| 00:00.07 | cusco | almost misread it as 'iphone' |
| 00:00.16 | ChannelZ-Wk | And don't count out amazon, they might have what you're looking for on Prime if you have that. |
| 00:00.34 | cusco | lol |
| 00:01.13 | ChannelZ-Wk | 335 for $98 |
| 00:01.57 | igcewieling1 | Click your heels together say "Hey hey NSA I need a Polycom today!" and then recite your credit card number |
| 00:02.12 | ChannelZ-Wk | All they'll do is buy porn |
| 00:03.21 | cusco | lol |
| 00:03.26 | smash` | i found a 331 for 89 on dell.com |
| 00:04.09 | smash` | Nice link though |
| 00:04.14 | smash` | they dont have pricing online? |
| 00:04.43 | smash` | whats the difference between the 331 and 332 and 335 |
| 00:05.25 | *** join/#asterisk ISO8601 (yano@freenode/staff/yano) |
| 00:05.38 | smash` | which one is the PoE? |
| 00:08.32 | [TK]D-Fender | there is no 332 |
| 00:08.38 | [TK]D-Fender | And the other 2 are both PoE |
| 00:08.42 | [TK]D-Fender | (802.3af) |
| 00:12.57 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 00:17.36 | *** join/#asterisk tyman (~tyman@12.226.100.130) |
| 00:18.29 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:19.56 | carbinemonoxide | Okay, with xmpp debugging enabled I see that asterisk actually sees the number I'm calling accept the call. |
| 00:20.54 | carbinemonoxide | http://pastie.org/private/mo6xjd91rrfvo5aj6oytg |
| 00:21.24 | tyman | Asterisk 11.5.1 w/DPMA 11.0_1.6.0 is throwing the following error http://pastie.org/8315425 to the console every few seconds. It only starts after ~24-48hrs after a core restart. A core restart is the only way to clear the error as a reload of the dpma module does nothing. So, looks like a asterisk core issue from here. Googling has been unproductive. |
| 00:24.52 | tyman | More possible helpful info: http://pastie.org/8315434 |
| 00:28.05 | smash` | Thanks, [TK]D-Fender. Just wanted to mkae sure the 331 was not 802.3af |
| 00:28.08 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 00:42.03 | [TK]D-Fender | smash`: it is |
| 00:49.12 | smash` | the 331 is POE? |
| 00:49.45 | [TK]D-Fender | yes |
| 00:50.31 | smash` | shit, what about the 321? |
| 00:51.55 | smash` | Nevermind I got it, Thank you info. |
| 00:55.02 | [TK]D-Fender | how is that BAD? |
| 00:55.18 | [TK]D-Fender | 32X,33X, ALL PoE. |
| 00:55.25 | [TK]D-Fender | EVERY modern Polycom is natively PoE |
| 01:08.59 | *** join/#asterisk smirker (~x@101.162.79.153) |
| 01:13.11 | smirker | Is it just me or is the REPLACE func incredibly buggy? |
| 01:13.33 | smirker | (at least in 1.8.15-cert2) |
| 01:14.26 | smirker | exten => s,n,Set(test=*101#) exten => s,n,NoOp(Result: ${REPLACE(test,*)}) ; Result: 11## (expected result 101#) |
| 01:14.30 | smirker | or am i misunderstanding something? |
| 01:14.55 | *** join/#asterisk smash` (smash@c-50-139-7-160.hsd1.or.comcast.net) |
| 01:23.39 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:23.46 | *** join/#asterisk suneye (~atcmmi@119.122.152.219) |
| 01:26.55 | igcewieling1 | smirker: you looked at "core show function REPLACE"? |
| 01:30.31 | smirker | igcewieling1: sure have. If i try to remove all * characters from *101#, it returns 11## where I would expect 101# |
| 01:31.37 | igcewieling1 | smirker: you'd be surprised at how many people don't look at the built in docs before asking here. I don't have any suggestions for you though. |
| 01:33.22 | smirker | igcewieling1: i've even read the source code for the function, and it seems right. maybe a * at the start of a string had special meaning. i'm not sure ;o |
| 01:34.22 | igcewieling1 | smirker: * and # is not usually special in Asterisk since it can be a DTMF digit too. |
| 01:34.41 | smirker | igcewieling1: that only makes sense |
| 01:34.49 | igcewieling1 | you can always try escaping it with a \ though I doubt it will help |
| 01:43.01 | *** join/#asterisk suneye (~atcmmi@50.2.43.42) |
| 01:51.58 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:58.16 | *** join/#asterisk techman97 (~me@68-117-53-142.dhcp.roch.mn.charter.com) |
| 01:59.28 | techman97 | good evening all - I'm banging my head against a wall here. Asterisk box was working fine earlier today - routing just fine to a SIP provider on an MPLS connection. Now, I can ping/traceroute/etc to the host still just fine, but outbound SIP packets will not even touch the connection. Inbound calls work no problem, just fine. |
| 01:59.33 | *** join/#asterisk iq (~iq@cab10-39.1scom.net) |
| 01:59.33 | iq | hi |
| 01:59.45 | techman97 | any thoughts on anything to check? |
| 02:15.37 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 02:19.44 | *** join/#asterisk nam3l3zz (~quassel@86-46-203-155-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 02:19.55 | nam3l3zz | hi |
| 02:20.17 | nam3l3zz | is there a way to transfer aux data via Dial app, if iax2 is used ? |
| 02:20.39 | *** join/#asterisk mintos (~mvaliyav@14.96.184.104) |
| 02:20.49 | nam3l3zz | aux data = 15 numeric chars |
| 02:21.28 | igcewieling1 | nam3l3zz: "core show functions like IAX" |
| 02:21.48 | *** join/#asterisk B (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 02:21.53 | igcewieling1 | you should do a "core show functions" or "core show applications" every once in a while |
| 02:22.39 | j4jackj | Guest45724: strange nick you had, it's nickserv registered to someone else so don't use it again |
| 02:24.46 | nam3l3zz | igcewieling1: iaxvar a bit like sip header thing, thanx |
| 02:27.43 | Guest45724 | Hi.I am using chan_dongle and dongle is also connected with server and is have confirmed with lsusb. |
| 02:28.57 | Spengler1 | is there a way to auto forward messages from one voicemail box to another? |
| 02:29.20 | Guest45724 | module is loading and reloading properly. but issue is when is see dongle show devices |
| 02:29.47 | Guest45724 | ID Group State RSSI Mode Submode Provider Name Model Firmware IMEI IMSI Number dc_3715_0092 0 Not connec 0 0 0 NONE Unknown |
| 02:31.13 | Guest45724 | this is the out put. Firstly i was working fine |
| 02:31.31 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 02:33.47 | nam3l3zz | Guest45724: if dongle show devices is empty, either chan_dongle.so is not loaded or chan_dongle.conf is not filled in right |
| 02:34.46 | nam3l3zz | Guest45724: use pastebin of some sort for pasting anything, it is more prefferable these days |
| 02:36.24 | *** join/#asterisk devand (~devand@66.222.231.104) |
| 02:37.13 | nam3l3zz | Guest45724: prior to trying to see the dongle in asterisk, connect ot its app port (/dev/ttyUSB*) via minicom and do any AT command (ie: ATI), to see if ur linux box actually sees the thing |
| 02:37.22 | nam3l3zz | *to |
| 02:41.57 | Guest45724 | ok.Actually it was working previously. Also can u share me pastebin link |
| 02:42.34 | Guest45724 | also this is the output of ls /dev/ttyUSB* |
| 02:44.11 | nam3l3zz | igcewieling1: can iax2 peers exchange with data, without initiating Dial ? |
| 02:45.06 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:45.58 | Guest45724 | nam3l3zz: please send me link of pastebin so that i can put my chan_dongle.conf file |
| 02:46.13 | igcewieling1 | Asterisk is not designed to do stuff "outside of a call" |
| 02:46.21 | igcewieling1 | ~pb |
| 02:46.21 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:46.32 | igcewieling1 | heh, nevermind |
| 02:47.45 | nam3l3zz | Guest45724: go to pastebin.org (as example), paste, submit, once that much is done, the page you will land on, will have in it url, the url others would need to assist |
| 02:48.16 | Guest45724 | nam3l3zz: thanks |
| 02:48.16 | ChannelZ | Did you atually register the nick Guest45724 ? |
| 02:48.51 | nam3l3zz | i dont care :) |
| 02:51.17 | Guest45724 | http://pastebin.com/Wi2ugbNw |
| 02:51.59 | nam3l3zz | igcewieling1: can a call incoming via iax2, be dropped with a custom reply/"exit code", not just busy/not available/etc ? |
| 02:53.23 | nam3l3zz | Guest45724: was this config auto generated, or you did it yourself ? |
| 02:53.35 | Guest45724 | plz check this is configuration of dongle_devices.conf which is include in dongle.conf |
| 02:54.53 | nam3l3zz | Guest45724: try avoiding an include, do it default way, while don't have an exotic setup, will make you an example chan_dongle.conf with ur imei/imsi codes |
| 02:54.58 | igcewieling1 | nam3l3zz: "core show application Hangup" Are you new to Asterisk? |
| 02:55.24 | nam3l3zz | igcewieling1: to telecommunications in general, to be honest :) thanx, ur quiet friendly |
| 02:55.25 | Guest45724 | http://pastebin.com/FBZATXhv |
| 02:55.31 | Guest45724 | this is dongle.conf |
| 02:55.51 | igcewieling1 | nam3l3zz: I'm an asshole, but you caught me in a good mood. |
| 02:56.06 | nam3l3zz | igcewieling1: am lucky so ;) |
| 02:56.13 | igcewieling1 | ~book |
| 02:56.13 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:56.50 | igcewieling1 | nam3l3zz: read the book, study "core show applications" and "core show functions" and the .conf.sample files included in Asterisk (at least the ones relevant to what you might be trying to do) and you'll be an expert in no time. |
| 02:56.53 | nam3l3zz | Guest45724: change imei & imsi to imei of your dongle & imsi of your sim card, start from assigning either |
| 02:57.15 | nam3l3zz | Guest45724: don't alter the rest, while you are not aware of what other variables do |
| 02:58.33 | Guest45724 | is any way to check imei of dongle remotely as well as imsi of sim card. Because it is on remote location |
| 02:59.09 | igcewieling1 | nam3l3zz: there are three basic kinds of questions here. 1) can't really expect you to know the answer, such as your IAX question 2) you should have read the docs, arguably your hangup question might be considered this and 3) don't even know the right questions to ask. |
| 02:59.15 | nam3l3zz | igcewieling1: sip heavily relies on nat, found just tiny bit of information about nat in the book. have book about sip, has plenty info on nat, abviously also o'reilly :) certain areas are not too well covered in the book imho |
| 02:59.22 | igcewieling1 | type 1 and 3 and the ones most of us enjoy answering |
| 02:59.51 | nam3l3zz | Guest45724: both can be done via a set of AT commands |
| 03:01.02 | igcewieling1 | NAT falls under type 1. Here is your answer: If Asteirsk is NOT behind NAT and your clients are, then set nat=yes in [general] or the device section of sip.conf. If Asterisk is behind NAT (regardless of if the client is behind NAT) then forward port 5060 UDP (NOT TCP) on your router to Asterisk, in sip.conf set localnet= and externip= to the correct values and set directmedia (canreinvite in older Asterisk versions) to no. |
| 03:01.15 | igcewieling1 | the above will solve 90% of NAT issue. |
| 03:01.31 | nam3l3zz | Guest45724: do just imei will do for a start, connect via mincom linux programm, to appropriate serial port , find out serial port via dmesg, most likely if have no other usb-serial convertors it is going to be /dev/ttyUSB0, ie: minicom -D /dev/ttyUSB0 |
| 03:02.07 | nam3l3zz | Guest45724: once in minicom, do ATI+Enter, following lines will show you imei, in most of the cases. what model is your dongle ? |
| 03:02.53 | igcewieling1 | oh! and question type 4) what you want to know or are trying to do is so uncommon nobody wants to help. Guest45724's questions are type 4 |
| 03:03.03 | ChannelZ | Meh. Doing RTCP debug, is "Fraction lost" meant to be a percent of bytes? |
| 03:03.48 | igcewieling1 | ChannelZ: likely percent of packets, not bytes |
| 03:04.07 | ChannelZ | great, that's even worse |
| 03:04.16 | ChannelZ | Fraction lost: 11 |
| 03:04.21 | igcewieling1 | nam3l3zz: most info on SIP and NAT on the internet is crap and unless specific to Asteirsk doesn't apply to Asterisk |
| 03:04.46 | igcewieling1 | ChannelZ: I do not know for sure, but it seems silly to keep track of bytes lost instead of packets list. |
| 03:05.43 | nam3l3zz | igcewieling1: not deep enough, or eternal fight between admins and more fundamentially prepared buddies, unfortunatelly I am looking for an advice from "asterisk operators" (expressions fully defines 95% of the "locals" in here) , I would fancy to talk to a guy, who would be able to say, I'd fancy to rewright asterisk in pure assembler just to make it slightly faster... :) |
| 03:05.56 | ChannelZ | Dunno. It says "Sent packets: xxx Sent octets: xxxxxx Fraction lost: xx" so it seems to know both |
| 03:06.13 | ChannelZ | In any case Vitelity seems screwed up. |
| 03:06.31 | nam3l3zz | igcewieling1: *expression |
| 03:06.50 | igcewieling1 | ChannelZ: if you want to test, 1-256-425-7814 is on vitelity and connected to an asterisk server. |
| 03:07.08 | igcewieling1 | you'll get a hello, then a ringing sound forever |
| 03:07.25 | igcewieling1 | great for telemarketers to call |
| 03:07.30 | *** join/#asterisk smash` (smash@c-50-139-7-160.hsd1.or.comcast.net) |
| 03:07.41 | ChannelZ | hmm let me think about that |
| 03:08.29 | igcewieling1 | nam3l3zz: I manage asterisk systems which handle around a million calls a month as well as more than 60 other asterisk boxes. does that qualify 8-) |
| 03:08.37 | ChannelZ | I'm not sure that helps since my connection through them is already screwed up |
| 03:08.59 | igcewieling1 | ChannelZ: I don't know either. |
| 03:09.42 | igcewieling1 | Granted, that isn't exactly a "large carrier", but it isn't small either |
| 03:10.47 | *** join/#asterisk tyman (~tyman@75-149-49-133-SFBA.hfc.comcastbusiness.net) |
| 03:10.59 | tyman | tyman: Asterisk 11.5.1 w/DPMA 11.0_1.6.0 is throwing the following error http://pastie.org/8315425 to the console every few seconds. It only starts after ~24-48hrs after a core restart. A core restart is the only way to clear the error as a reload of the dpma module does nothing. So, looks like a asterisk core issue from here. Googling has been unproductive... |
| 03:11.04 | smash` | Hey whats a Softphone that I can download to do test comms. |
| 03:11.14 | ChannelZ | Zoiper, Blink, 3CX, linphone |
| 03:11.15 | nam3l3zz | igcewieling1: about 15 years ago I used reverse engineer binary code of some quiet known software products these days, conclusion is, had much more spare time to develop myself then, including a phd degree in computer science :) |
| 03:12.24 | tyman | More possible helpful info: http://pastie.org/8315434 |
| 03:12.34 | ChannelZ | I just called myself from my cell and got 'Fraction lost: 15' before I even answered. Something must be jacked up. |
| 03:12.41 | igcewieling1 | nam3l3zz: *nod* I got too old to enjoy that sort of thing, tough I still enjoy doing perverted stuff with telecoms |
| 03:12.58 | igcewieling1 | ChannelZ: what version of Asteirsk, are you using direct media |
| 03:13.18 | ChannelZ | 11.5.1 and no |
| 03:13.40 | igcewieling1 | should work. do pings show packet loss? |
| 03:13.52 | ChannelZ | Yeah, just trying that now |
| 03:14.39 | ChannelZ | Heh yup. I wonder if Comcast has screwed something up. Testing some other hosts... |
| 03:14.57 | igcewieling1 | that's right blame the carrier first. *tease* |
| 03:15.10 | *** join/#asterisk sawgood1 (~sawgood@unaffiliated/sawgood) |
| 03:15.45 | ChannelZ | well I was doing direct SIP calls and it was fine.. g722 even (what bitrate does * use for g722?) |
| 03:15.47 | *** join/#asterisk tyman_ (~tyman@75-149-49-133-SFBA.hfc.comcastbusiness.net) |
| 03:15.59 | igcewieling1 | G722 is 64K |
| 03:16.05 | igcewieling1 | + overhead of course |
| 03:16.22 | ChannelZ | so same as ulaw basically. |
| 03:16.43 | igcewieling1 | *nod* but far far far better audio quality if you have g722 end to end |
| 03:17.18 | igcewieling1 | g722 is the most widely support "HD" codec. |
| 03:17.22 | ChannelZ | right. I'm doing g722 from here to work, SIP-to-SIP and it was sounding fine yet my PSTN calls SIP to Vitelity and out was choppy and barfing. |
| 03:17.30 | ChannelZ | Getting mixed results now. |
| 03:17.37 | nam3l3zz | igcewieling1: I was all the time into telecoms, but never had time to study it on a pro level, in my spare time I try developping existing knowledge... p.s. 4 a.m. in Ireland :) |
| 03:17.56 | igcewieling1 | nam3l3zz: have a guinness and go to bed. 8-) |
| 03:18.22 | nam3l3zz | igcewieling1: great idea, am in a lab, full of cameras, would fancy a pint thou... :) |
| 03:19.08 | igcewieling1 | you're in Ireland, I thought you could drink anywhere! |
| 03:19.41 | nam3l3zz | :))) |
| 03:20.20 | nam3l3zz | i stayed late at work, not to be explaining to the gf, why am I not in bed... :) |
| 03:21.31 | nam3l3zz | in ireland the most promoted fiber is via dsl phone line, "the guiness way..." |
| 03:21.46 | ChannelZ | hmm nope. It's either Vitelity or a route in between me and them. Getting average 6% packet loss to Vitelity. Barf. |
| 03:22.02 | ChannelZ | does the support ticket-a-majig |
| 03:22.14 | igcewieling1 | nam3l3zz: *nod* wired the whole country with fiver in the late 1980s/early 1990s, IIRC. |
| 03:22.24 | igcewieling1 | s/fiver/fiber |
| 03:23.10 | igcewieling1 | they converted the whole telecom network to digital, made phone hacking much harder. |
| 03:23.24 | nam3l3zz | but we've got the best guiness, brewed in ireland only for the local market :) |
| 03:23.41 | igcewieling1 | Guinness is the only thing I missed when I had to stop drinking. |
| 03:24.27 | nam3l3zz | guinness is nice, have a nice italian cigar for 20 minute walk home , which may commence shortly :) |
| 03:24.58 | igcewieling1 | In parts of the USA you could get arrested for that. |
| 03:25.12 | nam3l3zz | if i was in states like u r , would smoke carribean cigars, they cost there fuk all i d say |
| 03:25.21 | nam3l3zz | fek, hows that ? |
| 03:25.35 | igcewieling1 | nam3l3zz: some cities have outlawed smoking in public places |
| 03:26.06 | nam3l3zz | even places like a footpath on public street ? |
| 03:27.40 | igcewieling1 | http://en.wikipedia.org/wiki/List_of_smoking_bans_in_the_United_States Several cities have banned smoking on public streets |
| 03:28.18 | nam3l3zz | ireland was the first eu country to engage the public smoking ban thing, but streets aren't included, nor planned to be included as I hope, hard to smoke in London, especialy airports, I often fly throu, just few designated places outside the airports |
| 03:28.40 | nam3l3zz | sounds tough |
| 03:28.45 | igcewieling1 | nam3l3zz: remember many of the USAs original immigrants were too conservative for Victorian England |
| 03:28.58 | nam3l3zz | yep |
| 03:29.34 | igcewieling1 | Explains a lot when you think about it. LOL! |
| 03:29.48 | igcewieling1 | anyway, im off to sleep |
| 03:30.05 | nam3l3zz | i'll do the same shortly, thanx anyhow |
| 03:32.43 | ChannelZ | looks like a problem between level3 and vitelity |
| 03:36.00 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 03:36.57 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 03:40.46 | ChannelZ | heh shit now even VItelity's website is crawling |
| 03:41.18 | ChannelZ | Poof! "We are currently undergoing emergency maintanence. Please try back shortly." |
| 03:57.29 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
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| 05:04.05 | Spengler1 | vitelity back online |
| 05:04.19 | Guest45724 | nam3l3zz: thanks man for you to the point help issue of dongle has been resolved. Its working now |
| 05:04.44 | Spengler1 | everyone can relax. the dongle is working |
| 05:05.26 | Guest45724 | ha ha u also have faced issue in dongle |
| 05:05.43 | Spengler1 | no i just like the word dongle |
| 05:06.07 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 05:06.10 | Guest45724 | ha ha just try it then u will like this word more |
| 05:06.39 | Spengler1 | which dongle r u using? |
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| 06:42.26 | phix | dingle dongle? |
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| 08:13.32 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 08:19.33 | *** join/#asterisk TazzNZ (~pieter@2403:4d00:e000:2:e14e:ed67:dc02:4413) |
| 08:20.16 | TazzNZ | Hey Guys - I am looking for some help with XMPP and Asterisk 11 (AsteriskNow v3) and device_state |
| 08:20.16 | TazzNZ | please :) |
| 08:24.12 | ChannelZ | hmm |
| 08:28.45 | TazzNZ | hmm ? |
| 08:28.58 | TazzNZ | or is that un-related ? |
| 08:29.42 | ChannelZ | No.. what are you expecting from DEVICE_STATE |
| 08:30.22 | TazzNZ | I want presence across 2 servers |
| 08:30.34 | TazzNZ | (tbh, not worried about MWI) |
| 08:31.57 | TazzNZ | I have followed https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub |
| 08:32.02 | TazzNZ | without luck |
| 08:32.23 | TazzNZ | and then I found someone saying that they got it working using openfire XMPP server |
| 08:32.49 | TazzNZ | but I am not having any luck getting it to work |
| 08:36.18 | ChannelZ | Well that's a pretty large configuration... what bit isn't working? |
| 08:37.37 | TazzNZ | I believe the updates are going into openfire |
| 08:37.44 | TazzNZ | but the other server isn't seeing it |
| 08:37.51 | TazzNZ | e.g: |
| 08:37.59 | TazzNZ | core show hint 94166 |
| 08:38.01 | TazzNZ | <PROTECTED> |
| 08:38.03 | TazzNZ | 1 hint matching extension 94166 |
| 08:38.05 | TazzNZ | vs |
| 08:38.08 | TazzNZ | core show hint 94166 |
| 08:38.11 | TazzNZ | No hints matching extension 94166 |
| 08:38.18 | TazzNZ | first server vs second server |
| 08:38.45 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 08:39.10 | TazzNZ | if I enable xmpp debug, I can see the server messages going into xmpp |
| 08:39.14 | TazzNZ | "in a loop" |
| 08:39.34 | TazzNZ | like asterisk is publishing the state over and over |
| 08:40.28 | TazzNZ | ChannelZ: If you have a working config somewhere, I am more than happy to follow it :) |
| 08:40.42 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 08:44.40 | ChannelZ | well it seems like you have no hints on server 2 |
| 08:45.11 | TazzNZ | yip |
| 08:46.07 | ChannelZ | so how can it set the state for something that doesn't exist |
| 08:47.25 | TazzNZ | but doesn't "server2" see what "server1" is publishing ? |
| 08:47.58 | TazzNZ | via xmpp that is |
| 08:48.39 | ChannelZ | It might publish the state to XMPP, but your server2 has no hints to monitor. |
| 08:49.16 | TazzNZ | right - so server2 doesn't learn the hints from server1 via xmpp ? |
| 08:49.50 | ChannelZ | I doubt it |
| 08:50.56 | TazzNZ | let me see if I can try and create a hint on server2 that would match a hint on server1 |
| 08:52.25 | *** join/#asterisk felipealmeida (~user@177.158.58.78) |
| 08:52.37 | ChannelZ | The hint system is the hint system; XMPP is just a means of distributing the states, but the server still has to be made interested in a state to care. At least that's what I get out of this. |
| 08:53.29 | TazzNZ | I thought that since the docs said that you create a buddy system that they would exchange hints |
| 08:53.52 | TazzNZ | so I didn't think of the server needing to know about the state |
| 08:54.55 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 08:55.19 | gavimobile | using centos command line, how can I get information about what card my system is using? |
| 08:56.43 | ChannelZ | what card...in general? Or are you talking of DAHDI? or... |
| 08:58.20 | TazzNZ | ChannelZ: I think you where right ! :D |
| 08:58.35 | TazzNZ | core show hint 94336 |
| 08:58.37 | TazzNZ | <PROTECTED> |
| 08:58.39 | TazzNZ | 1 hint matching extension 94336 |
| 08:58.41 | TazzNZ | that is on the second server |
| 09:00.33 | TazzNZ | awesome - THANKS ChannelZ !!! |
| 09:00.56 | TazzNZ | just checked and that works |
| 09:01.08 | *** join/#asterisk hjf (hjf@unaffiliated/hjf) |
| 09:02.49 | ChannelZ | good |
| 09:07.47 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:21.11 | gavimobile | ChannelZ: I'm not sure what card I have period... |
| 09:21.52 | gavimobile | are there different tools for each card brand? |
| 09:23.52 | ChannelZ | I assume you're talking about a telephony card which is what I was asking |
| 09:24.34 | ChannelZ | dahdi_hardware will show you anything DAHDI can identify as something it can talk to |
| 09:24.48 | ChannelZ | otherwise lspci will show you everything the OS sees |
| 09:25.19 | gavimobile | ChannelZ: thanks |
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| 09:39.14 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 09:44.17 | WIMPy | Not just different tools. There are at least 10 completely divfferent kinds of drivers. |
| 09:48.53 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 09:54.13 | giany | hi |
| 09:54.18 | giany | i have something like this : exten => s,4,Set(_SIPADDHEADER23=Privacy: header\; session) |
| 09:54.53 | giany | how can I escape that ;? adding \ makes some providers say Bad request |
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| 11:50.52 | igcewieling1 | giany: use the SipAddHeader Application |
| 11:54.24 | phix | WIMPy, WIMPy, WIMPy! Hefty! Hefty! Hefty! |
| 11:57.56 | igcewieling1 | giany: I think the Privacy header can only have "id" or "none", but you'd have to look at the RFC to be sure. |
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| 13:00.34 | Katty | hi kiddos. |
| 13:03.59 | jmetro | <PROTECTED> |
| 13:04.30 | Katty | what's the word |
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| 13:10.55 | [TK]D-Fender | Katty: haven't you heard? |
| 13:11.28 | [TK]D-Fender | IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdISt |
| 13:11.29 | [TK]D-Fender | heWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD!IsaidTHEbirdBIRDbirdTHEbirdIStheWORD! |
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| 13:16.13 | boom^time | Why did family guy have to devolve into this. |
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| 13:31.04 | igcewieling | "They had this weird setup where they opened up their old phone, cut the speaker to the ringer, and spliced it to a overhead speaker so when the phone rang, the speaker rang instead of the phone" |
| 13:32.05 | [TK]D-Fender | igcewieling: I've suggested that exact thing in the past.... |
| 13:32.29 | igcewieling | [TK]D-Fender: heretic! |
| 13:32.44 | igcewieling | now they want to use this AND a viking overhead paging system at the same time. |
| 13:32.51 | igcewieling | any ideas? |
| 13:33.49 | [TK]D-Fender | Certainly doable.. in as much as they'll have ringing over the actual page making it hard to hear. If they want to be that dumb and the pay is worth it.... |
| 13:34.26 | igcewieling | they are that dumb and we are expected to do this as part of the already approved install fees |
| 13:37.08 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
| 13:37.41 | [TK]D-Fender | igcewieling: Well I guess that settles it. |
| 13:38.07 | [TK]D-Fender | igcewieling: Do it and when they complain that it isn't working out like expected ... time to jack up the "change" fee ;) |
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| 13:39.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:44.17 | igcewieling | [TK]D-Fender: Turns out the Viking units we use can operate in "ringer only mode" which should do what we need. |
| 13:45.12 | [TK]D-Fender | igcewieling: This would be pretty easy to do in software alone too. |
| 13:45.57 | igcewieling | replaces the Viking paging unit with a freight train horn |
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| 15:02.15 | jeffspeff | i think my google is broken. been searching for callerid lookup sources and can't find anything. i did find an article about callerid superfecta, but doesn't list any sources in it. |
| 15:02.15 | SuperNull | hey guys, is there a way to show the secret on 'sip show peer' ? |
| 15:02.40 | Qwell | SuperNull: modify chan_sip |
| 15:03.06 | SuperNull | okay. |
| 15:03.08 | SuperNull | i might do that ;) |
| 15:03.09 | SuperNull | lol |
| 15:03.25 | Qwell | It shouldn't be too hard. Just look for _sip_show_peer() |
| 15:03.30 | SuperNull | alright. |
| 15:03.33 | [TK]D-Fender | jeffspeff: CNAM <- |
| 15:03.34 | SuperNull | goes to look. |
| 15:03.53 | [TK]D-Fender | jeffspeff: And the name you mentioned is a FreePBX module name which is meaningless |
| 15:03.56 | igcewieling | jeffspeff: all the "free" CallerID lookup services are total and utter crap. |
| 15:04.44 | jeffspeff | yeah, i know it's a fpbx module, but i read through the article hoping it might show a few sources |
| 15:04.52 | jeffspeff | igcewieling, why are they all crap? |
| 15:04.53 | igcewieling | We use BulkCNAM. |
| 15:04.53 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 15:05.11 | igcewieling | jeffspeff: because telcos charge for cnam lookups -- they are not free. |
| 15:06.14 | navaismo | SuperNull, if your peer is a friend use sip show users |
| 15:06.53 | [TK]D-Fender | "sip show peers" will show "type=friend" as well\ |
| 15:07.19 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 15:07.32 | navaismo | i mean to see the secret |
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| 15:10.38 | SuperNull | OKAYYYY. |
| 15:11.30 | SuperNull | wtf. why can you do sip show users but 'sip show user' hides it.. GAHHHHHHYYYYYYY |
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| 15:12.58 | SuperNull | but thank you very much for tha navaismo |
| 15:13.01 | SuperNull | big help for real time.. |
| 15:13.29 | SuperNull | having issues with password caching. were in the process of converting to #exec <> database dumps.. its more reliable :-/ |
| 15:15.30 | igcewieling | SuperNull: your statements makes no sense. |
| 15:15.38 | igcewieling | you use #exec to do a database dump. |
| 15:16.03 | igcewieling | maybe you mean realtime, in which case, keep a bottle of vodka and a 100ct bottle of Tylenol handy, you'll need it. |
| 15:17.20 | SuperNull | no. |
| 15:17.52 | SuperNull | unfortunately everything was setup for real time so im literally dumping the realtime user table to sip.conf style output like you suggested ..? how is dat confusing .. you told me to .. i did it and it works awesome. |
| 15:23.14 | igcewieling | SuperNull: I thought you were saying you were converting from using exec to using a database (which I assumed is realtime) |
| 15:24.45 | igcewieling | SuperNull: don't know if I sent this to you before or not http://pastebin.ca/2448309 |
| 15:26.47 | SuperNull | igcewieling looks like mine only mine has some legacy 'fix ups' in it.. |
| 15:27.09 | SuperNull | i dont do the array key using assoc returns because of.. possible columns that make no sense (regserver.. stuff...) |
| 15:27.32 | igcewieling | SuperNull: *nod* our peers don't register so it isn't a concern for us |
| 15:27.55 | SuperNull | yep. |
| 15:28.11 | SuperNull | plus it makes it easy for the 1.4 stuff to still use the peer table.. and our software.. |
| 15:28.34 | SuperNull | im dedicated to switching to opensips/kamillio tho .. raw control of SIP is more important to me for these uses.. |
| 15:28.50 | SuperNull | at least for non media related stuff.. media related stuff will be asterisk 1.8/11. |
| 15:35.01 | igcewieling | We must route all media via Asterisk and all calls are custom routed, so Kamailio is less useful for us than for most. |
| 15:36.10 | SuperNull | you doing any transcoding ? |
| 15:39.01 | igcewieling | sometimes. The main reason we must send audio is contractual oblications with our main carrier. Another reason is if you want to send T.38 calls thru Asterisk audio MUST be routed through Asterisk too |
| 15:40.23 | igcewieling | s/main reason we must send audio/main reason we must send audio via Asterisk/s |
| 15:41.19 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 15:41.49 | file | the fact you have to do that is a bug |
| 15:42.19 | igcewieling | file: having to send audio via Asterisk in order for T.38 to work? |
| 15:42.25 | file | yes. |
| 15:44.02 | igcewieling | Cool. Maybe it will be fixed some day. In the meantime we have en entirely separate non-asterisk infrastructure for T.38 stuff. |
| 15:45.10 | igcewieling | SBC/Kamailio/Adtran Media gateways |
| 15:45.32 | file | https://issues.asterisk.org/jira/browse/ASTERISK-17273 |
| 15:46.33 | SuperNull | igcewieling so are you suggesting you know kamailio enough to set that up ;) |
| 15:46.49 | igcewieling | Created:23/Jan/11 6:59 AM I won't hold my breath. |
| 15:46.58 | igcewieling | SuperNull: no. we paid someone to do that |
| 15:47.02 | SuperNull | damn. |
| 15:47.05 | file | you don't have to, it's being worked on |
| 15:47.09 | SuperNull | i was gonna offer you money for some light training. |
| 15:47.18 | SuperNull | 'consulting' |
| 15:47.29 | igcewieling | though the kamailio is really only used for outbound t.38. inbound t.38 is direct from the SBC to the Adtrans |
| 15:48.05 | SuperNull | i see a patch on there ? |
| 15:48.08 | file | although the end result of that won't be having T.38 go direct, as that's just sheer madness |
| 15:49.20 | igcewieling | T.38 is such a tiny percentage of our calls it doesn't really matter all that much. Annoying for us, but nothing more. |
| 15:50.35 | igcewieling | file: why is not proxying T.38 data via Asterisk madness? |
| 15:50.50 | file | T.38 devices lie, and are generally broken in different ways |
| 15:51.03 | file | Asterisk normalizes/fixes |
| 15:51.43 | file | for very controlled circumstances it could work |
| 15:52.06 | igcewieling | we have ALL Adtran boxes or Asterisk as endpoints |
| 15:53.34 | igcewieling | We are not exactly happy with Adtran at the moment either. AOS 10.x will fail to boot if you configure an ntp server which is unreachable -- requires a tech dispatch to fix. |
| 16:00.24 | SuperNull | i love them adtran ta900 series ATAs. |
| 16:01.58 | SuperNull | the ip rtp quality-monitoring is ... NICEEEE |
| 16:02.10 | igcewieling | We usually do, but not when we have to dispatch a tech because of a bug |
| 16:02.11 | SuperNull | what do you guys do for mass ATA ? i know some people use cisco but.. |
| 16:02.20 | igcewieling | We use Adtrans |
| 16:02.23 | SuperNull | yeah |
| 16:02.27 | SuperNull | TA900 series is def bug prone.. |
| 16:02.33 | SuperNull | gotta keep up with firmwares for sure. |
| 16:02.39 | SuperNull | it was REALLY bad on older firmwares |
| 16:02.49 | SuperNull | igcewieling you using auto-config ? |
| 16:03.33 | igcewieling | I'm pretty sure the person who sets up the adtrans does not use auto-config |
| 16:04.25 | igcewieling | my the time I get to them they are already set up. |
| 16:05.40 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:05.50 | SuperNull | gotcha. |
| 16:05.55 | SuperNull | auto-config is the way to go man. |
| 16:06.49 | SuperNull | no way to 'fuck it up' if its only the ip info and a remote config.. we used to have issues with other techs and my self included fubaring the adtran to a non working state ... recovery is a billion times easier. |
| 16:07.16 | SuperNull | also if it fails its just a matter of renaming the file to the mac of the adtran .. (we pre-provisiong all our adtrans for dhcp and auto config with the mac address) |
| 16:08.51 | igcewieling | none of this stuff happened during install. We pushed out an update to the ntp server list and the access lists to the routers. the access-list did not allow access to one of the ntp servers. next time the router booted it crshes |
| 16:09.17 | igcewieling | how do you do DHCP over T-1? |
| 16:09.33 | SuperNull | ppp remote ip assignment... |
| 16:09.38 | SuperNull | but that is different ;) |
| 16:09.57 | SuperNull | ideally .. you configure just the required stuff to get it online.. we never hook adtrans direct to T1 unless its for PRI hand off. |
| 16:10.26 | SuperNull | igcewieling you ever use dialup ;) |
| 16:10.34 | SuperNull | dialup uses ppp remote ip assign. |
| 16:10.48 | SuperNull | i also run a dialup network .. of olden times. |
| 16:12.43 | jeffspeff | how would i use an option like this $ curl -H "Accept: text/pbx" https://api. in func_curl ? |
| 16:13.20 | *** join/#asterisk Guest76569 (~Freeaqing@91.214.168.110) |
| 16:22.42 | jeffspeff | let me rephrase... how can i send header information when using curl? i found where i can receive header info, but i need to send. |
| 16:25.02 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:25.12 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:25.52 | ChannelZ-Wk | Sheesh. Sometimes Vitelity support is awesome, and then sometimes I get the guy who doesn't seem to know how to read. |
| 16:27.10 | *** join/#asterisk vlad_starkov (~vlad_star@89.175.55.54) |
| 16:28.48 | navaismo | jeffspeff, maybe you need to rephrase like: "this shit of curl didn't work at all I have tried to send info but it never work, just sucks" and maybe you get more replies |
| 16:31.26 | kaldemar | or just the opposite. CURLOPT does not seem to offer a way to add headers to a request. |
| 16:37.31 | *** join/#asterisk vlad_starkov (~vlad_star@89.175.55.54) |
| 16:38.54 | *** join/#asterisk ocholetras (~Alberto@229.Red-81-37-121.dynamicIP.rima-tde.net) |
| 16:39.01 | ocholetras | hi all! |
| 16:41.29 | ocholetras | anyone to talk about right ways of identify our channels/userdevices |
| 16:41.44 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 16:43.18 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 16:44.05 | navaismo | ~ask |
| 16:44.06 | 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. |
| 16:44.53 | ocholetras | ok |
| 16:45.14 | [TK]D-Fender | ocholetras: Identify them when? Where? For what purpose? |
| 16:45.21 | ocholetras | On sip.conf |
| 16:46.17 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 16:46.26 | [TK]D-Fender | ocholetras: Please rephrase your request and be complete about what you're looking to do |
| 16:46.31 | ocholetras | A book says that a channel should be a channel, without context. Then you use that channel to send calls for charly and next week for john. It appears that using ext or username on channel definition at sip.conf its somthin like anti-pattern way |
| 16:46.48 | gmalsack | hey all - which do you think is heavier on the system.... AGI() or SYSTEM() |
| 16:47.02 | *** join/#asterisk vlad_starkov (~vlad_star@89.175.55.54) |
| 16:47.37 | gmalsack | [TK]D-Fender: Well good day sir... everything going well? |
| 16:48.00 | [TK]D-Fender | ocholetras: that is a DEVICE name, not a CHANNEL name |
| 16:48.02 | kaldemar | ocholetras: they are not channel definitions. they are devices or peers. |
| 16:48.09 | [TK]D-Fender | ocholetras: [fred] <- device |
| 16:48.14 | kaldemar | ocholetras: a channel is created when there is a call. |
| 16:48.23 | [TK]D-Fender | ocholetras: Dial(SIP/fred) |
| 16:48.44 | ocholetras | understood |
| 16:52.40 | ocholetras | Im readign "Asterisk the definitive guide fourht edition" |
| 16:52.45 | ocholetras | And they said: "While we haven’t discussed Asterisk dialplans yet, it is useful to be able to visualize the |
| 16:52.45 | ocholetras | relationship between the channel configuration files (sip.conf, iax.conf) and the dialplan |
| 16:52.45 | ocholetras | (extensions.conf)." |
| 16:52.52 | ocholetras | So tat is wrong concept, right? |
| 16:52.56 | ocholetras | that* |
| 16:53.36 | ocholetras | They call "channel configuration files" to sip.conf and iax.conf where actually you configure the devices.. |
| 16:53.40 | ocholetras | ><' |
| 16:53.41 | ChannelZ-Wk | well SIP is a channel driver |
| 16:54.03 | ocholetras | Could mean that... |
| 16:54.08 | ChannelZ-Wk | so to call it a "channel configuration file" is correct if not a bit confusing taken literally |
| 16:54.11 | [TK]D-Fender | configure how the channels work... by the type of channel.. |
| 16:54.27 | ChannelZ-Wk | But entries in sip.conf aren't themselves channels |
| 16:54.30 | gmalsack | yea, the terminology can be confusing a first. |
| 16:54.53 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 16:54.58 | [TK]D-Fender | ocholetras: you are throwing too many words around at a time for your own good. |
| 16:55.02 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 16:56.24 | ocholetras | [TK]D-Fender: sorry 4 that. Ill try to write slower.. :P |
| 16:57.23 | [TK]D-Fender | ocholetras: Break things down into smaller questions. |
| 16:57.45 | gmalsack | if you remember the old asterisk gui, it used user.conf for the devices and left the channel configuration files for only configuring the channels. not sure why they got away from that, I thought it was a good way to created a defined separation between the two. made it easier for newbies to comprehend... |
| 16:58.06 | gmalsack | * my worthless 2 cents... |
| 16:58.19 | file | it was not done particularly well |
| 16:58.22 | WIMPy | But only for a few channeltypes. |
| 16:58.31 | file | and it was invasive |
| 16:58.36 | igcewieling | I do not remember the old Asterisk GUI. Maybe the people on #asterisk-gui remember the old Asterisk GUI |
| 16:58.49 | gmalsack | lol |
| 16:58.56 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:58.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:59.15 | gmalsack | yea it has some problems. but I thought it was a good effort.... |
| 16:59.26 | [TK]D-Fender | igcewieling: Stop talking about them ... as though they were PLURAL :p |
| 16:59.58 | gmalsack | [TK]D-Fender: lol |
| 17:00.03 | [TK]D-Fender | igcewieling: there are 4 users in that channel. 1 Bot, 1 Op, 1 idler ... and ME |
| 17:00.15 | ocholetras | Another quote, to explain why i am so confused: "In Asterisk, all the system cares about is the channel name. There is really no concept of a user at all2 , and extensions are nothing more than triggers which initiate a sequence of instructions." |
| 17:00.16 | gmalsack | lmao |
| 17:00.17 | [TK]D-Fender | and I beat the Op in... |
| 17:01.10 | gmalsack | when I was at digium, I asked them why that project was abandoned. they said the guy (singular) working on it left... |
| 17:01.12 | gmalsack | lo |
| 17:01.14 | [TK]D-Fender | ocholetras: forget that sentence. It is pretty broken... |
| 17:01.15 | gmalsack | l |
| 17:01.49 | ocholetras | omg |
| 17:02.00 | ocholetras | [TK]D-Fender: so i wasted my money with the book |
| 17:02.15 | [TK]D-Fender | ocholetras: No, I said one SENTENCE was poorly worded |
| 17:02.28 | file | yes, Pari was working on it |
| 17:02.29 | [TK]D-Fender | ocholetras: Stop jumping to conclusions that the whole book is bad |
| 17:02.31 | file | (the GUI) |
| 17:02.32 | ocholetras | All the configuration examples and explanations goes around that concept of channel |
| 17:02.45 | Qwell | file: among many others |
| 17:03.17 | gmalsack | so it was developed by a team? or Pari and the community? |
| 17:04.06 | file | there wasn't a "team" really, but it's too long ago for me to remember |
| 17:04.35 | [TK]D-Fender | ocholetras: Just ask us for clarification on the bits you are unclear with. |
| 17:06.13 | ocholetras | ok |
| 17:07.43 | *** join/#asterisk tyman (~tyman@75-149-49-133-SFBA.hfc.comcastbusiness.net) |
| 17:10.05 | file | time flies when you are manipulating the space time continuum. |
| 17:32.02 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 17:32.19 | danfromuk | Hi. Is there any way to boost the volume of a SIP Channel? |
| 17:35.47 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_VOLUME |
| 17:36.43 | danfromuk | Thansk |
| 17:36.44 | danfromuk | Thanks |
| 17:37.46 | danfromuk | Is the default value 0? |
| 17:38.52 | jeffspeff | when executing a system command from dialplan that uses a channel variable, do i have to escape around the variable or something? |
| 17:38.54 | file | the default is that the dialplan function is not used, so the audio is untouched |
| 17:39.03 | UnixDev | I'm having an issue on asterisk AMI interface, 1.8.x branch, latest stable.. it seems not all ExtensionStatus events are being sent… what could be causing this? |
| 17:40.53 | danfromuk | file: when I say default value, does Set(VOLUME(TX)=3) boost the volume by 3? and therefore a subsequent Set(VOLUME(TX)=0) would put it back to the untouched state? |
| 17:41.41 | file | yes. |
| 17:42.00 | file | it may still force it through transcoding, though |
| 17:42.04 | danfromuk | Perfect, thanks. |
| 17:42.38 | file | it's not smart enough to know when to remove itself |
| 17:44.12 | [TK]D-Fender | it has to for transcoding... |
| 17:44.16 | [TK]D-Fender | force* |
| 17:44.29 | file | if you set both gains to 0 it doesn't have to |
| 17:45.56 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 17:45.58 | [TK]D-Fender | I suppose if it's smart enough to see the starting point |
| 17:49.13 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 18:08.05 | *** join/#asterisk felipealmeida (~user@139.82.86.17) |
| 18:22.43 | igcewieling | the proper place to increase the gain is on the endpoint. |
| 18:25.00 | jeffspeff | why doesn't it work when i do [context] exten=XXXX,1,DOwhatever exten=1234,n,DowhateverNext |
| 18:26.23 | navaismo | ~book |
| 18:26.23 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:27.01 | igcewieling | jeffspeff: because without the _ asterisk is looking for literally 4 Xs as the diaed number. |
| 18:27.08 | igcewieling | now go read the book |
| 18:27.23 | [TK]D-Fender | jeffspeff: first... you don't jsut use "n" on that second pattern and expect it to be relative to the exten above |
| 18:27.50 | jeffspeff | igcewieling, thanks, i feel like it's a monday |
| 18:28.01 | igcewieling | also god kills a kitten every time you mix patterns and non-patterns like that. I know freepbx does it that way. I feel it is a bad idea. |
| 18:30.00 | navaismo | you hate freepbx haha |
| 18:31.23 | jeffspeff | i've got a curl command setting the callerid(name) now, i want to set that on all the inbound DID's. my idea was was to exten=_XXXXXXXXXX,1,curlcommand exten=1231231234,n,Goto(somecontext,123,1) |
| 18:31.53 | igcewieling | navaismo: I used to, but not any more. there are things I feel are wrong with FreePBX. The overly complex dialplan is one of them, mixing patterns and literals are another |
| 18:33.01 | igcewieling | why not [somecontext] exten => _XXX,1,Gosub(calleridlookup,s,1) |
| 18:33.08 | igcewieling | some some such thing. |
| 18:34.25 | jeffspeff | just learned something new. never used gosub before |
| 18:35.13 | igcewieling | We handle 10,000 TNs and not a single one of them is a literal |
| 18:35.18 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 18:36.06 | igcewieling | jeffspeff: if you had read the book you'd know about gosub |
| 18:36.36 | jeffspeff | i've read many parts of the book, but it's not the type of thing you read from cover to cover |
| 18:39.22 | jeffspeff | igcewieling, if my curl command to set the cid name is only 1 line, what's the purpose of using gosub? |
| 18:40.41 | igcewieling | jeffspeff: little reason. however, most people do not just run 2 priorities for an extension, using gosub to handle stuff done of every extension is a good idea. |
| 18:41.03 | *** join/#asterisk Flametail (18ef4758@gateway/web/freenode/ip.24.239.71.88) |
| 18:43.01 | Flametail | Hello. I am wanting to use some old hardware to turn my computers into softphones with my landline. However I'd like to do this with existing hardware only..nothing new. Can Asterisk allow me to basically use a server with a modem to ring a softphone on my laptop, and my laptop to call out the landline? Without any extensions or voicemail service? |
| 18:43.37 | Flametail | If I don't answer just let the telco voicemail pick up, etc. Not trying to make a PBX, just turn my computers into more handsets. |
| 18:44.44 | igcewieling | Flametail: no. |
| 18:45.24 | Flametail | Any ideas what could? |
| 18:45.43 | igcewieling | Flametail: purchase a telephony card which is compatible with Asterisk |
| 18:47.23 | igcewieling | " Not trying to make a PBX, just turn my computers into more handsets." <-- that is making a PBX |
| 18:48.46 | Flametail | PBX as far as I know involves all the extensions and internal calling. I just want to be able to answer the phone with my laptop and make landline calls from laptop. Both using the modem of the remote server. |
| 18:50.11 | [TK]D-Fender | Flametail: Much much extra stuff you can do doesn't change that it is a PBX. Just the number of different things it does to process your call |
| 18:50.32 | [TK]D-Fender | How much* |
| 18:51.46 | igcewieling | Flametail: you seem to think Asterisk is compatible with modems. It isn;t. No matter how many different ways you ask the question the answer is still the same. If you want to interface with an analog telephone line, you need to purchase a compatible telephony interface card |
| 18:52.18 | navaismo | or a gateway |
| 18:52.53 | navaismo | or old modem with tiger chipset to be recognized as x100p |
| 18:53.09 | igcewieling | navaismo: you are dead to me. |
| 18:53.30 | Flametail | Ah. I'm not really from the modem era so to me, if I could call out to someone with the windows tools on a Win98 box, a modem can make calls :) I've got a U.S Robotics 56k. Though I just noticed it's a fax card so it may be useless after all.... |
| 18:53.33 | igcewieling | don't send the poor guy on a fruitless hunt for a device which has not been manufatured in 15 years. |
| 18:55.12 | navaismo | igcewieling, :'( this is a bad week so far, yesterday we lost "dos a cero", today i cant fix a css issue and now you just leave me alone :'( :'( |
| 18:56.34 | Flametail | Well, assuming the card can still be used ( I hadn't checked which card it was before, recently accuired this old thing), are there software alternatives to these devices? Something that can use the device and perhaps create a virtual telephony card for asterisk? |
| 18:57.03 | [TK]D-Fender | Flametail: No, it is a dead-end waste of time |
| 18:57.17 | [TK]D-Fender | Flametail: $30 could get you a device you could use for this |
| 18:58.35 | navaismo | like the x100p on ebay? |
| 18:58.58 | igcewieling | Flametail: Step 1: throw out the modem. |
| 18:59.59 | [TK]D-Fender | navaismo: No, SPA-3000 / SPA-3102 used |
| 19:00.01 | igcewieling | I am confident there are no x100p's on ebay. I admit there may be people on ebay lying and saying they have an x100p for sale, but we call those people "scammers" |
| 19:01.28 | coppice | so many people have asked about this over the last 13 years, but nobody ever wants to sit down and write a driver to completely repurpose a winmodem as a telephony card. if its a modem card with DSP on board there is nothing you can do, but a winmodem could be made to work with some effort, just as the x100p has |
| 19:02.12 | [TK]D-Fender | cause that effort is worth mor than aquiring a piece of hardware known to work as-is :) |
| 19:03.18 | igcewieling | anyone qualified to write such a driver is rich enough to buy an ATA |
| 19:03.22 | coppice | well, it used to be a more interesting thing to do, like using the modem in every notebook to form a portable demo system. however, so much time has passed notebooks no longer have modems |
| 19:07.28 | Flametail | I'll be hooking it up to a phone line later and trying to use Dialer(to check if the modem still works), If that works the card is capable of phone calls, but I guess still not compatable with Asterisk. Ah well. If dialer works I'll have one computer phone :D Though I guess I should have tested it before bother y'all. Silly me... |
| 19:08.57 | [TK]D-Fender | doesn't dialer only actually DIAL a number and you still ahve to use a parallel-wired phone handset on the other jack to speak? |
| 19:10.10 | Flametail | Not sure. My only experience with dialer is when I had an old Win98 box. That used an old mic I plugged into the soundcard. |
| 19:10.14 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-mzreabwufonzugza) |
| 19:11.32 | [TK]D-Fender | Flametail: have fun... |
| 19:12.04 | Flametail | Yeah. It's all a goof off, see if I can experiment anyway. |
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| 19:58.06 | jeffspeff | just wanted to share this with everybody https://www.opencnam.com/ has a "hobbyist" free plan that gives 60 lookups an hour to their cache. just from fiddling around with it, i've found that if a number isn't in their cache, it will be added in about 2 minutes. |
| 19:58.30 | jeffspeff | pretty simple API too |
| 20:00.45 | igcewieling | jeffspeff: let us know how it works out for you in the long run. |
| 20:00.53 | igcewieling | Like I said, we use BulkCNAM.com |
| 20:02.29 | jeffspeff | opencnam is cheaper per lookup |
| 20:03.58 | igcewieling | you apparently don'y buy in bulk |
| 20:04.34 | jeffspeff | their site say .005 for carriers which i assume would be bulk |
| 20:04.35 | igcewieling | I have nothing against opencnam, they are cool guys, but they are not the only ones out there. |
| 20:04.52 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 20:04.59 | jeffspeff | just my first experience with any of them |
| 20:05.29 | jeffspeff | i'm just looking at the bulkcnam site and comparing with what i read on the other sites. |
| 20:15.11 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:17.04 | *** join/#asterisk [Outcast] (~anonymous@50-200-130-22-static.hfc.comcastbusiness.net) |
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| 20:17.45 | [Outcast] | anyone here working with pinefrog-1.8? |
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| 20:47.49 | *** join/#asterisk magespawn (~quassel@41-133-171-128.dsl.mweb.co.za) |
| 20:48.22 | magespawn | good evening |
| 20:49.15 | navaismo | o/ |
| 20:53.41 | magespawn | i am new to asterisk, does anyone know if the Asterisk: The Definitive Guide is still available? |
| 20:54.13 | zerohalo | it is - it is in the 4th edition now |
| 20:54.31 | igcewieling | ~book |
| 20:54.31 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:55.32 | magespawn | the link on the wiki is abit out of date then |
| 20:55.32 | magespawn | ty |
| 20:55.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:57.14 | magespawn | i have just started a new job and inhereted a Asterisk system, so need to bring myself up to speed as quickly as i can |
| 20:57.43 | igcewieling | voip-info is the last place you should look for information, not the first. |
| 21:00.36 | magespawn | voip-info? |
| 21:00.36 | navaismo | mix-up, voip-info has more examples against asterisk wiki |
| 21:10.18 | *** join/#asterisk dongola7 (~dongola7@unaffiliated/blair/x-0911782) |
| 21:11.43 | magespawn | thanks for the info |
| 21:12.58 | igcewieling | which of the dozen or so asterisk wikis are you referring to? |
| 21:13.50 | navaismo | wiki.asterisk.org |
| 21:14.01 | magespawn | might not be the wiki let me get the a link |
| 21:15.14 | magespawn | it is here http://www.asterisk.org/community the link under the learn more heading |
| 21:15.25 | magespawn | so not the wiki |
| 21:16.01 | magespawn | http://ebookbrowse.com/oreilly-asterisk-the-definitive-guide-3rd-edition-apr-2011-pdf-d280447018 this specific link |
| 21:18.16 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 21:19.19 | magespawn | thanks again good night all |
| 21:20.30 | navaismo | evening* |
| 21:25.05 | ChannelZ-Wk | http://burner.com/asterisk-primer if you want to give it a spin |
| 21:30.35 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 21:36.46 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 22:07.28 | navaismo | someone add that to the bot please |
| 22:07.55 | navaismo | every 'newbie' will be slapped with that link.. ~shutup&check |
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| 23:28.50 | ChannelZ-Wk | infobot: primer is http://burner.com/asterisk-primer |
| 23:28.50 | infobot | ChannelZ-Wk: okay |
| 23:32.16 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 23:34.14 | navaismo | 1599 conversation sent to the trash. maybe its time to unsubscribe from asterisk mailing list :S |
| 23:41.05 | WIMPy | "digital PRIs like T1/E1 and ISDN" doesn't make sense. |
| 23:42.03 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 23:42.04 | Penguin | Because all PRIs are digital? |
| 23:42.21 | WIMPy | As well as ISDN. |
| 23:44.04 | Penguin | Would you prefer "digital circuits like T1/E1 and ISDN"? |
| 23:44.17 | ChannelZ-Wk | I'll just replace it with "thinymajigs" |
| 23:44.44 | WIMPy | And "hardware based techonogies [...] trhough a layer called DAHDI". That's only one example. There are many other channeltypes for telephony cards. |
| 23:44.54 | Penguin | Ya can't just tell us it's wrong... You have to tell us what makes it right. |
| 23:45.32 | WIMPy | Yes, that makes more sense. Maybe eben ISDN interfaces like PRI (T1/E1) and BRI? |
| 23:45.40 | ChannelZ-Wk | I guess you didn't read the introduction about it not being exhaustive. There are 100 additional things I could say about almost everything. |
| 23:46.16 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 23:46.18 | WIMPy | Yes, but it doesn't say it's just an example of many others. |
| 23:46.22 | Penguin | I think I like that phrase better. |
| 23:46.29 | Penguin | phrasing |
| 23:47.12 | hjf | ok i know this isn't 1999 but |
| 23:47.21 | hjf | can i use a motorola SM56 modem for FXO? |
| 23:47.30 | Penguin | As long as you intend to call a T1/E1 an ISDN interface. |
| 23:48.05 | WIMPy | hjf: no |
| 23:48.10 | hjf | :( |
| 23:48.32 | WIMPy | Well, a PRI is E1/T1/J1, but a E1/T1/J1 need not be a PRI. |
| 23:53.59 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:54.03 | WIMPy | So maybe just leave out the "(T1/E1)" bit. |
| 23:58.03 | ChannelZ-Wk | Going home. Will deal with it later |