| 00:08.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:08.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:13.23 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 00:16.24 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 00:22.58 | *** join/#asterisk ccrnp (~ccr@208-90-186-2.warp2biz.com) |
| 00:23.54 | ccrnp | hi Folks ! I am having issue with asterisk 1.4.42 where there are bunch of "channel.c: Exceptionally long voice queue length queuing to IAX2" WARNINGS |
| 00:24.19 | ccrnp | any suggestion and direction would be really helpful |
| 00:25.20 | s14ck | ccrnp: please make nopaste or smthg |
| 00:26.14 | s14ck | ccrnp: Im going reboot my system brb |
| 00:27.15 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
| 00:27.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:34.43 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 00:35.00 | *** join/#asterisk bettor (~cyco@oh-76-5-245-224.dhcp.embarqhsd.net) |
| 00:39.35 | ccrnp | no one in this room experienced "channel.c: Exceptionally long voice queue length queuing to IAX2" problem before ? |
| 00:39.46 | navaismo | nope |
| 00:44.36 | *** join/#asterisk s14ck (~s14ck@190.203.177.134) |
| 00:44.49 | s14ck | ccrnp: hey |
| 00:44.59 | s14ck | ccrnp: im back |
| 00:45.04 | ccrnp | welcome back |
| 00:45.09 | ccrnp | so any suggestions |
| 00:45.25 | s14ck | sorry I dont see your log |
| 00:45.31 | ccrnp | http://nopaste.info/689cf2a970.html |
| 00:45.36 | igcewieling | ccrnp: what version of asterisk? |
| 00:45.52 | ccrnp | its 1.4.42 |
| 00:46.46 | igcewieling | in my experience that is either network congestion of the server is not fast enough to keep up with the incoming audio |
| 00:47.00 | igcewieling | However, it has been a long time since I was crazy enough to use IAX2 |
| 00:47.03 | s14ck | ccrnp: line number? (I dont wanna read all file) |
| 00:47.27 | ccrnp | 7 ,12 and 16 |
| 00:47.36 | s14ck | ccrnp: thnks |
| 00:48.32 | s14ck | ccrnp: there is not sense |
| 00:50.38 | ccrnp | ? |
| 00:52.13 | s14ck | ccrnp: please, get log again but with debug mode active |
| 00:52.31 | s14ck | core set debug 9 |
| 00:53.14 | ccrnp | sure |
| 00:53.45 | s14ck | (Oh God, Im losing my powers at the cli) |
| 00:54.50 | s14ck | ccrnp: aditionally pay attention in your system performance (cpu, mem, blah blah) |
| 00:55.29 | s14ck | ccrnp: obviously in middle of the test |
| 01:01.14 | Smak | Is there a difference between SIP trunking and SIP termination? |
| 01:02.26 | s14ck | Smak: Oh, yes. |
| 01:03.08 | Smak | I am wanting to play around with asterisk and I am trying to find a termination service. |
| 01:03.54 | Smak | so that I can make outbound calls, I thought they were the same thing, but maybe I don't know know what I am looking for. |
| 01:06.42 | s14ck | Smak: depends than you need to get. |
| 01:08.17 | Smak | That looks like english, kinda |
| 01:08.43 | s14ck | Smak: ^^ |
| 01:10.34 | s14ck | ccrnp: are you alive? |
| 01:11.26 | *** join/#asterisk jpsharp (~jsharp@67.238.9.52) |
| 01:15.42 | ccrnp | sure |
| 01:20.47 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 01:40.00 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 01:43.25 | SeRi | guys I am trying to craete a statement that when 1102 is dialed it sends it to another context so I am using GotoIf but I am not sure how to catch the number whe passing though the context... Any Ideas? |
| 01:46.24 | igcewieling | exten => 1102,1,Goto(context,extension,priority) |
| 01:46.35 | carrar | Like: Goto(some-context-here,${EXTEN},1) |
| 01:47.35 | carrar | Then match the extension in the new context |
| 01:48.28 | carrar | or move the extension to a variable |
| 01:51.25 | SeRi | Thanks for the ideas guys.... |
| 01:55.48 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 02:00.20 | SeRi | guys I know this is not going to work but here the idea. |
| 02:00.24 | SeRi | http://pastebin.com/ncuVZ4ni |
| 02:00.40 | SeRi | I am trying to catch 1102 from 11XX |
| 02:00.54 | SeRi | and send that number to gotcha |
| 02:08.25 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-217-48.midsouth.biz.rr.com) |
| 02:08.50 | SeRi | any ideas? |
| 02:11.53 | WIMPy | Maybe I don't get the question, but why don;t you just write a exten => 1102? |
| 02:13.17 | igcewieling | SeRi: if you have exten => 1102 and exten => _11XX then 1102 will match when dialing 1102 because it is a more specific match |
| 02:13.20 | SeRi | WIMPY let me pastebin the whole thing and it will make better sense |
| 02:13.56 | SeRi | igcewieling: The problem is that on the range 11XX we have over 20 exten so we use 11XX |
| 02:14.19 | WIMPy | And how many of them are special? |
| 02:16.22 | SeRi | just 1102 |
| 02:16.25 | SeRi | http://pastebin.com/H7tTSReN |
| 02:16.29 | SeRi | WIMPy: ^^ |
| 02:16.45 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
| 02:16.48 | WIMPy | Then I guess my answer wil stay the same. |
| 02:17.22 | SeRi | Ok I see what you mean just put 1102 underneath 11XX? |
| 02:17.48 | WIMPy | Order doesn't matter. |
| 02:18.51 | WIMPy | You don't want the DB check on that extension, either? |
| 02:18.57 | SeRi | nope |
| 02:19.08 | SeRi | just skip it all and go to pr |
| 02:19.36 | WIMPy | Ok, because you can do more specific extensions wfor only certain priorities. |
| 02:19.51 | WIMPy | Ugly to read, but otherwise very efficient. |
| 02:20.18 | SeRi | so... |
| 02:20.32 | SeRi | let me see |
| 02:27.32 | SeRi | ok got it |
| 02:27.58 | SeRi | [$EXTEN]=1102 :) |
| 02:28.04 | SeRi | Thanks guys!!!!!! |
| 02:28.16 | WIMPy | What? |
| 02:29.10 | SeRi | exten => _11XX,1,NoOp() |
| 02:29.23 | SeRi | same => n,GotoIf($[${EXTEN}=1102]?pr) |
| 02:29.50 | WIMPy | So you're doing it within the pattern anyway. |
| 02:29.56 | SeRi | :P |
| 02:29.57 | SeRi | hahaha |
| 02:30.01 | SeRi | hard headed I guess |
| 02:31.57 | WIMPy | That's the shortest I can come up with. Let's see if you can figure that out :-) http://wimpy.yeti.dk/pastebin |
| 02:39.55 | WIMPy | Ooops. Didn't save. |
| 02:43.37 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:56.31 | SeRi | lol |
| 02:56.35 | SeRi | can you repaste? |
| 02:57.16 | SeRi | ah |
| 02:57.17 | SeRi | I see |
| 02:57.22 | SeRi | 1102 going back to 3 |
| 02:59.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:00.35 | SeRi | nope |
| 03:00.42 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 03:00.42 | *** mode/#asterisk [+o file] by ChanServ |
| 03:02.07 | SeRi | is the priority :) |
| 03:08.23 | ChannelZ | eh? |
| 03:12.30 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-162-251.pools.arcor-ip.net) |
| 03:13.44 | SeRi | lol. |
| 03:15.27 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
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| 03:16.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:17.40 | ChannelZ | goes back to scratching himself |
| 03:24.22 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.248.245) |
| 03:29.48 | Smak | I am wanting to play around with asterisk and I am trying to find a cheapish termination service. Any recommendations? |
| 03:30.20 | jpsharp | You'll get as many answers as there are people logged in here. |
| 03:33.05 | Smak | I wish. I am rather confused just using google, Not sure on exactly what I am getting into. I have found something that sounds like what I am looking for, for $0.002/minute, and yet others are talking $0.2/per minute. Some include voice mail and lots of other features. I was to understand all I really needed was termination. Kind of confusing. |
| 03:36.36 | *** join/#asterisk mintos (~mvaliyav@112.79.41.207) |
| 03:37.12 | jpsharp | If all you're doing is sending calls out, all you need is termination. For very low volume, you'll be averaging about 2 cents USD per minute for calls made to domestic US number. |
| 03:37.56 | Smak | So I am not looking for sip trunking? |
| 03:38.31 | jpsharp | Well, some people call their services sip trunking. You want SIP or IAX services if you want to play with asterisk. |
| 03:38.40 | jpsharp | Which service are you looking at? |
| 03:38.56 | jpsharp | Personally, I've used teliax.com and gafachi.com for my termination needs. |
| 03:39.04 | Smak | google and http://www.voip-info.org/wiki/view/SIP/IAX+Services+for+Asterisk |
| 03:39.46 | Smak | I am just looking to get started and play around with asterisk, not trying to set any kind of business anything. |
| 03:40.45 | jpsharp | Then I'd recommend either one I've used. They're pretty asterisk friendly. |
| 03:41.05 | Smak | I didn't see a US rate for https://www.teliax.com/ |
| 03:41.05 | jpsharp | And their rates are reasonable without needing to do a big prepayment. |
| 03:42.40 | jpsharp | 2-3 cents/minute depending on destination. |
| 03:42.46 | jpsharp | https://www.teliax.com/plans/4 |
| 03:43.13 | fling | Hello! |
| 03:43.18 | Smak | Greetins |
| 03:44.01 | Smak | Wholesale VoIP Termination to the United States at blended rates as low as $0.0066 I am guessing wholesale mean more minutes then I would use just playing around. |
| 03:44.23 | *** join/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 03:44.27 | jpsharp | You wont touch wholesale rates until you're passing at least 100K minutes/month. |
| 03:44.28 | Smak | That is: www.gafachi.com |
| 03:44.28 | *** part/#asterisk CRCinAU (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 03:44.33 | *** join/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 03:44.39 | fling | How to match every number from "+7 901 4500000" to "+7 901 4529999" ? |
| 03:44.46 | CRCinAU_ | Has anyone noticed asterisk 11.2.0 crashing on a RecieveFax? |
| 03:44.53 | CRCinAU_ | using the spandsp |
| 03:45.25 | CRCinAU_ | and wtf: |
| 03:45.28 | CRCinAU_ | 14:44 < jpsharp:#asterisk> You wont touch wholesale rates until you're passing at least 100K minutes/month. |
| 03:45.31 | CRCinAU_ | 14:44 < Smak:#asterisk> That is: www.gafachi.com |
| 03:46.08 | jpsharp | WTF wtf? |
| 03:46.11 | Smak | I just keep thinking 2 - 3 cents a min can really add up. I am only playing around at this point but. |
| 03:46.44 | fling | ChannelZ: hello |
| 03:46.51 | Smak | Greetings |
| 03:46.56 | CRCinAU_ | they came through to me as a server message o_O |
| 03:47.05 | fling | Smak: :p |
| 03:47.09 | CRCinAU_ | the only output I get is: |
| 03:47.10 | CRCinAU_ | <PROTECTED> |
| 03:47.10 | CRCinAU_ | <PROTECTED> |
| 03:47.13 | CRCinAU_ | <PROTECTED> |
| 03:47.16 | CRCinAU_ | asterisk*CLI> |
| 03:47.19 | CRCinAU_ | Disconnected from Asterisk server |
| 03:47.21 | CRCinAU_ | Attempting to reconnect for 30 seconds |
| 03:47.24 | CRCinAU_ | can't seem to find a core dump |
| 03:48.12 | CRCinAU_ | on second thoughts, I've found the core dump now :) |
| 03:52.08 | CRCinAU_ | also, I'm not getting any emails from JIRA :( |
| 03:56.37 | ChannelZ | sorry, hi fling |
| 03:56.51 | ChannelZ | ~pb |
| 03:56.51 | 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. |
| 03:57.01 | fling | Help me please with my dialplan http://dpaste.com/881509/ , it works but now I want to dial these numbers http://spravkaru.net/mobile/383/ via another sip peer, so I need to match these numbers with exten |
| 03:57.09 | fling | ChannelZ: :p |
| 03:57.14 | Smak | Why are you sorry, did you not hear what we was saying about you. |
| 03:57.25 | fling | ChannelZ: so I want to do it right, where to read about it? |
| 03:58.59 | ChannelZ | well you have two choices.. you can write lots of dialplan to properly match all the numbers you want vs the ones you don't, or I'd probably do it in some other language with an AGI.. |
| 03:59.13 | ChannelZ | possibly fetching all that crap from a database or something, if it's stuff that's going to change as well |
| 03:59.56 | ChannelZ | I don't speak russian so I have no idea what those columns say.. is it just the first 3 digits basically you're interested in? The 913, 923, etc |
| 04:01.35 | ChannelZ | CRCinAU_: This is FFA it looks like, do you have the right versions of the modules installed? (IE not some mismatched architecture or for the wrong version of Asterisk)? |
| 04:02.35 | CRCinAU_ | ffa? |
| 04:02.53 | Smak | free for all? |
| 04:03.07 | CRCinAU_ | btw: https://issues.asterisk.org/jira/browse/ASTERISK-20949 |
| 04:03.09 | CRCinAU_ | I haven't been able to find any workaround.... |
| 04:03.14 | CRCinAU_ | t38 on or off. |
| 04:03.21 | CRCinAU_ | either way, everything I seem to do causes a core dump :( |
| 04:03.33 | fling | ChannelZ: +7AAABBBBBBB, AAA is the first column, BBBBBBB is a number between second and third column |
| 04:04.09 | ChannelZ | Fax For Asterisk |
| 04:04.14 | CRCinAU_ | oh. |
| 04:04.23 | fling | ChannelZ: so the first line is all the numbers from "+7 901 4500000" to "+7 901 4529999" |
| 04:04.24 | ChannelZ | res_fax_digium or whatever it is |
| 04:04.26 | CRCinAU_ | no, its the stock standard res_fax_spandsp |
| 04:04.29 | ChannelZ | their licensed one |
| 04:04.44 | mjordan | CRCinAU_: A core dump isn't useful. You need to generate a backtrace |
| 04:04.48 | CRCinAU_ | I don't use it very often - but someone needs to send me a fax lol |
| 04:04.59 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 04:05.31 | CRCinAU_ | mjordan: thanks. I haven't had to submit a bug report since 1.2 ish - so how it all works these days is... well, nfi :) |
| 04:05.35 | mjordan | np :-) |
| 04:05.38 | mjordan | thanks for reporting it |
| 04:06.11 | CRCinAU_ | hell, for some reason, I still have my CRCinAU nick unable to talk in here after a heated disagreement years ago ;) |
| 04:06.33 | CRCinAU_ | and I mean, years ;) |
| 04:10.00 | ChannelZ | fling: well with so many ranges, it's kind of ugly regardless. But in general if it's something takes more than a few dialplan lines to do, I usually do it in an AGI so my dialplan doesn't get outrageous. The expression syntax in the dialplan gets ugly and it makes me crosseyed |
| 04:10.39 | fling | ChannelZ: ok, I'm reading about AGI |
| 04:11.00 | ChannelZ | it also assumes you hopefully know some other scripting language. |
| 04:11.20 | CRCinAU_ | hmmm |
| 04:11.23 | ChannelZ | But you could do it all in the dialplan. It's not "wrong", just harder to maintain in my humble opinion |
| 04:11.29 | CRCinAU_ | I wouldn't have thought about using AGI for outgoing stuff |
| 04:11.32 | CRCinAU_ | but I suppose it works. |
| 04:12.25 | ChannelZ | Well I'm saying mainly just for doing the logic he needs to do to figure out if the number being dialed falls within these 2 dozen different ranges or however many there are |
| 04:12.56 | ChannelZ | personal preference really |
| 04:13.05 | CRCinAU_ | true |
| 04:13.11 | CRCinAU_ | using perl + regex would work better ;) |
| 04:15.08 | CRCinAU_ | mjordan: so is a core dump not useful at all without the DONT_OPTIMIZE and BETTER_BACKTRACES options set? |
| 04:15.16 | mjordan | yes, unfortunately :-\ |
| 04:15.21 | mjordan | otherwise, there aren't any symbols |
| 04:15.22 | CRCinAU_ | bugger. |
| 04:15.38 | CRCinAU_ | ok, I'll have to try and rig up some way I can send faxes to asterisk. |
| 04:15.54 | CRCinAU_ | my testing was someone trying to send me something from a remote SIP point. |
| 04:16.06 | CRCinAU_ | so I'll see if I can use an ATA or something on a fax machine to duplicate |
| 04:19.46 | CRCinAU_ | almost built.... now just have to rebuild chan_sccp-b |
| 04:19.49 | fling | ChannelZ: what should AGI return? |
| 04:20.01 | CRCinAU_ | return? |
| 04:20.44 | fling | looks like I don't understand how it works |
| 04:20.47 | ChannelZ | well usually what I do is send an AGI command to set a variable which I can then act upon in the dialplan accordingly |
| 04:22.07 | fling | ChannelZ: so variable will be true if number match and false if dont? |
| 04:22.08 | ChannelZ | using the SET VARIABLE command.. or you can even use SET EXTENSION and SET PRIORITY to make it jump directly |
| 04:22.21 | ChannelZ | Yeah, or whatever method you come up with. |
| 04:22.23 | fling | ChannelZ: and the script will just check for number matching |
| 04:22.54 | fling | I want the fine example! :p |
| 04:23.16 | ChannelZ | Like I have a "call handler" script which decides what to do based on some records in a database, so I return an 'action' that makes more sense to me in the dialplan.. like 'ignore', or 'annoy', etc. |
| 04:28.25 | ChannelZ | I can send you an example in PHP which is what I usually script in |
| 04:30.28 | *** join/#asterisk sparrowjs (~Jon@c-76-25-87-128.hsd1.co.comcast.net) |
| 04:32.56 | CRCinAU_ | hmnmmm |
| 04:33.16 | CRCinAU_ | now it doesn't crash, but it seems the ReceiveFax exits without actually doing anything |
| 04:33.17 | CRCinAU_ | ie no file created |
| 04:34.19 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:35.39 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
| 04:36.33 | *** join/#asterisk voxter_ (~voxter@d23-16-70-150.bchsia.telus.net) |
| 04:39.36 | *** join/#asterisk sparrowjs (~Jon@c-76-25-87-128.hsd1.co.comcast.net) |
| 04:46.02 | CRCinAU_ | I'm not sure what else to debug on this :( |
| 04:46.59 | ChannelZ | EVERYTHING! |
| 04:47.28 | CRCinAU_ | debug all the thing? |
| 04:47.40 | CRCinAU_ | things even |
| 04:48.27 | ChannelZ | All the broken things anyway |
| 04:52.18 | *** join/#asterisk mintos (~mvaliyav@14.97.193.198) |
| 04:53.35 | fling | ChannelZ: I can simplify my dialplan a lot with this script, thanks :] |
| 04:55.00 | CRCinAU_ | for fucks sake. |
| 04:55.07 | CRCinAU_ | debug gives me nothing. |
| 04:55.11 | CRCinAU_ | verbose gives me nothing |
| 04:55.20 | CRCinAU_ | fax set debug on gives me nothing |
| 04:55.40 | CRCinAU_ | its almost like ReceiveFax is just going "#!/bin/bash\n exit 0" |
| 04:57.10 | ChannelZ | fling: sure good luck. Obviously assumes you even have PHP installed. And it would generally go in /var/lib/asterisk/agi-bin and needs chmod +x |
| 05:00.20 | fling | ChannelZ: I'm about to add a lot of checks in the script and it will return town name for a given number or nothing if no match |
| 05:03.32 | ChannelZ | well I'd test it first, I didn't. |
| 05:03.54 | fling | I'm not using it in production now, just thinking |
| 05:04.00 | fling | I will test it a lot |
| 05:20.05 | fling | ChannelZ: now I do not want using long lists, I want the app like geoip but for phone numbers :D |
| 05:20.15 | fling | ChannelZ: to make things even more simplier |
| 05:22.12 | apb1963_ | Greetings. I can't get my IVR welcome message to play anything other than silence. Log here: http://ix.io/41C |
| 05:26.01 | *** join/#asterisk voxter_ (~voxter@d23-16-70-150.bchsia.telus.net) |
| 05:36.08 | sparrowjs | Hey everyone |
| 05:44.45 | [TK]D-Fender | apb1963_: Asterisk isn't playing any prompts except "Goodbye" |
| 05:47.53 | apb1963_ | so you see the problem |
| 05:48.12 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 05:48.29 | [TK]D-Fender | apb1963_: Yes. It's a FreePBX one, not Asterisk |
| 05:48.43 | apb1963_ | ok |
| 05:49.12 | apb1963_ | :-) |
| 05:51.24 | igcewieling | Just a reminder everyone, backups are IMPORTANT. |
| 05:52.58 | [TK]D-Fender | Jesus Saves |
| 05:53.17 | [TK]D-Fender | The Devil keeps redundant off-site backups. |
| 05:54.11 | apb1963_ | Moses invests |
| 05:57.56 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 05:59.55 | igcewieling | Jesus saves at Wal-Mart. Buddha shops at Target |
| 06:01.21 | apb1963_ | Moses invests in real estate |
| 06:07.50 | ChannelZ | fling: that's up to you, put it in a database, etc. |
| 06:08.15 | fling | ChannelZ: I have several options! |
| 06:11.18 | fling | ChannelZ: are h.323 and sip complex and bad? |
| 06:21.35 | ChannelZ | No personal experience with h323 |
| 06:27.07 | *** join/#asterisk sevin (~sevin@c-50-132-127-195.hsd1.wa.comcast.net) |
| 06:58.20 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 07:03.01 | salz212 | If somebody can have a look at sip debugs, I am having an unusual problem with Asterisk 11 with one carrier. It advertises fax, image udptl etc in SDP to which asterisk reply and then call is hanged due to BYE received from carrier. |
| 07:03.16 | [TK]D-Fender | ~pb |
| 07:03.16 | 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. |
| 07:03.20 | [TK]D-Fender | ^ your friend |
| 07:04.01 | salz212 | http://pastebin.ca/2304138 |
| 07:06.37 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 07:07.31 | [TK]D-Fender | Cause No. 79 - service or option not implemented unspecified [Q.850] This cause is used to report a service or option not implemented event only when no other cause in the service or option not implemented class applies. |
| 07:09.15 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:09.52 | CRCinAU_ | hmmm |
| 07:10.03 | CRCinAU_ | I'm just sure ReceiveFax is screwed in 11.2 |
| 07:10.15 | CRCinAU_ | and 11.1.x |
| 07:10.31 | CRCinAU_ | it says receiving fax, then just dies with a non-zero result. |
| 07:10.39 | CRCinAU_ | no file written, no debug output |
| 07:11.33 | salz212 | exactly I have done my re-search and all the tweaking I could.. but its not working for me. |
| 07:11.52 | salz212 | more over I am not using fax I have even unload the modules .. |
| 07:12.21 | salz212 | is there any way to not send udptl in Answer()'s reply to carrier..? |
| 07:13.45 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:16.08 | CRCinAU_ | I lodged this: https://issues.asterisk.org/jira/browse/ASTERISK-20949 |
| 07:16.13 | CRCinAU_ | however, now its not core dumping :( |
| 07:16.23 | *** join/#asterisk bpietro (~bpietro@host137-13-static.226-95-b.business.telecomitalia.it) |
| 07:16.35 | CRCinAU_ | its just.... well... its... just exiting? |
| 07:16.41 | CRCinAU_ | but voice calls work fine |
| 07:18.24 | kaldemar | CRCinAU_: what "dies"? |
| 07:19.45 | kaldemar | CRCinAU_: also, asterisk will not dump core unless configured to do so with option -g for the asterisk binary or dumpcore=yes in asterisk.conf. |
| 07:21.28 | fling | Should I use IAX everywhere? |
| 07:21.41 | [TK]D-Fender | fling: Do you have a reason to? |
| 07:22.32 | fling | [TK]D-Fender: I want to connect two asterisk servers |
| 07:22.40 | CRCinAU_ | kaldemar: RecieveFax exits non-zero with no debug output, no verbose output, and no file written |
| 07:22.56 | salz212 | Fender: I am not using udptl fax etc neither I want it to be advertise by Asterisk 11 |
| 07:23.07 | [TK]D-Fender | fling: The generally an option with bosnus. If you need to care about bandwidth at least |
| 07:23.30 | fling | [TK]D-Fender: my sip friends are using ekiga, I'm waiting for iax2 support in ekiga because sip works bad with nat |
| 07:23.49 | [TK]D-Fender | fling: SIP generally works jsut fine |
| 07:24.07 | [TK]D-Fender | salz212: Then perhaps you should tell your peer not to support it |
| 07:24.31 | CRCinAU_ | its interesting that it core dumped when compiled normally, but doesn't core dump with DONT_OPTIMIZE and BETTER_BACKTRACES set |
| 07:25.01 | ChannelZ | IAX doesn't necessarily work better with NAT, depending on what your issue actually is, it just uses less ports to worry about. |
| 07:25.03 | CRCinAU_ | makes me very wtf. |
| 07:25.33 | ChannelZ | But if you have more than a few calls going between those two systems, it's beneficial to link them with IAX and use trunking |
| 07:25.33 | salz212 | I have it disabled in sip.conf. Secondly I do not have registeration on Asterisk it is on SIP Server. |
| 07:25.40 | kaldemar | CRCinAU_: precisely what do you mean by "exits non-zero"? show output. |
| 07:25.40 | fling | ChannelZ: also it is faster, should work better on gprs/3g connection |
| 07:26.10 | CRCinAU_ | kaldemar: https://issues.asterisk.org/jira/browse/ASTERISK-20949 |
| 07:26.14 | fling | s/on/over/ |
| 07:26.20 | *** join/#asterisk voxter_ (~voxter@d23-16-70-150.bchsia.telus.net) |
| 07:26.59 | fling | infobot: thanks :p you are awesome |
| 07:26.59 | infobot | fling: sure thing |
| 07:27.18 | ChannelZ | fondles infobot's ass |
| 07:27.53 | kaldemar | CRCinAU_: the "exited non-zero" is just a print that tells the fax,s extension has finished and dialplan is moving on. |
| 07:28.26 | CRCinAU_ | kaldemar: however, it doesn't move to the next step, it just goes to the h,1 entry |
| 07:28.33 | [TK]D-Fender | fling: IAX2 is not "faster" |
| 07:28.39 | CRCinAU_ | either way, ReceiveFax hasn't worked . |
| 07:28.50 | fling | [TK]D-Fender: but it is using less traffic! |
| 07:28.59 | CRCinAU_ | which is interesting as I've had it working for a long time... just not under 11.x |
| 07:29.36 | ChannelZ | fling: real time is real time. It can't be "faster" but it indeed does have less overhead as call volume increases |
| 07:29.41 | [TK]D-Fender | fling: If you are using trunk mode between 2 * servers and are running at least 2 calls, yes |
| 07:29.45 | fling | the problem is few sip clients are using crappy connection and slow atom cpus, and the best working codec is gsm somewhy |
| 07:30.14 | kaldemar | CRCinAU_: you don't have a next step in the extension. |
| 07:30.17 | [TK]D-Fender | fling: this won't be any savings for clients |
| 07:30.21 | fling | ChannelZ: [TK]D-Fender: I want to use iax2 between asterisk and clients too |
| 07:30.41 | ChannelZ | I want a million dollars |
| 07:30.44 | fling | And I probably will when ekiga will add iax2 support |
| 07:30.49 | fling | ChannelZ: me too |
| 07:31.00 | CRCinAU_ | kaldemar: in my debugging I do. |
| 07:31.05 | CRCinAU_ | however not in what is posted. |
| 07:31.10 | ChannelZ | Unfortunately IAX hasn't latched on much as a client protocol |
| 07:31.27 | CRCinAU_ | I have a NoOp afterwards which should print FAXOPT(status).... |
| 07:31.28 | fling | hmm hmm |
| 07:31.36 | ChannelZ | Some softphones is all. But it wasn't necessarily designed to be that anyway. |
| 07:31.56 | kaldemar | CRCinAU_: your issue does not even show asterisk crashing, you should add proof of your claims in the issue. |
| 07:32.10 | CRCinAU_ | the attached core dump not enough? |
| 07:32.36 | ChannelZ | speaking of core dumps... I could use one myself |
| 07:33.38 | [TK]D-Fender | checkout time, later all |
| 07:33.58 | kaldemar | fling: don't count on ekiga adding iax2 support. |
| 07:34.09 | fling | kaldemar: will not they add it? :p |
| 07:34.23 | kaldemar | CRCinAU_: you should actually show that asterisk does crash. |
| 07:34.32 | CRCinAU_ | kaldemar: the interesting part is why it doesn't core dump with DONT_OPTIMIZE and BETTER_BACKTRACES set |
| 07:34.46 | CRCinAU_ | I should recompile it without those and see if it core dumps agian |
| 07:35.13 | kaldemar | fling: well, considering that the note of IAX2 has been on the todo list for over 3 years and not many folks making phones are interested in IAX2, i would hold by breath waiting for it. |
| 07:35.30 | kaldemar | CRCinAU_: does it even crash? |
| 07:35.39 | fling | kaldemar: ok :] |
| 07:36.00 | CRCinAU_ | kaldemar: no - I just used /dev/urandom to get a core dump </sarcasm> :) |
| 07:36.02 | kaldemar | CRCinAU_: that will probably be the first question when someone reads your issue. |
| 07:36.35 | fling | another question: How to turn on online sip status? I want my clients to be able to set the status like Online/Away/DND, etc > http://wiki.ekiga.org/images/c/cb/250px-Ekiga_in_a_call.png |
| 07:36.40 | fling | it is not working now |
| 07:36.44 | CRCinAU_ | kaldemar: however, when I followed the guide to get a useful backtrace, part of it says to recompile asterisk with DONT_OPTIMIZE and BETTER_BACKTRACES set |
| 07:37.18 | CRCinAU_ | kaldemar: now that I set those options, it doesn't crash and core dump, but it fails to work. |
| 07:37.44 | CRCinAU_ | so. If I compile with those options UNSET, and it core dumps again, then we're on a whole new level of weird. |
| 07:40.55 | ChannelZ | FAX should be dead |
| 07:41.01 | CRCinAU_ | I agree. |
| 07:41.05 | CRCinAU_ | but sadly, it isn't. |
| 07:41.14 | CRCinAU_ | in fact, I use it at work multiple times a day |
| 07:41.15 | ChannelZ | Well it is for you. That's a start. |
| 07:42.33 | CRCinAU_ | ok - so now its built without DONT_OPTIMIZE and BETTER_BACKTRACES set |
| 07:42.41 | CRCinAU_ | if this thing core dumps.... o_O |
| 07:43.35 | CRCinAU_ | wtf. |
| 07:43.40 | CRCinAU_ | no core dump now, but: |
| 07:43.55 | CRCinAU_ | Status: FAILED, Error (if any): T38_NEG_ERROR |
| 07:47.40 | *** join/#asterisk brodz (~AndChat25@bb121-6-91-247.singnet.com.sg) |
| 07:48.12 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 07:48.25 | linocisco | I have Avaya BCM450 but which run out of VOIP phone licenses . Can we extend it with asterisk server and sip phones? with BCM450 as main PBX in place and another route or trunk to Asterisk to let asterisk's SIP clients to use BCM access |
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| 07:49.39 | CRCinAU_ | linocisco: buy more licenses. save yourself the pain. |
| 07:49.57 | ChannelZ | fling: those are probably SIP NOTIFY messages (if at all) but you'd have to figure out/find out what they are. |
| 07:50.31 | fling | ChannelZ: ok, thanks, I'm searching for it |
| 07:50.45 | linocisco | CRCinAU_, it will take time and I hate proprietary stuffs but according to HQ's decision, currently BCM is the standard and they will move to Cisco CUCM later. in the mean time, we have no licenses |
| 07:51.44 | linocisco | CRCinAU_, it will take time and I hate proprietary stuffs but according to HQ's decision, currently BCM is the standard and they will move to Cisco CUCM later. in the mean time, we have no licenses for VOIP phones. I am thinking to connect BCM with asterisk to give people voip phones access |
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| 07:53.57 | CRCinAU_ | linocisco: you poor man :( |
| 07:54.52 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:54.52 | kaldemar | linocisco: what protocols does the pbx speak? |
| 07:55.05 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:55.22 | linocisco | kaldemar, I dont know how to check |
| 07:56.01 | kaldemar | linocisco: really? how about asking nortel? |
| 07:56.22 | ChannelZ | Is secret! |
| 07:56.47 | linocisco | kaldemar, I am not sure. HQ guys came here and setup. everything is not clear and transparent |
| 07:56.49 | kaldemar | linocisco: first step is to find out what protocols can be used to communicate with it. |
| 07:57.05 | linocisco | kaldemar, can we trace it wireshark? |
| 07:57.26 | linocisco | kaldemar, what i am sure is that it has no sip licences |
| 07:57.29 | kaldemar | linocisco: otherwise you'll be asking your question for some weeks more without answers. |
| 07:57.45 | linocisco | kaldemar, that makes sense |
| 07:57.49 | ChannelZ | Some casual googling implies it can do SIP, but whether or not that's a separate licensed thing is unclear |
| 07:58.57 | kaldemar | linocisco: the pbx needs to have an interface that can be used with an asterisk box. you must find out what it has (and if they're usable) before going any further with this. |
| 08:01.10 | linocisco | yes. it has LAN interfaces |
| 08:01.35 | igcewieling | linocisco: A LAN interface only matters if the PBX supports SIP |
| 08:02.12 | ChannelZ | or something |
| 08:02.54 | ChannelZ | SPeaking of propriatary bullcrap, I have 3 3com 3102 phones here free to someone who can use them |
| 08:05.08 | linocisco | igcewieling, other interfaces are USB port which is connected to UPS |
| 08:05.42 | linocisco | ChannelZ, can't u flash it to make it only sip phones? |
| 08:06.56 | ChannelZ | from what I can tell no. What I'd been reading seemed to imply that they don't even hold their OS images, just a bootstrap, and it has to contact the server to get the rest of its brain (like permenant forced provisioning) |
| 08:07.10 | igcewieling | linocisco: T-1/E-1 CAS or T-1/E-1 PRI or Analog. Other than SIP those are your connectivity options. |
| 08:08.00 | kaldemar | and all the other protocols asterisk supports. |
| 08:13.51 | *** join/#asterisk PbxMan (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 08:14.16 | igcewieling | kaldemar: none of those are supported by most PBXs |
| 08:15.10 | kaldemar | igcewieling: some do support H.323. but it was just a note that those are not the only options. |
| 08:18.39 | CRCinAU_ | ahhh shit. |
| 08:18.43 | CRCinAU_ | who can close jira issues? ;) |
| 08:19.20 | kaldemar | CRCinAU_: #asterisk-bugs |
| 08:19.46 | wdoekes | CRCinAU_: which bug? |
| 08:19.58 | CRCinAU_ | https://issues.asterisk.org/jira/browse/ASTERISK-20949 |
| 08:20.08 | CRCinAU_ | I found a config problem on my end. |
| 08:20.20 | CRCinAU_ | however, I have NFI why it was core dumping earlier. :( |
| 08:20.26 | CRCinAU_ | but now I can't reproduce it |
| 08:21.12 | wdoekes | in any case.. attaching core dumps is never good. we'd need the extracted info from it |
| 08:22.18 | wdoekes | closed |
| 08:22.30 | CRCinAU_ | yeah - mjordan told me that. |
| 08:22.34 | linocisco | igcewieling, E1 PRI card. but it is used for VSAT to VSAT calls |
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| 08:23.01 | CRCinAU_ | then after I recompiled with DONT_OPTIMIZE and BETTER_BACKTRACES, the core dumps stopped |
| 08:23.19 | CRCinAU_ | even when disabling those, recompiling, and reinstalling, it now *still* doesn't core dump. |
| 08:23.31 | wdoekes | ok.. glitch in the matrix then ;) |
| 08:23.36 | CRCinAU_ | so I just put my hands in the air and accept it |
| 08:23.56 | CRCinAU_ | I even downloaded the tarball and started completely from scratch. |
| 08:24.10 | CRCinAU_ | still can't get it to core dump again. |
| 08:24.42 | wdoekes | thanks for the effort, then |
| 08:25.17 | CRCinAU_ | shrugs shoulders |
| 08:25.22 | CRCinAU_ | was worth a try |
| 08:27.34 | ChannelZ | was it newly built when you were originally having problems, or an existing install that had been working previously for awhile and then suddenly broke? |
| 08:27.58 | CRCinAU_ | existing install |
| 08:28.50 | ChannelZ | Maybe some other dependent library got updated in the interim and caused some wonkiness? |
| 08:31.11 | ChannelZ | anyhoo off to bed |
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| 09:03.28 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 09:03.30 | Rico29 | hi all |
| 09:03.52 | Rico29 | is there a way to disallow message leaving on a particular voicemail box ? |
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| 09:30.05 | Rico29 | anybody ? |
| 09:31.23 | kaldemar | don't allow your dialplan to do it. |
| 09:35.05 | Rico29 | to do what ? |
| 09:35.26 | Rico29 | I want to play the VM message tu the caller, but I don't want him to be able to leave a message |
| 09:35.41 | Rico29 | in myd dialplan I only call 'voicemail' app |
| 09:41.23 | kaldemar | Rico29: do you have a custom message? |
| 09:42.12 | kaldemar | Rico29: anyway, don't use the voicemail app if you don't want callers to be able to leave messages. |
| 09:42.17 | Rico29 | kaldemar> no, only unavailable message |
| 09:42.22 | Rico29 | ok |
| 09:42.23 | kaldemar | Rico29: if you want to play a message only, use Playback. |
| 09:42.24 | Rico29 | thanks |
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| 09:58.24 | Rico29 | kaldemar> But I want to play the message configured by the user on his voicemail |
| 09:58.33 | Rico29 | just disallow leaving of message |
| 09:58.54 | Rico29 | that would have been an useful option I think |
| 10:01.10 | kaldemar | you can use something else for that. voicemail is meant for voicemail, not for unavailability messages. |
| 10:01.24 | Rico29 | ok |
| 10:01.57 | kaldemar | an extension that records a message for a user is not a complex task. |
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| 11:19.18 | ghost75 | has someone knowledge how the voicemail perl script from digum works? |
| 11:19.51 | ghost75 | is it precaching the voicemail file before it presents them to browser? |
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| 11:28.40 | kaldemar | ghost75: doesn't look that way. |
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| 11:32.55 | salz212 | Failed to initialize UDPTL, declining image stream... and call drops after Answer... I have been trying to find a solution to this problem for quite some time now.. I am not even using FAX or any image for call only .. carrier send faxt udptl support in SDP to whcih asterisk reply with udptl in sdp after Answer and right after that Carrier send BYE with service not implemented.. |
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| 11:48.08 | ectospasm | salz212: did you contact your provider? IIRC, if they're not sending you a fax, they shouldn't advertise UDPTL in the SDP... |
| 11:49.28 | salz212 | yes we have, you know how carriers are.. they have not given a reasonable answer yet. I was wondering if I can fix it in Asterisk or OpenSIPs for my case. |
| 11:49.31 | dr0ck | and then cry 'not implemented' |
| 11:49.42 | salz212 | exactly. |
| 11:50.07 | ghost75 | kaldemar: directly presenting a link to the file you mean? |
| 11:50.13 | salz212 | but the point is Asterisk is not configured to support FAX in my case why is it.. sending it in SDP./ |
| 11:50.56 | salz212 | I have compared Asterisk 1.6.X and 11.X. The only differnce is Asterisk 1.6 do not Answer with any UDPTL field in SDP |
| 11:52.15 | salz212 | 11 does. |
| 11:53.11 | ghost75 | there are 2 different ways to send t.38 |
| 11:53.40 | kaldemar | salz212: does t38pt_udptl setting in sip.conf help? |
| 11:54.00 | salz212 | no its not.. its the default one .. commented. |
| 11:54.21 | ghost75 | maybe his provider uses rtp instead udtpl ? |
| 11:54.22 | salz212 | i don't want to send.. any fax thats the problem.. |
| 11:54.34 | salz212 | do you want to see the traces again? |
| 11:54.52 | dr0ck | default is prolly diff. between versions, set it to 'no' instead of leaving commented? |
| 11:57.29 | salz212 | i think I tried that but I will do that again.. but I am sure it in not going to work I have tweak many parameters.. |
| 11:58.06 | salz212 | I did set it in carrier peer t38pt_udptl=no as well other than sip.conf.. |
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| 12:20.30 | salz212 | no luck by even setting it to no in sip.conf as well. |
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| 12:22.00 | randulo | ~seen davevg |
| 12:22.04 | infobot | davevg <~davevg__@24.115.249.195.res-cmts.senj.ptd.net> was last seen on IRC in channel #asterisk-doc, 737d 21h 21m 28s ago, saying: 'just checked, <= works :)'. |
| 12:22.12 | wasanzy | am getting this error when starting asterisk 11.2: |
| 12:22.29 | wasanzy | error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 12:24.07 | *** part/#asterisk randulo (~randulo@pdpc/supporter/active/zeeek) |
| 12:24.11 | *** part/#asterisk bpietro (~bpietro@host137-13-static.226-95-b.business.telecomitalia.it) |
| 12:24.17 | ectospasm | wasanzy: check menuconfig, did it build that module? You might be missing dependencies... |
| 12:25.08 | wasanzy | what dependencies could that be? |
| 12:25.25 | ectospasm | wasanzy: you'll have to check menuconfig, which is why I mentioned it. |
| 12:26.29 | wasanzy | ok |
| 12:26.53 | ectospasm | I don't have any version of 11 compiled... I'm checking menuconfig right now. |
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| 12:28.45 | wasanzy | sorry how do I check the menuconfig? |
| 12:29.33 | kaldemar | wasanzy: how did you install? |
| 12:29.34 | WIMPy | How did you install Asterisk? |
| 12:29.46 | wasanzy | from source |
| 12:30.15 | wasanzy | I followed the instructions in the Asterisk The definite Guide book |
| 12:30.18 | ectospasm | looks like you might need libopenssl (or openssl-dev[el]) for res_crypto to be built. |
| 12:30.42 | ectospasm | I'd assume res_crypto is necessary for libasteriskssl.so |
| 12:31.39 | wasanzy | should I install res_crypto? |
| 12:32.27 | wasanzy | I have penssl-devel installed |
| 12:33.12 | wasanzy | am auto loading modules so how come it is complaining? |
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| 12:36.56 | wasanzy | res_crypto is installed when I check menuselect "make menuselect |
| 12:43.21 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:51.21 | wasanzy | I have solved that problem by doing ln -s /usr/lib/libasteriskssl.so.1 /usr/lib64 and ln -s /usr/lib/libasteriskssl.so /usr/lib64 |
| 12:51.39 | wasanzy | now I have another problem: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 12:51.58 | wasanzy | the path exist so I don't know why it is complaining |
| 12:52.17 | WIMPy | Permissions? |
| 12:53.58 | wasanzy | srwxr-xr-x 1 asteriskpbx asteriskpbx 0 Jan 18 12:50 /var/run/asterisk/asterisk.ctl |
| 12:55.26 | kaldemar | are you that user when trying to attach? |
| 12:55.46 | wasanzy | attach? or start? |
| 12:57.07 | kaldemar | wait, you got that when starting asterisk? |
| 12:58.30 | *** join/#asterisk timahvo1 (~rogue@197.180.202.167) |
| 12:58.49 | wasanzy | yes kaldemar |
| 12:59.15 | wasanzy | but when I check the status, asterisk seem to be running |
| 13:00.00 | WIMPy | What status? |
| 13:00.13 | kaldemar | VERSION=`${DAEMON} -rx 'core show version' || ${TRUE}` probably causes that, but it starts anyway. |
| 13:00.29 | kaldemar | interesting. that output did not occur on previous versions. |
| 13:01.47 | wasanzy | /etc/init.d/asterisk status I mean. |
| 13:02.05 | wasanzy | so what can I do about the error? |
| 13:02.54 | kaldemar | it's not an error really. |
| 13:04.32 | wasanzy | does it means it will not affect any thing? |
| 13:06.38 | kaldemar | yes. it it just a line of text to stderr. |
| 13:07.25 | wasanzy | then I guess that shouldn't be printing? |
| 13:07.29 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216.191.106.165) |
| 13:07.56 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:07.57 | kaldemar | wasanzy: yes, the init script is to blame. |
| 13:09.42 | wasanzy | I have to probably take that line off then. |
| 13:10.16 | kaldemar | don't do that. |
| 13:10.42 | wasanzy | ok |
| 13:10.48 | wasanzy | thank you so much |
| 13:10.49 | kaldemar | it servers a good purpose. better would be to modify it to VERSION=`${DAEMON} -rx 'core show version' 2>/dev/null || ${TRUE}` |
| 13:11.07 | wasanzy | ok |
| 13:11.10 | kaldemar | that way you won't see the error print but the script still can figure out if asterisk is already running. |
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| 13:12.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:14.00 | wasanzy | I have this instead: VERSION=`${AST_SBIN}/asterisk -rx 'core show version' |
| 13:14.28 | kaldemar | then modify that to direct stderr to /dev/null. |
| 13:15.28 | wasanzy | this line is really confusing me: if [ "`echo $VERSION | cut -c 1-8`" = "Asterisk" ]; then |
| 13:16.10 | Gugge | if character 1-8 in $VERSION is "Asterisk" then |
| 13:16.23 | wasanzy | oh ok |
| 13:16.37 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tveebcfrleitqllr) |
| 13:18.15 | wasanzy | VERSION=`${AST_SBIN}/asterisk -rx 'core show version'` 2>/dev/null |
| 13:18.36 | wasanzy | still getting: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 13:18.50 | Gugge | does it? |
| 13:19.08 | Gugge | put 2>/dev/null inside the ``'s |
| 13:19.12 | joris2 | hi all, I need to record sip calls with my asterisk installation. But not on the same machine, on a special recording machine, what is the best approche to do this? I need the recorded voice data as well as the call details (in mysql) |
| 13:19.13 | WIMPy | you might have to insert it before the ` |
| 13:19.14 | kaldemar | that has been changed in asterisk.c to 11.2.0, it now prints directly to stderr for some reason instead of using ast_log. |
| 13:20.40 | wasanzy | ok worked, thank you. |
| 13:22.51 | kaldemar | reason for the change is in commit r376447. |
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| 13:30.04 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-013-251.sc.res.rr.com) |
| 13:32.49 | kaldemar | wasanzy: https://issues.asterisk.org/jira/browse/ASTERISK-20945 |
| 13:37.01 | WIMPy | would have gone for killall. |
| 13:42.07 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 13:59.34 | joris2 | anyone? |
| 14:00.06 | joris2 | I was thinking aobout the RTP monitoring feature, but that doesn't seem to be very stable |
| 14:00.52 | [TK]D-Fender | joris2, Whatever way you want. Mount a remote filesystem. Copy after the end ("core show application monitor"). |
| 14:01.26 | [TK]D-Fender | joris2, And DB storage of CDR's is already well documented |
| 14:01.57 | joris2 | it's not just the CDR's. It's the actual data |
| 14:01.58 | *** join/#asterisk sustav (~vpp@nat/digium/x-liupuscooptalecw) |
| 14:02.10 | joris2 | copy after hangup is my alternate option |
| 14:02.31 | joris2 | I was wondering if there was an option to record it directly on another server |
| 14:02.51 | joris2 | something like the xorcom patch and oraka |
| 14:03.02 | joris2 | but then clean and some newer |
| 14:03.58 | [TK]D-Fender | Mount a remote filesystem. <- |
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| 14:04.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:04.27 | joris2 | I guess that's my only option left... |
| 14:04.32 | joris2 | thanks ! |
| 14:04.46 | *** join/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 14:05.55 | [TK]D-Fender | Store & move to remote would be a good idea. |
| 14:06.11 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 14:06.19 | joris2 | not directly store on the remote fs> |
| 14:06.20 | joris2 | ? |
| 14:07.20 | *** join/#asterisk Kyosh (~whoa@pool-72-89-93-13.nycmny.fios.verizon.net) |
| 14:07.31 | [TK]D-Fender | joris2, And when that link fails for whatever reason your attempts to record die with no recovery and possibly choke up the server? |
| 14:08.34 | joris2 | hmmm... I thought asterisk would discard the recording then... |
| 14:08.48 | joris2 | but, move after recording is a better idea :-) |
| 14:09.08 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 14:09.46 | [TK]D-Fender | Somehow "discarding" doesn't sound "good" |
| 14:10.05 | joris2 | hehe, true |
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| 14:19.35 | jzaw | ok ok where the f is the command line in vista ? |
| 14:19.41 | jzaw | i cant see it |
| 14:20.03 | joris2 | windowskey+r -> cmd ? |
| 14:20.07 | WIMPy | thinks that is intentional |
| 14:20.19 | killown | does anyone knows how to set up a x100p board? http://bpaste.net/show/dxwNobnvH5DsoB2fFedu/ no phone cable is connected to the board and even so the Alarm is Ok |
| 14:20.23 | jzaw | ta |
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| 14:20.53 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:20.56 | jzaw | is not a windows user and is only forced to do this at the point of a barrel of a semi auto |
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| 14:26.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:34.07 | igcewieling | killown: no. nobody uses that card. contact the mfgr. |
| 14:34.16 | *** join/#asterisk sustav (~vpp@nat/digium/x-tbglxcsaxuvzbxij) |
| 14:36.45 | killown | igcewieling, has you sure this is not an asterisk problem http://bpaste.net/show/NVUv1kGxLGXzG1acPL8c/ ? |
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| 14:37.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:37.19 | igcewieling | looks like a card driver problem |
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| 14:49.59 | *** part/#asterisk district (district@falseprophet.ca) |
| 14:50.57 | killown | so I found it http://www.voip-info.org/wiki/view/X100P+clone :P |
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| 15:03.27 | *** join/#asterisk jan_bee (~quassel@euve6285.server4you.net) |
| 15:07.47 | jan_bee | Hello, I am running asterisk 1.8.13.1 on Debian Squeeze and for some reason the memory usage keeps rising until the server goes out of memory. Does anyone here know how to debug such a problem? |
| 15:11.01 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:12.26 | [TK]D-Fender | jan_bee, Upgrade. You're 7 releases behind |
| 15:13.27 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 15:13.53 | jan_bee | It's the latest release from de squeeze backports repository.. |
| 15:15.41 | [TK]D-Fender | jan_bee, Yes, Debian is slow. |
| 15:17.13 | killown | dahdi -> included from /usr/src/dahdi-linux/drivers/dahdi/xpp/xpd.h:26, include/linux/types.h:36: error: previous declaration of ‘bool’ was here |
| 15:17.49 | killown | igcewieling, it's not a driver card problem, it's 100% a dahdi problem which is from asterisk |
| 15:18.46 | [TK]D-Fender | ... |
| 15:18.51 | [TK]D-Fender | DAHDI is the driver.... |
| 15:19.14 | killown | I downloaded this driver from asterisk repos |
| 15:19.31 | killown | not from the board manufacturer |
| 15:19.44 | jan_bee | [TK]D-Fender: It makes maintenance a bit easier though. |
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| 15:26.55 | igcewieling | killown: nobody is going to ever issue any bug fixes for the x100p driver. |
| 15:26.55 | *** join/#asterisk shadebob (~shadebob@41.205.200.53) |
| 15:27.01 | killown | yeah include/linux/types.h already have typedef _Bool bool; this not need to be declare one more time in xpd.h, I just removed that declaration and solved the problem |
| 15:27.12 | killown | igcewieling, you speak for everyone here? |
| 15:27.21 | igcewieling | you can kick and scream and whatever, but you simply are not going to get help. |
| 15:27.31 | igcewieling | killown: I speak about everyone, but not for them. |
| 15:27.32 | killown | it's what you are saying |
| 15:27.44 | killown | if you don't want to help me, you don't need to waste your time |
| 15:27.46 | leifmadsen | he's not wrong |
| 15:28.00 | leifmadsen | the x100p stopped being supported by nearly everyone about 4 years ago |
| 15:28.45 | [TK]D-Fender | And that is more than 5 years after Digium stopped ever distributing them, aside from the fact yours is a clone |
| 15:28.56 | leifmadsen | if you're doing development though in the channel drivers, you might want to try using #asterisk-dev |
| 15:29.05 | killown | [TK]D-Fender, not a clone, this is the original one |
| 15:29.32 | leifmadsen | that's like adding "Real Beef" on the McDonalds hamburger |
| 15:29.48 | igcewieling | They stopped making them about 14 years ago, didn't they? |
| 15:29.53 | [TK]D-Fender | That's "Real Beef (tm)" |
| 15:29.57 | killown | no problem people I will patch it for myself |
| 15:30.07 | leifmadsen | igcewieling: not sure... it was always just a particular modem that Digium sold I'm pretty sure |
| 15:30.57 | igcewieling | leifmadsen: they added a heat sink! |
| 15:31.03 | leifmadsen | was it a nice heat sink? |
| 15:31.11 | igcewieling | I did choose "they" carefully |
| 15:31.24 | igcewieling | leifmadsen: hand crafted, every one! |
| 15:32.02 | igcewieling | I bought 5 of them just before the OEM stopped making them back in the early 2000s |
| 15:32.35 | igcewieling | ROFL! Bicom killed my untar. |
| 15:32.40 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 15:32.40 | igcewieling | I wondered if they would. |
| 15:34.58 | igcewieling | *sigh* wrong window |
| 15:36.09 | [TK]D-Fender | igcewieling, Yes, made by the honest hard-working indigenous people of .... wherever. |
| 15:37.15 | igcewieling | When I make this stuff up it is the Maori. |
| 15:40.47 | [TK]D-Fender | igcewieling, https://www.youtube.com/watch?v=dN8vyO8ILD8 |
| 15:42.03 | igcewieling | Ah. |
| 15:42.16 | igcewieling | I have a TiVo, I don't see many commercials |
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| 15:47.16 | [TK]D-Fender | igcewieling, It isn't a commercial... officially ;) |
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| 15:55.52 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 15:55.57 | Ice_Strike | Hello |
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| 16:02.36 | Ice_Strike | Polycom use TFTP to connect to my server |
| 16:02.43 | Ice_Strike | to open a firewall, is this correct: -A INPUT -p udp -m udp --dport 5060 -j ACCEPT |
| 16:04.37 | pabelanger | EPP |
| 16:04.38 | Ice_Strike | sorry |
| 16:04.40 | Ice_Strike | O meant |
| 16:04.41 | pabelanger | change that to HTTP |
| 16:04.42 | Ice_Strike | -A INPUT -p udp -m udp --dport 69 -j ACCEPT |
| 16:04.43 | pabelanger | :D |
| 16:05.06 | Ice_Strike | According to documentation it use port 69 |
| 16:05.37 | pabelanger | yes, udp/69 is TFTP |
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| 16:11.29 | Ice_Strike | pabelanger can you see anythign wrong with this: http://codepad.org/Ms2lWFcn |
| 16:11.39 | Ice_Strike | polycom is not connecting |
| 16:12.49 | jeffspeff | channel variables don't follow to a different extension within the same context do they? |
| 16:14.56 | tzanger | woo, upgrading to 1.8.20.0 |
| 16:15.11 | [TK]D-Fender | <Ice_Strike> Polycom use TFTP to connect to my server <- Does it? Where did you set this? |
| 16:15.18 | tzanger | (from 1.8.7.1, 1.4.something before that) |
| 16:16.53 | killown | what is a cheaper fxo card supported by the asterisk? |
| 16:17.02 | Ice_Strike | [TK]D-Fender Yes. Menu -> Setting -> Advance -> Admin Setting -> Network Conf -> Server Menu |
| 16:19.25 | pabelanger | tzanger, We use 1.8.7.1 religiously. Why did you upgrade? |
| 16:19.32 | pabelanger | plus some backports |
| 16:21.43 | [TK]D-Fender | Ice_Strike, go check that you actually have the right files * permissions on them |
| 16:22.00 | Ice_Strike | It did work before |
| 16:22.09 | Ice_Strike | then I setup a firewall |
| 16:22.15 | Ice_Strike | so something stop working |
| 16:22.16 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:22.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:22.31 | tzanger | pabelanger: no specific reason. |
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| 16:24.35 | [TK]D-Fender | Ice_Strike, "TFTP typically uses UDP as its transport protocol, but it is not a requirement. Data transfer is initiated on port 69, but the data transfer ports are chosen independently by the sender and receiver during initialization of the connection. The ports are chosen at random according to the parameters of the networking stack, typically from the range of Ephemeral ports.[3]" |
| 16:24.53 | [TK]D-Fender | Ice_Strike, Looks like the data is on another port for ACK and should fail |
| 16:25.08 | pabelanger | tzanger, So, why bother upgrading? |
| 16:26.17 | jeffspeff | channel variables don't follow to a different extension within the same context do they? |
| 16:26.47 | [TK]D-Fender | jeffspeff, Yes |
| 16:27.15 | [TK]D-Fender | jeffspeff, there is no scope within the dialplan |
| 16:27.41 | tzanger | pabelanger: mostly just to see what's changed. I have an odd problem with voip.ms that I'm sure is their end but I want to see if the sip stack updates have any bearing on it |
| 16:27.45 | Ice_Strike | I see hmm |
| 16:28.04 | jeffspeff | when i try to call variables in the h extension, they show up as empty in my logs. |
| 16:29.08 | [TK]D-Fender | jeffspeff, "h" has special rules because of the state of the channel being dead |
| 16:30.08 | jeffspeff | so, how do i get something to run after the call is hung up that uses variables set when the channel was active? or are those the special rules you were talking about? |
| 16:31.43 | [TK]D-Fender | jeffspeff, Show us what you're doing |
| 16:32.16 | pabelanger | tzanger, Okay. Was mostly curious if you were having a specific issue and trying to fix it. |
| 16:36.37 | jeffspeff | [TK]D-Fender, http://pastebin.com/CZ02Lnhw |
| 16:37.38 | [TK]D-Fender | jeffspeff, And a complete call attempt please |
| 16:37.51 | jeffspeff | running a new one now |
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| 17:41.20 | DoSJustin | is it safe to have branching and System calls in the 'h' extension? |
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| 18:40.41 | igcewieling | DoSJustin: in my experience yes, if you mean goto/gosub/etc by "branching" |
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| 18:49.20 | b0ot | What is the most common DSCP value for VoIP? |
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| 18:52.32 | ChannelZ | 42 |
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| 18:54.02 | ChannelZ | I just made that up. |
| 18:54.15 | philippel_mac | Are you guys aware or ever going to fix your jira server so that Safari can browse the ticket system? I used to think it was just my laptop but just had someone else check and they have the same problem |
| 18:54.42 | philippel_mac | the problem is you aren't accepting Safari's certiciate |
| 18:57.01 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 18:58.38 | ChannelZ | http://lists.digium.com/pipermail/asterisk-dev/2011-July/050119.html |
| 19:00.13 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:03.12 | *** join/#asterisk phonebuff (~chatzilla@23.31.148.65) |
| 19:04.13 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 19:05.36 | *** join/#asterisk nix8n82 (~AndChat27@24.143.11.81) |
| 19:22.10 | b0ot | What is the default DSCP value that asterisk will tag voice with? |
| 19:26.31 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 19:27.03 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 19:27.17 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 19:35.02 | leifmadsen | philippel_mac: "you guys" is Digium in particular |
| 19:35.59 | philippel_mac | 'you guys' is whom ever is responsible for the issues. asterisk.org :) |
| 19:36.30 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.241.7) |
| 19:36.43 | philippel_mac | but posting this here, I was curious if a bunch of people on macs would come back and say 'not a problem here' or 'yeah broken here also' … I've checked it on two systems and it's been like this for quite a while |
| 19:37.46 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:40.11 | *** join/#asterisk nuken (~nuken@open.integrada.coop.br) |
| 19:41.29 | nuken | hi all |
| 19:41.52 | nuken | i'm using a sip trunk between cisco 2811 router and Asterisk |
| 19:42.15 | nuken | calls from cisco -> asterisk are so fine ! codecs ok, voice quality very nice |
| 19:42.48 | nuken | but, calls from asterisk -> cisco don't use my preferencial codec(ilbc) and i have bad voice quality |
| 19:43.00 | nuken | i'm using freepbx |
| 19:43.26 | nuken | cisco 2811 with two FXS ports |
| 19:43.40 | nuken | any idea ? |
| 19:45.48 | [TK]D-Fender | nuken, Stop making it a preference and make it the only choice. |
| 19:47.02 | nuken | [TK]D-Fender, i've already tried, and asterisk returns me that it can't do the call because there is no codec avaliable |
| 19:47.17 | [TK]D-Fender | nuken, Then fix your Cisco. |
| 19:47.37 | nuken | cisco -> asterisk ok in both situations |
| 19:47.50 | nuken | do you think that the problem can be in cisco router ? |
| 19:48.02 | [TK]D-Fender | When it won't accept your preference when you send it call... you need to make it do so. |
| 19:49.36 | nuken | humm ok.. i will try to fix my cisco config |
| 19:50.07 | carrar | I suspect the color of the cat5 sheilding |
| 19:56.27 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 20:02.49 | [TK]D-Fender | Cat5E natively supports fuchsia |
| 20:03.02 | [TK]D-Fender | I recommend upgrading |
| 20:06.24 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 20:07.42 | ChannelZ | Everyone knows red is faster. |
| 20:15.36 | *** join/#asterisk pcAngel (yoink@S0106c8be195a474c.vc.shawcable.net) |
| 20:16.11 | pcAngel | Hi guys, My SIP peers are going between unreachable and reachable, which has never been a problem before today. The peers are all in different networks and my configuration hasn't changed |
| 20:16.25 | timholum | ChannelZ: But blue has fewer issue's ( due to it being calmer :) ) |
| 20:16.26 | pcAngel | I feel like I've hit some kind of a critical mass with my number of peers and the amount of peers asterisk can qualify |
| 20:16.43 | pcAngel | Is there anything like that I should be aware of? |
| 20:16.49 | pcAngel | or does anyone have other ideas? |
| 20:16.57 | pcAngel | I'm on asterisk 10.7 |
| 20:18.55 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 20:19.04 | apb1963_ | OMG, you guys cannot be serious about red being faster |
| 20:19.41 | apb1963_ | It's well documented that violet is fastest: http://www.google.com/imgres?imgurl=http://juliank.com/english/aura-body/files-aura/Human%2520Energy%2520Filed/color_wavelength_frequency.png&imgrefurl=http://juliank.com/english/aura-body/files-aura/Human%2520Energy%2520Filed/Human%2520Energy%2520Field.htm&h=162&w=451&sz=48&tbnid=YP-FAEbAa3zJQM:&tbnh=46&tbnw=129&zoom=1&usg=__RO-wcjlvkKaRZDxgEpAOYYxjF5E=&docid=47GrLitgMZ9TYM&sa=X&ei=2qz5UNuQKeyzigLM7YC4Dw&ve |
| 20:20.45 | timholum | apb1963_ thats light, electrons work differently :) |
| 20:21.08 | apb1963_ | Are you saying colors are made up of electrons? |
| 20:21.18 | timholum | yup |
| 20:21.21 | timholum | must be |
| 20:23.47 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 20:25.10 | apb1963_ | I tried to convince the audacity people to map sound to light a number of years ago... but they weren't listening. Here's one of my favorite web pages that I just found a few seconds ago: http://www.lunarplanner.com/Harmonics/planetary-harmonics.html |
| 20:26.43 | apb1963_ | If you're looking at it.. be sure to scroll down. |
| 20:28.12 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:40.51 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-013-251.sc.res.rr.com) |
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| 20:42.29 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 20:47.58 | pcAngel | You develop "favorite" webpages pretty quickly.. a few seconds |
| 20:48.02 | pcAngel | Ugh |
| 20:52.04 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 20:57.16 | *** join/#asterisk mwally (~mwally@mwally.opencnam.com) |
| 21:09.51 | *** join/#asterisk minotaur01 (~minotaur0@bas9-hamilton14-3096720286.dsl.bell.ca) |
| 21:11.17 | apb1963_ | It's a good page. |
| 21:12.53 | apb1963_ | It has everything... light... musical notes.... words....pictures.... what more could you ask for from a web page? |
| 21:13.20 | apb1963_ | And colors... can't forget the colors. |
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| 21:28.25 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.180.25) |
| 21:38.15 | ccrnp | Hi Folks ! I have this weird error on asterisk 1.4.32 can anyone help me with this, I see hannel.c: Exceptionally long voice queue length queuing to IAX2/XXXXXX followed by chan_iax2.c: I should never be called! |
| 21:43.04 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:51.40 | ChannelZ | Hamster Dance. |
| 21:53.59 | fubada | hi |
| 21:54.08 | fubada | i cant figure out how to define alerts in 4.2 |
| 21:54.12 | ccrnp | guys! anyone to leas me |
| 21:54.15 | ccrnp | *lead |
| 21:54.16 | fubada | did it change since 3? |
| 21:56.46 | fubada | ops wrong channel |
| 22:01.24 | *** join/#asterisk hwgasdfasdf (~andrew@ip98-177-170-15.ph.ph.cox.net) |
| 22:01.36 | hwgasdfasdf | hi |
| 22:02.10 | hwgasdfasdf | does anyone else have problems with motif / xmpp where outgoing calls stop working after a while? They just ring and ring? |
| 22:02.29 | hwgasdfasdf | trying to figure out where the problem is, and just discovered unloading chan_motif crashes asterisk completely |
| 22:03.42 | navaismo | ccrnp, is that aiax trunk between servers? |
| 22:04.06 | hwgasdfasdf | then of course after reloading asterisk, everything works again, which is a problem for figuring out what is busted |
| 22:04.52 | hwgasdfasdf | old jingle / gtalk on another server works fine, always |
| 22:09.46 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:13.52 | ChannelZ | ccrnp: 1.4 is pretty old. Do you use Pickup a lot? |
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| 22:20.45 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:30.51 | ccrnp | navaismo, yeah its a iax trunk between servers |
| 22:31.08 | ccrnp | navaismo, no not that often |
| 22:51.08 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 22:51.55 | navaismo | ccrnp, you have enable the trunk=yes? if so happens if you disable it trunk=no? |
| 23:01.07 | sawgood | This might be 'dumb' (real dumb) with the luck I've had asking for help in the last 2 days, but ... |
| 23:01.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.40 | sawgood | from the Asterisk CLI (1.8.7.2) is there anything I can watch for that says, "Shawn I sent a SIP notify to x298 to turn on the MWI light because voicemail is present" |
| 23:02.13 | sawgood | MWI light is not coming on (no email attachments) just voicemail on a 1.8.7.2 box out to a local Yealink phone on my desk |
| 23:02.58 | sawgood | on the CLI, I do see other VM activity, just nothing after the phone is hung up talking leaving the VM (nothing else at core set verbose 5) |
| 23:03.58 | sawgood | voicemail show users for default |
| 23:03.58 | sawgood | shows one vm present for x298 |
| 23:05.04 | *** join/#asterisk sustav (~vpp@nat/digium/x-esjzqfxcdiieoqur) |
| 23:05.45 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:06.20 | *** join/#asterisk sparrowjs (~Jon@216.241.45.82) |
| 23:07.56 | sawgood | Here is my [general] context for /etc/voicemail.conf |
| 23:07.58 | sawgood | http://pastebin.com/UZrv3gaw |
| 23:08.54 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:f892:5ce8:80e6:da8a) |
| 23:15.07 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:15.19 | kaldemar | sawgood: define mailbox in sip.conf. you'll see a notify in sip debug. |
| 23:15.48 | sawgood | ty |
| 23:30.47 | sawgood | http://pastebin.com/WzKRsdJE |
| 23:31.52 | sawgood | I connected to Asterisk box (yes it does run FreePBX as well), turned on sip set debug peer 298, make a call to voicemail (left a message), hung up the phone, waited 10 more seconds and turned off sip set debug peer 298 (I do not think the paste has any other traffic) |
| 23:33.22 | ChannelZ | did you sip reload and rereg the phone for fun? |
| 23:33.28 | sawgood | no |
| 23:33.34 | ChannelZ | no to which? |
| 23:33.44 | sawgood | no sip reload |
| 23:33.48 | sawgood | I did it just now though |
| 23:34.02 | ChannelZ | * should send a notify after the message. |
| 23:34.11 | ChannelZ | a new message I should say |
| 23:34.31 | ChannelZ | Only if the phone specifically subscribes to MWI will the phone magically know after a reboot for instance |
| 23:34.45 | sawgood | I waited about 10-15 seconds after hanging up (looking for that) and only 1 message comes in after hang up (after 5 seconds from hang up I'd say) |
| 23:34.54 | ChannelZ | yeah but you said you didn't reload your sip config after having added the mailbox for that peer, yes? |
| 23:35.12 | sawgood | With this case, I have other boxes and identical phones (pre-provisioned with the same settings) ... they work 100% with MWI |
| 23:35.48 | sawgood | I can do the test over if that would help? |
| 23:36.53 | ChannelZ | If you have a mailbox=xxx set for the peer in question, MWI will only turn on after a new message is left. |
| 23:37.19 | sawgood | oh ... is that right ... I did a sip reload then ... I'll try |
| 23:37.27 | *** join/#asterisk ryan42 (unix@doc-72-47-18-129.brenham.tx.cebridge.net) |
| 23:38.21 | sawgood | ok voicemail shows ZERO messages for the extension. I did a SIP reload, and I'll try again now |
| 23:39.08 | ChannelZ | sip show peer XYZ should also show the mailbox it's monitoring for that peer |
| 23:40.12 | sawgood | <PROTECTED> |
| 23:40.34 | ChannelZ | is "device" a voicemail context? |
| 23:40.42 | sawgood | default 298 LAB Yealink 0 |
| 23:40.52 | sawgood | well the mailbox line was part of the output from sip show peer 298 |
| 23:41.09 | sawgood | I think 'default' is the context 'no'? |
| 23:41.51 | ChannelZ | Usually, but wondering why yours says "device". Is that what you put in sip.conf for that peer's mailbox? |
| 23:41.59 | sawgood | on a box, which the MWI works full time, this is the output I have from sip show peer 402 |
| 23:42.03 | sawgood | <PROTECTED> |
| 23:42.11 | ChannelZ | See the difference? |
| 23:42.19 | sawgood | actually, I can 'see' that ... wow! |
| 23:42.50 | sawgood | I'll look on another box which does work too to confirm BRB! |
| 23:44.26 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 23:44.28 | sawgood | I looked at two other boxes (both with working VM to MWI) and they both had Mailbox : 201@device |
| 23:45.16 | ChannelZ | it's not necessarily wrong if you actually have a voicemail context called "device" that some peers are in and one called "default" that others are in. |
| 23:45.45 | ChannelZ | However at least by what you've shown, mailbox 298 is in default. |
| 23:45.59 | sawgood | wow, I changed it to 292@default (and the little green light comes on now) |
| 23:46.21 | ChannelZ | It's possible the other ones set to "device" are only working because you've configured the phones themselves with the right mailbox/context and they are subscribing themselves. |
| 23:47.21 | ccrnp | navaismo, I haven't set the value for trunk |
| 23:54.56 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 23:56.10 | sawgood | Is there a way from the CLI, to issue a command to see which end points have their MWI 'on' (what I say) is the green button lit on the phone |
| 23:56.23 | sawgood | can I issue a subscribtion request to see who has their MWI light on? |
| 23:58.24 | ChannelZ | you can arbitrarily turn them on and off but I'm not sure about asking. |
| 23:59.28 | sawgood | cool: I'd like to try to turn off the green light on my phone (any tips)? |
