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| 00:43.09 | KNERD | Engrish |
| 00:47.40 | phix | back |
| 00:48.22 | phix | Yeah I had Italian and French class, but that was a life time ago :) |
| 00:49.07 | phix | salut davlefou :) That's about all I can say |
| 00:49.55 | phix | Dovid: yeah KNERD is right, Engrish is the best you will get out of him :) |
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| 00:59.30 | [TK]D-Fender | "Good night kids" |
| 00:59.42 | jaytee | turning in early? |
| 00:59.48 | jaytee | :-) |
| 00:59.55 | [TK]D-Fender | No, translating davlefou |
| 01:00.06 | jaytee | I know, just kiddin |
| 01:00.40 | [TK]D-Fender | jaytee: It's hard to tell the line between "humour" and "insufficient medication" sometimes ;) |
| 01:00.49 | jaytee | I don't speak french but that statement was so obvious even I didn't have to use babelfish |
| 01:01.12 | jaytee | ugh, the lousy version of Dune is on BBCA |
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| 01:09.47 | duri | anybody with OBI110s around ? |
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| 01:46.25 | Chorca | hey guys. Still having some issues with the slin16 input format, wondering if anyone knows how i can get the codecs to give me more output, or how i can further debug a codec mismatch |
| 01:53.00 | Chorca | Getting the following error: [Jan 15 20:51:37] WARNING[2930]: channel.c:4986 ast_write: Codec mismatch on channel SIP/chrispoly-00000002 setting write format to slin from slin16 native formats (g722) |
| 01:53.22 | Chorca | no matter what format I set moh to in musiconhold.conf, that's the error I get. |
| 01:53.58 | Chorca | This is on 10.1.0, while the exact same file on 10.0.0, plays back fine without that error. |
| 01:55.08 | Chorca | I'm trying to figure out how to pull more debugging out of the codecs or something, or the reason it's setting them, what the mismatch might be, anything. |
| 01:55.26 | Chorca | i set verbose and debug to 100 but i'm not seeing any kind of useful messages. |
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| 02:13.49 | ChannelZ | What do you mean what format you set moh to? |
| 02:14.19 | Chorca | in moh.conf, the format= line, i have set to format=wav16 |
| 02:14.32 | Chorca | core show file formats says that wav16 is slin16 |
| 02:15.22 | ChannelZ | and 10.1.0 is only RC, entirely possible something is wrong with it |
| 02:16.17 | Chorca | soxi says the file is 16000hz, 256kbit 16-bit Signed Integer PCM |
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| 02:16.42 | Chorca | it plays back after that error, at half speed |
| 02:17.32 | Chorca | and a previous error i got while fooling around revealed an "expected 8000Hz" message, which leads me to believe that somehow the formats are trying to all use 8000Hz and failing out |
| 02:17.49 | ChannelZ | is your file called .wav16? |
| 02:17.54 | Chorca | correct |
| 02:19.42 | Chorca | Hmm.. looks like it may just be limited to the wav16 format |
| 02:20.09 | Chorca | I just tried at 44khz with format=sln44 and an extension of .sln44, and it's working now, no errors. |
| 02:20.39 | Chorca | if i remember the wav16 format was added in by someone, so it may have been broken by other code in 10.1 |
| 02:21.07 | Chorca | well, added previously to the other high-bitrate slin formats |
| 02:21.14 | ChannelZ | not sure. slin16 is 16-bit but I don't know if it's limited in samplerates to specifics |
| 02:22.37 | Chorca | it's odd, cause the same file works on another machine. it transcodes through g.722, and so i removed the resample library to try and get the same codec chain on this machine, which happened, but it did the same thing, so it seems like the input codec isn't reading the file properly. |
| 02:22.44 | Chorca | it's dropping back to slin at 8khz |
| 02:23.48 | ChannelZ | well that it plays at half speed implies it's supposed to still be 8khz though why it works on another machine I don't know |
| 02:24.10 | ChannelZ | http://forums.asterisk.org/viewtopic.php?t=74366 |
| 02:24.31 | ChannelZ | that said not sure of the status of 10.x |
| 02:36.06 | Chorca | yeah, not sure there either |
| 02:37.24 | Chorca | I mean, g.722 only runs at 7khz anyway |
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| 02:37.40 | Chorca | looking at the translations, it runs slin44 -> slin16 -> g722 |
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| 03:13.07 | MacOS | Anybody here ? |
| 03:15.16 | Stiliyan | seems like everybody is sleeping :) |
| 03:15.40 | WIMPy | No bodies. Only lost souls. |
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| 03:17.55 | Stiliyan | Hehe .. sounds very optimistic :) |
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| 03:27.58 | michael-i | Hi all. I know there is a manager command SIPNotify but is there a way to build a Notify message dynamically from the CLI or fire from dial plan? |
| 03:30.45 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 04:07.10 | ChannelZ | michael-i: well there is the 'sip notify' command on the console |
| 04:07.58 | michael-i | ChannelZ: right, but it doesn't let you construct events dynamically. You have to choose from the ones in sip_notify.conf |
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| 05:24.10 | phix | Gang! |
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| 05:24.49 | prasanthprabalan | Hi all |
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| 05:25.22 | prasanthprabalan | what is the minimum hardware requirement of a 32 line PRI card |
| 05:25.44 | prasanthprabalan | Like how much ram it will take for a single line |
| 05:25.57 | prasanthprabalan | *memory usage |
| 05:26.01 | prasanthprabalan | ?? |
| 05:31.19 | [TK]D-Fender | prasanthprabalan: I've seen 8 port PRI cards... where have you found a 32 port card? |
| 05:32.18 | coppice | why would anyone want a 32 port PRI card? wouldn't an OC<something small> card make more sense at such a density? |
| 05:33.11 | *** join/#asterisk dev-ast (~dev-ast@cpe-76-183-171-186.tx.res.rr.com) |
| 05:33.30 | dev-ast | hi |
| 05:33.40 | [TK]D-Fender | coppice: "Sense" didn't come bundled with the package :) |
| 05:33.46 | prasanthprabalan | its not 32 port card |
| 05:34.14 | prasanthprabalan | it is a TE122 degium card which will support E1 32 lines |
| 05:34.40 | coppice | that's a 30 line card, not 32 |
| 05:34.44 | [TK]D-Fender | Last I checked, E1 was 31 time-slices (PRI = 30B +1d) |
| 05:34.46 | [TK]D-Fender | Not 32 |
| 05:35.32 | [TK]D-Fender | prasanthprabalan: And don't count PRI's like that. That is a single E1. And frankly a P3 could do that fine |
| 05:36.14 | prasanthprabalan | ok what might be the minimum ram and processor required |
| 05:36.46 | [TK]D-Fender | prasanthprabalan: What are you trying to get away with? |
| 05:36.54 | coppice | that really depends on the kind of processing you intend to do |
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| 05:38.17 | prasanthprabalan | Ya i want it to be done with Intel processor |
| 05:39.09 | prasanthprabalan | I want to know the minimum hardware requirement so that the system wont gives problem because of hardware shortage |
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| 05:40.24 | *** join/#asterisk dev-ast (~dev-ast@cpe-76-183-171-186.tx.res.rr.com) |
| 05:43.27 | phix | hey, this may not be an asterisk specific question, but does any one know of a good free or commerical application for faxing from windows to hylafax / asterisk? |
| 05:44.45 | [TK]D-Fender | prasanthprabalan: You are asking a vague and dangerous question with no scope of what kind of equipment you are expecting to try to use. You are asking for trouble. |
| 05:45.13 | [TK]D-Fender | prasanthprabalan: And there is more in play than jsut the card. There is no "number" we can give you. it depends. |
| 05:45.37 | prasanthprabalan | Ok i got this data while searching.. |
| 05:45.42 | [TK]D-Fender | phix: There are plenty of connector agents for windows... |
| 05:45.44 | prasanthprabalan | 800- Mhz Pentium 3 processor |
| 05:45.44 | coppice | phix: the hylafax web site lists several windows clients for interacting with hylafax |
| 05:45.44 | prasanthprabalan | 64 MB RAM |
| 05:45.44 | prasanthprabalan | PCI/PCI-e Slot |
| 05:45.59 | [TK]D-Fender | prasanthprabalan: That MIGHT do, but it depends on what you're going to be doing. |
| 05:46.09 | prasanthprabalan | This is the minimum req for Degium TE122 card |
| 05:46.22 | [TK]D-Fender | prasanthprabalan: I don't think it has fully sunk in yet. How you process your calls can completely kill your server. |
| 05:46.47 | [TK]D-Fender | prasanthprabalan: If you want advise that wont get you screwed when you're off the mark you'd better provide clearer details |
| 05:47.01 | [TK]D-Fender | prasanthprabalan: Because you are doing a bad job trying to dogde the requirements |
| 05:53.29 | prasanthprabalan | Ok thanks for the reply |
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| 06:51.33 | phix | coppice: Yeah they look old and unsupported / maintained |
| 06:52.09 | phix | [TK]D-Fender: What is the best one to use that works in windows 7 too? I usually use WHFC but it is unmaintained now and doens't work under windows 7 |
| 06:52.36 | [TK]D-Fender | phix: Never fully got around to this stuff myself... |
| 06:52.40 | [TK]D-Fender | and on that note.. bed time... |
| 06:52.45 | [TK]D-Fender | BBL |
| 06:53.02 | coppice | phix: I thought WHFC had a new maintainer now. try googling |
| 06:53.11 | phix | coppice: it does? ok I will google it |
| 06:53.49 | coppice | there are commercial clients, but I guess that's not what you are looking for |
| 06:53.54 | phix | lol West Ham Football Club apparantly :) |
| 06:54.46 | phix | coppice: I am after what ever, I dont care if it is commerical, I just want one that works :) supports cover sheets and address / phone book, pref one that syncs with gmail contacts but that isn't a major requirement, that would be gravy |
| 06:54.57 | phix | but I guess no one bothers with faxing these days |
| 06:55.16 | phix | except maybe lawyers as they can charge $10 or more a page |
| 06:56.04 | coppice | FAXing is still very important in some circles, typically ones driven by legal concerns, but medical use is huge too |
| 06:57.10 | coppice | you might be amazed how many people use iaxmodem + hylafax or spandsp to handle millions of FAXes a day, and the number handling something like 100k must be huge |
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| 07:02.01 | phix | ah ok, yeah I suppose you are right, as faxing is more of a legal document than an email |
| 07:02.50 | phix | yeah I am using iaxmodem + hylafax, I just need a good windows client, I use gfax for linux |
| 07:06.14 | phix | ok coppice so do you have an opionion of a good free or commerical windows faxing client? |
| 07:06.40 | phix | ifax is first on google search :\ |
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| 07:07.57 | phix | nice 30 day free trial, might give that a go |
| 07:09.53 | coppice | its always hard to find these things when you actually need to :-\ |
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| 07:11.31 | coppice | ah, this is the one I was thinking of with fresh win 7 maintainence - http://sourceforge.net/tracker/?func=detail&aid=2935174&group_id=91606&atid=597727 |
| 07:15.06 | phix | interesting |
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| 07:42.21 | IsUp | morning |
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| 07:59.22 | *** join/#asterisk michel2019 (~michel.fr@193.227.186.146) |
| 08:00.06 | michel2019 | When I make asterisk -rx "core show channels verbose", one of the headers is Extensions....Is that the destination number dialed by user? |
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| 08:07.49 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:07.51 | schmidts | good morning |
| 08:12.18 | ollii | moin |
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| 08:16.48 | IsUp | michel2019: yes |
| 08:17.34 | jondot | anyone has a clue about how to set up eSpeak TTS like google translate did? |
| 08:19.21 | michel2019 | IsUp, but under the Data header (Outgoing Line) I have this: SIP/X.X.X.X/+9193080 |
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| 08:19.44 | michel2019 | IsUp, and what is saved into my database is this number that begin by +91 |
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| 08:21.00 | IsUp | michel2019: '(Outgoing Line)' is normal. that means channel is bridged. |
| 08:21.16 | schmidts | jondot imho you need a 3rd party software like festival for it |
| 08:21.29 | IsUp | michel2019: whats your goal? |
| 08:22.14 | jondot | schmidts, i understand google uses eSpeak (not festival) |
| 08:22.33 | jondot | they just have a good quality of a voice that i don't get when i jsut sudo apt-get install espeak |
| 08:23.01 | schmidts | jondot hmm maybe this is because google does a little bit more than install espeak;) |
| 08:23.09 | jondot | they do? |
| 08:23.38 | jondot | or in other question - since eSpeak is open source, do we (small people) get a chance to see the improvement they did? |
| 08:23.40 | schmidts | jondot maybe they allready have bought the company which made espeak :D |
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| 08:30.22 | michel2019 | IsUp, I just need to know which is the destination number sent to carrier....As each call has 2 channels opened...In Outgoing channel I need to know the number sent to carrier |
| 08:32.58 | coppice | jondot: have you tried the open source espeak quality? |
| 08:33.10 | jondot | yes, i'm only trying espeak now |
| 08:33.24 | jondot | the quality is not natural at all |
| 08:33.32 | coppice | it sounds pretty horrible, doesn't it? |
| 08:33.51 | jondot | with festival i was able to download HMMS voices which are a bit more natural, but i don't trust festival on the server side so much. it feels heavy |
| 08:34.03 | jondot | i also saw there is considerable lag when using asterisk (from other people) |
| 08:34.08 | coppice | it has broad language coverage, but I don't see much else that's positive about it |
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| 08:34.25 | jondot | exactly. i've been researching TTS like a dog for the last couple of days. |
| 08:35.20 | jondot | so far the most natural is festival + half a day of tweaking, or AT&T natural voice which is a dedicated system - linux version is not even for sale for server side |
| 08:35.36 | *** join/#asterisk beaver_rrr (beaver@crabhost.org) |
| 08:36.03 | beaver_rrr | why can asterisk tell me "603 undefined reason" as sip registration error? |
| 08:36.44 | jondot | coppice, by the way MARY (java TTS) is excellent, but very little information about linux installation / integration |
| 08:37.26 | wdoekes2 | beaver_rrr: afaics, asterisk 1.8 doesn't |
| 08:37.34 | coppice | the sound quality of most of these packages has more to do with the quality of the voice model than the software itself |
| 08:37.56 | jondot | thats true |
| 08:38.25 | wdoekes2 | beaver_rrr: is it your asterisk registering somewhere? in which case it's the "somewhere" that's telling your 603 |
| 08:39.17 | beaver_rrr | oh yes, it registers on provider's server |
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| 08:41.02 | wdoekes2 | do you get a 401 from them first? |
| 08:42.05 | beaver_rrr | nope |
| 08:42.57 | wdoekes2 | pb your register line (without the secret, optionally with IPs scrambled) and the sip conversation (optionally with IPs scambled) |
| 08:43.01 | beaver_rrr | http://pastebin.com/JkDtYKtF |
| 08:43.07 | beaver_rrr | that's what i get |
| 08:43.43 | beaver_rrr | http://pastebin.com/1xUMymqP |
| 08:43.51 | beaver_rrr | that's register line |
| 08:44.16 | wdoekes2 | and if you replace suntel with 160xxx ? |
| 08:46.01 | wdoekes2 | (and it seems odd that asterisk would call a 603 a timeout.. so I'm wondering if those are two separate issues) |
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| 08:47.27 | michael-i | I know this goes against all beautiful design but: Is there a way to execute AMI commands via extensions.conf? Just trying to get some proof-of-concept SIPNotify messages working and need to construct them dynamically. |
| 08:47.51 | beaver_rrr | i suppose NAT problems here, also dont know how to deal with dynamic external ip |
| 08:49.25 | IsUp | beaver_rrr: enable 'sip debug' and do 'sip reload' |
| 08:49.56 | IsUp | beaver_rrr: and make sure your 'register =>' line is under [general], not your peer. |
| 08:50.52 | wdoekes2 | michael-i: if you can execute AMI from the command line, you can do so from a System() call |
| 08:51.25 | beaver_rrr | it is under [general] |
| 08:51.58 | IsUp | beaver_rrr: okay, do sip debug. and pastebin output |
| 08:52.17 | michael-i | wdoekes2: yup…just looking for a generic dial plan application which takes a nice long string and drops it into the AMI |
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| 08:56.12 | beaver_rrr | IsUp: not sure if copied everything, but here > http://pastebin.com/SZwkg8Hx |
| 08:58.29 | jzaw | morn |
| 09:00.05 | IsUp | beaver_rrr: sounds like a NAT issue. try with setting 'externip=' and do 'sip reload' |
| 09:02.38 | IsUp | beaver_rrr: sorry, it's 'externaddr' in 1.8. also you have to set 'localnet=' |
| 09:03.11 | beaver_rrr | im not sure, what to do if i have dynamic external ip. Im using no-ip service to access, but not sure if it gonna work with asterisk |
| 09:04.05 | kaldemar | michael-i: you could also use an AGI to execute the sip notify CLI command and configure your notifications in sip_notify.conf |
| 09:04.36 | IsUp | beaver_rrr: 'externhost/externrefresh' could be a solution in your case |
| 09:04.56 | beaver_rrr | ok, thanks, gonna try :) |
| 09:05.30 | michael-i | kaldemar: also true…but you missed the "dynamic" part. I need to fire off messages that are call specific and not a known set. Firing off from System() to call sip notify's already works. Just not dynamic messaging |
| 09:05.37 | michael-i | s/messaging/messages |
| 09:06.54 | michael-i | If I could do dialplan variable substitution in those messages in sip_notify.conf my problem would also be solved. |
| 09:08.44 | IsUp | kaldemar: does grandstream supports "notifications" for rebooting or anything? |
| 09:10.37 | kaldemar | IsUp: no idea. |
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| 09:21.02 | IsUp | hm ive been testing sip notify on grandstream. it returns "401 Unauthorized" when i sent a ootify packet |
| 09:22.49 | IsUp | *notify |
| 09:24.22 | wdoekes2 | michael-i: how do you mean dynamic messages do not work? you wanted to do AMI from the dialplan, with System you can. and there you can supply all the variables you need. it's not clean, but you said you wanted proof-of-concept |
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| 09:46.34 | michael-i | wdoekes2: with System I can do CLI, not AMI…unless you mean something else |
| 09:55.22 | wdoekes2 | I mean that with System you can do anything |
| 09:55.24 | *** join/#asterisk gego (~gego@b238085.customer.hansenet.de) |
| 09:56.07 | wdoekes2 | as long as you have the right apps/scripts (in this case, a script doing ami for you) |
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| 10:01.53 | michael-i | wdoekes2: gotcha, yeah..I've been looking at BAMI ;) bash ami on github. Could be my quick fix |
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| 11:17.35 | IsUp | when i sent NOTIFY to phone, it returns 'Unauthorized'. looks like phone need auth. again to accept NOTIFY packet. how can i auth for re-sending packet? |
| 11:17.46 | IsUp | im using 'sip notify' in CLI |
| 11:18.29 | EmleyMoor | Is there a way I can remove a "keyless" entry from the astdb? I have accidentally managed to add one |
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| 11:21.08 | irroot | EmleyMoor using v10 ?? you can use sqlite command line tool once asterisk is stoped |
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| 11:21.49 | EmleyMoor | No, 1.6.2.9 |
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| 11:31.53 | EmleyMoor | Resolved for now but any hints to make it easier should I need to again would be appreciated |
| 11:34.02 | EmleyMoor | b |
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| 14:04.57 | Katty | HAI GUYS |
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| 14:05.47 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 14:06.03 | schmidts | Hi Katty |
| 14:06.13 | _Corey_ | Katty: reboot, repeat as required |
| 14:06.29 | Katty | we need more original answers. |
| 14:07.12 | _Corey_ | *yawn* ... not on one cup of coffee ;) |
| 14:07.25 | schmidts | Katty this is cause you are wearing nylon stockings, the electric voltage from them, jump over to your network cable and make asterisk work wrong |
| 14:07.30 | Katty | applies need moar coffee sign to _Corey_ |
| 14:08.23 | Katty | schmidts: *hee* |
| 14:09.14 | schmidts | Katty hehe its not my fault BOFH told me to say this |
| 14:09.27 | [TK]D-Fender | EJUMACATION! I HAZ! |
| 14:10.03 | [TK]D-Fender | Hrm, minor targeting failure... |
| 14:11.32 | Katty | that's ok. |
| 14:11.52 | schmidts | Katty what about this: new guy cross-connected phone lines with ac power bus. |
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| 14:13.38 | _Corey_ | schmidts: I had a customer plug their POTS lines into their network switch once... :) |
| 14:14.09 | _Corey_ | (because it's all VoIP after all...) |
| 14:15.23 | schmidts | _Corey_ remember me of a customer which connected the port from his ATA (RJ11 jack) to the DSL Uplink of his router (RJ 11 Jack) and also the network cable and he wondered why this doesnt work ;) |
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| 14:16.58 | _Corey_ | :) |
| 14:17.30 | Katty | schmidts: i think you assume i know more about hardware than i do |
| 14:17.35 | Katty | schmidts: i'm much more of a software person |
| 14:17.47 | Katty | schmidts: however connecting massive ammounts of power to something doesn't /seem/ like a good idea. |
| 14:18.17 | schmidts | Katty i was talking about this: http://pgl.yoyo.org/bofh/ |
| 14:18.30 | schmidts | thinks Katty didnt know BOFH? :D |
| 14:18.45 | Katty | i know about BOFH |
| 14:19.03 | schmidts | ok ;) |
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| 14:32.05 | Katty | who is in the know about Orlando |
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| 14:47.42 | n3hxs | Orlando... somewhere in FL |
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| 15:28.12 | raden | morning Katty (: |
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| 15:39.21 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:40.21 | asteriskATmarmuD | hi guys I am trying to set up music on hold for meet me rooms. I know the option "M" but how do I set the music on hold file. will the "default" class be played? |
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| 15:42.44 | Katty | hi raden |
| 15:42.56 | raden | hugs Katty |
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| 15:46.31 | raden | hows Katty |
| 15:48.24 | [TK]D-Fender | asteriskATmarmuD, "core show application meetme" |
| 15:49.24 | asteriskATmarmuD | [TK]D-Fender: 'M' -- enable music on hold when the conference has a single caller - I know that |
| 15:49.38 | [TK]D-Fender | asteriskATmarmuD, read it again |
| 15:49.40 | asteriskATmarmuD | [TK]D-Fender: but how is the specific file determined? |
| 15:50.17 | *** join/#asterisk BuenGenio (~Gene@203.145.92.214) |
| 15:50.20 | [TK]D-Fender | asteriskATmarmuD, " M[(class)]: Enable music on hold when the conference has a single caller. Optionally, specify a musiconhold class to use. If one is not provided, it will use the channel's currently set music class, or 'default'. |
| 15:50.58 | [TK]D-Fender | asteriskATmarmuD, and it's never a "file", it's always a class. |
| 15:51.00 | asteriskATmarmuD | [TK]D-Fender: thanks, I don't get this output |
| 15:51.54 | asteriskATmarmuD | [TK]D-Fender: man you're pedantic, but absolutely right! THX! |
| 15:53.44 | [TK]D-Fender | asteriskATmarmuD, What version of * are you using, and what was it installed from? |
| 15:54.34 | asteriskATmarmuD | [TK]D-Fender: Connected to Asterisk 1.4.39.1 |
| 15:54.56 | asteriskATmarmuD | [TK]D-Fender: I know, not the current one :) |
| 15:55.13 | [TK]D-Fender | Still not current for its very dated branch, and an odd release |
| 15:55.53 | asteriskATmarmuD | [TK]D-Fender: I am not proud of it eigther, but we got our reasons ;) |
| 15:56.39 | asteriskATmarmuD | [TK]D-Fender: I don't know it app meetme supports"M[(class)]" already |
| 15:57.00 | [TK]D-Fender | Should... |
| 15:58.29 | *** join/#asterisk Diffen (~diffen@c-a27ce555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 16:00.13 | Diffen | Evning, im trying to start asterisk after i have done a restart now in the asterisk cli. unfortunately asterisk doesnt want to start. the message i get when im running asterisk -vr are : Unable to open pid file '/var/run/asterisk/asterisk.pid': Read-only file system |
| 16:00.23 | Diffen | im in as a sudoer... |
| 16:03.27 | WIntermeW | omg..the freeswitch dialplan is a pain in the ass for those previously used to *.. :s |
| 16:03.51 | asteriskATmarmuD | :D |
| 16:03.56 | leifmadsen | welcome to XML I suppose |
| 16:04.07 | WIntermeW | yep :D |
| 16:04.45 | WIMPy | ouch |
| 16:05.05 | [TK]D-Fender | Diffen, the entire filesystem is read-only. You're system seems to be in somewhat critical shape... |
| 16:08.34 | Diffen | tk i guess so. there have been a big problem at the datacenter for two hours now |
| 16:08.49 | Diffen | and when i could connect to it again i couldnt start anything. |
| 16:09.09 | WIMPy | Repair your FS. |
| 16:09.12 | Diffen | i even tried to reboot the whole server by typing shutdown -r now but nothing happends |
| 16:09.15 | Diffen | oh man |
| 16:09.19 | Diffen | wimpy dont say that :D |
| 16:09.36 | WIMPy | You have to. |
| 16:09.45 | [TK]D-Fender | Pul plug. Reinsert plug. Turn back on. |
| 16:10.24 | WIMPy | That's where the magig SysRq key comes handy ... If you've got a keyboard. |
| 16:10.57 | [TK]D-Fender | "Datacenter" |
| 16:11.00 | Diffen | :D |
| 16:11.17 | WIMPy | IP KVM? ILO? Whatever. |
| 16:11.40 | Diffen | nothing for me. but i guess that the datacenter have that posibility |
| 16:14.39 | Diffen | crap. i wonder what happend at the datacenter…. nothing has been written in any logfiles for two hours... |
| 16:14.43 | Diffen | not good |
| 16:15.54 | [TK]D-Fender | Diffen, "Read-Only Filesystem" may have something to do with that. |
| 16:15.58 | ChannelZ | Are they on a UPS (or your machine?) |
| 16:16.07 | Diffen | true :D |
| 16:16.08 | [TK]D-Fender | Diffen, If you can't remount it live, then you'll need to reboot |
| 16:16.13 | ChannelZ | Probably took a power hit, corrupted a filesystem, and it rebooted read-only |
| 16:16.28 | Diffen | ive told them to reboot it. |
| 16:16.35 | Diffen | do you guys know what may cause that? |
| 16:16.42 | ChannelZ | I just said |
| 16:16.43 | Diffen | like in big terms |
| 16:17.12 | _Corey_ | "big" terms? |
| 16:17.31 | ChannelZ | Big picture: Your server is broken |
| 16:17.40 | Diffen | ok :D |
| 16:17.58 | ChannelZ | type "reboot" and see what happens |
| 16:18.20 | ChannelZ | but to know for sure you've probably got to be there looking at the screen when it boots |
| 16:19.23 | Diffen | i really hope that this will be solved on a reboot |
| 16:19.25 | Diffen | :D |
| 16:19.37 | Diffen | nothing happends when i type reboot |
| 16:19.57 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:19.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 16:20.24 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 16:22.16 | ChannelZ | there's always "init 6" |
| 16:23.20 | Diffen | init 6? |
| 16:23.33 | WIMPy | More like reboot -df |
| 16:23.45 | WIMPy | Are you sure the box rebooted before? |
| 16:23.49 | ChannelZ | what is -d ? |
| 16:23.54 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 16:24.02 | WIMPy | Don't write wtmp. |
| 16:24.11 | Diffen | the uptime for it was 264 days or so… but i have rebooted it before |
| 16:24.14 | WIMPy | Wouldn't work on a RO FS. |
| 16:24.41 | WIMPy | Sounds like an remergency remount,ro then. |
| 16:24.55 | Diffen | ... |
| 16:24.57 | Diffen | ugh |
| 16:31.36 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 16:31.53 | sereal-work | in asterisk 10, what's the setCallerID function? |
| 16:32.14 | sereal-work | seems like they got rid of it |
| 16:32.38 | WIMPy | 'core show function CALLERID' |
| 16:32.44 | ChannelZ | Set(CALLERID(num)=12345) etc |
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| 16:33.22 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 16:33.26 | [TK]D-Fender | sereal-work, "SetCallerID was deprecated in * 1.2 |
| 16:33.36 | [TK]D-Fender | sereal-work, Welcome to 2005 |
| 16:33.39 | [TK]D-Fender | :p |
| 16:34.21 | sereal-work | really? what should I be using now? |
| 16:34.28 | sereal-work | (my asterisk 1.4 system still used it) |
| 16:35.12 | [TK]D-Fender | sereal-work, "core show function CALLERID" as you've been told by others. |
| 16:35.27 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:35.47 | sereal-work | ugh, sorry i'm stupid - I just saw WIMPy and ChannelZ message |
| 16:35.54 | sereal-work | guess I need more coffee :P |
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| 16:43.28 | Diffen | if anyone is intersted the datacenter rebooted the server and got this message. /dev/VolGroup00/LogVol00: UNEXPECTED INCONSISTENCY; RUN fsck MANUALLY. |
| 16:44.45 | WIMPy | That was expected. |
| 16:45.34 | Diffen | ok |
| 16:45.44 | Diffen | ill hope im in your prayers tonight :D |
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| 17:04.22 | *** join/#asterisk niros (~nir@bzq-218-196-74.red.bezeqint.net) |
| 17:05.16 | niros | Helllo,is it possible to disable MANAGER Commands to queuelog ? (e.g 1326733379|MANAGER|1001|SIP/501|REMOVEMEMBER) ? |
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| 17:26.56 | ChannelZ | Diffen: ....so did you? |
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| 17:34.55 | Diffen | channelz its rebooted and they say its ongoing service... |
| 17:35.35 | Diffen | but im downloading the backup from a remote site and so if i have to i will restore that on a different server |
| 17:38.06 | cmendes0101 | If I add port= to a user in sip.conf. Should sip show peers show that same port? |
| 17:38.38 | pdtpatrick | Question -- anyone seen this error before? it is affecting voicemails. Thanks |
| 17:38.45 | pdtpatrick | http://paste.pocoo.org/show/535744/ |
| 17:44.05 | sereal-work | what does failed to extend from 1024 to 1306 mean? |
| 17:48.15 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 17:48.28 | nickfennell | hi all. Anyone know of an issue where asterisk isn't clearing extensions in-use |
| 17:48.38 | sereal-work | the last two digits of 13 change |
| 17:49.58 | p3nguin | cmendes0101: If the type is user, sip show peers will not show the device at all. |
| 17:50.35 | nickfennell | seems to be this issue, https://issues.asterisk.org/view.php?id=1031 |
| 17:56.25 | Buuyo | Hi all. What could cause a bit of a low volume squealing and swishing sound to come out of an asterisk setup with a Wildcard TDM410P? I can't usually hear it over a dialtone, but when I'm in a menu where it's mostly quiet, or in a call it's always there, and quite noticeable. |
| 17:57.07 | Buuyo | It's not present over SIP. Just through the analog ports. |
| 17:57.31 | nickfennell | makes a guess at shared IRQ |
| 17:59.13 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 18:01.36 | Buuyo | A shared IRQ would do that? I definitely see something using the same IRQ. |
| 18:02.00 | nickfennell | I can't say that it would but I'm sure it could. |
| 18:02.33 | nickfennell | If you've got a lot of low level interference from something else on the system then it could bleed through. I guess it depends on the quality of the motherboard |
| 18:02.42 | Buuyo | But I guess the theme here is that it's the card in one way or another. It might not help that I got it for cheap from ebay. :) |
| 18:02.58 | nickfennell | Yeah, you may have just found out why it was cheap and on eBay ;) |
| 18:03.05 | Buuyo | heh |
| 18:03.25 | nickfennell | Is it an FXO card? |
| 18:03.45 | Buuyo | It's got those expandable modules. I've got 2 FXS and 2 FXO plugged in. |
| 18:03.52 | nickfennell | oh right |
| 18:04.00 | nickfennell | do you get noise when you call between the FXS ports? |
| 18:05.02 | *** join/#asterisk serxoz (~serxoz@21.169.20.95.dynamic.jazztel.es) |
| 18:05.24 | Buuyo | not sure. i'll have to check. |
| 18:05.28 | serxoz | hi! |
| 18:05.56 | *** join/#asterisk neurosys (~neurosys@69.198.141.150) |
| 18:06.51 | serxoz | somebody knows how to detect busy tone from operator mundo-r from spain? i've a spa 3102 and it doesn't detect when pstn hangup? :S |
| 18:07.00 | Buuyo | nickfennell: nope |
| 18:07.29 | nickfennell | no noise on FXS? |
| 18:07.34 | Buuyo | no noise |
| 18:08.04 | Buuyo | oh look it just kicked in :| |
| 18:08.17 | Buuyo | I think I've just discovered that it's intermittent |
| 18:08.32 | nickfennell | lol |
| 18:08.41 | nickfennell | You've also discovered that it's nothing to do with the FXO line |
| 18:08.46 | Buuyo | darn cheap ebay hardware. :) |
| 18:08.51 | nickfennell | How much it cost you? |
| 18:08.54 | Buuyo | 60 bucks |
| 18:09.09 | Buuyo | from hong kong no less |
| 18:09.18 | Buuyo | so it's probably a faulty counterfeit |
| 18:10.04 | Buuyo | But I'll see if I can put it on its own irq |
| 18:10.51 | *** join/#asterisk binbash_ (~peter@a80-127-248-129.adsl.xs4all.nl) |
| 18:12.57 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 18:14.36 | serxoz | sipura spa 3102 are getting me crazy :S all calls marked as answered? |
| 18:15.11 | p3nguin | Calls coming in on the FXO port? |
| 18:16.30 | *** join/#asterisk mahlon (mahlon@martini.nu) |
| 18:17.31 | serxoz | no, calls from voip to pstn |
| 18:17.48 | serxoz | seems it doesn't detect busy tone |
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| 18:18.31 | ChannelZ | analog usually doesn't |
| 18:19.13 | serxoz | and this operator doesn't send reversal polarity... |
| 18:19.45 | serxoz | it's a pity cause i've got 60 minutes to call mobile for free |
| 18:20.10 | serxoz | and if all calls are answered i can't know time consumed |
| 18:20.43 | serxoz | it's a home pbx :D |
| 18:23.49 | *** join/#asterisk cyford (~allen@184-77-45-149.gar.clearwire-wmx.net) |
| 18:25.03 | serxoz | i'm going to buy an axe, see you! |
| 18:25.21 | *** part/#asterisk serxoz (~serxoz@21.169.20.95.dynamic.jazztel.es) |
| 18:25.34 | ChannelZ | enjoy |
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| 19:00.48 | pdtpatrick | Question -- if I'm going to install asterisk 1.8 from the repo (debian), currently the wiki shows that the only dependency is python-software-properties. But if i want things like pgsql for CDR and res_jabber, do i have to install the dependencies myself and then when i install the package from the repo, it will automatically install those modules as long as the dependencies are satisfied? |
| 19:01.56 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 19:02.55 | Goldwing | Q: i have a asterisk 10.0.0 test setup, and i gave one extension voice mail. Called the extension and recorded a voicemail, now when i listen to the voicemail i should be able to delete it by pressing 7, but it doesn't delete it... |
| 19:05.32 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
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| 19:19.23 | pabelanger | pdtpatrick: most dependencies are already defined within the asterisk package, anything in the addons folder should be an additional package. EG: asterisk-mysql |
| 19:19.57 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 19:21.13 | pabelanger | pdtpatrick: you can use apt-rdepends to check them out. |
| 19:21.52 | pdtpatrick | this is all I'm seeing in the repo |
| 19:21.52 | pdtpatrick | http://paste.pocoo.org/show/535796/ |
| 19:23.00 | pabelanger | right, asterisk is the main package. The other packages just break out some functionality from it |
| 19:25.55 | pdtpatrick | http://paste.ubuntu.com/806555/ |
| 19:26.08 | pdtpatrick | don't see anything jabber related or postgresql |
| 19:27.02 | pdtpatrick | nvm |
| 19:27.05 | pdtpatrick | it is in asterisk-modules |
| 19:27.08 | pdtpatrick | http://paste.ubuntu.com/806556/ |
| 19:27.27 | pdtpatrick | pabelanger: good stuff man. Thanks a million |
| 19:28.10 | pabelanger | pdtpatrick: ya, I forgot tzafrir_laptop moved them into the new package |
| 19:29.16 | pdtpatrick | oh snap, there's now a Python API for the AMI .. I'm impressed :) |
| 19:29.22 | pdtpatrick | python-asterisk - Asterisk Manager API interface module for Python |
| 19:30.09 | pabelanger | yup, I believe it is starpy |
| 19:30.27 | pabelanger | maybe not |
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| 19:33.16 | pdtpatrick | oh man good stuff, u just cut my time in half :) |
| 19:35.58 | *** join/#asterisk mahables (~Jason@c-76-101-131-88.hsd1.fl.comcast.net) |
| 19:36.41 | mahables | Anyone care to chat about a queue issue in 1.4.42? |
| 19:39.12 | mahables | ...issue has to do with, don't laugh, local channels and device state. |
| 19:40.35 | *** join/#asterisk cyford (~allen@184-77-45-149.gar.clearwire-wmx.net) |
| 19:48.15 | pabelanger | ~ask |
| 19:48.15 | 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. |
| 19:48.20 | pabelanger | mahables: ^ |
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| 20:07.21 | mahables | Sorry guys :-) Anyways, to be more specific, I'm running into an issue with device state on local channels in queues... if I add a member in a queue like "queue add member LOCAL/200@toremote/n to 12000 penalty 1 as 200 state_interface LOCAL/200@toremote/n" this will sometimes result in the member receiving two calls at once from the same queue. This won't happen until a load of approximately 20-30 simultaneous calls. |
| 20:10.26 | mahables | Also, there are times where wrapup is ignored, and a member gets a call immediately following the previous call from the same queue. |
| 20:14.08 | mahables | If I change the state_interface to SIP/200_R, the double call issue does not occurr, but wrapup time is still ignored at times. |
| 20:20.02 | mahables | I was kinda just hoping to find someone that was willing to give me some insight on device state in 1.4 |
| 20:22.29 | *** part/#asterisk Buuyo (electrum@ieee1003.1-1988.posi.xxx) |
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| 20:23.23 | *** join/#asterisk twanny796 (~twanny@46.11.98.111) |
| 20:23.47 | drmessano | Anyone have app_swift running in Asterisk 10? |
| 20:25.44 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 20:26.24 | *** join/#asterisk FainaUkraina (~Gene@203.145.92.214) |
| 20:37.11 | k-man | I'm after a desk phone to use with asterisk, something thats quite nice and I could pick up cheaply on ebay, any recommendations? I needs to be easy to configure via a web interface. i've tried the cisco tftp approach and its too difficult |
| 20:40.52 | pabelanger | ~polycom |
| 20:40.52 | infobot | methinks polycom is The Polycom Song by dialing sip:polycom@leifmadsen.com or ISN 7659*460. Polycom phone are devices that are favoured by much of the community and range in price from under $100 and upwards. |
| 20:41.04 | pabelanger | ~cheap |
| 20:41.05 | infobot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:41.40 | leifmadsen | k-man: don't use a hard phone then -- use a soft phone |
| 20:42.42 | k-man | thanks |
| 20:42.57 | eZz | a soon will be mind phones |
| 20:43.03 | k-man | leifmadsen, imho a soft phone is just not as nice as a hard phone |
| 20:43.17 | leifmadsen | agreed |
| 20:43.17 | eZz | no need an additional equipment |
| 20:43.17 | leifmadsen | but it's better than a crappy hard phone |
| 20:43.18 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 20:43.18 | *** mode/#asterisk [+o file] by ChanServ |
| 20:43.22 | eZz | :-D |
| 20:43.37 | leifmadsen | k-man: don't cheap out -- make sure you get something like a Polycom IP335 at a minimum |
| 20:43.44 | leifmadsen | if cost is an issue, then just stick with the softphones |
| 20:43.49 | k-man | leifmadsen, I don't mind spending a few dollars, I bought a cisco 7942 which seems nice but configuring it is beyond the time I have to invest in it |
| 20:43.55 | leifmadsen | or get an ATA adapter and use an analog phone |
| 20:44.15 | eZz | brr |
| 20:44.32 | leifmadsen | k-man: ya don't use Cisco phones because they are designed to be used with Cisco systems that generate the configs |
| 20:45.26 | k-man | yes as I discovered :) |
| 20:45.39 | ChannelZ | We are the network! |
| 20:45.51 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 20:45.54 | ChannelZ | or whatever that dopey tagline was. |
| 20:46.25 | F2Knight | Q: Woke up today and Jabber was having connection issues to Google Voice.. anyone else with problem? |
| 20:46.38 | F2Knight | res_jabber.c:3112 aji_initialize: JABBER ERROR: No Connection |
| 20:47.05 | ChannelZ | I had a few of those at work but not at home |
| 20:47.23 | ChannelZ | but I had a shit-ton of them at home yesterday. Not sure if my cable went out or what, it was at 5am |
| 20:47.29 | jzaw | re jabber |
| 20:47.36 | ChannelZ | it happens from time to time regardless |
| 20:47.40 | jzaw | i have the same problem but with my own xmpp server |
| 20:47.51 | jzaw | i can connect and talk via google voice - * np |
| 20:48.22 | jzaw | but not my own ejabberd |
| 20:48.33 | F2Knight | ChannelZ, happens... okay but this has been happening for hours... um.. The bigger problem is attempting to dial over it causes asterisk to segafault and reload |
| 20:48.37 | jzaw | could someone discuss with me to help me trace the issue? |
| 20:49.27 | jzaw | F2Knight: one thing that instantly segfaults my * with respect to jabber.conf is ipv6 |
| 20:50.21 | jzaw | i either have to put in an ipv4 ip or make sure the fdqn doesnt resolve to an ipv6 ip |
| 20:50.32 | jzaw | ie i strip out the ipv6 ip from my dns |
| 20:51.42 | ChannelZ | hmm I've never had mine crash |
| 20:51.50 | F2Knight | jzaw, i don't have any IP6 settings defined... heck for that matter whos giving you an IP 6 address ? :) |
| 20:52.07 | jzaw | my ips .... the best :P |
| 20:52.15 | jzaw | s/ips/isp |
| 20:52.33 | F2Knight | suppose so :) I got an IP6 address when i connect to Hamachi lol |
| 20:52.56 | jzaw | ive got a /64 but can have as many of those as i like |
| 20:53.37 | jzaw | ive got a /29 ipv4 and could have pretty much any more i like ... as long as i can justify it |
| 20:53.56 | k-man | what about the polycom ip330? |
| 20:55.26 | F2Knight | that reminds me... I just added hamachi to that server yesterday.. let me turn it off see if that has anything to do with it. |
| 20:55.27 | *** join/#asterisk frinnst (~misfit@unaffiliated/frinnst) |
| 20:56.14 | F2Knight | jzaw, I have a whole class C block but of IP-4, that was a pain in the ass to get. |
| 20:58.34 | *** join/#asterisk Mahmoud (~foo@unaffiliated/mahmoud) |
| 20:58.37 | F2Knight | SOLVED: sorta.... seems when connecting the server over hamachi Jabber does not want to load. |
| 20:59.04 | Mahmoud | can we have asterisk vs. freeswitch? i've been away since freeswitch's epic victories. wondering if anything has changed the balance? |
| 21:00.13 | [TK]D-Fender | k-man, Where are you located? |
| 21:00.37 | [TK]D-Fender | Mahmoud, I don't remember a fight let alone a "winner", let alone "epic" |
| 21:00.49 | [TK]D-Fender | Mahmoud, Check your FUD at the door please.. |
| 21:00.54 | F2Knight | Mahmoud, FreeSWITCH always sounded nice... but the leaning curve and XML configs always kept me away from it., |
| 21:01.18 | Mahmoud | xml is one reason i went to freeSWITCH |
| 21:01.38 | k-man | [TK]D-Fender, australia |
| 21:01.55 | F2Knight | I would be interested in knowing why? XML has always seemed to me to be a lot of extra overhead, and a real pain to edit by hand |
| 21:02.01 | Mahmoud | i was wondering if there was major arch changes in asterisk since then? |
| 21:02.02 | [TK]D-Fender | k-man, Go for the Cisco/Link SPA series then. Far more cost effective in your areal |
| 21:02.07 | [TK]D-Fender | area* |
| 21:02.23 | k-man | [TK]D-Fender, ah ok - I already have one spa942 which is not too bad |
| 21:02.36 | Mahmoud | F2Knight: xml is meant to be used by other software imo. good to have that flexibility |
| 21:02.37 | [TK]D-Fender | Mahmoud, We don't know "then" was for you, an no, not "major", several incremental |
| 21:02.38 | k-man | thanks for the pointer [TK]D-Fender |
| 21:02.53 | Mahmoud | [TK]D-Fender: then == 2-3 years |
| 21:03.27 | [TK]D-Fender | Mahmoud, Nothing "huge", but 2 whole new branches came out. |
| 21:03.46 | drmessano | I don't recall any epic victories by FreeSWITCH, but whatever |
| 21:04.32 | F2Knight | Mahmoud, I guess you could say that Asterisk is more mature and has a more active community, and is by far still more popular PBX, software out there. |
| 21:04.46 | jzaw | [20:58:37] <F2Knight> SOLVED: sorta.... seems when connecting the server over hamachi Jabber does not want to load. |
| 21:04.46 | jzaw | ipv6 issue eh? |
| 21:04.55 | phix | Hey gang! |
| 21:06.04 | F2Knight | but as for the XML, no one has been able to provide a convinsing reason to me... other then what seems like personal reasons. I fail to see the great magic and love of it. but would like to understand and see,, then maybe I would appreciate it more. |
| 21:06.17 | F2Knight | jzaw, not sure.. going to try to reeneable with IP4 only |
| 21:06.19 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:07.22 | jzaw | still doesnt give me a clue about whats up with mine |
| 21:07.30 | drmessano | I only saw the use of XML as something else to add to a feature list containing popular buzzwords. Every argument of Freeswitch vs Asterisk seems to include "Yeah, but FS uses XML for its configs".. but when asked "Ok, so how is that better?" theres crickets |
| 21:07.43 | jzaw | though my ejabber is listening on ipv6 so i might have to remove that |
| 21:09.15 | F2Knight | drmessano, thats kinda the way I feel, and Well I don't get the reason for XML anyways, as to me json seems better, XML to me seems like a lot of bloat. |
| 21:09.35 | jzaw | so hard to read sometimes |
| 21:09.44 | F2Knight | One thing I can say for FreeSWITCH over Asterisk, is that is uses sophia (sp*) for the SIP stack. |
| 21:09.45 | k-man | [TK]D-Fender, out of interest, what is it about the linksys spa phones that make then better for .au? |
| 21:09.50 | F2Knight | and not a hack up one that asterisk has |
| 21:09.58 | [TK]D-Fender | k-man, Read what I said again.. |
| 21:10.11 | k-man | oh- cost effective! thanks [TK]D-Fender |
| 21:10.19 | [TK]D-Fender | chan_bigprint.so :) |
| 21:10.47 | [TK]D-Fender | Polycom > All ... and in AU is hit with a disproportionate import tax |
| 21:10.59 | drmessano | Asterisk's SIP stack has gotten a lot better in the last few releases |
| 21:12.09 | [TK]D-Fender | In North America there is little reason to choose anything other than Polycom, elsewhere.. you may find the reverse. The SPA series is web configurable, and provisionable, stable, acceptable build & firmware quality and workable featureset |
| 21:16.25 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 21:17.01 | Cubber | is there a new way of using festival with asterisk? using the Festival() command in extensions.conf does not seem to work anymore |
| 21:17.34 | F2Knight | jzaw, seems that was the issue. Started it up with only ipv4 support |
| 21:17.47 | F2Knight | seems to work well now, able to make calls again. |
| 21:17.54 | jzaw | :) |
| 21:18.38 | jzaw | ive re established bindipv6only and removed inet6 in the listen statement in ejabberd.cfg |
| 21:18.47 | jzaw | and now * is segaulting |
| 21:21.13 | jzaw | +1 for freeswitch |
| 21:21.23 | jzaw | freeswitch does ipv6 :/ |
| 21:21.41 | drmessano | Which version of Asterisk? |
| 21:22.02 | k-man | [TK]D-Fender, thanks for outlining that |
| 21:22.14 | jzaw | svn 10 |
| 21:23.04 | drmessano | Hmmm |
| 21:23.09 | jzaw | imsmr |
| 21:23.18 | drmessano | That's probably not a good idea |
| 21:23.29 | [TK]D-Fender | checkout time, BBIAB |
| 21:23.30 | jzaw | on my other box my syslog was telling me that pthreads was involved |
| 21:23.47 | *** join/#asterisk Diffen (~diffen@c-a27ce555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 21:23.58 | jzaw | drmessano: same was happing in svn 1.8 |
| 21:25.26 | drmessano | Do you have all your bindaddr statements locked down? |
| 21:25.35 | jzaw | i did |
| 21:25.41 | jzaw | ive been removing them |
| 21:25.46 | jzaw | ill put them back |
| 21:25.59 | jzaw | but its only when i involve jabber.conf |
| 21:26.11 | jzaw | if i set that up with gtalk.conf |
| 21:26.19 | jzaw | its fine i can voice talk |
| 21:27.35 | drmessano | ah |
| 21:29.23 | *** join/#asterisk cptcvman (~robert@cpe-174-098-127-210.triad.res.rr.com) |
| 21:29.39 | F2Knight | jzaw, any special reason for running svn? |
| 21:30.07 | jzaw | im running under debian squeeze |
| 21:30.22 | jzaw | and have never found the debs too stable/featureful |
| 21:30.24 | F2Knight | jzaw, are you trying to use ejabbere to provide gtalk support? |
| 21:30.50 | jzaw | i wanted to provide presence info |
| 21:31.05 | jzaw | and also to voice talk via my own _user_name_ |
| 21:31.11 | jzaw | rather than be tied to google |
| 21:31.13 | F2Knight | as for the stable debs, no they are not uptodate. however getting stable tarball is a better choice.. 1.8.8.1 currently, |
| 21:31.46 | F2Knight | I was having lots of Gtalk/Google Voice issues using the SVN of 1.8 and 10, dropped down to stable 1.8.8.0 and it worked right out of the box. |
| 21:31.47 | jzaw | i suppse svn tag eh? |
| 21:31.54 | drmessano | I've never had problems with SVN |
| 21:31.58 | jzaw | gtalk works np at all |
| 21:32.05 | jzaw | worked first time |
| 21:32.09 | drmessano | Every rare now and then a bad commit, but SVN is never a bad choice |
| 21:32.19 | jzaw | i wanted a working conf that i could simply _convert_ to my own |
| 21:32.26 | pabelanger | ~backtrace |
| 21:32.26 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:32.29 | drmessano | Tarballs are nothing but snapshots |
| 21:32.35 | pabelanger | jzaw: ^ |
| 21:32.35 | cptcvman | I'm having trouble getting Asterisk to use the queue_log. Is there some Asterisk CLI command I can use to view information about logging? |
| 21:33.05 | F2Knight | okay the Gtalk / Google Voice you have to use the jinggle/gtalk/jabber provided as from what I have been told Googles version is not fully compat with other software. |
| 21:33.17 | jzaw | pabelanger: ive never done a backtrace ... must admit it would help f someone could point me to a howto (backtrace) |
| 21:33.24 | jzaw | s/f/if |
| 21:33.53 | pabelanger | jzaw: I take it you did not read the URL that was linked? |
| 21:34.04 | jzaw | ah soz ... being blind |
| 21:34.25 | jzaw | perfect ... shall read |
| 21:34.26 | jzaw | thanks |
| 21:37.21 | *** join/#asterisk rjvvliet_ (~rjvvliet@178-85-122-187.dynamic.upc.nl) |
| 21:41.09 | cptcvman | Does asterisk 1.6.2 not use the queue_log anymore? |
| 21:44.00 | F2Knight | cptcvman, look at http://www.the-asterisk-book.com/unstable/applikationen-queuelog.html |
| 21:45.35 | cptcvman | Why the F2 key? |
| 21:45.44 | cptcvman | Or did I read the name incorrectly? |
| 21:51.18 | jzaw | drmessano: pabelanger F2Knight |
| 21:51.18 | jzaw | Program received signal SIGSEGV, Segmentation fault. |
| 21:51.18 | jzaw | [Switching to Thread 0x40b11470 (LWP 2062)] |
| 21:51.18 | jzaw | 0x404b64e4 in pthread_mutex_lock () from /lib/libpthread.so.0 |
| 21:51.53 | pabelanger | jzaw: pb the whole backtrace |
| 21:52.14 | jzaw | will do |
| 21:52.34 | F2Knight | ~pb |
| 21:52.34 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 21:52.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:53.59 | jzaw | damn i thought i compiled * with dont_optimise |
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| 21:55.32 | jzaw | say pabelanger when in gdb and its just segfaulted |
| 21:55.42 | *** join/#asterisk Eitan (~Eitan@12.192.84.98) |
| 21:55.46 | jzaw | and its back to the gdb prom how do i get it to write out the backtrace? |
| 21:56.17 | pabelanger | jzaw: all the information is in the URL |
| 21:57.10 | jzaw | i tried that |
| 21:57.11 | jzaw | gdb /usr/sbin/asterisk |
| 21:57.14 | jzaw | set args |
| 21:57.15 | jzaw | and run |
| 21:57.23 | jzaw | thought it might auto write out |
| 21:57.29 | jzaw | ill do it long hand then |
| 21:57.31 | jzaw | cheers |
| 22:00.01 | jzaw | http://pastebin.com/nFNqS4pM |
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| 22:15.08 | paulc | Does a goto clear the gosub stack? |
| 22:17.05 | [TK]D-Fender | no |
| 22:17.20 | [TK]D-Fender | Meatballs are ordered separately |
| 22:20.15 | leifmadsen | paulc: no, all information is maintained in memory -- you can see what is saved for the channel using DumpChan() |
| 22:20.47 | paulc | @leifmadsen: *doffs hat* thank you sir! |
| 22:20.56 | paulc | [TK]D-Fender: Mmmm Meatballs :) |
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| 22:39.10 | banditti | spa2102 or obi110 or???? |
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| 23:03.04 | pdtpatrick | Question -- why is asterisk-modules available for ubuntu and not for debian in the repo? Is it called something else for debian ? |
| 23:07.40 | pdtpatrick | nvm .. looks like they bundled it into asterisk package |
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| 23:19.46 | paulc | banditti: the Obi110's pretty nice.. and has FXO as well as FXS (which the 2102 doesn't, I don't think?) |
| 23:19.53 | F2Knight | pdtpatrick, yea Ubuntu's package breaks it all up. Would like to get a hold of a build script for making current deps though. :) |
| 23:20.02 | F2Knight | like the DKMS features |
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| 23:50.42 | phix | paulc: That an external device or a card? |
| 23:52.37 | paulc | phix: It's an external device. www.obihai.com for details - it's a 1 FXO 1 FXS ATA with 2 SIP registrations, also connects to ObiTalk for hop on/hop off within your "circle of trust", and supports Google Voice. It's pretty neat. |
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