| 00:32.11 | *** join/#asterisk italorossi (~Adium@187.61.182.39) |
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| 00:49.58 | ectospasm | hmmm, is there a way to record which side of a PRI call hangs up in the CDR? Essentially I'm looking for a way to record which side sends the DISCONNECT message. Using Asterisk 1.8.11-cert4 |
| 00:50.49 | WIMPy | That's not PRI specific. |
| 00:51.35 | WIMPy | You can use the option to Dial to continue in the dialplan if either side hangs up and then see if the next priority is hit to find out if it was the caller or the callee. |
| 00:53.18 | ectospasm | hmmm... which option to dial is that? |
| 00:53.47 | WIMPy | There is one for each side IIRC. |
| 00:54.15 | WIMPy | I use g |
| 00:56.15 | ectospasm | but that appears only for the destination channel, and according to this: http://forums.asterisk.org/viewtopic.php?p=177440 it may break custom CDR variables... |
| 01:00.27 | WIMPy | Unfortuntly it doesn't say what might not work there. |
| 01:01.04 | ectospasm | true enough... |
| 01:01.13 | WIMPy | I surely get to h for both channels anyway. |
| 01:05.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:05.55 | ectospasm | right, but how do ${HANGUPCAUSE} and ${DIALSTATUS} change based on which side hung up? |
| 01:06.07 | WIMPy | Not |
| 01:10.27 | ectospasm | I suppose I could set up a dialplan variable... if the destination hangs up, set the variable... in the hangup exten, I could see if that variable is set, and branch accordingly. |
| 01:11.25 | WIMPy | No. h is always executed. It's the next priority after Dial that is only executed when the allee hangs up. |
| 01:11.37 | WIMPy | With DIALSTATUS=ANSWER |
| 01:11.39 | ectospasm | No, I know 'h' is always executed |
| 01:12.23 | ectospasm | ...but if the 'g' dialplan option causes a certain variable to be set, 'h' can test whether it's set and then process accordingly... |
| 01:12.39 | WIMPy | Ah, that way. Yes. |
| 01:13.03 | WIMPy | misinterpreted the ... . |
| 01:13.35 | ectospasm | it's OK, thanks for helping me think this through. |
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| 01:15.43 | WIMPy | has been applying for a HANGUPLOCATION for some time. The cahnnel could be part of that. |
| 01:16.15 | ectospasm | is that pre-defined variable, or did you define it? |
| 01:16.32 | WIMPy | I think campaign is the right word. |
| 01:16.47 | WIMPy | It doesn't exist so far. |
| 01:17.18 | WIMPy | But the PSTN doesn't only give you a cause but also a location that may change the sense of the cause. |
| 01:19.49 | ectospasm | well, with PRI the direction is definitely known... I'm only worried about cause code 16 for now... |
| 01:20.07 | WIMPy | Off course it is known there. |
| 01:20.39 | WIMPy | Normally answered calls will end with 16. (or preferrably 0x10) |
| 01:21.18 | WIMPy | Other causes are possible, but I don;t know how widely they are implemented even if they's apply. |
| 01:22.36 | luckman212 | can someone tell me what the cel_tds.so module is useful for? |
| 01:23.02 | WIMPy | T write CEL to a MS-SQL. |
| 01:23.06 | WIMPy | To... |
| 01:23.11 | ectospasm | ...or Sybase... |
| 01:23.42 | luckman212 | ah... for some reason I just compiled a new trunk (1.8) build from svn, and it was kicking out errors about the conf of that module, never saw those before |
| 01:23.55 | luckman212 | So I disabled it in modules.conf -- no harm it seems, I wasn't using it |
| 01:24.50 | luckman212 | chan_jingle.so is another one that started whining, also disabled ... I think 'motif' is the replacement? but only for trunk/11? |
| 01:25.11 | WIMPy | Sounds correct. |
| 01:26.27 | luckman212 | is it normal for asterisk to cause the internal PC speaker "bell" to beep once when it starts up? |
| 01:26.51 | WIMPy | Hasn;t happened for me so far. |
| 01:26.57 | luckman212 | I always thought that indicated some type of error, but I have scoured through /var/log/asterisk/full and nothing jumps out |
| 01:27.01 | luckman212 | Hrmm |
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| 02:06.50 | luckman212 | Hmm.. I rebooted the box and it no longer beeps when stopping/starting. I think it was the res_curl module that wouldn't unload |
| 02:06.58 | luckman212 | I was getting some warnings about that in the logs |
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| 03:15.58 | ipengineer | Does anyone know if there is anything that can be done to the queue_log file to 'reset' the queue stats? I am running asterisk 1.4.29 the reload function doesn't reset the stats |
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| 03:34.36 | ChannelZ | queue reset stats ? |
| 03:34.58 | ChannelZ | (on the console) |
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| 04:13.52 | tedryder | Is there a web conference module or plugin/etc. for asterisk? |
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| 04:35.11 | tedryder | Is there a web conference module or plugin/etc. for asterisk? |
| 04:50.00 | carrar | When you finish writting one, please let us know |
| 04:50.33 | deo | [TK]D-Fender: what is it again the site you gave me yesterday? about som reading materials regarding asterisk?? thanks |
| 04:50.45 | deo | good day [TK]D-Fender |
| 04:50.47 | deo | ^ |
| 04:56.15 | jmls | ok - we have an extension that wants calls recording. easy to do with mixmonitor |
| 04:56.38 | jmls | a) which leg do you record on - A or B |
| 04:56.55 | jmls | I am now thinking of transfers. |
| 04:57.32 | jmls | A calls B. recording starts on A. B transfers call to C. no problem |
| 04:58.07 | jmls | A calls B. recording starts on A. A transfers call to C. what happens with the recording ? |
| 05:03.24 | tedryder | Well I know BigBlueButton use to use Asterisk |
| 05:03.32 | tedryder | They use FreeSwitch now |
| 05:03.51 | tedryder | I didn't know if I just was unaware of one |
| 05:07.05 | jmls | ahhhh. http://lists.digium.com/pipermail/asterisk-users/2011-August/265434.html |
| 05:11.55 | [TK]D-Fender | ~book |
| 05:11.55 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 05:11.58 | [TK]D-Fender | deo: ^^ |
| 05:13.50 | jmls | wait |
| 05:14.04 | jmls | doesn't fix the issue, according to http://www.mail-archive.com/asterisk-users@lists.digium.com/msg265380.html |
| 05:14.36 | *** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158) |
| 05:14.40 | jmls | a (record call) calls B. a transfers to C. recording stops |
| 05:15.01 | jmls | even with AUDIOHOOK_INHERIT set by a |
| 05:15.31 | RiceCracker | Is anyone aware of codec translation issues between Asterisk 1.2 and Asterisk 1.8 over an IAX2 connection??? |
| 05:16.15 | jmls | which is rather pointless, as if a initiated the recording, then it will always be the owner of mixmonitor, despite any transfers by B . therefore, in this scenario, AUDIOHOOK_INHERIT is useless |
| 05:16.58 | jmls | surely if a transfers to c, then c should inherit AUDIOHOOK_INHERIT from A and therefore mixmonitor should continue |
| 05:18.00 | [TK]D-Fender | RiceCracker: transport should have nothing to do with payload |
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| 05:27.37 | deo | thanks [TK]D-Fender |
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| 06:03.59 | v0lZy | lo |
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| 06:36.07 | bulkorok | hi |
| 07:08.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:08.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 07:41.26 | atan | I'm looking at the Linksys SPA400 because it has 4 FXO. Are there any other, 'better' products that are in the same price range? |
| 07:58.42 | fredericve | Does anyone know if it is possible to set variables on a channel that is already up? |
| 08:00.28 | kaldemar | sure. |
| 08:01.21 | fredericve | usecase: an external caller calls into asterisk and it is answered by a SIP peer. The person that answers the calls can then type some combination on his phone (e.g. #812345), which calls a Set(CDR(accountcode)=12345). And the 12345 can be any combination of numbers |
| 08:02.28 | kaldemar | fredericve: see [applicationmap] in the sample features.conf |
| 08:03.46 | kaldemar | making the 12345 part dynamic might not work directly through a feature. |
| 08:04.15 | *** part/#asterisk Aelia (~any@cl-368.gva-01.ch.sixxs.net) |
| 08:04.42 | fredericve | kaldemar: I was indeed looking at that but the 12345 part is the difficulty |
| 08:05.20 | fenrus | win 19 |
| 08:08.17 | kaldemar | from a feature you could try executing a macro that reads the number part and uses AMI to set the variable on the channel. that's one thought. there might be a better way to do this since getting the target channel requires some extra effort. |
| 08:10.58 | fredericve | No that won't work either. From the features.conf sample file: |
| 08:11.02 | fredericve | ; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk |
| 08:11.02 | fredericve | ; applications. When applications are used in extensions.conf, they are executed |
| 08:11.02 | fredericve | ; by the PBX core. In this case, these applications are executed outside of the |
| 08:11.02 | fredericve | ; PBX core, so it does *not* make sense to use any application which has any |
| 08:11.02 | fredericve | ; concept of dialplan flow. Examples of this would be things like Macro, Goto, |
| 08:11.03 | fredericve | ; Background, WaitExten, and many more. |
| 08:11.57 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:12.09 | fredericve | I fear the only possibilty will be through an external application that does an AMI call |
| 08:12.14 | awk | <PROTECTED> |
| 08:13.11 | kaldemar | fredericve: i know it's not recommended by the comment but you still *can* execute a macro. |
| 08:13.11 | fredericve | kaldemar: Thanks for your help. I'm gonna try the mailing lists |
| 08:16.59 | fredericve | kaldemar: Let me try. I'll come back with the result |
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| 08:46.10 | awk | tzafrir_laptop I guess you the only one who can answer this, please advise |
| 08:47.14 | tzafrir_laptop | awk, I seem to have missed the question |
| 08:47.25 | awk | http://pastebin.com/krQLj1rv |
| 08:47.34 | awk | That is the best way I can explain |
| 08:48.04 | awk | I have managed to get this working on audiocodes, which they seem to be using the OpenSBC, menu for menu |
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| 09:10.08 | fredericve | kaldemar: I can call the macro, but When trying to read some digits with the Read application asterisk does not seem to receive them |
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| 09:39.13 | bitwize | fredericve: Do you send the digits via DTMF 2833? |
| 09:39.48 | bitwize | I had the same problem when sending in-band... |
| 09:41.34 | fredericve | bitwize: yes. all phones are configure to use rfc2833 |
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| 10:01.34 | *** join/#asterisk arakno (~raul@net-93-64-173-18.cust.dsl.vodafone.it) |
| 10:04.25 | arakno | hallo there: one question! | analog fax | ----> | asterisk |, is there anyway that asterisk can handle the fax "routing" it via email without losing the analog fax machine ? |
| 10:05.43 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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| 10:09.05 | kaldemar | fredericve: must be a problem with your DTMF settings. |
| 10:09.16 | bulkorok | arakno: you need an analog to T38 gateway between them |
| 10:14.09 | arakno | bulkorok: where can i study this solution? |
| 10:14.56 | arakno | and overall how much is analog fax supported (like ecm and so) |
| 10:15.21 | kaldemar | fredericve: which is odd if you can start the feature in the first place. is audio going through asterisk when the channels are bridged? |
| 10:18.02 | arakno | bulkorok: help me understand | analog fax | --> | t38 gateway | --> | asterisk | -----> provider supporting t38 fax |
| 10:25.59 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul/x-4918593) |
| 10:26.28 | bulkorok | arakno: asterisk supports t38 passthrough... |
| 10:27.47 | bulkorok | arakno: you need sth like the cisco ATA 187 |
| 10:28.16 | arakno | yeah but what on the other side? I mean in Italy we dont have t38 support |
| 10:29.05 | bulkorok | arakno: so how do you make calls? isdn? analog? |
| 10:31.54 | *** join/#asterisk Anderson-D (~anderson@mail.primocollect.com.ua) |
| 10:32.02 | Anderson-D | Hi. I have a question |
| 10:32.37 | Anderson-D | I use asterisk 1.8.7.0 @ CentOS 5.7 (64-bit) |
| 10:32.52 | Anderson-D | I installed asterisk18-tds and asterisk18-odbc packages |
| 10:33.30 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 10:33.36 | Anderson-D | But 'module load res_odbc' tells me "Module res_odbc.so was not compiled with the same compile-time options as this version of Asterisk" |
| 10:34.00 | Anderson-D | What am I doing wrong? I installed everything from the asterisk repo, all libs are 64-bit |
| 10:34.02 | Chainsaw | Anderson-D: That would suggest that the Asterisk package and the add-ons are not from the same repository. |
| 10:34.11 | Chainsaw | Anderson-D: Or that the packager was asleep at the switch. |
| 10:34.28 | Chainsaw | Anderson-D: General sentiment in this channel tends to be against distro packages anyway. You may want to compile from source. |
| 10:35.11 | Anderson-D | Chainsaw, thank you for response! I'm gonna check if everything was installed from same repo. |
| 10:35.28 | Chainsaw | Anderson-D: Something tells me it wasn't, if the compiler was different. |
| 10:35.43 | Anderson-D | I do agree with you that building from source is better, I'll do that if nothing helps |
| 10:36.04 | Chainsaw | Anderson-D: But you may want to report a CentOS bug if this is all from their own repos. |
| 10:36.19 | Chainsaw | Anderson-D: (Or to whoever supplied the packaged builds to you) |
| 10:36.36 | Anderson-D | Yeah, I guess I'll finally submit a bug |
| 10:38.41 | Anderson-D | Chainsaw, you were right. There were few asterisk repos (1.4, 1.8, current etc) |
| 10:39.46 | arakno | bulkorok: analog |
| 10:40.08 | Chainsaw | Anderson-D: Best not to mix them. 1.8 is a good bet if you want a Just Works(TM) experience right now, with few updates. |
| 10:40.37 | arakno | bulkorok: what i need is a fax to mail goodie in between |
| 10:40.39 | bulkorok | sarakno: so you want to send faxes with an analog faxmachine via asterisk to an analog phone-line!? |
| 10:41.10 | arakno | bulkorok: yes, but i want to "handle" voip the fax in between |
| 10:41.25 | arakno | "handle the fax in voip" |
| 10:41.40 | bulkorok | then you need a second analog to voip adapter beteen asterisk and the phone line |
| 10:41.54 | arakno | so a sort of a/d d/a |
| 10:42.14 | bulkorok | it shoeld be more a/t38 t38/a but... yes |
| 10:42.28 | arakno | can you suggest me the device? |
| 10:42.56 | bulkorok | we have some cisco ata ... |
| 10:43.25 | arakno | cisco ata 187.. i see |
| 10:43.55 | bulkorok | digium offers some analgo cards to install in a "real" server... http://www1.digium.com/en/products/telephony-cards/analog |
| 10:46.35 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:46.44 | arakno | thank ya |
| 10:46.49 | bulkorok | sure |
| 10:50.44 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 10:55.06 | Anderson-D | Another problem: after update asterisk doesn't write logs to /var/log/asterisk/messages |
| 10:55.15 | Anderson-D | s/update/reinstall/ |
| 10:55.38 | *** join/#asterisk aossama (~ao@ec2-50-17-241-55.compute-1.amazonaws.com) |
| 10:56.34 | Chainsaw | Anderson-D: That would suggest you have some adjustments to make in /etc/asterisk |
| 10:57.24 | Anderson-D | astlogdir => /var/log/asterisk |
| 10:59.32 | Anderson-D | Alright, fixed by uploading old configs |
| 11:04.54 | wonderworld | are there any problems to think of when virtualizing Asterisk with XEN (no telephony hardware, SIP only) |
| 11:08.30 | wonderworld | i was wondering if timing might be an issue. the system will need to handle a large confbridge 10 mainly |
| 11:09.00 | bulkorok | wonderworld: espacially for conference timing is needed... |
| 11:09.29 | Chainsaw | wonderworld: Additional overhead. Timing-sensitive workloads don't virtualise well at the best of times. |
| 11:09.55 | Chainsaw | wonderworld: Asterisk 10 will cope a little better then say... a 1.4 with a dahdi dummy timer. |
| 11:10.11 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 11:10.30 | Chainsaw | Hi Faustov. |
| 11:10.35 | Chainsaw | But still, I wouldn't. |
| 11:10.39 | wonderworld | thanks, i think i need to rethink things then. probably a dedicated platform will serve me better |
| 11:11.42 | Assid | hmm.. strange issue.. i have an asterisk server in the cloud.. which actually has all the vendors connected.. there is a local asterisk server on site (planning to have this on each site).. when the user makes an outgoing call.. it connects to the asterisk in the cloud server and then makes the call to the vendor from there |
| 11:11.54 | Assid | the issue i face.. when i hangup.. the call continues on |
| 11:12.29 | Assid | i am using iax to interconnect the 2 boxes |
| 11:18.16 | wonderworld | what do you mean by "the call continues on"? |
| 11:18.45 | WIMPy | And where? |
| 11:19.21 | wonderworld | you have [SIP phone] -> [local ASt] -> [IAX] -> [Remote Ast] -> [Provider] -> [PSTN] -> [Phone] , right? |
| 11:19.29 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 11:19.43 | wonderworld | you'd need to find out, where in the chain the call stays active |
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| 11:21.45 | Chainsaw | Thunderstorm in the cloud. |
| 11:22.23 | wonderworld | billing nightmare i'd say |
| 11:42.53 | *** part/#asterisk deo (~dnepangue@203.177.214.75) |
| 11:51.08 | *** join/#asterisk rolandow (~roland@92.68.81.83) |
| 11:51.13 | rolandow | hi guys! |
| 11:51.38 | rolandow | i was wondering if somebody here has eny exprience with discount voip providers like callpirates.com .. |
| 11:51.51 | rolandow | they offer really low prices on cellphone calls .. what's the catch? |
| 11:53.27 | rolandow | and they all look like voipbuster anyways |
| 11:58.10 | plundra | What's the name of that softphone with a blue interface? :) |
| 12:00.59 | *** join/#asterisk teff (~teff@client-80-1-233-47.bsh-bng-011.adsl.virginmedia.net) |
| 12:03.35 | plundra | (Ah! Found it, Zoiper) |
| 12:04.45 | mtbf | Hi guys, I am using asterisk 1.8.13.0, I wanted to obtain also recordings separated by channel (-in, -out .wav) files, so I replaced the sox binary with my own overlay, but I noticed it doesn't change anything even when I remove it, so this means MixMonitor does not use /usr/bin/sox binary and there's no soxmix command, does it have anything build in? |
| 12:11.30 | kaldemar | mtbf: use Monitor instead of MixMonitor. |
| 12:14.08 | mtbf | kaldemar: Thanks, but I don't want to mess in dialplans out there, there is no other workaround? |
| 12:16.05 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xixjghemosqmskjb) |
| 12:16.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:19.37 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:28.56 | *** join/#asterisk bchia (~Adium@nat/digium/x-dwojwmsbahwaqxwx) |
| 12:40.52 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 12:52.05 | rolandow | nobody can tell me about dellmont services? :) |
| 12:52.46 | *** join/#asterisk fredericve (~fes@wan.gw.belgacom.office.wavre.escaux.com) |
| 12:53.29 | fredericve | kaldemar: at first I was using directmedia=yes, but even when it is turned off it does not work |
| 12:53.33 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 12:53.41 | jmls | afternoon all |
| 12:53.47 | jmls | cdr question follows ;) |
| 12:54.17 | jmls | I want to be able to add a custom field to the cdr if a call is pickedup |
| 12:54.32 | kaldemar | fredericve: do you see debug prints for DTMF if you put "dtmf" in the console line in logger.conf? |
| 12:54.36 | jmls | however, how can I tell if the call has been picked up ? |
| 12:55.45 | jmls | or just answered by the target extension |
| 12:57.30 | jmls | and, if I knew that, how could I change / add a field to the current cdr record on the first leg ? |
| 12:57.44 | [TK]D-Fender | jmls, CDR(userfield) |
| 12:58.21 | kaldemar | fredericve: "logger reload" is naturally needed after that change. |
| 12:59.28 | jmls | [TK]D-Fender: I presume that has to be *after* the call pickup ? |
| 12:59.52 | [TK]D-Fender | jmls, You can set it any time... |
| 13:00.16 | *** join/#asterisk serafie (~erin@nat/digium/x-gbnzebxqnevsgajp) |
| 13:00.38 | jmls | weeeeelllllll, the reason why I asked in the first place was that it didn't seem to work if I set CDR(myfield)=foo before the pickupchan() |
| 13:00.48 | specing | What are ports 15750 and 15751 being used for? |
| 13:01.21 | specing | <PROTECTED> |
| 13:01.21 | kaldemar | specing: if your configs are somewhat default, those land in the RTP port range. |
| 13:02.08 | leifmadsen | specing: well that sounds like ports in the RTP range (default 10000->20000 as defined in rtp.conf) |
| 13:02.14 | leifmadsen | ... what kaldemar said |
| 13:02.32 | [TK]D-Fender | jmls, you can set it any time. |
| 13:02.37 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:02.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:02.55 | fredericve | kaldemar: yes the DTMF's are logged |
| 13:03.40 | kaldemar | fredericve: can you pastebin a CLI output of a call with verbosity and DTMF debug and the Macro extension? |
| 13:04.08 | specing | leifmadsen, kaldemar hmm ok |
| 13:04.25 | specing | 150 modules loaded ;_; |
| 13:05.46 | jmls | so, I have this in my log : |
| 13:06.01 | jmls | -- Executing [**2001@from_internal:6] Set("SIP/C8Sk8dy41YUt-0000001a", "CDR(pickup)=2003") in new stack |
| 13:06.52 | jmls | and no pickup field in the cdr record. The inbound channel has "Executing [s@record_call:7] Set("SIP/502807e9b9e737036d000018-00000018", "CDR(recording)=502807e9b9e737036d000018_20120830-140218_07803034440_2001-1346331737.24.wav49") in new stack" |
| 13:07.05 | jmls | and the cdr does have the recording field in the cdr record |
| 13:08.17 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:08.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:12.21 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 13:12.22 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:12.54 | fredericve | kaldemar: http://pastebin.com/mU5nLG0h |
| 13:13.36 | fredericve | I removed some of the dialplan because it's quite large |
| 13:13.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:14.11 | fredericve | The relevant stuff is still there |
| 13:16.26 | bitglue | so how does one typically store the target number for server-managed forwarding? |
| 13:16.47 | bitglue | astdb? |
| 13:17.12 | WIMPy | That would be the obvious choice. |
| 13:17.41 | WIMPy | And by definition, forwarding is server/switch based. |
| 13:18.10 | bitglue | my phone has a dialog labeled "Target when Forwarding", so I don't know about that. |
| 13:18.33 | bitglue | i'll have a hard time convincing my users it's not true |
| 13:22.59 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-hwgixbtrvudgbtmj) |
| 13:24.41 | specing | Chainsaw: Do you(Gentoo) modify the asterisk configs in the repo? |
| 13:24.59 | Chainsaw | specing: No, they get installed unadulterated when USE="samples" is set. |
| 13:25.21 | Chainsaw | specing: And if samples is not set, they are not installed to avoid frustrating "no to everything" etc-update runs. |
| 13:26.17 | specing | Oh ok |
| 13:26.42 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1003) |
| 13:26.52 | specing | Im just plunging through extensions.conf and wondering if I can ask about it here |
| 13:27.09 | leifmadsen | yes? |
| 13:27.19 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:27.29 | specing | I found the [demo] section |
| 13:27.36 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1005) |
| 13:28.02 | specing | adjust the context of my testing client to it and now Im wondering why Im getting nothing from it :| |
| 13:28.16 | leifmadsen | not enough information provided |
| 13:28.27 | Chainsaw | specing: Main thing is to set debug & verbose to 9. |
| 13:28.32 | Chainsaw | specing: So you can see what's going on. |
| 13:29.10 | specing | All I get is SIP read from UDP:10.0.0.8:5080 |
| 13:29.29 | specing | I've tried dialing 1000 as it is in the .conf |
| 13:30.27 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1005) |
| 13:30.57 | specing | From what I see the 's' in "exten => s,1,..." means start |
| 13:31.25 | specing | so it is executed before going to the extension I actually dialed? |
| 13:31.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:31.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:32.09 | [TK]D-Fender | specing, No |
| 13:32.13 | [TK]D-Fender | ~stdextens |
| 13:32.13 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 13:32.46 | leifmadsen | 's' is not a catch all |
| 13:32.55 | specing | ok |
| 13:34.41 | specing | I think I should go on with reading the book instead of poking at stuff in *.conf |
| 13:35.30 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:36.12 | [TK]D-Fender | specing, You should go completely through teh dialplan section at least |
| 13:45.05 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 13:45.53 | bitglue | what's the difference between the "i" and "I" options to Dial()? |
| 13:46.22 | bitglue | ie, what's the difference between "forwarding requests" and "line update requests" or "redirecting party update requests"? |
| 13:50.54 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 13:50.58 | bitglue | also, is there a way to match *any* extension that's not special (t, i, etc), including alphanumeric extensions? |
| 13:51.16 | WIMPy | I is just about updating metadata. |
| 13:51.34 | bitglue | ah |
| 13:51.42 | leifmadsen | bitglue: _[a-zA-Z0-9].,1,NoOp() |
| 13:51.51 | Assid | hmm... |
| 13:52.03 | Assid | [TK]D-Fender: you got a few mins for me ? |
| 13:52.03 | leifmadsen | bitglue: to do what you'd want, replace . with !, and don't include the special extensions in your matching |
| 13:52.12 | bitglue | leifmadsen: that doesn't match "z" |
| 13:52.13 | Assid | you remember how im connecting my nec to asterisk and then to the telco ? |
| 13:52.22 | leifmadsen | bitglue: like I said, replace . with ! and it would |
| 13:52.29 | bitglue | leifmadsen: but then it would match "t" |
| 13:52.39 | leifmadsen | like I said, remove the special extensions from the match |
| 13:52.46 | bitglue | leifmadsen: but then it wouldn't match "tz" |
| 13:52.47 | Assid | for whatever reason.. call conference and redial list on the NEC handsets dont register correctly.. i think its cause the NEC is looking for a certain type of dialtone |
| 13:53.22 | [TK]D-Fender | Assid, shoot |
| 13:53.23 | leifmadsen | bitglue: correct, so you would then place the example I gave you and make it call the same GoSub() you're going to create |
| 13:53.28 | WIMPy | Assid: Dialtone has no meaning. |
| 13:53.49 | bitglue | maybe i don't /actually/ care though, because i'm trying to match the names of my sip peers, which correspond to unix usernames, which all happen to be two or more characters |
| 13:53.53 | leifmadsen | _[a-su-zA-Z0-9]!,1,GoSub() |
| 13:54.05 | leifmadsen | _[a-zA-Z0-9].,1,GoSub() |
| 13:54.18 | Assid | WIMPy: if i connect the telco's pri .. everything automagically works.. like call conference button.. and the redial list on the phone |
| 13:54.28 | leifmadsen | bitglue: also, the specific extension is going to match higher priority than the pattern match |
| 13:54.39 | leifmadsen | so if exten => t,1,NoOp() exists, the pattern match won't match it |
| 13:54.42 | leifmadsen | in the same context |
| 13:54.45 | Assid | WIMPy: the channel status is ok.. everything seems fine and dandy.. otherwise |
| 13:55.04 | Katty | GOOD MORNING |
| 13:55.16 | leifmadsen | Katty: OMG CAN YOU FIX MY ASTRIX?!?!?!?! |
| 13:55.31 | leifmadsen | IT'S BR)KED AND I'M LOOSING MONEYZ! |
| 13:55.47 | *** join/#asterisk defswork (~andy@cpc17-sutt4-2-0-cust175.perr.cable.virginmedia.com) |
| 13:55.49 | leifmadsen | I NEED VERY GOOD HELP ! |
| 13:55.52 | WIMPy | Assid: If it uses external conferences that won't work with Asterisk. But I already told you. |
| 13:56.32 | Assid | WIMPy: the NEC digital handsets which are directly connected to the NEC .. shuold work right ? |
| 13:56.35 | Katty | infobot: good morning |
| 13:56.36 | infobot | Good morning, good! Have a cookie... oh no! The cookie jar is dry! |
| 13:56.40 | [TK]D-Fender | leifmadsen, #psychology |
| 13:56.43 | Assid | im not talking about it using asterisk's meetme |
| 13:56.53 | leifmadsen | infobot: Katty morning |
| 13:57.07 | leifmadsen | huh |
| 13:57.14 | Katty | i think info bot is broken. |
| 13:57.14 | leifmadsen | so ya, looks like it no work like it looks |
| 13:57.25 | leifmadsen | Katty: LIKE MY ASTRIX?!?!?! PLZ 2 BE HELPING ME?! |
| 13:58.30 | chuckf | leifmadsen: ~book |
| 13:58.34 | Katty | http://www.youtube.com/watch?v=8Ts5PU_KYwA <- my favorite morning. |
| 13:58.36 | WIMPy | Assid: I don't know what the NEC would do. |
| 13:59.13 | chuckf | everyone ready for the four day weekend? |
| 13:59.21 | leifmadsen | chuckf: I hope you were joking :) |
| 13:59.29 | Katty | chuckf: i wish! |
| 13:59.38 | Katty | chuckf: i've not even had word that we're off monday here. |
| 13:59.40 | leifmadsen | four day?!? who let this joker off on Friday? |
| 13:59.52 | chuckf | leifmadsen: yeah, it was a joke |
| 13:59.56 | Katty | chuckf: but i'm not important enough for anyone to bother updating me on anything. i'm always the last to know ;) |
| 14:00.04 | leifmadsen | chuckf: ok cool :) otherwise, it'd have been hilarious. |
| 14:00.14 | leifmadsen | see what I did there?! :) |
| 14:00.28 | chuckf | Katty: I'd say take monday off no matter what they say |
| 14:00.55 | chuckf | I'm going to the indy car race this weekend with an all access pass:) |
| 14:01.28 | Katty | i have plans for saturday and sunday, but no plans for Monday yet. |
| 14:01.35 | Katty | the annual Japanese festival is saturday in St. Louis |
| 14:02.24 | chuckf | that could be fun |
| 14:02.48 | Katty | it always is. |
| 14:02.49 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 14:07.49 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 14:14.59 | bitglue | what would most people expect if DND and forwarding is on? Or should I make that state impossible? |
| 14:17.21 | *** join/#asterisk vlad_sta_ (~vlad_star@213.79.102.163) |
| 14:17.39 | *** join/#asterisk GameGamer43 (users.5533@id-5533.hampstead.irccloud.com) |
| 14:17.57 | Assid | hmm.. i get quite a bit of these : [Aug 30 19:45:57] WARNING[17814]: app_dial.c:1472 wait_for_answer: Unable to write frametype: 2 |
| 14:18.21 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-94-17.user.veloxzone.com.br) |
| 14:22.06 | specing | I really think the asterisk book could ease up on the password warnings |
| 14:22.53 | specing | Im guessing it is mostly meant for system administrators and as such they probably know what they are doing and one warning should be enaugh |
| 14:22.59 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 14:23.07 | specing | the book is literaly littered with the red sections |
| 14:23.15 | Katty | literally littered |
| 14:23.21 | pabelanger | specing: complain to leifmadsen |
| 14:23.26 | pabelanger | hides |
| 14:23.42 | Katty | leifmadsen: HOW DARE YOU PUT RED SECTIONS IN THE BOOK |
| 14:23.47 | leifmadsen | Katty: freak out |
| 14:23.52 | pabelanger | .dance |
| 14:24.04 | leifmadsen | specing: the point is that not everyone is going to start at the beginning and work through |
| 14:24.06 | specing | Katty: I have a ... problem with l's |
| 14:24.12 | leifmadsen | it's meant to be jumped throughout |
| 14:24.15 | pabelanger | actually.... (╯°□°）╯︵ ┻━┻ |
| 14:24.18 | mjordan | leifmadsen: you should replace it with "you may get h4x0r3d" |
| 14:24.22 | wonderworld | well, having your asterisk hacked can be real expensive.... |
| 14:24.23 | leifmadsen | pabelanger: OMG YES |
| 14:24.29 | leifmadsen | wonderworld: that |
| 14:24.38 | leifmadsen | in fact, I don't think we have enough warnings |
| 14:25.01 | specing | I'll open up the sip ports and hang an iptables LOG there, then |
| 14:25.04 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 14:25.17 | specing | I want to see just how much garbage there really is there |
| 14:25.25 | wonderworld | just re-read the book btw. but i have an older version. unsure if rebuying it would be neccesary |
| 14:25.31 | Katty | adds a banana peel to specing's book. |
| 14:25.53 | leifmadsen | wonderworld: depends what version you have.... A:TDG 3e was essentially a re-write |
| 14:25.54 | specing | Katty: Im reading the online version ;P |
| 14:26.02 | wonderworld | let me check |
| 14:26.24 | wonderworld | 2005 First Edition ;) |
| 14:26.35 | leifmadsen | wonderworld: omg ya... the 3e isn't even the same book :) |
| 14:26.40 | leifmadsen | it's twice as thick too |
| 14:26.42 | wonderworld | hehe |
| 14:26.48 | wonderworld | nice, i'll go for it |
| 14:27.04 | leifmadsen | note 4e should be out early 2013 (Asterisk 11) |
| 14:27.15 | wonderworld | ok, i'll wait then |
| 14:27.15 | leifmadsen | 3e is for Asterisk 1.8 |
| 14:27.26 | leifmadsen | 4e will just be an update/refresh of 3e |
| 14:27.36 | leifmadsen | at which point we'll probably quit updating it |
| 14:27.41 | specing | RTP is part of SIP, right? |
| 14:27.44 | leifmadsen | no |
| 14:27.50 | leifmadsen | RTP is a separate protocol |
| 14:27.59 | leifmadsen | SIP is a transmission protocol, RTP is for the media |
| 14:28.15 | leifmadsen | SIP happens to use RTP to send the media |
| 14:28.41 | specing | So it is kinda a part of SIP |
| 14:29.22 | leifmadsen | no. |
| 14:29.32 | leifmadsen | it is used by SIP. |
| 14:29.37 | leifmadsen | it's not a part of it at all. |
| 14:30.29 | drmessano | Wait, if you'e going to stop updating it.. |
| 14:30.32 | drmessano | That means |
| 14:30.42 | wonderworld | is there any tool to debug calls (i.e. a better graphical layout / summary for the SIP-messages being transfered) ? |
| 14:30.54 | drmessano | Asterisk syntax and API will not change every again? |
| 14:30.58 | drmessano | OMG |
| 14:31.04 | drmessano | ever* |
| 14:31.11 | specing | leifmadsen: That is like saying a castle is made of lego bricks but they arent a part of it... |
| 14:31.20 | leifmadsen | no it's not |
| 14:31.25 | specing | :) |
| 14:31.33 | drmessano | I knew it.. I knew Asterisk 11 was gonna be the one |
| 14:31.34 | leifmadsen | SIP doesn't need to use RTP |
| 14:31.43 | wonderworld | specing: it's like saying HTMl is part of PHP |
| 14:31.48 | leifmadsen | wonderworld: that |
| 14:31.49 | wonderworld | specing: it's like saying HTMl is part of HTTP even |
| 14:32.47 | specing | Why? PHP is not made of HTML... |
| 14:33.03 | wonderworld | and RTP is not made of SIP |
| 14:33.11 | drmessano | The relationship is more like SMTP/POP or SMTP/IMAP |
| 14:33.14 | leifmadsen | they are separate RFCs |
| 14:33.30 | drmessano | They are used in conjunction but not a part of each other |
| 14:33.43 | specing | wonderworld: you got it the other way around |
| 14:33.43 | wdoekes | SMTP/MIME would be better |
| 14:33.57 | drmessano | You can have SIP without RTP.. If you're not using Audio |
| 14:34.41 | specing | invisible castles ;P |
| 14:34.56 | wonderworld | H323 is using RTP as well for the data stream, isn't it? |
| 14:35.03 | drmessano | Lettuce is not a Taco, but its hard to eat a taco without lettuce |
| 14:35.05 | drmessano | There.. Done |
| 14:35.38 | specing | doesen't know what a "taco" is |
| 14:35.54 | drmessano | I believe there is an RFC |
| 14:35.59 | wonderworld | it's a telco in tazmania |
| 14:36.02 | specing | haha |
| 14:36.49 | *** join/#asterisk ideaman55 (~ideaman55@173-10-29-218-BusName-utah.ut.hfc.comcastbusiness.net) |
| 14:37.09 | bitglue | so, if i make a call with Dial(...U(confirm-call)), where confirm-call is a thing that prompts the called party to press 1 to accept the call, while the called party is being prompted, the caller does not hear ringing. How do I fix that? |
| 14:37.26 | bitglue | i mean, the caller does not hear ringing while the called party is being prompted to accept. |
| 14:37.27 | leifmadsen | bitglue: use 'r' |
| 14:37.52 | leifmadsen | U(confirm-call)r |
| 14:38.01 | wonderworld | the toll-free number you have dialed is not toll-free if dialed from outside the united states :( |
| 14:38.12 | [TK]D-Fender | That only forces ringing where progress isn't passed, but this pause is * based. |
| 14:38.17 | [TK]D-Fender | "r" should not work |
| 14:38.46 | bitglue | i'm testing now |
| 14:39.20 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 14:40.53 | bitglue | r does work |
| 14:42.03 | Kobaz | anyone have any problems with 'old' polycoms randomly dropping calls |
| 14:42.11 | Kobaz | 331's about a year old |
| 14:42.36 | Katty | steals leifmadsen's timtam |
| 14:42.39 | Kobaz | had one phone that kept dropping calls and replaced it, and then it stopped dropping calls |
| 14:42.42 | Katty | makes a timtamslam |
| 14:43.08 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:43.12 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:43.29 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:43.37 | pabelanger | Kobaz: Did you upgrade the firmware since we last spoke? |
| 14:43.45 | Kobaz | yeah |
| 14:43.57 | Kobaz | not at this particular office but at a branch i did the upgrade |
| 14:44.06 | Kobaz | no more complaints of missing rings from them so far |
| 14:44.12 | pabelanger | \o/ |
| 14:44.19 | pabelanger | For the dropped calls, is the phone rebooting? |
| 14:44.38 | Kobaz | the calls just go away |
| 14:44.54 | Katty | ponders friday night shenanigans |
| 14:44.54 | pabelanger | leifmadsen: ^ same issue about phantom ringing |
| 14:45.02 | pabelanger | which we already fixed |
| 14:45.03 | leifmadsen | pabelanger: maybe? |
| 14:45.09 | leifmadsen | I can't remember what we did :) |
| 14:45.11 | pabelanger | Yup |
| 14:45.16 | pabelanger | we did a firmware upgrade to phone |
| 14:45.19 | leifmadsen | ah right |
| 14:45.20 | pabelanger | no more issue |
| 14:45.26 | Kobaz | crazy bugs |
| 14:45.40 | Kobaz | i always thought that the polycom phones were really flakey with that |
| 14:45.52 | Kobaz | like you ring them and then they return a reject code |
| 14:45.58 | pabelanger | Kobaz: But I have a site that is dropping calls because the phone reboots. I suspect it is a PoE issue with lack of power. I moved them to a power brick and waiting to hear back |
| 14:46.19 | Kobaz | but no one was on that phone when it was suposed to be ringing |
| 14:46.29 | Kobaz | an idle phone shouldnt be rejecting calls |
| 14:46.46 | pabelanger | unless a background process is happening? |
| 14:46.54 | pabelanger | polling for new firmwares or configs? |
| 14:47.50 | Katty | s/polling/trolling/ |
| 14:48.26 | bitglue | hmmm so problem with server-managed DND: missed calls do not show up in the handset's log. Maybe there's a way around this? I suspect it may be specific to my handset, but I don't know where to start looking. |
| 14:48.46 | bitglue | unless there's a generic SIP way to place a call, without actually placing it. |
| 14:49.02 | *** join/#asterisk AviMarcus (~avi@192.117.240.103) |
| 14:49.03 | AviMarcus | Anyone heard of endstream before? Or used didlogic for termination? |
| 14:49.07 | Kobaz | maybe |
| 14:49.09 | Kobaz | but i dunno |
| 14:49.11 | AviMarcus | oh. 'lo :P |
| 14:49.18 | Kobaz | even if it's doing some sort of background thing it should take the call |
| 14:49.23 | Kobaz | and pause the background thing |
| 14:49.56 | pabelanger | Kobaz: should and do are two different things that we don't have control over :p |
| 14:54.24 | *** part/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 14:55.15 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 14:56.38 | *** join/#asterisk tstyle (~tstyle@96.53.116.78) |
| 15:00.31 | awk | Got the SBC working :D |
| 15:00.35 | awk | w000p w00000p. |
| 15:05.30 | *** join/#asterisk DaneoShiga (~dshiga@kraz.dreamhost.com) |
| 15:05.48 | DaneoShiga | can i use hostnames on manager.conf on asterisk 1.8? |
| 15:05.51 | *** join/#asterisk tedryder (~tedryder@nc-184-3-98-228.dhcp.embarqhsd.net) |
| 15:07.24 | DaneoShiga | on the permit rule? |
| 15:07.31 | specing | dialplan doesen't work, again :( |
| 15:07.47 | tedryder | Does anyone know if there is a way to interact with an Asterisk meetme through a web interface--talk and listen? |
| 15:10.26 | Katty | hmmmmmmm |
| 15:10.35 | Katty | seems like i recall something someone built for that |
| 15:10.41 | Katty | thinkthink |
| 15:10.48 | Katty | seems like maybe it was part of that trixbox software bundle |
| 15:11.11 | tedryder | Thanks for that. |
| 15:11.18 | tedryder | I will take a look. |
| 15:11.22 | Katty | webmeetme 3? |
| 15:11.35 | Katty | maybe that was fonality |
| 15:11.45 | tedryder | I think that only has controls for admin stuff. |
| 15:11.49 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 15:12.50 | specing | I think linphone may somehow be broken |
| 15:15.13 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 15:15.52 | leifmadsen | faceplams |
| 15:16.00 | leifmadsen | s/faceplams/facepalms |
| 15:16.10 | chuckf | knows linphone is broken somehow |
| 15:16.29 | chuckf | but I suspect you're doing something wrong |
| 15:17.53 | specing | I think linphone is connecting via IPv4 but asterisk is bound to IPv6 |
| 15:18.18 | specing | udp6 0 0 :::5060 :::* 25876/asterisk |
| 15:18.26 | specing | nc 127.0.0.1 5060 |
| 15:18.27 | specing | (UNKNOWN) [127.0.0.1] 5060 (?) : Connection refused |
| 15:18.31 | wdoekes | nc -u |
| 15:18.43 | specing | oh |
| 15:18.46 | chuckf | that's not linphone being broken |
| 15:19.27 | specing | Then I guess I misconfigured asterisk |
| 15:19.39 | *** join/#asterisk j4m3s_ (~j4m3s@adsl-98-64-244-108.mia.bellsouth.net) |
| 15:19.46 | wdoekes | sipsak -vvvs sip:localhost |
| 15:20.09 | specing | sipsak? |
| 15:20.22 | specing | searches |
| 15:20.30 | wdoekes | types |
| 15:20.33 | drmessano | Did you check the box for use ipv6 instead of ipv4 |
| 15:20.44 | specing | drmessano: ? |
| 15:20.54 | drmessano | http://blogs.voxeo.com/speakingofstandards/2011/04/08/how-to-make-sip-calls-over-ipv6-using-linphone-on-mac-windows-linux/ |
| 15:23.18 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:31.38 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 15:34.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:40.29 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:40.32 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 15:40.32 | *** join/#asterisk bchia (~Adium@nat/digium/x-dwojwmsbahwaqxwx) |
| 15:40.32 | *** join/#asterisk teff (~teff@client-80-1-233-47.bsh-bng-011.adsl.virginmedia.net) |
| 15:40.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 15:40.33 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85ed.bcn.adamo.es) |
| 15:40.33 | *** join/#asterisk john_____ (~john_____@smusa.sg) |
| 15:40.33 | *** join/#asterisk n3hxs (~ed@ann100fw01.answernet.com) |
| 15:44.00 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:44.08 | bmoraca_work | dpma is going to give me a friggin ulcer |
| 15:45.05 | bmoraca_work | finally got it working 100% on my test machine...moved the config over to a production machine and now it's having the same issue it had at first... |
| 15:45.25 | bmoraca_work | the proxy extension in dpma_message_context hangs after the first priority |
| 15:46.25 | bmoraca_work | this time, though, downgrading to 1.0.2 didn't fix the issue |
| 15:50.19 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:51.55 | *** join/#asterisk navaismo (~navaismo@189.144.212.238) |
| 15:54.35 | bmoraca_work | i think i figured it out, though...i believe it's related to NAT |
| 15:58.18 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:01.46 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 16:05.48 | ideaman55 | IAX over satellite with bad audio and Jitter problems on the local side according to iax show netstats....Any reccomendations? |
| 16:06.08 | WIMPy | Use a cable. |
| 16:07.27 | WIMPy | Although I've been tol that it doesn;t work that bad. |
| 16:10.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:10.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:13.28 | xSmurf | how do I find out what is causing a high cpu usage? I see one of the threads is constantly using 25% of the CPU (even though there are no active channels) |
| 16:15.20 | roe | I am seeing mention of an iax3 protocol, but I can't actually find the rfc |
| 16:15.41 | WIMPy | Where? |
| 16:16.02 | roe | http://fonlogic.wordpress.com/2010/04/01/new-feature-in-asterisk-1-8-virtual-communication-clouds/ |
| 16:17.27 | WIMPy | And that in 1.8? Sounds like a type, but it doesn't look like one. |
| 16:17.39 | WIMPy | typo |
| 16:17.46 | roe | that's why I figured I'd ask |
| 16:21.08 | WIMPy | Oh, maybe the date is the clue. |
| 16:21.15 | *** join/#asterisk nephfl (614c6ed2@gateway/web/freenode/ip.97.76.110.210) |
| 16:21.32 | roe | haha |
| 16:21.42 | roe | feels stupid |
| 16:23.15 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.37) |
| 16:23.37 | lorsungcu | allergies |
| 16:28.20 | Katty | hands lorsungcu tissues. |
| 16:28.30 | Katty | Qwell: SO i heard isaac is supposed to roll through |
| 16:28.31 | lorsungcu | blah i need a new nose. |
| 16:28.34 | Qwell | orly? |
| 16:28.55 | Katty | Qwell: but on second glance alabama doesn't appear to be in its rollthrough path |
| 16:29.06 | Qwell | we're also practically TN |
| 16:29.19 | Katty | north eastern TN may get some rain |
| 16:29.26 | Katty | hmm |
| 16:29.29 | Katty | no, i take that back |
| 16:29.38 | Katty | you may get some "additional rain" |
| 16:29.50 | *** join/#asterisk k3asd` (~k3asd@host23-87-dynamic.6-87-r.retail.telecomitalia.it) |
| 16:30.20 | Katty | Qwell: http://i.imwx.com/images/maps/truvu/map_specnews27_ltst_4namus_enus_650x366.jpg |
| 16:30.41 | lorsungcu | my brother is in baton rouge |
| 16:30.45 | Qwell | Katty: we're above the 1 |
| 16:31.07 | Katty | Qwell: and saint louis is in a yellow pocket |
| 16:31.15 | Katty | Qwell: right where i'll be saturday for the japanese festival. |
| 16:31.23 | Katty | boooo |
| 16:32.01 | *** join/#asterisk j4m3s_ (~j4m3s@adsl-98-64-244-108.mia.bellsouth.net) |
| 16:36.21 | *** join/#asterisk lorsungcu (~anonymous@50-77-55-81-static.hfc.comcastbusiness.net) |
| 16:37.54 | *** join/#asterisk mattp (~mattp@fuchs.unfinished.org.uk) |
| 16:40.56 | *** join/#asterisk kessius (~kessius@189.4.61.222) |
| 16:42.04 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:42.04 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:44.20 | bitglue | i'm reading some document (http://wiki.snom.com/FAQ/How_to_change_settings_using_SIP_message) that talks about using sipsak to generate a SIP event. Is there a more convienent way to do that within the Dialplan, besides using System and invoking sipsak? |
| 16:46.07 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:49.03 | Qwell | bitglue: Sure, Asterisk 10 added support for sending SIP MESSAGEs |
| 16:49.48 | bitglue | hm, i'm running 1.8 :( |
| 16:49.51 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_MESSAGE and https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend |
| 16:49.55 | Qwell | sucks. upgrade. |
| 16:50.06 | bitglue | well, maybe digium could get it packaged for debian. |
| 16:50.07 | Qwell | (or you could cheat and use Certified Asterisk 1.8.11) |
| 16:50.24 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |
| 16:50.35 | bitglue | i'm not really keen on maintaining my own builds if i can help it |
| 16:54.42 | mattp | how would I go about making call-handling decisions based on the status of a IAX2 trunk? (presumably something related to GotoIf in the dialplan) |
| 16:56.47 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:56.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:57.29 | lorsungcu | my right clicker stopped working :/ |
| 17:02.23 | Katty | OH NOES |
| 17:02.30 | Katty | hate when that happens |
| 17:03.00 | ChannelZ | ironical. My left clicker stopped working the other day. |
| 17:03.14 | Qwell | coincidental* |
| 17:03.32 | Katty | an asteroid, mister president. |
| 17:03.36 | Qwell | glomps Katty |
| 17:03.46 | Katty | hai. |
| 17:03.51 | Katty | i haz a boat. |
| 17:03.54 | Qwell | orly? |
| 17:04.02 | Katty | we can es cap e isaac. |
| 17:04.16 | Katty | plus she will hold gear |
| 17:04.18 | Katty | ALL THE GEARS |
| 17:04.22 | Katty | not to exceed 1000lbs. |
| 17:04.30 | Qwell | so, SOME OF THE GEARS |
| 17:04.35 | Katty | yes. |
| 17:04.51 | Katty | i also have a floaty cooler we can tow-behind. |
| 17:05.10 | Katty | and a pirate flag. |
| 17:05.15 | Katty | cause what boat is complete without a pirate flag? |
| 17:05.22 | Katty | no boat of mine, clearly. |
| 17:05.26 | bitglue | is there something like System() but that isn't subject to shell injection attacks? |
| 17:05.46 | Qwell | bitglue: don't allow arbitrary arguments.. |
| 17:05.49 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:06.08 | specing | it still isn't working |
| 17:06.20 | Katty | neither is my tax money. |
| 17:06.26 | Katty | cries. |
| 17:06.28 | specing | I'be removed all the ipv6 from the configuration |
| 17:06.34 | specing | have* |
| 17:06.39 | bitglue | Qwell: well, that means i can't allow any user input to affect anything System() does, which is rather limiting... |
| 17:06.59 | *** join/#asterisk Tim_Toady (~fuzzy@193.92.239.195.dsl.dyn.forthnet.gr) |
| 17:07.02 | Katty | Qwell: we need our PADI certifications for when the world floods. |
| 17:07.14 | specing | oh well I'll just gist spam |
| 17:07.21 | Qwell | bitglue: Welcome to the world of executing arbitrary commands. |
| 17:08.05 | bitglue | Qwell: you know, most programming languages have a way to invoke exec(), which isn't subject to the shell. Failing that, they usually have a function to quote things for a shell. |
| 17:08.17 | Qwell | Patches welcome. |
| 17:08.27 | bitglue | Qwell: "no" would have been a perfectly fine answer |
| 17:08.35 | Katty | bitglue: you seem cranky today. |
| 17:08.40 | Katty | gets bitglue a mt dew |
| 17:08.42 | bitglue | Katty: my asterisk is broken. |
| 17:09.01 | Katty | bitglue: perkup cranky pants, people in the world are dying from malnutrition. |
| 17:10.03 | bitglue | Katty: well see, i have this shell script that feeds people, and i was implementing an IVR to allow people to feed them. The problem is, I'm afraid someone will select "press 1 to send food to Africa; rm -rf #", so I'm kinda stuck. |
| 17:10.32 | Katty | it's always good to have perspective. |
| 17:10.46 | Katty | and you can always hire someone if you need help Right Now! |
| 17:14.16 | specing | https://gist.github.com/73d0828a70e0f00722c7 -- what is broken here? |
| 17:16.05 | Katty | drmessano: how's that dog house. |
| 17:16.53 | bitglue | specing: it looks like 10.0.0.8 thinks "jaK" is a valid SIP message. |
| 17:17.14 | Qwell | wtf is the "jaK" from? |
| 17:17.23 | specing | I have no idea |
| 17:17.36 | Qwell | Your shit be broken, yo. |
| 17:17.49 | specing | the computer hosting asterisk is the same as the one with linphone |
| 17:17.57 | specing | 10.0.0.8 is eth0 |
| 17:18.08 | Qwell | find another client. that one is clear all jaK'd up. |
| 17:18.11 | Qwell | clearly |
| 17:19.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:19.56 | specing | lol |
| 17:20.19 | specing | well I can't seem to be able to bing ekiga to something other than 5060 and jitsi is java |
| 17:20.24 | specing | bind* |
| 17:21.18 | specing | Qwell: it worked yesterday... |
| 17:21.27 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:22.23 | specing | Now sip debug is full of CSeq N REGISTER followed bi jaKs |
| 17:23.57 | leifmadsen | you could always make asterisk listen on something other than 5060 rather than the client |
| 17:26.55 | specing | I don't think these clients will allow me to change the connect port... |
| 17:27.08 | specing | since you can't even configure the listen on ekiga... |
| 17:27.20 | WIMPy | <specing> I think I should go on with reading the book instead of poking at |
| 17:27.24 | WIMPy | stuff in *.conf |
| 17:27.27 | WIMPy | [15:35] nickfennell has joined #asterisk (~nickfenne@unaffiliated/nickfennell) |
| 17:27.32 | WIMPy | Sorry. |
| 17:27.39 | WIMPy | Desk overcluttered :-( |
| 17:28.25 | drmessano | Katty: It's ok. I got a slap on the hand |
| 17:29.57 | specing | I think I'll nuke the linphone userconfig |
| 17:30.58 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 17:34.25 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 17:35.42 | bitglue | high-fives #asterisk |
| 17:36.10 | bitglue | syncing between handset and asterisk's idea of the DND setting = complete |
| 17:39.20 | Katty | drmessano: phew. |
| 17:41.49 | drmessano | I kinda missed something before I left the office last night |
| 17:42.14 | Qwell | drmessano: forget to remove another sponge from a patient, doc? |
| 17:42.20 | specing | I still don't know what is wrong :| |
| 17:42.27 | specing | Ah right, sipsak |
| 17:42.34 | drmessano | I had already had a convo with one of our traffic/billing girls and told her we wouldn't miss ANY spots last night during the upgrade.. and if we did, I would load them back in |
| 17:43.11 | drmessano | I left.. and 20 mins later, we ran a sync and it blew out the last batch of them for the 2am hour on all stations |
| 17:43.13 | drmessano | OOPS |
| 17:43.49 | specing | Had a rough day? xD |
| 17:44.00 | drmessano | Rough night |
| 17:44.07 | specing | or that |
| 17:44.22 | drmessano | Welll, technically, wasnt rough until I came in today and had to apologize for the missed spots I didnt know were missed |
| 17:44.36 | drmessano | Granted, I cost us like $100 in revenue, but whatever |
| 17:45.50 | drmessano | I hate not being perfect. Thought I had all the bases covered |
| 17:45.55 | drmessano | meh |
| 17:46.44 | drmessano | I guess it's better than the unstable PBX problem i've been dealing with for ages now |
| 17:51.09 | *** join/#asterisk NOT_guru (~chatzilla@24-241-103-142.static.stls.mo.charter.com) |
| 17:51.22 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 17:57.35 | specing | I have reinited linphone and now it works |
| 17:57.46 | specing | though I still get jaKs through |
| 17:58.09 | Qwell | I bet that's a stupid ping or something. |
| 17:58.18 | specing | hehe |
| 17:58.36 | Qwell | No, really. |
| 18:00.06 | drmessano | It's jaK!!! :) ... eXosip's udp keep alive. |
| 18:00.11 | drmessano | Thats a paste |
| 18:00.23 | drmessano | http://web.archiveorange.com/archive/v/YpFJNYnnzv8RBN0clvyn |
| 18:00.26 | drmessano | Youre welcome |
| 18:00.58 | Qwell | boom, nailed it |
| 18:01.42 | Qwell | and Klaus knows what he's talking about. |
| 18:02.53 | drmessano | http://git.savannah.gnu.org/cgit/exosip.git/tree/src/eXtl_udp.c <-- Line 790 |
| 18:02.58 | drmessano | Who has access? |
| 18:03.07 | drmessano | Lets fix this, open source gorilla style |
| 18:03.34 | drmessano | I want to write the comment for the commit |
| 18:03.57 | drmessano | "Changed to CRLF because we're not jaKing around" |
| 18:05.17 | specing | ha ha |
| 18:06.27 | drmessano | Weird.. Wesley Snipes hasn't been in any movies in 2 years |
| 18:06.30 | drmessano | Oh, wait |
| 18:09.45 | Qwell | drmessano: Can't submit patches on savannah? |
| 18:09.46 | leifmadsen | heh |
| 18:10.36 | specing | I see using "99luftballons" for an extension name is officially endorsed. |
| 18:12.31 | drmessano | Qwell: No clue.. |
| 18:12.48 | drmessano | I was just asking if someone had a key. I guess you can knock if you like |
| 18:13.23 | Qwell | If you can't commit it, fork it. |
| 18:14.10 | drmessano | Not sure that this whole jaK thing is worth it |
| 18:14.19 | drmessano | Screw linphone, TBH |
| 18:14.25 | leifmadsen | +1 |
| 18:14.49 | drmessano | I hope their repo crashes and their backups contain CRC errors |
| 18:14.50 | drmessano | YEAH |
| 18:14.50 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 18:16.22 | specing | Who made all these sounds in /var/lib/asterisk/sounds/? |
| 18:16.30 | Katty | Me. |
| 18:16.34 | specing | they are certainly comprehensive |
| 18:16.53 | Katty | (not really). |
| 18:17.17 | Qwell | specing: There should be a CREDITS file in the same dir. |
| 18:17.17 | leifmadsen | well, for english, Allison Smith made them |
| 18:19.09 | bitglue | so, if i changed CALLERID just prior to Queue() so that the queue members know which queue is calling them, where can in unchange CALLERID so that if the call is then transferred, it does not have the queue name in it? |
| 18:19.36 | drmessano | One of these days I should send off for a few prompts |
| 18:21.45 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:24.25 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-brmrrwiuocluxniy) |
| 18:24.40 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-jitahduvihrhvzfc) |
| 18:25.58 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 18:31.53 | specing | I look forward to recording myself and then have myself bounce around the PBX system :P |
| 18:32.05 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 18:32.42 | specing | Goto(99luftballons,threat1,1) :) |
| 18:36.32 | *** join/#asterisk twanny796 (~twanny@46.11.10.19) |
| 18:38.09 | jpsharp | Every extension a Captain Kirk? |
| 18:38.48 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 18:44.17 | pigpen | Dammit Jim! |
| 18:44.21 | pigpen | couldn't resist. |
| 18:44.37 | *** join/#asterisk dougsk (dougsk__@66.45.165.167) |
| 18:45.17 | jpsharp | I'm not Jim. Thats my dad! |
| 18:45.44 | pigpen | Anyway, got a quickie. I have some users that want to be able to have a call come in, answer, put it on hold and pick it up on another extension, put it on hold, pick it up on another extension, etc... |
| 18:45.58 | dougsk | hello there, any recommendation for a small hardware appliance with one span T1 for testing purposes? |
| 18:46.19 | pigpen | Now, call parking comes to mine, but the hold, pickup, hold pickup introduces too many keystrokes |
| 18:46.32 | pigpen | Is what I want SLA? |
| 18:46.42 | pigpen | dougsk, small or cheap? |
| 18:46.54 | [TK]D-Fender | dougsk, The words "testing purpose" have no meaning. It either is a T1 interface or it isn't. There is no "Light". |
| 18:47.06 | dougsk | pigpen, big is okay since cheap is better :D |
| 18:47.25 | leifmadsen | pigpen: no -- call parking |
| 18:47.29 | pigpen | audiocodes or digium |
| 18:47.52 | pigpen | leifmadsen, so I can just "put it on hold back whence it came? |
| 18:47.56 | dougsk | [TK]D-Fender, get that no worries. |
| 18:48.11 | leifmadsen | pigpen: no -- I guess that's SLA, but it's going to be significantly more complicated to setup |
| 18:48.43 | pigpen | yeah. I am very familiar with parking. but didn't know much about sla |
| 18:49.12 | leifmadsen | in Asterisk, it's basically a meetme room being monitored via state |
| 18:49.31 | pigpen | right. |
| 18:50.06 | pigpen | and it does work well, most of the time. If it does bork, it is usually the "watcher" device fault. Fix with a reboot |
| 18:51.50 | pigpen | thanks again! |
| 18:51.54 | dougsk | pigpen, [TK]D-Fender, I was looking at something like the following, wrong direction? http://www.ebay.com/itm/1U-rack-mount-IP-PBX-1-E1-T1-port-Elastix-PBX-VoIP-PBX-voip-server-phone-system-/160849411154?pt=LH_DefaultDomain_0&hash=item25735f3c52#ht_2500wt_1163 |
| 18:52.09 | leifmadsen | pigpen: np :) |
| 18:52.39 | specing | looks kinda cheap |
| 18:52.45 | specing | *grin* |
| 18:53.00 | dougsk | yeah it's definitely cheap, that's why I'm looking at it :D |
| 18:53.10 | [TK]D-Fender | dougsk, If anything goes wrong the words "zero customer support" come to mind |
| 18:53.11 | Qwell | ~cheap |
| 18:53.11 | infobot | well, 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. |
| 18:53.29 | pigpen | pick up a 1u atom supermicro, put a single t1 in it, load up linux & asterisk < done |
| 18:53.47 | drmessano | Yep |
| 18:53.49 | pigpen | Supermicro makes a nice D525, has been very reliable. |
| 18:53.52 | bitglue | so uh....it seems like if i try to run a command that doesn't exist with System(), my call is terminated. Is that intended behavior? How am I to check SYSTEMSTATUS if so? |
| 18:53.53 | drmessano | Supermicro are teh shiznit |
| 18:53.55 | dougsk | infobot, nah it's okay, it will never hit production. pigpen, yeah that seems to be the best answer at this point. |
| 18:54.17 | drmessano | I have (2) Supermicro boxes in our mobile operation center. Everything "just works" |
| 18:54.26 | Qwell | dougsk: If you're going to use different hardware in production, then your testing is useless. |
| 18:54.29 | pigpen | we have deployed about 75 or so with 1 bad on delivery |
| 18:55.07 | dougsk | Qwell, quite right, and that's not the point of it. The LEC will not turn up the circuits unless I have a pbx on site. |
| 18:55.20 | dougsk | I just need to turn up a pbx so they can loop in the smartjack and walk away |
| 18:55.33 | dougsk | then two weeks later come back with real hardware |
| 18:55.48 | specing | Qwell: What if I am a cheapskate? Where do I get really cheap hardware for cheapskates? |
| 18:55.53 | [TK]D-Fender | dougsk, Maybe it'll work. Maybe it's cause problems. Who knows. |
| 18:56.08 | [TK]D-Fender | dougsk, We don't trust the vendor. YMMV |
| 18:56.13 | leifmadsen | you don't get hardware then |
| 19:00.15 | chuckf | specing: you get the cheap hardware for cheapskates from china |
| 19:00.53 | specing | :) |
| 19:03.03 | drmessano | and you don't annoy the kind folks on IRC when your crap system starts acting up and you swear it is an Asterisk bug |
| 19:03.09 | drmessano | Because, yeah |
| 19:03.20 | pigpen | haha...^^^ |
| 19:03.30 | dougsk | nice |
| 19:03.44 | pigpen | drmessano, take a bow. That was good |
| 19:04.01 | drmessano | bows |
| 19:04.06 | navaismo | argh chinaroby really sucks, I knew one customer with a clone tdm card and it need patches only for old dahdi version and always complaining about quality |
| 19:04.43 | drmessano | I saw their 4 port FXO card is $89. HAHAHAHHA... I was nervous about buying a fully loaded OpenVox card for $170. Guess I OVERPAID |
| 19:04.56 | navaismo | I finished the relationship |
| 19:05.46 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 19:06.01 | navaismo | OpenvoX is better than chinaroby |
| 19:06.14 | pigpen | any thoughts about the Cisco 7931 ip phone (I prefer polycom, but it is a button issue) |
| 19:06.20 | drmessano | It's pretty bad when someone makes a cheap enough chinese clone to make buying an X100P clone modem POS card the "expensive route" |
| 19:06.51 | drmessano | I wonder how much a 1 port chinaroby card runs |
| 19:06.58 | specing | drmessano: what if it is indeed an asterisk bug? |
| 19:07.38 | drmessano | specing: It's probably not. |
| 19:07.45 | specing | heh |
| 19:08.16 | navaismo | OMG sip phone for 50USD haha and stupid me saving for a D40 |
| 19:08.50 | specing | Why is there a picture with western people on the openvox homepage if they are in China? |
| 19:09.08 | Qwell | specing: Because nobody would knowingly buy that garbage, so they have to hide that fact? |
| 19:09.23 | drmessano | WHY R MY NIFTY FIVE DOLLAR CARD NOT BEING SEEN BY THE AKERISK <--- Probably not a DAHDI bug |
| 19:09.50 | drmessano | WHY ARE MY CARD NOT DIALTONE MUCH OFTEN <--- Probably not a bug either |
| 19:10.17 | navaismo | use the echo too |
| 19:11.06 | drmessano | I R HEER ECHO AND I BOUGHT HARDWIRE ECHO CASELER???? <--- I'm sure that $3 echo cancellation is as good as Digiums |
| 19:12.06 | Roelt | drmessano, do you know why my androidiphonewindowsphone sip client doesn't work with edge in the middle of nowhere under a bush in a tree up a creek? |
| 19:12.47 | Roelt | oh, and the battery is flat and i'm using freepbx on windows with a vmware inside osx/2 |
| 19:12.48 | drmessano | Blame NAT... Because everyone knows SIP doesn't work through a NAT |
| 19:12.58 | Roelt | bloody rain |
| 19:14.11 | drmessano | CHINAROBY SAY SMOKE ARE AKERISK BUG AND MUCH ARGUE WITH -DEV TO FIX?!? <--- Best $485 you ever spent |
| 19:14.51 | drmessano | Need to do a series of "You might be a cheapskate if.." with a nod to Jeff Foxworthy |
| 19:15.15 | drmessano | If you ever bought a PBX that was cheaper than a Grandstream, you might be a cheapskate |
| 19:15.28 | Qwell | s/PBX that was cheaper than a // |
| 19:15.31 | specing | Qwell: fair enaugh. |
| 19:15.39 | drmessano | lol |
| 19:16.10 | drmessano | Are those things RoHS compliant? |
| 19:16.15 | drmessano | Wait, HAHAHAHHAHA |
| 19:17.54 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-38-44.user.veloxzone.com.br) |
| 19:17.58 | drmessano | Check out the guy on the banner at http://www.chinaroby.com/ |
| 19:18.03 | navaismo | impressions on http://openvox.cn/en/products/ippbx/icalldroid.html?page=shop.product_details&flypage=flypage.tpl&product_id=109 |
| 19:18.03 | drmessano | Holding the cell phone |
| 19:18.09 | drmessano | Yeah, his PBX stopped working |
| 19:18.39 | navaismo | voiptoday say "OpenVox Leads Asterisk Home Unified Communication." |
| 19:18.41 | Qwell | navaismo: See that .cn in the domain name? |
| 19:19.28 | navaismo | yep |
| 19:19.43 | Qwell | Have you been completely ignoring everything we've been saying the last half hour? |
| 19:19.52 | drmessano | I bought an OpenVOX 4-port.. 2 FXO, 2 FXS.. it was OK-ish. I found it was damn near impossible to get the echo out of the thing, and I was using a relatively decent line |
| 19:20.09 | chuckf | See that .cn in the domain name? << There's your problem |
| 19:20.25 | drmessano | I ended up using the FXS ports for alarm system closures lol |
| 19:21.01 | navaismo | well I used an IX100 before, and so far was a good choice for small soho and that make me think openvox was better than chinaroby |
| 19:21.28 | dougsk | drmessano, "smoke are asterisk bug" LMAO -- okay okay I give I picked up a digium card :D |
| 19:22.41 | drmessano | I have had a Digium T1 card connected to a Rhino Channel bank for well over a year.. and it's been incredibly rock solid with all the call abuse |
| 19:24.16 | jpsharp | I have a customer who's been running a Digium T1 card + Carrier Access AccessBank II for almost 8 years now. |
| 19:25.51 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 19:26.04 | drmessano | We probably set up 2500 calls an hour, answer maybe 20 of those.. Can't recall any issues |
| 19:26.46 | navaismo | another astricon missed :'( |
| 19:27.25 | drmessano | I kinda want that Chinaroby phone |
| 19:27.32 | drmessano | Little afraid of the lead poisoning |
| 19:28.33 | _Corey_ | Jeez.. those ChinaRoby guys even rip off the Asterisk trademark on their circuit boards |
| 19:30.23 | [TK]D-Fender | In china Copyright is interpreted as "It's your right to copy" |
| 19:32.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 19:32.40 | navaismo | LOL |
| 19:33.07 | drmessano | Yes, but they call it "Circle Star" |
| 19:33.34 | drmessano | and you cant sue someone for using a CIRCLE and a STAR, gosh darnit |
| 19:34.09 | drmessano | OMG... that would be a great name for an Asterisk fork |
| 19:34.22 | drmessano | I mean.. AHEM URM UHHHH |
| 19:36.18 | jpsharp | Splat |
| 19:37.36 | specing | drmessano: What is the problem with it not being RoHS? |
| 19:37.50 | specing | Atleast soldering shit to it becomes easier |
| 19:38.26 | specing | i've spent a full hour to solder a serial line to my rohs tablet. |
| 19:40.19 | drmessano | Because those lead using bastards are spoling our children, and our childrens-childrens water and food supplies. |
| 19:40.23 | drmessano | Also, a little heavier |
| 19:43.47 | jaytee | it's not just the lead from China that's a problem.... it's the cadmium too! |
| 19:44.33 | specing | lead is everywhere, lol |
| 19:44.47 | specing | just not where it would be most usefull - on electronics |
| 19:46.13 | drmessano | What do you have against tin? |
| 19:46.24 | drmessano | Is 63/37 not good enough for you? |
| 19:51.17 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 19:52.16 | sekil | anyone know why stupid snom 320 in 6.x fw won't reply when being 401d by *? |
| 19:55.25 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:06.51 | *** join/#asterisk WolWid (WolWid@pD9ED4B7C.dip.t-dialin.net) |
| 20:09.20 | sekil | or..is there a way to accept register w/o challenge? |
| 20:09.38 | n3hxs | Meh, tin whiskers.... will kill your electronics... in the future. |
| 20:30.11 | sekil | grrr |
| 20:30.32 | sekil | does anyone know how to tell * to blindly accepts registers |
| 20:36.40 | *** join/#asterisk str8uplinux (~dschuett@mail.lonemountaintruck.com) |
| 20:37.10 | pigpen | does anybody know if digium supports xml contacts via ftp yet? |
| 20:37.14 | pigpen | I don't remember. |
| 20:37.19 | str8uplinux | is the mysql database/table in 10.7 the same as used in 1.8? |
| 20:37.36 | pigpen | str8uplinux, shoud be. |
| 20:38.05 | str8uplinux | ok, i just wanted to make sure. |
| 20:38.07 | str8uplinux | thanks |
| 20:41.08 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 20:44.12 | bmoraca_work | man, DPMA is really bothering me right now |
| 20:44.13 | bmoraca_work | urg |
| 20:44.23 | bmoraca_work | works perfectly on my testing server |
| 20:44.32 | bmoraca_work | but not on the production server |
| 20:46.34 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:48.12 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.1 (2012/08/30), 1.8.15.1 (2012/08/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:50.11 | specing | libpri is 2011? better make a new release right now! |
| 20:50.55 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:55.45 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 20:55.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:56.33 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-7-26.se.biz.rr.com) |
| 20:57.24 | leifmadsen | nah. |
| 20:58.59 | bchia | bmoraca_work: what's not working for you? |
| 20:59.23 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-7-161.broadband.tenet.odessa.ua) |
| 20:59.25 | WIMPy | sekil: Don't you think v6 is a little dated? |
| 21:00.32 | sekil | I think that anything is better than upgrading snoms |
| 21:01.01 | *** join/#asterisk Goldwing (~Goldwing@84.245.47.128) |
| 21:01.13 | WIMPy | why? |
| 21:02.15 | xSmurf | any clues are to why asterisk takes 25% of the CPU in one thread while there are open channels? |
| 21:02.26 | sekil | cause it sucks...had major issues when upgraded 6-7 fw version... |
| 21:02.36 | sekil | even had to rma couple phones.. |
| 21:02.55 | sekil | altogether fw upgrade process on snoms scares me.. |
| 21:03.23 | Qwell | xSmurf: There are literally hundreds of things that it could be doing. |
| 21:03.28 | WIMPy | Worked automatically ofr me way back then. Just pointed it to the URL from the wiki and had to wait for quite a long time. |
| 21:03.46 | xSmurf | Qwell: have so clues to where I should look? |
| 21:03.53 | xSmurf | it's 25~30% constantly |
| 21:04.15 | Qwell | Show me that it's Asterisk causing it, first. |
| 21:04.16 | *** join/#asterisk Hectaman (~hectaman@unaffiliated/hectaman) |
| 21:04.48 | xSmurf | 1 S asterisk - - 20942 21 - 49 - - - - - 0 Aug29 - 04:00:31 - |
| 21:04.48 | bmoraca_work | bchia: Asterisk never gets past the first priority in the proxy extension under dpma_message_context |
| 21:04.52 | xSmurf | 21% |
| 21:05.00 | Qwell | xSmurf: pastebin the output of top |
| 21:05.04 | Qwell | press 1 first |
| 21:05.10 | xSmurf | that was fair more useful |
| 21:05.20 | xSmurf | that's from ps -LlFm -p `pidof asterisk` |
| 21:05.25 | xSmurf | all other threads are at 0 |
| 21:05.36 | xSmurf | 20897 asterisk -11 0 736m 33m 17m S 25.2 6.6 245:17.68 asterisk |
| 21:05.58 | bmoraca_work | works perfectly on my test machine but not on my others... |
| 21:06.00 | bmoraca_work | sucks |
| 21:06.11 | Qwell | bmoraca_work: Are you still getting the 401s? |
| 21:06.23 | bmoraca_work | Qwell: no, I got that fixed |
| 21:06.30 | bmoraca_work | Qwell: but i'm back to my initial problem |
| 21:06.33 | bmoraca_work | well |
| 21:06.33 | Qwell | show me a sip debug then |
| 21:06.34 | bmoraca_work | sort of |
| 21:06.38 | bmoraca_work | it works perfectly on my test box |
| 21:07.02 | bmoraca_work | but when i went to make it live on my production box, it's doing the thing where it hangs after the first priority |
| 21:07.30 | xSmurf | I was going to pstack but the amd64 pkg is missing :/ |
| 21:08.07 | Qwell | Or you could show me what I asked for. :) |
| 21:08.17 | xSmurf | how is top going to help over what I pasted? |
| 21:08.22 | bmoraca_work | i'm working on it |
| 21:08.26 | Qwell | bmoraca_work: not you |
| 21:08.30 | bmoraca_work | oh, sorry :) |
| 21:08.52 | *** part/#asterisk Hectaman (~hectaman@unaffiliated/hectaman) |
| 21:09.20 | xSmurf | http://pastie.org/private/xkhmxoyrcm2pdxccyfhq |
| 21:09.46 | xSmurf | http://pastie.org/private/6rhg57tj1wsu4dlkbrh0g |
| 21:10.32 | Qwell | xSmurf: press 1 first |
| 21:12.17 | xSmurf | there's a single cpu |
| 21:12.29 | xSmurf | it's a xen instance |
| 21:13.01 | xSmurf | you want the munin graph to? |
| 21:13.14 | xSmurf | http://stats.koumbit.net/cgi-bin/munin-cgi-graph/koumbit.net/voice1.koumbit.net/cpu-day.png |
| 21:13.16 | Qwell | okay, so, figure out what is causing it to hang in the kernel. I feel like we've already had this discussion before. |
| 21:13.24 | bmoraca_work | Qwell: http://pastebin.com/ZsnaGaVa if you've got a moment or are at all interested...also, downgrading to 1.0.2 yields the same result |
| 21:14.32 | xSmurf | maybe related to [1134301.860021] dahdi: Detected time shift. ? |
| 21:15.13 | Qwell | xSmurf: Do you have DAHDI hardware? |
| 21:15.18 | xSmurf | no |
| 21:15.43 | xSmurf | but we use meetme |
| 21:16.04 | sekil | found out |
| 21:17.59 | Qwell | ~asterisk debugging |
| 21:18.01 | Qwell | ~asteriskdebugging |
| 21:18.05 | Qwell | stupid bot |
| 21:18.11 | [TK]D-Fender | ~debug |
| 21:18.11 | infobot | ACTION DeBuggers $1 |
| 21:18.13 | Qwell | bmoraca_work: core set debug 10, and enable debug in logger.conf |
| 21:18.15 | [TK]D-Fender | ~debuggin |
| 21:18.18 | [TK]D-Fender | ~debugging |
| 21:18.18 | infobot | if debugging is the process of removing bugs, then programming must be the process of putting them in. |
| 21:18.22 | [TK]D-Fender | :/ |
| 21:18.31 | [TK]D-Fender | ~collectdebug |
| 21:18.31 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:18.37 | [TK]D-Fender | Ah, there we go |
| 21:18.56 | Qwell | bmoraca_work: ^^ steps 1/2 |
| 21:19.22 | Qwell | err, set it to go to console though |
| 21:19.29 | bmoraca_work | hmmmmmm |
| 21:19.50 | bmoraca_work | the difference between the two systems is that avahi is running on the one that works... |
| 21:19.52 | Qwell | also core show channels |
| 21:20.05 | Qwell | that shouldn't matter |
| 21:20.10 | Qwell | it's not even getting to DPMA |
| 21:21.15 | *** join/#asterisk F2Knight (~Adium@70-89-188-5-or.portland.hfc.comcastbusiness.net) |
| 21:21.18 | bmoraca_work | you're right, it didn't matter...but i had to try :P |
| 21:21.25 | bmoraca_work | one sec, lemme get the info |
| 21:21.35 | bmoraca_work | this system is fairly heavily used, so might take a minute |
| 21:21.47 | Qwell | try to get the core show channels after the priority 1 execution, but before the sip dialog goes away |
| 21:23.32 | bmoraca_work | so you want console log? |
| 21:23.45 | Qwell | debug on console would be best |
| 21:23.58 | Qwell | just add a ,debug at the end of the console => line |
| 21:24.41 | bmoraca_work | holy crap |
| 21:24.45 | bmoraca_work | that's a lot of info |
| 21:24.50 | Qwell | yes, yes it is |
| 21:24.59 | Qwell | feel free to throw it to a file instead :p |
| 21:25.07 | bmoraca_work | i think i'll need to |
| 21:25.44 | *** join/#asterisk jpcansa (~JP@201.200.132.10) |
| 21:27.04 | jpcansa | can anyone recommend me a equipment to provide E1s, i mean from ds4 to e1 |
| 21:27.52 | WIMPy | jpcansa: From or to where? |
| 21:28.18 | xSmurf | for my part this is pretty much all I have in the debug log http://pastie.org/private/23kxdrdvdp4hqvorybgxia |
| 21:28.35 | Qwell | xSmurf: It's not Asterisk. It's in the kernel. |
| 21:28.46 | Qwell | You were using that weird USB GSM card before. Did you remove that? |
| 21:28.58 | xSmurf | but it's an asterisk thread using up the cpu |
| 21:29.04 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 21:29.05 | jpcansa | WIMPy, i get for example a DS4 from my provider and then i provide E1s to my customers in the same building |
| 21:29.16 | bmoraca_work | Qwell: i'm going to filter out all the dsp.c debug lines...ok? |
| 21:29.22 | Qwell | bmoraca_work: yep |
| 21:29.48 | WIMPy | jpcansa: RAD has all sorts of (de)multiplexers for about any situation. |
| 21:29.56 | xSmurf | at best it's dahdi module fucking up |
| 21:30.01 | xSmurf | which is still asterisk related |
| 21:30.04 | Qwell | xSmurf: Are you using meetme for that call? |
| 21:30.09 | xSmurf | as I said, yes |
| 21:30.27 | Qwell | What versions of DAHDI and Asterisk? |
| 21:31.09 | jpcansa | thanks WIMPy , ill take a look at rad.com |
| 21:31.15 | xSmurf | asterisk 1.6 |
| 21:31.21 | Qwell | rolls his eyes |
| 21:31.24 | Qwell | ~upgrade asterisk |
| 21:31.24 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 21:31.27 | xSmurf | no |
| 21:32.06 | xSmurf | dahdi 2.3.0.1 |
| 21:32.07 | bmoraca_work | Qwell: http://pastebin.com/TzXwUD8x |
| 21:32.15 | bmoraca_work | core show channels did not show the channel |
| 21:32.22 | xSmurf | I'll upgrade when there is a stable pkg |
| 21:32.23 | bmoraca_work | though the sip dialog ended immediately |
| 21:32.33 | xSmurf | I'm not gonna start compiling and maintaining stuff in production |
| 21:32.42 | Qwell | xSmurf: Enjoy your evening then. |
| 21:32.48 | xSmurf | is tired of having this discussion |
| 21:32.57 | xSmurf | if you want people to upgrade so bad, maintain packages |
| 21:33.06 | xSmurf | it's not to the end user to maintain packages |
| 21:33.10 | bmoraca_work | upgrading isn't that hard... |
| 21:33.35 | xSmurf | bmoraca_work: having to check every day to make sure there aren't any upgrades is unacceptable |
| 21:33.41 | xSmurf | no security updates |
| 21:34.07 | bmoraca_work | meh |
| 21:34.09 | Qwell | umm. You realize that there has been an announcement list, for like 8 years? |
| 21:34.14 | bmoraca_work | i've never really worried about that |
| 21:34.21 | Qwell | One that announces every single release since 1.4.0? |
| 21:35.08 | Qwell | I stand corrected. 1.2.0-beta1 |
| 21:35.19 | xSmurf | Qwell: that still means having to manually track everything, something that's not feasable for the end user |
| 21:35.27 | jpsharp | You mean there's versions past 1.2? |
| 21:35.44 | Qwell | xSmurf: Have a good day. |
| 21:35.53 | bmoraca_work | xSmurf: i don't know that there is a pbx that automatically updates itself... |
| 21:35.59 | Qwell | bmoraca_work: this is blowing my mind here |
| 21:36.07 | bmoraca_work | Qwell: makes no sense, huh? :( |
| 21:36.12 | Qwell | wait |
| 21:36.13 | xSmurf | bmoraca_work: in a production environment, yes security updates get done automatically |
| 21:36.19 | Qwell | when you did core show channels...did it output *nothing*? |
| 21:36.21 | bmoraca_work | i copied the config files 100% from the one that works |
| 21:36.45 | bmoraca_work | Qwell: it outputted the two calls, but nothing related to the phone i'm trying to provision |
| 21:36.52 | xSmurf | bmoraca_work: I'll see you maintain rack full of services up to date without them |
| 21:37.02 | Qwell | huh |
| 21:37.15 | bmoraca_work | xSmurf: 99% sure that CUCM doesn't update automatically. |
| 21:37.28 | bmoraca_work | either way, that's not a fun convo |
| 21:37.33 | xSmurf | that's one of the reason we don't use Cisco |
| 21:37.38 | xSmurf | (there are plenty of others) |
| 21:37.49 | bmoraca_work | Qwell: i tried all 3 versions of DPMA available for download |
| 21:37.56 | bmoraca_work | SELinux is off |
| 21:38.00 | bmoraca_work | (well, permissive) |
| 21:38.09 | Qwell | That channel is just disappearing. |
| 21:38.14 | bmoraca_work | yeah |
| 21:38.21 | Qwell | I don't even see it going away. |
| 21:38.48 | bmoraca_work | i see the destruction being scheduled |
| 21:38.54 | Qwell | that's the SIP dialog |
| 21:39.00 | Qwell | I mean the actual channel that's executing the dialplan. |
| 21:39.01 | bmoraca_work | ohh, the channel itself |
| 21:39.11 | bmoraca_work | yeah, nothing about that in the log |
| 21:39.14 | bmoraca_work | but let me check again |
| 21:41.08 | bmoraca_work | oh, interesting |
| 21:41.21 | bmoraca_work | i have one [Aug 30 14:15:44] VERBOSE[12833] pbx.c: -- Auto fallthrough, channel 'Message/ast_msg_queue' status is 'UNKNOWN' in my log |
| 21:41.23 | bmoraca_work | but only one |
| 21:41.34 | Qwell | O.o |
| 21:41.45 | bmoraca_work | and it's not preceeded by anything useful |
| 21:42.11 | bmoraca_work | oh |
| 21:42.12 | bmoraca_work | nm |
| 21:42.20 | bmoraca_work | that was from when i unloaded the dpma module |
| 21:42.23 | bmoraca_work | and tried it |
| 21:42.27 | bmoraca_work | irrelevant! |
| 21:42.37 | Qwell | ahh |
| 21:42.46 | bmoraca_work | cat full | grep Message/ast_msg_queue <== the command i ran... |
| 21:44.32 | Qwell | bmoraca_work: Are you able to test with 10.6.0-digiumphones and DPMA 1.2.0? |
| 21:44.40 | bmoraca_work | not until later tonight |
| 21:44.54 | Qwell | if you get a chance, I'd try that. I really don't know what to say at this point. |
| 21:44.57 | bmoraca_work | server is in production |
| 21:45.00 | bmoraca_work | ok |
| 21:45.02 | bmoraca_work | i'll try it |
| 21:45.14 | Qwell | I mean, that's just such an abrupt stop |
| 21:45.23 | Qwell | It's a Set() ffs |
| 21:45.46 | bmoraca_work | i know |
| 21:45.46 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 21:46.22 | bmoraca_work | i'll downgrade the test machine to 10.6.0-digiumphones |
| 21:46.24 | bmoraca_work | just for fun |
| 21:46.26 | bmoraca_work | see if it works |
| 21:46.48 | Qwell | well, let me know tomorrow |
| 21:47.00 | bmoraca_work | oh i will :P |
| 21:48.04 | bmoraca_work | wtf, did 10.7.1 get released today? |
| 21:48.11 | bmoraca_work | yes, lol |
| 21:48.12 | Qwell | bmoraca_work: about an hour ago |
| 21:48.27 | bmoraca_work | should i try that or should i try 10.6? |
| 21:48.29 | Qwell | If you were on the asterisk-announce list, you'd know that. ;) |
| 21:48.33 | bmoraca_work | rofl |
| 21:48.34 | Qwell | it's a security fix |
| 21:48.37 | bmoraca_work | ahh |
| 21:48.39 | bmoraca_work | nm then |
| 21:53.54 | *** join/#asterisk Ta^3 (~tacvbo@fixed-203-101-206.iusacell.net) |
| 21:56.16 | bmoraca_work | Qwell: just fyi, i removed NAT from the equation and it did not resolve the issue |
| 22:00.07 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xixjghemosqmskjb) |
| 22:07.43 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:09.35 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 22:17.10 | FuriousGeorge | what is the official distro and docs for * install these days? |
| 22:17.20 | FuriousGeorge | and HI ALL :) |
| 22:17.59 | *** part/#asterisk WolWid (WolWid@pD9ED4B7C.dip.t-dialin.net) |
| 22:18.34 | FuriousGeorge | I shouldn't have said "official disto". I should have said "preferred" |
| 22:18.49 | FuriousGeorge | e.g. CentOS? |
| 22:19.22 | WIMPy | Don't ask others. Use what you're comfortable with. |
| 22:21.18 | FuriousGeorge | WIMPy: reason I ask is because if you search "asterisk installation" on the google machine these days, this is the third result, and it's the 1st result from asterisk.org: http://www.asterisk.org/downloads/yum |
| 22:22.43 | FuriousGeorge | as you nsee that's for 1.6. i dont care about the distro, but I'm having weird issues and so I wanna start again from scratch just as a asanity check |
| 22:23.22 | WIMPy | First of all, you don't want to use anything older than 1.8. |
| 22:24.07 | FuriousGeorge | so i've been told in this #. wish i had known that before following what appear to be the official docs on *.org |
| 22:24.08 | WIMPy | And if you don't mind compiling yourself, I'd do that. |
| 22:24.20 | FuriousGeorge | i dont mind |
| 22:25.08 | FuriousGeorge | WIMPy: any thoughts on this: http://pve.proxmox.com/wiki/DebPBX |
| 22:25.14 | FuriousGeorge | happen to be using proxmox anyway |
| 22:25.42 | FuriousGeorge | (gentoo refugee, and BeOS before that ;)) |
| 22:28.24 | *** part/#asterisk str8uplinux (~dschuett@mail.lonemountaintruck.com) |
| 22:29.45 | FuriousGeorge | shoot, just noticed that's 1.4 |
| 22:31.09 | FuriousGeorge | so im gonna go with CentOS and * compile and if anyone has any reason that these two should never be wed i holy matrimony speak now pls |
| 22:32.08 | WIMPy | Well, I could mention that we learned bad things about RedHat kernels some weeks ago. But that will only be of interest if you want to use mISDN. |
| 22:33.09 | FuriousGeorge | no ISDN here. but I will be doing it in a Proxmox OpenVZ node if that makes a difference to you or anyone |
| 22:33.13 | *** part/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 22:37.28 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 22:38.09 | *** join/#asterisk wonderworld (~ww@dsdf-4db5d2e9.pool.mediaWays.net) |
| 22:54.36 | *** join/#asterisk darkdrgn2k (~darkdrgn@69-165-131-20.dsl.teksavvy.com) |
| 22:54.52 | darkdrgn2k | hi |
| 22:55.02 | darkdrgn2k | cany anyone suggest an IP door intercom |
| 22:55.05 | darkdrgn2k | can even |
| 22:55.58 | *** join/#asterisk italorossi (~Adium@187.61.182.39) |
| 22:59.47 | navaismo | 2Helios IP from 2N, ITS PANTEL or Surix |
| 23:02.08 | darkdrgn2k | 2helios seems to complicated for my need |
| 23:02.20 | darkdrgn2k | i just need a "press a button" and the phone rings |
| 23:02.40 | *** part/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 23:03.16 | _Corey_ | darkdrgn2k: Go with one of the analog ones connected to an ATA... you'll save a lot of money |
| 23:03.45 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 23:04.22 | darkdrgn2k | any suggestions on those? |
| 23:04.46 | _Corey_ | Bogon and Valcom both make them, if I'm not mistaken |
| 23:05.21 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:05.23 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-23-220.bois.qwest.net) |
| 23:06.05 | navaismo | helios IP has a one button model, but you can go to its pantel or surix too |
| 23:06.07 | _Corey_ | s/Bogon/Bogen/ |
| 23:08.32 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.37) |
| 23:32.25 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
