| 00:08.51 | *** join/#asterisk rilliam (~kris@adsl-67-116-254-228.dsl.pltn13.pacbell.net) |
| 00:08.55 | rilliam | hey guys is there a channel for freepbx for noobs? |
| 00:09.19 | WIMPy | #freepbx? |
| 00:09.31 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 00:15.14 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 00:15.24 | roxdragon | How to install asterisk ultimate release on debian squeeze? |
| 00:16.52 | Gugge | roxdragon: what ultimate release? |
| 00:17.43 | roxdragon | Gugge, release of asterisk |
| 00:17.47 | roxdragon | 1.8? |
| 00:18.31 | russellb | it's kind of like Windows Vista Premium Signature Bill Gates Edition |
| 00:18.47 | russellb | but it's the Asterisk Ultimate Edition |
| 00:18.50 | Gugge | 1.8.ultimate :) |
| 00:19.35 | russellb | you can install from source ... or in a few days we'll have a debian package repository available with 1.8 packages :-) |
| 00:19.48 | russellb | for a source install, see the Installation chapter in this book |
| 00:19.50 | russellb | ~newbook |
| 00:19.50 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 00:20.24 | roxdragon | Gugge, it's possible installation asterisk 1.8 on debian squeeze? |
| 00:20.43 | russellb | didn't I just answer that? |
| 00:21.26 | Gugge | it looks like it |
| 00:22.08 | roxdragon | there's the package? no? |
| 00:22.20 | roxdragon | on repository |
| 00:22.28 | russellb | not yet |
| 00:22.57 | roxdragon | what version is in the repo? |
| 00:30.10 | roxdragon | if I install asterisk from the repo, then can upgrade to version 1.8 without compile?? |
| 00:33.07 | p3nguin | I have to assume that ultimate release means the latest, most recent version available. |
| 00:42.00 | roxdragon | p3nguin, version use? |
| 00:49.12 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:59.41 | roxdragon | p3nguin, ping |
| 01:03.44 | roxdragon | it's possibile don't make asterisk? |
| 01:04.53 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:05.14 | russellb | roxdragon: what language do you speak natively? There might be an easier place for you to get help |
| 01:05.26 | roxdragon | i'am italian sorry |
| 01:05.56 | russellb | Hm, ok, no #asterisk-it it seems ... |
| 01:06.53 | roxdragon | russellb, what are the commands to install asterisk on debian without filling in anything? |
| 01:07.03 | russellb | I don't understand the question |
| 01:07.10 | roxdragon | there is no oneon that channel |
| 01:07.13 | russellb | I don't know what "without filling in anything" means |
| 01:07.25 | russellb | there is an installation chapter available in this book: |
| 01:07.27 | russellb | ~newbook |
| 01:07.27 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 01:08.02 | roxdragon | without compile asterisk * russellb |
| 01:08.20 | russellb | that is not an option right now if you want asterisk 1.8 |
| 01:09.27 | roxdragon | if I install Asterisk 1.6 from repo , I can upgrade later to 1.8? without compile |
| 01:09.34 | russellb | yes |
| 01:09.42 | roxdragon | ok :) |
| 01:10.23 | roxdragon | I have to install addons? dahdi? |
| 01:10.39 | roxdragon | What are those packages? |
| 01:11.03 | russellb | if you don't know that you need them, you probably don't need them |
| 01:12.13 | roxdragon | What does dahdi? |
| 01:13.37 | jaytee | used for analog or T1/E1, not needed if you're just running SIP |
| 01:14.03 | jaytee | and timing if you want to use MeetMe |
| 01:14.19 | roxdragon | and addons? |
| 01:15.09 | jaytee | extra stuff for other voip protocols, mysql_cdr connector for call detail, etc. |
| 01:15.19 | russellb | stuff you probably don't need :-) |
| 01:15.24 | jaytee | yeah |
| 01:16.13 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-57.telkomadsl.co.za) |
| 01:16.16 | roxdragon | I have to install the audio codec? |
| 01:18.31 | jaytee | if I recall back when I installed once from packages it installed all the non-licensed codecs for you and you just allow or deny them per device in the config files |
| 01:18.47 | jaytee | but I usually compile from source |
| 01:19.06 | jaytee | and you can choose which ones to load |
| 01:19.26 | roxdragon | i used apt-get install asterisk |
| 01:19.44 | jaytee | that makes sense for a debian package install |
| 01:20.09 | jaytee | I'd do something like yum -y install asterisk but that's not how I roll :-) |
| 01:20.44 | roxdragon | ok ok :) |
| 01:22.26 | jaytee | so where in Italy are you? |
| 01:28.58 | jaytee | russellb, so there's an Asterisk Cookbook in the works following along behind The Definitive Guide? |
| 01:32.46 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
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| 01:42.51 | roxdragon | jaytee, the nokia n95 is not registered to asterisk |
| 01:43.10 | roxdragon | on asterisk 1.4 it's ok but 1.6 no |
| 01:43.14 | roxdragon | don't work |
| 01:54.37 | roxdragon | help Scheduling destruction of SIP dialog '1287194928@192.168.1.4' in 32000 ms (Method: REGISTER) |
| 01:54.37 | roxdragon | Really destroying SIP dialog '302602635@192.168.1.4' Method: OPTIONS |
| 01:54.37 | roxdragon | Really destroying SIP dialog '1287194928@192.168.1.4' Method: REGISTER |
| 01:54.56 | roxdragon | 192.168.1.4 is a nokia... don't work |
| 02:00.49 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 02:01.57 | joobie | guys when i dial my polycom ip 7000, i seem to have a bit more delay on the phone than the regular ip 330's |
| 02:02.04 | joobie | is terhe anything i can do to try and improve this? |
| 02:03.16 | p3nguin | Sell the 7000 and buy another 330. |
| 02:04.55 | p3nguin | Is there a popular SMS gateway to use with Asterisk? I'm going to need to send text messages from Asterisk. |
| 02:05.29 | joobie | p3nguin, only problem is the 330 aint a conference phone :P |
| 02:05.51 | p3nguin | Heh, that could pose a problem. I'm not a Polycom user, so I had no idea. |
| 02:06.11 | joobie | p3nguin, i just tapped into my local SMS provider (smsglobal.com.au) |
| 02:06.17 | p3nguin | If I ever need a conference phone, that's probably the kind I'd get, though. |
| 02:06.19 | joobie | wrote a custom script and used AGI |
| 02:06.48 | p3nguin | joobie: Anything that can be shared, or is it restricted code? |
| 02:09.17 | roxdragon | help nokia don't work with asterisk |
| 02:09.18 | roxdragon | http://paste.ubuntu.com/581893/ |
| 02:16.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 02:20.16 | russellb | jaytee: yep |
| 02:20.51 | russellb | jaytee: just finished the first version ... ebook only for now, and it's quite short to start with (50 pages or so). It's on ofps.oreilly.com for review right now |
| 02:27.57 | joobie | p3nguin, our script has a lot of other stuff within.. i dont mind distro'ing it if you are stuck - but i'll need to strip out a lot of other stuff around it before |
| 02:28.15 | joobie | p3nguin, most of these SMS gateways provide you with sample scripts you can use to do this though |
| 02:28.19 | joobie | i know smsglobal do |
| 02:29.18 | roxdragon | From: "asterisk" <sip:asterisk@192.168.1.3>;tag=as07cb30c0 |
| 02:29.18 | roxdragon | To: <sip:192.168.1.4> |
| 02:29.20 | roxdragon | qhy? |
| 02:29.23 | roxdragon | why? |
| 02:29.47 | roxdragon | 192.168.1.4 = internal " 402 " |
| 02:30.13 | p3nguin | joobie: Maybe you can just point me in the right direction. I need to start sending SMS from Asterisk. The project requires that when a person calls in and presses a defined key, it sends a pre-written message by text to cell phone. I'm an engineer, not a programmer, so I can set it up but I can't write code if it's very complicated. |
| 02:36.11 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 02:37.51 | *** join/#asterisk jaminja (~jaminja@unaffiliated/jaminja) |
| 02:38.50 | joobie | p3nguin, where are you located? |
| 02:39.00 | *** join/#asterisk coppice (~chatzilla@m121-203-211-159.smartone-vodafone.com) |
| 02:39.08 | joobie | p3nguin, first try to find a sms service (like SMS global) that is in your area |
| 02:39.29 | joobie | then check if they have scripts you can use which interface to their SMS gateway (95% of the places that offer this service will have this) |
| 02:40.18 | joobie | the scripts will likely support arguements when you call teh script, so eg "./script <mobile number> <message>". Then juse use AGI in asterisk to interface the script to your dialplan when someone presses the number |
| 03:00.05 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 03:21.04 | p3nguin | joobie: I'm in the US. I don't really know what I'm looking for. |
| 03:29.48 | joobie | p3nguin, look for sms gateways in the US |
| 03:30.18 | joobie | then look at which ones of those have perl / bash scripts that you can use to send SMS's with |
| 03:30.26 | joobie | (they will have sample scripts) |
| 03:30.39 | joobie | then set tha tup on your asterisk box and integrate to asterisk via AGI |
| 03:30.47 | p3nguin | joobie: Is this typically a service that I have to pay for, or is it free? |
| 03:32.32 | joobie | you have to pay for it |
| 03:34.04 | p3nguin | I'm seeing 3 - 16 cents per SMS message. |
| 03:34.27 | p3nguin | Typically 3-5 cents. 16 cents isn't that common. |
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| 04:50.21 | Dovid | morning ev1 |
| 05:09.39 | Corydon76-home | p3nguin: honestly, you shouldn't have to pay anything for SMS. It was designed to run in unused space on the telephone grid. |
| 05:10.26 | Corydon76-home | That's why there's a limit of 160 characters per message |
| 05:10.40 | p3nguin | This project will potentially generate a lot of SMS messages, so that would be nice if I don't have to pay anything. |
| 05:17.31 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 05:29.10 | Corydon76-home | Welcome to the wonderful world of telephony, where executives will jump at the chance to serve you garbage and make you pay for it |
| 05:30.15 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:30.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 06:07.02 | drcode | hi all |
| 06:07.25 | drcode | I want to install asteriks on ubuntu 10.04 |
| 06:07.47 | drcode | do I need mysql or I can use light sql like sqlite? |
| 06:08.06 | p3nguin | Do you /want/ a database? |
| 06:08.18 | p3nguin | You don't need one for Asterisk. |
| 06:09.05 | drcode | ok |
| 06:09.26 | drcode | I have temp users that register |
| 06:09.34 | p3nguin | Oh yeah? |
| 06:09.41 | drcode | dose astriks save those users |
| 06:09.57 | drcode | or I can tell ast that those users are temp? |
| 06:09.58 | p3nguin | Do you mean Asterisk? |
| 06:10.04 | drcode | yes |
| 06:10.59 | p3nguin | I'm not entirely sure what you're talking about. No devices can register unless you configure them in Asterisk. |
| 06:11.16 | p3nguin | Those configured devices can come and go as they please. |
| 06:12.18 | drcode | I mean iax users |
| 06:12.30 | drcode | I have iax users that sign to astriks |
| 06:12.31 | p3nguin | You probably means iax2 devices. |
| 06:12.45 | drcode | by default users are temporary? |
| 06:12.46 | p3nguin | And there you go with this astriks thing again. |
| 06:12.55 | p3nguin | This channel is for Asterisk. |
| 06:12.58 | p3nguin | not astriks |
| 06:12.58 | drcode | oops sorry\ |
| 06:13.04 | drcode | asterisk |
| 06:13.30 | p3nguin | You'll configure the devices on Asterisk. The device can come and go as necessary. |
| 06:13.47 | drcode | ok |
| 06:13.50 | drcode | thanx p3nguin |
| 06:14.16 | p3nguin | I'll assume you mean "thanks," since I did take my time to explain this stuff to you. |
| 06:14.40 | p3nguin | If you have more questions about Asterisk, this is the place to ask. |
| 06:18.17 | drcode | ok |
| 06:18.42 | drcode | I am looking also for video confrence h264 with sip |
| 06:19.33 | drcode | asterisk can also work as video confrence? |
| 06:19.38 | kaldemar | drcode: asterisk does not support video conferencing at the moment. |
| 06:20.15 | Corydon76-home | And probably will never, without video mixing hardware |
| 06:20.37 | Corydon76-home | Patent issues are the main problem |
| 06:22.24 | drcode | I see |
| 06:22.43 | drcode | I didn't know that video mixing is patent |
| 06:23.48 | kaldemar | it isn't, but codecs are. |
| 06:23.59 | drcode | I see |
| 06:24.05 | drcode | thanks to all |
| 06:33.14 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 06:42.32 | coppice | if google succeed in proven VP8 is patent free, maybe things will change. the video codecs need a lot of CPU power, though |
| 06:42.47 | coppice | s/proven/proving |
| 06:44.04 | Corydon76-home | coppice: yeah, but the capability of a single machine is continually improving, in terms of processing power. It might not be terribly doable today, but in 18 months? |
| 06:44.23 | Corydon76-home | It's certainly a lot more doable today than it was just 2 years ago |
| 06:45.13 | coppice | it depends a lot on the resolution you expect. CPUs are fairly good at decompressing video, but compressing it is still a challenge |
| 06:45.18 | Corydon76-home | but yes, processing power is the secondary issue |
| 06:45.46 | Corydon76-home | We can deal with the processing issue, if we first deal with the patent issue |
| 06:46.25 | coppice | MPEG-LA now oversees 1700 patents on codecs :-\ |
| 06:46.54 | Corydon76-home | yeah, exactly... a minefield, to be sure |
| 06:47.14 | *** join/#asterisk manji (~manjiki@ppp-94-65-221-48.home.otenet.gr) |
| 06:47.56 | coppice | even something like G.729 is a pain. it seems to have a well defined way to licence the patents, but you get lawyers letter from some extra people nobody warned you about one you start shipping product :-\ |
| 06:48.02 | Corydon76-home | Is that 1700 number counting duplicate patents filed under different patent jurisdictions? |
| 06:48.34 | coppice | I don't know. I also have no idea how many may be nearing expiry, and how many are fairly fresh |
| 06:48.51 | Corydon76-home | I thought the license on G.729 included patent indemnification |
| 06:49.38 | Corydon76-home | Safest bet is simply to wait until the last one expires. For g.729, that's sometime in 2014, I think |
| 06:49.52 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:49.53 | coppice | indemnification against additional patents? nope. There is at least one on G.729B which is very solid, and which the pool doesn't warn you about |
| 06:50.09 | Corydon76-home | Joyful |
| 06:50.33 | coppice | its later than that. I think G.723.1 is clear in 2014 |
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| 07:00.04 | Corydon76-home | I thought G.723.1 was actually clear in 2012 |
| 07:02.07 | kaldemar | do you guys happen to have any sources at hand for those thoughts? it's pretty hard to find any handy material for patent expirations. |
| 07:03.13 | Corydon76-home | kaldemar: it will depend upon the actual jurisdiction. Different countries have differing patent systems. |
| 07:03.59 | Corydon76-home | And so-called "submarine patents" are at play, too |
| 07:05.06 | Corydon76-home | kaldemar: You can do your own research, or you can hire a patent attorney to wade through the patents, which is what you'd need to do if you have any money/business at stake |
| 07:10.12 | kaldemar | the amount of available material and the required effort to find all of it is excruciating. |
| 07:11.44 | Corydon76-home | kaldemar: I don't disagree. But it's been awhile since I did the research |
| 07:12.11 | Corydon76-home | I was prohibited while working for Digium from doing any such research, for liability reasons |
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| 07:16.54 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:16.56 | schmidts | good morning |
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| 07:25.21 | *** join/#asterisk Tim_Toady (~moi@77.49.233.214) |
| 07:40.18 | coppice | Corydon76-home: We did a check recently, and G.723.1 still seemed to be encumbered until 2014. Someone else said 2012, so we looked to see if we could do a free G.723.1 ready for next year, but it turned out to be later. Its a pain to find the relevant dates, though |
| 07:41.13 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 07:45.55 | coppice | Corydon76-home: It looks like G.729 may be clear in 2016, but nobody uses that. they use G.729A, which I think has additional and later expiring patents on it |
| 07:47.00 | Corydon76-home | coppice: I wonder how many might use the bare G.729, if they didn't have to pay any royalties at that time |
| 07:47.41 | coppice | well, it doubles the CPU load, but it sounds better too |
| 07:48.01 | Corydon76-home | When it comes to those consortiums, you usually end up either having to pay all of them according to their scheme |
| 07:48.32 | Corydon76-home | and then if you have the patent license, you might as well use everything you have a license for |
| 07:49.09 | coppice | not really. you incrementally pay for most of the G.729 annexes, which seem to have extra patents on each of them. G.729 and G.729A are licenced as a bundle, though |
| 07:51.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:52.31 | creativx | hmm.. if i dont specify a timeout in queue(), a call will never ever timeout in a queue no matter what i set under timeout= in queue.conf? |
| 07:52.31 | Corydon76-home | No, I meant that you're paying fees even to organizations whose patents have expired... |
| 07:53.43 | Corydon76-home | creativx: the call will never timeout OUT of a queue, no |
| 07:54.11 | coppice | when the patents expire the remaining stake holders kick out the looser, but they don't reduce the price to you |
| 07:54.14 | Corydon76-home | creativx: the timeout within queues.conf is to specify timeouts for individual queue members |
| 07:54.35 | creativx | Corydon76-home: mkay, then I've understood it correctly |
| 07:54.47 | creativx | we had an issue with a sales queue with only one member.. who went to lunch |
| 07:54.47 | Corydon76-home | creativx: two different timeouts, not duplicates |
| 07:54.58 | creativx | yeah I see |
| 07:55.31 | Corydon76-home | coppice: Ah, okay. Yeah, the price never goes down. |
| 07:55.43 | creativx | I've got a macro that wraps the queue() app, so I gotta think up some way to have some queues time out and some not |
| 07:55.57 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 07:56.20 | coppice | and I don't think they actually agree to inform you about patent expiration. its buyer beware |
| 07:56.39 | Corydon76-home | Dog-eat-dog world |
| 07:57.55 | Corydon76-home | I'm not all that surprised, really. (You want to give me money? Okay!) |
| 07:58.18 | coppice | I'm more concerned about fresh new patents. e.g. G.711.0 looks potentially interesting, but with at least 9 patents on it, it seems like a none starter |
| 07:58.31 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 07:59.58 | Corydon76-home | What's the .0 entail? Better encoding/decoding of ulaw/alaw? Or higher bandwidth (like what G.722 provides)? |
| 08:10.50 | *** join/#asterisk lftsy (~lftsy@212-147-16-108.fix.access.vtx.ch) |
| 08:12.50 | coppice | G.711.1 is a weird wideband extension for G.711. G.711.0 is further lossless compression of signals that have already been subjected to lossy G.711 coding |
| 08:12.54 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:14.18 | coppice | being lossless, its variable bit rate, which tends to defeat encryption. That detracts somewhat |
| 08:19.26 | *** join/#asterisk bip (~bip@unaffiliated/bip) |
| 08:19.46 | bip | how do I find the dadhi version I'm running ? |
| 08:19.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 08:23.02 | Tim_Toady | bip from ur package manager or if its installed by hand you can try something like : strings /lib/modules/`uname -r`/dahdi/dahdi.ko | grep version |
| 08:23.45 | kaldemar | or just "dahdi show version" in asterisk CLI. |
| 08:23.49 | bip | thanks kaldemar |
| 08:24.15 | bip | do you jnow if any files shows the asteriskNow version as well ? |
| 08:24.31 | kaldemar | no, i've never used it. |
| 08:25.12 | bip | I m writing the digium support and i want to be sure I'm givin em every possible information |
| 08:25.34 | bip | DAHDI Version: 2.3.0.1 Echo Canceller: MG2 |
| 08:26.43 | Tim_Toady | bip cat /etc/asterisknow-version |
| 08:27.49 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 08:27.52 | bip | thanks Tim_Toady |
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| 08:52.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 09:12.17 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 09:12.20 | roxdragon | hi all |
| 09:16.09 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 09:22.39 | *** join/#asterisk recluze (~recluze@175.145.106.157) |
| 09:22.51 | recluze | I have a strange problem with Asterisk on a zap channel. The call connects, rings a few times and then gets disconnected -- but nothing on the full log to indicate the hangup ... http://pastebin.com/d2shTmqQ |
| 09:22.59 | recluze | any hints on how to troubleshoot this? |
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| 09:35.11 | roxdragon | help please [Mar 18 10:33:46] NOTICE[1475]: chan_sip.c:20152 handle_request_invite: Call from '401' to extension '403' rejected because extension not found in context 'ipPhones'. |
| 09:35.45 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 09:35.50 | joobie | hey boys |
| 09:36.09 | joobie | my soundpoint ip 7000 phone is a bit delayed.. if i make a call on the soundpoint 330 it's not so bad |
| 09:36.20 | joobie | any ideas as to what i should look into tweaking on the phone? |
| 09:38.56 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 09:41.22 | kaldemar | roxdragon: you have defined context ipPhones for device 401 in sip.conf. your ipPhones context in extensions.conf does not have an extension that matches 403. |
| 09:42.42 | roxdragon | kaldemar, http://paste.ubuntu.com/581992/ |
| 09:43.13 | roxdragon | What's wrong? |
| 09:44.26 | roxdragon | i use asterisk 1.6 |
| 09:46.40 | kaldemar | roxdragon: did you reload dialplan? |
| 09:48.00 | kaldemar | and remove all samples that you don't use. |
| 09:48.43 | kaldemar | also [general] is commented out. |
| 09:49.15 | roxdragon | as reloading the dialplan? |
| 09:49.25 | roxdragon | what's the command |
| 09:49.37 | kaldemar | "dialplan reload" is the command |
| 09:49.54 | kaldemar | ;[general] -> [general] |
| 09:49.54 | roxdragon | Server*CLI> dialplan reload |
| 09:49.55 | roxdragon | No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) |
| 09:50.34 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 09:51.10 | roxdragon | command don't work |
| 09:52.39 | kaldemar | something is really wrong then. enable the general context and restart asterisk. |
| 09:53.54 | roxdragon | ok ;[general] -> [general] |
| 09:54.11 | roxdragon | how to restart asterisk? /etc/init.d/asterisk restart ? |
| 09:54.43 | kaldemar | yes. |
| 09:55.30 | roxdragon | kaldemar, [Mar 18 10:55:13] NOTICE[1687]: chan_sip.c:20152 handle_request_invite: Call from '401' to extension '403' rejected because extension not found in context 'ipPhones'. |
| 09:56.05 | kaldemar | what does "dialplan show" in CLI show? |
| 09:56.57 | roxdragon | http://paste.ubuntu.com/582001/ kaldemar |
| 09:57.31 | kaldemar | interesting. looks like your asterisk doesn't read extensions.conf at all. |
| 09:57.51 | Zhad | kal> thanks for your help yesterday. |
| 09:58.27 | Zhad | went the Dial(,,G()) route in the end |
| 09:58.34 | roxdragon | why ??? kaldemar |
| 09:58.36 | roxdragon | :( |
| 09:58.53 | kaldemar | roxdragon: it does by default. how did you install asterisk? |
| 09:58.57 | Zhad | bets that at somepoint extensions.conf will be renamed dialplan.conf :-) |
| 09:59.25 | kaldemar | the command to reload dialplan used to be "extensions reload". |
| 09:59.32 | roxdragon | kaldemar, apt-get install asterisk |
| 09:59.44 | kaldemar | then all dialplan related commands were put under "dialplan". |
| 10:00.07 | kaldemar | roxdragon: on what system? |
| 10:00.12 | Zhad | It will be something else that catches all the people that don't read UPGRADING.txt |
| 10:01.21 | roxdragon | kaldemar, debian squeeze... the command for dialplan is Server*CLI> reload extconfig |
| 10:01.21 | roxdragon | Server*CLI> |
| 10:02.16 | kaldemar | "reload extconfig"? |
| 10:02.19 | roxdragon | yes |
| 10:02.53 | roxdragon | Connected to Asterisk 1.6.2.9-2+squeeze1 currently running on Server (pid = 1671) |
| 10:03.10 | Zhad | 1.6.2.9 is fairly old now |
| 10:03.41 | kaldemar | roxdragon: where did you find that command? |
| 10:04.17 | kaldemar | roxdragon: and what does "dialplan reload" tell you? |
| 10:05.02 | roxdragon | kaldemar, on a forum |
| 10:05.32 | roxdragon | Zhad, Only install software from the repo |
| 10:05.50 | kaldemar | roxdragon: that command will not help you. |
| 10:06.01 | kaldemar | if it even is a command. |
| 10:06.15 | Zhad | Is there not a newer one in backports? |
| 10:06.25 | kaldemar | roxdragon: what does "module show like pbx_config" give you? |
| 10:06.52 | *** part/#asterisk recluze (~recluze@175.145.106.157) |
| 10:06.57 | *** join/#asterisk tallship (~kvirc@cpe-76-172-48-131.socal.res.rr.com) |
| 10:07.41 | roxdragon | not yet Zhad |
| 10:07.43 | roxdragon | kaldemar, pbx_config.so Text Extension Configuration 0 |
| 10:07.52 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 10:08.18 | roxdragon | we can try to make him upload the file extensions.conf us? in manual mode? |
| 10:08.20 | kaldemar | roxdragon: and "module reload pbx_config.so"? |
| 10:08.39 | roxdragon | Server*CLI> "module reload pbx_config.so |
| 10:08.39 | roxdragon | No such command '"module reload pbx_config.so' (type 'core show help module reload pbx_config.so' for other possible commands) |
| 10:08.46 | tallship | Anyone know what voipjet was talking about in their announcement that all kinds of Asterisk boxes are getting hacked? |
| 10:08.47 | kaldemar | without "'s |
| 10:09.52 | Sertys | tallship: huh? |
| 10:09.55 | Sertys | where'd u get that? |
| 10:11.00 | roxdragon | what's the command write? |
| 10:11.02 | kaldemar | roxdragon: no " in the command, try again without them. |
| 10:11.12 | roxdragon | module reload pbx_config.so |
| 10:11.14 | roxdragon | ? |
| 10:11.14 | kaldemar | roxdragon: module reload pbx_config.so |
| 10:11.19 | roxdragon | ok |
| 10:11.45 | roxdragon | Server*CLI> module reload pbx_config.so |
| 10:11.45 | roxdragon | [Mar 18 11:11:33] NOTICE[1715]: loader.c:686 ast_module_reload: The module 'pbx_config.so' was not properly initialized. Before reloading the module, you must run "module load pbx_config.so" and fix whatever is preventing the module from being initialized. |
| 10:13.12 | wdoekes2 | roxdragon: and? |
| 10:13.17 | kaldemar | roxdragon: module load pbx_config.so |
| 10:15.18 | roxdragon | Server*CLI> module load pbx_config.so |
| 10:15.18 | roxdragon | Unable to load module pbx_config.so |
| 10:15.19 | roxdragon | Command ' module load pbx_config.so' failed. |
| 10:18.23 | kaldemar | roxdragon: remove everything from extensions.conf except [general] and the stuff that you put there yourself. then try again. |
| 10:20.19 | roxdragon | ok... i have restart asterisk and module load pbx_config.so |
| 10:20.20 | roxdragon | Unable to load module pbx_config.so |
| 10:20.20 | roxdragon | Command 'module load pbx_config.so' failed. |
| 10:20.37 | roxdragon | kaldemar, http://asteriskfaqs.org/2011/02/18/asterisk-users/ast_compile_ael2.html ????????? |
| 10:21.47 | tallship | Sertys: it's a bulletin dated 16 March as soon as I login. |
| 10:22.19 | tallship | dunno. It's the only place I've seen any mention of it. |
| 10:22.31 | kaldemar | roxdragon: you are not using ael. |
| 10:22.42 | Sertys | is ti possible there's a new * xploit out there? |
| 10:23.11 | tallship | That's kinda what they inferred. |
| 10:23.21 | roxdragon | :( |
| 10:23.44 | kaldemar | roxdragon: and the ael module seems to be working fine on your install. |
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| 10:24.49 | roxdragon | <PROTECTED> |
| 10:25.45 | kaldemar | roxdragon: try "module unload pbx_config.so" and "module load pbx_config.so". if it still does not work, pastebin your extensions.conf gain. there must be something wrong with it. |
| 10:26.17 | roxdragon | http://paste.ubuntu.com/582009/ kaldemar |
| 10:26.21 | roxdragon | ok one moment |
| 10:27.21 | roxdragon | kaldemar, http://paste.ubuntu.com/582011/ |
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| 10:28.14 | roxdragon | my extensions.conf http://paste.ubuntu.com/582012/ kaldemar |
| 10:29.33 | roxdragon | kaldemar, ; This configuration file is reloaded |
| 10:29.34 | roxdragon | ; - With the "dialplan reload" command in the CLI |
| 10:29.37 | roxdragon | :\ |
| 10:30.37 | *** join/#asterisk SeTTleR (~bernd@p5DDED197.dip.t-dialin.net) |
| 10:31.27 | kaldemar | roxdragon: you removed your configurations from extensions.conf. put them back. remove the SAMPLES that you did not write to the file. |
| 10:33.12 | roxdragon | that is the original file |
| 10:33.12 | roxdragon | <PROTECTED> |
| 10:36.20 | kaldemar | roxdragon: it does not work. you need to fix it. |
| 10:36.38 | roxdragon | remove all file? |
| 10:37.02 | roxdragon | Remove all contents of the file? |
| 10:37.26 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 10:37.43 | kaldemar | roxdragon: no. not all. remove the sample contexts that you did not write. leave [general] in the file. then add your own contexts such as [ipPhone]. |
| 10:38.04 | Zhad | Okay, maybe I'm not quite finished with the paging |
| 10:39.08 | Zhad | Using Dial(phone1&phone2&console) etc. as soon as one pciks up, the others don't and Page doesn't have a ,,G() option. |
| 10:39.14 | roxdragon | include => dundi-e164-canonical |
| 10:39.14 | roxdragon | include => dundi-e164-customers |
| 10:39.14 | roxdragon | include => dundi-e164-via-pstn |
| 10:39.18 | roxdragon | this? |
| 10:41.13 | kaldemar | roxdragon: yes, that also. |
| 10:42.20 | Zhad | maybe wrap the whole thing around an internal dial |
| 10:47.34 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
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| 10:48.00 | roxdragon | kaldemar, [globals] |
| 10:48.00 | roxdragon | CONSOLE=Console/dsp; Console interface for demo |
| 10:48.00 | roxdragon | ;CONSOLE=DAHDI/1 |
| 10:48.01 | roxdragon | ??? |
| 10:48.11 | roxdragon | comment all? |
| 10:48.20 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 10:48.46 | Zhad | okay, wrapping it in a Dial doesn't work, it ignores the Dial options and still stops when it stops |
| 10:49.26 | kaldemar | roxdragon: remove it if you don't use it. |
| 10:51.04 | roxdragon | I do not even use a |
| 10:51.21 | *** join/#asterisk zooz (~zooz@host86-163-12-128.range86-163.btcentralplus.com) |
| 10:51.32 | Severian | iax2 show peers shows me dallasoffice (Unspecified) (D) 255.255.255.255 0 (T) Unmonitored |
| 10:51.32 | Severian | 1 iax2 peers [0 online, 0 offline, 1 unmonitored] I believe the Unspecified might be why my calls are not going through, but I can't figure out why it is unspecified. Is this likely to be a real problem? I can post more details. |
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| 10:52.26 | roxdragon | kaldemar, finish |
| 10:52.43 | roxdragon | i do? |
| 10:58.05 | kaldemar | roxdragon: try to load the module again. if it doesn't work, i don't know what's wrong. |
| 11:00.25 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 11:00.27 | roxdragon | http://paste.ubuntu.com/582024/ kaldemar |
| 11:03.47 | Zhad | is there a way of calling multiple extensions at the same time and conferencing them all without using Page? |
| 11:05.08 | Zhad | and then being able to close the call when caller hangs up |
| 11:05.14 | Zhad | qwithout using Page |
| 11:05.39 | Severian | maybe I should rephrase the problem. I am trying to place a call that should go from 1 asterisk server to another. I keep getting " dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)" and I have not been able to figure out why. There are currently no calls between the servers, because they are both my servers and I am just trying to learn. |
| 11:06.28 | Zhad | what does module show like iax report? |
| 11:06.55 | kaldemar | Zhad: you can do that with an extension that originates calls to a meetme conference, and mark the meetme to close down when the caller exits. |
| 11:07.28 | Severian | chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 1 modules loaded. |
| 11:07.50 | Zhad | kald> makes sense |
| 11:08.16 | bip | http://pastebin.com/a2ykGJfC |
| 11:08.19 | kaldemar | Severian: the other server hasn't registered to you and it is defined as host=dynamic. that's why the address is unspecified. |
| 11:08.43 | roxdragon | :( |
| 11:08.47 | bip | users complains calls get dropped, the pastebin above shows a last call that was dropped |
| 11:09.05 | Zhad | kal> wouldn't the first person bridges to that conference be marked as the caller though, which would need to happen before the atucal caller calls? |
| 11:09.14 | Zhad | s/bridges/bridged/; |
| 11:09.15 | bip | does it give any hint about what might have happened ? |
| 11:09.17 | Severian | kaldemar, How do I make it register? I put a line in the iax.conf in the general section. Is that enough? |
| 11:09.31 | Zhad | s/atucal/actual/; |
| 11:09.35 | kaldemar | Severian: yes, put a register line in iax.conf. |
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| 11:10.26 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 11:10.28 | roxdragon | kaldemar, you recommend installing the 1.8? |
| 11:10.35 | roxdragon | from source? |
| 11:11.08 | kaldemar | roxdragon: depends. i recommend installing from source though, be it 1.8 or 1.6.X. |
| 11:11.26 | roxdragon | i have asterisk 1.6 |
| 11:11.41 | kaldemar | roxdragon: there must be something strange about the package or something related to it on your system. |
| 11:12.31 | roxdragon | the system is clean. Yesterday I installed debian |
| 11:13.05 | kaldemar | Zhad: core show application meetme: "x: Close the conference when last marked user exits" and "A: Set marked mode." |
| 11:13.16 | Zhad | thanks |
| 11:13.18 | roxdragon | kaldemar, if I install from source, then I upgrade? |
| 11:13.34 | *** join/#asterisk defswork (~andy@mx1.3gcomms.co.uk) |
| 11:13.37 | kaldemar | Zhad: use separate extensions or arguments for the caller and callees. |
| 11:13.47 | kaldemar | roxdragon: do not understand. |
| 11:14.50 | roxdragon | Would you help me build in the 1.8? |
| 11:14.55 | Zhad | Actually, it still wouldn't fix the problem that I'm currently having |
| 11:15.41 | Zhad | the Meetme conference generated by Page, doesn't drop the Console/dsp channel quickly enough to Originate a call back to it, but it does drop it too quickly to play anything on it. |
| 11:17.18 | roxdragon | kaldemar, Asterisk 1.8.3.2 |
| 11:17.18 | roxdragon | Source Tarball ? |
| 11:17.30 | kaldemar | roxdragon: download a source package, run contrib/scripts/install_prereq to install missing dependencies, then run ./configure && make && make install. |
| 11:18.01 | kaldemar | roxdragon: http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 11:18.43 | roxdragon | http://www.asterisk.org/downloads kaldemar |
| 11:18.48 | roxdragon | kaldemar, Asterisk 1.8.3.2 |
| 11:18.54 | roxdragon | tar gz it's ? |
| 11:18.56 | roxdragon | ok |
| 11:20.05 | Zhad | hmm Wait doesn't work if there isn't a channel open. |
| 11:24.48 | roxdragon | kaldemar, i have download source 1.8.3.2 but install_prereq not found |
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| 11:29.51 | kaldemar | roxdragon: you need to uncompress the package. google for some installation instructions. |
| 11:30.00 | *** join/#asterisk jkprg (~jarda@ip-62-245-93-150.net.upcbroadband.cz) |
| 11:32.11 | roxdragon | kaldemar, http://paste.ubuntu.com/582043/ |
| 11:34.01 | kaldemar | roxdragon: you paste already tells you what to do. |
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| 12:02.29 | Zhad | ended up playing in aplay |
| 12:02.43 | Shazaum | hi, I have a problem with recordings, they are accelerated, it has to do with SOX? |
| 12:03.07 | roxdragon | kaldemar, Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 12:03.23 | *** join/#asterisk dotcomstu (~stuart@92.40.138.106.threembb.co.uk) |
| 12:08.30 | Shazaum | :( |
| 12:09.04 | kaldemar | roxdragon: you must start asterisk |
| 12:14.31 | Severian | I think I get it. If your IP address is not dynamic, you don't need to register. You just need to create a section, maybe called a context, in the iax2.conf file for the asterisk machine that is sending you calls. Does that sound right? |
| 12:16.50 | Shazaum | echo "I have a problem with recordings, they are accelerated, it has to do with SOX?" |
| 12:18.42 | Gugge | Shazaum: how do you record? |
| 12:22.44 | slim_ | hello all, i'm search for sip server , that can make chat audio/video and text also can integrate with microsoft ocs and telephone integration not needed my question is asterisk is the suitable application for this ? |
| 12:23.09 | ssureshot | morning,, what ways can I get more information on why a phone isnt' registering with asterisk? I can check VM with the phone but cannot dial the extension for the second phone |
| 12:23.32 | ssureshot | upgrade from 1.2 > 1.8 |
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| 12:27.04 | Shazaum | Gugge, mixmonitor(/var/spool/asterisk/monitor/files/tests.wav|b) |
| 12:28.26 | Shazaum | Gugge, im recording with the default options mixmonitor |
| 12:34.59 | Shazaum | well |
| 12:35.01 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:35.15 | Shazaum | Gugge, "issue 17078" - =/ tks |
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| 13:33.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:37.24 | *** join/#asterisk Tim_Toady (~moi@77.49.233.214.dsl.dyn.forthnet.gr) |
| 13:37.43 | ssureshot | if I can call extensions that don't have phones on it.. ieee.. I have a pot where I throw telemarketers... I can ring that and get the message... I can check vm for users with the phone.. but I cannot call extension to extension |
| 13:37.55 | ssureshot | what should I be looking for? |
| 13:38.24 | ssureshot | phones are cisco 7940 and they have an x by the extension on the display |
| 13:39.45 | kaldemar | ssureshot: CLI when you make a call. |
| 13:39.52 | ssureshot | my configs all look correct, phone options all match my sip.conf and my extensions.conf was converted over with few changes like zap to dahdi |
| 13:40.58 | ssureshot | http://inetpro.org/pastebin/10864 << is the cli |
| 13:42.18 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:44.06 | ssureshot | when I show sip peers, my 2 extensions I'm working with show unspecified / unmonitored... |
| 13:49.21 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:51.03 | kaldemar | ssureshot: make them register to asterisk. are the phone settings correct? |
| 13:52.07 | ssureshot | kaldemar: settings are corect as far as I can tell, have copied my tftpboot files and asterisk settings from my old system |
| 13:52.35 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:53.00 | kaldemar | ssureshot: reboot them and watch sip debug for any REGISTER messages from the phones. if you don't see any, asterisk is not the issue. |
| 13:53.01 | ssureshot | dhcp, tftp, etc.. sip file / extensions were transferred,, I've double checked the phone settings to sip file... all looks good |
| 13:53.11 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 13:53.47 | ssureshot | kaldemar: I will post the sip log ina few seconds.. I don't know what I'm looking for |
| 13:54.39 | roxdragon | kaldemar, root@Server:~# asterisk -r |
| 13:54.39 | roxdragon | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 13:54.39 | roxdragon | root@Server:~# /etc/init.d/asterisk restart |
| 13:54.39 | roxdragon | -bash: /etc/init.d/asterisk: No such file or directory |
| 13:56.41 | roxdragon | kaldemar, http://paste.ubuntu.com/582095/ |
| 13:56.56 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 13:59.06 | kaldemar | roxdragon: "make config" will install an init script for you. |
| 13:59.59 | ssureshot | kaldemar: here is the sip debug log http://inetpro.org/pastebin/10865 |
| 14:00.08 | ssureshot | I'm not quite sure whatI'm looking for |
| 14:00.41 | roxdragon | update-rc.d: using dependency based boot sequencing |
| 14:00.46 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:00.47 | roxdragon | kaldemar, |
| 14:02.17 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 14:03.59 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 14:05.23 | *** join/#asterisk karmic_dude (~karmic@63.214.236.169) |
| 14:06.45 | ssureshot | looks like it just keeps trying to register but I'm not seeing any reason why |
| 14:08.07 | kaldemar | ssureshot: did you upgrade from 1.6.X to 1.8? |
| 14:08.26 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 14:09.13 | *** join/#asterisk ajkaanbal (~ajkaanbal@190.146.84.212) |
| 14:09.47 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:09.47 | ssureshot | I'm upgrading from 1.2 > 1.8 :) |
| 14:09.59 | kaldemar | ssureshot: ah, even a larger jump. set pedantic=no under [general] in sip.conf. |
| 14:10.50 | roxdragon | kaldemar, ok it's works :D but...... |
| 14:11.54 | kaldemar | the default value for pedantic was changed from no to yes in 1.8. it enables tag checking in headers and makes asterisk ignore requests that it doesn't like. |
| 14:12.38 | kaldemar | roxdragon: but? |
| 14:12.40 | garymc | Anyone know if this is a fix for transfered calls not showing or even recording at all? http://www.freepbx.org/forum/freepbx/users/call-monitoring-stops-when-call-is-tranferred-internally#comment-26562 |
| 14:12.50 | garymc | whoops forget the freepbx link |
| 14:13.00 | garymc | but this is a asterisk issue |
| 14:13.34 | ssureshot | kaldemar: wow you are the man... any other changes I should look to make with the jump from versions? |
| 14:13.36 | garymc | im running asterisk version 1.6.2.10 |
| 14:13.36 | kaldemar | garymc: what is? |
| 14:13.47 | ssureshot | kaldemar: I thank you so much my friend |
| 14:14.06 | garymc | kaldemar : transfered calls not recording or listing in my ARI |
| 14:14.26 | kaldemar | ssureshot: read through UPGRADE*.txt in the source package. those files have the configuration changes that you need to address when upgrading. |
| 14:14.30 | kaldemar | ssureshot: you're welcome. |
| 14:15.13 | ssureshot | will do,, I've read through that a few times but I'll roll through it again for good measure |
| 14:15.43 | kaldemar | garymc: maybe that's an ARI issue? |
| 14:15.59 | garymc | ok, but they dont seem to be recording at all |
| 14:16.46 | garymc | in my asterisk monitor folder |
| 14:16.58 | kaldemar | garymc: what makes your configurations? |
| 14:18.33 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:18.48 | garymc | freepbx |
| 14:18.57 | garymc | kaldemar: freepbx |
| 14:22.17 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 14:23.04 | roxdragon | kaldemar, could you help me fix the dialplan? this is the last problem! |
| 14:24.18 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 14:24.35 | garymc | Kaldemar : iam told if there is no recording in my monitor folder of the transfered call , then it is an asterisk issue not a freepbx one :S |
| 14:24.52 | Cadey | Bit of a non-asterisk question here. Are there many network admins in there that use windows servers and also asterisk their PBX of choice |
| 14:25.05 | Cadey | *in here |
| 14:25.53 | jaytee | Cadey, I use asterisk but the majority of the servers I deal with are Windows servers. |
| 14:26.25 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:27.08 | Cadey | cool, would you be interested in windows baised asterisk tools such as client that let you pick extensions and have it display whats going on (channels on the extension, incomming or outgoing, what they are dialing and who is calling) |
| 14:27.12 | Cadey | things like that? |
| 14:27.48 | roxdragon | [Mar 18 15:27:26] WARNING[1865]: chan_sip.c:5226 sip_call: No audio format found to offer. Cancelling call to 401 |
| 14:28.41 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 14:29.03 | kaldemar | garymc: if you're told that there is no recording in your monitor folder, then there is a file missing. it can be because of various reasons. i'd look for the reason in the Monitor app the last. |
| 14:29.57 | kaldemar | Cadey: there are such tools already. |
| 14:30.07 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 14:30.20 | Cadey | kaldemar: yeah but some of them are a little ropy |
| 14:30.29 | Cadey | ropey |
| 14:30.47 | _Corey_ | ropey? That's a new one |
| 14:31.34 | Cadey | ropey, as in a bit lacking, dodgy etc |
| 14:32.09 | _Corey_ | I assumed it was a pejorative... :) |
| 14:32.36 | garymc | kaldemar: ok so what do I do? |
| 14:32.54 | garymc | we cant get call recordings for transfered calls and its a very serious matter now |
| 14:33.27 | _Corey_ | Cadey: There's always a market for good software... Depending on what you're looking for, there are some good web-based tools though |
| 14:34.30 | kaldemar | garymc: you haven't told much about the scenario. not even how you trigger call recording. see if it even happens when you transfer a call. you need to narrow the problem down to something. |
| 14:34.57 | garymc | ok so i need a cli output |
| 14:35.08 | roxdragon | kaldemar, ? |
| 14:35.14 | coppice | There's usually a good market for ropey software, when suitably marketed |
| 14:35.31 | Cadey | _Corey_ : I guess the reason im asking is because ive already started making an AMI proxy service for windows which also allows other more custom connections. So the AMI Proxy as well as jsut being a proxy to relay messages will also dish out diffrent messages |
| 14:35.39 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:35.39 | roxdragon | !codec |
| 14:35.51 | roxdragon | ~codec |
| 14:35.51 | infobot | A codec is a device or program capable of encoding and/or decoding a digital data stream or signal. The word codec may be a combination of any of the following: 'compressor-decompressor', 'coder-decoder', or 'compression/decompression algorithm'. |
| 14:36.04 | kaldemar | garymc: a CLI output is a good start. |
| 14:36.17 | garymc | ok |
| 14:36.28 | kaldemar | is a ropey software the kind that hangs? |
| 14:36.36 | _Corey_ | coppice: and shiny black consumer devices lacking buttons... ;) |
| 14:37.58 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 14:39.26 | coppice | also available in white |
| 14:40.25 | _Corey_ | haha |
| 14:43.05 | garymc | kaldemar : http://pastebin.com/P6MM90zu called into office from my mobile. answered call on ext201 and transfered to ext200 |
| 14:43.11 | garymc | no recording of transfered call |
| 14:43.24 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:44.56 | kaldemar | garymc: btw, did it work before? and if it did, what changed? |
| 14:47.51 | garymc | no it never worked |
| 14:49.58 | *** join/#asterisk oquidave (~oquidave@41.190.129.127) |
| 14:50.07 | oquidave | hello people |
| 14:50.23 | oquidave | am configuring a TDM card for asterisk |
| 14:51.03 | oquidave | am wondering what does the RED alarm mean when i type dahdi show status? |
| 14:52.13 | oquidave | all the lights on the digium card are showing RED, YET i have connected a T1 line to one of the ports |
| 14:52.55 | kaldemar | garymc: freepbx is your issue. |
| 14:52.55 | oquidave | i have checked my wiring and everything seems fine....what can i do? thanks |
| 14:54.22 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 14:55.14 | kaldemar | garymc: the MixMonitor app was recording Local/201@from-queue-5dac;2 and SIP/201-000008eb to /var/spool/asterisk/monitor/20110318-143907-1300459147.14066.WAV |
| 14:55.41 | garymc | yes |
| 14:55.55 | kaldemar | garymc: once SIP/201-000008eb dropped, you got a "== MixMonitor close filestream" |
| 14:57.05 | kaldemar | garymc: did you get the aforementioned file? |
| 14:57.26 | garymc | yes, but the transfered part isnt there |
| 14:58.03 | kaldemar | garymc: why would it if nothing tells asterisk to record it? |
| 14:58.23 | garymc | so you defo think its a freepbx issue? |
| 14:58.46 | kaldemar | garymc: more a freepbx lack of feature. |
| 15:00.43 | kaldemar | garymc: and for future encounters, don't harass people with 755 lines of CLI output to see a single call. it was quite humorous for a friday afternoon though. this time. |
| 15:01.04 | garymc | hmmm ok |
| 15:01.22 | garymc | I thought showing all of it would be more of a precise thing to help me :S |
| 15:01.45 | *** join/#asterisk munson (~munson@99.188.100.194) |
| 15:02.01 | munson | any cisco 7942g gurus that could possibly point me in right direction to get these phones enabled in my asterisk/freepbx/freeswitch whatever box. I d/l the SIP fw from cisco and phone is still looking for a P03-8-12-00 file which the fw didn't come with that file |
| 15:03.04 | garymc | kaldemar : freepbx lack of feature <- Sure. What is missing then? What is the option/command? Where should it be placed relative to the rest of thing to make this operable? |
| 15:03.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 15:07.40 | *** join/#asterisk sled-dog (~luser@adsl-074-165-241-009.sip.msy.bellsouth.net) |
| 15:07.46 | kaldemar | garymc: a Monitor/MixMonitor in the dialplan for example. |
| 15:08.33 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 15:08.36 | saliak | I'm trying to get asterisk up and running. I've done what I think it's a successful install, and am trying to connect a zoiper client to it. I setup a sip.conf extension, but when i try to connect with zoiper it sits there for a while and seems to fail. is there a log file that'll give me some insight into what's going on? |
| 15:08.38 | kaldemar | garymc: there were 4 different local channels generating the noise. next time show the relevant parts. |
| 15:09.08 | garymc | i dont know what you mean |
| 15:09.13 | Zhad | Did sipmodem ever get finished? |
| 15:09.47 | kaldemar | saliak: attach to asterisk CLI with "asterisk -vvvvr" and if that doesn't give you enough hints, enable sip debug with command "sip debug". |
| 15:10.57 | saliak | ah, there's the answer. asterisks seems to be segfaulting |
| 15:11.11 | saliak | so i guess that explains why it's not connecting |
| 15:12.56 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
| 15:13.09 | coppice | Zhad: is anything ever really finished? |
| 15:17.47 | Zhad | True |
| 15:18.10 | Zhad | Some things (at the moment I'm thinking of Sage) should never have been started |
| 15:19.30 | Zhad | I've just discovered through it's ODBC driver, if you execute a simple SQL query with a join in, the single line result can take over 3 minutes to get, during which time no-one else can use it. |
| 15:19.49 | Zhad | a similar query in postgresql would take less than 200ms |
| 15:20.12 | Zhad | stops irrelevant venting |
| 15:20.58 | coppice | don't worry. its a lot better than the venting in Fukushima |
| 15:24.45 | Zhad | It will actually be quicker to run a separate query to get the other column data than to include it in the original query, it's insane. |
| 15:24.54 | saliak | anyone had luck installing asterisk from the apt repository in ubuntu? When i try, it lists a bunch of packages that it can't get, upon which asterisk depends (http://pastebin.com/UzHjqWpc). Is there some other repository i need to add? |
| 15:25.14 | Zhad | Is your mirror fresh? |
| 15:25.49 | Zhad | stopped using his ISPs mirror for debian packages when he noticed some were missing/out dated |
| 15:26.12 | saliak | i think so. apt-get update does that, eh? |
| 15:26.38 | Zhad | that updates your local package descriptors with what's in the repository |
| 15:27.14 | Zhad | If you are using one of the servers listed on ubuntu's site, you should be okay. |
| 15:27.17 | saliak | sot he question is if the repositories i'm using are up to date? |
| 15:27.26 | saliak | yeah, i'm using the ones from the default source.list |
| 15:27.36 | Zhad | yup, odd problem, but it does happen. |
| 15:28.22 | saliak | anyone know of a repository that does have everything for asterisk? |
| 15:29.12 | Zhad | the one you've got will have. |
| 15:30.48 | Zhad | what does apt-get install -f report? |
| 15:31.11 | saliak | 0 upgraded, 0 newly installed, 0 to remove and 2 not upgraded. |
| 15:32.49 | Zhad | what do cat /etc/apt/sources.list and sb_release -a report ? |
| 15:33.40 | Zhad | should warn you, he is musch more experienced with debian than ubuntu. |
| 15:35.50 | saliak | http://pastebin.com/sJN662XY |
| 15:36.12 | saliak | zhad: http://pastebin.com/sJN662XY. sb_release isn't a command on my system, and i'm not sure where to find it |
| 15:37.13 | Zhad | lsb_release |
| 15:37.21 | Zhad | sorry, typo |
| 15:38.05 | saliak | http://pastebin.com/j44NWrCR |
| 15:40.08 | Zhad | You should check with someone who is more au fait with ubuntu, but it looks like you have repositories there for 2 different releases |
| 15:40.19 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 15:40.42 | Zhad | lines 20-23 look like they refer to an older release than the one you are using. |
| 15:41.37 | Zhad | (in sources.lst) |
| 15:41.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 15:41.55 | Zhad | list even |
| 15:43.46 | Zhad | I'd comment out those lines, run apt-get update and then try apt-get install asterisk. |
| 15:44.33 | Zhad | It looks like you have an Ubuntu 7 install that you upgraded to 10, but left some repositories that related to the v7 release in there. |
| 15:46.16 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:46.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 16:01.06 | *** join/#asterisk hairyraven (~nobody@95.73.138.47) |
| 16:03.54 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 16:05.12 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 16:09.46 | saliak | ah, i see |
| 16:12.03 | saliak | Zhad: bingo. that was it. i totally missed all the "gusty"'s in there. changing to lucid let it find what it needed |
| 16:15.38 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:19.07 | raden_work | Naikrovek, yo bro |
| 16:19.28 | Naikrovek | raden_work: heya |
| 16:19.36 | raden_work | hows it going |
| 16:19.42 | Naikrovek | same |
| 16:20.09 | raden_work | lol |
| 16:20.17 | raden_work | here too |
| 16:20.26 | raden_work | working on setting up a 5 GHZ wifi bridge today |
| 16:21.20 | raden_work | one thing I never thought of was router placement Internet-modem-bridge AP-Bridge station-router or Internet-modem-router-bridge AP-Bridge station |
| 16:21.26 | raden_work | decisions decisions LOL |
| 16:21.39 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:21.47 | Naikrovek | net -> modem -> router -> the rest |
| 16:21.57 | Naikrovek | ... i think |
| 16:24.41 | raden_work | would make most sense |
| 16:25.05 | raden_work | Cause to do anything with radios i wont be able to access them without unplugging router etc.... |
| 16:26.34 | raden_work | then again technically once there setup I sould never have to mess with them ( technically ) |
| 16:28.08 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 16:29.12 | benngard | [Mar 18 17:33:12] WARNING[3104]: res_fax_spandsp.c:367 spandsp_log: WARNING T.30 ECM carrier not found <- is that warning some "danger" or can i just ignore it? |
| 16:30.42 | FlaPer87 | when compiling an external module, how can I add extra libs that should be linked? |
| 16:30.59 | *** join/#asterisk neothedeveloper (~chatzilla@122.170.17.56) |
| 16:31.14 | neothedeveloper | hello |
| 16:31.38 | neothedeveloper | is it possible not to play DTMF tone to caller? |
| 16:32.06 | pabelanger | yes. What are you doing? |
| 16:32.30 | pabelanger | s/yes/depends |
| 16:32.53 | neothedeveloper | what I am looking for is, I have access DIAL number |
| 16:33.36 | neothedeveloper | caller will dial the number |
| 16:33.56 | neothedeveloper | then when system asks for DNID |
| 16:34.06 | neothedeveloper | when caller presses DTMFs |
| 16:34.28 | neothedeveloper | asterisk should not send DTMF tone to caller |
| 16:34.36 | neothedeveloper | instead it play silence |
| 16:34.48 | pabelanger | neothedeveloper: Do an attended transfer |
| 16:34.52 | neothedeveloper | pardon for my english :) |
| 16:35.03 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 16:35.13 | neothedeveloper | well I will need blind |
| 16:36.13 | neothedeveloper | pabelanger:in which case it's possible & how? |
| 16:37.56 | pabelanger | From what you described, I would setup an attended transfer in Asterisk, then disable MOH when Asterisk puts your user on hold. |
| 16:39.11 | neothedeveloper | well I am talking about to disable touch tones which are being played to caller |
| 16:40.04 | pabelanger | If the caller is generating the DTMF, then asterisk has nothing to do with it. It is the local phone. |
| 16:41.04 | neothedeveloper | pabelanger:thanks |
| 16:43.33 | *** part/#asterisk slim_ (~slim_@41.239.35.81) |
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| 17:07.47 | saliak | if i've installed asterisk correctly, and can get a sip device to connect (and be in the "demo" context), shouldn't extension 1234 playback some audio to me? |
| 17:11.46 | p3nguin | It will... ONLY if you have created extension 1234 to Playback() some audio file. |
| 17:16.40 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 17:20.21 | *** join/#asterisk [netman] (~netman@23.Red-83-40-2.dynamicIP.rima-tde.net) |
| 17:22.52 | saliak | Yeah, i think that's the case for the demo context. at least it looks like there's a "#" extension that plays back something (demo-thanks). unfortunately i don't hear anything. just trying to see if it's setup correctly. looks like my sip softphone connects ok, but whatever extension i dial it never connects. i'm guessing i have something basic wrong? http://pastebin.com/y4m8BvEd |
| 17:28.47 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 17:29.17 | p3nguin | saliak: You have several extensions available to call, but I don't know if you can call the s extension that starts the demo. |
| 17:29.38 | *** join/#asterisk timholum (~timh@68-117-120-138.static.eucl.wi.charter.com) |
| 17:29.45 | timholum | is there a way to make essencially a catch all for extentions in a dial plan? |
| 17:30.19 | p3nguin | Yes. But it isn't usually recommended unless you know what you are doing. |
| 17:31.11 | p3nguin | timholum: Use the pattern of _. for matching any one or more characters, including the standard extensions of s, h, i, and t. |
| 17:31.30 | timholum | p3nguin: Thanks |
| 17:31.47 | *** join/#asterisk [netman] (~netman@199.Red-83-41-0.dynamicIP.rima-tde.net) |
| 17:32.30 | p3nguin | Be prepared for it to cause a problem from time to time because it matches all standard extensions unless you explicitly define them elsewhere. |
| 17:33.33 | timholum | the system I am modifying only ever calls one extentions |
| 17:33.36 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:34.00 | timholum | I am modifiying the voice backend to bigbluebutton |
| 17:34.16 | saliak | p3nguin: is there an example of just something stupid simple that I can use to make sure that i've got the most basic of my setup working? i guess a dialpan that, whatever, you do, plays back the tt-monkeys or something like that? I assume that if my softphone can connect, my sip.conf is correct? i'm a little up in the air if i'm interpreting how the dialplan context works correctly |
| 17:35.38 | *** join/#asterisk cusco (~tralala@pcmedic.pt) |
| 17:35.40 | cusco | hello |
| 17:35.42 | p3nguin | saliak: Looking at your extensions, you should be able to call extension 2 and hear something useful. |
| 17:36.05 | cusco | Im having this constant problem with dahdi. It is failing to use channel 202 as D-Chan |
| 17:36.12 | cusco | about 2 times per hour |
| 17:36.21 | cusco | thise is causing current calls to hang up |
| 17:36.24 | cusco | :( |
| 17:36.58 | cusco | now this pri line is on a 4 span digium card, where 2 other spans belong to another telco and the 4th one is empty |
| 17:37.08 | cusco | I believe I'm having timer issues |
| 17:37.23 | cusco | log at: http://paste.debian.net/111144/ |
| 17:37.37 | cusco | could some one point me in the right direction? |
| 17:37.51 | cusco | is there a problem having two telcos in one pri card? |
| 17:38.56 | p3nguin | saliak: The context that you assign to your device determines an "entry point" for the call to start in the dial plan. |
| 17:40.04 | _Corey_ | cusco: I've heard about people having issues where carrier timing will be problematic, though I've not had any issues using more than one carrier at the few sites I've tried it. (With Digium cards) |
| 17:40.25 | p3nguin | saliak: Your context is demo, so any extension that you call will be looked for in the [demo] context. If there is no match, you'll either be disconnected or you'll get congestion tones. |
| 17:40.56 | _Corey_ | cusco: Is this a new turnup or something that had been working? |
| 17:43.38 | saliak | p3nguin: that was my understanding, but my testing doesn't seem to reflect that. with the demo context, i can dial any extension and it just says "dialing" and sits there. i have the asterisk console started, but it just says "Using SIP RTP CoS mark5" every time i make a call, and nothing else (i'm assuming because the call doesn't connect) |
| 17:44.37 | p3nguin | saliak: Run "core set verbose 4" and make another call. |
| 17:44.39 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 17:45.13 | *** join/#asterisk [netman] (~netman@142.Red-80-39-55.staticIP.rima-tde.net) |
| 17:45.40 | p3nguin | What does this mean when starting Asterisk 1.8.2.3? Unable to access the running directory (Permission denied). Changing to '/' for compatibility. |
| 17:46.08 | p3nguin | Does that mean the dir where I ran it from? |
| 17:46.08 | saliak | p3nguin : i had it set at 12 (started as asterisk -vvvvvvvvvvvvr), but even at 4, it doesn't say anything.. eventually the call seems to time out and it gives up (but i think that's my client) |
| 17:46.56 | p3nguin | saliak: Sounds broken to me. If a call was starting, it should show it on the CLI. Maybe a sip debug would help. sip set debug on |
| 17:48.51 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 17:53.57 | cusco | _Corey_: hi sorry for the delay. This is old... However we weren't using (much) this telco |
| 17:54.03 | cusco | now that we do, we notice these problems |
| 17:55.14 | munson | anyone here familiar with SIP and the 7942g cisco ip phone? phone is asking for a P0S3-8-12-00 from tftp server but the fw i got from cisco doesn't come with that file ;( |
| 17:55.36 | cusco | what can I do regarding timers? |
| 17:56.09 | p3nguin | munson: What version of SIP firmware do you have? |
| 17:56.23 | munson | the latest but lemme check what one was it |
| 17:56.28 | ssureshot | there is a long pause when I hit the VM button on my phone while it waits for you to press your extension before it auto senses the extension your at.. can I shorten this? |
| 17:57.37 | _Corey_ | cusco: Is the (presumed) timing issue with the new carrier affecting calls on the other one? |
| 17:58.12 | munson | p3nguin, cmterm-7942_7962-sip.9-1-1SR1.zip |
| 17:58.23 | saliak | p3nguin: interesting. when i setup an iax connection it works! of course, my client seems to crash every 20 seconds, but that's probably a different issue? do you have to setup a codec for sip connections somewhere? when i make a iax call it shows GSM as the codec, but for SIP it's unkonwn. is that something that's set in a conf file? |
| 17:58.30 | p3nguin | munson: You've extracted the firmware files to the tftpd root? |
| 17:58.36 | munson | yes |
| 17:58.47 | munson | got all the files there on my freeswitch/bluebox server |
| 17:59.07 | munson | but phone is asking for that P0S3-8-12-00 file |
| 17:59.28 | p3nguin | saliak: You should always define the codec, for both SIP and IAX2. disallow=all and then allow the one you want to use, e.g., allow=ulaw |
| 17:59.41 | citywok | ~itsplist-us |
| 17:59.42 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:00.09 | p3nguin | munson: Do you have your SIPDefault.cnf and SIP<MAC>.cnf files handy? |
| 18:00.36 | saliak | p3nguin : that was it. thanks! |
| 18:00.51 | p3nguin | munson: Those files need to have the image_version set appropriately. |
| 18:01.22 | munson | ya they are on the server, created from the endpointmanager-1.1.0 from bluebox/freeswitch....but even then the endpointmanager only has the 7940/7960 listed |
| 18:01.42 | p3nguin | The format is probably the same. |
| 18:01.56 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 18:02.10 | p3nguin | I'd generate it for the 7940 and then manually edit for my own needs. |
| 18:02.14 | munson | ya that is what i read in a forum but like i said the phone is asking for that one file which the cisco fw didn't comne with |
| 18:02.20 | munson | kk |
| 18:02.39 | p3nguin | The image_version setting tells the phone which firmware to load. |
| 18:03.21 | p3nguin | It would have said image_version:P0S3-08-12-00 |
| 18:03.41 | p3nguin | And it has to be changed to match your files' version. |
| 18:04.55 | munson | ya lookin at the SIPDefaults.cnf it still is pointing to that P0S3-8-12-00 on first line on image_version: |
| 18:05.09 | p3nguin | Yep. Change that. |
| 18:05.20 | munson | change it? |
| 18:06.00 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:06.00 | munson | i have .sbn files and .loads as well as the OS79XX.TXT and ringlist files and SEP/SIP.cnf and .xml files |
| 18:06.02 | munson | but no bin |
| 18:06.10 | p3nguin | You'll want to look at your firmware files to get the name so you can put the correct version there. Maybe image_version:P0S3-09-1-00, for example. |
| 18:06.25 | p3nguin | You'll have to adjust the OS79XX.TXT file, too. |
| 18:06.36 | cusco | _Corey_: could be not sure... |
| 18:06.50 | p3nguin | Don't expect image_version:P0S3-09-1-00 to be right. I'm just using that to show you how you have to change it. |
| 18:07.09 | p3nguin | Look on your tftpd where you extracted that firmware zip file. |
| 18:07.24 | munson | ya but no P0S3 nada in the cisco fw files |
| 18:07.39 | cusco | /sys/devices/system/clocksource/clocksource0/available_clocksource has tsc hpet acpi_pm jiffies, current has tsc |
| 18:07.54 | p3nguin | You should have the firmware files ending in .bin, .sbn, .sb2, and .loads |
| 18:08.32 | p3nguin | I have, for example, P003-08-11-00.{bin,sbn,sb2,loads} |
| 18:08.53 | p3nguin | Wait, that's slightly incorrect. |
| 18:09.00 | munson | no P0S* files in fw |
| 18:09.25 | p3nguin | P003-08-11-00.{bin,sbn} and P0S3-08-11-00.{sb2,loads} |
| 18:09.46 | p3nguin | Okay, then what files *are* in your firmware archive file? |
| 18:10.22 | munson | yup no files i have apps42*.sbn, snu42*.sbn,cvm42*.sbn,dsp42*.sbn,dsp42*.sbn,jar42*.sbn and loads are sip42,term42,term62.loads |
| 18:10.41 | p3nguin | saliak: The SIP config fixed up after setting the codec in the peer definition? |
| 18:11.46 | cusco | _Corey_: also seems that I'm only using res_timing_dahdi.so --> http://paste.debian.net/111148/ |
| 18:11.53 | p3nguin | Alright, so you do have firmware files. Now you just have to google for what needs to go in your OS79XX.TXT and SIP*.cnf files. |
| 18:12.18 | saliak | p3nguin: yeah. the issue now seems to be that the calls are dropped (between a laptop on my local network and my local asterisk server) after 30-50 seconds. still trying to get more data on that. again, it's not clear if it's the server, or maybe the softphone i'm using |
| 18:12.57 | p3nguin | saliak: The verbose output could be useful. Pastebin the verbose output of a call. |
| 18:14.53 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 18:15.28 | *** join/#asterisk [netman] (~netman@64.Red-83-41-7.dynamicIP.rima-tde.net) |
| 18:18.02 | saliak | p3nguin : seems to be intermittent. for some reason can't get it to happen now. it doesn't spit out any errors in particular, the line just goes silent (but doesn't disconnect). i think i need a control softphone |
| 18:20.27 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 18:21.24 | saliak | here's the output from two calls, one successful and one that goes silent. nothing prints out till i hang up (at which point, it looks like a successful call) - http://pastebin.com/TmaJAHFf |
| 18:21.29 | munson | should i just call cisco and demand a damn fw file that works lmao |
| 18:22.31 | munson | cuz in the SIP fw file cmterm-7942_7962-sip.9-1-1SR1.zip doesn't have any P0S*.bin or .sbn or nada |
| 18:22.34 | p3nguin | munson: Let me see what I have available. You have a 7942G? |
| 18:22.38 | munson | yes |
| 18:23.45 | munson | been on cisco for awhile now and they say the 7942g isn't supported for "asterisk" or any other pbx system but i didn't ask for support i'm asking for some P0S* file that the phone is asking for lol |
| 18:24.13 | p3nguin | Just tell them you're using a 3rd party call control device. Don't ask for support for Asterisk. |
| 18:25.08 | p3nguin | I see what you mean about the weird file names. I'm looking at SIP 8.5.3 for that phone. |
| 18:27.24 | cusco | question: If I make changes to /etd/dahdi/system.conf, I need to unload and reload res_dahdi or can I just reload it?? |
| 18:27.53 | munson | hmm i can try the 8version but didn't know why its not in the 9...hell i'll d/l each and every one ;) |
| 18:28.47 | p3nguin | I don't have much info about a 7942/7962. |
| 18:29.13 | munson | kk |
| 18:29.48 | munson | i'm on cisco site now checkin diff fw's and maybe they have one that has a P0S* file |
| 18:30.36 | p3nguin | You won't find that. |
| 18:31.34 | munson | hmm |
| 18:31.38 | p3nguin | You're going to have to get google to tell you what to define in SIPDefault.cnf. |
| 18:31.40 | *** join/#asterisk [netman] (~netman@122.Red-83-41-11.dynamicIP.rima-tde.net) |
| 18:33.34 | munson | ya and thats find a P0S3-8-12-00 file that these files are wanting to point to |
| 18:33.57 | munson | at least the SIPDefault.cnf file and others are wanting |
| 18:34.47 | p3nguin | Google doesn't realize that a 794*2* is different from a 794*0* ? |
| 18:35.46 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 18:37.32 | munson | thats what i'm trying to figure out |
| 18:38.08 | munson | done searches and i thought 7940 was for the whole series but is just for that phone |
| 18:40.34 | Nugget | 79x0 is completely unlike the other cisco phones |
| 18:40.34 | Nugget | different firmware, different config format, different behavior |
| 18:41.08 | munson | well i mean last resort i guess i can always use their call manager setup but don't want that |
| 18:44.14 | p3nguin | 7940 means 7940 and 7960 only. 7941 means 7941 and 7961 only. Et cetera. |
| 18:44.25 | munson | yup ;( |
| 18:44.35 | p3nguin | 7900 is the only one that encompasses the entire series. |
| 18:46.45 | munson | maybe bought the wrong phones lol ;( |
| 18:47.15 | munson | teacher got em with the new cisco2821 and FX0 moduels and 2 4port PoE switches and some other ports i never worked with |
| 18:48.44 | munson | i was truly hoping it was on the endpointmanager's on bluebox or freepbx |
| 18:59.21 | *** join/#asterisk nighty^ (~nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
| 19:01.31 | p3nguin | I personally use 7940/7960 and a lonely 7912 with Asterisk with no problems. You could go that route or the new 500 series. |
| 19:01.53 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:02.03 | chandoo | hi |
| 19:02.26 | chandoo | i have little echo in ekiga, i am thinking it is feedback from the mic |
| 19:02.29 | chandoo | how to fix this |
| 19:02.54 | p3nguin | Turn down the volume! |
| 19:03.46 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 19:08.56 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:11.30 | chandoo | p3nguin:} any speical audio codecs i have to enable, right now i have Speex, PCMU, PCMA enabled under audio-codecs |
| 19:12.15 | p3nguin | Enable the one you want to use, the one that is configured in asterisk for your peer (phone). |
| 19:13.30 | chandoo | p3nguin:} no idea, first time using sip |
| 19:13.43 | chandoo | i don't what i want |
| 19:13.50 | chandoo | i don't know what i want |
| 19:14.29 | p3nguin | In asterisk sip.conf, you had to create a peer definition for your phone. You included in it, disallow=all and allow=ulaw or allow=something-else. Whichever codec you decided to allow, that is the one to configure the softphone to use. |
| 19:14.56 | *** join/#asterisk nathan7 (nathan@unaffiliated/nathan7) |
| 19:14.58 | chandoo | i think it fixed, i enabled echo cancellation in the preferences |
| 19:15.23 | chandoo | p3nguin:} all i have is allvoi provider account trying to use ekiga in fedora |
| 19:15.39 | p3nguin | Are you trying to tell me that you aren't using Asterisk? |
| 19:15.57 | chandoo | p3nguin:} i am sorry , no i am using asterisk |
| 19:16.06 | chandoo | p3nguin:} i am sorry , no i am not using asterisk |
| 19:16.16 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 19:16.18 | p3nguin | Then you're in the wrong channel. This is an Asterisk channel. |
| 19:16.31 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 19:16.34 | chandoo | is there channel for ekiga |
| 19:16.41 | p3nguin | no clue |
| 19:16.54 | chandoo | p3nguin:} do you recommend any sip clients for linux |
| 19:17.09 | p3nguin | a phone |
| 19:17.19 | p3nguin | I mean, a soft phone? |
| 19:17.20 | chandoo | for linux desktop |
| 19:17.25 | chandoo | yes softphone |
| 19:17.26 | p3nguin | Twinkle |
| 19:17.44 | p3nguin | It's the one I use. |
| 19:17.54 | p3nguin | It's the phone I use when I don't have a phone. |
| 19:18.49 | chandoo | p3nguin:} thanks :) Tinkle looks good, let me try it out |
| 19:18.57 | p3nguin | tinkle, lol |
| 19:19.25 | joobie | hey p3nguin |
| 19:19.33 | joobie | how did you go with that sms stuff? |
| 19:19.35 | p3nguin | How's it goin'? |
| 19:19.54 | joobie | aite.. just got up |
| 19:20.05 | joobie | u? |
| 19:20.08 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 19:20.09 | chandoo | p3nguin:} sorry typo twinklee |
| 19:20.38 | saliak | i'm trying to setup broadvoice up as my trunk for asterisk. i've got asterisk running on the same computer i use as a firewall (using shorewall). do i need to open up 5060 to the server, or does asterisk initiate the connection to the broadvoice servers? i've followed their install instructions at http://www.broadvoice.com/support_install_asterisk.html |
| 19:21.01 | p3nguin | I'm trying to build up an embedded system for Asterisk. Having troubles understanding some file system operations. |
| 19:22.26 | joobie | an embedded system for asterisk in what sense? |
| 19:22.34 | joobie | what file system operations are you stuck with? |
| 19:22.49 | p3nguin | saliak: Allow UDP port 5060 and the UDP port range defined in rtp.conf. |
| 19:23.57 | chandoo | guyz once you are done with your conversation, can i install asterisk and use it in any way, all i have is SIP account with All Voi service provider, |
| 19:24.07 | chandoo | i am not sure if my question make any sense |
| 19:24.38 | *** join/#asterisk cerberus_za (~coert@196-210-142-16.dynamic.isadsl.co.za) |
| 19:24.43 | chandoo | notices twinkle is for kde, |
| 19:24.46 | p3nguin | joobie: I've a flash memory module for the hard drive, I put ext2 on it, and I want to mount it with noatime to reduce writeback to preserve the flash memory. I can add noatime in the fstab, so it says defaults,noatime. I wanted to know if there was a way to make noatime a file system default, so I didn't need to include it in the fstab... |
| 19:25.19 | *** join/#asterisk phyburn (~phyburn@wsip-70-165-35-234.oc.oc.cox.net) |
| 19:25.49 | p3nguin | joobie: But there's apparently no way to do that, because noatime isn't one of the supported options that tune2fs can add as a default filesystem option. So I tried the extended option. That didn't appear to work, either. |
| 19:26.09 | p3nguin | I'm probably just not understanding something. |
| 19:27.00 | joobie | p3nguin, why cant you not use fstab for this? |
| 19:27.15 | p3nguin | If I specify noatime as an option in the fs superblock and don't specify it in the mount options, should it still show up in mtab/mounts? |
| 19:27.28 | p3nguin | I can. The question wasn't if I can or can't use the fstab. |
| 19:27.45 | p3nguin | The question was if I can make noatime a default fs option or not. |
| 19:28.22 | joobie | hmm |
| 19:28.22 | p3nguin | The initial answer was no, but then I was shown the extended options. |
| 19:28.38 | joobie | possibly with the -A flag with chattr |
| 19:28.47 | joobie | like err.. chattr -R -A / |
| 19:28.50 | joobie | but err |
| 19:28.53 | joobie | i would just use fstab |
| 19:29.07 | joobie | when your system boots, it will read fstab and mount the FS without atime |
| 19:29.09 | p3nguin | Uh, we're not dealing with files and directories. |
| 19:29.28 | p3nguin | I'm talking about a file system. chattr does not work on the fs level. |
| 19:29.38 | joobie | yes it does |
| 19:30.14 | p3nguin | You think chattr [options] /dev/sda1 is going to work? |
| 19:30.21 | joobie | no |
| 19:30.21 | p3nguin | I don't. |
| 19:30.28 | tzanger | /dev/sda1 is not an fs |
| 19:30.32 | joobie | p3nguin, /dev/sda1 is your block device |
| 19:30.46 | p3nguin | No, but it's the volume containing the fs. |
| 19:30.56 | p3nguin | I'm not new to Linux. |
| 19:30.56 | joobie | if you are using ext2 on sda1, mount it, then run chattr -A -R on the fs |
| 19:31.07 | tzanger | p3nguin: for someone who's not new to linux, you are sure talking like a n00b. |
| 19:31.19 | tzanger | p3nguin: /dev/sda1 is a block device. it's not anything else. |
| 19:31.20 | p3nguin | Once you mount it, you're dealing with the files, not on the fs level. |
| 19:31.33 | chandoo | i want to make some test calls to some one , how to do that, any one available for testing |
| 19:31.39 | tzanger | when you mount it, you will deal with the filesystem it contains by accessing them from the mount point, not the device node. |
| 19:31.47 | joobie | p3nguin, bascially just use fstab for this |
| 19:31.58 | joobie | p3nguin, mtab reflects your active mounts |
| 19:32.12 | joobie | when you mount the fs, fstab will be used |
| 19:32.43 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 19:33.00 | p3nguin | grep sda1 /proc/mounts |
| 19:33.00 | p3nguin | /dev/sda1 / ext2 rw,noatime,errors=continue 0 0 |
| 19:33.29 | p3nguin | This is fine. It is a result of using noatime in the fstab. |
| 19:33.37 | p3nguin | THAT WAS NEVER IN QUESTION. |
| 19:33.37 | joobie | nod |
| 19:34.11 | joobie | p3nguin, <p3nguin> I'm probably just not understanding something. |
| 19:34.15 | joobie | you are right. |
| 19:34.23 | p3nguin | The question was: If I set a default option of noatime ON THE FUCKING FILE SYSTEM, will it be used if I do not use noatime in the fstab or the mount command? |
| 19:34.36 | joobie | ahh |
| 19:34.39 | joobie | that was not the original Q :P |
| 19:34.46 | p3nguin | It doesn't show up in "mount" if I do it that way. |
| 19:34.59 | p3nguin | But I don't know if it is silent by doing it that way. |
| 19:35.27 | p3nguin | Because I do not understand file systems in that way. |
| 19:35.29 | joobie | how are you setting the default option of noatime without fstab ? |
| 19:35.38 | p3nguin | tune2fs |
| 19:35.52 | joobie | hmm |
| 19:36.06 | joobie | im not sure if it will be persistant |
| 19:36.08 | p3nguin | Default mount options: (none) |
| 19:36.11 | p3nguin | Mount options: noatime |
| 19:36.14 | joobie | you can do this via tune2fs whilst the FS is mounted ya? |
| 19:36.17 | p3nguin | Oh, it's persistent. |
| 19:36.34 | p3nguin | Sure, I can change the info if the fs is mounted or not. |
| 19:37.15 | joobie | ok |
| 19:37.25 | joobie | i'm not sure. |
| 19:37.48 | joobie | if you use tune2fs, not all the options will be shown in /proc/mounts |
| 19:37.49 | p3nguin | If I set it like that and take it off the fstab (or mount command), when I mount -a / it no longer shows noatime in "mount". I just don't know if that means noatime was not used, or if noatime is used but on a different level. |
| 19:37.51 | joobie | but err |
| 19:37.54 | joobie | you can test this |
| 19:38.05 | joobie | just mount the fs without atime and with tune2fs |
| 19:38.13 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 19:38.14 | joobie | cat a test file |
| 19:38.21 | joobie | then ls -lu the test file |
| 19:38.27 | joobie | see if the atime has been updated |
| 19:38.43 | p3nguin | I'm pretty sure I'm going to end up using noatime in the fstab, but I'd like to have a definitive answer to this. |
| 19:38.53 | joobie | well test it |
| 19:38.56 | joobie | it's a pretty simple test case |
| 19:38.59 | p3nguin | That's a good idea. I'll test it now. |
| 19:39.09 | p3nguin | Yep, good idea. |
| 19:39.25 | p3nguin | tune2fs -E mount_opts=noatime /dev/sda1 |
| 19:39.53 | munson | kk logging off i'll test some more stuff out monday |
| 19:40.05 | WIMPy | Interesting topic. |
| 19:40.10 | p3nguin | changed the fstab |
| 19:40.11 | munson | thx p3nguin for help |
| 19:40.16 | p3nguin | mount -o remount / |
| 19:40.19 | WIMPy | But I think it works. It does for error behaviour. |
| 19:40.38 | p3nguin | grep sda1 /proc/mounts |
| 19:40.39 | p3nguin | /dev/sda1 / ext2 rw,relatime,errors=continue 0 0 |
| 19:41.09 | p3nguin | There's probably no reason to test since we know the atime will get updated. |
| 19:41.33 | p3nguin | Am I thinking incorrectly about that? |
| 19:42.59 | joobie | p3nguin, personally i would use fstab for this |
| 19:43.01 | p3nguin | Well, I catted a file and ls -lu does not show a current time. |
| 19:43.20 | joobie | p3nguin, if you want to do it differently, just test it |
| 19:43.22 | p3nguin | I'm sure I'll end up using the fstab, but I still want a definitive answer about this topic. |
| 19:43.51 | p3nguin | mounts shows relatime, and the access time for the file I accessed did not update. |
| 19:43.55 | joobie | .. /proc/mounts will not show all options you configure within tune2fs |
| 19:43.56 | joobie | it can't |
| 19:44.04 | joobie | so it may / may not work |
| 19:44.15 | joobie | ok |
| 19:44.16 | p3nguin | exactly! |
| 19:44.17 | joobie | then its' fine |
| 19:45.14 | WIMPy | And don't forget nodiratime. |
| 19:45.41 | p3nguin | If I'm using noatime, nodiratime shouldn't do anything. At least that was my understanding. |
| 19:46.02 | p3nguin | If I'm wrong about that, show me so I'll know. |
| 19:46.07 | WIMPy | Mine is different. |
| 19:46.31 | WIMPy | I thinkt diratime is on even if you use noatime to keep some mail stuff working. |
| 19:47.10 | WIMPy | Is there even a diratime without no? |
| 19:47.18 | joobie | noatime implies nodoratime |
| 19:47.29 | joobie | .. at least in modern day kernels (2.6) |
| 19:47.30 | p3nguin | That was my thought. |
| 19:48.41 | WIMPy | The man says diratime is the default. What sense does that make then? |
| 19:48.42 | p3nguin | Maybe I said that wrong. Instead of saying nodiratime shouldn't do anything, I should have said nodiratime isn't needed because it doesn't change anything additional. |
| 19:49.30 | p3nguin | I think noatime trumps diratime. |
| 19:49.53 | p3nguin | noatime means don't update atime at all on the fs. That would include diratimes. |
| 19:50.27 | p3nguin | I've misunderstood things before, so I could be mistaken on that. |
| 19:50.48 | joobie | WIMPy, it is the default if you don't specify "noatime" |
| 19:51.00 | WIMPy | So essentially on a modern kernel, noatime actuelly has the same effect as nodiratime. |
| 19:51.13 | p3nguin | it includes it, at least. |
| 19:51.20 | p3nguin | It does more than just nodiratime. |
| 19:51.43 | p3nguin | nodiratime would affect only directories, where regular files would still get updated. |
| 19:51.45 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:52.10 | WIMPy | The current default seems to be noatime,diratime. |
| 19:52.15 | p3nguin | noatime would prevent updating for all types of files. |
| 19:53.35 | p3nguin | I seem to be in the presence of some people who know a bit about hardware... |
| 19:53.59 | p3nguin | This flash memory I have has a write endurance specification of: |
| 19:54.10 | p3nguin | 8years@100gbytes write and erase per day at 32GB |
| 19:54.14 | *** join/#asterisk tracep (~tracep@107.7.25.234) |
| 19:54.16 | *** join/#asterisk blatz (~no@coal.obleton.com) |
| 19:54.21 | p3nguin | What does that mean in REAL WORLD terms? |
| 19:54.43 | blatz | i am looking for help with realtime odbc. |
| 19:54.46 | p3nguin | Does that mean they tested a 32GB device? |
| 19:55.14 | tracep | need help with setting outbound CID on Asterisk 1.4 via config files, anyone willing to help? thx! |
| 19:55.15 | p3nguin | Does that mean my 4GB device won't last for 8 years if I write and erease 100 GB per day? |
| 19:55.31 | p3nguin | tracep: callerid=Your Name <123> |
| 19:55.31 | WIMPy | NFI, but the usual numbers are 10K writes on consumer grade, 100K writes on industrial grade flash. |
| 19:55.52 | joobie | http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=blob;f=fs/inode.c;h=9910c039f026254a9caea134a5a7ac8d40c9c76f;hb=HEAD |
| 19:56.12 | WIMPy | Flash is best used RO. |
| 19:56.16 | blatz | i am getting... "Realtime mapping for 'meetme' found to engine 'odbc', but the engine is not available" |
| 19:56.19 | joobie | have a look at the function at line 1503 |
| 19:56.20 | tracep | p3nguin: thanks, but where? user CID is overriding and going outbound instead of main # |
| 19:56.44 | blatz | but odbc show displays the right settings |
| 19:56.50 | p3nguin | tracep: Oh, I see your problem. I'll help you... just a moment. |
| 19:56.52 | blatz | and says connected |
| 19:57.03 | joobie | afaik this is why noatime implies nodiratime |
| 19:57.06 | tracep | T1 provider won't place calls without BTN set as outbound CID |
| 19:57.09 | coppice | its rare to see a sector failure with a flash disk. the whole bloody thing dies instead "-\ |
| 19:58.00 | p3nguin | tracep: Add a variable in your phone's peer definition: setvar=externalCID=3149691077 |
| 19:58.12 | p3nguin | tracep: And then use this in dial plan: ExecIf($[${IF($["${externalCID}" != ""]?1)}],Set,CALLERID(num)=${externalCID}) |
| 19:58.28 | p3nguin | tracep: Put that before the Dial() on your outbound extension. |
| 19:59.15 | p3nguin | tracep: If the externalCID variable exists, set the CALLERID(num) to that value before Dial()ing. |
| 20:00.16 | p3nguin | tracep: That's how I handle the external CID being different from the callerid setting for the device. |
| 20:02.17 | tracep | p3nguin: this is a fairly simple setup (v1.4) with minor dialplan, but attempts at setting the callerid before dial and in the zapata.conf is still being replaced by the user calling calling out |
| 20:02.32 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 20:02.56 | blatz | need help with odbc realtime |
| 20:03.12 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 20:03.15 | p3nguin | tracep: Is the phone you're dialing outbound with an analog phone? |
| 20:03.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:03.47 | p3nguin | tracep: And does zap support "setvar=variableName=value"? |
| 20:03.57 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 20:04.00 | tracep | no. SIP based (linksys SPA942) to * and to new T1 |
| 20:04.02 | p3nguin | tracep: If so, what I gave you will work. |
| 20:04.28 | *** part/#asterisk blatz (~no@coal.obleton.com) |
| 20:04.32 | p3nguin | tracep: Okay, put the setvar line that I gave you into the phone's definition in sip.conf. Run sip reload after saving sip.conf. |
| 20:04.41 | *** join/#asterisk clu3 (~steve@186.1.193.254) |
| 20:05.09 | p3nguin | tracep: Then change extensions.conf to reflect the command I gave you. Put it right before the Dial() command for outbound calls. |
| 20:05.16 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 20:06.04 | p3nguin | tracep: If you still can't get it done, paste your phone's peer definition into the pastebin and I'll change it for you. Similarly, paste your outbound calling context into the pastebin and I'll also change that for you. |
| 20:06.04 | tracep | the person who set this system up years ago is using the extensions.ael to do config |
| 20:06.04 | p3nguin | ~pb |
| 20:06.04 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:06.41 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 20:06.50 | ClintGoudie-Nice | top of the day to you all |
| 20:06.53 | p3nguin | Does ael support the same applications? My bet is "yes!" |
| 20:06.56 | tracep | sip.conf change is done. is the command the same even if using extensions.ael instead of extensions.conf? |
| 20:07.12 | p3nguin | No clue. Show me a snippet of the ael you have. |
| 20:07.53 | p3nguin | The application and its syntax can't change... as far as I can tell. |
| 20:08.19 | p3nguin | It's going to be the syntax of the extensions that will be different. |
| 20:09.35 | tracep | http://pastebin.com/hTMkwue8 |
| 20:09.50 | ClintGoudie-Nice | Is it possible to configure an extension to automatically forward to a different extension after a certain number of rings or if the extension is busy? |
| 20:10.02 | ClintGoudie-Nice | I'd rather do it on an extension basis, but I can do it system wide if I need |
| 20:11.53 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:12.04 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 20:12.19 | sequencer | hi all, i need some help with a dropped call issue |
| 20:12.24 | sequencer | who should i be asking ? |
| 20:12.42 | tracep | p3nguin: any thoughts on the snippet? |
| 20:12.58 | p3nguin | tracep: http://pastebin.com/h8XKLFs1 |
| 20:13.58 | p3nguin | tracep: As long as you have set the variable like I said, it should work just fine. |
| 20:14.15 | tracep | p3nguin: thanks! giving it a shot and let you know |
| 20:14.46 | p3nguin | tracep: By the way, I changed your pattern matches to a more sensible pattern for NANP. |
| 20:15.02 | tracep | p3nguin: yep, saw that. thx |
| 20:19.14 | tracep | p3nguin: hmmm...ok CID is right now, but provider is saying dialed # is not show |
| 20:19.48 | p3nguin | tracep: Ugh. I can't do everything! |
| 20:20.18 | p3nguin | That might be a zap setting. I don't use zap, so I don't know. |
| 20:20.45 | tracep | p3nguin: k, thx for your help on OCID though! |
| 20:22.14 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 20:24.23 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 20:26.16 | *** join/#asterisk dr__ (~dr@79.113.188.74) |
| 20:37.06 | titter | Is there a way to see all connections to the CLI? |
| 20:37.51 | Chainsaw | titter: They normally show up as a "UNIX connection", and you get reports of any connect/disconnect as long as you are on. |
| 20:38.30 | titter | Yup, just wondering if there is something actually watching these connections already there |
| 20:40.24 | ClintGoudie-Nice | an the follow me module hooks me up. |
| 20:40.25 | *** part/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 20:41.51 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:48.27 | p3nguin | wimpy: Did you have an idea what my write endurance specs could mean to me in a real world application? |
| 20:48.47 | WIMPy | NFI, but the usual numbers are 10K writes on consumer grade, 100K writes on industrial grade flash. |
| 20:49.09 | p3nguin | This is labeled as industrial. |
| 20:49.15 | p3nguin | sticker on it |
| 20:49.59 | p3nguin | I did not put swap on the flash memory, even though the installer really wanted me to... so is 100K writes a lot, relatively speaking? |
| 20:50.42 | WIMPy | Depends on what you're doing. |
| 20:51.04 | p3nguin | I'm building an Asterisk PBX. |
| 20:51.14 | WIMPy | Flash is best used RO. |
| 20:51.20 | p3nguin | I can't do that. |
| 20:51.41 | p3nguin | I have logs that need to be written and config files that often need changing. |
| 20:51.54 | WIMPy | Logs are evil. |
| 20:52.30 | p3nguin | I could try RO and see how it goes. I can write cdr and recordings to another computer on the network. |
| 20:53.03 | p3nguin | I'd like to keep it all self-contained, but maybe it's not going to be practical. |
| 20:53.04 | WIMPy | That might be a good idea. |
| 20:54.36 | p3nguin | The project is coming along nicely. I'm pleased with the progress I've made in only a couple of hours. |
| 20:55.08 | sequencer | i got the pastebin for this trouble |
| 20:55.11 | sequencer | http://pastebin.com/0j9DFyf8 |
| 20:55.16 | sequencer | any ideas ? |
| 20:56.38 | sequencer | also am not sure what id this : 169. CSeq: 102 CANCEL |
| 20:59.27 | sequencer | any help would be much appreciated :) |
| 21:03.47 | *** join/#asterisk boch (bed20889@gateway/web/freenode/ip.190.210.8.137) |
| 21:04.37 | boch | hi all |
| 21:05.16 | boch | could you give me a hand? im not being able to register my asterisk using jabber to openfire server... it does not outputs any debug of error.. it just keeps disconnected when list connected users |
| 21:06.22 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 21:08.00 | wdoekes2 | sequencer: at 115. the call is cancelled by the polycom |
| 21:08.31 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 21:08.58 | sequencer | wdoekes2 the user didnt cancel the call, so what can be the issue? |
| 21:09.06 | sequencer | malunction from the polycom itself ? |
| 21:09.13 | sequencer | malfunction* |
| 21:09.24 | titter | So if I have terrible routing to my Asterisk server from only a few locations, but can get much better routing from an area close to those locations, nearly half the latency and no packet loss to Asterisk ... how would you do it? I am thinking SER and rtpproxy to just route the sip and rtp to Asterisk? Can Asterisk simply route SIP/RTP to another Asterisk server? |
| 21:10.13 | wdoekes2 | hard to say.. but you're missing the start of the dialog in your pb |
| 21:10.49 | wdoekes2 | but it looks like the polycom doesn't like the (content of the) 183 message |
| 21:11.03 | wdoekes2 | (the one at 86.) |
| 21:11.12 | sequencer | let me see, ill try to get it, ill have to dig into 15k of lines now.. |
| 21:11.35 | sequencer | hmm.. what does that mean ? |
| 21:11.42 | sequencer | SIP/2.0 183 Session Progress |
| 21:11.44 | wdoekes2 | you could try to enable debugging on your polycom (logging to syslog?) if it exists |
| 21:12.01 | wdoekes2 | so called "early media" |
| 21:12.16 | sequencer | i have around 60 polycoms, it would be a hurricane |
| 21:12.52 | wdoekes2 | a dialog is set up as follows --> INVITE <-- 100 trying <-- 180 ringing <-- 183 progress <-- 200 --> ACK |
| 21:13.10 | wdoekes2 | the 1xx responses are optional.. the 200 signals that the dialog has begun |
| 21:13.36 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:13.54 | wdoekes2 | 183 is used to start media before the call is up.. e.g. to tell the user about the costs of the call |
| 21:14.35 | wdoekes2 | well.. if you're debugging one phone, you wouldn't need to enable syslog for all of them, now would you? |
| 21:15.02 | sequencer | yes, because the dropped calls are a random issue |
| 21:15.13 | sequencer | we need to be aware of them on all extensions |
| 21:16.24 | *** join/#asterisk slum (~s@173-9-8-170-BusName-boston.ma.boston.hfc.comcastbusiness.net) |
| 21:16.49 | slum | need help, how can I dial a number from the CLI to see if my SIP registration worked? |
| 21:17.15 | p3nguin | You don't need SIP registration to work to be able to make calls. |
| 21:17.23 | p3nguin | Just so you know. |
| 21:17.35 | slum | ok but I want to see if I can make an outbound call |
| 21:18.02 | p3nguin | Did you create extensions that route calls through a peer that has the capability of doing that? |
| 21:18.02 | slum | I have the context set up |
| 21:18.09 | slum | yes |
| 21:18.14 | p3nguin | Dial the number. |
| 21:18.23 | wdoekes2 | sequencer: well.. I don't see anything immediately wrong with the 183.. esp. not without seeing the original INVITE that starts the dialog. blame points at the polycom for now |
| 21:18.35 | slum | p3nguin, I can't connect a softphone |
| 21:18.47 | p3nguin | Create a sip peer for it. |
| 21:19.03 | slum | How do I do that? |
| 21:19.20 | p3nguin | In sip.conf, create an entry for your phone, just like you had to do for your ITSP. |
| 21:19.38 | slum | ok |
| 21:19.44 | p3nguin | I'm starting to get the feeling that you haven't read The Book. |
| 21:20.01 | slum | nope |
| 21:20.05 | slum | link? |
| 21:20.12 | p3nguin | ~newbook |
| 21:20.13 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 21:20.14 | p3nguin | ~book |
| 21:20.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 21:20.25 | p3nguin | new one and old one, respectively |
| 21:20.58 | wdoekes2 | boch: I don't know about jabber.. but you could try tcpdump if the jabber module doesn't support packets dumps. |
| 21:21.22 | wdoekes2 | boch: and then there's 'core set debug N filename.c' |
| 21:21.47 | *** part/#asterisk tracep (~tracep@107.7.25.234) |
| 21:22.12 | slum | thanks |
| 21:22.45 | wdoekes2 | titter: set+rtpproxy sounds like the way to go |
| 21:22.51 | wdoekes2 | s/set/ser/ |
| 21:23.31 | sequencer | wdoekes2 i am getting the pastebin for the call sart for you ... |
| 21:28.20 | sequencer | wdoekes2 i am not sure if this is sufficient http://pastebin.com/RGseZqqj |
| 21:29.39 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:33.41 | wdoekes2 | was that a failed call too? |
| 21:33.49 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 21:33.57 | sequencer | its the same call, this is the beginning (start) of the call |
| 21:34.24 | sequencer | the call was in progress for some time then it just drops while parties are on the phone |
| 21:35.30 | wdoekes2 | how long is "some time"? and why isn't the peer starting the dialog with 200? |
| 21:36.02 | sequencer | it varies between seconds and about 20-30 minutes |
| 21:36.13 | sequencer | why , i dont know.. let me check |
| 21:36.25 | titter | wdoekes2: thanks, and there goes my weekend lol. |
| 21:36.54 | wdoekes2 | if you know what you're doing, it won't take the whole weekend ;) |
| 21:37.43 | titter | Never messed with either of other two ... so mostly it will be reading and looking at examples. Since I am doing nothing but routing everything on ... I don't think it will be bad |
| 21:39.03 | sequencer | cant see any 200 signal |
| 21:39.07 | titter | Now the question, I wonder how this would work in a cloud hosted setup if I wanted to expand/cut costs and provide multiple geographical locations to our offices. |
| 21:39.46 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 21:39.47 | wdoekes2 | titter: http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6/modules/nathelper/examples/alg.cfg?revision=7553&content-type=text%2Fplain |
| 21:40.44 | wdoekes2 | sequencer: asterisk doesn't support 100rel: calls without a 200 within a certain amount of time should get regarded as down |
| 21:41.02 | wdoekes2 | but then we're talking minutes, not seconds |
| 21:41.20 | sequencer | hmmm.. could that be the issue of dropped calls ? |
| 21:41.27 | wdoekes2 | titter: and now it will take the weekend and then some ;) |
| 21:41.28 | sequencer | how would i solve this ? |
| 21:41.29 | titter | wdoekes2: <3 |
| 21:41.38 | *** join/#asterisk bmg505 (~leon@196-209-120-116.dynamic.isadsl.co.za) |
| 21:42.32 | wdoekes2 | sequencer: answer() the call in your asterisk config |
| 21:42.41 | *** join/#asterisk jkroon (~jkroon@dsl-241-247-239.telkomadsl.co.za) |
| 21:42.48 | sequencer | am using freepbx |
| 21:42.49 | wdoekes2 | this would be simple, if you weren't using freepbx ;) |
| 21:42.58 | wdoekes2 | #freepbx |
| 21:44.36 | sequencer | ok, what should i ask for in specific ? |
| 21:44.47 | sequencer | i am an * newbie |
| 21:46.47 | wdoekes2 | no clue ;) if you need help rm -rf'ing your freepbx install, we can help you though |
| 21:47.22 | sequencer | ive been having these issues with false fax detection |
| 21:47.23 | roxdragon | exist an client SIP for asterisk? |
| 21:47.29 | roxdragon | Type OS android |
| 21:47.54 | wdoekes2 | any SIP client should work |
| 21:48.04 | sequencer | i disabled faxes and things went good, when i enabled faxdetect=yes on the sip trunk it started giving these troubles back again |
| 21:48.05 | wdoekes2 | I believe there are plenty to choose from |
| 21:48.41 | wdoekes2 | sequencer: if you traced the problems to faxdetect, you really should've mentioned that |
| 21:48.50 | wdoekes2 | I don't have any experience with that though |
| 21:52.04 | sequencer | its fine though, we will see how it goes |
| 21:58.26 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 21:59.12 | titter | Another random one, anyone ever hide the UA or at least the version number on Asterisk? |
| 22:00.33 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 22:00.40 | wdoekes2 | I believe there is a setting somewhere, titter |
| 22:00.53 | ClintGoudie-Nice | Is there a way to set the type of transfer (bridged vs blind) on a sip trunk? |
| 22:01.05 | wdoekes2 | ;useragent=Asterisk PBX ; Allows you to change the user agent string |
| 22:02.16 | titter | Thank |
| 22:02.19 | wdoekes2 | ClintGoudie-Nice: isn't the trunk supposed to be initiating the transfer? |
| 22:06.34 | titter | wdoekes2: Thanks for that, found it on Google as you said it. sipv shows unknown which makes me happy. |
| 22:14.22 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 22:15.13 | ClintGoudie-Nice | wdoekes2: I get a call in from a cisco call manager trunk, it rings a phone, and then from follow me, after a certain number of rings it gets transferred to another trunk. When that happens it drops the call instead of completing the transfer. Previously I've seen this happen with ccm due to the transfer mode. |
| 22:15.24 | *** join/#asterisk tstorm (~tstorm@173-164-230-21-SFBA.hfc.comcastbusiness.net) |
| 22:21.43 | roxdragon | how to install codec g722? |
| 22:23.48 | fauxalliance | roxdragon, copy it to the modules directory and add load it via the CLI |
| 22:24.44 | *** part/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 22:24.48 | roxdragon | fauxalliance, how to? |
| 22:24.59 | roxdragon | where is the module? |
| 22:26.49 | WIMPy | Where all other modules are, as well. |
| 22:27.31 | WIMPy | By defalut you will have it. |
| 22:27.42 | fauxalliance | roxdragon, 'locate modules |grep asterisk' |
| 22:28.00 | *** join/#asterisk clu3 (~steve@186.1.193.254) |
| 22:30.23 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 22:30.59 | roxdragon | thanks |
| 22:31.06 | roxdragon | don't work MOH |
| 22:45.48 | *** join/#asterisk phyburn (~phyburn@wsip-70-165-35-234.oc.oc.cox.net) |
| 22:46.52 | brainiac | exit |
| 22:47.10 | roxdragon | ~moh |
| 22:47.10 | infobot | it has been said that moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf, or originally from http://www.freeplaymusic.com |
| 22:47.35 | *** join/#asterisk DaneM (~Dane@c-24-10-55-70.hsd1.ca.comcast.net) |
| 22:48.15 | *** join/#asterisk cyborg-one (1000@85-238-110-10.broadband.tenet.odessa.ua) |
| 22:50.35 | DaneM | Hello, everybody. I'm thinking about getting back into making asterisk phone systems using Asterisk. The last time I attempted it (and before that, too), the documentation was a bit sketchy and often out of date, depending on where I looked (about 2-3 years ago). Can anyone tell me where the best place to find reliable, up-to-date instructions on best practices and so forth would be? I also need to know what the best modern hardware for |
| 22:51.09 | paulc | DaneM: The Asterisk wiki is probably a good place to start: https://wiki.asterisk.org/wiki/dashboard.action |
| 22:51.55 | roxdragon | help |
| 22:51.56 | roxdragon | [Mar 18 23:50:25] WARNING[3870]: file.c:644 ast_openstream_full: File /var/lib/asterisk/moh/romance does not exist in any format |
| 22:51.56 | roxdragon | [Mar 18 23:50:25] WARNING[3870]: res_musiconhold.c:325 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/romance': No such file or directory |
| 22:52.13 | DaneM | paulc: Thanks! I'll start there. Do you happen to know off the top of your head what the most reliable hardware for 4 FXO ports is? I've had trouble finding that out in the past. |
| 22:52.33 | paulc | DaneM: Do you mean server hardware? Or telephony hardware to serve 4 phone lines? |
| 22:53.24 | DaneM | paulc: Server hardware: 4 incoming POTS lines from AT&T. It'll go out to local phones via VOIP. |
| 22:54.07 | DaneM | I'll probably be using typical PC hardware for the server. |
| 22:54.18 | *** join/#asterisk garymc (~chatzilla@host86-145-43-73.range86-145.btcentralplus.com) |
| 22:54.43 | paulc | DaneM: Can't comment on the POTS lines, we mainly use SIP, but I think the Digium boards are generally considered the best. Or alternatively an FXO/SIP gateway (Adtran springs to mind, can't remember the other one I'm thinking of) |
| 22:55.01 | paulc | Server wise.. HP, Dell, etc.. the voip-info.org wiki has some other people's success stories. |
| 22:55.40 | DaneM | paulc: thanks again. I'll check that out, too. The last few attempts involved buying expensive boards only to find them outdated and out of driver support :-p |
| 22:56.00 | paulc | DaneM: Buy your boards from Digium and they come with support, warranty, etc, and you won't have any problems :-) |
| 22:56.25 | DaneM | paulc: nodnod. Thanks. I had previously bought digium boards on ebay...a mistake, it seems |
| 22:56.55 | paulc | As the saying goes... "You gets what you pays for" - lots of knock-off/clone boards on there I think |
| 22:57.27 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 22:57.59 | *** join/#asterisk phyburn (~phyburn@wsip-70-165-35-234.oc.oc.cox.net) |
| 22:58.18 | DaneM | paulc: definitely. Learned my lesson (I think....) |
| 22:58.54 | DaneM | Incidentally, which IRC client do you folks prefer. I'm using pidgin, but it doesn't seem to be well-designed for this. |
| 22:59.39 | paulc | DaneM: I SSH into my Linux box, then run irssi in a terminal shell session. On windows I previously had success with X-Chat. |
| 23:00.32 | DaneM | Cool. I'm dual-booting (Windows/Linux), so unfortunately ssh isn't really good for me :-p I'll try Xchat. |
| 23:00.59 | DaneM | paulc: Thanks again. |
| 23:01.25 | *** part/#asterisk DaneM (~Dane@c-24-10-55-70.hsd1.ca.comcast.net) |
| 23:02.02 | *** join/#asterisk KNERD (~KNERD@adsl-99-96-118-71.dsl.hrlntx.sbcglobal.net) |
| 23:02.11 | paulc | no worries :) |
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| 23:07.34 | gurra | anyone tried to add a phone number on e164.org lately? the verification call is not working for me... |
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| 23:28.26 | *** join/#asterisk Mezevenf (~Mezevenf@mail.kenlee.com.au) |
| 23:28.38 | Mezevenf | hey guys |
| 23:29.58 | Mezevenf | anyone got a sec to give me some pointers with a B410P? |
| 23:30.08 | WIMPy | ~ask |
| 23:30.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:31.34 | Mezevenf | Ok I have a B410P that I want to integrate with an existing PBX, I have it jumpered to NT mode and configured as much as it requires as far as I can tell. Should the lights on the card be green when under NT mode and plugged into the PBX? |
| 23:32.40 | WIMPy | I don't know about the specific feature of the b410p, but most probably yes. |
| 23:32.59 | WIMPy | But possibly only after the first connection attempt. |
| 23:33.12 | Mezevenf | Should I be incorporating the existing NT1 devices with the B410P or bypassing them entirely? |
| 23:33.21 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 23:33.45 | WIMPy | No, just a direct connection. |
| 23:34.14 | WIMPy | Well, does the b410p reconfigure the port when jumpering NT mode? |
| 23:35.31 | Mezevenf | As far as I can tell the only differentiation in config between TE and NT is setting the CPE settings as termination mode is commented out elsewhere |
| 23:36.00 | WIMPy | was asking about the physical connection. |
| 23:36.08 | WIMPy | I'm just having a look at the manual. |
| 23:36.21 | Mezevenf | I assume so as the jumpers cross the wires |
| 23:36.34 | WIMPy | Jes, looks like they do. |
| 23:37.00 | Mezevenf | bascially I have NT1 boxes from Telstra here which the system was plugged into. I'm now bypassing them and plugging into the B410P directly |
| 23:37.13 | WIMPy | Yes, it clearly states it 'eliminates the need to use a cross over cable'. |
| 23:37.44 | WIMPy | Yes, just put the plug that was in the NT in to the card. |
| 23:38.39 | Mezevenf | Should I be expecting green lights without any config at all or only after correct config? |
| 23:39.21 | WIMPy | You need correct config and a working link. |
| 23:39.55 | Mezevenf | Link to the PBX? as its not linking to any BRI carrier line |
| 23:40.09 | WIMPy | Yes, the link to the PBX. |
| 23:40.17 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:40.47 | Mezevenf | Do you know the pinout of the B410P in NT by any chance? As the existing PBX does not use regular RJ45 |
| 23:41.59 | WIMPy | NT is just Tx and Rx crossed over. |
| 23:42.15 | WIMPy | It should be the same as the NT. |
| 23:42.59 | *** join/#asterisk cyborg-one (1000@188.115.188.104) |
| 23:43.09 | Mezevenf | The only thing that has me concerned is that the system itself only uses 2pairs per existing NT1 connection, so its not a full blown cable |
| 23:43.44 | Mezevenf | but just a single 8pair comes from the PBX to handle 4 existing NT1 boxes |
| 23:43.53 | *** join/#asterisk n1x0n (nixon@n1x0n-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 23:43.56 | WIMPy | Yes, it's 3456 only. 12 and 78 are only for additional power supply. But I've never seen that being used anywhere. |
| 23:44.17 | Mezevenf | Ok, thank you for your help WIMPy, much appreciated! |
| 23:46.18 | n1x0n | Hello, is it possible to have asterisk listening on more then one port ? My cellurar operator blocks port 5060 :-/ I did some googling and testing byt it always bind on one port - any suggestions much appreciated |
| 23:46.32 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:46.36 | WIMPy | n1x0n: No |
| 23:46.45 | WIMPy | But you can use iptables, off course. |
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| 23:47.45 | n1x0n | WIMPy: mm yeah that was my backup plan , thx |
| 23:48.08 | neurosys | can someone tell me what version * began using CoS? |
| 23:49.46 | WIMPy | Mezevenf: Do you use ptp or ptmp? |
| 23:51.11 | Mezevenf | using PTP atm |
| 23:51.33 | WIMPy | ok |
| 23:53.59 | Mezevenf | ok 100ohm dip brought up green |
| 23:54.30 | Mezevenf | since I'm using this as outbound only, should I setup a custom extension per line? What would you recommend? |
| 23:56.12 | WIMPy | Extension for the BRIs? |
| 23:56.46 | WIMPy | If you don't want to send calls TO the PBX you don't need extensions for it. |
| 23:57.55 | WIMPy | If you don't have termination elsewhere, you want that enabled. The manual is actually misleading there. |
| 23:58.14 | WIMPy | Or just wrong. |