.19 | kazaa_lite | hi all |
| 12:58.35 | kazaa_lite | has anyone used Grandstream GXP1200 IP Phone? any feedback about it would be appriciated? |
| 12:58.44 | ManxPower-work | ~gs |
| 12:58.45 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 12:59.18 | arossouw | granstream is useless, trust me if you wanna go cheap rather by digital to analog converter |
| 13:00.40 | kazaa_lite | what are the recommened phones? |
| 13:01.30 | arossouw | Polycom |
| 13:01.34 | TommyBotten | I like Aastra a lot. Snom and polycom also come highly recommended. |
| 13:01.38 | arossouw | their rather expensive |
| 13:01.51 | arossouw | personally dont like snom, they keep losing their passwords |
| 13:01.52 | *** join/#asterisk voipmonk (~shido6@67.204.1.83) |
| 13:02.00 | TommyBotten | really? |
| 13:02.13 | TommyBotten | I haven't had that issue at all |
| 13:02.32 | arossouw | ive had with snom 6.5.13 -> 6.5.18 snom 300 phones |
| 13:02.41 | arossouw | maybe cause you use power of ethernet |
| 13:03.38 | arossouw | TommyBotten: i regularly have to log into the pabx and reset the passwords for extensions |
| 13:03.44 | ManxPower-work | Linksys seems to be known as a low end phone that works |
| 13:04.55 | [TK]D-Fender | arossouw: Given where you are Linksys is probably a decent choice |
| 13:05.06 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 13:05.16 | *** join/#asterisk adeel (~adeeln@c-67-174-36-109.hsd1.ca.comcast.net) |
| 13:05.31 | arossouw | [TK]D-Fender: same price range as Snom ? |
| 13:06.08 | [TK]D-Fender | arossouw: Don't know for sure in your area. Go look |
| 13:06.18 | arossouw | [TK]D-Fender: saving costs is very important for company i work for, so far ITA adapters is the cheapest choice |
| 13:06.37 | ManxPower-work | arossouw, Then Asterisk may not be the correct solution for you. |
| 13:07.02 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 13:07.06 | *** join/#asterisk nightrid3r (kvirc@41.214.128.56) |
| 13:07.20 | Katty | drags in |
| 13:08.10 | arossouw | ManxPower-work: i agree, if you want to go asterisk you need to at least by decent equipment |
| 13:08.20 | Katty | it's johnny gray's birthday! |
| 13:08.23 | arossouw | s/by/buy |
| 13:08.24 | *** join/#asterisk boch (~fran@200.61.191.9) |
| 13:08.44 | adeel | between polycom/aastra/snom, which boasts a better featureset/bang-for-the-buck? i've been using polycom phones for the last few years and was wondering how they compare |
| 13:08.50 | ManxPower-work | arossouw, Asterisk may save you money in the long term, but it does not always do so in the short term. |
| 13:09.01 | ManxPower-work | ~phones |
| 13:09.02 | infobot | 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, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 13:09.13 | boch | hello, is it possible to play DTMF to calling party like D() option to Dial() app does? |
| 13:09.25 | adeel | ManxPower-work, thanks |
| 13:09.38 | ManxPower-work | boch, yes. SendDTMF before the Dial |
| 13:09.48 | *** join/#asterisk nightrid3r (kvirc@41.214.128.56) |
| 13:09.57 | arossouw | dont like Wildcard TDM Analog cards |
| 13:10.04 | ManxPower-work | nightrid3r, next time use pastebin.ca instead of flooding the channel |
| 13:10.48 | Katty | i don't like none of dem dere wildcard tdm cards |
| 13:10.50 | nightrid3r | ? |
| 13:10.55 | Katty | ^- arossouw |
| 13:11.17 | ManxPower-work | * nightrid3r has quit (Excess Flood) |
| 13:11.17 | shader | my office currently has an old pbx system that loses caller id information/incoming phone number when the caller uses the name directory. Can asterisk preserve that information? |
| 13:11.21 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 13:11.42 | ManxPower-work | shader, that is really a phone thing, not an Asterisk thing |
| 13:11.45 | boch | ManxPower-work, but i want the channels to be linked immediately after DTMF, like D() option does. If I use SendDTMF i am not sure if call will succeed |
| 13:11.55 | nightrid3r | i always have that when i connect 1st time, then it reconnects and all goes well, dunno why |
| 13:12.04 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 13:12.08 | [TK]D-Fender | shader: Preserve what? INformation on another PBX? |
| 13:12.25 | shader | [TK]D-Fender: caller id information |
| 13:12.34 | [TK]D-Fender | shader: Why would * LOSE information? |
| 13:12.42 | shader | lol |
| 13:12.48 | shader | I don't know, but our current system does |
| 13:13.07 | shader | though ManxPower-work says it might be because of the phones we're using |
| 13:13.11 | Katty | shader: that is most unfortunate. |
| 13:13.11 | arossouw | dont like troubleshooting voip issues |
| 13:13.27 | Katty | shader: you should bring your company up to date in the real world. |
| 13:13.34 | [TK]D-Fender | shader: Its the system. |
| 13:13.38 | ManxPower-work | shader, oh, I don't know if it's a phone issue on your random, undisclosed PBX brand/model |
| 13:13.38 | shader | Katty: that's what I'm working on |
| 13:13.45 | Katty | shader: excellent |
| 13:13.49 | shader | ManxPower-work: lol |
| 13:13.55 | ManxPower-work | But if you were using Asterisk, then it's a phone issue |
| 13:14.00 | shader | it's a Siemens |
| 13:14.07 | shader | of unknown age |
| 13:14.16 | ManxPower-work | Just remember Asterisk isn't really a PBX. |
| 13:14.46 | ManxPower-work | Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 13:14.47 | Katty | yeah it's really a coffee machine |
| 13:15.31 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 13:16.00 | [TK]D-Fender | No, it's a JUKEBOX |
| 13:16.04 | [TK]D-Fender | spins up the hits |
| 13:17.39 | *** join/#asterisk codebanshee (~codebansh@81.171.245.107) |
| 13:19.10 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.87.110) |
| 13:20.26 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 13:26.18 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 13:26.40 | shader | do you have to restart asterisk when you update conf files? |
| 13:27.08 | [TK]D-Fender | shader: Depends which |
| 13:27.27 | smooth_p[work] | hey Katty :> |
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| 13:31.00 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 13:32.36 | arossouw | shader: normally asterisk -rx "reload" works |
| 13:32.43 | shader | ok |
| 13:32.58 | arossouw | shader: if you change zapata and modules , i think you need to restart then |
| 13:33.29 | shader | what about errors in conf files? |
| 13:34.00 | Katty | hi penguin :> |
| 13:34.56 | arossouw | i think thats what /var/log/asterisk/messages are for |
| 13:38.10 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:38.20 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:38.38 | kazaa_lite | polycom does not mention some decent range of codecs its phones support |
| 13:38.57 | tzafrir | arossouw, 'dahdi restart' also works, if it exists (asterisk >= 1.4.22) |
| 13:38.59 | [TK]D-Fender | kazaa_lite: Polycom lists EXACTLY what they support |
| 13:39.06 | kazaa_lite | cool |
| 13:39.13 | Katty | chugs some caffeine |
| 13:39.22 | arossouw | tzafrir: noted :-) |
| 13:39.25 | kazaa_lite | then snom phones seem to be much better interms of features |
| 13:39.49 | arossouw | is it possible to monitor isdn lines, like if there is an answer on the line? |
| 13:40.01 | Katty | you can Monitor() whatever you want |
| 13:40.05 | arossouw | thinking of using nagios |
| 13:40.12 | Katty | or mixmonitor() if that's your thing |
| 13:40.35 | Katty | and you can do anything you want with it afterwards with bash |
| 13:41.35 | arossouw | cool, but not allowed to record stuff, legal reasons |
| 13:41.54 | [TK]D-Fender | [09:39]<kazaa_lite>then snom phones seem to be much better interms of features <- like? |
| 13:42.02 | arossouw | thinkin of a tool that can call the isdn numbers and see whether is a response |
| 13:42.14 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 13:42.20 | [TK]D-Fender | arossouw: Considered trying Asterisk? |
| 13:42.22 | Katty | hi benngard |
| 13:42.39 | benngard | hi katty |
| 13:42.41 | kazaa_lite | i was just going to through their websites and seem snom mentions a lot of things supported |
| 13:42.44 | asteriskATmarmuD | arossouw: we got such a tool working with the AVM fritz card and CAPI |
| 13:42.54 | Katty | what's a nice catchy oldish song i can listen to |
| 13:42.56 | kazaa_lite | attractive thing was support of codecs........ |
| 13:43.02 | [TK]D-Fender | kazaa_lite: Sorry, could you be a little more vague? |
| 13:43.22 | Katty | i'm thinking 60s |
| 13:43.27 | kazaa_lite | snom has larger number of audio formats supported than polycom |
| 13:43.33 | [TK]D-Fender | kazaa_lite: And which codec that they support do you actually care about? |
| 13:43.49 | Katty | ooh here we go: only the lonely |
| 13:43.49 | benngard | as long as i have g711 i am satisfied |
| 13:43.55 | kazaa_lite | nothing special... but more number of codecs |
| 13:44.18 | Katty | i have a snom at home upstairs in the bedroom |
| 13:44.22 | Katty | and a polycom downstairs in the basement |
| 13:44.33 | kazaa_lite | [TK]D-Fender: are you doing some survey on customer feedback?:P |
| 13:44.33 | arossouw | lol [TK]D-Fender vague answer |
| 13:44.46 | Katty | i know what my preference is |
| 13:45.01 | Katty | but that obviously isn't going to the be same for everyone |
| 13:45.21 | arossouw | [TK]D-Fender: you should see these ppl i work for, vague is understatement |
| 13:45.24 | kazaa_lite | [TK]D-Fender: ?? |
| 13:45.34 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:45.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:45.40 | Katty | hi leif |
| 13:45.41 | [TK]D-Fender | kazaa_lite: No, just keeping things in perspective |
| 13:45.52 | leifmadsen | yo! |
| 13:46.01 | kazaa_lite | [TK]D-Fender: cool |
| 13:46.07 | [TK]D-Fender | kazaa_lite: Adding a mitllion things you'll never use doesn't necessarily add "valuie" |
| 13:46.12 | [TK]D-Fender | value* |
| 13:46.40 | ManxPower-work | To everyone that thinks running a hybrid Asterisk/Existing PBX type of setuo: "The most dangerous strategy is to jump a chasm in two leaps." --Benjamin Disraeli |
| 13:46.43 | Katty | ohhh my love.........my darling!!! |
| 13:46.44 | kazaa_lite | [TK]D-Fender: but i might like to try things... and then if my purchase is limited.... then i made a bad choice |
| 13:47.12 | Katty | goodness |
| 13:47.19 | kazaa_lite | [TK]D-Fender: so better i buy something which has good feedback and more things in it.... |
| 13:47.22 | Katty | i have a whole herd in my front yard |
| 13:47.29 | Chainsaw | leifmadsen: Could you update the topic with the new versions please? |
| 13:47.37 | leifmadsen | Chainsaw: for you? never |
| 13:47.49 | Katty | would you do it for a scooby snack? |
| 13:47.49 | Chainsaw | leifmadsen: I'd do it myself, but it's mode +t. |
| 13:48.10 | [TK]D-Fender | kazaa_lite: Polycom wins on quality pretty much hands-down. They are also rather strong on features, but are beat in certain areas by different makes & models. The problem is the trade-off is rarely worth it |
| 13:48.23 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 13:48.29 | leifmadsen | done! |
| 13:48.29 | Katty | kazaa_lite: get one of each. try them out |
| 13:48.49 | Chainsaw | leifmadsen: Excellent, thanks :) |
| 13:49.02 | leifmadsen | Chainsaw: thanks for the reminder -- it's normally part of the release process |
| 13:49.08 | dddh | should I use kall8? |
| 13:49.14 | kazaa_lite | Katty: dont have enough money to buy a phone just to compare the difference:P may be sometime later when i am rich i could do that:P |
| 13:49.24 | Chainsaw | leifmadsen: Yeah, I always look at the topic here to see whether I'm up to date. So now I was late :) |
| 13:49.31 | Katty | kazaa_lite: that's no excuse |
| 13:49.37 | Katty | kazaa_lite: return the one you don't like for a full refund |
| 13:49.48 | leifmadsen | heh |
| 13:49.52 | kazaa_lite | Katty: that seems bad:P |
| 13:50.00 | Katty | no, that's why they made return policies |
| 13:50.03 | leifmadsen | asterisk.org is probably the best place to check for the latest versions and release announcement |
| 13:50.15 | Katty | in other mews. |
| 13:50.25 | Katty | sargent general has to decided to take up residence INSIDE the feeder. |
| 13:50.44 | Katty | rather than sitting on the little stand and poping the lid up, he's just decided to crawl inside the feeder and park it |
| 13:51.02 | leifmadsen | Katty: http://www.dailyhaha.com/_vids/super_troopers_meow.htm |
| 13:51.34 | Katty | :> |
| 13:51.37 | Katty | i've seen that :>> |
| 13:51.44 | aurs | have anyone here tried the webenabled = yes / http.conf trick with the manager api? I'm using it, but the xml isn't always correct.. missing a few lines etc. anyone else seen that? |
| 13:53.41 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 13:54.43 | Katty | hmmm, visiting female downy woodpecker (= |
| 13:55.52 | Katty | Kobaz: that molded in area that i has grass growing in it right now. if i spray the area with roundup, how long should i wait before planting the blueberry bush? |
| 13:56.06 | Katty | s/has/have has/ |
| 13:56.51 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:58.52 | aurs | sometimes, all I get from the SIPPeers command is this: <response type="object" id="unknown"><generic event="PeerlistComplete" listitems="2625"/></response> |
| 13:59.01 | aurs | so no PeerEntry lines |
| 13:59.19 | aurs | or I might get some PeerEntry lines, but not all of them |
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| 14:04.51 | ddickenson | Hello all, I have quite a bit of asterisk installs at work all of which are using dahdi connections to the pstn. I would like, however, at home for my side business to have some sort of either sip/iax2 trunk coming in that will give me one DID and a couple of incoming connections for as cheap as possible. There will not be much traffic on this circuit but would like at least 2-3 concurrent calls |
| 14:05.16 | ddickenson | anyone have any favorite vendors? |
| 14:05.21 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 14:05.33 | niekvlessert | question: i would like to have a fake sip phone ringing |
| 14:05.45 | niekvlessert | seems very simple, but i have no idea how to do it |
| 14:05.47 | niekvlessert | :) |
| 14:06.04 | *** join/#asterisk Jhirley (~Jhirley@mail.mmdlaw.com) |
| 14:06.23 | niekvlessert | i want a blf from a number which is answered and can be pickedup by a blf key on other phones |
| 14:06.44 | *** join/#asterisk beek_ (~klinebl@pdpc/supporter/bronze/beek) |
| 14:06.58 | shader | does anyone know the pricing for Skyp-for-Asterisk> |
| 14:06.59 | shader | ? |
| 14:07.37 | Jhirley | 0/ Peeps |
| 14:07.44 | Katty | hi beek |
| 14:07.46 | [TK]D-Fender | niekvlessert: BLF != "pickup" |
| 14:07.56 | beek_ | morning Katty |
| 14:08.08 | [TK]D-Fender | niekvlessert: And there is no "fake SIP". You can dial a LOCAL CHANNEL that won't actually get answered if you like |
| 14:08.12 | niekvlessert | [TK]D-Fender: i know, blf is working fine, pickup is the problem |
| 14:08.25 | [TK]D-Fender | shader: www.digium.com |
| 14:08.27 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:08.43 | niekvlessert | [TK]D-Fender: ah! how would it do that? dial(local/<which number>,20)? |
| 14:08.57 | niekvlessert | but that local is no phone |
| 14:08.59 | [TK]D-Fender | niekvlessert: Local channels = call into dialplan |
| 14:09.01 | niekvlessert | that's no problem? |
| 14:09.19 | niekvlessert | [TK]D-Fender: ok |
| 14:09.24 | [TK]D-Fender | niekvlessert: Local/exten@context |
| 14:09.26 | Jhirley | Katty: its almost easter, you know what that mean, its time to bit the heads off of little yelllow chickens ! |
| 14:09.42 | [TK]D-Fender | niekvlessert: so make an exten that will wait around and not answer |
| 14:10.08 | niekvlessert | ok i believe i'm getting it 50%, lemme try something |
| 14:10.09 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:10.12 | niekvlessert | [TK]D-Fender: thanks already |
| 14:10.59 | *** join/#asterisk sourcode (~code@ppp-58-8-117-48.revip2.asianet.co.th) |
| 14:13.50 | Katty | Jhirley: i do not like crunchy chick |
| 14:14.39 | Jhirley | you gotta get them fresh, so they are soft ? |
| 14:20.40 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:22.00 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 14:24.12 | niekvlessert | [TK]D-Fender: how do we call pickup a local channel? it won't work with local/999@loggedin & 999/loggedin |
| 14:24.47 | [TK]D-Fender | niekvlessert: How are you picking up calls currently? |
| 14:24.49 | niekvlessert | [TK]D-Fender: local show channels shows the channel fine like this |
| 14:26.02 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:3c09:49ec:2ac0:762d) |
| 14:26.30 | niekvlessert | [TK]D-Fender: exten => _**.,n,Pickup(${EXTEN:2}@loggedin) |
| 14:26.42 | niekvlessert | **999 is send just fine |
| 14:27.09 | ManxPower-work | maybe Local/(${EXTEN:2}@loggedin) ? |
| 14:27.18 | [TK]D-Fender | niekvlessert: pastebin a complete sample with configs |
| 14:27.19 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 14:27.24 | niekvlessert | ok hold on |
| 14:27.28 | a1fa | grr... my sexterix is not doing any csv logging |
| 14:27.45 | a1fa | it is enabled in cdr.conf, and the module is loaded |
| 14:27.50 | a1fa | anything else that needs to happen? |
| 14:30.02 | *** join/#asterisk generalhan (~asd@about/windows/staff/generalhan) |
| 14:30.46 | cusco | hi |
| 14:31.07 | cusco | with realtime configuration, a queue has strategy random |
| 14:31.17 | cusco | but the call seems to go always to the same operator... |
| 14:33.09 | Katty | onlyyyy youuuu |
| 14:33.16 | Katty | can make all this world seem rightttttt |
| 14:33.31 | Katty | smooth_p[work]: only youu can make the darkness brightttttt |
| 14:33.42 | shader | what sip trunk service do y'all use? |
| 14:33.58 | Katty | infobot: itsp-list? |
| 14:33.59 | infobot | [itsp-list] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 14:34.04 | ManxPower-work | ~siptrunk |
| 14:34.05 | infobot | siptrunk is, like, something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 14:34.09 | smooth_p[work] | HEY Katty |
| 14:34.14 | smooth_p[work] | oops caps |
| 14:34.14 | Katty | smooth_p[work]: herroes. |
| 14:34.17 | Katty | YEAH |
| 14:34.20 | Katty | SHAME ON YOUR CAPS |
| 14:34.30 | smooth_p[work] | :S |
| 14:34.41 | smooth_p[work] | shines some light in the channel |
| 14:34.47 | Katty | cheers |
| 14:34.53 | Katty | smooth_p[work]: have you seen crittercam this morning? |
| 14:34.56 | smooth_p[work] | nope |
| 14:35.02 | Katty | smooth_p[work]: there are some great antics goin on |
| 14:35.07 | smooth_p[work] | yeah? |
| 14:35.09 | Katty | yep |
| 14:35.21 | smooth_p[work] | like? |
| 14:35.35 | smooth_p[work] | or wait its still on? |
| 14:35.40 | Katty | well i put up a new bird feeder, and the squirrels keep inching out on this ITTY BITTY branch to try and get to it |
| 14:35.46 | Katty | or climb up the pole to reach it, and slide back down |
| 14:35.50 | Katty | yep it's still on |
| 14:36.11 | smooth_p[work] | haha |
| 14:36.21 | shader | ManxPower-work: is that impleying that you can't get sip trunks to work with * ? |
| 14:36.31 | Katty | smooth_p[work]: http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 14:36.45 | Katty | smooth_p[work]: i also have one very hungry downy woodpecker |
| 14:36.56 | smooth_p[work] | oooh nice |
| 14:37.10 | Katty | you know the feeders on the tree? |
| 14:37.18 | Katty | it's a little box with a lid the squirrels push up |
| 14:37.26 | smooth_p[work] | ah yeah |
| 14:37.27 | Katty | well one of them was sitting /inside/ the feeder eating this morning |
| 14:37.37 | Katty | actually, he's still in there |
| 14:37.39 | Katty | on the left tree |
| 14:38.09 | Katty | rather than sitting on the platform thing and pushing the lid up |
| 14:38.40 | rttrey | I assume the new bird feeder is the one in the middle there. i see the squirrels tryin to climb up the pole and slide back down |
| 14:38.57 | Katty | rttrey: yeah they also inch out on that little branch above it |
| 14:39.03 | Katty | rttrey: suprisingly they're making it up there |
| 14:39.17 | Katty | rttrey: not that i mind or anything |
| 14:39.18 | beek | loads up the URL to the crittercam |
| 14:39.30 | Katty | beek: http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 14:39.51 | rttrey | Katty: cool setup you have. looks like a squirrel managed to stay up |
| 14:40.04 | Katty | rttrey: they're smart little critters. |
| 14:40.09 | Katty | rttrey: and very persistent |
| 14:40.23 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 14:42.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:42.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:43.14 | Katty | had 7 in the yard saturday morning |
| 14:43.52 | smooth_p[work] | they need to learn some tricks |
| 14:43.53 | smooth_p[work] | :p |
| 14:44.03 | *** join/#asterisk mhilmi (~no@modemcable107.95-22-96.mc.videotron.ca) |
| 14:44.13 | Katty | you kiddin me? |
| 14:44.18 | Katty | they're entertaining as it is |
| 14:44.35 | Katty | i had one little guy on the feeder saturday who was sittin there eatin, and another one droped down from the branch. |
| 14:44.40 | Katty | they both got spooked and went flyin |
| 14:45.32 | a1fa | infobot csv? |
| 14:45.33 | infobot | methinks csv is comma separated values |
| 14:46.00 | a1fa | no shit sherlock |
| 14:46.07 | a1fa | has anyone have problems with * not logging ? |
| 14:46.12 | Katty | nope |
| 14:46.37 | Katty | well there was that one time i forgot to enable in cdrconf |
| 14:46.46 | Katty | but that wasn't an asterisk problem, that was a me forgetting problem |
| 14:47.15 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:47.19 | a1fa | its should be enabled by default |
| 14:47.29 | a1fa | strangely, i removed the file, and restarted * and its working now |
| 14:47.35 | a1fa | i wonder how it got disabled in the first place |
| 14:48.52 | mhilmi | Our asterisk box is connected to the POTS via a punchbox. Does anyone know of or seen any device that would allow you to switch between connections on the punchbox easily? IE when asterisk goes down, flick a switch and old PBX gets reconnected. |
| 14:49.10 | mhilmi | er punchbox = punch block |
| 14:51.57 | *** part/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 14:53.52 | ManxPower-work | shader, no, it means that saying "sip trunk" makes as much sense as saying "sip easter bunny". Stop using the term. |
| 14:54.06 | ManxPower-work | They are "peers" |
| 14:56.37 | shader | ManxPower-work: ah |
| 14:57.04 | shader | well, it's hard to stop using the term, given that most businesses seem to sell sip peerage under the name "sip trunk" |
| 14:57.18 | shader | is there a better search term I could use? |
| 14:57.21 | Katty | it's the sales & marketing folk who invited the term |
| 14:57.37 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:57.41 | Katty | i think it's the only way they could wrap their brain around the concept |
| 14:58.08 | shader | yep, especially since the standard sales model seems to be paying for number of simultaneous connections |
| 14:58.11 | shader | i.e. "trunks" |
| 14:58.28 | shader | so, since they're using sip, they must be sip trunks, right? |
| 14:58.43 | *** join/#asterisk mazpe (~mazpe@ec2-174-129-37-13.compute-1.amazonaws.com) |
| 14:59.07 | Katty | i wonder what kind of stuff i say about my car that is just absolutely riddiculus |
| 14:59.16 | ManxPower-work | shader, no. a trunk is a dedicated circuit. SIP does not have dedicated circuits. Marketing can call it anything they want, that does not make it true. They are SIP peers. |
| 14:59.18 | mazpe | anyone knows of a guide or suggestion to create a vlan and QoS for VoIP? |
| 14:59.20 | Katty | andi don't even know it ;) |
| 14:59.49 | ManxPower-work | mazpe, do you know how to do QoS and VLAN for non-VoIP? |
| 15:00.15 | ManxPower-work | Learn that first. 8-| |
| 15:00.17 | mazpe | no |
| 15:00.21 | shader | Katty: probably not that much, given that car companies aren't trying to sell you very many pieces individually |
| 15:00.29 | niekvlessert | http://pastebin.com/f2EH4yDk |
| 15:00.35 | shader | and they don't have to come up with odd service pricing plans |
| 15:01.07 | Katty | erm |
| 15:01.21 | [TK]D-Fender | niekvlessert: "core show application pickup" |
| 15:01.26 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:02.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:02.56 | Katty | wondderrrboyyyy |
| 15:03.01 | Katty | what is the secret of your powerrrrrr |
| 15:03.04 | thehar | lol |
| 15:03.56 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 15:15.47 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 15:16.17 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 15:16.54 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 15:19.13 | niekvlessert | [TK]D-Fender: what do you mean? |
| 15:19.32 | niekvlessert | group problem? |
| 15:19.49 | niekvlessert | 999 is a member of loggedin |
| 15:20.09 | niekvlessert | not a member... part of |
| 15:20.17 | niekvlessert | how do we get local in a group?? |
| 15:20.44 | *** join/#asterisk Geminizer (~ryan@cpe-76-180-27-4.buffalo.res.rr.com) |
| 15:20.52 | niekvlessert | local is not a user... |
| 15:21.09 | [TK]D-Fender | niekvlessert: PickupExten <- |
| 15:21.21 | niekvlessert | ok |
| 15:22.39 | niekvlessert | that's not an application |
| 15:22.55 | rubberneck | I am running Asterisk 1.6.1.9, how can I determine if I can use the MixMonitor application to record in MP3 format? |
| 15:23.03 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:23.04 | wcselby | o/ |
| 15:23.27 | Katty | :>>>>>>>>>>>>> |
| 15:23.29 | *** join/#asterisk dzup (dzup@unaffiliated/dzup) |
| 15:23.31 | Katty | hugs wcselby |
| 15:23.44 | wcselby | :) |
| 15:24.16 | Geminizer | Hello all. In a dialplan context in which the user is prompted to enter a number between 1 and 100, it would be preferable to not have the user enter 3 digits for any possible number entered (e.g. 001, 010, 100)... is there way to construct a dialplan such that entering 1, 10, or 100 is acceptable ? |
| 15:24.39 | wcselby | digit timeout |
| 15:25.06 | wcselby | or, perhaps the proper way |
| 15:25.09 | Geminizer | perhaps follow the number entered with a # sign to indicate that number entry is over, but what pattern would that be? |
| 15:25.20 | wcselby | Geminizer - Set(TIMEOUT(digit)=2) |
| 15:25.36 | niekvlessert | [TK]D-Fender: that's not an application |
| 15:25.51 | wcselby | gives the person 2 seconds to enter digits, then goes on |
| 15:26.19 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:26.51 | Geminizer | wcselby: so have extensions matches _X, _XX, _XXX, etc ? |
| 15:27.59 | wcselby | Geminizer - that may work, depending on how you do things. |
| 15:28.10 | wcselby | are you planning on having a 100 option IVR or something? |
| 15:28.57 | kaldemar | Geminizer: core show application Read |
| 15:29.00 | *** join/#asterisk mog (~mog@c-71-228-185-24.hsd1.al.comcast.net) |
| 15:29.30 | Geminizer | well, more so a number to accept how many seconds they want for making a call, where the number can be anything from 100 down to 1 |
| 15:30.28 | niekvlessert | [TK]D-Fender: plz help, willing to pay :) |
| 15:30.40 | wcselby | Geminizer - look at Read() |
| 15:30.43 | wcselby | core show application Read |
| 15:30.52 | wcselby | would probably be better for what you're wanting to do |
| 15:31.20 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 15:31.33 | Geminizer | ok, I will look into that... thanks guys |
| 15:32.07 | wcselby | niekvlessert - what are you trying to do? |
| 15:32.42 | LemensTS | Anyone know of where i can find scripts if phones cant callout/in or register? |
| 15:33.13 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 15:33.44 | Nugget | OK, what the flip is a Digium T10i? |
| 15:34.12 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:34.51 | Katty | telnet |
| 15:34.54 | Katty | :< |
| 15:36.02 | Kobaz | it's one step up from the T9i |
| 15:37.42 | Katty | oh |
| 15:37.43 | Katty | you |
| 15:37.46 | Katty | i was looking for you earlier |
| 15:38.11 | Nugget | hides |
| 15:38.12 | Katty | hoping to get some more advice from the planty expert |
| 15:38.25 | Katty | hugs Nugget |
| 15:38.29 | Katty | puts Nugget back under his rock |
| 15:38.48 | smooth_p[work] | the nugget will rott there |
| 15:38.50 | smooth_p[work] | :p |
| 15:39.05 | Katty | don't under estimate the power of the Nugget |
| 15:39.19 | niekvlessert | wcselby: i want to make it possible to pickup a blinking blf by pressing that button, but that extension ringing is not an actual phone, it can be everything, but blf & call pickup have to possible |
| 15:39.37 | smooth_p[work] | kk |
| 15:39.48 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 15:41.43 | wcselby | niekvlessert - I think that really depends on the model phone you have |
| 15:41.59 | wcselby | niekvlessert - what did [TK]D-Fender suggest you lookup? |
| 15:42.47 | *** join/#asterisk aceio (~c2cbd7fe@gateway/web/freenode/x-lluyyycrnagpbuhw) |
| 15:43.59 | niekvlessert | use local channel |
| 15:44.09 | niekvlessert | but we can't do call pickup on a local channel up to now |
| 15:44.22 | niekvlessert | pickupexten he said, but that's not a function |
| 15:44.25 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 15:46.52 | [TK]D-Fender | niekvlessert: Your pastebin is very incomplete. We don't see the rest of the pertinent dialplan, nor the status dump and dialplan execution prior to the pickup attempt |
| 15:47.40 | niekvlessert | ok, last try |
| 15:47.56 | niekvlessert | if this won't work we put a phone in the closet with ringtone off ;) |
| 15:53.40 | wcselby | niekvlessert - what type of phones do you have? |
| 15:54.05 | niekvlessert | aastra + snom |
| 15:54.11 | niekvlessert | but we want only 1 line on a phone |
| 15:54.27 | wcselby | what version of asterisk are you using? |
| 15:54.31 | niekvlessert | 1.4.26.2 |
| 15:55.24 | wcselby | niekvlessert - did you look at "core show application pickup" ? |
| 15:55.39 | niekvlessert | sure we did, what do you want to |
| 15:55.44 | niekvlessert | know |
| 15:55.50 | niekvlessert | callgroup=group and stuff |
| 15:56.19 | *** join/#asterisk vader-- (~me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 15:56.21 | vader-- | hello |
| 15:56.39 | vader-- | my demo adtran total access 924e came in today |
| 15:57.09 | bmoraca_work | nice |
| 15:57.19 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:57.22 | bmoraca_work | i like them, they're fun |
| 15:58.39 | vader-- | i don't like the rack ears they sent |
| 15:58.41 | vader-- | they are dumb |
| 15:58.57 | bmoraca_work | they don't have rack ears normally |
| 15:59.02 | bmoraca_work | they're meant for wall-mounting |
| 15:59.08 | bmoraca_work | you can buy rack ears separate ly |
| 16:00.56 | vader-- | yucky, wall mounting hehe |
| 16:01.11 | vader-- | im trying to decide if i want to play around with FreePBX or SipX |
| 16:01.14 | bmoraca_work | i agree...i'd prefer regular 19" rack ears...but, meh |
| 16:01.15 | vader-- | sipx looks cool |
| 16:02.59 | niekvlessert | wcselby: what about application pickup? |
| 16:04.09 | *** join/#asterisk QbY (~QbY@c-24-126-145-123.hsd1.ga.comcast.net) |
| 16:05.33 | wcselby | niekvlessert - i was curious if you had looked at it as a solution to your needs...? |
| 16:05.46 | QbY | i'm making a bit of a unique realtime/database deployment. Is it possible to tell asterisk to register based on credentials that come from somehwere other than sip_buddies |
| 16:06.59 | niekvlessert | wcselby: i know we can pickup but we cannot pickup a local channel or a zap channel, we can't get it done |
| 16:07.05 | niekvlessert | pickup works fine between sip |
| 16:07.22 | niekvlessert | but there's no sip channel going on |
| 16:08.58 | wcselby | niekvlessert - okay, well, try providing [TK]D-Fender with the information he requested and someone here maybe able to help you more |
| 16:11.41 | niekvlessert | wcselby: i will soon, but i'm working a bunch of things now. :) it's evening soon overhere, so I will have time then |
| 16:20.03 | Naikrovek | ooh |
| 16:20.16 | Naikrovek | i think we've gotten big enough that freepbx can no longer serve us |
| 16:20.28 | Naikrovek | finally i can justify time to get vanilla * |
| 16:20.34 | *** join/#asterisk BreezBl0k (~BreezBl0k@5e0ede22.bb.sky.com) |
| 16:20.50 | Naikrovek | and learn what all you people have been talking about |
| 16:21.32 | BreezBl0k | Hi I have an extension which i can ring other extensions with no problems but when they try and call me it goes straight to VM |
| 16:21.50 | BreezBl0k | The phone is connect through NAT to asterisk |
| 16:22.02 | Naikrovek | your phone is connected through nat? |
| 16:22.27 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.26.154) |
| 16:22.35 | Naikrovek | sounds like * can't reach your phone system so it goes straight to voicemail |
| 16:22.38 | Naikrovek | ~sipnat |
| 16:22.39 | infobot | [~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 |
| 16:23.18 | BreezBl0k | the funny thing is i can ring another phone which is also behind NAT and it works no one way audio issues its perfect |
| 16:26.18 | Naikrovek | yeah |
| 16:26.20 | Naikrovek | that's NAT for ya |
| 16:26.26 | Naikrovek | it's weird on occaision |
| 16:26.32 | Naikrovek | occasion* |
| 16:26.42 | Akiraa | how often is the sip/nat question asked per day? |
| 16:26.49 | LemensTS | . |
| 16:26.50 | Naikrovek | at least 10 |
| 16:27.00 | Naikrovek | but i don't mind answering it |
| 16:30.12 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
| 16:31.09 | *** join/#asterisk GameGamer43 (~GameGamer@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:31.55 | BreezBl0k | hmmm ive re registered it and its working |
| 16:32.32 | BreezBl0k | bring on IPv6 :) |
| 16:33.20 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 16:40.53 | Naikrovek | yeah |
| 16:41.03 | Naikrovek | the problem is that the NAT hole is being forgotten by your router |
| 16:41.20 | Naikrovek | so once that happens, calls will no longer be able to reach you |
| 16:41.26 | Naikrovek | even though you'll be able to call out just fine |
| 16:44.43 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:47.42 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:50.09 | *** join/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 16:50.09 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:53.50 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 16:55.44 | *** join/#asterisk nickaugust (~anonymous@rrcs-71-42-53-182.se.biz.rr.com) |
| 16:55.53 | *** join/#asterisk hmmhesays (~hmmhesays@24-116-107-203.cpe.cableone.net) |
| 16:56.46 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 16:59.44 | smooth_p[work] | http://www.phonebooth.com/ |
| 17:00.57 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:01.34 | hmmhesays | looks like a crappy clone of the other thousands of hosted pbx platforms out there |
| 17:01.47 | Kobaz | heh |
| 17:01.53 | smooth_p[work] | links to "crappy clone of the other thousands of hosted pbx platforms" |
| 17:01.57 | smooth_p[work] | pls k thx |
| 17:02.08 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 17:02.20 | smooth_p[work] | 200 mins a month for free is pretty good Id think |
| 17:02.27 | ghenry | hi, when you leave a voicemail, pressing * prompts you for the voicemail password |
| 17:02.35 | ghenry | is that an inbuilt feature? |
| 17:03.04 | smooth_p[work] | oh well its just inbound calls |
| 17:03.25 | [TK]D-Fender | ghenry: No. |
| 17:04.02 | ghenry | [TK]D-Fender: OK, must be a FreePBX thing |
| 17:04.11 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 17:06.31 | Akiraa | I suppose there is no good hosted pbx solution out there |
| 17:07.02 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:08.06 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 17:10.26 | Katty | has anyone ever experienced having to reboot their polycom phone due to call volume? 200 incoming calls per day, and about 4 reboots. symptoms being the person calling cannot hear the person answering the incoming call, but the person answering can hear the caller. |
| 17:10.56 | Naikrovek | haven't ever had that kind of volume. check the memory on the device before you reboot to see where it's at, i'd say |
| 17:11.25 | smooth_p[work] | does anyone know if theres a free trial for the Skype for Asterisk? |
| 17:12.05 | Chainsaw | smooth_p[work]: They're in it for the money, sorry. |
| 17:12.22 | Katty | Naikrovek: well. |
| 17:12.29 | Katty | Naikrovek: i've swapped around a few phones. |
| 17:12.37 | Katty | Naikrovek: doesn't seem to fix the issue. :/ |
| 17:12.43 | Naikrovek | different models? |
| 17:12.45 | Katty | Naikrovek: which leads me to believe that it's not actually the phone |
| 17:12.54 | Katty | Naikrovek: hmmm, perhaps not different models. |
| 17:12.56 | Naikrovek | oh |
| 17:12.58 | smooth_p[work] | Chainsaw: hrm |
| 17:13.31 | Katty | Naikrovek: they've all be 5xx somethings |
| 17:13.33 | Chainsaw | smooth_p[work]: Then again, making a purchase puts an obligation to deliver service on Digium. I'm sure you could terminate the contract if it fails to operate. |
| 17:13.52 | Naikrovek | Katty: ah. i don't know the details of those models. call polycom? |
| 17:14.03 | Katty | Naikrovek: yes i was going to do that here this afternoon |
| 17:14.19 | Chainsaw | Katty: I've had that, it was the RJ11 from the handset. |
| 17:14.22 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 17:14.26 | [TK]D-Fender | [13:11]<smooth_p[work]>does anyone know if theres a free trial for the Skype for Asterisk? <- no |
| 17:14.28 | Katty | oh really? |
| 17:14.38 | Chainsaw | Katty: Yeah, on a 670. After trying an incompatible headset. |
| 17:14.44 | Katty | Chainsaw: oh |
| 17:14.51 | Katty | Chainsaw: hrmm, no headsets with these |
| 17:14.53 | Chainsaw | Katty: Physically unplugging & replugging the handset connector fixed it up. No amount of powercycling did. |
| 17:14.54 | Katty | Chainsaw: which rules that out |
| 17:15.02 | Naikrovek | Katty: he said "handset" the first time |
| 17:15.03 | Chainsaw | Katty: Perhaps worth a go. |
| 17:15.04 | Naikrovek | then "headset" |
| 17:15.11 | Katty | yes perhaps |
| 17:15.17 | Katty | won't hurt anything, that's for sure |
| 17:15.35 | Chainsaw | Katty: I unplugged, replugged with the power on, then did a soft reboot. |
| 17:15.43 | Katty | k |
| 17:17.31 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 17:19.12 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 17:22.59 | sawgood | I have a series of related questions ... If I have a fresh build of CentOS 5.4, and I then want to add Asterisk on top of this (with no FreePBX) ... are there existing RPM packages for 1.6.1 and/or 1.6.2, or are the only RPM versions available now still 1.6.0 builds? |
| 17:23.10 | Qwell | sawgood: Only 1.6.0 for now |
| 17:23.22 | sawgood | qwell: thank you ... so |
| 17:23.24 | Qwell | 1.6.2 will be packaged Real Soon Now |
| 17:23.39 | sawgood | knowing this ... lets say I do that ... CentOS 5.4 with Asterisk 1.6.0 ... right |
| 17:23.50 | sawgood | then, Digium releases a RPM for 1.6.1 ... |
| 17:24.12 | sawgood | would I be able to upgrade from 1.6.0 to 1.6.1 (since these are two different versions)? |
| 17:24.19 | Qwell | That's the plan |
| 17:24.35 | sawgood | what about 1.6.0 to 1.6.2.x |
| 17:24.44 | Qwell | it would skip 1.6.1 |
| 17:25.02 | sawgood | everything in 1.6.1 is in 1.6.2 I heard? |
| 17:25.12 | Qwell | in theory |
| 17:25.19 | Kobaz | except depricated stuff that was removed |
| 17:25.51 | sawgood | So, if I have 1.6.0 now ... from an AsteriskNOW 1.5 build ... does that mean I have the Asterisk addons, or is that an additional RPM I can install? |
| 17:25.53 | Kobaz | and bugs of course too... all the bugs in 1.6.1 are also in 1.6.2, for your convenience |
| 17:26.11 | Qwell | sawgood: -addons is also packaged |
| 17:26.16 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 17:26.20 | Qwell | sawgood: i386 or x86_64? |
| 17:26.26 | sawgood | i386 |
| 17:26.52 | Qwell | rpm -i http://packages.asterisk.org/centos/5/current/i386/RPMS/asterisknow-version-1.5.0-1_centos5.noarch.rpm |
| 17:26.59 | Qwell | yum install asterisk16 asterisk16-addons |
| 17:27.04 | sawgood | So, how long as the official RPM for 1.6.0 been out? |
| 17:27.06 | leifmadsen | in theory, nothing that is deprecated should be removed |
| 17:27.12 | Qwell | leifmadsen: in theory |
| 17:27.13 | sawgood | thanks |
| 17:27.14 | leifmadsen | that is a policay change :) |
| 17:27.23 | Kobaz | leifmadsen: some stuff does get removed |
| 17:27.25 | leifmadsen | Qwell: unless of course something is so severely broken that it makes no sense to use :) |
| 17:27.36 | leifmadsen | Kobaz: hence "in theory" |
| 17:27.39 | Kobaz | heh |
| 17:27.43 | sawgood | I meant ... how long in time has the 1.6.0 official RPM been out to the public? |
| 17:27.49 | Kobaz | like show channels no longer works in 1.6.2 |
| 17:27.51 | Kobaz | you need core show channels |
| 17:27.53 | Qwell | sawgood: a while |
| 17:28.03 | sawgood | more than 1 year? |
| 17:28.03 | Qwell | Nov or so? |
| 17:28.16 | Qwell | err |
| 17:28.18 | Qwell | Nov 08 |
| 17:28.24 | Kobaz | actually... all of 'show' was removed in/before 1.6.2 |
| 17:28.28 | sawgood | 1.5 years |
| 17:28.30 | leifmadsen | Kobaz: that was something that started in 1.4 but was never finished -- an in fact "show channels" DOES work if you enable CLI aliases. We even created the templates for backwards compatibility for your convenience |
| 17:28.35 | *** join/#asterisk dynamicpulse (~dynamicpu@adsl-99-172-50-102.dsl.emhril.sbcglobal.net) |
| 17:28.39 | Kobaz | leifmadsen: interesting |
| 17:29.07 | Kobaz | i need to make a new alias |
| 17:29.09 | leifmadsen | Kobaz: yes, junky, mvanbaak and myself spent 3 days getting CLI updated (based on the work started by file) |
| 17:29.10 | Kobaz | core fix bugs |
| 17:29.11 | sawgood | so, 1.6.0 RPM has been out for 1.5 years ... and nobody has made a 'updated' RPM for 1.6.1.x or 1.6.2.x |
| 17:29.14 | leifmadsen | Kobaz: :) |
| 17:29.21 | Qwell | sawgood: correct |
| 17:29.29 | Katty | i really hate dealing with bank of america |
| 17:29.35 | leifmadsen | sawgood: that's because the recommended way to install is via source |
| 17:29.36 | sawgood | Qwell: how can this be the case ... seems like a very long time? |
| 17:29.55 | leifmadsen | sawgood: lack of interest from the community when other methods are readily available and more highly supported |
| 17:30.17 | leifmadsen | sawgood: installing from RPM is not useful if you end up with crash and want to report a back, because your backtrace will not be useful (as an example) |
| 17:31.02 | leifmadsen | in addition, resources have been better spent on resolving bugs and testing new features, rather than the time spent supporting RPMs |
| 17:31.07 | sawgood | leifmadsen: sounds cool to me ... but the 'junior market' ... the people who 'want' Asterisk but have no clue .. are semi-limited if they do not know how to use src code |
| 17:31.28 | Qwell | sawgood: those same people probably don't want/need/use all the newest features |
| 17:31.41 | *** join/#asterisk slinksh0t (~slinksh0t@74.115.208.59) |
| 17:31.44 | [TK]D-Fender | sawgood: If you can't handle compiling * then you sure as hell can't handle CONFIGURING it. |
| 17:31.46 | leifmadsen | well, Asterisk isn't simple to begin with, so those who are most successful with it are capable of compiling it. The market you speak of is better suiting using something like PBX in a Flash or trixbox |
| 17:32.00 | Kobaz | heh |
| 17:32.02 | Qwell | glares at leifmadsen |
| 17:32.03 | dddh | should 1.6.2.5 be upgraded to 1.6.2.6? |
| 17:32.16 | leifmadsen | dddh: if your in house testing passes, then sure |
| 17:32.24 | Kobaz | i found that once i became familiar with the c source code, i became much better at using asterisk |
| 17:32.32 | Chainsaw | dddh: Some useful fixes in there. I think it's worth your time. |
| 17:32.38 | dddh | leifmadsen: ok |
| 17:32.39 | leifmadsen | eyes Qwell in a laser beam manner |
| 17:32.46 | Chainsaw | dddh: My downstream patchset is finally shrinking :) |
| 17:32.58 | dddh | Chainsaw: I am afraid g729 may not work when version will change ;) |
| 17:33.00 | Kobaz | my downstream patchset is growing nicely |
| 17:33.18 | sawgood | from a 'sales' point of view ... (meaning the power of the software) ... what does 1.6.1 or 1.6.2 offer the public which would justify using more than 1.6.0? |
| 17:33.41 | sawgood | point being here is value add ... 1.6.0 like the pack ... or 1.6.2 as a value add |
| 17:33.46 | Chainsaw | sawgood: Much better T38 support. |
| 17:34.01 | Chainsaw | sawgood: If you care about faxing at all, you probably want 1.6.2 |
| 17:34.15 | sawgood | nice nick Chainsaw |
| 17:34.17 | sawgood | funny! |
| 17:35.20 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 17:35.37 | drfreeze | Hello |
| 17:35.47 | sawgood | Ok ... one more side question ... if I spent a long time setting up CentOS 5.4 with Asterisk 1.6.2 from src ... and all is working 'cool' ... would imaging the hard drive with a product like Aronics be an effective way to maange the inital install? |
| 17:36.11 | Qwell | You're outside the scope of this channel now |
| 17:36.15 | drfreeze | I've got a strange situation |
| 17:36.41 | drfreeze | I have a client that goes thru and automated system at an insurance company |
| 17:36.53 | drfreeze | At one point they ask for a representative. |
| 17:37.06 | leifmadsen | ~asterisk-versioning |
| 17:37.10 | leifmadsen | ~asterisk-versions |
| 17:37.11 | infobot | rumour has it, asterisk-versions is http://www.asterisk.org/asterisk-versions |
| 17:37.14 | leifmadsen | sawgood: see above |
| 17:37.21 | drfreeze | When they do, they are connected to an inbound call back to their office |
| 17:38.08 | drfreeze | The dial command being used to call the number is a plain dial command, no options, so I don't think there is any callee controlling of the phone system going on |
| 17:38.45 | drfreeze | When the transfer happens, the AGI script ends (which should hang up the call) and the inbound call starts |
| 17:38.49 | drfreeze | http://pastie.textmate.org/private/f4fegureg1m3o5fvicw |
| 17:39.39 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 17:39.57 | drfreeze | Anyone have some suggestions as to what is going on? |
| 17:41.04 | sawgood | ok ... sub-variable ... is Asterisk 1.6.2.x 'meant' to be setup from src on a system with no Asterisk already, or can you have Asterisk 1.6.0 (from RPM) and then 'build' from scr to 1.6.2? |
| 17:41.33 | ChannelZ | I wouldn't |
| 17:41.54 | leifmadsen | you can once you remove the RPM :) |
| 17:42.05 | [TK]D-Fender | sawgood: Doesn't matter. Wipe the modules folder and upgrade right over whatever you had prior |
| 17:42.07 | leifmadsen | I can just see problems caused by overwriting the RPM |
| 17:42.08 | sawgood | you guys are hard core! |
| 17:42.14 | leifmadsen | not really... :) |
| 17:42.31 | ChannelZ | well yeah remote the rpm first |
| 17:42.35 | leifmadsen | svn co http://svn.asterisk.org/svn/asterisk/tags/1.6.2.6 && cd 1.6.2.6 && ./configure && make install |
| 17:42.52 | ChannelZ | Else some time down the road it might decide to upgrade its self and stomp on top of _you_ |
| 17:43.02 | sawgood | ah! |
| 17:43.53 | sawgood | So, if someone 'says' they are running Asterisk 1.6.2 ... is it 'assumed' they are using FreePBX, not using FreePBX, or it is never assumed? |
| 17:44.00 | leifmadsen | 1.6.2 means nothing |
| 17:44.03 | ChannelZ | not assumed |
| 17:44.07 | leifmadsen | it is ambiguous |
| 17:44.11 | ChannelZ | I don't even assume they're running 1.6.2 |
| 17:44.13 | leifmadsen | 1.6.2.6 is a version |
| 17:44.14 | ChannelZ | :P |
| 17:44.19 | [TK]D-Fender | sawgood: Nothing is assumed |
| 17:44.30 | Qwell | sawgood: If somebody says they're driving a Ford, what color should be assumed? |
| 17:44.39 | Beave | blue |
| 17:44.41 | [TK]D-Fender | sawgood: Except maybe that they are running Asterisk 1.6.2.X |
| 17:44.41 | leifmadsen | Qwell: black |
| 17:44.47 | shader | Or what brand even? |
| 17:44.48 | ChannelZ | Rust |
| 17:44.50 | shader | lol |
| 17:44.54 | Beave | hah |
| 17:45.07 | sawgood | So, "officially" is FreePBX the GUI for Asterisk (are they partenered in some way)? |
| 17:45.13 | [TK]D-Fender | sawgood: No |
| 17:45.26 | Qwell | FreePBX is a GUI for Asterisk |
| 17:45.29 | [TK]D-Fender | sawgood: 3rd party product |
| 17:45.29 | shader | though AsteriskNOW uses it I think |
| 17:45.42 | sawgood | 3rd party ... got it! |
| 17:45.52 | ChannelZ | call it bundled |
| 17:46.07 | sawgood | this cat is driving me CRAZY! ... he has learned how to beg for treats by scratching on the desk real fast! |
| 17:46.16 | leifmadsen | o.O |
| 17:46.29 | leifmadsen | ok, that's my cue to do something else :) |
| 17:46.30 | ChannelZ | Two words: airsoft gun |
| 17:46.46 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:46.46 | [TK]D-Fender | sawgood: Then every time it does, grab a spray bottle full of water and spritz the cat. It will soon learn something quite different |
| 17:47.01 | Kobaz | heh, my cousin uses the spray bottle on her dog |
| 17:47.03 | sawgood | its the girlfriends cat ... so gotta be nice! |
| 17:47.05 | ChannelZ | [TK]D-Fender: to love water? :) |
| 17:47.15 | [TK]D-Fender | ChannelZ: OR ELSE |
| 17:47.46 | sawgood | I'm the main one who gives him treats, so he keeps coming to me (he has learned) ... maybe he can do the src of Asterisk for me ... |
| 17:48.25 | ChannelZ | http://www.youtube.com/watch?v=8KswnjMa-MQ |
| 17:48.32 | ChannelZ | This one has a drinking problem |
| 17:48.49 | drfreeze | [TK]D-Fender: Did you see my post above? |
| 17:49.44 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 17:49.47 | [TK]D-Fender | drfreeze: that pastebin shows nothing useful |
| 17:49.53 | sawgood | that cat video was funny |
| 17:50.29 | *** join/#asterisk sgimeno (~chatzilla@62.32.225.37) |
| 17:51.01 | drfreeze | [TK]D-Fender: Neither does the Master.log. The inbound call (from the transfer) is not recorded |
| 17:51.33 | sawgood | If I have Asterisk 1.6.x build from src .... will 'everything' related to the dialplan go into extensions.conf, or are there many subfiles like extensions_custom.conf when using FreePBX? |
| 17:51.46 | Chainsaw | sawgood: I have everything in a single file. |
| 17:51.47 | drfreeze | I'm trying to determine where the call comes from. If it is really the callee calling back into the callers office, or something else. |
| 17:52.53 | sawgood | Chainsaw: very nice ... same thing for the extensions in sip.conf? |
| 17:52.53 | Chainsaw | sawgood: I have sip.conf seperated out. I use includes there. |
| 17:52.54 | drfreeze | Since it is the inbound dialplan, I think that would mean the call must be originating from outside |
| 17:52.54 | sawgood | I understand the need for having more than one file ... many reasons ... but good God FreePBX has really gone overboard |
| 17:52.59 | [TK]D-Fender | sawgood: Its your dialplan, do whatever you want with it. |
| 17:53.26 | [TK]D-Fender | sawgood: No, you don't NEED them. And I disagree on their going "overboard" |
| 17:54.51 | sawgood | [TK]D-Fender: to me, if one is using FreePBX ... you have to find the right file which is not over-written by FreePBX, so your changes are not lost ... I am having a hard time trying to figure out which file is the 'last stop' on the food chain, to where I can know 100% for sure my edits will work |
| 17:55.27 | ChannelZ | The point of the GUI is so you don't have to |
| 17:55.36 | sawgood | I guess once I complete my 'master list' of filenames for FreePBX, this will become a piece of cake |
| 17:55.36 | ChannelZ | So if you WANT to, that probably means the GUI is not serving you well, so why use it? |
| 17:55.53 | [TK]D-Fender | sawgood: Yuo have failed to understand the point of FreePBX. You aren't supposed to have to #&$^ with the config files manually! |
| 17:56.10 | *** join/#asterisk jlpicard1701e (~jlpicard1@LCaen-151-92-24-3.w217-128.abo.wanadoo.fr) |
| 17:56.19 | jlpicard1701e | uh.... hi? |
| 17:56.23 | [TK]D-Fender | sawgood: You seem to think that FreePBX is some sort of valid starting point to learning *. It is NOT. It is a starting point to STOPPING from learning * |
| 17:56.36 | sawgood | Well, sometimes a 'customer' will have a fucntioning system which only needs a tweak or two (and outside of FreePBX is easier than inside of it) |
| 17:56.54 | ChannelZ | until they untweak it |
| 17:56.54 | sawgood | [TK]D-Fender: you are 100% correct ... thank you ... |
| 17:57.00 | [TK]D-Fender | sawgood: Well if you're the integrator, then you should already know better. |
| 17:57.13 | shader | how would you set up voicemail transcription? |
| 17:57.17 | jlpicard1701e | I'm a french user and I'd like to configure the keyboard in french plz..... how could i do it? |
| 17:57.34 | [TK]D-Fender | jlpicard1701e: What "keybaord"? This isn't a distro support channel |
| 17:57.39 | sawgood | I deal with many IP PBX boxes .... most do not have an Asterisk front end ... I come in after the inital install |
| 17:57.42 | bmoraca_work | shader: get a really good speech-to-text program and use externnotify |
| 17:57.46 | ChannelZ | scrape off the lettering on the keys and get a sharpie |
| 17:58.47 | jlpicard1701e | [TK]D-Fender> I just installed asterisk on a Vbox in order to test it.... and I'd like configure the keyboard... |
| 17:58.53 | sawgood | I'm pretty sure you can buy a French keyboard ... I've seen many non English keyboards before (Chinese ones are always a trip to use) |
| 17:59.15 | [TK]D-Fender | jlpicard1701e: Distro's aren't supported here, only * itself. Go to #centos and ask there |
| 17:59.26 | [TK]D-Fender | jlpicard1701e: That is what AsteriskNOW is based on |
| 17:59.39 | jlpicard1701e | [TK]D-Fender> thanks! |
| 18:01.01 | sawgood | Have you ever known someone who has to look at their keys to type? I had a co-worker like that ... when he would take breaks, other co-workers would swap the keys around on his keyboard to piss him off |
| 18:01.04 | sawgood | ha! |
| 18:02.30 | rttrey | lol yah we do that to the new guys that come in to my job |
| 18:04.48 | hardwire | if anybody is looking for a te410p + echocan can you contact me in privmsg. I'm selling some new and used equipment. |
| 18:05.58 | coppice | sawgood: Chinese keyboards are fine for English speakers |
| 18:06.21 | anonymouz666 | what about Chinese ATAs? |
| 18:11.13 | Katty | sighs |
| 18:11.27 | Katty | 3 weeks to see a specialist |
| 18:11.59 | coppice | well it wouldn't be special if you could do it every day, would it? |
| 18:12.06 | Katty | :P |
| 18:12.07 | sawgood | my question is how did Katty ... sigh (what keyboard syntax produced that output) I've forgotten how to do that |
| 18:12.11 | Katty | that's enough out of you smarty pants |
| 18:12.25 | *** part/#asterisk jlpicard1701e (~jlpicard1@LCaen-151-92-24-3.w217-128.abo.wanadoo.fr) |
| 18:12.30 | Katty | <PROTECTED> |
| 18:12.37 | coppice | I'm not wearing any pants |
| 18:12.38 | sawgood | like this |
| 18:12.43 | Katty | coppice: k |
| 18:12.44 | sawgood | ah! ty! |
| 18:13.07 | Katty | my tinnitus from those 3 or 4 days of prozac has continued for 5 months |
| 18:13.20 | Katty | nearly 6 months now. |
| 18:13.42 | sawgood | you were on Prozac for only 3 or 4 days? |
| 18:13.50 | sawgood | it takes a lot longer than that to kick in |
| 18:14.09 | Katty | prozac is an ototoxic drug |
| 18:14.23 | Katty | and it does not take 3 or 4 days for adverse side effects to kick in |
| 18:14.45 | sawgood | oh ... you meant the 'bad' stuff ... I was talking about the 'good' stuff |
| 18:14.46 | sawgood | sorry! |
| 18:14.55 | Katty | a normal Ringing of the Ears reaction should have been gone by now, i would think |
| 18:15.01 | Katty | unless it i has done perminent hearing damage |
| 18:15.08 | sawgood | to stop hearing voices is at least a 6-8 week process ... ha ha ha |
| 18:15.21 | Katty | funny how it can cause both perminent hear damage, and temporary tinnitus both |
| 18:15.27 | coppice | Katty: nothing biological is permanent |
| 18:15.27 | Katty | tho i'm not sure how temporary 6 months really is |
| 18:15.35 | Katty | coppice: well of course it isn't |
| 18:15.43 | Katty | coppice: self awareness is just a skiwshy meat side effect |
| 18:15.45 | aceio | hi all |
| 18:16.21 | Katty | coppice: but, regardless, they wanna put me through a hearing test. |
| 18:16.34 | aceio | just finished compling asterisk |
| 18:16.36 | Katty | coppice: to see if my hearing has been severely impacted--which i don't think it can |
| 18:16.50 | Katty | coppice: erm, i don't think it has. just due to the fact that everyone still sounds as annoying as ever ;) |
| 18:17.55 | aceio | <PROTECTED> |
| 18:18.09 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.22 | aceio | will this work ? |
| 18:20.01 | Katty | coppice: the test will give them some clue as to where the 'ringing' is coming from.. |
| 18:20.37 | Katty | coppice: if no significant hearing damage has been done, and the ringing is from the inner ear...i just have plain ole tinnitus |
| 18:20.57 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 18:22.26 | coppice | I'd like to know where my ringing is coming from. I had ringing, and it turned out I had massive amounts of earwax. during the clearing out of the earwax the whistling went away several times, but when all the wax was gone I ended with a stable state of (thankfully very gentle) whistling |
| 18:25.43 | wcselby | Naikrovek - you about? |
| 18:26.04 | wcselby | did you say you had a script for generating large -directory.xml files for polycom phones? |
| 18:26.49 | ecrane | ever do that ear candle thing? At the end you can see lots of stuff come out.. or maybe the wax came from the candle.. hard to tell... |
| 18:27.21 | Katty | coppice: i don't think mine is caused by ear wax. |
| 18:27.42 | Katty | coppice: after i stopped taking prozac (after the 4th pill) i waited about a month for the tinnitus to go away. when it didn't i went to see an ENT |
| 18:27.55 | Katty | coppice: she poked about my ears and looked for anything abnormal without much luck... |
| 18:28.03 | Katty | coppice: if it was waxy...i'm guessing she would have found it |
| 18:28.25 | bmoraca_work | has anyone else noticed that 1.6.2.0 puts the color codes in the log files? is there anyway to turn that off or is it fixed in a later version? |
| 18:28.51 | beek | WTF? Is this #asterwax? |
| 18:29.04 | coppice | well, I'm pretty sure mine is related in some way, as I had short periods of totall silence during the several sessions it took to clean all the crap out |
| 18:29.26 | coppice | beek: this is a listening related issue. its highly relevamt |
| 18:29.46 | beek | Are you getting any jitter on the ringing? |
| 18:29.54 | beek | Any dropped ringing packets? |
| 18:30.02 | beek | One-way ringing audio? |
| 18:30.29 | LemensTS | bmoraca_work: wouldnt that have to do with your editors configuration |
| 18:30.43 | Katty | coppice: eek :< |
| 18:30.48 | Katty | coppice: that's scary |
| 18:31.00 | bmoraca_work | LemensTS: it's plain text. it puts the color codes ([0m, for instance) in the log files. |
| 18:31.05 | Katty | coppice: did you panic when everything went silent? |
| 18:31.41 | coppice | I don't mean the whole world went silent. just the whistling stopped |
| 18:32.16 | LemensTS | bmoraca_work: ah gotchya |
| 18:32.33 | bmoraca_work | it seems to be a "feature"...albeit not one that's very useful or disablable |
| 18:32.52 | bmoraca_work | doesn't want to have to patch logger.c |
| 18:33.22 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 18:34.19 | bmoraca_work | ahhh |
| 18:34.26 | bmoraca_work | 1.6.2.6 fixes it...from 3 days ago |
| 18:35.37 | bmoraca_work | oh, no, wait...looks like i lied |
| 18:36.00 | bmoraca_work | guess i need to add it to my log archiving daily cron job |
| 18:40.26 | hardwire | bmoraca_work: what happened? |
| 18:40.44 | hardwire | haha |
| 18:40.47 | hardwire | nevermind |
| 18:40.48 | hardwire | that rules |
| 18:40.49 | bmoraca_work | nothing...someone decided it'd be a good idea to log the console color codes to the log files |
| 18:40.52 | hardwire | I've never seen that before |
| 18:40.53 | Katty | so i can still hear up to the 16Khz area |
| 18:41.00 | hardwire | I'm using 1.6.2.5 |
| 18:41.02 | Katty | -42DB |
| 18:41.15 | hardwire | I also turn off color tho |
| 18:41.19 | hardwire | now I'm afraid to turn it on |
| 18:41.21 | bmoraca_work | ahhh |
| 18:41.34 | Katty | but i can't hear much below 90Hz |
| 18:41.45 | Katty | -6DB |
| 18:42.58 | Katty | or perhaps these speakers can't produce audio below 90Hz |
| 18:43.01 | Katty | that's also a possibility |
| 18:45.45 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 18:46.45 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 18:51.33 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 18:52.01 | adnc | hello, is it possible to find out which pattern did match to the active calls? |
| 18:53.09 | LemensTS | adnc: do a verbose line before the priority of the dial cmd |
| 18:53.25 | adnc | well the call is already active |
| 18:53.48 | LemensTS | Cant make another call? |
| 18:53.53 | adnc | no |
| 18:54.09 | adnc | i just would like to check if the proper route has been taken |
| 18:56.54 | adnc | core show channels shows everything |
| 18:57.58 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 18:59.49 | *** join/#asterisk Alagar (~Administr@122.164.36.10) |
| 19:00.01 | ManxPower-work | <PROTECTED> |
| 19:00.15 | Qwell | What's invalid about it? |
| 19:00.38 | ManxPower-work | Qwell, someone said an SDP with no codec is invalid. |
| 19:00.39 | adnc | isn't it a regular INVITE? |
| 19:00.47 | *** join/#asterisk slinksh0t_ (~slinksh0t@64.120.149.85) |
| 19:01.02 | Qwell | ManxPower-work: There is a "codec" |
| 19:01.03 | ManxPower-work | The polycom sends back SIP/2.0 488 Not Acceptable Here |
| 19:01.20 | ManxPower-work | Qwell, only rfc2833, right? |
| 19:01.27 | Qwell | yeah :p |
| 19:01.45 | Qwell | I don't know if that's valid or not. |
| 19:01.51 | ManxPower-work | Why is asterisk not including the other codecs in the invite (this is a transfer, the peer has codecs set) |
| 19:01.55 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:02.37 | ManxPower-work | Maybe we should just roll back to 1.4.23.1 |
| 19:02.38 | wcselby | why when I reboot my polycom 601, it won't upload it's contact directory into a MAC-directory.xml file? |
| 19:02.47 | Katty | in this line: WMMDDYY:HHMM what does the W stand for? |
| 19:02.48 | ManxPower-work | wcselby, because you screwed something up. |
| 19:03.16 | wcselby | ManxPower-work - that's entirely possible. but what? |
| 19:03.28 | tzafrir | day of the week? |
| 19:03.28 | Katty | oh, nevermind. it's weekday |
| 19:03.30 | wcselby | i also don't have any -boot or -app files for this particular phone |
| 19:03.31 | ManxPower-work | wcselby, could be dozens of things. |
| 19:03.44 | anonymouz666 | ManxPower-work: port 0 on m= attribute is also very strange. I'd say that 18 is rtpmap for g729 |
| 19:04.04 | ManxPower-work | usually it's a permissions problem because the user PlcmSpIp does not have permission to write to the directory |
| 19:04.35 | wcselby | works with all the other phones....all should have the same settings. i'll go recheck. |
| 19:04.57 | ManxPower-work | anonymouz666, we have g729 licenses, g729 codec is 256, not 18 |
| 19:05.04 | wcselby | well yeah it's the same username in the ftp logs as the other phones |
| 19:06.12 | ManxPower-work | wcselby, then your phone config is messed up. |
| 19:06.21 | ManxPower-work | check the MAC.cfg file |
| 19:09.18 | wcselby | blah |
| 19:09.24 | wcselby | everything is setup exactly the same way as the others |
| 19:09.32 | wcselby | i think maybe we'll look at swappign out the phone |
| 19:09.33 | ManxPower-work | wcselby, if it was, then it would be working |
| 19:10.13 | anonymouz666 | ManxPower-work: 256? |
| 19:10.26 | ManxPower-work | anonymouz666, "core show codecs" |
| 19:10.27 | anonymouz666 | is that the output from show codecs? |
| 19:10.31 | anonymouz666 | ooh |
| 19:11.16 | *** join/#asterisk Z_God (~julius@wlan236133.mobiel.utwente.nl) |
| 19:11.42 | ManxPower-work | I have about 3 hours to fix before we roll back to 1.4.23.1 |
| 19:15.21 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 19:15.39 | anonymouz666 | you won't see 256 as a static payload type, but I think it is missing the a= attribute to describe the g729... something like a=rtpmap:18 G729/8000 |
| 19:16.00 | ManxPower-work | anonymouz666, so it looks like abug? |
| 19:16.30 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.225.167.dsl.dyn.forthnet.gr) |
| 19:17.11 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.47 | ManxPower-work | anonymouz666, only happens on an attended transfers |
| 19:19.21 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 19:21.56 | wcselby | ManxPower-work - what version of * are you running? |
| 19:22.12 | ManxPower-work | wcselby, 1.4.30 |
| 19:22.23 | wcselby | ahh |
| 19:33.44 | wcselby | fuckin' a |
| 19:33.46 | wcselby | sorry |
| 19:33.55 | wcselby | it was a permissions issue... |
| 19:39.21 | wcselby | that.....and you don't need to reboot a phone to upload the MAC-directory.xml file...you can just edit a contact. |
| 19:40.00 | Netgeeks | Anyone here familiar with/has used ConfBridge app? |
| 19:40.08 | *** join/#asterisk citrus (~citrus@wsip-98-173-200-235.sb.sd.cox.net) |
| 19:41.53 | citrus | need some help with a new setup not working with inbound call, asterisk shows Asterisk1*CLI> |
| 19:41.54 | citrus | <PROTECTED> |
| 19:41.56 | citrus | <PROTECTED> |
| 19:41.58 | citrus | <PROTECTED> |
| 19:41.59 | citrus | but thats it. never dumps it into any context or anything |
| 19:42.09 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 19:42.22 | ManxPower-work | wcselby, It usually is |
| 19:42.35 | ManxPower-work | citrus, "sip set debug on" |
| 19:43.55 | citrus | ok |
| 19:44.01 | citrus | i got an output |
| 19:44.43 | bmoraca_work | ~pb |
| 19:44.44 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:45.14 | citrus | http://pastebin.com/d6sNCCg8 |
| 19:46.00 | bmoraca_work | is that all? |
| 19:46.02 | [TK]D-Fender | citrus: under [nexvortex] set "insecure=port,invite" |
| 19:46.09 | bmoraca_work | if so, your peer is incorrectly configured |
| 19:46.46 | bmoraca_work | i gotta do some reading up on SER...i'm not familiar enough with it... |
| 19:47.16 | ManxPower-work | Is the correct command to check out the latest non-released Asterisk 1.4? "svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4-SVN" |
| 19:48.45 | LemensTS | i think i use co instead of checkout not sure if it matters |
| 19:49.14 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 19:50.05 | *** join/#asterisk sulex (~sulex@host-78-14-170-90.cust-adsl.tiscali.it) |
| 19:50.22 | outtolunc | manx, that should pull it.. i see stuff last update like 63 minutes ago |
| 19:50.57 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 19:51.46 | bmoraca_work | i thought the ./branches directory had all of the different source branches (1.6.2.0, 1.6.2.1, etc) and not just the latest... |
| 19:51.52 | bmoraca_work | curious |
| 19:52.07 | Katty | WAHHHHHHHHHHHHH |
| 19:52.17 | Katty | screams, rips hair out, shreds curtains, etc |
| 19:52.27 | ManxPower-work | bmoraca_work, I've not had to try SVN is *years* |
| 19:52.32 | Katty | breathes. |
| 19:52.33 | ManxPower-work | stupid bug |
| 19:53.02 | bmoraca_work | why do you need to now? |
| 19:53.21 | ManxPower-work | bmoraca_work, because I suspect a bug in chan_sip.c. |
| 19:53.27 | bmoraca_work | ahhhh |
| 19:53.31 | *** join/#asterisk Gestahlt (~chatzilla@HSI-KBW-078-042-049-043.hsi3.kabel-badenwuerttemberg.de) |
| 19:53.55 | ManxPower-work | bmoraca_work, http://pastebin.ca/1842669 |
| 19:54.06 | Gestahlt | Hi |
| 19:54.07 | *** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net) |
| 19:54.14 | Gestahlt | I have troubles getting chan_lcr to work |
| 19:54.25 | Gestahlt | either i set up something wrong or the dialstring is wrong |
| 19:54.55 | bmoraca_work | ManxPower-work: what's wrong with it? |
| 19:55.07 | Gestahlt | i use this syntax: LCR/Ext/$OUTNUM$ |
| 19:55.19 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 19:55.23 | ManxPower-work | bmoraca_work, I've been told that it is an invalid packet. The phone rejects it. |
| 19:55.52 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:55.56 | ManxPower-work | Gestahlt, Asterisk variables are ${VARNAME} not $VARNAME$ |
| 19:56.23 | bmoraca_work | ManxPower-work: his syntax is a custom trunk in freepbx |
| 19:56.32 | Gestahlt | Ah sorry. I use Freepbx as GUI. There i have to enter for my B1 (Which works great) CAPI/ISDN1/$OUTNUM$ |
| 19:56.43 | Gestahlt | :) exactly bmoraca |
| 19:56.45 | ManxPower-work | bmoraca_work, Ah. I wonder why he is wasting his time here. |
| 19:56.47 | ManxPower-work | ~freepbx |
| 19:56.48 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:56.57 | Gestahlt | Yeah that is ok |
| 19:57.04 | ManxPower-work | Gestahlt, no, it's not. |
| 19:57.05 | bmoraca_work | ManxPower-work: i don't know enough about SIP to know why that would be an invalid packet...but to my untrained eye, it looks normal |
| 19:57.07 | Gestahlt | I made a custom config for LCR |
| 19:57.18 | ManxPower-work | bmoraca_work, notice the lack of codec stuff |
| 19:57.41 | Gestahlt | So i have to deal with the regular asterisk dialplan |
| 19:57.43 | ManxPower-work | I don't know enough about SIP to really know if it's not valid or not, but the polycoms think it's invalid. |
| 19:57.51 | ManxPower-work | Gestahlt, if you want help here, yes. |
| 19:57.53 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 19:57.59 | bmoraca_work | ManxPower-work: this isn't the first SIP header in the call, though. that might have been handled in the initial invite...although i didn't think that phones did any sort of authentication |
| 19:58.06 | ManxPower-work | Or you can go to the proper place for FreePBX support. |
| 19:58.24 | Gestahlt | im in both places. I consider my chan_lcr issue moer asterisk related |
| 19:58.30 | ManxPower-work | bmoraca_work, We only have the problem when a specific sequence of events happen during a transfer. |
| 19:58.38 | ManxPower-work | Gestahlt, is chan_lcr included in Asterisk? |
| 19:58.41 | Gestahlt | aye |
| 19:58.46 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:58.47 | ManxPower-work | Must be 1.6 |
| 19:58.48 | Gestahlt | it shows even active |
| 19:58.49 | Gestahlt | 1.6.2 |
| 19:58.57 | Gestahlt | i compiled chan_lcr from the sources |
| 19:58.59 | Gestahlt | it worked before |
| 19:59.07 | Qwell | ManxPower-work: No it is not. |
| 19:59.08 | Gestahlt | but then i had a mess with my sql |
| 19:59.12 | bmoraca_work | ahh, during transfer...that'd explain the higher call sequence number |
| 19:59.19 | Gestahlt | ah included |
| 19:59.22 | Gestahlt | sorry read to fast |
| 19:59.30 | Gestahlt | and my dialplan went poof |
| 19:59.35 | Gestahlt | so i had to redo everything |
| 19:59.43 | Gestahlt | everything works now except for the chan_lcr |
| 19:59.45 | ruben23 | hi, if i re-compile again asterisk and upgrade to higher version, used dahdi instead of zaptel, what should i do- do i need to remove zaptel..? |
| 19:59.53 | ManxPower-work | bmoraca_work, only during an attended answer where the dialed number is answered before the transfer completes. |
| 20:00.09 | ManxPower-work | ruben23, You should read UPGRADE*.txt. Didn't I tell you this before? |
| 20:00.10 | Gestahlt | ah wait.. focus |
| 20:01.03 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 20:01.24 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 20:01.27 | ruben23 | ManxPower-work: not yet..where i can find this txt..? |
| 20:01.46 | ManxPower-work | ruben23, the Asterisk source tarball, where all the official docs live. |
| 20:01.54 | Gestahlt | Ok. I compiled and installed asterisk 1.6.2. I also compiled and installed chan_lcr (1.5/1.6? Latest). I had it running before. When i dial out it, the asterisk tells me Chan unavial = trunk failure or something like that |
| 20:02.10 | Gestahlt | when i dial in i dont get any output from the cli |
| 20:02.11 | ManxPower-work | Gestahlt, I wish you the BEST of luck. |
| 20:02.17 | Gestahlt | Ok |
| 20:02.22 | Gestahlt | i need skill |
| 20:02.24 | Gestahlt | not luck |
| 20:02.26 | sun28 | moin |
| 20:02.43 | ManxPower-work | no, you need luck. I suspect not a single person here will help you with chan_lcr |
| 20:03.01 | Gestahlt | is it broken? |
| 20:03.20 | Gestahlt | or just because it isnt integrated into asterisk? |
| 20:03.28 | Gestahlt | or both? |
| 20:03.36 | ManxPower-work | Gestahlt, I have no idea. I just know I've been using Asterisk for 10 years and only heard of people using chan_lcr a couple of times. |
| 20:04.03 | redax | hi. |
| 20:04.10 | ManxPower-work | So, I wish you the BEST of luck. |
| 20:04.16 | ManxPower-work | redax, UPGRADE*.txt |
| 20:04.17 | Gestahlt | Yeah Manx.. i notice that while researching. All information you get is a mixture between obsolete and completly wrong |
| 20:04.18 | bmoraca_work | Gestahlt: most people who use LCR have developed their own lookups for it. |
| 20:04.52 | redax | ManxPower-work: hm? |
| 20:05.12 | bmoraca_work | with FUNC_ODBC and stored procedures, it would be trivial to create a very fast lookup strictly in dialplan |
| 20:05.23 | ManxPower-work | redax, seems most questions people are asking recently is "what do I have to know when I upgrade Asterisk". The answer is UPGRADE*.txt |
| 20:05.38 | redax | heh :D thanks. |
| 20:05.39 | ruben23 | ManxPower-work:ill be getting the upgrade text of the old version or the new verison i plan to run..? |
| 20:05.48 | Gestahlt | btw: Any good module for using mobile phones for GSM trunks? |
| 20:05.51 | ManxPower-work | ruben23, read them all |
| 20:07.36 | redax | actually I wanted to ask a little bit different thing, like is there a guideline, what is the recommended linux kernel configuration for asterisk ? |
| 20:08.41 | shader | has anyone here used Mottovoip.nl's service? |
| 20:09.18 | *** part/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 20:10.08 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:10.56 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 20:10.58 | ManxPower-work | redax, whatever comes with your distro |
| 20:12.14 | Gestahlt | redax: depends what you need.. if you have fritz card PCI hardware.. you run out of luck with newer distros |
| 20:12.58 | redax | actually I have ISDN cards, but mISDN driven |
| 20:14.10 | redax | especially I'd like to know the timer frequency settings... default is 250Hz.... but elastix which using CentOS 5.3, uses 1000Hz Timer.. |
| 20:14.41 | Gestahlt | redax: Please tell me you use chan_lcr |
| 20:15.01 | Gestahlt | its driving me nuts |
| 20:15.34 | redax | hm. haven't known we have chan_lcr. is that related to least cost routing? |
| 20:17.21 | redax | ManxPower-work: so, any generic kernel config should work well, even for faxing? :-o |
| 20:17.36 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 20:17.43 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.48 | ManxPower-work | what makes you think you can do anything in the kernel setup that would impact faxing? |
| 20:19.19 | redax | never though, until today evening... I've installed an * 1.6.0.25 for our office, and faxes from our 100km far away office doesn't comes at all. *(sip trunk via the offices) |
| 20:19.51 | redax | meanwhile an elastix box (ALIX 3D3 -- so not a craftwerk box) takes all the faxes |
| 20:20.41 | redax | the strange thing, it hangs up when the modem training voices started... t38 enabled both end, allowed only g711a codec. |
| 20:20.57 | redax | both using SPA-2102 as ATA |
| 20:22.31 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 20:22.36 | redax | any idea? :-o |
| 20:22.39 | Gestahlt | redax: No, chan_lcr is the replacement for chan_misdn |
| 20:22.48 | Gestahlt | redax: in mISDN2 |
| 20:23.14 | redax | hm. cool. does it working at least better than mISDN ? |
| 20:23.41 | redax | I had millions of problem with mISDN in the last few years. |
| 20:24.50 | Gestahlt | redax: I got it working once but then i messed up my asterisk box.. |
| 20:24.56 | Gestahlt | redax: while it worked, it worked fine |
| 20:25.13 | Gestahlt | redax: and now i try to get it working again.. which is why i am here |
| 20:25.48 | adnc | is there a documentationdatabase for asteriskmodules? |
| 20:26.16 | ManxPower-work | adnc, "core show applications" "core show functions" |
| 20:26.21 | Gestahlt | redax: but i just found a good howto which seems like my working configuration |
| 20:26.45 | adnc | ManxPower-work, for example res_esel.so? |
| 20:26.48 | *** join/#asterisk boch (~fran@200.61.191.9) |
| 20:27.00 | ManxPower-work | Gestahlt, I STRONGLY doubt mISDN and LCR are related. |
| 20:27.08 | ManxPower-work | adnc, vague questions get vague answers. |
| 20:27.25 | ManxPower-work | I don't have a res_esel.so on my version of Asterisk |
| 20:27.26 | Gestahlt | manx: not LCR (least cost routing) chan_lcr (linux call router) |
| 20:27.35 | Gestahlt | manx: pls check it out on the mISDN homepage |
| 20:27.45 | ManxPower-work | Gestahlt, Golly, beave! Could have told us that earlier |
| 20:27.50 | adnc | ManxPower-work, soso |
| 20:27.59 | Gestahlt | manx: chan_lcr is mISDN2.. its the replacement for chan_misdn |
| 20:28.19 | ManxPower-work | adnc, you don't mean res_ael.so, do you? |
| 20:28.24 | redax | just in time... |
| 20:28.29 | adnc | ManxPower-work, no, res_esel.so |
| 20:28.37 | adnc | and some others which debian loads by default |
| 20:28.41 | Gestahlt | redax: just fyi DO NOT USE THE GIT Source |
| 20:28.46 | redax | but if the module framework is named as `misdn' what's wrong with the name chan_misdn |
| 20:28.47 | Gestahlt | Its broken |
| 20:28.49 | ManxPower-work | adnc, we don't support packages here, you knopw that. |
| 20:28.53 | adnc | i wanted to take off those i don't need |
| 20:28.56 | Gestahlt | download the tarball and compile it from there |
| 20:29.08 | adnc | ManxPower-work, i didn't ask support for packages |
| 20:29.09 | redax | heh, mISDN git sources is like russian roulette |
| 20:29.18 | Gestahlt | redax: no clue.. i think because there was added a lot of functionality |
| 20:29.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:30.56 | adnc | simply where documentation about modules can be found. and this question is pure asterisk |
| 20:31.40 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 20:33.30 | redax | Gestahlt: I'm fear of the name mISDN :D |
| 20:34.06 | redax | but will give a chance |
| 20:34.32 | Gestahlt | redax: mISDN added the word "fear" to my vocabulary |
| 20:34.59 | Gestahlt | redax: im so frigging happy with the AVM Fritz B1 |
| 20:35.06 | Gestahlt | redax: almost no effort to configure |
| 20:35.39 | *** join/#asterisk kazaa_lite (~eddie@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 20:36.52 | Gestahlt | redax: But beware of mISDN2 howtos.. they might mislead you |
| 20:37.21 | shader | anyone here use asterisk with google voice? |
| 20:37.34 | redax | Gestahlt: but must say, I'm using Billion Tiny USB TA in a few places, which is working without problems... |
| 20:37.47 | redax | I had serious problems with WC B410P |
| 20:37.54 | shader | can you set your caller id information with asterisk, or does that have to be done by google? |
| 20:40.53 | idespinner | anyone here ever seen a TDMoE(redfone) channel on asterisk go "yellow alarm" before? |
| 20:41.24 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 20:42.14 | pabelanger | anybody know the CPU of the AA55? |
| 20:42.25 | pabelanger | AA50* |
| 20:45.52 | idespinner | pabelanger, /proc/cpuinfo on an AA60 reports: Intel(R) Celeron(R) CPU 220 @ 1.20GHz |
| 20:46.03 | idespinner | err sorry |
| 20:46.08 | idespinner | misread you |
| 20:46.34 | idespinner | on the AA50 its a blackfin |
| 20:46.55 | idespinner | but it also has a dedicated DSP |
| 20:47.36 | idespinner | pabelanger, http://pastebin.com/kdnfVet2 |
| 20:48.51 | beek | Gang... I'm sure that I remember reading this in one of the config files, but I'll be damned if I can find it. |
| 20:48.59 | pabelanger | idespinner: Thanks for the output |
| 20:49.15 | beek | Isn't there a way to have Asterisk run a system command after it starts? |
| 20:50.21 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 20:52.27 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 20:56.43 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:58.16 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:59.36 | *** join/#asterisk brezular (~brezular@adsl-dyn178.91-127-122.t-com.sk) |
| 21:01.57 | *** part/#asterisk kazaa_lite (~eddie@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 21:09.17 | shader | beek: edit the init script? |
| 21:09.29 | shader | (I don't know, never tried) |
| 21:10.00 | beek | shader: That's the brute force way, but possible. |
| 21:11.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 21:14.10 | shader | is there anything in asteriskNOW that can't be installed via yum on CentOS? |
| 21:14.22 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:14.24 | shader | or is it just preconfigured better? |
| 21:15.22 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 21:15.44 | *** join/#asterisk kazaa_lite (~eddie@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 21:15.57 | kazaa_lite | hi all... how can i set calls per second for asterisk? |
| 21:18.13 | idespinner | running into a minor issue with color on the asterisk console |
| 21:18.18 | b14ck | what do you mean set calls per second? |
| 21:18.18 | idespinner | ...No entry for terminal type "xterm" |
| 21:18.44 | idespinner | and theres no color, any ideas? ive tried setting $term to xterm-color |
| 21:19.40 | idespinner | kazaa_lite, us the -M argument for the asterisk executable |
| 21:19.54 | idespinner | "asterisk -M <value>" Limit the maximum number of calls to the specified value |
| 21:20.28 | kazaa_lite | cool and how can i test it from CLI? |
| 21:20.40 | kazaa_lite | like for freeswitch i have fsctl cps |
| 21:20.44 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 21:21.31 | citrus | how do i dump asterisk console output to a file? |
| 21:21.36 | Cresl1n | redax: you there still? |
| 21:21.52 | b14ck | citrus, asterisk -r > file |
| 21:21.54 | b14ck | =p |
| 21:21.58 | citrus | :) |
| 21:22.06 | idespinner | citrus, or alternatively check out logger.conf |
| 21:22.20 | idespinner | in /etc/asterisk/logger.conf |
| 21:26.24 | sawgood | I have a box with 1.4.21.2 on it ... from the Asterisk CLI ... I can see my *65 feature code parsing .... but I cannot hear any audio (if I call my SIP trunk number) ... the line rings, but I get no audio in either direction |
| 21:26.48 | sawgood | I know this points to RTP 10000-20000, but the admin of network says the ports are open and forwarded |
| 21:26.56 | citrus | i am having an issue with voice, when a call comes in i pick up on the sip client but we can't talk to eachother at all, i think its an RTP problem but i am unsure, http://pastebin.com/tcXqeFCD |
| 21:28.15 | Katty | hi |
| 21:28.29 | sawgood | how can I see 'which' port the RTP traffic is coming in/out on? |
| 21:28.37 | sawgood | can I do this from the Asterisk CLI? |
| 21:29.07 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 21:29.14 | Katty | my asterisk does not work at all how to fix pls |
| 21:30.54 | dennisG | katty, do you get an error ? |
| 21:31.16 | dennisG | and do you see any proces thingy of asterisk? |
| 21:31.24 | Katty | that's funny |
| 21:31.45 | dennisG | haha why katty? |
| 21:32.16 | dennisG | because your asterisk work like a charm? :P |
| 21:32.19 | sawgood | any way to determine which UDP port traffic is flowing on ... outside of having to do a tcpdump? |
| 21:32.34 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:32.54 | dennisG | sawgood, just use tshark / wireshark ? |
| 21:33.05 | dennisG | it's live data.. |
| 21:33.24 | dennisG | or check your firewall/router/layer 3 switch :) |
| 21:33.42 | idespinner | sawgood, dont see any useful data in core show channel or sip show channel |
| 21:33.48 | idespinner | you may have to use tcpdump |
| 21:34.17 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:34.26 | Katty | dennisG: there is no error. |
| 21:34.32 | idespinner | or enable RTP debugging |
| 21:34.58 | Katty | or you could just check your firewall logs for policy violations |
| 21:35.10 | Katty | that'd be too easy tho |
| 21:35.41 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
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| 21:38.01 | Katty | dennisG: please stop sending me notices. |
| 21:38.11 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:39.07 | p3nguin | notices katty |
| 21:39.11 | Katty | ohai |
| 21:39.16 | p3nguin | ;) |
| 21:39.32 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:40.08 | Katty | p3nguin: i had to go work on an icky samsung 7100 box this afternoon |
| 21:41.09 | p3nguin | Were you at least successful in your works? |
| 21:41.22 | Katty | of course. |
| 21:41.27 | Katty | irsmrt |
| 21:42.46 | outtolunc | she bought a vowel <G> |
| 21:42.49 | Katty | p3nguin: they have a known issue of not being able to hold time. |
| 21:42.50 | outtolunc | ducks |
| 21:43.16 | Katty | p3nguin: they lose about 1.5 minutes per month. no fix...something about the time chip they use |
| 21:43.52 | sawgood | I did a tcpdump 'capture' of the *65 feature code attempt, and it shows the UDP port being used around 14600 (which is cool) ... but the destination IP address is a 192.x.x.x (which is my PC out the WAN) |
| 21:44.04 | sawgood | So, is my concern of no audio a NAT problem maybe? |
| 21:44.27 | Katty | two words. |
| 21:44.27 | p3nguin | No way to replace the part that's junk? |
| 21:44.29 | Katty | firewall. logs. |
| 21:44.48 | Katty | p3nguin: this is a an appliance we're talkin about |
| 21:44.56 | citrus | do have to specify my internal and external IP in asterisk 1.6.2 |
| 21:44.57 | sawgood | I see other non RTP packets coming to my PC on its WAN correct public IP (ssh traffic) ... but RTP traffic is using the 192.x.x.x address |
| 21:45.13 | p3nguin | I would crack open the case and see about changing the part if it's faulty. |
| 21:45.14 | Katty | p3nguin: i ain't gonna rip it open and rip out a chip and all that jazz |
| 21:45.20 | p3nguin | Ah, I would. |
| 21:45.25 | Katty | you'r ealso not a samsung partner |
| 21:45.39 | idespinner | sawgood, sounds like a nat problem for sure |
| 21:45.54 | idespinner | sawgood, do a sip show channel xxxxxxx |
| 21:46.09 | idespinner | and check out the recievedaddress and theoretical address |
| 21:46.16 | Katty | p3nguin: besides, i might get /gasp/ dirt under my nails. |
| 21:48.24 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 21:48.48 | sawgood | idespinner one sec |
| 21:50.57 | sawgood | What is the syntax for xxxxxx |
| 21:51.12 | vader-- | any of you guys use SIPx? |
| 21:51.16 | Katty | syntax error near 'for' |
| 21:51.31 | idespinner | its [tab] |
| 21:52.00 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 21:52.59 | sawgood | idespinner: I did a sip show channels command |
| 21:53.00 | sawgood | I see nothing with my WAN IP in the output |
| 21:53.00 | Katty | heh |
| 21:53.00 | sawgood | should I do this with a call in progress? |
| 21:53.08 | Katty | goes home |
| 21:53.54 | sawgood | I started a call, and I see my 192.x.x.x address in the SIP show channels window |
| 21:53.55 | idespinner | sawgood, yes, call in progress |
| 21:54.13 | sawgood | I see my 192 address not my public IP |
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| 21:55.38 | *** part/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:57.02 | p3nguin | sawgood: Describe your setup from the networking point-of-view. |
| 22:00.32 | sawgood | idespinner: working now ... I needed a sip_nat.conf file |
| 22:00.36 | sawgood | whew! |
| 22:00.37 | hmmhesays | Katty, you never call, you never write |
| 22:01.10 | idespinner | sip_nat.conf? ive never heard of that |
| 22:01.20 | idespinner | but whatever works you know... |
| 22:01.30 | idespinner | are you running freepbx or something? |
| 22:01.57 | sawgood | idespinner: yes |
| 22:02.57 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 22:04.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 22:05.00 | QbY | in Dial(SIP/xxxx) -- how do i specify the username and password |
| 22:06.20 | [TK]D-Fender | QbY: Dial(SIP/user:pass@host/extentodial) |
| 22:06.21 | idespinner | QbY, by the peer definition in SIP.conf |
| 22:06.25 | korihor | how set dev_state on realtime SIP? |
| 22:06.42 | [TK]D-Fender | QbY: which you shouldn't do in Dial, but rather using a peer entry |
| 22:07.07 | QbY | is it not possible to pass the credentials in the dial string? |
| 22:07.12 | korihor | this is always 'Not in use' for realtime SIP |
| 22:08.00 | QbY | i'm feeding the dial cmd with values from a db.. |
| 22:08.01 | [TK]D-Fender | QbY: I just gave you the format... |
| 22:08.15 | QbY | ok, awesome.. didn't see it |
| 22:09.42 | adnc | i can not find any documentation about format_ogg_vorbis.so could someone please point me to one if there is a documentation to it? |
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| 22:17.52 | mythicalbox | just to confirm, ss7 over e1 doesn't exist right, the signaling would be something like esf,b8zs, etc |
| 22:19.39 | tzafrir | mythicalbox, why would you say that? |
| 22:22.43 | *** part/#asterisk slinksh0t (~slinksh0t@64.120.149.85) |
| 22:23.13 | mythicalbox | asked a client what protocol will be provided for the e1 they are having provisioned, and they replied "ss7" |
| 22:23.57 | [TK]D-Fender | adnc: What is there to document? |
| 22:24.09 | vader-- | hmm been trying to find people who use or have used sipx |
| 22:24.16 | vader-- | SipXecs |
| 22:24.27 | vader-- | looks interesting |
| 22:24.55 | adnc | [TK]D-Fender, i do get a warning saying that the ogg file can not be used for streaming, i wonder if there is something to configure |
| 22:27.46 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 22:27.49 | [TK]D-Fender | adnc: No... there is no streaming, just like it says. fixed file only |
| 22:29.59 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 22:34.40 | joako | I want to record calls in asterisk and use some web-based GUI to review them. Is there such a thing? |
| 22:35.33 | [TK]D-Fender | joako: #freepbx <- |
| 22:36.10 | Kobaz | 994M /var/log/asterisk/full |
| 22:36.14 | Kobaz | time to log rotate i guess |
| 22:37.48 | joako | [TK]D-Fender, Isn't freepbx to manage all of asterisk? I was hoping there is something only to manage the recordings. |
| 22:39.18 | [TK]D-Fender | joako: its a bunch of files.. no mapping into anything else necessarily. You could just point Apache at it an be done with it. |
| 22:39.36 | [TK]D-Fender | joako: there is no assumed file naming structure... or way to tie to anythingt else |
| 22:42.37 | *** join/#asterisk Daviey (~Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 22:43.59 | joako | I found something it I guess you name the recording ${UNIQUEID} and it looks up the CDR info.. |
| 22:44.46 | Kobaz | you can name the recording whatever you want, but yes uniqueid is quite useful |
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| 22:49.36 | *** join/#asterisk `paul (~kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 22:49.55 | `paul | how do i include a file in extensions.ael ? |
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| 22:54.34 | fifer | I'm still working on an issue I have with Aastra 6731i phones and dtmf |
| 22:55.08 | fifer | I'm now in * 1.6.0.26 but had the issue in 1.4.30 |
| 22:55.10 | fifer | I |
| 22:55.54 | *** join/#asterisk dzup (dzup@unaffiliated/dzup) |
| 22:55.55 | fifer | I have tried all the settings for DTMF on the phone, I have other phones, including Aastras that work fine, it is just these 6731i's of which I'm testing on 3, all doing the same thing |
| 22:56.18 | fifer | I have dtmf debugging turned on in the log and while I can see the dtmf from the other phones, nothing from these. |
| 22:56.46 | fifer | I'm just trying to see if anyone has seen this issue. |
| 22:57.09 | fifer | I have found a single bug report on * but it is unclear to me if it is the same issue and what the status is. |
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| 23:44.23 | QbY | we have a TE405P connected to an Adit 600. With lines going to subscriber premises, not Telco. We can't get the Adit to output dialtone--what's the first place you'd look? |
| 23:45.26 | paulc | in the manual? |
| 23:45.27 | paulc | giggles |
| 23:45.34 | paulc | sorry - I'm in a pissy mood and work's doing my head in |
| 23:46.03 | QbY | same here |
| 23:46.05 | idespinner | QbY, What kinda lines, analog? fxo/fxs? I would plug a butt-set to the lines to check for dialtone |
| 23:46.10 | paulc | are you seeing anything on the console when a subscriber goes off hook? |
| 23:47.13 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 23:48.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:48.46 | sawgood | so, I have a Asterisk 1.4.21.2 box with FreePBX which has Polycom end points ... when you hit the speakerphone you get dial tone (I have a route for 7 + 11 digits on outbound dialing) ... the phone 'cuts' off after getting to the last two remaining needed digits |
| 23:49.03 | idespinner | sawgood, digitmap on polycom phones.... |
| 23:49.10 | sawgood | it automatically stops accepting input and starts a system prompt, "all circuits are busy" |
| 23:49.28 | sawgood | idespinner: Is this an easy fix? |
| 23:49.34 | idespinner | yes |
| 23:49.48 | sawgood | log into the GUI of the phone to change the digitmap? |
| 23:49.59 | idespinner | how are you provisioning? |
| 23:50.16 | sawgood | I'm not really sure ... I'm just helping someone out with this case ... |
| 23:50.36 | idespinner | if the phones are manually configured, yes, log into each phone |
| 23:51.19 | idespinner | its under sip -> local settings -> digitmap |
| 23:51.21 | sawgood | Is digitmap a 'number' or syntax I put in the GUI |
| 23:51.32 | idespinner | syntax |
| 23:51.57 | idespinner | http://www.voip-info.org/wiki/view/Polycom+Phones#Digitmapreference |
| 23:52.03 | sawgood | thank you |
| 23:52.09 | QbY | idespinner: they are fxo to the customer--have plugged in a buttset and it worked.. |
| 23:52.23 | QbY | we upgraded from 1.2 to 1.6--everything owrked fine under 1.2 |
| 23:52.32 | QbY | but with dahdi, they don't see to wanna agree |
| 23:52.46 | idespinner | dahdi config files changed abit between dahdi and zaptel |
| 23:53.06 | *** join/#asterisk aandrade (~aandrade@189.34.124.123) |
| 23:53.20 | QbY | yeah, comparing the two now |
| 23:53.31 | idespinner | QbY, http://www.voip-info.org/wiki/view/DAHDI#ConversionfromZaptel |
| 23:53.48 | idespinner | i know, some people probably think i'm evil for quoting voip info so much |
| 23:54.03 | Corydon76-dig | EVIL!!! |
| 23:54.11 | idespinner | yea yea go ahead and say it |
| 23:54.14 | Corydon76-dig | points at idespinner |
| 23:54.16 | idespinner | you know your thinking it |
| 23:54.50 | Corydon76-dig | EVIL MINDREADER!!! |
| 23:57.35 | fifer | I'm still working on an issue I have with Aastra 67xxi phones and dtmf |
| 23:57.37 | fifer | I'm now in * 1.6.0.26 but had the issue in 1.4.30 |
| 23:58.01 | fifer | I have tried all the settings for DTMF on the phone, I have other phones, including Aastras that work fine, it is just these 6731i's of which I'm testing on 3, and a 6755i, all doing the same thing |
| 23:59.47 | idespinner | maybe factory default the phones? |