| 00:00.28 | _ShrikE | Happy New Year everyone! |
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| 00:33.53 | dacs | ~books |
| 00:34.48 | dacs | ~book |
| 00:34.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:42.36 | jblack | Is the following legal? lblackall=${lblack}&${lblackmyth} |
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| 00:57.57 | etfonhomey | [TK]D-Fender, got much experience with call queues? |
| 00:59.07 | etfonhomey | Or anyone else? |
| 00:59.45 | dacs | etfonhomey: stick with [TK]D-Fender he is the master |
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| 01:00.04 | etfonhomey | I noticed you guys became friends earlier. |
| 01:00.07 | dacs | [TK]D-Fender: lol |
| 01:00.21 | dacs | etfonhomey: yep |
| 01:00.41 | etfonhomey | He's helped me with many * / Polycom problems. |
| 01:00.51 | dacs | etfonhomey: do you know how i can restart my * |
| 01:01.07 | etfonhomey | From the * CLI? |
| 01:01.17 | etfonhomey | restart gracefully |
| 01:01.44 | etfonhomey | If not at the * CLI, get there and then do the restart gracefully |
| 01:01.58 | dacs | nope i can't get to the CLI , because * not started yet |
| 01:02.17 | etfonhomey | Kill the process? |
| 01:02.39 | dacs | and how to start it again, this is what i am asking? |
| 01:02.45 | etfonhomey | "asterisk" |
| 01:03.24 | Corydon76-dig | You should probably use "safe_asterisk" to start it |
| 01:03.38 | etfonhomey | If it's already running, you'll get "Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect" |
| 01:03.56 | etfonhomey | Corydon76-dig, got much experience with call queues? |
| 01:03.57 | Corydon76-dig | or asterisk -Rc |
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| 01:04.13 | mrdigital-deskto | hi |
| 01:04.22 | Corydon76-dig | etfonhomey: uh, no? |
| 01:04.37 | dacs | mpg123: no process killed |
| 01:04.37 | dacs | Asterisk ended with exit status 1 |
| 01:04.37 | dacs | Asterisk died with code 1. |
| 01:04.58 | Corydon76-dig | dacs: it's probably already running |
| 01:05.04 | dacs | Automatically restarting Asterisk. |
| 01:05.15 | Corydon76-dig | dacs: killall safe_asterisk |
| 01:05.32 | etfonhomey | Corydon76-dig, oops, got a sec to help me with a small call queue problem? |
| 01:09.31 | dacs | now i am in the CLI |
| 01:09.57 | etfonhomey | OK, ok, here's my call queue issue: Agent A logins in to receive calls and is the only logged in agent. Agent A makes an outgoing call from his phone. While Agent A is on the phone to his outgoing call, a call comes into the queue. |
| 01:10.22 | etfonhomey | The call gets sent to Agent A even though he is still on the phone with the outgoing call he placed. |
| 01:11.34 | etfonhomey | What setting am I missing that tells the queue not to send the call to Agent A. I've set ringinuse=no |
| 01:12.37 | etfonhomey | in queues.conf |
| 01:15.30 | dacs | how to tell if SIP messages are making it from the Asterisk server to my ATA |
| 01:15.51 | etfonhomey | Does your ATA have a web interface and a log? |
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| 01:17.07 | dacs | etfonhomey: my phone was ringing when i call it using just the ATA, now i config my ATA to work with *, when i call i don't hit my * |
| 01:17.22 | dacs | 'Phone does't ring |
| 01:19.05 | etfonhomey | sip set debug peer nameofyouratainsip.conf |
| 01:19.24 | etfonhomey | And see if what messages you get in the CLI. |
| 01:19.34 | k-man | how do you associate a particular extension number with a particular phone? |
| 01:20.27 | etfonhomey | extensions.conf |
| 01:22.50 | k-man | ok |
| 01:24.07 | k-man | how do i reload the extensions.conf file? |
| 01:24.55 | etfonhomey | extensions reload |
| 01:25.02 | etfonhomey | oops |
| 01:25.11 | etfonhomey | or is it dialplan reload |
| 01:25.51 | etfonhomey | or both? |
| 01:28.01 | etfonhomey | No one with any clues on my call queue problem? |
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| 01:48.30 | jblack | k-man: asterisk -r, then dialplan reload |
| 01:55.37 | k-man | thanks |
| 01:56.17 | jblack | There it is |
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| 02:06.24 | clusco | hie everyone! |
| 02:07.45 | jblack | hi |
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| 02:09.27 | k-man | i've tried to set up a test extension (500) to echo |
| 02:09.47 | k-man | but when i dial the number, asterisk says extension not found |
| 02:10.34 | clusco | emmm..... |
| 02:10.44 | clusco | try create 1 more extension |
| 02:11.44 | k-man | to do the same thing? |
| 02:12.27 | k-man | when i reload the dialplan, it says adding 500 |
| 02:12.59 | k-man | do i have to add something to allow my extension to dial 500? |
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| 02:13.52 | k-man | ie, my extension is called [jason] |
| 02:14.04 | k-man | do i have to allow jason to dial 500 somehow? |
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| 02:26.54 | clusco | hie everyone.... |
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| 03:02.21 | jblack | k-man: ping |
| 03:08.10 | jblack | k-man: Well, when you come back, I can answer your question, if you dont' already have the answer |
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| 03:56.44 | HaMYaI | Hi, I am having problem with agi->get_data() when the caller enters the number so fast |
| 03:56.51 | HaMYaI | asterisk will just hangup |
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| 04:28.13 | BearPerson | I know this will sound silly, but - is there a software modem utility for asterisk? |
| 04:29.20 | BearPerson | I'm wondering if it's possible to abuse one of my voip numbers to provide myself with (horribly slow, but cheap) net access when I'm out somewhere with just a laptop, modem, and a phone line ;) |
| 04:31.56 | jblack | why don't you look at mgetty? |
| 04:32.09 | jblack | that takes a real number, but <shrug> |
| 04:33.59 | BearPerson | mgetty still seems to need some kind of modem |
| 04:34.41 | dmz | i've not had much luck with voip & modems |
| 04:34.48 | BearPerson | what I'm looking for is something that provides a terminal [device] on one end and eats/produces sound sent through asterisk on the other |
| 04:35.09 | dmz | voip quality tends to not be where it needs to be for modulation to work properly w/modems |
| 04:35.24 | dmz | bugger, just upgraded a box from 1.2 -> 1.4 and now extensions can't call each other |
| 04:35.28 | BearPerson | I admit that tunneling modulated data through voip is somewhat redundant, but the result could be neat ;) |
| 04:35.52 | dmz | it would be really neat, and if you can get it working post on a wiki somewhere :) |
| 04:36.00 | BearPerson | :) |
| 04:36.37 | BearPerson | "be your own ISP with asterisk and a spare voip number" ;) |
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| 05:04.29 | Nugget | how does that make you an ISP? |
| 05:05.00 | Nugget | oh, n/m, I see. |
| 05:07.14 | mrdigital-deskto | linuxmce uses asterisk cool |
| 05:10.07 | etfonhomey | OK. Who wants to help me with a minor call queue issue? |
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| 05:36.01 | [TK]D-Fender | etfonhomey, What is it? |
| 05:36.35 | etfonhomey | There you are. |
| 05:36.54 | etfonhomey | OK. Here goes: |
| 05:37.24 | etfonhomey | I have one (and only one) agent logged into the queue to receive calls. |
| 05:37.57 | *** join/#asterisk cyberscan (n=cybersca@dblt-216-227-19-216.gtcom.net) |
| 05:38.27 | etfonhomey | That agent makes a call to anywhere and while on that call, a call comes into the queue. |
| 05:38.32 | [TK]D-Fender | etfonhomey, pastebin it all |
| 05:38.52 | etfonhomey | ringinuse=no :) Too late just one more line |
| 05:39.13 | etfonhomey | The call gets delivered to the agent even though they are on the phone. |
| 05:40.15 | [TK]D-Fender | etfonhomey, Good... now pastebin it all. |
| 05:40.21 | etfonhomey | Oh. |
| 05:40.41 | etfonhomey | Thought you wanted me to pastebin my explanation cause it was long winded. |
| 05:41.07 | etfonhomey | What do you need? agents.conf, extensions.conf, and queues.conf and verbose CLI output? |
| 05:41.26 | [TK]D-Fender | etfonhomey, Everything re;event |
| 05:42.03 | etfonhomey | Is it my connection or is pastebin.ca down? |
| 05:43.24 | etfonhomey | ~pb |
| 05:43.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 05:52.07 | etfonhomey | [TK]D-Fender, http://pastebin.com/d5d1becf1 That's agents.conf, queues.conf, and extensions.conf, gonna recreate the issue and paste the CLI verbose output now. |
| 05:53.15 | [TK]D-Fender | etfonhomey, * can't know that a callback agent is on the phone. It has no idea what that dialplan that is to be called will DO. |
| 05:54.28 | [TK]D-Fender | etfonhomey, and secondly, FFS stop using extesn with VOICEMAIL FOLLOWING for your agents! |
| 05:54.38 | etfonhomey | Why is there the "ringinuse" option? |
| 05:54.57 | [TK]D-Fender | etfonhomey, only functional for DEVICES. Chan_local is not a device. |
| 05:55.23 | etfonhomey | so the agents are considered chan_sip? |
| 05:56.40 | *** join/#asterisk mrdigital (n=kjhh@65-78-103-248.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 05:57.19 | [TK]D-Fender | etfonhomey, No, agents as called through agentcallbacklogin is jsut a local channel. not a SIP channel. |
| 05:57.43 | [TK]D-Fender | etfonhomey, its all jsut dumb dialplan. For all * knows, it'll just call Wait(100000000) and sit around doing NOTHING. |
| 05:58.46 | etfonhomey | [TK]D-Fender, what I want to accomplish is that if this situation does occur that the call will wait until the agent is off the phone. |
| 05:59.01 | etfonhomey | [TK]D-Fender, any work around? |
| 05:59.01 | jblack | Ah. I see agent is french for "slave wage treadmill of neverending death" |
| 05:59.45 | [TK]D-Fender | jblack, no its for "Tabarnac y-a trop d'anglophones ici." |
| 05:59.48 | [TK]D-Fender | ;) |
| 06:00.14 | [TK]D-Fender | etfonhomey, well if they're in a queue it'll jsut keep calling till it gets through. |
| 06:00.32 | etfonhomey | [TK]D-Fender, as long as I don't have that voicemail in there, right? |
| 06:00.33 | [TK]D-Fender | etfonhomey, depending on your timeouts. That and not using extens with VOICEMAIL ATTACHED DAMMIT :p |
| 06:00.38 | etfonhomey | :) |
| 06:00.54 | [TK]D-Fender | etfonhomey, haven't I bludgeoned you for this earlier already? :) |
| 06:00.59 | jblack | Heh. Google tells me that means "Tabarnac are too many English here." |
| 06:01.38 | etfonhomey | [TK]D-Fender, I'm sure. What's your best practices for doing VM for internal extensions? |
| 06:01.57 | [TK]D-Fender | jblack, French tend to use religious words as swear words considering their deep involvelement with the church. hence the use of "tabernacle" (translated). |
| 06:02.25 | [TK]D-Fender | jblack, you might jsut as well say "holy shit" or "fuck" in its place for your understanding :) |
| 06:02.34 | jblack | Ahhhh. |
| 06:02.53 | etfonhomey | That was the only part of that comment that I wasn't sure about... |
| 06:02.58 | [TK]D-Fender | welcome to #linguistics |
| 06:03.17 | jblack | Yes... Fuck the americans, the great cockroach of the internet. |
| 06:03.26 | etfonhomey | :) I can read and translate French pretty well, but have never been immersed in it. |
| 06:03.50 | [TK]D-Fender | etfonhomey, the problem is you didn't hold your head under long enough ;) |
| 06:04.20 | jblack | et... I have voice mail for internal extensions, and I didn't have to go through any weird sort of agent stuff. |
| 06:04.38 | jblack | Are you doing something weird, or could a newbie like me be able to lend you a hand? |
| 06:04.41 | etfonhomey | I was in Paris last summer and would say something that they understood, but then would just stare glass eyed when I tried to slow down their response so that I could understand it. |
| 06:05.42 | [TK]D-Fender | jblack, he's using call Queue for distribution of queued up callers to the first available agent. Not a basic dial. |
| 06:06.06 | [TK]D-Fender | jblack, This functionality is what is assumed when someone says the want to set up an "inbound call center" |
| 06:06.06 | jblack | Ok, so he's doing what he's supposed to be doing, then subverting it somehow |
| 06:06.19 | [TK]D-Fender | jblack, Correct. |
| 06:06.23 | etfonhomey | [TK]D-Fender, looking through the CLI output, I see where it registers as Local. |
| 06:06.32 | [TK]D-Fender | jblack, "queues.conf" , "show application queue" |
| 06:07.00 | [TK]D-Fender | etfonhomey, Congratulations, now make you loging use another context where the extens DON'T lead to an "Answer" condition. |
| 06:07.00 | jblack | Yeah. I understood agent when I looked it up. I used to pity and cry for the monkeys every time I went out to chain smoke |
| 06:07.04 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 06:07.41 | [TK]D-Fender | jblack, not healthy.... consider a resolution for it... |
| 06:07.53 | etfonhomey | [TK]D-Fender, "loging"? |
| 06:08.11 | etfonhomey | Ah, logins! |
| 06:08.34 | jblack | as it's already the 2nd, I'll consider it for next year |
| 06:09.58 | [TK]D-Fender | jblack, Its the 1st in PST still... |
| 06:10.48 | etfonhomey | [TK]D-Fender, If I use a Goto to jump to another context, will ${CALLERID(num)} still be the same? |
| 06:11.27 | [TK]D-Fender | etfonhomey, just DON'T ok? stop asking for pain and just double it up. |
| 06:13.10 | jblack | Ahh, luck for me that I'm in EST. :) |
| 06:13.22 | jblack | Or unlucky as you'd likely phrase it |
| 06:13.30 | jblack | et: Yes. |
| 06:13.36 | etfonhomey | [TK]D-Fender, How do I set it up so that the phone can have two contexts as you suggest? |
| 06:14.05 | [TK]D-Fender | jblack, pray for your tar-charred lungs and those around you :| |
| 06:14.13 | etfonhomey | [TK]D-Fender, btw, what did you do to dacs? |
| 06:14.48 | [TK]D-Fender | etfonhomey, that isn't a PHONE having 2 contexts, thats your agent dialing INTO a separate context to ones that point to the same DEVICES. |
| 06:14.53 | jblack | Uh-uh. I pray for no one. I already know what my tombstone will read, and I smoke outside these days. Even when it's <0c out |
| 06:15.06 | [TK]D-Fender | etfonhomey, And please stop calling a SIP device an "extension". |
| 06:17.00 | etfonhomey | [TK]D-Fender, How do I have my agent dial into separate contexts? |
| 06:17.01 | [TK]D-Fender | jblack, Like the spinster who wanted on her tombstone "Born a virgin, lived a virgin, died a virgin". Instead the Tombston engraver (having been an ex-post office worker) decided to abbreviate as "Returned unopened" :p |
| 06:17.16 | [TK]D-Fender | etfonhomey, go look where your LOGIN points to <--- |
| 06:17.26 | jblack | lol. I like that one! |
| 06:17.52 | etfonhomey | Doh! |
| 06:18.24 | etfonhomey | I'm surprised you had that much patience... |
| 06:19.07 | [TK]D-Fender | ~h2so4 |
| 06:19.08 | jbot | [~H2SO4] "John was here but is no more, for what he thought was H2O was H2SO4" |
| 06:19.15 | [TK]D-Fender | ^^^ |
| 06:21.40 | etfonhomey | [TK]D-Fender, do you need a "hint" in every context in which a SIP device could change status for presence? |
| 06:22.21 | [TK]D-Fender | etfonhomey, You need it in the context used for subscriptions (where the phone's context or "subscribecontext") points to. |
| 06:24.10 | etfonhomey | [TK]D-Fender, When would the "ringinuse=no" in queues.conf actually prevent a call from being delivered to an agent? |
| 06:25.04 | [TK]D-Fender | etfonhomey, it wouldn't. For final repetition, * has NOT IDEA what that dialplan will execute and can have no sense as to what device to VERIFY. |
| 06:25.27 | [TK]D-Fender | etfonhomey, it was not meant for Agents at all. That means will not work. Period. |
| 06:25.50 | etfonhomey | [TK]D-Fender, so what is the purpose of the option? |
| 06:25.53 | [TK]D-Fender | etfonhomey, Dr. Phil and Oprah are available for bookings if you are having trouble coping :) |
| 06:26.02 | [TK]D-Fender | etfonhomey, for fixed devices obviously. |
| 06:26.12 | etfonhomey | [TK]D-Fender, such as? |
| 06:26.22 | [TK]D-Fender | etfonhomey, member => sip/100 <- yippy-kai-yay |
| 06:26.48 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 06:26.59 | jblack | Heh. A guy in montreal sold the snowbank in front of his house $3550CDN. That's worth something like $300k US these days. |
| 06:27.04 | etfonhomey | [TK]D-Fender, gotcha, now I understand what you mean by fixed. |
| 06:27.40 | [TK]D-Fender | jblack, nope, dollar is pretty much par currently... and how did he sell a snowback for that? |
| 06:27.58 | [TK]D-Fender | jblack, I've got a tone fo the white shit outside.... I could be making a killing! |
| 06:28.08 | jblack | I was being sarcastic. On ebay, as a joke, http://www.edmontonsun.com/News/Canada/2007/12/30/4746067.html |
| 06:28.14 | jblack | Thought you'd get a kick out of it |
| 06:28.49 | etfonhomey | [TK]D-Fender, in call queues, do fixed devices still need to login? |
| 06:29.35 | [TK]D-Fender | etfonhomey, no they don't |
| 06:30.18 | [TK]D-Fender | etfonhomey, Go read up on the new options with 1.4 for "addqueuemember", etc. THESE will allow you to do the sort of stuff you want. |
| 06:30.26 | [TK]D-Fender | jblack, interesting. |
| 06:30.48 | dacs | [TK]D-Fender: do you ever sleep man?:) |
| 06:30.50 | [TK]D-Fender | jblack, Alberta : Where the land is so flat you can see your dog running away from you for DAYS <- |
| 06:31.04 | [TK]D-Fender | dacs, yup, heading out momentarily |
| 06:31.22 | *** join/#asterisk tengulre (n=tengulre@124.42.50.54) |
| 06:31.39 | jblack | I've never made it up to alberta. Just Montreal and Vancouver |
| 06:31.42 | dacs | chees & crackers |
| 06:32.00 | [TK]D-Fender | jblack, Next time you're up here we'll grab a beer |
| 06:32.22 | jblack | I'll take you up on that sometime. |
| 06:32.23 | etfonhomey | [TK]D-Fender, thanks for your help AGAIN. |
| 06:32.28 | dacs | [TK]D-Fender: dream that you are helping me nicely from now on |
| 06:32.29 | dacs | lol |
| 06:32.47 | dacs | [TK]D-Fender: have a wonderful night |
| 06:32.57 | jblack | dacs: I haven't been here much yet, but everyone I've ever seen him lart deserved it. |
| 06:33.35 | jblack | I think he's pretty patient for a guy that keeps teaching people how not to stick forks into toasters |
| 06:33.47 | [TK]D-Fender | jblack, And the rest only delusionally believe they were larted in the first place. over-sensitive bunch! |
| 06:33.48 | dacs | <PROTECTED> |
| 06:34.34 | [TK]D-Fender | jblack, keep the change ;) |
| 06:34.43 | jblack | speaking of forks and toasters, I still have a little smoke coming out of mine. Nothing that can't hold. |
| 06:35.00 | etfonhomey | jblack, out of your fork or your toaster? |
| 06:35.22 | jblack | Actually, a more correct analogy is that after having spent so much time poking a fork into my toaster, my toaster now no longer browns both sides of the bread. |
| 06:36.00 | jblack | et: Neither one nor the other. |
| 06:36.18 | jblack | Think you might be able to lend me a hand? It could be hard to find. |
| 06:36.52 | jblack | Or incredibly simple. It's either obvious, or the result of something I did when I started off that seemed smart, but was incredibly stupid. |
| 06:37.04 | etfonhomey | [TK]D-Fender and jblack, good night! |
| 06:37.06 | jblack | Thus breaking assumptions that you may have. |
| 06:37.13 | jblack | Or not. sleep well et. |
| 06:37.37 | [TK]D-Fender | jblack, todays final deep thought : If a buttered piece of toast always lands butter-side down, and a cat always lands on its feet : What happens when you push a cat with a piece of buttered toast tied to its back off the counter? :) |
| 06:38.24 | jblack | Would you believe that I have personal observance in the answer to that question? |
| 06:38.58 | [TK]D-Fender | jblack, jsut food for thought.... the toast... NOT the cat :) |
| 06:38.59 | jblack | When you get down to it, it depends on how hard you push the cat at the floor. |
| 06:39.11 | [TK]D-Fender | Alas I am off. Later all. |
| 06:39.11 | jblack | Darn, cause I heard cat tastes like chicken |
| 06:39.15 | jblack | sleep well. |
| 06:39.43 | jblack | I'm still having trouble with blind transfers. Anyone care to lend a hand? |
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| 07:04.19 | jblack | is gsm known by another name as well? I can't seem to find it in x-lite's configuration |
| 07:12.08 | jblack | Ok. I think I know what's going wrong. |
| 07:12.44 | jblack | For my blind transfers. I think that the calls are at different protocols, and * is refusing to do the transfer |
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| 08:09.36 | dacs | all sleep |
| 08:09.41 | dacs | ? |
| 08:12.27 | tzafrir_laptop | sure |
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| 08:23.55 | dacs | tzafrir_laptop: can you take a look at my sip show peer and tell me why i can't recive a call please |
| 08:24.32 | tzafrir_laptop | 'sip show peers' is generlly lees relevant than 'sip show users' |
| 08:24.56 | dacs | ~paste |
| 08:24.57 | jbot | hmm... paste is http://rafb.net/paste/ |
| 08:26.22 | *** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.wa.comcast.net) |
| 08:26.27 | tzafrir_laptop | note that 'sip show users' contains the passwords of your users |
| 08:26.30 | obnauticus | anyone here good with chan_mobile? |
| 08:26.49 | tzafrir_laptop | ~anyone |
| 08:26.50 | jbot | *** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do. |
| 08:27.06 | obnauticus | okay tzafrir_laptop who should I highlight then? |
| 08:27.13 | dacs | tzafrir_laptop: http://rafb.net/p/NP9JB167.html |
| 08:27.33 | obnauticus | Unless you can come up with a solution i think that saying that asking `anyone' is fine :| |
| 08:27.50 | tzafrir_laptop | dacs, the point is: take a look atthe output there. Is the username / password correct? |
| 08:28.13 | tzafrir_laptop | also: if you enable sip debug, do you see something happening? |
| 08:28.44 | tzafrir_laptop | obnauticus, the point is that you haven't actually asked |
| 08:29.05 | obnauticus | Well with my previous attempts in this channel it seems not many people use chan_mobile. |
| 08:29.11 | *** join/#asterisk Micc (n=mic@c-67-168-123-42.hsd1.wa.comcast.net) |
| 08:29.25 | dacs | tzafrir_laptop: i just enabled sip debug |
| 08:29.41 | obnauticus | but the whole question with details provided are in this paste: http://rafb.net/p/jZ0zX145.html |
| 08:29.47 | dacs | where can i see if it is doing anything , just try to call |
| 08:30.30 | tzafrir_laptop | just call, yes |
| 08:30.44 | tzafrir_laptop | If anything happens, you'll see lots of junk |
| 08:30.56 | tzafrir_laptop | (and probably won't be able to make sense of it) |
| 08:31.18 | dacs | nothing is happening, not even my phone is not ringing |
| 08:31.33 | tzafrir_laptop | Bhave you tried to call? |
| 08:31.47 | dacs | yes |
| 08:32.00 | tzafrir_laptop | If you don't see anything with sip debug on, nothing got to Asterisk |
| 08:32.25 | tzafrir_laptop | Either you call the arong (IP) address, or there's a firewall in your way |
| 08:32.48 | tzafrir_laptop | s/arong/wrong/ |
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| 08:34.34 | tzafrir_laptop | s/er/&123/ |
| 08:35.39 | Frek818 | clear |
| 08:35.50 | tzafrir_laptop | s/er/$&123/ |
| 08:36.03 | tzafrir_laptop | no luck |
| 08:37.04 | Frek818 | Hella |
| 08:37.15 | Frek818 | s/Hella/Hello/ |
| 08:38.23 | Frek818 | tzafrir_laptop: What problem are you having? |
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| 08:42.03 | tzafrir_laptop | Wanted to check some extra capabilities |
| 08:42.53 | tzafrir_laptop | s/some [a-z]*/a few/ |
| 08:44.10 | dacs | hell |
| 08:44.21 | dacs | s/hell/help/ |
| 08:44.27 | dacs | :) |
| 08:44.53 | tzafrir_laptop | well, even ed could do better |
| 08:45.34 | tzafrir_laptop | ~ed |
| 08:45.35 | jbot | methinks ed is the standard UNIX line editor. It underlies vi, and is closely related to sed. Unlike most editors, works even without cursor addressing, and can be quickly learned. |
| 08:46.46 | tzafrir_laptop | d$ |
| 08:48.07 | Nugget | vi is an editor that has two modes: one which beeps at you and one which doesn't. |
| 08:51.46 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 08:52.54 | dacs | tzafrir_laptop: can you check you pm |
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| 08:54.26 | tzafrir_laptop | dacs, are you sure you set the IP address right? |
| 08:54.52 | tzafrir_laptop | next thing, run a sniffer (e.g: tcpdump) on the asterisk server |
| 08:56.21 | dacs | yes |
| 08:58.16 | tzafrir_laptop | so do you see any traffic from your ATA when you call? |
| 08:58.24 | dacs | no |
| 08:59.06 | tzafrir_laptop | Check the IP address / port on your ATA |
| 09:02.45 | tzafrir_laptop | are both Asterisk and the ATA on the same LAN? |
| 09:02.57 | tzafrir_laptop | Is there any firewall between them? |
| 09:03.50 | dacs | yes to all except the lat question. |
| 09:04.00 | dacs | s/lat/last/ |
| 09:04.34 | tzafrir_laptop | Any chance it filters SIP transport? |
| 09:05.07 | dacs | i don't think so |
| 09:05.31 | tzafrir_laptop | Can you check that with a computer? |
| 09:05.50 | dacs | what do you want me to do |
| 09:06.23 | tzafrir_laptop | echo test | nc -u asteriskserver 5060 |
| 09:06.42 | tzafrir_laptop | run that from a computer behind the same firewall |
| 09:07.06 | tzafrir_laptop | and check if you see that in tcpdump on your Asterisk |
| 09:07.23 | tzafrir_laptop | tcpdump 'udp port 5060' |
| 09:07.38 | tzafrir_laptop | or maybe: tcpdump -n 'udp port 5060' |
| 09:07.52 | tzafrir_laptop | And make sure you run it on the correct network interface |
| 09:09.47 | *** join/#asterisk kbrooks (n=kbrooks@d235-130-238.home1.cgocable.net) |
| 09:11.53 | dacs | echo test | nc -u asteriskserver 5060 is there is a same command as this for windows |
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| 09:21.11 | R1ck | anyone know if siemens phones work well with Asterisk? |
| 09:21.25 | mtryfoss | gigaset wireless ? |
| 09:21.48 | R1ck | not wireless, just the gigaset s450ip |
| 09:22.11 | dacs | ~anyone |
| 09:22.11 | jbot | *** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do. |
| 09:22.36 | mtryfoss | yes, it works fine |
| 09:22.44 | R1ck | cool thanks |
| 09:23.19 | mtryfoss | however, stay away from the version with integrated router |
| 09:35.35 | dacs | good night all |
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| 09:43.28 | hi365 | im trying to write: if extension is 900-998 go to qeng. is this correct? |
| 09:43.28 | hi365 | Gosubif($["${AMPUSER}" >= 900] & $["${AMPUSER}" <= 998]]?qeng) |
| 09:46.35 | hi365 | for some reason the first cindition is alwyas false (0) |
| 10:08.17 | hi365 | how in the world is this true???!!! |
| 10:08.22 | hi365 | GosubIf("Local/*46@from-internal-ec2b,2", ""900" = "999"]?qop|1") |
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| 10:13.23 | tzafrir_laptop | = ? |
| 10:19.35 | *** join/#asterisk dickyjoe (n=richardl@124.149.64.142) |
| 10:19.39 | dickyjoe | Hello all |
| 10:20.02 | dickyjoe | Can someone help me with ENUM lookups on asterisk 1.4.16.2 on a centos machine |
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| 10:25.43 | dickyjoe | anyone home? |
| 10:37.56 | *** join/#asterisk stony (n=stony@p57B38701.dip0.t-ipconnect.de) |
| 10:37.59 | stony | hi |
| 10:38.15 | stony | is it possible to readjust the volume of the voip-connection in asterisk ? |
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| 11:23.25 | badcfe | when i include a context in another, may i have a timeout extension in the included context? |
| 11:23.56 | badcfe | will this timeout be treated in scope as if in the included context itself? |
| 11:26.07 | tzafrir_laptop | basically, yes |
| 11:26.30 | tzafrir_laptop | if there was already a 't' extension there, things might get messy |
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| 12:42.23 | Winkie | is there anything cheaper than the IAXy when trying to get a single FXS port? it doesn't have to be a standalone device |
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| 12:54.28 | Sniper_linux | Hello All |
| 12:54.45 | Sniper_linux | I need to ask about earlier media when making a VOIP call |
| 12:55.00 | Sniper_linux | Someone has any idea about earlier media? |
| 12:57.07 | JT | an idea would be to ask the question :) |
| 12:59.50 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 13:00.06 | tzafrir_laptop | Winkie: I saw one made by "x100p.com" . There's also one made by atcom |
| 13:00.48 | *** join/#asterisk obnauticus (n=obnautic@c-24-22-14-101.hsd1.wa.comcast.net) |
| 13:00.55 | obnauticus | !seen dseeb_ |
| 13:01.00 | obnauticus | :( |
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| 13:29.18 | zeeesh | anybody there ... i m getting problem with sip call .. i m just getting one way trafic .. a receiver can just hear my voice .. but i m unable to hear his voice .. one way trafice .. how to troubleshoot ???? |
| 13:29.34 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 13:34.14 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 13:35.26 | fiXXXerMet | I setup asterisk at my home and had 2 IP phones working (they were registered and could call each other). I've setup port forwarding on my firewall (opened up 5060, 4569 and 10001-20000 and forwarded them to my asterisk box) at home, brought the phones to work, and now they aren't registered. Anything else that I need to open up? |
| 13:35.38 | shido6 | :) |
| 13:35.56 | shido6 | zeeesh its either a router or firewall issue |
| 13:35.59 | Qwell | 5060 udp? |
| 13:36.12 | fiXXXerMet | All UDP, yse |
| 13:36.43 | Qwell | fiXXXerMet: are the phones trying to connect to an internal IP? |
| 13:36.47 | shido6 | fiXXXerMet: this is a new network you have to open up the same ports at work |
| 13:36.53 | fiXXXerMet | Qwell: Nope. |
| 13:37.08 | fiXXXerMet | shido6: So I need to setup rules to the firewall here, as well? |
| 13:37.10 | Qwell | port forwards inside a LAN don't work if you use an external address |
| 13:37.14 | shido6 | indeed |
| 13:37.24 | fiXXXerMet | Qwell: What do you mean? |
| 13:37.32 | Qwell | ie; 192.168.1.2 connecting to 64.4.4.4 which forwards to 192.168.1.1 |
| 13:37.35 | Qwell | will not work |
| 13:37.51 | zeeesh | <shido6>: i m doing peer calling .. so route issue is finised .. within my place .. i hv too different peers in 2 diffferent pc .. its working fine .. but the other peer is registered from another country .. he can hear my voice .. but i can't ... ? |
| 13:38.06 | fiXXXerMet | I have the phones connecting to a hostname, which resolves to a public address, which goes to my firewall, which goes to my server? |
| 13:38.20 | Qwell | oh, the asterisk box is at home |
| 13:38.31 | fiXXXerMet | Yup |
| 13:38.54 | shido6 | if you have one way audio an RTP port is being blocked or not going through a firewall or router |
| 13:38.54 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
| 13:39.01 | fiXXXerMet | shido6: What ports do I need to forward at work? |
| 13:39.16 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 13:39.17 | shido6 | how many phones are at work using SIP? |
| 13:39.24 | fiXXXerMet | I have just 2 now. |
| 13:39.45 | fiXXXerMet | We'll have 50+ when everything is deployed, but right now I awnt these two to call each other, through my box at home, while at work |
| 13:40.11 | shido6 | frankly I would use 5060 for one phone and 5061(both UDP ) for the other phone then 10001 - 20000 UDP and make a quick change in one of the phones to use 5061 instead of 5060 |
| 13:40.15 | shido6 | :) |
| 13:40.37 | fiXXXerMet | Hmmm |
| 13:40.47 | shido6 | then check rtp.conf to make sure you you're using that port range for RTP |
| 13:41.33 | fiXXXerMet | That's right |
| 13:41.53 | fiXXXerMet | So for the work firewall, I need to open 5060 and 5061 from the asterisk IP to the phones IP? |
| 13:42.08 | shido6 | just do 5060 UDP for now |
| 13:42.23 | fiXXXerMet | ok |
| 13:44.40 | fiXXXerMet | These SonicWall firewalls are such a pita |
| 13:44.40 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
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| 13:45.55 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:50.54 | fiXXXerMet | Registration State:Failed |
| 13:51.52 | fiXXXerMet | I don't see anything in /var/log/asterisk |
| 13:54.21 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:54.21 | FlatFoot | afternoon all |
| 13:54.25 | fiXXXerMet | hai |
| 13:54.33 | FlatFoot | anyone had much to do with ... |
| 13:54.35 | FlatFoot | <PROTECTED> |
| 14:01.14 | shido6 | back |
| 14:02.22 | shido6 | whats up FlatFoot? |
| 14:03.00 | FlatFoot | shido6: not a lot just trying to find if anyone had much exp with this unit before i buy one |
| 14:03.27 | shido6 | theres some info on the other unit |
| 14:03.41 | shido6 | err |
| 14:03.43 | shido6 | the 2n Ateus |
| 14:03.45 | FlatFoot | what the MV-372 ? |
| 14:04.04 | shido6 | voiceblue, stargate, bluestar, bluetower, etc |
| 14:04.14 | *** join/#asterisk brpvieira (n=bernardo@c9118288.static.bhz.virtua.com.br) |
| 14:04.45 | FlatFoot | was just wondering about call quality |
| 14:04.55 | FlatFoot | wether it is any good basically |
| 14:09.12 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:11.50 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:16.06 | fiXXXerMet | shido6: My firewall has 5060 udp open, and the work firewall has everything from lan, going out, open. Still, my phone won't register? |
| 14:16.14 | fiXXXerMet | Nothing in the logs either except "failed" from the phone. |
| 14:16.32 | *** join/#asterisk ddunavant (n=David@66.170.97.28) |
| 14:16.33 | shido6 | going out is nice |
| 14:16.37 | shido6 | but what about going IN |
| 14:17.09 | shido6 | 5060 is forwarded to what ip on the router? |
| 14:17.13 | fiXXXerMet | Well, isn't the connection originating from inside? |
| 14:17.21 | shido6 | it should be forwarded to one of your phones |
| 14:17.27 | fiXXXerMet | ok |
| 14:17.37 | *** join/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177562067.dsl.bell.ca) |
| 14:18.30 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:19.20 | fiXXXerMet | shido6: I have 5060 going from wan to lan, and lan to wan, all open |
| 14:19.24 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 14:19.31 | fiXXXerMet | Do I only need 5060 udp? |
| 14:20.51 | tzafrir_laptop | if you upgrade sox on debian to 14, be sure to install libsox-fmt-all |
| 14:21.07 | tzafrir_laptop | otherwise you might be srprised that sox doesn't know of the format gsm |
| 14:22.14 | [TK]D-Fender | fiXXXerMet: YOU NEED FAR MORE FOR nat SETUP. rEAD THIS NOW : |
| 14:22.16 | [TK]D-Fender | ~SIPNAT |
| 14:22.16 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:22.25 | [TK]D-Fender | darn caps.... |
| 14:24.14 | *** join/#asterisk mog (n=mog@nat/digium/x-e5ce4ae83d11b6e4) |
| 14:24.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:24.42 | *** join/#asterisk VijayG (i=VijayG@58.68.47.115) |
| 14:30.07 | lilalinux | is it possible to administrate the phonebook of the Siemens Gigaset SL75 WLAN sip phone? |
| 14:30.11 | lilalinux | via asterisk |
| 14:33.05 | [TK]D-Fender | lilalinux: Well * can be used to trigger a SCRIPT that can do something perhaps. |
| 14:33.29 | [TK]D-Fender | lilalinux: but directly, no. There is no miracle "phone directory protocol". |
| 14:33.56 | [TK]D-Fender | lilalinux: So ask yourself what ways your phone can get its directory information from |
| 14:34.12 | *** join/#asterisk suvir (n=chatzill@ppp-124.120.141.180.revip2.asianet.co.th) |
| 14:34.48 | *** join/#asterisk glen2 (n=glen@62.100.170.6) |
| 14:35.10 | mvanbaak | anyone knows of a SIP softphone for the Playstation portable ? |
| 14:40.34 | R1ck | how do I set an ISDN MSN number on outgoing calls for a zap device? |
| 14:41.12 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-ba471829ddb094f1) |
| 14:41.20 | fujin | mvanbaak: it doesn't have a microphone |
| 14:41.45 | mvanbaak | fujin: the lite+slim can handle a headset |
| 14:41.54 | fujin | ah. |
| 14:42.02 | fujin | I've got an oldschool PSP. originally was 1.5 |
| 14:42.06 | mvanbaak | ah |
| 14:42.09 | fujin | It's a little more piratey now. |
| 14:42.12 | mvanbaak | yeah, that one cannot handle it |
| 14:42.30 | mvanbaak | I bought a lite+slim one today |
| 14:42.36 | mvanbaak | for my wife |
| 14:42.47 | mvanbaak | but would be cool to have a softphone on it for me ;) |
| 14:43.43 | *** join/#asterisk docelmo (n=vircuser@c-68-32-135-157.hsd1.de.comcast.net) |
| 14:43.56 | *** join/#asterisk ming_zym (n=ming_zym@124.14.237.85) |
| 14:56.57 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 15:00.25 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-186.static.twtelecom.net) |
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| 15:01.28 | *** join/#asterisk PasDNick (n=tonio@LPuteaux-151-43-27-85.w193-252.abo.wanadoo.fr) |
| 15:02.38 | Neil_L | Hi everyone, I need to record all the incoming and outgoing phone calls at a call center and wondered if there was a good front end already written for asterisk? |
| 15:02.47 | fiXXXerMet | [TK]D-Fender: I have sip.conf setup right (I think - I followed the links you sent me) and I have port forwarding setup, but my phone is still failing to register. |
| 15:03.01 | [TK]D-Fender | fiXXXerMet: PASTEBIN is your friend. |
| 15:03.03 | [TK]D-Fender | ~pb |
| 15:03.03 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:03.12 | fiXXXerMet | What do you want me to paste? |
| 15:03.21 | [TK]D-Fender | fiXXXerMet: your sip.conf masking only passwords. |
| 15:03.42 | [TK]D-Fender | fiXXXerMet: then if that checks out we'll move on to SIP debug analysis |
| 15:03.46 | fiXXXerMet | ok. |
| 15:05.17 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:05.46 | *** join/#asterisk sholden (n=sholden@adsl-070-155-153-142.sip.btr.bellsouth.net) |
| 15:06.25 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 15:06.34 | ZaVoid | morning guys |
| 15:07.46 | fors1 | Hi! My norwegian employer is opening a new office in Mountain View, CA. I've been given the task to get internet and phone connection up and running. I want a integrated T1 for that office. Problem is finding a provider. I've tried AT&T, but they couldn't offer integrated T1 in that area. Suggestions? |
| 15:08.23 | *** part/#asterisk sholden (n=sholden@adsl-070-155-153-142.sip.btr.bellsouth.net) |
| 15:08.57 | ZaVoid | fors1 did you try like bandwidth.com to find a provider? |
| 15:09.18 | ZaVoid | http://www.bandwidth.com/ |
| 15:09.27 | fiXXXerMet | [TK]D-Fender: http://pastebin.com/d55064632 |
| 15:10.07 | fors1 | ZaVoid: no, I haven't seen this one. I tried an other service like this, but didn't get any quotes back. I'll give this a try. Thanks :) |
| 15:10.13 | [TK]D-Fender | fiXXXerMet: please permanently remove all commented lines and pastebin again. |
| 15:10.16 | ZaVoid | yep |
| 15:10.26 | ZaVoid | hey [TK]D-Fender how was your holiday? |
| 15:10.35 | R1ck | anyone know anything about zapata with MSN numbers? |
| 15:10.39 | [TK]D-Fender | ZaVoid: All too short.... |
| 15:10.44 | ZaVoid | i hear ya |
| 15:10.54 | fiXXXerMet | [TK]D-Fender: I did with grep -v ';' but since there are comments on 'uncommented' lines, do you know how to do that? |
| 15:11.26 | [TK]D-Fender | fiXXXerMet: do it line by line if you have to. Jsut get rid of the useless filler. |
| 15:16.51 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:17.10 | hi365 | file: ping |
| 15:17.50 | file | hi365: pong |
| 15:19.13 | dmz | fiXXXerMet grep -v "^;" will clear out the lines that start w/comments |
| 15:19.48 | fiXXXerMet | [TK]D-Fender: http://pastebin.com/d3ec0e399 |
| 15:20.13 | hi365 | ide love to give ssh access, but im working on the box atm. (11654) will you be around in an hour or two? |
| 15:20.33 | hi365 | tahnsk |
| 15:20.35 | fiXXXerMet | dmz: Thanks :) |
| 15:20.36 | [TK]D-Fender | fiXXXerMet: looks mostly fine..and your PHONES entry? |
| 15:21.02 | dmz | np :) |
| 15:21.09 | dmz | hey anyone here use app-conference w/1.4? |
| 15:22.44 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 15:22.57 | fiXXXerMet | [TK]D-Fender: I guess I don't have one...... I used asterisk gui to create two extensions. They worked locally, though they don't work remotely (outside of the LAN) |
| 15:23.15 | fiXXXerMet | Would that be the [A] and [B] options at http://www.aocomputing.net/?p=3 ? |
| 15:23.35 | [TK]D-Fender | fiXXXerMet: is your remote phone behind a NAT of its own? |
| 15:24.20 | fiXXXerMet | yes |
| 15:25.26 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:25.32 | [TK]D-Fender | fiXXXerMet: then make sure it is set to "nat=yes" as well. Then if things still aren't working, enable sip debug, restart the phone and pastebin the failed attempt at verbose 10 |
| 15:26.25 | fiXXXerMet | [TK]D-Fender: nat=yes is already set. I'll enable debug and then get back to you. |
| 15:26.39 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-8ff806d541976255) |
| 15:26.39 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:26.45 | [TK]D-Fender | fiXXXerMet: you need it in the PHONE's entry as well, not just under [general]" |
| 15:26.53 | fiXXXerMet | ah |
| 15:26.57 | [TK]D-Fender | fiXXXerMet: * needs to know if it can trust the return info the phone is sending |
| 15:27.09 | fiXXXerMet | Could you show me an example of this phone's entry? |
| 15:28.46 | *** join/#asterisk wakku (n=eurulo@unaffiliated/wakku) |
| 15:29.25 | fiXXXerMet | [TK]D-Fender: I mean, guess they're the [A] and [B] sections as shown at http://www.aocomputing.net/?p=3 but I don't know how those get assigned to a phone. |
| 15:29.59 | [TK]D-Fender | fiXXXerMet: do sip show peer [yourphonesentrywithoutbraces]" |
| 15:30.17 | [TK]D-Fender | fiXXXerMet: and see if its set. Then do the test I told you should follow |
| 15:30.37 | Dr-Linux | i installed new version 1.4.16, all went fine, but i'm unable to install mysql driver from asterisk-addons 1.4.5. What's wrong with it? |
| 15:31.22 | mvanbaak | Dr-Linux: please pastebin the error |
| 15:32.00 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:32.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:32.10 | Dr-Linux | mvanbaak: http://phpfi.com/286392 |
| 15:32.46 | Dr-Linux | mvanbaak: it doesn't even try to bother installing mysql module :S |
| 15:33.05 | *** join/#asterisk CapRicORN^80 (n=you@207.176.6.68) |
| 15:33.26 | mvanbaak | instead of 'make install' do a 'make menuselect', deselect the chan_ooh323, save, run ./configure and run make install |
| 15:33.50 | De_Mon | I just had a user report that the person he called couldn't hear him 10min into a call. |
| 15:34.01 | mvanbaak | the error is not with the mysql driver, it's abouth the chan_oh323 driver |
| 15:34.26 | mvanbaak | De_Mon: what tech ? |
| 15:34.39 | mvanbaak | sip/iax/zap/h323/skinny/etc ? |
| 15:34.40 | De_Mon | it was a SIP (user) <-> Asterisk <-> SIP itsp (called party) |
| 15:35.04 | mvanbaak | any NAT in the game ? |
| 15:35.25 | De_Mon | yeah, but the call was working fine the first 10min |
| 15:35.27 | Dr-Linux | mvanbaak: cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory << is not a problem i can fix it, but i don't really need it, i need mysql drvier |
| 15:35.42 | mvanbaak | Dr-Linux: so do as I say |
| 15:35.45 | file | make menuselect, unselect chan_ooh323 |
| 15:35.48 | mvanbaak | 16:33 < mvanbaak> instead of 'make install' do a 'make menuselect', deselect the chan_ooh323, save, run ./configure and run make install |
| 15:36.02 | De_Mon | mvanbaak didn't you say that in -dev? |
| 15:36.05 | mvanbaak | De_Mon: maybe the firewall is expiring the NAT state ? |
| 15:36.14 | mvanbaak | De_Mon: no, I said it here |
| 15:36.22 | Dr-Linux | i see |
| 15:36.25 | Dr-Linux | hold on |
| 15:37.38 | *** join/#asterisk Defraz (n=tim@fw.fuzecore.com) |
| 15:37.49 | fiXXXerMet | [TK]D-Fender: http://pastebin.com/m296e8cba Those lines repeat a number of times |
| 15:37.54 | De_Mon | oh, nm that question came up in dev yesterday |
| 15:38.11 | Dr-Linux | [root@i2c-RHEL-PBX1 asterisk-addons-1.4.5]# make menuselect |
| 15:38.11 | Dr-Linux | make: *** No rule to make target `makeopts', needed by `menuselect/menuselect'. Stop. |
| 15:38.16 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 15:38.38 | De_Mon | mvanbaak hrrrm you might be on to something there. |
| 15:38.38 | mvanbaak | run: ./configure |
| 15:38.41 | [TK]D-Fender | fiXXXerMet: SIP/2.0 401 Unauthorized <- bad user/pass |
| 15:38.45 | Dr-Linux | mvanbaak: ok |
| 15:39.32 | mvanbaak | Dr-Linux: after that run: make menuselect |
| 15:40.22 | fiXXXerMet | [TK]D-Fender: The password hasn't changed since it worked internally, though I change reset it and it still fails. What about the "Transmitting (no NAT) to 192.168.0.66:5060" in the logs? |
| 15:40.24 | Dr-Linux | i got a new page |
| 15:40.27 | fiXXXerMet | "no NAT"? |
| 15:40.29 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 15:40.31 | [TK]D-Fender | fiXXXerMet: Also <--- SIP read from 69.85.26.2:41691 ---> comes in from the OUTSIDE, and then "Sending to 192.168.0.66 : 5060 (no NAT)" is sends to the INTERNAL IP the phone seems to provide. it sis NOT set properly |
| 15:40.34 | mvanbaak | De_Mon: I got that once |
| 15:40.38 | Dr-Linux | mvanbaak: how i can deselect chan_ohh323? |
| 15:41.12 | [TK]D-Fender | fiXXXerMet: Yes, that is also bad. so your phone's auth is wrong and * was not told it was behind NAT. As those settings are buried in users.conf thatnks to the GUI I cannot help you there. |
| 15:41.27 | badcfe | |is MacroExit() the way to return from a macro? |
| 15:41.33 | mvanbaak | Dr-Linux: go to the item and hit the spacebar |
| 15:41.58 | mvanbaak | Dr-Linux: after that, hit the s key to save the changes |
| 15:42.06 | De_Mon | mvanbaak is there some sort of setting that would control that sort of NAT behavior? |
| 15:42.13 | De_Mon | in the firewall |
| 15:42.15 | badcfe | or will a macro "return" by itself once theres no more extentions in it? |
| 15:42.36 | badcfe | i wander if an explisit MacroExit should be done or if its not needed or not good to do it |
| 15:42.43 | mvanbaak | De_Mon: I have no idea. I fixed it by adding some static redirections. |
| 15:42.44 | Dr-Linux | mvanbaak: i've only this > http://phpfi.com/286549 |
| 15:42.47 | Dr-Linux | nothing else |
| 15:42.49 | mvanbaak | De_Mon: not all setups can do that |
| 15:43.11 | mvanbaak | Dr-Linux: what does it read on top ? |
| 15:43.31 | mvanbaak | 'Press 'h' for help.' |
| 15:43.36 | mvanbaak | ok |
| 15:43.47 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:43.52 | mvanbaak | the arrows show you what option is highlighted |
| 15:43.59 | mvanbaak | with <Enter> you can select it |
| 15:44.03 | Dr-Linux | mvanbaak: i've all this |
| 15:44.04 | mvanbaak | so do that in this case |
| 15:44.07 | [TK]D-Fender | badcfe: is the instruction page for its use not clear enough? |
| 15:44.10 | Dr-Linux | yeah got it |
| 15:44.24 | mvanbaak | Dr-Linux: there you can select h323 and deselect it with the spacebar |
| 15:44.47 | mvanbaak | if that's done, hit the <- key on your keyboard to return to this welcome screen |
| 15:44.55 | mvanbaak | then hit the s key to save and exit |
| 15:45.00 | Dr-Linux | yeah |
| 15:45.04 | fiXXXerMet | [TK]D-Fender: I set NAT for both phones in users.conf and they both now have a dialtone and are registered. So, hurah to that. However, when I try to dial each other's extension, I get "Call from '' to extension '6001' rejected because extension not found." |
| 15:45.08 | Dr-Linux | mvanbaak: i got this after hitting enter >> http://phpfi.com/286550 |
| 15:45.31 | [TK]D-Fender | fiXXXerMet: dialplan errors. go debug. The messages are pretty blatant. |
| 15:45.42 | mvanbaak | Dr-Linux: yeah, so navigate to number 4 with the down arrow key |
| 15:45.48 | mvanbaak | ah |
| 15:45.58 | Dr-Linux | what's "XXX" ? |
| 15:45.59 | mvanbaak | and you dont have libmysqlclient-dev installed |
| 15:46.03 | [TK]D-Fender | fiXXXerMet: And keep in mind the GUI plays games with your settings... this isn't a support channel if things get messy... |
| 15:46.08 | mvanbaak | that's why you cannot select the mysql stuff |
| 15:46.27 | *** join/#asterisk reber (i=remi@san13-2-82-244-36-122.fbx.proxad.net) |
| 15:46.31 | [TK]D-Fender | Dr-Linux: 90% of your Inbox :) |
| 15:46.41 | mvanbaak | lol [TK]D-Fender |
| 15:46.59 | mvanbaak | the biggest part of your Movies/ directory ;) |
| 15:47.12 | mvanbaak | ok, I have to run |
| 15:47.13 | mvanbaak | latero |
| 15:47.18 | [TK]D-Fender | Dr-Linux: the other 10% are credit-fix scams to help clear the debt from your pr0n overexpenditures :p |
| 15:47.38 | Dr-Linux | [TK]D-Fender: Happy new year, |
| 15:47.50 | [TK]D-Fender | Dr-Linux: thanks.... I'm going to need it. |
| 15:47.53 | Dr-Linux | [TK]D-Fender: i'm happy to see you cheering for the first time |
| 15:47.58 | Dr-Linux | :) |
| 15:50.52 | badcfe | [TK]D-Fender: hmm, seems that since i have autofallthrough=no i should always do an explisit MacroExit |
| 15:51.15 | [TK]D-Fender | badcfe: no need |
| 15:51.39 | [TK]D-Fender | badcfe: autofallthrough should have no impact on macros |
| 15:59.26 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 16:01.53 | *** join/#asterisk prophety (i=prophety@bas6-montreal28-1177927016.dsl.bell.ca) |
| 16:02.39 | jblack | Hello, world |
| 16:02.40 | *** join/#asterisk scurb (n=scurb@c-13afe355.104-15-64736c13.cust.bredbandsbolaget.se) |
| 16:03.28 | [TK]D-Fender | jblack: mornin' |
| 16:04.57 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:04.57 | jblack | I think I got a hint about what's wrong with transferring on my system. |
| 16:04.57 | jblack | Not everything is using the same protocol. |
| 16:05.41 | asteriskmonkey | anyone doing voicemail storage on an nfs mount with 1.4 and 1.2? |
| 16:05.42 | [TK]D-Fender | jblack: Shouldn't be an issue. |
| 16:05.54 | [TK]D-Fender | jblack: pastebin some stuff up for us to examine |
| 16:05.58 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.6) |
| 16:06.04 | jblack | Oh, good, because I don't think I can fix that. |
| 16:06.31 | jblack | Sure, I'll pastebin some stuff, but according to the dialplan, the lines are transferring. Just as that happens, both ends say the call is cleared by remote user |
| 16:06.39 | jblack | dialplan debug, that is |
| 16:07.34 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 16:09.06 | jblack | Do I have to enable transfer explicitely anywhere else, other than t and T in Dial() ? |
| 16:10.11 | De_Mon | it has to be mapped in features.conf |
| 16:10.26 | jblack | Thats' interesting |
| 16:11.10 | prophety | hi everyone, i have a T1 and i would like to reroute or transfer my sip outgoing calls (from the callback) to my sip incoming calls channels |
| 16:11.18 | jblack | I never explicitely said that # maps to blind transfer in features.conf, but the chick says "Transfer" when I press # |
| 16:11.19 | prophety | is it possible ? |
| 16:11.42 | De_Mon | jblack it may be the default setting |
| 16:12.37 | [TK]D-Fender | jblack: You only need "tT'when you are on a phone that doesn't support its OWN transfers (like shit phones like X-Lite) |
| 16:13.12 | [TK]D-Fender | prophety: you can send your calls any which way you want. |
| 16:13.46 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 16:14.24 | *** part/#asterisk harpal (n=Harpal@124.125.79.212) |
| 16:15.58 | jblack | I plan on buying shit phones. |
| 16:16.31 | jblack | Ok, here's a log from 2 calls as seen by dialplan |
| 16:16.33 | jblack | http://pastebin.com/d6287dd7a |
| 16:16.58 | jblack | The first one is the broken transfer, the second one (seperated by a dozen newlines) is calling 1001 directly, showing the intended result |
| 16:18.46 | jblack | re shit phones.. I ordered a http://www.voipsupply.com/product_info.php?products_id=2127 yesterday, with the intent of getting cordless phones. |
| 16:19.23 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:20.39 | [TK]D-Fender | jblack: Oh wait.. this is where it was dropping the call even though it looked like it should be fine, right? |
| 16:21.01 | [TK]D-Fender | jblack: EW, GrandSuck |
| 16:21.14 | [TK]D-Fender | jblack: What are you testing with? |
| 16:21.25 | De_Mon | jblack btw... http://www.telephonydepot.com/product_p/105-056-4004.htm |
| 16:23.54 | jblack | Yeah, same thing. |
| 16:24.09 | R1ck | anyone know the Syntax required by KPN (dutch telco provider) for MSN numbers? |
| 16:24.56 | jblack | I'm testing from sip:/jblackwin to sip:/jblack are both ext 1000. Calls between them generally work fine. |
| 16:24.59 | [TK]D-Fender | jblack: What phones? |
| 16:25.20 | jblack | jblack is ekiga, jblackwin is x-lite. |
| 16:25.44 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 16:26.35 | jblack | sip:/lblack goes to 1001 in the dialplan (which is off, so it's dropping to vm) |
| 16:26.49 | [TK]D-Fender | jblack: try Zoiper, it supports normal SIP transfers, and FORGET that DTMF transfer garbage. |
| 16:27.35 | jblack | You did catch that I plan on adding analog phones to the mix over the next few days? |
| 16:27.54 | Alexandre_fr | Hey folks |
| 16:28.43 | [TK]D-Fender | ~zoiper |
| 16:28.43 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 16:28.53 | jblack | Yeah, I'm downloading it now |
| 16:29.00 | Alexandre_fr | So it's better to transfer with a phone which have is own transfer than using asterisk capacity ? |
| 16:29.00 | [TK]D-Fender | jblack: Yes, with that GS bleh-box |
| 16:29.28 | [TK]D-Fender | Alexandre_fr: Its all * "capacity", just that listening in on DTNF for it is BS. |
| 16:29.32 | [TK]D-Fender | DTMF* |
| 16:30.02 | Alexandre_fr | ok, and it's not working very well ? |
| 16:30.03 | jblack | I wasn't sure what to get. Among the two suggestions, that was $115. The other suggestion was about 500 bucks. |
| 16:32.17 | [TK]D-Fender | jblack: Linksys SPA-8000 +/- $240 for 8 port |
| 16:32.44 | [TK]D-Fender | jblack: or for jsut 4 ports, you'd be better off with 2 x SPA-2102 @ $70 |
| 16:33.08 | [TK]D-Fender | Alexandre_fr: complicates your dialplan, adds issues, etc. |
| 16:33.30 | jblack | I'll try to cancel my order with grandstream. |
| 16:33.36 | [TK]D-Fender | Alexandre_fr: its useful if you NEED it. |
| 16:34.10 | Alexandre_fr | ok |
| 16:34.30 | jblack | Zoiper can do transfers. |
| 16:36.14 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:36.55 | jblack | I think I still need blind transfer to work for analog phones, won't I? |
| 16:37.51 | [TK]D-Fender | jblack: the ATA should provide that functioanlity, not * |
| 16:38.39 | [TK]D-Fender | jblack: the SPA series are quite full-featured and offer jsut about everything you can imagine implementing on an analog device. |
| 16:38.50 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:41.21 | jblack | Ok. I'll get a SPA |
| 16:41.58 | *** part/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 16:43.46 | *** join/#asterisk jobalcaen (n=joel@205.200.27.58) |
| 16:44.16 | jobalcaen | does anyone know how to update trixo using the package manager |
| 16:44.23 | Qwell | trixo? |
| 16:44.34 | jobalcaen | I know how to get there but I dont know which packages I need to get me to 2.4.0 |
| 16:44.39 | De_Mon | trixo isn't something that comes with asterisk... |
| 16:44.41 | jobalcaen | sorry..i mean trixbox |
| 16:44.54 | jobalcaen | shit |
| 16:45.29 | jobalcaen | sorry...I find it confusing sometimes |
| 16:45.50 | jobalcaen | wasnt trixbox formely known as asterisk@home |
| 16:45.58 | Qwell | asterisk@home != asterisk |
| 16:46.11 | russellb | ~trixbox |
| 16:46.12 | jbot | from memory, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 16:46.16 | jobalcaen | i'm ill informed i guess |
| 16:46.30 | jobalcaen | thanks |
| 16:46.32 | russellb | np |
| 16:48.24 | prophety | <PROTECTED> |
| 16:48.24 | prophety | <PROTECTED> |
| 16:48.24 | prophety | <PROTECTED> |
| 16:48.53 | *** join/#asterisk mikecx (n=mikecx@pool-70-104-112-56.chi.dsl-w.verizon.net) |
| 16:48.57 | prophety | most of the people told me that's not possible |
| 16:49.05 | [TK]D-Fender | prophety: "show application transfer" <---- |
| 16:49.16 | De_Mon | 11:13AM <[TK]D-Fender> prophety: you can send your calls any which way you want. |
| 16:50.21 | mvanbaak | asteriskmonkey: I do |
| 16:50.26 | mvanbaak | oops |
| 16:51.10 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:51.29 | mikecx | is there any good way to watch what's happen to my faxes and why the fax machine is not able to connect once the fax has been detected by the fax machine. I'm using zapata fxo/fxs for the fax and had it working once before |
| 16:51.36 | asteriskmonkey | mvanbaak so you do run nfs ? |
| 16:53.21 | mvanbaak | asteriskmonkey: yup |
| 16:53.55 | asteriskmonkey | have you tested what happens when the nfs mount is inaccessable in 1.4 or 1.2? |
| 16:54.43 | mvanbaak | no, because it never is ;) |
| 16:54.57 | asteriskmonkey | lol |
| 16:55.02 | mvanbaak | really |
| 16:55.06 | asteriskmonkey | ok looks like i have some extended testing to do |
| 16:55.22 | asteriskmonkey | I wanted to know if asterisk crashed if unavailable |
| 16:55.26 | mvanbaak | our NFS server is a 2-node netapp cluster |
| 16:55.49 | asteriskmonkey | yes, im running a bunch of freeBSD nas heads here :/ |
| 16:55.59 | asteriskmonkey | going into some iscsis wooo |
| 16:56.43 | mvanbaak | ah |
| 16:56.45 | CapRicORN^80 | [TK]D-Fender: hi |
| 16:57.04 | CapRicORN^80 | you were saying that i have to define 55555 in my extension . |
| 16:57.10 | [TK]D-Fender | CapRicORN^80: So... learned how to read what you did in your dialplan yesterday? |
| 16:57.14 | *** join/#asterisk didz_ (n=voce@201.19.73.107) |
| 16:57.15 | CapRicORN^80 | yes |
| 16:57.23 | CapRicORN^80 | learned for almost 3 hours |
| 16:57.31 | CapRicORN^80 | but still need more |
| 16:57.43 | [TK]D-Fender | CapRicORN^80: I was saying that you should realize that you didn't have an extren in your dialplan to handle the # you dialed. |
| 16:57.53 | jblack | Brian Hyrek, 716-250-1990 |
| 16:58.04 | CapRicORN^80 | anyways i am trying the asterisktoft book |
| 16:58.08 | CapRicORN^80 | and now trying exten => 613,1,echo(IAX2/iaxfwd/613) |
| 16:58.11 | CapRicORN^80 | sorry |
| 16:58.15 | CapRicORN^80 | this wrong one |
| 16:58.15 | [TK]D-Fender | CapRicORN^80: If you still can't see it from that context you have a real issue |
| 16:58.20 | CapRicORN^80 | and now trying exten => 613,1,Dial(IAX2/iaxfwd/613) |
| 16:58.28 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:58.53 | CapRicORN^80 | now i have configured every thing right and now i am trying 613 to test . but again getting error |
| 16:59.14 | *** join/#asterisk CrashSys (n=kumba@t1.databalance.com) |
| 16:59.32 | CrashSys | Does anyone know if there's a sip-header I can pass to a polycom before sending a call to make it NOT forward the call? |
| 16:59.33 | jblack | Just got off the phone with voipsupply.com. Those are some of the most chatty people I've ever talked to. |
| 16:59.56 | [TK]D-Fender | CapRicORN^80: Pastebin it.... |
| 17:00.01 | CapRicORN^80 | ok |
| 17:00.14 | [TK]D-Fender | CapRicORN^80: the CLI + sip debug, AND your dialplan. |
| 17:00.30 | [TK]D-Fender | CrashSys: There isn't. |
| 17:00.32 | CapRicORN^80 | ok |
| 17:00.41 | *** join/#asterisk inforx (n=inforx@S0106006097940f68.vw.shawcable.net) |
| 17:00.50 | [TK]D-Fender | CrashSys: in SIP, "the phone is king". As in each enpoint can play games to its own liking. |
| 17:00.59 | jblack | [tk]: You should have heard the call.. "About that linksys that was recommended to me.. That does use ethernet, right?" |
| 17:01.06 | CrashSys | crap... |
| 17:01.14 | jblack | After I had already ordered it. ;) |
| 17:01.18 | inforx | how can one find out what codecs remote IAX endpoint is supporting ? |
| 17:01.26 | [TK]D-Fender | CrashSys: You can I believe disable the feature COMPLETELY in provisioning IIRC |
| 17:01.32 | mvanbaak | inforx: send an invite to it |
| 17:01.41 | [TK]D-Fender | inforx: Send them an invite |
| 17:01.46 | mvanbaak | inforx: in the IAX debug you'll see what they support |
| 17:02.00 | inforx | I set iax debug, but it doesn't say |
| 17:02.06 | tzanger | hello everyone |
| 17:02.10 | inforx | how do I send them invite mvanbaak ? |
| 17:02.12 | CrashSys | d-fender: Yeah, but then they cant forward the ext to their cell phone... problem is all phones ring on incoming, and if one person words they are all SOL... |
| 17:02.25 | CrashSys | d-fender: Does the i option in 1.4 get around this? |
| 17:02.38 | [TK]D-Fender | CrashSys: Tell them to stop |
| 17:02.45 | mvanbaak | inforx: exten => something,1,Dial(IAX2/remotebox/exten-on-remote-box) |
| 17:02.52 | [TK]D-Fender | CrashSys: And make a dialplan means for forwarding. |
| 17:02.54 | CrashSys | Yeah well, i'm in a situation where a salesman sold the thing... |
| 17:02.58 | [TK]D-Fender | BRB, lunch |
| 17:03.40 | jblack | Yay. They stopped shipment on the other one. Life is good |
| 17:04.23 | CapRicORN^80 | [TK]D-Fender: http://pastebin.com/ma7d024a |
| 17:04.42 | inforx | mvanbaak, I have that already, but not sure how I see the codecs on there, PREF_CODECS Is empty when I debug and make the call |
| 17:07.49 | nDuff | CrashSys: disable on-phone forwarding completely through provisioning, and allow it to be done through your stdexten (or equivalent). |
| 17:08.26 | CrashSys | Yeah, but then I get no visual indication of forwarding... will probably just do that and tell them tough shiznit... |
| 17:08.59 | CrashSys | I guess the 'i' option in the 1.4 dial command wont prevent this... |
| 17:10.54 | file | if the 'i' option is used Asterisk will dial the device, the device will say to forward to somewhere else, Asterisk will ignore that and stop the dialing attempt to that specific device |
| 17:11.35 | CrashSys | which is what polycom does... so that's worth a shot then... |
| 17:11.49 | *** part/#asterisk jobalcaen (n=joel@205.200.27.58) |
| 17:12.15 | *** join/#asterisk NovceGuru (i=shelby@ballmung.easymac.org) |
| 17:13.10 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 17:13.46 | *** join/#asterisk nohup_ (n=nohup@crack.nohup.nl) |
| 17:13.51 | nohup_ | hello :) |
| 17:14.21 | CrashSys | I have a feature request of digium... can you make wget friendly download links? :D |
| 17:14.26 | nohup_ | is there a function i can use in extensions.conf to make asterisk send a dtmf digit to the calling party ? |
| 17:14.40 | CrashSys | nohup: senddtmf |
| 17:15.01 | nohup_ | exten => bla, n, Senddtmf(0) ? |
| 17:15.03 | nohup_ | like that ? |
| 17:15.30 | CrashSys | http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF |
| 17:15.33 | CrashSys | Yes, basically |
| 17:16.02 | nohup_ | awesome, thanks :) |
| 17:16.14 | CapRicORN^80 | [TK]D-Fender: you there ? |
| 17:18.15 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:20.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:21.24 | [TK]D-Fender | CapRicORN^80: Looking for 613 in tutorial (domain 203.175.75.32) ----SIP/2.0 404 Not Found |
| 17:21.41 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:21.46 | [TK]D-Fender | CapRicORN^80: Read the big print.... |
| 17:21.58 | CapRicORN^80 | big print ? |
| 17:22.17 | [TK]D-Fender | CrashSys: for visual notification, you can use a line-key for presence + Custom DeviceState |
| 17:22.48 | [TK]D-Fender | CapRicORN^80: the error message I pasted tells you exactly what its looking for and that its not finding it. And it SHOULDN'T |
| 17:22.54 | mvanbaak | CapRicORN^80: you have to put extension 613 inte your tutorial context |
| 17:23.14 | CapRicORN^80 | ok listen |
| 17:23.28 | CapRicORN^80 | how can i test fwd ? |
| 17:23.43 | CapRicORN^80 | i am getting voicemail messages from fwd on my email |
| 17:23.59 | [TK]D-Fender | CapRicORN^80: look at the extensions you put in your contexts vs what you are dialing. What you dialed yesterday wouldn't match anything. |
| 17:24.16 | [TK]D-Fender | CapRicORN^80: and you couldn't even tell me which line you thought was SUPPOSED to match. |
| 17:29.47 | CapRicORN^80 | [TK]D-Fender: wait |
| 17:30.06 | CapRicORN^80 | sorry i didnt made changes in sip.conf |
| 17:30.27 | [TK]D-Fender | CapRicORN^80: Go fix all of your contexts and extens. |
| 17:30.29 | CapRicORN^80 | i was using tutorial instead of internal |
| 17:30.44 | CapRicORN^80 | now i can call 613 |
| 17:32.07 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 17:33.15 | *** join/#asterisk envisean (n=envisean@wsip-64-58-162-191.oc.oc.cox.net) |
| 17:33.27 | af_ | asterisk run on linux 2.4 kernels? |
| 17:34.21 | mvanbaak | yes |
| 17:34.27 | mvanbaak | but zaptel may not |
| 17:34.46 | CapRicORN^80 | [TK]D-Fender: if i have to check fwd . i mean i want to do test call to some other user |
| 17:35.04 | CapRicORN^80 | is there any number ? |
| 17:35.14 | [TK]D-Fender | CapRicORN^80: Yes I know. your problem was never with FWD. it was always dialplan errors. |
| 17:35.33 | [TK]D-Fender | CapRicORN^80: Go fix the last of those and we'll see if you even have a problem with FWD at all. |
| 17:35.42 | [TK]D-Fender | af_: yes, and yes |
| 17:35.55 | CapRicORN^80 | i have |
| 17:36.00 | CapRicORN^80 | i can 613 now |
| 17:36.11 | CapRicORN^80 | dial * |
| 17:37.31 | rob0 | I used to use zaptel on 2.4.x, doesn't that still work? Might have to manually set device nodes, but that's the only problem I know of. |
| 17:37.51 | [TK]D-Fender | rob0: Slackware = 100% happy |
| 17:40.45 | inforx | does anyone know if I need something special for g.729a pass thru mode ? I added allow=g729 into sip.conf and set g729a as pref codec on the spa unit |
| 17:40.50 | mvanbaak | I dont like slackware |
| 17:40.55 | mvanbaak | and slackware doesn't like me |
| 17:41.31 | mvanbaak | inforx: pass thru mode should work fine without any special need |
| 17:41.41 | [TK]D-Fender | inforx: Make sure both ends of your call are using the same codec and * doesn't need to transcode (recording, etc), and you'll be fine |
| 17:44.06 | inforx | mvanbaak, I am getting chan_sip.c:5420 process_sdp: No compatible codecs, not accepting this offer! |
| 17:44.27 | inforx | it would appear its trying to transcode |
| 17:46.11 | mvanbaak | yup |
| 17:46.28 | mvanbaak | means the other side of the call does not accept g729a |
| 17:47.31 | [TK]D-Fender | inforx: This is the point where you should be pastebinning the complete CLI output of your failed call from beginning to end including SIP debug. |
| 17:47.52 | [TK]D-Fender | inforx: so we can see exactly where the mismatch is happening. |
| 17:47.55 | [TK]D-Fender | ~pb |
| 17:47.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:47.57 | [TK]D-Fender | ^^^^^^^ |
| 17:52.11 | *** join/#asterisk saftsack (n=oliver@p4FC75706.dip.t-dialin.net) |
| 17:53.18 | jblack | [tk] : You should have heard my call when I canceled the grandstream and ordered the linksys. After I finished up doing all the ordering, I said "Uh, btw, this uses ethernet, right?". |
| 17:53.33 | jblack | So, thank you for being the sort that makes no mistakes. =) |
| 17:54.07 | [TK]D-Fender | jblack: Exactly what model did you order? |
| 17:54.47 | jblack | Exactly the one you recommended, the Linksys SPA-8000 |
| 17:54.52 | *** join/#asterisk karleeto (n=karl@207.191.91.242) |
| 17:55.20 | fiXXXerMet | Any recommended softphones for linux? |
| 17:55.21 | [TK]D-Fender | jblack: Cool. |
| 17:55.27 | [TK]D-Fender | fiXXXerMet: ... |
| 17:55.30 | karleeto | hello people |
| 17:55.31 | [TK]D-Fender | ~ekiga |
| 17:55.31 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 17:55.35 | [TK]D-Fender | ~zoiper |
| 17:55.36 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 17:55.42 | [TK]D-Fender | ~linphone |
| 17:55.42 | jbot | rumour has it, linphone is a SIP VOIP phone. To configure it to use sip.handhelds.org, ask ibot about linphone config . not working with fwd.pulver.com |
| 17:56.23 | jblack | I never could get linphone to authenticate over sip to *. Thats why I ended up with ekiga |
| 17:56.29 | inforx | can I use Dial app with pass thru mode ( g.729a ) ? |
| 17:56.37 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 17:56.37 | fiXXXerMet | Sweet - ekiga is installed in ubuntu by default. |
| 17:56.50 | [TK]D-Fender | inforx: Yes. how else ddo you think calls get paced out of *? |
| 17:57.01 | [TK]D-Fender | inforx: Now please provide the pastebin I requested |
| 17:57.04 | jblack | fixxermet: Sure. Ubuntu even has the kitchensync |
| 17:57.16 | [TK]D-Fender | jblack: nice.... |
| 17:57.19 | inforx | hm I was reading http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru |
| 17:57.39 | inforx | my config is really simple, all I have is diasallow=all and allow=g729 in sip.conf |
| 17:57.49 | jblack | inforx: You're not new to asterisk, are you? |
| 17:57.51 | [TK]D-Fender | inforx: Stop reading, and show us what you're DOING so we can sonve this fast. |
| 17:57.52 | inforx | then just dial string in extensions.conf |
| 17:57.59 | [TK]D-Fender | solve* |
| 17:58.08 | mvanbaak | food |
| 18:01.38 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 18:01.58 | NovceGuru | Hey guys, if I can get my land line transfered to a VSP that can port it, could I then (likely) transfer to a VSP that can't port it directly from my PSTN provider? |
| 18:02.21 | De_Mon | ~vsp |
| 18:02.21 | jbot | rumour has it, vsp is a VoIP Service Provider |
| 18:03.37 | inforx | [TK]D-Fender, http://pastebin.com/m32cc3acb |
| 18:05.34 | *** join/#asterisk UserReg_CL (n=dede@200.113.129.63) |
| 18:05.37 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:05.42 | jblack | Ok guys, so I'm giving up on the blind dialing, and want to get Features to work (*1 to automon, *2 for attended transfer, *3 to park calls, etc). I have uncommented them under [featuremap], but pressing *2 are being ignored |
| 18:05.49 | UserReg_CL | hi... good year all !!! |
| 18:05.52 | [TK]D-Fender | inforx: your [sipura1] is using Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 18:06.17 | jblack | Is there more to it than just removing those comment markers? |
| 18:06.20 | UserReg_CL | one question please... need show only user sip conect (from console) |
| 18:06.27 | [TK]D-Fender | inforx: So * says that account only supports gsm|ulaw|alaw|h263 , and the PHONE itself says G729 ONLY. |
| 18:06.32 | inforx | well on sipura I set the g729.a as preferred codec [TK]D-Fender |
| 18:06.33 | [TK]D-Fender | inforx: Fix your sip.conf entry! |
| 18:06.51 | inforx | ah |
| 18:06.53 | inforx | ok |
| 18:07.00 | [TK]D-Fender | inforx: sure you told your SIPURA to use G.729, but you didn't tell * to LET it. |
| 18:07.25 | jblack | userreg: I don't understand. You want to know which sip users are connected? does "sip show peers" give you what you want? |
| 18:07.55 | [T]ank | i had a php script crash on me, and now asterisk is acting up. I am wondering if it is the manager that is not working properly. The symptoms I am having are that when I do a show channels it does not show me the summary of total channels. and if I try to do anything after that I have to exit out of the CLI and go back in to make anything work again. it is like asterisk is trying to crash. I know that if i were to restart asterisk it would fix it, but I ha |
| 18:08.26 | jblack | [t]ank: You got cut off at "...fix it, but I ha" |
| 18:08.54 | [T]ank | I know that if i were to restart asterisk it would fix it, but I have a lot of calls on and do not want to do that if I have other options. Any advice would be appreciated. |
| 18:09.22 | UserReg_CL | jblack: thank... sorry... need know which users is talk (or InUse) only |
| 18:09.39 | jblack | Did you run a ps aux to see if the script is still running? Perhaps you could kill it, thus returning control |
| 18:09.52 | [T]ank | yeah, i have killed a number of scripts. |
| 18:09.54 | [T]ank | it helped. |
| 18:10.00 | jblack | Ok. |
| 18:10.01 | [T]ank | but not 100% fixed in asterisk |
| 18:10.07 | [T]ank | i did a reload manager also |
| 18:10.15 | [T]ank | since the php scripts use AGI |
| 18:10.24 | jblack | I think I remember a timeout option that causes stuck things to get dumped. |
| 18:10.30 | jblack | I can look for it, but it'll take me a minute |
| 18:10.36 | inforx | [TK]D-Fender, I am getting now chan_sip.c:3670 sip_write: Asked to transmit frame type 2, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 18:10.44 | [TK]D-Fender | UserReg_CL: "show application chanisavail" |
| 18:10.49 | [T]ank | you thinking of rtptimeout? |
| 18:10.51 | jblack | tank: Is this agi, or system ? |
| 18:10.57 | [TK]D-Fender | inforx: ENTIRE call attempt please... |
| 18:11.04 | [T]ank | i "think" it is AGI |
| 18:11.13 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 18:11.26 | [T]ank | it is AGI that I am using and php that crashed, so I am assuming the issue with asterisk is AGI related |
| 18:11.27 | inforx | [TK]D-Fender, it's flooding my screens with that |
| 18:11.38 | jblack | Ok, gimme a moment |
| 18:11.51 | jblack | Look at the set_time_limit() option for php |
| 18:12.02 | [T]ank | in php? |
| 18:12.05 | [T]ank | or asterisk |
| 18:12.08 | jblack | php |
| 18:12.10 | [T]ank | ok |
| 18:12.31 | jblack | That'll tell the php interpreter to kill the script if it runs for longer than that. |
| 18:12.42 | [TK]D-Fender | inforx: Can't help you until you pastebin the whole call. from the beginning. |
| 18:12.53 | UserReg_CL | Thank Fender |
| 18:13.36 | jblack | [t]: by the way, when the scripts get stuck, are they stuck in an idle loop, or a busy loop? |
| 18:13.37 | inforx | [TK]D-Fender, i figured it, it was in iax.conf |
| 18:14.15 | jblack | [t]ank: by the way, when the scripts get stuck, are they stuck in an idle loop, or a busy loop? |
| 18:14.16 | inforx | another question, for IAX trunking I only need to enable port 4569 on the firewall right ? |
| 18:14.57 | [TK]D-Fender | inforx: Yes, it uses only 4569 UDP |
| 18:15.17 | [T]ank | i did not look. we killed them before knowing for. I would GUESS busy, as the cpu was running at 100% for that PID |
| 18:16.09 | jblack | Ok. Well, verify that. If it's a busy limit, you can also look at setting up ulimits in the php script. |
| 18:16.36 | jblack | That's moe complicated, but it'll let you base suicide on computer time, rather than wallclock time |
| 18:17.13 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 18:17.50 | jblack | And make your life easier by setting up a test suite for your php scripts, so that you catch bugs when you run it, instead of when asterisk runs it. ;) |
| 18:18.22 | [T]ank | this is one of those things that has been working well for a long period of time, and for whatever reason it crashed the other day and we just found out. |
| 18:18.47 | jblack | rarity doesn't turn a bug into a feature. :) |
| 18:18.58 | CrashSys | works for M$ |
| 18:19.12 | jblack | Bugs for M$ are rare? |
| 18:19.22 | CrashSys | when they turn from a feature into a bug |
| 18:19.38 | CrashSys | like vista's aeroglass "feature" |
| 18:19.46 | CrashSys | is pretty much a large flawed bug in the operating system now |
| 18:20.05 | CrashSys | Ehh... guess it's not that rare... |
| 18:20.45 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:21.39 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:21.55 | hi365 | file ping |
| 18:22.10 | file | pong |
| 18:22.15 | hi365 | pm? |
| 18:23.31 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 18:25.44 | *** join/#asterisk beek (n=klinebl@pool-96-245-14-242.phlapa.fios.verizon.net) |
| 18:27.53 | *** join/#asterisk alexcf (i=alexcf@online0.sov.netsumo.com) |
| 18:27.54 | alexcf | hi |
| 18:28.05 | jblack | Oh cute. zoiper isn't able to do dtmf tones for me |
| 18:28.11 | alexcf | first off.. :p im not coder/linux guru :p |
| 18:28.28 | *** join/#asterisk lewis333 (n=lewis@207.97.163.106) |
| 18:28.48 | alexcf | http://pastebin.com/m1ef5918a |
| 18:28.52 | alexcf | can anyone help me with that? |
| 18:29.11 | lewis333 | Looking for help connecting a Polycom IP 4000 to an Altigen system - anyone got any experience? |
| 18:29.36 | jblack | show me fxotune.c lines 130-140 ? |
| 18:30.23 | [TK]D-Fender | jblack: it works fine. |
| 18:31.04 | alexcf | http://pastebin.com/m4b83960b |
| 18:31.10 | jblack | I'll sip debug it. The visual impression I'm getting is that it's not sending dtfm2833 |
| 18:31.12 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 18:32.13 | alexcf | also, dunno if anyone has had the problem, but when a call comes into the queue and it gets terminated as a ZOMBIE in the queue_log, it stalls asterisk and i have to restart it |
| 18:32.27 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583308.dsl.bell.ca) |
| 18:32.33 | alexcf | 1199296165|1199296160.16|noc|Local/1574@default-25a8,1<ZOMBIE>|COMPLETECALLER|5|0 |
| 18:32.37 | alexcf | 1199296184|1199296184.21|switchboard|NONE|ENTERQUEUE||0751 |
| 18:32.39 | alexcf | like so |
| 18:32.44 | jblack | alexcf: What compiler are you using? |
| 18:32.59 | alexcf | urr |
| 18:33.03 | alexcf | how would i find out? |
| 18:33.10 | jblack | See line 134? That's C99. |
| 18:33.36 | jblack | If you're using an old compiler or a compiler pretending to be old, it won't like that. |
| 18:33.37 | alexcf | ok |
| 18:33.41 | alexcf | ah |
| 18:33.48 | alexcf | gcc version 2.95.4 20011002 (Debian prerelease) |
| 18:33.52 | alexcf | so yea, old |
| 18:34.10 | alexcf | jwh: voip0 is lame :p |
| 18:34.15 | jwh | alexcf: it certainly is |
| 18:34.40 | alexcf | (jwh used to work for the company i work for) |
| 18:35.03 | jwh | at least yours doesn't sit in STOP state randomly :P |
| 18:35.05 | *** join/#asterisk RoyK (n=roy@ip-131-23-149-91.dialup.ice.no) |
| 18:35.06 | alexcf | asterisk barf'd something chronic on debian etch too |
| 18:35.19 | alexcf | (luckily i hadn't formatted the old machine) |
| 18:35.27 | jblack | I'll look at my dtmf problem when I get back |
| 18:36.16 | alexcf | http://www.asteriskguru.com/archives/asterisk-dev-queue-transfer-server-stability-vt62012.html |
| 18:36.22 | alexcf | but yea, jwh, that's the issue :p |
| 18:37.36 | jwh | still 1.2? |
| 18:38.32 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:39.02 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 18:39.05 | shadebob | Hi, |
| 18:40.48 | alexcf | yea |
| 18:41.00 | teknoprep | anyone here use SIP on Teliax before and asterisk ? |
| 18:41.05 | teknoprep | i am having odd problems |
| 18:41.19 | jblack | I'm using IAX on Teliax. |
| 18:41.39 | jblack | I'm not sure why you'd use sip instead of iax, since they support it. |
| 18:41.43 | beek | I too use IAX on Teliax... no problem. |
| 18:42.14 | teknoprep | i don't have a Zaptel card |
| 18:42.19 | teknoprep | so it uses ZTdummy |
| 18:42.45 | jblack | You don't need a zaptel card to do iax. :) |
| 18:43.18 | jblack | They even provide an iax context to log in with iax over ip. |
| 18:43.27 | Qwell | iax over ip? |
| 18:43.30 | Qwell | as opposed to? |
| 18:43.37 | jblack | No idea. :) |
| 18:44.07 | Qwell | file: iax2 over serial. get on that. |
| 18:44.15 | file | yes sir |
| 18:44.24 | mog | iax2 over zap Qwell |
| 18:44.31 | Qwell | mog: I was thinking that |
| 18:44.46 | mog | we need switchtype=iax2 get creslin on that |
| 18:45.13 | Qwell | PRI is like...iax2 trunking. |
| 18:45.39 | Qwell | erm, nm, trunking does the audio too, doesn't it? |
| 18:45.43 | *** join/#asterisk vrtk (n=bb@189.21.178.20) |
| 18:47.09 | marlow | mog: iax2 over zap is easy .. HDLC trunk :) |
| 18:47.30 | Qwell | hdlc != ip? |
| 18:47.47 | marlow | Qwell: hdlc is lower layer |
| 18:47.57 | Qwell | gotcha |
| 18:48.01 | marlow | Qwell : data framing for pri |
| 18:48.04 | [T]ank | is there a way to show channels in order of how long they have been connected? |
| 18:48.22 | marlow | Qwell : you can on asterisk use part of PRI for voice and part of it for a data trunk |
| 18:48.43 | Qwell | yes, I knew that, just didn't know much more than that. heh |
| 18:50.53 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 18:54.37 | [T]ank | i have killed all of the php scripts completely. So now, is it possible to recover asterisk without having to restart it. It is taking about 3 minutes for it to complete a "show channels" request. There are only about 50 calls in the system so it should be instant. |
| 18:56.49 | marlow | [T]ank : you can always do a graceful restart, which means it'll only restart once all calls are gone |
| 18:57.20 | _ShrikE | once a graceful restart has been issued, doest asterisk continue taking new calls? |
| 18:57.31 | _ShrikE | doest = does |
| 18:57.31 | [T]ank | well.. there will be calls until we close, so I can just reboot once we close. |
| 19:01.04 | marlow | stop gracefully: Gracefully shut down Asterisk, i.e. stop receiving new calls and shut down at empty call volume |
| 19:03.23 | marlow | you can also use: |
| 19:03.25 | marlow | stop when convenient: Shut down Asterisk at empty call volume |
| 19:03.42 | marlow | which means it doesn't stop until no calls are on the system, but it still takes calls in |
| 19:03.54 | marlow | it just waits for an idle point |
| 19:04.37 | _ShrikE | with a stop gracefully, what does asterisk issue if a new call comes in via sip? 503? |
| 19:06.16 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 19:06.34 | dacs | good morning all |
| 19:07.59 | [T]ank | marlow: i was hoping to find someway to do this without restarting. |
| 19:08.18 | [T]ank | getting 50 people off the phone with our call volume is going to be a nightmare. |
| 19:08.48 | Alan_Hicks | Just a quick question for you guys. I don't see anything in sip.cfg that jumps out at me. I'm using Polycom IP 320 phones and I was wondering if there's any way to make a distinctive ring on transfers. |
| 19:09.13 | Alan_Hicks | If you could just point me in the right direction I'd appreciate it. No hand-holding required. ;-) |
| 19:09.46 | jblack | alan_hick: Forgot to feed the monkeys that live in the phones and play special tones, eh? |
| 19:10.05 | CrashSys | alan_hicks: voip-info.org... search for "polycom auto-answer" |
| 19:10.19 | CrashSys | it'll tell you how to add a sip header to make the phone change ringers/etc... |
| 19:10.34 | Alan_Hicks | jblack: Damn! I forgot to feed them over the holidays! |
| 19:11.07 | Alan_Hicks | CrashSys: That's just it though, the phones handle the transfer themselves, no config necessary for asterisk (at least to my knowledge). |
| 19:11.18 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 19:11.52 | jblack | That'll do it every time. :) |
| 19:12.15 | lewis333 | there a channel for Altigen systems? |
| 19:12.25 | Alan_Hicks | CrashSys: I configured auto-answer correctly for paging and did a distinctive ring. |
| 19:12.49 | Alan_Hicks | But transfers as near as I can tell, don't touch the dialplan, so there's no way to add a SIP header. |
| 19:14.46 | Alan_Hicks | Think that's bad? They're projecting cotton at 68 cent in March. |
| 19:15.51 | jblack | Yeah, it may close at 100 today. It's 99.59 right now. |
| 19:16.07 | syzygyBSD | fuck it, I am moving away |
| 19:16.18 | syzygyBSD | oops, sorry, |
| 19:16.27 | syzygyBSD | s/&#*$/a nicer word/ |
| 19:16.56 | fiXXXerMet | Is there any way to output sip debug to a file? |
| 19:16.58 | jblack | I think this is good news. Let the markets crash, let oil go to 3k, let inflation turn the dollar to nothing, causing a great depression. Then NOBODY will have a phone, and i won't need to debug my asterisk installation! |
| 19:17.02 | jblack | sweet! |
| 19:17.20 | *** join/#asterisk scrllock (i=p0m@35.10.222.74) |
| 19:17.55 | syzygyBSD | jblack: you forgot about all the people that are trying to figure out what happend will be calling eachother, because they can't afford to go anywhere with the price of oil |
| 19:18.57 | tzafrir_home | Hey, they will still be able to use strings and cans |
| 19:19.18 | jblack | tza: Can I use IAX2 on strings? |
| 19:19.38 | *** part/#asterisk myiagy (n=Jose@200.215.59.133) |
| 19:22.38 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 19:25.41 | *** join/#asterisk scrllock (i=p0m@35.10.222.74) |
| 19:26.14 | jblack | I suppose it's possible... string is made from plants. Plants have branches, which come off of trunks. IAX does trunking. |
| 19:26.56 | jblack | and now that I think about it, I do have a * on my christmas tree. |
| 19:27.43 | syzygyBSD | lol |
| 19:28.23 | jblack | Great. as if asterisk weren't already hard enough to configure, now I've got to worry about poking my eye out with pine needles. |
| 19:28.43 | syzygyBSD | good thing you can configure it remotely |
| 19:28.51 | jblack | With a string! |
| 19:29.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:29.15 | jblack | Pull on the string hard enough, and I can change my tree from a vertical to a horizontal configuration |
| 19:29.57 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 19:32.10 | *** part/#asterisk inforx (n=inforx@S0106006097940f68.vw.shawcable.net) |
| 19:32.58 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 19:33.18 | *** join/#asterisk CapRicORN^80 (n=you@202.61.62.110) |
| 19:34.03 | jblack | Since things are quiet... What do you guys think about medicine made out of sperm? seriously. http://www.msnbc.msn.com/id/22333518/ |
| 19:34.36 | syzygyBSD | as long as I get some of the profits I am all for it |
| 19:34.39 | vrtk | $100 a barrel! |
| 19:34.42 | vrtk | whole shift!! |
| 19:35.38 | vrtk | thanks god i've got legs |
| 19:36.07 | jblack | heh. The soles of your shoes are made out of oil too. |
| 19:37.33 | syzygyBSD | not mine, I have wooden shoes |
| 19:38.17 | vrtk | you just had to remember that, hadn't you ? |
| 19:39.10 | jblack | I'd suggest you swim to freedom instead.. but you'd have to do it without those nifty goggles. |
| 19:39.21 | *** join/#asterisk glen2 (n=glen@87-194-2-134.bethere.co.uk) |
| 19:39.53 | vrtk | where did you get this info about cotton prices ? |
| 19:40.24 | *** join/#asterisk egecko (n=sam@cpe-76-176-206-245.san.res.rr.com) |
| 19:40.26 | *** join/#asterisk Curi (n=creinero@201.239.234.148) |
| 19:40.42 | Curi | Hi all |
| 19:40.52 | jblack | Ohhh. This is better. RMS and TdR are in a namecalling match, including things like "your big fat lying mouth." |
| 19:42.15 | JayTee52 | where? |
| 19:42.23 | jblack | http://kerneltrap.org/mailarchive/openbsd-misc/2008/1/2/533804 |
| 19:42.36 | jblack | 2008/1/2. Heh. Happy new year |
| 19:43.07 | *** join/#asterisk EdNagy (n=moose@72.168.135.209) |
| 19:43.11 | Curi | Has anyone had a problem with asterisk (i'm using 1.4.16.1) when it tries to register a sip account in another server it times out, but the box actually receives the packet and it's asterisk that's not reading the queue. If i do a netstat -na i see that the recv-q keeps piling up |
| 19:43.43 | jblack | curi: Sounds like a firewall problem to me |
| 19:44.09 | Curi | but the server is receiving-- i see like if ipchains were running right? |
| 19:44.11 | envisean | hey guys, does anyone know where I can get some pretty standard IVR prompts? |
| 19:44.36 | syzygyBSD | asterisk-sounds |
| 19:44.49 | envisean | i got the ivr module installed for freepbx |
| 19:44.57 | envisean | i have all the default sounds |
| 19:45.05 | envisean | but i meant like a default greeting, and also the phone book |
| 19:45.21 | syzygyBSD | I don't know.. I never do anything with freepbx |
| 19:45.50 | tzafrir_home | envisean, that is a freepbx question, not an asterisk question |
| 19:46.02 | envisean | ok thanks tzafrir_home |
| 19:46.03 | tzafrir_home | freepbx questions got to #freepbx |
| 19:46.21 | jblack | envisean: How about yo' mamma? |
| 19:46.36 | jblack | Give her a visit and shove a microphone in front of her face. ;) |
| 19:47.03 | envisean | jblack: wow |
| 19:47.06 | tzafrir_home | jblack, and if you read carefully there, you'll see RMS uses no web browser |
| 19:47.13 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:47.50 | Curi | jblack: but wouldn't a firewall just drop the packet? i see the packet geting in, it's just that asterisk is not reading it from the queue |
| 19:47.51 | jblack | Or combs. |
| 19:49.35 | jblack | curi: What a firewall does with a packet is variable. How do you know that the packets are getting to asterisk, and not getting dropped/rejected/redirected/etc ? |
| 19:49.43 | jblack | Did you strace asterisk to verify that? |
| 19:50.11 | jblack | Or run sip debug ? |
| 19:50.22 | Curi | jblack: i'm running tethereal/wireshark and see the packet geting in the interface |
| 19:51.00 | Curi | jblack: but then with a sip debug i only see the packets that asterisk sent, not the ones that came in |
| 19:51.36 | Curi | jblack: and when i do a netstat -na i see that the recv-q for the udp 5060 port is filling up, because asterisk is not reading the packets |
| 19:52.46 | jblack | Hmmm. You checked lsof -i, to make sure it's listening on the right interface? |
| 19:53.07 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:54.52 | Curi | jblack: it's listening in |
| 19:55.05 | Curi | jblack: *:5060 |
| 19:55.21 | jblack | I'm out of ideas |
| 19:55.51 | fiXXXerMet | Is there any way to output sip debug to a file? |
| 19:55.53 | Curi | jblack: me too :( |
| 19:57.05 | jblack | try "asterisk -rvvvvvvvvvvv | tee logfile" |
| 19:58.47 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 20:02.32 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 20:05.02 | *** join/#asterisk mikecx (n=mikecx@pool-70-104-112-56.chi.dsl-w.verizon.net) |
| 20:05.46 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 20:05.51 | mikecx | faxes using tdm800P with fxo/fxs lines are detected by the fax machine, it then tries to connect and fails. my dialplan http://pastebin.com/m7469a8e5 (ignore the comments) |
| 20:06.18 | *** join/#asterisk jdspencer (n=jdspence@12.37.95.91) |
| 20:07.06 | jdspencer | Can anyone point me to or tell me what the bus speed for the Digium TE41x cards would be? (The 3.3V PCI-X ones) |
| 20:07.12 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 20:09.55 | Alan_Hicks | CrashSys: I configured auto-answer correctly for paging and did a distinctive ring. |
| 20:10.04 | Alan_Hicks | Oops. :-) Ignore. |
| 20:10.26 | Alan_Hicks | Just a quick question for you guys. I don't see anything in sip.cfg that jumps out at me. I'm using Polycom IP 320 phones and I was wondering if there's any way to make a distinctive ring on transfers. |
| 20:11.05 | Alan_Hicks | These phones handle Transfers via one of their soft buttons. |
| 20:11.43 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:11.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:19.02 | [TK]D-Fender | Alan_Hicks: look at *'s standard variables. there is one for "blind transfers" as an indication. There may be one to indicate an attended transfer is in progress. |
| 20:19.17 | [TK]D-Fender | Alan_Hicks: You'd have to add some dialplan checks to see if you need to add a header based on that. |
| 20:19.25 | [TK]D-Fender | Alan_Hicks: All around of course. |
| 20:19.36 | file | with SIP you don't know it's an attended transfer until you actually complete the transfer... fyi |
| 20:19.43 | *** part/#asterisk Curi (n=creinero@201.239.234.148) |
| 20:19.51 | Alan_Hicks | [TK]D-Fender: See that's just the point though, I haven't done anything with Transfers in *. I'm just using the phone's built-in transfer soft-button. |
| 20:20.04 | [TK]D-Fender | file: Yeah, thats what I though, but I wanted to leave that open... |
| 20:20.18 | [TK]D-Fender | Alan_Hicks: I didn't say through * DTMF. |
| 20:20.33 | Alan_Hicks | ~DTMF |
| 20:20.34 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 20:20.44 | *** join/#asterisk greekguy8888 (n=alex@c-76-118-201-12.hsd1.ma.comcast.net) |
| 20:20.50 | greekguy8888 | hola everyone |
| 20:21.07 | greekguy8888 | have a question for the ages |
| 20:21.21 | greekguy8888 | anyone have experience with * queues and transfers? |
| 20:21.37 | Alan_Hicks | [TK]D-Fender: I'm not sure I follow you. What does DTMF have to do with it? |
| 20:21.38 | [TK]D-Fender | greekguy8888: Yeah... it ties up your agents until the transferred call terminates |
| 20:21.50 | greekguy8888 | unless you use the # right? |
| 20:21.55 | greekguy8888 | anyway around this? |
| 20:22.23 | [TK]D-Fender | Alan_Hicks: you phone's soft-keys transfer capability have nothing to do with the "tT" transfer options via DTMF (following your comment " I haven't done anything with Transfers in *. ") |
| 20:22.38 | [TK]D-Fender | Alan_Hicks: Wasn't sure exactly what you meant so I tried to head that thought off at the pass |
| 20:22.41 | greekguy8888 | is there a way to block the soft key use in polycoms? |
| 20:22.54 | greekguy8888 | i know you can remap keys |
| 20:23.02 | greekguy8888 | but remapping screwa all other functions of that key |
| 20:23.07 | Alan_Hicks | [TK]D-Fender: Ok. I'm not entirely sure what I mean either half the time. :^) |
| 20:23.12 | *** part/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 20:23.30 | [TK]D-Fender | Alan_Hicks: So together we know either everything you want, or nothing at all :) |
| 20:23.50 | [TK]D-Fender | greekguy8888: only stops the HARD keys. You can't stop the soft keys |
| 20:23.55 | Alan_Hicks | [TK]D-Fender: I'm afraid it's "nothing at all" far too often. :^) |
| 20:24.11 | [TK]D-Fender | greekguy8888: So functions you can disable in provisioning. |
| 20:24.20 | [TK]D-Fender | greekguy8888: Go Check the admin guide |
| 20:24.23 | greekguy8888 | so in an enterprise deployment, you have a key of death on the phone and pray no one uses it? |
| 20:24.55 | greekguy8888 | i remapped the hardkey just fine, its the availability of the softkey that scares me |
| 20:25.21 | Alan_Hicks | [TK]D-Fender: Ah! Think I get it now. The Polycom soft button is jsut sending a standard transfer DTMF signal to asterisk which then dials the transferee party. |
| 20:25.35 | [TK]D-Fender | Alan_Hicks: NO. the exact opposite |
| 20:25.36 | greekguy8888 | well no |
| 20:25.47 | greekguy8888 | it transfers only the sip channel when you use the polycom transfer |
| 20:25.56 | [TK]D-Fender | greekguy8888: you mean "Reject" I presume... |
| 20:25.56 | Alan_Hicks | [TK]D-Fender: Enlighten me then please. :-) I've apparently gotten it all wrong again. |
| 20:26.06 | greekguy8888 | if there si an agent channel attatched, its not acted upon, that's why the agent looks like it's in use |
| 20:26.18 | [TK]D-Fender | Alan_Hicks: SIP transfers are seperate from *-managed DTMF initiated transfers |
| 20:26.27 | greekguy8888 | yes |
| 20:26.33 | greekguy8888 | much different |
| 20:26.43 | [TK]D-Fender | greekguy8888: Yeah, I wish they'd free up the agent as well... |
| 20:26.46 | greekguy8888 | the transfer actually happens via a redirect via the buttons i believe |
| 20:26.59 | [TK]D-Fender | greekguy8888: Tahts in various states of "repair" depending on when you check the bug tracker. |
| 20:27.07 | Alan_Hicks | [TK]D-Fender: Then how do I transfer to an outbound line via my wctdm card? That's not SIP. |
| 20:27.13 | greekguy8888 | yeah i checked it till my eyes rolled into the back of my head |
| 20:27.23 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 20:27.48 | [TK]D-Fender | Alan_Hicks: That is often done via DTMF features, but should also (for FXS) be able to trigered through flash. |
| 20:28.13 | [TK]D-Fender | greekguy8888: Great view of a cavernous mass? :) |
| 20:28.14 | Alan_Hicks | I'm so confused. :-) |
| 20:28.23 | greekguy8888 | lol |
| 20:28.32 | greekguy8888 | so there's nop hope for a workaround then i guess |
| 20:28.35 | Alan_Hicks | Let me make an example here, and maybe that will help me understand. |
| 20:29.11 | Alan_Hicks | Caller dials my PSTN number. Asterisk answers and dials my internal SIP phones. SIP/alan picks up the line. |
| 20:29.46 | Alan_Hicks | SIP/alan decides to transfer the call to another PSTN number. The user hits the softbutton labelled "Transfer" and dials 555-5555 (whatever). |
| 20:30.08 | Alan_Hicks | This can't be a SIP transfer, right? * has to dial out the wctdm. |
| 20:30.18 | greekguy8888 | it is a sip transfer |
| 20:30.20 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 20:30.22 | [TK]D-Fender | greekguy8888: Well there's hope, but I'm not a core coder, perhaps you could get involved with those who are and see if you can't nudge the process along. |
| 20:30.38 | greekguy8888 | any * admins on? |
| 20:31.05 | [TK]D-Fender | Alan_Hicks: Yes. which is a SIP transfer, not an * DTMF transfer. Same end result, except that you phone does the work, and has a nice interface. |
| 20:31.26 | greekguy8888 | the thing that sucks is, i can't be the only one dealing with this... there are like 4-5 enterprise opensource releases of *, and nothing, i mean nothing, looks like there will be a fix to this |
| 20:31.44 | greekguy8888 | cmon, how do they deal with this on switchvox? |
| 20:31.48 | Alan_Hicks | Ok. So to do what I want to do, I need to enable DTMF transfers. Say user dials *8 or something, then the number or extension. |
| 20:32.06 | Alan_Hicks | * would intercept the *8 and act on it according to the dial-plan, right? |
| 20:32.12 | greekguy8888 | alan, yes use your features.conf file and make sura all dials have the tT on the end of them |
| 20:32.24 | [TK]D-Fender | Alan_Hicks: you don't need *'s involvement to transfer a call via a Polycom. |
| 20:32.38 | greekguy8888 | unless you are transferring from a queue, no |
| 20:32.41 | greekguy8888 | :) |
| 20:32.45 | Alan_Hicks | [TK]D-Fender: Apparently I do if I want the call to have a distinctive ring. |
| 20:32.55 | greekguy8888 | in that case yes |
| 20:32.58 | greekguy8888 | but again |
| 20:33.02 | [TK]D-Fender | greekguy8888: Oh yeah! If you want to transfer a call out of a queue is you call your Agent via a Local channel you can use the "tT" transfer and that WILL free up the agent. |
| 20:33.05 | greekguy8888 | you can do this with feature code |
| 20:33.12 | [TK]D-Fender | greekguy8888: Horribly unnatural, but it works from what I recall. |
| 20:33.23 | greekguy8888 | send it into another context where you can set distinctive ring b4 you dial |
| 20:33.49 | greekguy8888 | really? |
| 20:33.50 | Alan_Hicks | greekguy8888: I was thinking I would just use a macro or something, like I did for distinctive rings on paging. |
| 20:34.00 | greekguy8888 | so fo instance instead of using sip/wkjhsdfgdf |
| 20:34.02 | [TK]D-Fender | greekguy8888: Yeah, I nearly forgot that bit... go try |
| 20:34.08 | greekguy8888 | which is ext 333 |
| 20:34.11 | [TK]D-Fender | greekguy8888: yeah. |
| 20:34.14 | greekguy8888 | i would instead set the membername to |
| 20:34.16 | greekguy8888 | Local/333 |
| 20:34.17 | *** join/#asterisk Absorto (n=user@189.141.94.36) |
| 20:34.22 | greekguy8888 | ? |
| 20:34.22 | Alan_Hicks | I'll have to read up on features.conf though. Thanks for all the help, and the nice cluebat strike to the forehead. That always helps. |
| 20:34.25 | greekguy8888 | like that? |
| 20:34.40 | [TK]D-Fender | greekguy8888: member => Local/333@contextwithmyexten/n |
| 20:34.41 | mikecx | asterisk does not like me |
| 20:34.55 | greekguy8888 | fender thanks for the clue, let me try that out |
| 20:34.56 | [TK]D-Fender | greekguy8888: and make sure that the dial you use enabled those dial features |
| 20:35.15 | greekguy8888 | you mean the tT |
| 20:35.19 | greekguy8888 | right> |
| 20:35.39 | [TK]D-Fender | greekguy8888: Yes |
| 20:36.13 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:38.06 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-198-162-26.hsd1.tx.comcast.net) |
| 20:38.48 | *** join/#asterisk pagec (n=pagec@cpe-66-108-229-212.nyc.res.rr.com) |
| 20:39.09 | pagec | anyone having problems with Polycom 601 phones not wanting to boot since newyears? |
| 20:39.25 | Qwell | y2k8 bug? |
| 20:39.30 | Qwell | hangover? |
| 20:39.50 | pagec | sorta looking like it :( |
| 20:40.46 | Qwell | pagec: to be honest, I'd call Polycom and ask them.. they may have had similar reports |
| 20:41.30 | mikecx | anyone have problems with faxes over fxo/fxs lines refusing to connect? |
| 20:41.52 | pagec | Qwell: tried, unless you have a reseller pin you cannot get them on the phone |
| 20:42.02 | Qwell | call your reseller? |
| 20:42.49 | pagec | Qwell: did, CDW is being slow to respond, and ppl tend to get pissing when the phones aren't working |
| 20:42.59 | Qwell | yuck |
| 20:44.09 | *** join/#asterisk p4c0 (n=Paks@200.124.22.34) |
| 20:45.26 | p4c0 | hello, i'm using asterisk to connect to my voip service provider, however when i try to make outgoing calls asterisk invites doesn't have user/password, and server replies with forbidden, is there a way to force this? even if i'm registered with the voip service provider?? |
| 20:46.57 | pagec | p4c0: which server is replying with fobidden, your server to your telephone (or soft phone) or their server to your server? |
| 20:47.47 | Qwell | registering has nothing to do with making calls |
| 20:47.55 | p4c0 | pagec, the server of the provider (to my invite request for outgoing calls, i debug them and noticed that there's no user/password there) |
| 20:48.49 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:49.13 | pagec | did your provider suppy you a user name and password when you paid/signed up for their service to make outgoing calls with? |
| 20:49.17 | p4c0 | Qwell, how can i make my invites to have the username/password? or to resend them after a forbidden from server? |
| 20:49.35 | p4c0 | pagec, yes, the same as for registry |
| 20:49.45 | *** join/#asterisk javb (n=javb@190.80.201.55) |
| 20:50.48 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 20:50.48 | javb | hi, i had asterisk 1.2, everything PERFECT. now, after upgrading to 1.4, i cant transfer, or even make a new call, while i`m using the other line of the phone (polycom 330), the notice i get is: "[Jan 2 04:46:32] NOTICE[1931]: chan_sip.c:13794 handle_request_invite: Failed to authenticate user "102" <sip:102@10.0.0.55>;tag=A5C8F780-3952FB9D" |
| 20:50.53 | javb | Any ideas? |
| 20:51.09 | javb | No changes to any config file |
| 20:51.15 | p4c0 | it should be of type peer right? |
| 20:51.37 | javb | uuh? |
| 20:51.38 | pagec | p4c0: friend (at least for my provider) |
| 20:51.49 | p4c0 | pagec, i'll try, thanks |
| 20:52.05 | Alan_Hicks | Hmm... wonder if I can reconfigure what the Polycom's transfer soft-button does? |
| 20:52.10 | pagec | p4c0: who is your provider? |
| 20:52.43 | p4c0 | pagec, some local one... optynex.com |
| 20:52.45 | pagec | javq have you tried doing something like 'sip show users' and seeing if "102" is a user? |
| 20:53.11 | pagec | javb: ^ |
| 20:53.49 | *** join/#asterisk marcan (i=1337@128.Red-83-61-125.dynamicIP.rima-tde.net) |
| 20:54.15 | pagec | p4c0: well i cannot find their web site, but they probably probably have a sip config howto |
| 20:54.27 | javb | here is "sip show users" http://pastebin.com/m2f94f78a |
| 20:54.35 | pagec | soemtimes you need a specific user name in addition to the password/name (or other settings) |
| 20:54.48 | p4c0 | pagec, i have done it |
| 20:54.53 | pagec | the username being the [section header[ |
| 20:55.02 | pagec | glad to see it's working then |
| 20:55.28 | greekguy8888 | fender u still here? |
| 20:55.30 | mikecx | grrrr, why wont this damn fax answer |
| 20:55.37 | javb | this is so weird, everything was ok in 1.2 |
| 20:55.46 | asteriskmonkey | in asterisk 1.2x has anyone expericend an issue where you have a 2 box setup.. 1 has the pri lines and the other is sip only, if you have a phone on the pri box you can make calls perfectly and on the sip only box calls can be made but international calls come back with a trouble cause 41 |
| 20:56.01 | pagec | javb: yeah you have the user 102 with password pass |
| 20:56.13 | jblack | Ok. So i'm reading up much more carefully in features.conf. As I understand it, there's 4 parts. |
| 20:56.43 | jblack | The first part is to set something up in application map.. name => buttonsequence, Command to run |
| 20:56.47 | asteriskmonkey | also get a 500 internal server error back on the sip only box |
| 20:56.56 | jblack | Then, in featuremap, you then do commandname => buttonsequence |
| 20:57.06 | javb | pagec, iknow, i repeat, this is out of the question, i have not made ANY change... wow... and googling doesnt seem to help... :/ |
| 20:57.06 | *** join/#asterisk gaero (n=theo@unaffiliated/balkantools) |
| 20:57.13 | gaero | hi there |
| 20:57.25 | jblack | Then, one adjusts DYNAMIC_FEATURES, possibly through extensions.conf globals, to run on FEATURENAME#FEATURENAME .... |
| 20:57.36 | jblack | Then, finally, one adjusts dial options. |
| 20:57.36 | gaero | somebody have eard about RMS norm of telecommunications ? |
| 20:57.38 | jblack | Is that correct? |
| 20:57.43 | greekguy8888 | anyone have knowledge on queues and transfers using the sip transfer on polycoms? |
| 20:57.47 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 20:58.25 | pagec | javb: i would install a softphone and get that working from scratch. there is probably some little thing that is off which getting a new connection to work will highlight to you |
| 20:58.45 | [TK]D-Fender | jblack: Yup |
| 20:58.59 | pagec | there are a few differences between 1.2 and 1.4, i don't know if i woudl expect 1.4 to work perfectly off of 1.2 stuff |
| 20:59.15 | jblack | Ok, good. |
| 20:59.20 | p4c0 | pagec, humm the textual reply from provider is "SIP/2.0 403 Forbidden (Not Proxy/Gateway)" just after a "SIP/2.0 100 Trying" however there's no user/passwor on the invite that i'm sending |
| 20:59.37 | jblack | then I ahve some questions. |
| 20:59.54 | asteriskmonkey | i have 2 boxes 1 has a pri the other dosnt,using the same context that the sip only box is registerd too on the pri box international calls can be made, yet on the sip only its gets a 500 error, but local works fine.. odd... |
| 21:00.04 | jblack | First, it seems redundant to have featurename => keysequence in both featuremap and applicationmap |
| 21:00.29 | jblack | Is it actually redundant and a way to potentially get onself into trouble, or is there a benefit to it? |
| 21:01.02 | [TK]D-Fender | jblack: Featuremap is for *'s interal offered features. Application map is for really custom stuff you want to do |
| 21:01.22 | jblack | Ahhh. |
| 21:01.24 | [TK]D-Fender | jblack: like "attended transfer" is not a dialplan feature. |
| 21:01.48 | [TK]D-Fender | jblack: You can use applicationmap to allow you to play back some sort of scripted message to your caller for instance. |
| 21:01.54 | marlow | which is, why it's a bitch, when you've got mixed devices |
| 21:02.16 | [TK]D-Fender | jblack: Or as a cool example : Record a bunch of DTMF tones in a sequence and trigger them as a macro |
| 21:02.24 | [TK]D-Fender | jblack: To speed a login or something |
| 21:03.01 | [TK]D-Fender | jblack: Or to Flash an analog line for example to access PSTN call-waiting. |
| 21:03.36 | jblack | Ok. Looking through the * book, I don't see a Dial flag for atxfer. |
| 21:03.47 | pagec | p4c0: idk, you may want to call your provider then |
| 21:03.52 | jblack | How do I find what the dial flag is? |
| 21:04.04 | p4c0 | pagec, yes |
| 21:04.12 | [TK]D-Fender | jblack: "tT" each allow access to atxfer if you set it in DYNAMIC_FEATURES |
| 21:04.26 | jblack | so t and T are for both blind and attended |
| 21:04.33 | [TK]D-Fender | jblack: But remember this is only for devices taht don't have this functionality all by themselves. |
| 21:04.39 | [TK]D-Fender | jblack: Exactly |
| 21:04.46 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 21:05.03 | jblack | Understood. Let devices do things on their own when possible. |
| 21:05.41 | ManxPower | [TK]D-Fender: do you use Adtran channel banks? |
| 21:06.02 | [TK]D-Fender | ManxPower: Nope, only used Rhino. You know my avversion to using T1 tech for analog needs |
| 21:06.11 | asteriskmonkey | ManPower: ever had an issue like this? , i have 2 boxes 1 has a pri the other dosnt,using the same context that the sip only box is registerd too on the pri box international calls can be made, yet on the sip only its gets a 500 error, but local works fine |
| 21:06.38 | ManxPower | asteriskmonkey: sorry, I'm not helping people on the channel for a while. |
| 21:06.47 | asteriskmonkey | k |
| 21:06.53 | ManxPower | [TK]D-Fender: we had one lock up on monday. really odd. |
| 21:07.50 | javb | ok, i cant do it with the softphone, but no with the polycom set |
| 21:09.10 | pagec | javb: then i would look at the polycoms configuration. there are various pages on the web on how to setup polycoms to work with asterisk |
| 21:09.28 | mvanbaak | ManxPower: gheh, why not ? |
| 21:09.40 | javb | pagec, my 12 polycoms here WERE working great! |
| 21:09.53 | pagec | javb: be sure the search for the specific model polycom you have |
| 21:10.13 | pagec | javb: for better or worse you went with 1.4 and it is a new day |
| 21:13.07 | *** part/#asterisk gaero (n=theo@unaffiliated/balkantools) |
| 21:16.47 | javb | in sip debug, i see i`m getting something like "User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049" |
| 21:17.09 | javb | maybe there is the authenti.. problem ? |
| 21:17.53 | ManxPower | mvanbaak: I'm being an asshole. |
| 21:18.28 | mvanbaak | ManxPower: yeah. but that was true even back in the days that you did help ppl ;) |
| 21:24.03 | [TK]D-Fender | alrighty. Gheckout time.. off to go guitar-shopping. |
| 21:24.05 | [TK]D-Fender | BBL |
| 21:24.26 | *** join/#asterisk mrdigital (n=kjhh@pool-72-81-78-17.phlapa.east.verizon.net) |
| 21:25.24 | ManxPower | 8-) |
| 21:25.26 | pagec | javb: the polycom is your phone, perhaps the problem is with your phone and the asterisk server, not your provider? |
| 21:26.41 | javb | pagec, not my provider |
| 21:27.17 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.17 (2008/01/02), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.7.1 (2007/12/13), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 21:29.51 | lirakis | later all |
| 21:29.59 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:34.12 | *** join/#asterisk PepOSX (n=pepOSX--@201.243.87.221) |
| 21:41.37 | greekguy8888 | anyone have knowledge on queues and transfers using the sip transfer on polycoms? |
| 21:42.29 | ManxPower | mvanbaak: Mostly I got tired of the quality of the users around here. I decided to take a break from them. |
| 21:43.07 | ManxPower | If enough people do that maybe Digium will put a support person here. |
| 21:44.16 | mvanbaak | that would be nice eh ? |
| 21:45.55 | ManxPower | I'm just grumpy. |
| 21:48.54 | *** join/#asterisk phillipmarlowe (n=marlowe@208.66.88.252) |
| 21:49.59 | nhuisman_work | anyone here using rpath? |
| 21:51.10 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 21:52.56 | *** join/#asterisk darthahmed (n=none@host86-142-74-140.range86-142.btcentralplus.com) |
| 21:54.56 | ManxPower | mvanbaak: Digium has an unofficial presence on this channel, just not an official one dedicated to support. |
| 21:55.22 | mvanbaak | yeah |
| 21:55.27 | mvanbaak | and I understand why |
| 21:55.38 | mvanbaak | we give support on IRC as well with our company |
| 21:55.55 | mvanbaak | and you will need a fulltime gal/guy on the channel |
| 21:55.59 | mvanbaak | that's just too much money |
| 21:57.42 | greekguy8888 | anyone have knowledge on queues and transfers using the sip transfer on polycoms? |
| 22:01.14 | tzafrir_home | nhuisman_work, #rpath ? |
| 22:02.32 | darthahmed | hi everyone |
| 22:08.53 | jblack | what's with these polycoms, that everyone is always asking for help on? |
| 22:09.20 | jblack | Are they either bad devices or incredibly common? |
| 22:11.03 | darthahmed | @jblack they r a pain to configuure |
| 22:11.04 | BBHoss | jblack, non-trivial configuration |
| 22:11.53 | darthahmed | and not remotely worth the pain |
| 22:12.57 | BBHoss | darthahmed, they are nice, well built |
| 22:13.12 | BBHoss | but lets not start that shit again |
| 22:13.26 | darthahmed | :) |
| 22:15.41 | hmmhesays | does zaptel check for dialtone before trying to send a call out a particular fxo channel? |
| 22:18.38 | nhuisman_work | man, digium better fix their damn iso they are shipping |
| 22:18.55 | nhuisman_work | it has a python error on the installer which prevents you from choosing anything except express install |
| 22:19.07 | nhuisman_work | and then the config file for the repository on their rpath linux is wrong so you can't update it |
| 22:21.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:22.55 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 22:24.06 | fiXXXerMet | Problem with my phones. I am able to call other extensions, including my own, and I can get a dial tone, but after making/receiving certain calls (like, other extensions), I do not have any ring tone after dialing the number? |
| 22:24.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-244-43-191.dsl.hstntx.swbell.net) |
| 22:24.54 | nhuisman_work | could someone do me a favor and browse to http://ifa.hawaii.edu |
| 22:24.57 | nhuisman_work | tell me if it is working |
| 22:25.05 | fiXXXerMet | Not working. |
| 22:25.08 | nhuisman_work | ah good |
| 22:25.13 | nhuisman_work | at least it isn't just me |
| 22:25.25 | jblack | Nobody and no thing should work in hawaii. They should just go to the beach. |
| 22:25.43 | nhuisman_work | tell that to my asterisk install which isn't working yet :P |
| 22:25.44 | jblack | No answer here. PRobably took my beach advice |
| 22:25.57 | dacs | is there is a way i can access my ATA webconfiguration remotly, if my publick ip is xx.xx.xx.xx and my ATA is 192.168.1.100 |
| 22:26.15 | _ShrikE | maybe its boycotting the warriors display at the sugar bowl. |
| 22:26.31 | nhuisman_work | man they got owned |
| 22:26.35 | nhuisman_work | what was their display? |
| 22:26.41 | nhuisman_work | or do you mean their performance |
| 22:26.42 | dacs | s/publick/public/ |
| 22:26.53 | _ShrikE | yeah.. i felt kind of bad for them... yeah i meant performance. |
| 22:27.23 | *** join/#asterisk PepOSX (n=pepOSX--@190.78.221.19) |
| 22:27.30 | _ShrikE | dacs: that would need to be configured in your firewall. |
| 22:27.45 | *** join/#asterisk twitchnln (n=twitch@cpe-orncorp.dktc.atl.oneringnetworks.net) |
| 22:27.58 | *** part/#asterisk twitchnln (n=twitch@cpe-orncorp.dktc.atl.oneringnetworks.net) |
| 22:28.38 | dacs | _ShrikE: how, can you plese point some guide for me |
| 22:28.56 | _ShrikE | read the port forwarding section of your firewall manual |
| 22:29.16 | BBHoss | dacs, or you could use ssh tunneling/vpn |
| 22:32.52 | dacs | _ShrikE: BBHoss thank you guys, will look into it |
| 22:34.54 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 22:38.15 | *** join/#asterisk DaveCanoe (n=Dave@H49.C20.B96.tor.eicat.ca) |
| 22:38.33 | dacs | i got a question about the * book, chapter 4. Initial Config of *. it is listed that if i am going to use SIP insteade of FXS/FXO or IAX i should skip to SIP config section,pg 82 "Just as we did with the extensions.conf file; run the following commands |
| 22:39.00 | dacs | in your bash shell:" i am having a hard time getting my extention to work. |
| 22:39.26 | [TK]D-Fender | dacs, perhaps you could show us the actual problem. Pastebin is your friend. |
| 22:39.54 | [TK]D-Fender | ~pb |
| 22:39.55 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:40.06 | dacs | [TK]D-Fender: i don't understand how to config the extention.conf, the syntax of it |
| 22:41.06 | [TK]D-Fender | dacs, thats what Chapter 5 is all about. This is the absolutely most important part of * to learn. |
| 22:41.53 | [TK]D-Fender | dacs, its all explained in there in detail. |
| 22:42.38 | dacs | [TK]D-Fender: so i should just contiue reading , because i thought after i config'd my sip.conf i should confige the extention.conf so that i can get * to work |
| 22:42.52 | jblack | Personally, I think the rest of the book is a cover for Chapter 5 and appendix B |
| 22:43.13 | [TK]D-Fender | dacs, wel yes, extensions.conf is next in line. But if you don't understand it at all well you're just going to have to read now aren't you? |
| 22:43.46 | dacs | yep, i just want to make sure i am on the right path |
| 22:43.52 | [TK]D-Fender | jblack, largely yes |
| 22:44.14 | dacs | [TK]D-Fender: thank you , i will continue to read. |
| 22:44.22 | [TK]D-Fender | dacs, well you need to prove that your work in sip.conf is right so you're going to need to have something to be able to dial. |
| 22:44.30 | dacs | i just can't wait to get my system up and running |
| 22:44.41 | jblack | dacs: Heh. this sounds so familiar to me. |
| 22:44.51 | dacs | jblack: lol |
| 22:44.58 | dacs | [TK]D-Fender: you right |
| 22:45.16 | jblack | Do you understand what contexts are yet? |
| 22:45.28 | dacs | jblack: nope |
| 22:45.40 | jblack | Ok. I know exactly where you're at. |
| 22:45.47 | dacs | hahah |
| 22:45.52 | jblack | You've already played with them some, if you did sip.conf and iax.conf. |
| 22:45.54 | dacs | go read from the begining |
| 22:45.58 | jblack | Basically, they're logical groupings for things. |
| 22:45.59 | [TK]D-Fender | dacs, Chapter 5 awaits you. You can't really shortcut this with dedicated training. |
| 22:46.40 | dacs | jblack: not yet iax i just setup my sip.conf |
| 22:46.45 | jblack | Ok. |
| 22:46.55 | dacs | i will just continue reading |
| 22:47.43 | jblack | That's best. I'll be here for hours, if you get confused by anything you read. |
| 22:48.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584480.dsl.bell.ca) |
| 22:48.34 | dacs | jblack: thank you , i will come here and ask you then, i really appreciate it |
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| 22:52.14 | De_Mon | I never read a book about asterisk and I turned out *okay* |
| 22:52.33 | De_Mon | but I'm special so pretend I didn't say anything... :X |
| 22:53.30 | darthahmed | does anyone have any guides for upgrading from 1.2 to 1.4? |
| 22:54.37 | darthahmed | chanzap is doing my head in |
| 22:54.50 | De_Mon | darthahmed asterisk documentation includes that info in the UPGRADE.txt file |
| 22:55.16 | darthahmed | thanks De_Mon |
| 22:55.39 | [TK]D-Fender | De_Mon, thats why you get to ride the "little bus" ;) |
| 22:55.52 | De_Mon | ^_~ |
| 22:56.05 | darthahmed | i follwed that, but it doesnt say anything regarding install order |
| 22:56.17 | darthahmed | or version issues |
| 22:56.31 | darthahmed | i just got the latest versions of zap,lib and * |
| 22:56.40 | darthahmed | and compiled in that order |
| 22:56.44 | [TK]D-Fender | darthahmed, go download THE BOOK. It describes in detail how to insttall *. |
| 22:56.46 | [TK]D-Fender | ~book |
| 22:56.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:57.12 | darthahmed | is that 1.4 based? |
| 22:57.19 | darthahmed | i got the old book |
| 22:57.20 | jblack | Oh, I didn't know there was a pdf for it. |
| 22:57.24 | [TK]D-Fender | darthahmed, in order : libpri, zaptel, asterisk, then add-ons |
| 22:57.34 | darthahmed | aaaaah |
| 22:57.39 | [TK]D-Fender | darthahmed, yes its for 1.4 |
| 22:57.39 | darthahmed | so lib first? |
| 22:57.48 | darthahmed | thans man!!!! |
| 22:58.14 | jblack | They should put in a chapter about how to make friends, so that people will actually call. |
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| 22:59.00 | De_Mon | we should start asking for the ISBN of "The book" before providing support ;) |
| 22:59.10 | darthahmed | :) |
| 23:00.00 | darthahmed | i'll just ask botman :) |
| 23:00.18 | [TK]D-Fender | jblack, thats a different book by Dale Carnegie |
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| 23:00.58 | jblack | Win friends? I have that somewhere. |
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| 23:06.36 | darthahmed | book says zaptel lib then * |
| 23:07.07 | tzafrir_home | jblack, now that should be trivial: echo 'friends:' >> Makefile; make friends |
| 23:07.22 | tzafrir_home | darthahmed, the order between them doesn't matter |
| 23:07.41 | darthahmed | i know it shouldnt |
| 23:07.44 | tzafrir_home | libpri is generally trivial and faster to build, so start from it |
| 23:07.52 | darthahmed | as long as lib preceeds * |
| 23:08.13 | jblack | tzafrir: Don't forget to touch and strip them |
| 23:08.30 | darthahmed | just being careful here cos of the problem i had earlier today |
| 23:09.02 | darthahmed | i did zap,lib,* and when i ran menuselect chanzap was disabled |
| 23:09.14 | darthahmed | i taught it migh have to do with the compile order |
| 23:09.33 | De_Mon | BajaEd you touch and strip ALL your friends? |
| 23:09.39 | De_Mon | whoops jblack |
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| 23:13.15 | jblack | De_Mon: Having already established that I don't have any? Yes. |
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| 23:35.53 | De_Mon | jblack well, I don't think you'll keep any friends with that particular policy |
| 23:36.35 | De_Mon | http://pastebin.com/m3e0012fa |
| 23:37.00 | De_Mon | remember how I wanted a search/replace function a while ago? Well, I came up with a great implimentation |
| 23:38.37 | De_Mon | I just wish there was a way I could get by without setting all those variables before hand |
| 23:40.59 | CrashSys | Anyone had good luck takin a 1.2 dialplan to 1.4? |
| 23:41.19 | [TK]D-Fender | CrashSys, yup |
| 23:41.33 | CrashSys | Any noteworthy gotcha's? or pretty much fires right up? |
| 23:41.35 | [TK]D-Fender | CrashSys, if your setup was fully 1.2 compliant, then it'll be fine in 1.4 for the mostpart |
| 23:41.48 | [TK]D-Fender | CrashSys, ready upgrade.txt and the other docs. |
| 23:42.05 | CrashSys | will do |
| 23:45.00 | CrashSys | you just use 1.2 sounds with 1.4 or is it included in the 1.4 tarball? |
| 23:48.05 | [TK]D-Fender | CrashSys, upgrade the whole pile. |
| 23:48.16 | [TK]D-Fender | CrashSys, and like Nike said "Just Do It". |
| 23:48.50 | CrashSys | lol |
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| 23:56.24 | NovceGuru | im glad broadvoice's site goes down in the middle of signing up :( |
| 23:56.55 | riddlebox | lol |
| 23:57.21 | outtolunc | before or after the payment <G> |
| 23:57.31 | NovceGuru | outtolunc: after HAHA |
| 23:57.35 | outtolunc | sweet |
| 23:59.04 | NovceGuru | We are sorry. |
| 23:59.05 | NovceGuru | We have encountered an error. |
| 23:59.05 | NovceGuru | A detailed report have been sent to our engineering department. |
| 23:59.36 | NovceGuru | ffs |