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| 02:06.24 | drmessano | I wonder if there is a record for running fairly frequent updated release branch SVN and dodging major bugs |
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| 02:23.11 | shmaltz | where did zap show status go? |
| 02:23.57 | p3nguin | You likely lost it when you stopped using zaptel and started using dahdi. |
| 02:25.24 | shmaltz | p3nguin, thanks, now whats the equi command in 1.6.x |
| 02:25.25 | shmaltz | ? |
| 02:25.50 | p3nguin | Take a guess. |
| 02:26.20 | shmaltz | p3nguin, I'm done guessing have spent too much time on this already |
| 02:26.24 | shmaltz | where is load modules? |
| 02:26.29 | TJNII | likes p3nguin's style tonight |
| 02:26.29 | p3nguin | If you don't use zaptel because it was replaced with dahdi, and you wanted to use zaptel show status... |
| 02:26.58 | shmaltz | p3nguin, dahdi does't work |
| 02:27.10 | shmaltz | how do I see what modles are loaded? |
| 02:27.21 | p3nguin | module show |
| 02:28.09 | p3nguin | You could also run "module show like dahdi" to see what's there. |
| 02:30.12 | shmaltz | p3nguin, I got this: |
| 02:30.14 | shmaltz | Module Description Use Count |
| 02:30.16 | shmaltz | chan_dahdi.so DAHDI Telephony Driver w/PRI 0 |
| 02:30.24 | shmaltz | but dahdi blah doesn't work |
| 02:33.43 | shmaltz | ~pb |
| 02:33.44 | 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.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. |
| 02:34.25 | shmaltz | why am I getting this: |
| 02:34.27 | shmaltz | http://pastebin.com/5m7J85ud |
| 02:37.47 | drmessano | is dahdi running? |
| 02:39.37 | shmaltz | drmessano, no and I beleive because of that errror |
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| 02:40.10 | drmessano | No, I didn't ask if it was loaded in Asterisk |
| 02:40.40 | shmaltz | drmessano, |
| 02:40.42 | shmaltz | ls /dev/dahdi/ |
| 02:40.43 | shmaltz | 1 2 3 4 channel ctl pseudo timer |
| 02:41.25 | shmaltz | drmessano, the bottom of this post has my dmesg output: |
| 02:41.27 | shmaltz | http://pastebin.com/XaXNNGLt |
| 02:43.08 | drmessano | HAve you rebooted during all this, and if so, have you rebuild dahdi after the reboot? |
| 02:43.13 | drmessano | and asterisk? |
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| 02:46.18 | shmaltz | drmessano, thanks for your help, didn't realize they renamed ztcfg to dahdi_cfg |
| 02:46.25 | shmaltz | took me a while to realize :P |
| 02:46.47 | ManxPower | Read The Readme |
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| 02:49.58 | drmessano | http://svnview.digium.com/svn/asterisk/tags/1.6.2.0/Zaptel-to-DAHDI.txt |
| 02:50.40 | shmaltz | ManxPower, I did, I couldn't find it there |
| 02:50.49 | shmaltz | drmessano, thats a little usefull one |
| 02:50.53 | shmaltz | is that part of the tar |
| 02:50.56 | shmaltz | is checking |
| 02:52.04 | shmaltz | nope doesnt' exist in the tar |
| 02:53.25 | drmessano | It's been in every tarball for 2 years |
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| 03:00.19 | shmaltz | drmessano, I can't find it in the one I have |
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| 03:02.35 | drmessano | It's in the top level directory |
| 03:03.06 | shmaltz | root@bunim:/usr/src/dahdi-linux-complete-2.3.0+2.3.0# ls |
| 03:03.08 | shmaltz | ChangeLog Makefile README build_tools/ linux/ tools/ |
| 03:03.20 | drmessano | [22:49] <drmessano> http://svnview.digium.com/svn/asterisk/tags/1.6.2.0/Zaptel-to-DAHDI.txt |
| 03:03.24 | drmessano | ^^^^^ Asterisk |
| 03:04.13 | shmaltz | oh it's here: |
| 03:04.15 | shmaltz | /usr/src/dahdi-linux-complete-2.3.0+2.3.0/linux# ls |
| 03:04.17 | shmaltz | ChangeLog LICENSE LICENSE.LGPL Makefile README UPGRADE.txt build_tools/ doc/ drivers/ include/ |
| 03:04.19 | shmaltz | ChangeLog has it |
| 03:04.26 | shmaltz | let me check asterisk |
| 03:04.45 | shmaltz | yest it is, thank you drmessano |
| 03:05.56 | shmaltz | is compiling spandsp for the first time |
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| 04:04.21 | joobie | guys, how can i set the music on hold context for a sip handset? seems to default to 'default' |
| 04:04.29 | joobie | .. when i press the hold button on the phone |
| 04:37.11 | p3nguin | joobie: Try setting mohinterpret and mohsuggest either globally or in the peer definition. |
| 04:38.20 | [TK]D-Fender | joobie: musicclass=someotherclass |
| 04:42.53 | ManxPower | Try reading sip.conf.sample! |
| 04:42.55 | p3nguin | That must be new. |
| 04:43.57 | p3nguin | And since no version was specified during the inquiry, I have to assume the version being used is the exact version I use. |
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| 05:24.24 | joobie | thanks TK |
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| 05:45.48 | joobie | TK, doesnt seem to work on 1.4? |
| 05:46.39 | joobie | hmm.. i can see it can be used in the dialplan |
| 05:46.47 | joobie | TK, is that meant to be for sip.conf |
| 05:46.48 | joobie | ? |
| 05:47.08 | [TK]D-Fender | joobie: Clearly |
| 05:47.49 | [TK]D-Fender | [00:04]<joobie>guys, how can i set the music on hold context for a sip handset? seems to default to 'default' <- You asked phone-level, so that's what I gave you |
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| 06:29.57 | joobie | that's dialplan level |
| 06:30.11 | joobie | or maybe that's how i interpreted it |
| 06:30.14 | joobie | all good TK. |
| 06:30.22 | joobie | the option i used was mohsuggest |
| 06:30.24 | joobie | in sip.conf |
| 06:30.25 | joobie | cheers |
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| 06:47.20 | vk4akp | The caller ID problem for the TDM400P card has been solved. |
| 06:47.43 | vk4akp | Simple one line entry in Xaptel.conf |
| 06:47.53 | vk4akp | Zaptel.conf* |
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| 07:06.52 | shamelessn00b | Hello |
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| 07:29.28 | ChannelZ | ohell |
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| 07:40.37 | shamelessn00b | sup ChannelZ |
| 07:41.09 | shamelessn00b | ZlennahC |
| 07:41.21 | shamelessn00b | o.o |
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| 07:48.24 | ChannelZ | not a lot |
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| 07:51.08 | shamelessn00b | same, working on sphinx4 these days |
| 07:51.49 | shamelessn00b | Improving accuracy in noisy environments |
| 07:52.13 | *** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty) |
| 07:52.14 | shamelessn00b | and then integrating with asterisk, using zanzibar(an mrcp related project) |
| 07:52.16 | ChannelZ | Cool. I have no idea what that is. |
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| 07:56.36 | ChannelZ | Oh. Voice recognition |
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| 08:04.56 | shamelessn00b | yeah |
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| 08:57.54 | Boter | hey everybody |
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| 08:59.13 | Boter | where can i read info about compiling and setting asterisk to work with some ip phone i got? |
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| 09:03.40 | Boter | i guess i need sip? |
| 09:03.42 | UQlev | Boter: 1st make sure your asteris work with any softphone |
| 09:06.25 | Boter | i just compiled asterisk |
| 09:06.37 | Boter | so probably i need first some pbx? |
| 09:06.37 | Boter | eg freepbx? |
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| 09:20.19 | Boter | ok going through configs |
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| 09:26.31 | Boter | hm |
| 09:26.34 | Boter | x-fire logins |
| 09:26.40 | Boter | call failed: not found |
| 09:26.44 | Boter | that's what i get |
| 09:28.06 | Boter | oh ok my bad |
| 09:28.08 | Boter | reading more :) |
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| 10:07.51 | Infin1ty | what's the best way to log all incoming call notfication, i'm trying to extensions.conf but i can't seem to use the System() correctly |
| 10:08.15 | Chainsaw | Infin1ty: Generally that's done with CDR (Call Dispatch Reporting). |
| 10:10.32 | Infin1ty | oh i see, can i tell it to call an external app as well? |
| 10:17.26 | Chainsaw | Infin1ty: There's a asterisk-cdr_shell that should be able to do that. Not sure whether that still works on 1.6 though. |
| 10:20.48 | Faustov | [May 24 11:28:10] WARNING[13627]: channel.c:3697 set_format: Unable to find a codec translation path from 0x400 (ilbc) to 0x8 (alaw) |
| 10:20.56 | Faustov | I'm seeing this on some incoming calls |
| 10:21.03 | Faustov | any idea how I could handle it? |
| 10:21.24 | Faustov | where/how are "codec translation paths" configured? |
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| 10:26.36 | Chainsaw | Faustov: That means you allowed ilbc on one endpoint, but not the other. |
| 10:26.54 | Chainsaw | Faustov: So instead of just passing through ilbc, Asterisk would have to translate between them. It is telling you that it doesn't have hardware or software that can do it. |
| 10:27.09 | Faustov | Chainsaw: i've allowed ilbc on the server i'm observing this |
| 10:27.25 | Chainsaw | Faustov: Sure. But it is wanting to talk to something else that can only do alaw. |
| 10:28.15 | Faustov | hmmm |
| 10:28.58 | Faustov | but alaw is first in my codec order |
| 10:29.05 | Faustov | and ilbc is last |
| 10:29.12 | Chainsaw | Faustov: Ordering is not as significant as you think it is I'm afraid. |
| 10:29.32 | Chainsaw | Faustov: If you allow ilbc, the device may decide to use ilbc. If the rest of your network can't do it, perhaps it's best to leave it off. |
| 10:30.06 | Faustov | how is the codec chosen then? as far as I've observed in the sip debug, the "highest match" from allowed codecs on both lists is chosen |
| 10:30.10 | Faustov | is this not the case? |
| 10:30.40 | Chainsaw | Faustov: The case is that ilbc got chosen on one of your endpoints. |
| 10:31.03 | Chainsaw | Faustov: Your other endpoint can't do it, and this is why Asterisk would have to translate. It can't, because there's no hardware/software capable of doing so. Hence your warning. |
| 10:31.14 | Faustov | mhm ok |
| 10:31.24 | Faustov | i'll review configure logs |
| 10:31.29 | Chainsaw | Faustov: Cheap fix: Disable ilbc. Expensive fix: Buy licenses for transcoding. Even more expensive fix: Buy a transcoding board that can do it. |
| 10:31.34 | Faustov | maybe one end doesn't have it compiled in |
| 10:31.54 | Chainsaw | Faustov: That makes sense, yes. |
| 10:32.02 | *** join/#asterisk Jumpie (n3rdz@ip68-230-28-186.ph.ph.cox.net) |
| 10:32.17 | Jumpie | heh i tried to make my gmail theme the green on black terminal theme |
| 10:32.21 | Jumpie | thought i could handle it |
| 10:32.26 | Jumpie | melted my mind |
| 10:33.24 | Chainsaw | Jumpie: It really melted your mind on the old screens. They're not allowed anymore. |
| 10:33.45 | Chainsaw | Jumpie: I recommend amber on black over green on black, there's a reason those amber ones were more expensive. |
| 10:34.52 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:36.12 | Chainsaw | Infin1ty: Another option would be to write a CDR module yourself: http://www.russellbryant.net/blog/2008/06/20/how-to-write-an-asterisk-module-part-2/ |
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| 11:11.34 | shamelessn00b | '-' |
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| 11:21.22 | Dovid | hi. anyone know of any G729 issues against specific switches where the audip dies (and not the rtp) |
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| 11:33.55 | joobie | Dovid, never heard of it |
| 11:34.15 | joobie | i'd blame the switch tho |
| 11:34.18 | joobie | rather than g729 |
| 11:34.31 | joobie | dont forget - your switch was made in china |
| 11:34.38 | Faustov | is there any ready way to obtain statistics for codecs? As in, check how often which codec has been chosen for a call? |
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| 11:48.29 | Dovid | joobie: Were not talking about a simple switch. talking about the big boys |
| 11:48.37 | Dovid | when I use G711U then there is no issue |
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| 12:07.23 | joobie | Dovid, what switch? |
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| 12:11.29 | file | everyone! SAFETY DANCE! |
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| 12:36.03 | Lantizia | Hey has anyone done much with the Aastra XML scripts? |
| 12:37.16 | Lantizia | I've got a simple AastraIPPhoneInputScreen file generating by a perl script to just take a number and then load the same URL with the extra parameter |
| 12:37.31 | Lantizia | It works fine... but destroyOnExit="yes" doesn't get rid of the screen after it's done :S |
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| 12:57.25 | joelsolanki | hello all |
| 12:57.53 | joelsolanki | linksys pap2 --> sip proxy switch --> provider |
| 12:58.06 | joelsolanki | linksys using g723 codec on both lines. it give choppy voice |
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| 12:58.42 | joelsolanki | if i keep asterisk between sip proxy switch and provider and do transcoding for all g723 to g279 calls, will that fix the voice quality problem ? |
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| 13:33.57 | Dovid | joelsolanki: There could be a lot of reasons for the issue. 1) Codec issues, 2) Bandwidth issue (which is mot likely the case) |
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| 13:44.30 | tlir | is 60 the lowest asterisk value for the extension's registration expiration? |
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| 13:45.07 | tlir | I've set it to 1 second in my sip extension and I verified that Expire : 1 is passed in the sip packet |
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| 13:51.01 | ManxPower | ~answers |
| 13:51.02 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 13:55.14 | devmod | Hello, any pointers on getting video to work on h323? |
| 13:59.04 | ManxPower | I don't think anyone here right now is crazy enough to try. H323 support in Asterisk is horrible |
| 13:59.59 | devmod | haha |
| 14:00.25 | devmod | i dont think there are any good h323-sip gw for free out there either |
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| 14:02.11 | tDOTzillaCHYEA | is anybody familiar with google voice and ipkall? |
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| 14:08.17 | tDOTzillaCHYEA | is anybody in here? |
| 14:10.50 | Baylink1 | Nope. |
| 14:10.55 | Baylink1 | ?ask |
| 14:11.01 | Baylink1 | <sigh> |
| 14:11.03 | Baylink1 | ~ask |
| 14:11.04 | 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. |
| 14:11.06 | Faustov | :D |
| 14:11.25 | Baylink1 | FPBX's bot uses the opposite one, I can never keep them straight. |
| 14:12.22 | Faustov | is there any way to obtain statistics for codecs? As in, check what codecs are being chosen most of the time? |
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| 14:15.02 | tDOTzillaCHYEA | CHYEAH! |
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| 14:22.35 | devmod | lol 1.6.2.8-rc1/ 261557 2 weeks lmadsen Update ChangeLog. Tick tock on the clock. Shoutouts to kpfleming and DJ Funky F... |
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| 14:41.28 | Lantizia | Hey, how much do you think a 2nd hand TE110P/TE120P would tell for? |
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| 14:42.06 | Baylink1 | You check eBay? :-) |
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| 14:42.15 | Lantizia | Baylink1, yeah can't see any |
| 14:42.29 | Baylink1 | Did you look in Completed Auctions, too? |
| 14:42.36 | Lantizia | ah no :) |
| 14:44.11 | Lantizia | Baylink1, still nothing :S this is odd |
| 14:44.42 | Baylink1 | A bit. |
| 14:44.47 | Baylink1 | Which card is that? |
| 14:45.02 | Lantizia | http://completed.shop.ebay.co.uk/i.html?_nkw=digum&_in_kw=1&_ex_kw=&_sacat=See-All-Categories&_okw=digum&_oexkw=&LH_Complete=1&_udlo=&_udhi=&_samilow=&_samihi=&_sadis=200&_fpos=Postcode&LH_SALE_CURRENCY=0&_sop=12&_dmd=1&_ipg=50&_rdc=1 |
| 14:45.16 | Lantizia | thats just a search for all completed listings with the word digium in it |
| 14:47.44 | Baylink1 | Yeah, no, I just don't remember the description of those cards; which ones are they? |
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| 14:48.45 | Lantizia | oh single ISDN30 |
| 14:49.02 | Baylink1 | So, T-1/E-1 cards. |
| 14:49.06 | Lantizia | yes |
| 14:49.40 | Baylink1 | I have *seen* 1-port PRI cards down as far as a couple hundred bucks. All I buy these days are A104 Sangomas; new, I pay about $1300 for those. |
| 14:49.58 | Baylink1 | If you don't need one this minute, put a recurring search in on eBay and wait. |
| 14:50.27 | Lantizia | we want to sell two TE110P's (single PRI)... and 2 quadBRI cards |
| 14:50.38 | Lantizia | So we can get ISDN to SIP gateways instead |
| 14:50.49 | Lantizia | The cards are too much of a pain in the ass |
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| 14:56.16 | jtexter3 | I have a system where sometimes I start getting one-way audio on my PRI lines |
| 14:56.44 | jtexter3 | During this time, I see "chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel xxx" for several channels |
| 14:56.57 | jtexter3 | Anyone seen this before? |
| 14:57.07 | jtexter3 | Unplugging the PRI and plugging back in seems to clear it up |
| 14:59.05 | Baylink1 | Have you turned up debugging on the span in question? |
| 14:59.34 | tzafrir_laptop | jtexter3, what type of device is it? What version of Asterisk? |
| 14:59.59 | tzafrir_laptop | In the middle of a call? |
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| 15:03.13 | jtexter3 | tzafrir_laptop: Sangoma 8 span, Asterisk 1.6.2.7 with libpri-trunk |
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| 15:04.07 | jtexter3 | Baylink1: Not yet, that's my next step |
| 15:04.21 | Baylink1 | Do you only have the one span? |
| 15:04.50 | jtexter3 | Baylink1: No, I have 18 |
| 15:05.12 | jtexter3 | The only other thing I see in the log is "chan_dahdi.c: Ring requested on channel 0/18 already in use or previously requested on span 19. Attempting to renegotiate channel." |
| 15:05.18 | Baylink1 | Is it only the one span causing the problem? |
| 15:05.37 | Baylink1 | If so, what's your carrier spread? |
| 15:06.33 | jtexter3 | Seems to affect the spans from one carrier, AT&T |
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| 15:08.45 | Baylink1 | Are you sure that the SWITCHTYPE you have set matches what they expect (which *should* be 5E, but you never know...)? Often, incorrectly set SWITCHTYPEs will still *work*, but less reliably in edge-cases. |
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| 15:31.25 | devmod | any ideas on getting video to work on h323? |
| 15:32.55 | [TK]D-Fender | devmod: Tried allowing the codecs? |
| 15:33.19 | devmod | yes, but nothing happened. I can't find any up to date info on it either :/ |
| 15:33.29 | devmod | audio works fine btw |
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| 15:41.32 | raden_work | devmod, what happens exactly ? |
| 15:41.48 | Kobaz | how do i lessen the number of post-voicemail cycles there are.... for instance I have maxsecs to be 120... but the line is stuck in voicemail for at least 5 minutes cycleing through vm-review.ulaw and vm-reachoper.ulaw about 20 times before it actually hangs up |
| 15:42.09 | devmod | raden_work: I get audio but no video between endpoints |
| 15:42.42 | raden_work | devmod, are u sure thats the codecs asterisk is using ? |
| 15:42.59 | devmod | i have only enabled h263 |
| 15:43.14 | devmod | for sip calls audio and video works fine |
| 15:44.03 | [TK]D-Fender | Kobaz: maxlen, not maxsecs IIRC |
| 15:44.28 | [TK]D-Fender | Kobaz: And it sounds like you have a disconnect supervision issue |
| 15:44.33 | Kobaz | [TK]D-Fender: the voicemail recording time limiting is working |
| 15:44.43 | Kobaz | [TK]D-Fender: yes it is a disconnect supervision issue |
| 15:44.57 | [TK]D-Fender | Kobaz: You could also remove the review option. |
| 15:44.58 | Kobaz | [TK]D-Fender: it's minsecs and maxsecs in 1.6.0 |
| 15:45.03 | Kobaz | okay, that works |
| 15:45.07 | [TK]D-Fender | Kobaz: that will cut back the looping at the end |
| 15:45.09 | Kobaz | yeah |
| 15:45.12 | Kobaz | the looping is killing me |
| 15:45.37 | Kobaz | the stupid thing is disconnect supervision works about 50% of the time |
| 15:45.53 | Kobaz | i don't really want to bother troubleshooting since a t1 is going in, in two weeks |
| 15:47.22 | Kobaz | s - Skip the playback of instructions for leaving a message to the calling party |
| 15:47.30 | Kobaz | is that it? i don't see any options for skipping review |
| 15:47.33 | Kobaz | i can just comment out the code |
| 15:47.34 | [TK]D-Fender | Kobaz: Disable the review then |
| 15:47.48 | [TK]D-Fender | Kobaz: CONFIG option, not app call |
| 15:47.57 | Kobaz | oh, voicemail config, k |
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| 15:50.58 | KavanS | does anyone know how "color" would be disabled on the asterisk console? |
| 15:50.58 | KavanS | previous version of 1.4 had color, now this later version does not |
| 15:50.58 | KavanS | did I miss something during the compile stage? |
| 15:50.58 | KavanS | (this is also a new OS, so I suppose it could be a terminal setting) |
| 15:50.58 | Kobaz | k, turned off review |
| 15:50.58 | Kobaz | yay |
| 15:51.02 | Kobaz | [TK]D-Fender: i really, really, really.... hate analot |
| 15:51.04 | Kobaz | analog |
| 15:51.15 | doolittlework | hi there i am using the monitor command, works like a charm or not, when i playback the file i can hear the far side voice clear but my voice is almost like it is breaking up, note that while i am on the call there is no such problem only on playback, any suggestions where to start faultfinding |
| 15:51.30 | devmod | is /trunk/addons not added to the release tarballs ? |
| 15:51.33 | Kobaz | there's a line cross issue too... every so often people hear really loud fax beeps and stuff |
| 15:51.34 | doolittlework | using snom phone with siptrunk out using g729 codeex |
| 15:52.33 | Kobaz | doolittlework: don't use monitor, use mixmonitor |
| 15:52.48 | [TK]D-Fender | devmod: Trunk is what will become 1.8 |
| 15:53.06 | devmod | [TK]D-Fender: ohh ok got it |
| 15:53.21 | Kobaz | doolittlework: monitor does it's recording in the audio bridging thread... so the audio will be affected by disk activity... mixmonitor does recording in a seperate thread |
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| 15:57.01 | grandpapadot | Hey guys, how many (rough estimate) non-transcoded G.729 channels can a asterisk 1.4/1.6 system handle on a quade-core system? Most calls in a queue, only 25 agents though. |
| 15:57.27 | doolittlework | Kobaz: i will look into that but i am also experiencing that klik klik sound on the ringtone of outgoing calls, can this affect it, if i make an outbound call i get the click but after connect voice is normal both sides. |
| 15:58.01 | devmod | [TK]D-Fender: can the trunk be considered relatively stable? |
| 15:58.05 | Kobaz | doolittlework: you may get audio artifacts when using Monitor() |
| 15:58.05 | grandpapadot | .. we're expecting to handle 1000-1500 calls |
| 15:58.07 | Kobaz | devmod: hell no |
| 15:58.18 | devmod | Kobaz: |
| 15:58.20 | [TK]D-Fender | devmod: Of course not |
| 15:58.35 | Kobaz | trunk = latest bleeding edge |
| 15:58.39 | [TK]D-Fender | grandpapadot: More than enough |
| 15:58.41 | devmod | oh all right |
| 15:58.43 | Kobaz | and they dont call it the bleeding edge for nothing |
| 15:58.52 | Kobaz | bring lots of bandaids |
| 15:59.07 | devmod | I was interested in ooh323 channel, i wonder if it is modular enough to work on 1.6 |
| 15:59.08 | grandpapadot | [TK]D-Fender: 1000-1500 simultaneous? Also, would you recommend 1.4.latest or 1.6.current for this setup? |
| 15:59.28 | Kobaz | ooh323 will build on 1.6 |
| 15:59.36 | [TK]D-Fender | grandpapadot: Uh.. what happened to 25? |
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| 15:59.47 | Kobaz | grandpapadot: if you want something super solid, 1.4 |
| 15:59.52 | [TK]D-Fender | grandpapadot: 1000.... WTF are they doing? |
| 15:59.56 | Kobaz | 1.2 even better |
| 15:59.58 | grandpapadot | [TK]D-Fender: 25 agents, 1000-1500 inbound g.729 sip calls, no transcoding |
| 16:00.12 | Kobaz | how can 25 agents do 1000+ concurrent calls? |
| 16:00.13 | grandpapadot | [TK]D-Fender: it's a call center, garth brooks concert ticket sales line |
| 16:00.17 | devmod | Kobaz: all right, will test it thanks |
| 16:00.22 | [TK]D-Fender | grandpapadot: Then you might get by on 1 serioulsy beefy box. |
| 16:00.35 | Kobaz | grandpapadot: 1000 people waiting in queue? |
| 16:00.38 | grandpapadot | [TK]D-Fender: they expect it to sell out fast, but the callers need to hear the prompts ... |
| 16:01.11 | Kobaz | grandpapadot: get 3-4 asterisk boxes |
| 16:01.40 | grandpapadot | Kobaz: yea, that's the idea, but I was hoping for some real-world thoughts on how much a 1.4 vs 1.6 quade core box could simultaneously handle. |
| 16:02.01 | Kobaz | there's some anecdotal load tests on the voip wiki |
| 16:02.04 | grandpapadot | So what's a reasonable non-transcoding per-box number? 500? 750? |
| 16:02.14 | Kobaz | some people say you can put about 500 per box |
| 16:02.25 | Kobaz | i've never went past more than 100 or so |
| 16:02.40 | lirakis | grandpapadot, it totally depends on what you want to do |
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| 16:02.59 | lirakis | grandpapadot, any kind of audio mixing or transcoding will kill the machine |
| 16:03.07 | grandpapadot | lirakis: play a prompt, put them in a queue, NO TRANSCODING or MIXING |
| 16:03.17 | lirakis | grandpapadot, MOH while in queue? |
| 16:03.33 | grandpapadot | lirakis: Yep, but native, already in the channel format. |
| 16:03.41 | lirakis | grandpapadot, still .. requires lots of resources |
| 16:03.41 | Kobaz | i wonder how many concurrent calls the american idol servers handle |
| 16:03.56 | doolittlework | thanks kobaz did the trick, thx dude |
| 16:03.57 | [TK]D-Fender | and no RECORDING <- |
| 16:04.11 | lirakis | .. i setup a call center system taht did i think ... 50 simultaneous calls (up and in queue.. so total) before MOH got jitter. |
| 16:04.15 | lirakis | all g711 |
| 16:04.15 | lirakis | ulaw |
| 16:04.27 | Kobaz | doolittlework: Monitor() is legacy code, and really doesn't work well |
| 16:04.40 | lirakis | and this was on .. not a quad... but .. some pentium 3ghz xeon thing... a year or two ago |
| 16:04.47 | grandpapadot | Well, we're getting 100-150 now on our other 1.4 systems, so I know it's a lot more than that ... |
| 16:04.51 | Kobaz | doolittlework: you know how long it took me to figure out that problem you're having? like 5 months |
| 16:05.17 | Kobaz | lirakis: sip or dahdi? |
| 16:05.19 | lirakis | grandpapadot, im just saying .. it totally depends on whats going on .. your codecs.. etc. etc. so .. as Kobaz said... any metrics are anecdotal |
| 16:05.24 | lirakis | Kobaz, sip |
| 16:05.31 | Kobaz | what sort of machine? |
| 16:05.48 | grandpapadot | lirakis: no transcoding, play a prompt, stack a queue, 25 agents |
| 16:06.02 | grandpapadot | Kobaz: quade core i5 |
| 16:06.07 | *** join/#asterisk slashtom (~tom@k-rad.co.uk) |
| 16:06.11 | lirakis | Kobaz, dont remember exactly ... it was 2 years ago .. i believe a 3ghz dual xeon ... not sure of the ram ... prob. 6-8gb or some thing |
| 16:06.19 | grandpapadot | 4gb |
| 16:06.21 | Kobaz | mm |
| 16:06.39 | Kobaz | xeon isn't all that great for multiprocessing though |
| 16:06.51 | Kobaz | hyperthreading was a hack before multi-core |
| 16:06.59 | lirakis | Kobaz, yeah .. like i said .. a while ago (one of my first "real" installs) |
| 16:07.04 | *** join/#asterisk fishloa (fishloa@87-194-32-209.bethere.co.uk) |
| 16:07.05 | grandpapadot | Kobaz: hyperthreading is back with the i7 |
| 16:07.13 | lirakis | so im sure i did some stupid stuff too |
| 16:07.14 | lirakis | ;) |
| 16:07.26 | *** join/#asterisk ccd (~default@ip70-176-18-1.ph.ph.cox.net) |
| 16:07.35 | Kobaz | i would turn off hyperthreading and just run on the bare cores |
| 16:07.46 | Kobaz | unless they made huge improvements since the originals |
| 16:08.17 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:09.07 | fishloa | could somebody help me with DADHI/Wildcard AEX410/asterisk-1.6.2.7. Incoming calls work find, with CID, outgoing only works intermitently, with DADHI not actually dialling |
| 16:09.09 | *** join/#asterisk lordoxide (~chatzilla@206.183.2.183) |
| 16:09.50 | grandpapadot | ok, we'll just start at 250 and work our way up real-time |
| 16:10.34 | lordoxide | sup all, quick question. We are looking for some parsing software, or settings in asterisk that can log all verbose/debug information per call, in individual files. Any have any advise, basically id like to be able to look at a directory with files named "[callerid timestamp].log |
| 16:11.10 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 16:11.21 | fishloa | any DADHI gurus here? who could help fix a UK dialout issue? |
| 16:11.33 | [TK]D-Fender | lordoxide: Not really possible |
| 16:11.38 | *** part/#asterisk tlir (~tlir@bzq-84-110-103-46.red.bezeqint.net) |
| 16:12.18 | lordoxide | [TK]D-Fender: not even with 3rd party software, it would make troubleshooting issues post occurence so much easier =/ |
| 16:12.59 | [TK]D-Fender | lordoxide: the information isn't clearly marked in a way to parse out an entire call |
| 16:14.52 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:15.11 | Kobaz | lordoxide: we have a completely custom logging system, and everything that needs logging is written in agi |
| 16:15.30 | grandpapadot | Last question, I assume 1.6 is generally more optimized code so go with that for high call volume setups? |
| 16:15.59 | Kobaz | grandpapadot: the main issue is going to be stability really |
| 16:16.30 | grandpapadot | Kobaz: so for stability w/high call volume, go with trusty 1.4 or use 1.6? |
| 16:16.49 | grandpapadot | We have good luck with 1.4, but rarely see over 200 channels per switch. |
| 16:16.59 | [TK]D-Fender | Kobaz: Everything that you felt was worth it... and possible. |
| 16:17.12 | [TK]D-Fender | Kobaz: That isn't really comprehensive though... and can't be. |
| 16:17.38 | [TK]D-Fender | grandpapadot: 1.6 isn't a specific branch. There are *3* |
| 16:17.45 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
| 16:18.04 | Kobaz | grandpapadot: here's my anecdotal evidence... i had a 1.4 system up for 390 days before the first crash |
| 16:18.19 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:18.25 | grandpapadot | [TK]D-Fender: which branch would you trust the most for maximum calls vs lowest risk of becoming unstable? |
| 16:18.28 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 16:18.32 | Kobaz | grandpapadot: my longest 1.6.0.x system has been up for about 3 months |
| 16:18.41 | Chainsaw | tzafrir_laptop: https://issues.asterisk.org/view.php?id=17382 |
| 16:18.43 | Chainsaw | tzafrir_laptop: https://issues.asterisk.org/view.php?id=17383 |
| 16:18.57 | Kobaz | grandpapadot: my system that has the most crashes is also 1.6.0, and it rarely is up for more than a few days |
| 16:19.12 | Qwell | Chainsaw: is this another one of those Gentoo kernel patches that F things up? |
| 16:19.14 | lordoxide | Kobaz: we run an agi script and can log specific results by unique id or caller id currently, but certain output like when asterisk is actually dialing, responses based on 5xx errors, are hard to capture. I was hopting to find some port that could tie into the asterisk application and save all the debug/verbose information to individual files |
| 16:19.27 | Kobaz | lordoxide: good luck... heh |
| 16:19.33 | Chainsaw | Qwell: No, this is the upstream kernel moving quicker then you anticipated. |
| 16:19.57 | Chainsaw | Qwell: Vanilla kernel.org; what is to become 2.6.35-rc1. You can take them now, or you can take them later when things start breaking. Up to you. |
| 16:20.09 | [TK]D-Fender | lordoxide: that was the "no" I was talking about |
| 16:20.53 | *** part/#asterisk jtexter3 (~jtexter3@72.242.229.213) |
| 16:20.58 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:23.09 | *** join/#asterisk ruchir (~ruchir@117.196.98.139) |
| 16:23.13 | ruchir | hi all |
| 16:23.15 | *** part/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 16:23.17 | ruchir | i'm having weired issue |
| 16:23.23 | ruchir | we've been using agi scripts for billing |
| 16:23.31 | ruchir | script has some db operations and complex features |
| 16:23.46 | ruchir | however over some days, i see lots of agi scripts being hang |
| 16:23.55 | ruchir | and asterisk stops responding to new calls and rejectrs them |
| 16:23.57 | ruchir | we use php agi scripts |
| 16:24.01 | ruchir | any help? |
| 16:24.05 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 16:25.17 | Kobaz | fix your php scripts |
| 16:29.04 | *** join/#asterisk Trixboxer (~praju@115.113.145.85) |
| 16:29.18 | ruchir | what could be wrong |
| 16:29.22 | ruchir | its simple db lookup |
| 16:29.25 | ruchir | and features |
| 16:29.26 | ruchir | dial |
| 16:29.28 | ruchir | hangup |
| 16:29.29 | ruchir | cdr |
| 16:29.32 | ruchir | billing |
| 16:29.33 | ruchir | thats it |
| 16:29.39 | Kobaz | is your enter key stuck? |
| 16:29.54 | ruchir | nope |
| 16:29.56 | ruchir | why? |
| 16:30.01 | Corydon76-dig | ruchir: are you handling the HUP signal correctly? |
| 16:30.01 | Tim_Toady | lol |
| 16:30.10 | ruchir | i also noticed who msgs being printed repeatedly |
| 16:30.11 | Chainsaw | ruchir: You seem to overuse it. Please don't. |
| 16:30.16 | Kobaz | it's |
| 16:30.17 | Kobaz | reakky |
| 16:30.19 | Kobaz | rude |
| 16:30.19 | Kobaz | to |
| 16:30.21 | Kobaz | use |
| 16:30.21 | ruchir | i dont knw how its happening |
| 16:30.23 | Kobaz | one |
| 16:30.26 | Kobaz | word |
| 16:30.28 | Kobaz | per |
| 16:30.31 | Kobaz | line |
| 16:30.38 | ruchir | let me restart |
| 16:30.40 | chuckf | yeah |
| 16:30.54 | KavanS | ruchir, I was wondering if you are familiar with CONTROL-M |
| 16:31.11 | *** join/#asterisk ruchir (~ruchir@117.196.98.139) |
| 16:31.15 | ruchir | back |
| 16:31.35 | ruchir | we're doing db lookup, some call features, cdr, billing, etc from php agi |
| 16:32.06 | ruchir | as far as i know no infinite loops |
| 16:32.08 | ruchir | except dial |
| 16:32.14 | ruchir | as until hangup, it will stay there |
| 16:32.16 | Kobaz | control-m as in carrage return |
| 16:32.50 | Kobaz | KavanS: ? |
| 16:32.57 | KavanS | Kobaz, yep :P |
| 16:33.03 | KavanS | I was kidding... |
| 16:33.14 | Kobaz | ruchir: like i said, you need to add lots of debugging and see where your script is failing |
| 16:33.21 | ruchir | i see |
| 16:33.28 | ruchir | ok np we;ll check further |
| 16:33.41 | [TK]D-Fender | ruchir: You offer no real details to debug with |
| 16:33.48 | Kobaz | ruchir: it's like walking to a car shop *without* your car, and saying, it doesn't work... fix it... |
| 16:34.25 | ruchir | Kobaz: i understand but that is the only info i could get from prodyction system |
| 16:34.48 | KavanS | anyone have any clues on this asterisk color console thing? |
| 16:34.49 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:34.59 | KavanS | I'm reading I could modify the init script...but that's for people who want to disable color :\ |
| 16:35.00 | ruchir | nevermind i'll try to setup test environment |
| 16:35.16 | Kobaz | ruchir: good plan |
| 16:35.23 | ruchir | and get back |
| 16:35.27 | ruchir | thx for help so far |
| 16:41.57 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-ufhrzouvmzgpcsrx) |
| 16:42.09 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:42.47 | Corydon76-dig | KavanS: If you're running an older version of Asterisk, and you start up Asterisk at boot time, you won't have color |
| 16:43.31 | KavanS | ok, I am using asterisk via an init script, but I'm not starting it up at boot time - it's a cluster to be exact |
| 16:43.39 | *** join/#asterisk xa0z (~Interex@75-129-243-246.dhcp.mtvr.il.charter.com) |
| 16:43.46 | KavanS | so you're suggesting my issue is related to the init script? |
| 16:44.04 | Corydon76-dig | KavanS: It's specifically related to having the TERM env variable not set |
| 16:44.10 | xa0z | Anyone here with a Cisco 7970 or 7971 know of a way to register to a hostname, rather than an IP address, using any SIP image? |
| 16:44.16 | Corydon76-dig | It's set in logins, but not at boot |
| 16:45.07 | KavanS | ok |
| 16:45.13 | Corydon76-dig | If you set the TERM env variable to "vt100" or another terminal that permits color, then the Asterisk console will generate color |
| 16:45.51 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:46.06 | Corydon76-dig | This was revised to make Asterisk always generate color and strip the control codes for terminals which don't support it |
| 16:46.48 | KavanS | Corydon76-dig, ok roger that, thank you much for the explanation, I will do some further googling :) |
| 16:47.46 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:00.09 | xa0z | Or not :/ |
| 17:01.10 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 17:07.50 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:11.35 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 17:17.25 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-sufbtxrsnggxwlzk) |
| 17:18.09 | shader | hello |
| 17:18.27 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:23.42 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:24.12 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-177-17.mia.bellsouth.net) |
| 17:39.57 | *** join/#asterisk datarecal (~data@2607:fb98:4::2) |
| 17:40.16 | datarecal | trying to figure out what this error means : chan_sip.c:3114 auto_congest: Auto-congesting SIP/1962941343-00000014 i am getting the "all ciruits are busy now" |
| 17:42.56 | tzafrir_laptop | Chainsaw, sorry for the delay. I generally rather at least for -rc1 for the dust to begin to settle |
| 17:43.18 | Chainsaw | tzafrir_laptop: They look rather permanent to me, but as I said, up to you. It breaks now. |
| 17:44.15 | tzafrir_laptop | I figure the #include one will just go in. As for the other one: I wonder if there's a way to avoid a versioned case. |
| 17:44.26 | Chainsaw | tzafrir_laptop: Not really, the rename was only done in 2.6.34 |
| 17:44.26 | tzafrir_laptop | Or at least keep it separate |
| 17:45.55 | tzafrir_laptop | Anyway, they look good. I figure I'll commit them (at least to trunk) them when -rc1 will be out |
| 17:46.10 | Chainsaw | tzafrir_laptop: Cheers. |
| 17:47.13 | *** join/#asterisk errr (~errr@fedora/errr) |
| 17:47.41 | Trixboxer | datarecal: May be due to no trunks available |
| 17:47.53 | xa0z | Anyone here with a Cisco 7970 or 7971 know of a way to register to a hostname, rather than an IP address, using any SIP image? |
| 17:48.01 | datarecal | triboxer what does that mean |
| 17:48.22 | errr | How can I change the Return-path header set when a voicemail is sent to a user via email? servermail = set and that works but does not modify that return path header.. |
| 17:48.55 | Trixboxer | datarecal: are you trying to dial an extension or some outside number ? |
| 17:49.05 | datarecal | outside number |
| 17:49.14 | Trixboxer | datarecal: To dial outside you must have a trunk |
| 17:49.19 | Trixboxer | registered |
| 17:49.34 | datarecal | yeah i just checked my provider it says registered:nop |
| 17:49.36 | datarecal | no* |
| 17:49.47 | Trixboxer | and also an outbound route which can send calls to trunks |
| 17:49.48 | Trixboxer | yeah |
| 17:49.56 | datarecal | hmm wonder why it wouldnt be registered |
| 17:50.01 | Trixboxer | so concentrate on registered = yes 1st |
| 17:50.22 | Trixboxer | might be password or registration string problem |
| 17:51.38 | datarecal | hmm doesnt say much in the full log |
| 17:52.00 | [TK]D-Fender | You don't normally need to register to place calls |
| 17:52.12 | datarecal | it works sporatically |
| 17:52.19 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:53.41 | datarecal | triboxer any thoughts on where i can look to diagnose the problem |
| 17:54.25 | Trixboxer | datarecal: Are you using same trunk on two different PBX ? |
| 17:54.42 | datarecal | no one pbx |
| 17:55.24 | Trixboxer | Is it always registered = no or sometimes ? |
| 17:55.35 | datarecal | its usually always registered |
| 17:55.54 | Trixboxer | so, you have any outbound route ? |
| 17:56.13 | datarecal | i can make calls but 50% of the time it does that congestion |
| 17:56.27 | datarecal | same number |
| 17:56.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:57.15 | Trixboxer | so it must be provider error or your internet connection.. |
| 17:57.59 | datarecal | http://paste2.org/p/849037 |
| 17:58.46 | bmoraca_work | you've got a SIP problem. enable SIP debug and see what's going on. probably a NAT problem |
| 17:59.01 | bmoraca_work | and why the HECK is it so damn COLD out! |
| 18:01.04 | errr | bmoraca_work: clearly you are not in south texas :-) |
| 18:01.42 | bmoraca_work | nope, central California. it's 56 degrees out. not supposed to get above 80 for another 2 weeks. i can't remember in the last 15 years a time it was this cold this late in the year |
| 18:02.17 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-177-17.mia.bellsouth.net) |
| 18:03.34 | shader | does DB_EXIST assign to DB_RESULT if the entry exists? |
| 18:04.07 | *** join/#asterisk rocksfrow_work (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:07.13 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 18:08.13 | datarecal | what is the file where you can specify your outbound ip |
| 18:08.52 | rocksfrow_work | question... |
| 18:09.04 | rocksfrow_work | does asterisk rotate queue_log automatically at all? |
| 18:09.49 | bmoraca_work | shader: according to "core show function DB_EXISTS", yes |
| 18:10.39 | bmoraca_work | datarecal: in what context? |
| 18:11.00 | *** part/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 18:11.04 | [TK]D-Fender | rocksfrow_work: no |
| 18:11.12 | rocksfrow_work | [TK]D-Fender: awesome. thanks. |
| 18:11.33 | [TK]D-Fender | datarecal: sip.conf for sip, iax.conf for iax, etc |
| 18:12.17 | datarecal | and put externip=xxx.xxx.xxx.xxx |
| 18:14.24 | bmoraca_work | datarecal: localnet and externip are required for proper NAT traversal if your server is behind a NAT |
| 18:15.31 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-ttatwgrqmfeivxgx) |
| 18:16.57 | datarecal | ok i see now it says registration timed out, trying again |
| 18:17.01 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 18:19.27 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 18:25.08 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 18:35.03 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 18:35.33 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 18:39.23 | *** part/#asterisk fishloa (fishloa@87-194-32-209.bethere.co.uk) |
| 18:39.59 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 18:45.41 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 18:50.38 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-rbmnlxikcoonpboj) |
| 18:51.09 | shader | when features.conf says that the default for call transfers is #, does that mean that it is automatically enabled and bound to # unless you change it? |
| 18:55.09 | p3nguin | shader: It means that DTMF transfers are initiated by the # key if you are using t or T in your Dial() command. |
| 18:57.27 | shader | ok |
| 18:57.44 | *** part/#asterisk rocksfrow_work (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:58.21 | shader | what context does it use for the transfer? |
| 18:58.38 | p3nguin | the one where the Dial() happens. |
| 18:58.42 | *** join/#asterisk coreyf1513 (~coreyf151@24-177-5-103.dhcp.nwtn.ct.charter.com) |
| 18:59.08 | p3nguin | Oh, you mean what context you can call... the one for your phone. |
| 18:59.47 | p3nguin | You can't use a context that you don't have access to. |
| 19:01.58 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 19:04.19 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 19:05.36 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 19:05.39 | shader | can a phone have multiple contexts? |
| 19:05.47 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 19:06.20 | p3nguin | No. |
| 19:06.29 | p3nguin | But the context that it does have can include other contexts. |
| 19:06.35 | p3nguin | So effectively, yes. |
| 19:06.40 | Kobaz | shader: what would you expect it to do? |
| 19:06.43 | p3nguin | literally, no. |
| 19:07.00 | coreyf1513 | I have a system with a DAHDI PRI interface (T4XXP), calls seem to work find but I'm having issues receiving proper q931 hangup codes for outbound calls with errors, so i'm only receiving 3 codes - no answer, busy or drop (for calls that were answered). It seems that I receive all the error codes as progress codes, and the call appears to have dial timeout/no answer. I want Asterisk to immediately hang-up and return the error |
| 19:08.25 | *** join/#asterisk [8 (~recipe@207-9-95-178.pool.ukrtel.net) |
| 19:09.31 | shader | what's the asterisk cli command for converting audio? |
| 19:10.51 | shader | hmm |
| 19:12.17 | pabelanger | shader: sox |
| 19:12.26 | Qwell | file convert? |
| 19:12.42 | pabelanger | shader: tho sox is an external application |
| 19:12.57 | shader | I think I got it, though the result wasn't what I was expecting |
| 19:13.42 | shader | I used to have a problem where going from one menu to another had a long delay before the second did anything, like play the audio message |
| 19:14.38 | shader | for some reason I think converting it from gsm to gsm fixed the delay, even though it took 0ms to convert the file |
| 19:15.29 | p3nguin | There was nothing to convert, so of course it was fast. |
| 19:15.55 | p3nguin | The delay you're experiencing is probably dialplan processing rather than audio file problems. |
| 19:16.54 | shader | ok |
| 19:17.10 | shader | any idea why the delay disappeared? |
| 19:17.14 | p3nguin | Don't use overlapping extensions and don't provide more possibilities than necessary, and I would imaging it will process much sooner. E.g., don't allow things like exten => 1,... as well as exten => _1XX,... |
| 19:17.22 | p3nguin | No idea. |
| 19:17.28 | coreyf1513 | i attempted to patch chan_dahdi.c in a way that i hoped would accomplish this, though I only managed to make Dial return the q931 codes I wanted and claim to have hung-up the dahdi channel, though the channel became permenantly unavailable (until asterisk was restarted).. |
| 19:18.00 | p3nguin | If 1 is the only thing available, it won't wait for you to dial more digits. |
| 19:18.21 | shader | p3nguin: oh, I think that it might have been overlapping extensions |
| 19:18.55 | p3nguin | Asterisk tries to make the best match, so overlaps will cause a delay. |
| 19:18.58 | shader | but I recently removed the second one |
| 19:19.08 | shader | 8 vs 800 |
| 19:19.22 | p3nguin | If you later removed the 800, the delay should have disappeared. |
| 19:19.47 | p3nguin | If you are including more contexts, that's also going to introduce delay while those other contexts are checked for overlaps. |
| 19:20.23 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 19:20.37 | shader | ok |
| 19:20.38 | p3nguin | The default digit timeout is 5 seconds. You can shorten it if you want. |
| 19:21.05 | p3nguin | If you had a 5-second delay, that was most certainly why. |
| 19:22.09 | shader | is there a way to specify make options via the command line instead of menuselect? |
| 19:22.39 | Corydon76-dig | shader: nope |
| 19:22.57 | shader | so there's no way to configure asterisk via a bash script? darn |
| 19:23.06 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:23.27 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 19:23.29 | p3nguin | You could probably generate your own Makefile. |
| 19:23.47 | shader | hmm |
| 19:23.57 | p3nguin | Sounds like a horrible undertaking, though. |
| 19:24.00 | shader | yeah |
| 19:24.11 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 19:24.29 | p3nguin | What's the reason you can't just use the regular menuselect method? |
| 19:25.02 | shader | I was hoping to make a bash script to automatically install and configure asterisk as a means of backing up the system |
| 19:25.28 | shader | so far it works pretty well, but I'm about to add a feature which I believe requires menuselect |
| 19:25.56 | p3nguin | Package the software after you've built it. |
| 19:26.13 | p3nguin | Then just backup the package file(s). |
| 19:26.16 | coreyf1513 | shader: you might take a look at the source packages from http://packages.asterisk.org |
| 19:26.34 | doolittlework | scp * root@196.211.34.66:/usr/src/mysql-backup/24-05-2010 |
| 19:26.48 | p3nguin | scp? Why wouldn't you use rsync? |
| 19:26.50 | doolittlework | lol sry wrong terminal |
| 19:27.27 | shader | impeccable timing doolittlework |
| 19:27.59 | Tim_Toady | shader generate ur own menuselect.makeopts file and use it for each build |
| 19:28.01 | ujjain | maye scp is slow :p |
| 19:28.03 | ujjain | faster |
| 19:28.06 | shader | ok |
| 19:28.24 | shader | Tim_Toady: so the menuselect.makeopts file is all I need? |
| 19:28.38 | Tim_Toady | i think so |
| 19:29.06 | Tim_Toady | i tried it some time ago, i think its the only file you need to copy |
| 19:29.10 | p3nguin | Wouldn't that be convenient? |
| 19:30.59 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 19:31.52 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 19:35.35 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 19:37.21 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:38.55 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 19:40.11 | Corydon76-dig | shader: I misspoke, you can use the menuselect utility in CLI form |
| 19:43.53 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:44.47 | shader | Corydon76-dig: oh? |
| 19:48.35 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 19:48.43 | Corydon76-dig | --enable <option> and --disable <option> |
| 19:48.47 | Corydon76-dig | One at a time, though |
| 19:49.49 | shader | ok |
| 19:49.50 | shader | thanks |
| 19:49.52 | Jumpie | hmm..dont most ip phones, be it 1 or 2 ethernet ports, have a 100mbit port? |
| 19:50.04 | Jumpie | gotta customer wanting 150 phones all with gigabit ports |
| 19:51.02 | Jumpie | seems excessive, unless the bridged one for computer is |
| 19:51.03 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 19:51.54 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 19:53.27 | shader | Corydon76-dig: what's the whole command? "# make menuselect --enable <option>" is giving me the make help information |
| 19:53.58 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 19:55.27 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:01.44 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:02.27 | *** join/#asterisk nomed (~nomed@ulteo/nomed) |
| 20:02.44 | nomed | hi all |
| 20:03.00 | nomed | is there anyone that is using asterisk with telecom italia ? |
| 20:04.52 | pabelanger | So, why does fullcontact only exist in Asterisk realtime? |
| 20:06.16 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 20:13.05 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 20:15.28 | Katty | ohai |
| 20:15.43 | Katty | my asterisk does not work at all |
| 20:15.44 | Katty | how to fix pls |
| 20:16.13 | bmoraca_work | Katty: grab a hammer, take it outside, and beat the asterisk box with it |
| 20:16.14 | emora | Katty: It would help if you can give some details. |
| 20:16.18 | Katty | aww |
| 20:16.21 | Katty | you thought i was serious |
| 20:16.23 | Katty | that's so cute |
| 20:16.27 | bmoraca_work | lol |
| 20:16.33 | Katty | hugs bmoraca_work |
| 20:16.37 | Katty | bmoraca_work: hello sunshine! |
| 20:17.06 | bmoraca_work | Katty: there's no sunshine in central California today...it's cold, cloudy, and windy. blech |
| 20:18.52 | beek | hugs Katty |
| 20:20.52 | *** join/#asterisk QaDeS (~mklaus@p4FC71616.dip0.t-ipconnect.de) |
| 20:23.20 | Katty | bmoraca_work: but you're there! that makes it instantly brighter! (= |
| 20:23.20 | raden_work | hey Katty :) |
| 20:23.25 | Katty | hugs beek |
| 20:23.27 | Katty | hi raden (= |
| 20:23.28 | raden_work | evening bmoraca_work |
| 20:23.41 | pabelanger | Katty: PC Load letter error? |
| 20:23.47 | bmoraca_work | Katty: rofl...tell that to my wife...she thinks i'm evil |
| 20:24.10 | Katty | bmoraca_work: well if you didn't have so much baggage >.< |
| 20:24.12 | Katty | bmoraca_work: <3 |
| 20:24.17 | Katty | pabelanger: bu...wha? |
| 20:24.20 | bmoraca_work | hah! |
| 20:24.47 | raden_work | LMAO |
| 20:25.03 | Katty | pabelanger: i did not get the memo |
| 20:25.05 | Katty | pabelanger: i mean meme |
| 20:25.13 | *** join/#asterisk ManxPower (~manxpower@106.sub-75-234-24.myvzw.com) |
| 20:26.05 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:26.24 | Katty | HAI MANX |
| 20:29.46 | Katty | so where is everyone this afternoon |
| 20:29.46 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:30.54 | raden_work | WISCONSIN :P |
| 20:31.30 | Katty | oooh |
| 20:31.32 | Katty | they have cheese |
| 20:31.34 | Katty | i'll bring the whine. |
| 20:31.53 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 20:32.02 | *** join/#asterisk brezular (~brezular@2002:4e62:ae74:1234:211:9ff:fe83:2173) |
| 20:32.25 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:33.27 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:38.47 | Jumpie | hmm it seems for an 'average' user low end phone, going gigabit ports isnt cost effective |
| 20:41.43 | Katty | you mean on the phone |
| 20:41.46 | Katty | or the switch |
| 20:42.19 | *** join/#asterisk DrDamnit (~michael@173-165-161-161-atlanta.hfc.comcastbusiness.net) |
| 20:42.53 | *** join/#asterisk cslamar (~cslamar@ampache/staff/cslamar) |
| 20:43.20 | DrDamnit | I am converting a zaptel installation to a dahdi installation. I cannot seem to get the TE 122 and TDM800 to show up. What am I missing? |
| 20:43.35 | cslamar | hello, has anyone using ImageMagick's convert to create tiffs that can be used for faxing in asterisk? |
| 20:45.37 | DrDamnit | If you're going to create TIFFs for use with fax, they have to be conform to some basic specs... give me a minute and I can dig them up. |
| 20:46.13 | cslamar | thanks |
| 20:46.22 | DrDamnit | I resolved this by outputing text and piping it to enscript to create a postscript file. Then I used a ghostscript command to convert the postscript into a Tiff. |
| 20:47.08 | cslamar | yea i've been able to get gs to convert a pdf to the correct tiff, but there is background noise in the image and it's unsuable |
| 20:47.54 | DrDamnit | Faxable images must have 204x98 or 204x196 DPI resolution and must have a 1728 pixel width. This is more based on the fax standards than anything else. |
| 20:47.54 | DrDamnit | Welcome to fax! The cutting edge technology of 1955. |
| 20:48.12 | cslamar | haha |
| 20:48.28 | DrDamnit | My fax machine has a flux capacitor. |
| 20:48.56 | cslamar | were you able to to convert pdfs to tiffs using gs? |
| 20:49.18 | DrDamnit | I actually didn't do this. I was coaching someone else on how to do it. These were the answers. |
| 20:49.32 | cslamar | gotcha |
| 20:49.52 | DrDamnit | tiff; however, should be a nice "intermediate" format that should be easy to get to. |
| 20:50.05 | DrDamnit | kind of like csv for between excel and mysql, etc... |
| 20:50.17 | DrDamnit | Anyone in here understand DAHDI? |
| 20:50.32 | cslamar | well thanks anyways |
| 20:50.36 | *** part/#asterisk cslamar (~cslamar@ampache/staff/cslamar) |
| 20:50.37 | DrDamnit | you're welcome. |
| 20:51.11 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 20:55.08 | *** join/#asterisk pgrace (~pgrace@2001:470:8a93:2:20c:29ff:fee9:9689) |
| 20:56.30 | pgrace | I'm trying to link asterisk and exchange unified messaging. I'm getting a 302 from UM (expected, I have turned on promiscuous redirect) but I've got a problem. It appears that asterisk is trying to directly forward my softphone to the Contact field, but my phone is udp while the connection to the UM box is tcp. I have set canreinvite=no in my softphone profile but this seems to persist. any ideas? |
| 20:57.37 | DrDamnit | I have installed DAHDI, and run genconf and dahdi_cfg -v. in the CLI, asterisk shows the cards are there and working (dahdi show status), but dahdi show channels only shows pseudo. What did I do wrong? |
| 20:59.56 | Jumpie | DrDamnit i had better luck and reliability just using an ata for faxes |
| 20:59.56 | Jumpie | hehe |
| 21:00.38 | DrDamnit | Jumpie, I agree. My DAHDI question is not related to my response for the faxing to cslamar |
| 21:00.51 | shader | anyone know of a cheap usb sip handset that works with mac? unlikely, I know |
| 21:01.52 | DrDamnit | shader, if you can do it, you can make any handset work with mac provided you can do some type of internet connection sharing. We do it with Windows and Linux laptops all the time. |
| 21:01.59 | DrDamnit | Not the answer to your question, but.... it works. |
| 21:02.37 | DrDamnit | or... just any cheap headset with a SIP softphone would work. |
| 21:03.18 | beek | DrDamnit: What does your /etc/dahdi/systemconf look like? What kind of card is this? |
| 21:03.57 | Jumpie | fun times with dahdi |
| 21:04.11 | DrDamnit | beek, lemme copy / paste for you... |
| 21:04.27 | Jumpie | use pastebin plz |
| 21:05.06 | DrDamnit | Jumpie, thanks for the reminder. |
| 21:05.08 | DrDamnit | beek, http://pastebin.com/eU5A5WSb |
| 21:06.56 | *** join/#asterisk scalex000 (~chatzilla@190.166.189.165) |
| 21:07.23 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 21:08.03 | beek | DrDamnit: And you're saying that 'dahdi show channels' is blank? |
| 21:08.27 | Deeewayne | waves to Katty |
| 21:09.15 | DrDamnit | beek: yes. it only shows the psudo channel |
| 21:09.32 | DrDamnit | beek: dahdi show status shows the cards though. and the dahdi_tools show the cards with no alarms. |
| 21:11.37 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 21:11.39 | beek | Wait a minute... why is your card defined like it's in Europe. A PRI in the US should only have 23 B channels and one D channel. |
| 21:12.25 | DrDamnit | yes |
| 21:12.30 | DrDamnit | in US. Let me check.... |
| 21:14.49 | beek | I'd say that you need to rerun the configurator. |
| 21:14.50 | DrDamnit | I already corrected that once, because I need the TDM card as the first 8 channels, you can see that here: http://pastebin.com/1hXxsrjM |
| 21:14.55 | DrDamnit | it doesn't work either. |
| 21:16.49 | beek | DrDamnit: Pastebin your /etc/asterisk/chan_dahdi.conf file |
| 21:17.13 | *** join/#asterisk jhirley (~jhirley@adsl-159-225-76.mia.bellsouth.net) |
| 21:17.49 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 21:18.12 | DrDamnit | Just one line for that file: /etc/asterisk/chan_dahdi.conf |
| 21:18.17 | DrDamnit | #include dahdi-channels.conf |
| 21:19.33 | beek | DrDamnit: Do you have a dahdi-channels.conf file? |
| 21:19.59 | DrDamnit | yes. I am pastebinning it now. |
| 21:20.16 | DrDamnit | http://pastebin.com/WAP5Dxnp |
| 21:20.53 | beek | It's FUBARed too... |
| 21:21.17 | beek | Rerun your configurator and ensure that everything is configured for PRI 23B, 1D |
| 21:21.48 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:22.02 | DrDamnit | ok |
| 21:24.26 | scalex000 | hi guys, I need help with something trivial on queue |
| 21:24.35 | beek | ~ask |
| 21:24.36 | 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. |
| 21:24.40 | scalex000 | ok |
| 21:25.34 | scalex000 | I setup monitor-format and monitor-type on queues.conf, I need to use mixmonitor function on extensions.conf too to record conversation |
| 21:27.17 | *** join/#asterisk miamiseb (~deigo@208.76.35.132) |
| 21:27.24 | DrDamnit | beek: I am working on it. cleaning some stuff up first. Thanks for your patience. |
| 21:27.42 | beek | ... which will run out in about ten minutes when I have to leave. |
| 21:27.56 | DrDamnit | ok. going as fast as i can. |
| 21:28.39 | DrDamnit | beek: I had a moment of weaknes and tried to get zaptel working. That failed (obviously), and now I am reinstalling DAHDI. |
| 21:29.09 | beek | DrDamnit: The transition from Zaptel to DAHDI is basically 1-1 |
| 21:29.31 | DrDamnit | beek: Yes. For some reason, I am having a moment of retardedness. |
| 21:29.49 | beek | been there, done that. |
| 21:30.39 | DrDamnit | beek: we should start a club. Re-comiled DAHDI, and it sees the devices. step 1/3 complete. Reconfiguring asterisk for a recompile. |
| 21:30.52 | beek | scalex000: use the monitor() application. |
| 21:31.07 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 21:31.07 | scalex000 | ok |
| 21:31.11 | scalex000 | beek, thx |
| 21:31.17 | DrDamnit | beek: chan_dahid is present and selected. running make. |
| 21:31.30 | ManxPower | Why don't you do a "core show applications" and see all the cool apps asterisk has for you to use |
| 21:31.30 | beek | scalex000: Wait a minute... no. |
| 21:31.42 | miamiseb | bah, fop2 is only triggering custom_popup on the first button. |
| 21:31.59 | ManxPower | miamiseb, We don't really care. Perhaps they will care on a channel that supports FOP? |
| 21:32.08 | beek | scalex000: You need to set the MONITOR_FILENAME before queueing the call. |
| 21:32.12 | DrDamnit | beek: what files should I remove so they can be re-generated by dahdi tools? |
| 21:32.33 | scalex000 | beek, I got it |
| 21:32.37 | miamiseb | I wasn't looking for support, just relying a problem I'm encoutering, much like someone might mention a cat stomping on their keyboard. |
| 21:32.43 | miamiseb | relaying* |
| 21:32.51 | miamiseb | and encountering* |
| 21:33.00 | miamiseb | I'm going to sign up for the moment of retardness club. |
| 21:33.04 | beek | DrDamnit: /etc/system.conf and /etc/asterisk/chan_dahdi.conf should do it. |
| 21:33.07 | ManxPower | miamiseb, perhaps I'm a little trigger happy because of all the idiots looking for FreePBX support here. |
| 21:33.33 | DrDamnit | beek: because you may leave before I am ready... what are the files I should setup, and in what order to get this to work? just those two, right? |
| 21:33.40 | miamiseb | cool, no problem by me. |
| 21:33.49 | DrDamnit | FYI, I am on asterisk 1.4, most current trunk svn. |
| 21:33.52 | beek | DrDamnit: yep. |
| 21:34.16 | ManxPower | DrDamnit, It is impossible for both parts of that statement to be true |
| 21:34.31 | DrDamnit | sorry |
| 21:34.34 | DrDamnit | I meant branch. |
| 21:34.37 | ManxPower | You are either on asterisk 1.4.?? or you are on trunk SVN. You can't be both |
| 21:34.44 | beek | /etc/dahdi/system.conf is for the DAHDI system. /etc/asterisk/chan_dahdi.conf is for asterisk's use thereof. |
| 21:35.03 | DrDamnit | equivalent to zaptel.conf and zapata.conf, respectively. |
| 21:35.27 | ManxPower | DrDamnit, you're one of those that sometimes puts salt in your coffee because you are not paying attention, aren't you? |
| 21:35.39 | DrDamnit | beek: is the syntax the same? can I copy / paste from zaptel.conf -> system.conf, and then zapata.conf -> chan_dahdi.conf? |
| 21:36.07 | beek | Ummm... that sounds like a recipe for disaster. |
| 21:36.15 | beek | Why not use the tools to do it right? |
| 21:36.17 | DrDamnit | ManxPower: probably. I have been upgrading this machine for 9 hours (lots of custom crap). |
| 21:36.19 | DrDamnit | beek: ok. |
| 21:36.28 | DrDamnit | I'll use the tools then change the dialplan. |
| 21:36.38 | beek | DrDamnit: I was just getting ready to say that! |
| 21:36.52 | DrDamnit | run genonf, then dahdi_cfg -v, and that should be it, right? |
| 21:37.00 | DrDamnit | running make install |
| 21:37.43 | DrDamnit | starting dahdi |
| 21:38.28 | DrDamnit | running gen conf... |
| 21:38.30 | DrDamnit | restarting dahdi |
| 21:39.10 | DrDamnit | both cards up, configured. no alarms. |
| 21:39.42 | DrDamnit | chan_dahdi.conf is blank. |
| 21:39.54 | DrDamnit | beek: chan_dahdi.conf is blank. |
| 21:40.12 | DrDamnit | beek: dahdi_cfg -v configured everything properly, but chan_dahdi.conf is blank. |
| 21:40.45 | beek | Do you have the dahdi-tools compiled? |
| 21:40.57 | DrDamnit | yes |
| 21:41.22 | DrDamnit | did I skip a command? |
| 21:41.32 | *** join/#asterisk jetlag (jetlag@pool-173-61-216-196.cmdnnj.east.verizon.net) |
| 21:41.58 | ChannelZ | chan_dahdi.conf is the default config -- I think dahdi_cfg writes a different name, dahdi_channels.conf or something like that |
| 21:42.10 | DrDamnit | so I should just include that file then.... |
| 21:42.14 | ChannelZ | you have to specifically choose to use it, typically by #including it |
| 21:42.21 | scalex000 | hey this is important warning, WARNING[27265] chan_sip.c: Asked to transmit frame type 256, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8) |
| 21:43.01 | beek | DrDamnit: Doesn't the dahdi_cfg -v output tell you what file it writes (it's been over a year since I've done it) |
| 21:43.09 | ChannelZ | I meant dahdi_genconf above not dahdi_cfg |
| 21:43.29 | beek | Add some more 'v's for additional verbosity |
| 21:44.11 | beek | But you said that system.conf is there. |
| 21:44.36 | DrDamnit | pastebinning them for you... |
| 21:44.39 | ChannelZ | dahdi_cfg configures the hardware based on /etc/dahdi/system.conf - dahdi_genconf writes a channels file for Asterisk |
| 21:44.56 | DrDamnit | dahdi-channels.conf: http://pastebin.com/spam.php?i=8S7DS6vu |
| 21:45.47 | DrDamnit | system.conf: http://pastebin.com/eCCajBGg |
| 21:46.19 | DrDamnit | so run dahdi_cfg first, then dahdi_genconf second? |
| 21:46.43 | ChannelZ | no |
| 21:47.07 | ChannelZ | dahdi_cfg gets run for you (or should) when you start the drivers. It configures the hardware. |
| 21:47.18 | *** join/#asterisk Yon (Yon@2002:50d9:f410::50d9:f410) |
| 21:47.20 | DrDamnit | *CLI> [May 24 17:46:59] WARNING[4941]: config.c:1115 process_text_line: parse error: No category context for line 12 of /etc/asterisk/dahdi-channels.conf |
| 21:47.25 | ChannelZ | dahdi_genconf is meant to be run once, to generate your channels file for asterisk, which you can use or tweak as necessary. |
| 21:47.47 | beek | DrDamnit: That card is erroneously being configured as an E1 card, not a T1 card. |
| 21:47.51 | ChannelZ | read the top of the file |
| 21:47.53 | DrDamnit | ChannelZ: so I was doing it correctly, genconf first, dahdi_cfg second. |
| 21:48.03 | ChannelZ | "This is not intended to be a complete chan_dahdi.conf." |
| 21:48.27 | DrDamnit | beek: please elaborate...wrong channel count? |
| 21:48.33 | beek | DrDamnit: It's not hard to configure chan_dahdi.conf manually and that's what I did. |
| 21:48.57 | beek | DrDamnit: hang on... |
| 21:49.00 | beek | I'm wrong. |
| 21:49.04 | beek | It's correct. |
| 21:49.17 | beek | 1-8 is your TDM card. 9- is your T1. You're okay. |
| 21:49.49 | DrDamnit | beek: you are correct. 1-8 is a TDM, and 9+ is a T1. |
| 21:49.58 | DrDamnit | 5ess, national, pri_cpe, all correct. |
| 21:50.17 | DrDamnit | sorry.... should be 5ess on switchtype. |
| 21:50.34 | miamiseb | I think what ChannelZ was trying to convey to you is that you need a fuller chan-dahdi.conf, one which would include a context, and likely #include dahdi-channels |
| 21:51.01 | DrDamnit | beek: what's this? *CLI> [May 24 17:50:34] WARNING[4941]: config.c:1115 process_text_line: parse error: No category context for line 12 of /etc/asterisk/dahdi-channels.conf |
| 21:51.11 | ChannelZ | Read what I and miamiseb just said |
| 21:51.36 | ChannelZ | dahdi_genconf only creates a fragment of what chan_dahdi.conf should be |
| 21:51.37 | beek | I hate to bail but I'm now running late for an appointment. GL |
| 21:51.38 | DrDamnit | ChannelZ: Yes. I included the dahdi-channels.conf. Is there something I am missing there? |
| 21:51.46 | DrDamnit | beek: thanks for your help. |
| 21:51.52 | ChannelZ | Yea you're obviously missing things in your chan_dahdi.conf |
| 21:51.57 | ChannelZ | pastebin it |
| 21:52.05 | miamiseb | DrDamnit, http://www.voip-info.org/wiki/view/chan_dahdi.conf would be a sample chan_dahdi, than you can modify |
| 21:52.35 | ChannelZ | or look at the one that came with Asterisk... |
| 21:52.51 | ChannelZ | at minumum you need a [channels] block |
| 21:54.36 | ManxPower | I always look at the chan_dahdi.conf.sample in the configs/ directory of the Asterisk source. Much easier than relying on a frequently out of date and sometimes just plain wrong page on some wijki |
| 21:55.16 | miamiseb | that seems like good advice. I've been led in the wrong direction more than once by voip-info, but it's also been invaluable in getting some stuff working |
| 21:55.46 | miamiseb | of the top of my head, cisco 7960s and 70s in sip mode. They've got good tftp config files and such |
| 21:56.58 | *** join/#asterisk alerios (~alerios@190.144.75.22) |
| 21:57.09 | ManxPower | I am mainly referring to Asterisk/Digium/DAHDI/Zaptel/etc. |
| 21:57.17 | miamiseb | nods. |
| 21:57.21 | ManxPower | I have found VoIP Info to be useful for non-Asterisk stuff. |
| 21:57.34 | miamiseb | Thats why I said it was good advice, if you've got up to date samples in the source tree, might as well use those. |
| 21:57.41 | DrDamnit | ChannelZ: I am re-doing my chan_dahdi.conf. http://pastebin.com/T9dS698N |
| 21:57.49 | ManxPower | miamiseb, but nobody does that. they just ignore the docs in the souce |
| 21:58.12 | DrDamnit | ChannelZ: I'll post the original zaptel.conf |
| 21:58.17 | ManxPower | It's a good thing to or all of us would be totally bored because do few people would need help. Hey, I could get a life if that happened1 |
| 21:58.57 | DrDamnit | ChannelZ: This is the original zaptel.conf I am trying to convert: http://pastebin.com/uBppmjvE |
| 21:59.05 | miamiseb | lmao @ ManxPower |
| 21:59.29 | DrDamnit | To all of you that are helping me through me fit of retardedness, thanks. |
| 22:00.08 | ChannelZ | DrDamnit: That again is a fragment of a config file |
| 22:00.23 | DrDamnit | in zaptel.conf? |
| 22:00.27 | DrDamnit | or the other? |
| 22:00.56 | ChannelZ | either of them. |
| 22:01.21 | ChannelZ | The thing you just pastebin'd above looks like the default chan_dahdi.conf but doesn't include the channels file you generated with dahdi_genconf, so I dont know what you are doing |
| 22:01.39 | DrDamnit | ok... I'll fix that. |
| 22:02.04 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:02.07 | DrDamnit | root@voip:/opt/astbackup-20100523~/usr/astbackup/etc# dahdi_genconf -vvvvvvvvvvvv |
| 22:02.07 | DrDamnit | Default parameters from /etc/dahdi/genconf_parameters |
| 22:02.07 | DrDamnit | Generating /etc/dahdi/system.conf |
| 22:02.07 | DrDamnit | Generating /etc/asterisk/dahdi-channels.conf |
| 22:02.18 | DrDamnit | it generates dahdi-channels.conf |
| 22:02.20 | ChannelZ | you don't need to keep using dahdi_genconf over and over |
| 22:03.00 | ChannelZ | I'll say once more: dahdi_cfg gets run once when the DAHDI drivers start in order to configure the hardware channels, per /etc/dahdi/system.conf |
| 22:03.21 | ChannelZ | Then once you start Asterisk, assuming it's compiled right for DAHDI, it tries to load /etc/asterisk/chan_dahdi.conf |
| 22:04.32 | DrDamnit | ChannelZ: you're right. (*light goes on*). so since I just ran genconf, I need to re-run dahdi_cfg, then restart asterisk? |
| 22:04.33 | ChannelZ | You can either copy/paste the channels generated by dahdi_genconf in /etc/asterisk/dahdi-channels.con INTO /etc/asterisk/chan_dahdi.conf, or #include /etc/asterisk/dahdi-channels.conf at the end of /etc/asterisk/chan_dahdi.conf |
| 22:04.49 | ChannelZ | sighs |
| 22:04.58 | DrDamnit | Sorry for the frustration. |
| 22:05.09 | DrDamnit | I am not a noob, I swear. I just have a mental block on this today |
| 22:05.46 | ChannelZ | if you are using the DAHDI init scripts ('/etc/init.d/dahdi start', etc) it runs dahdi_cfg for you. You should almost never need to run it yourself. |
| 22:06.00 | DrDamnit | ok. So, restart Dahdi then? |
| 22:06.49 | ChannelZ | you probably don't even need to do that unless you've been changing slots and such all this time.. you keep running dahdi_genconfig but if your hardware config hasn't changed it's just re-writing the same thing over and over |
| 22:06.54 | miamiseb | you only need to reload the module if you want to re-read the config file |
| 22:07.16 | DrDamnit | miamiseb: I'll do that now. From the CLI: dahdi restart? |
| 22:07.58 | miamiseb | module reload chan_dahdi.so |
| 22:08.38 | ChannelZ | well I don't think he's even gotten that far because his config is not right |
| 22:09.09 | DrDamnit | If I could, I'd buy you all a beer. 31 channels are working. |
| 22:09.49 | DrDamnit | you people are awesome. i really appreciate this. |
| 22:10.19 | miamiseb | No problem, hope it all works well for you. |
| 22:10.44 | miamiseb | I'd like to take this moment to curse packed javascript. Thanks. |
| 22:10.58 | DrDamnit | I completely screwed this up by copy /pasting old zaptel configs in here, and it took me several tries to get it out. Thanks for all your patience everyone. |
| 22:13.59 | miamiseb | I've always lamented losing something from my clipboard that was irreplacable, after this latest loss of a variable name, I'm installing a clipboard management tool |
| 22:14.50 | *** join/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
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| 22:18.51 | bmoraca_work | distinctive ring on cisco 7940s sucks |
| 22:19.02 | raden_work | lol |
| 22:19.22 | bmoraca_work | it works...but just barely |
| 22:19.45 | bmoraca_work | turns "ring...pause" into "ring, ring...pause" |
| 22:19.56 | p3nguin | It's much better if you use SCCP. |
| 22:20.12 | Nugget | for sufficiently small vallues of "much" |
| 22:20.39 | p3nguin | Then you get to choose from five different ring types. |
| 22:21.01 | bmoraca_work | if i am going to use SCCP, i'll use callmanager |
| 22:21.10 | p3nguin | Whatever. |
| 22:21.59 | p3nguin | chan_sccp is quite satisfactory. |
| 22:22.11 | miamiseb | I use both SIP cisco's connected to asterisk and sccp connected to call manager, and I much prefer the call manager setup than messing about with tftp xml config files. |
| 22:23.00 | p3nguin | It's not any more work to configure the phones for SCCP on Asterisk than it is to configure them for SIP. |
| 22:23.15 | p3nguin | And it's not like you change them often, anyway. |
| 22:23.18 | miamiseb | Right, I just don't like either in comparision to the ease of getting them up natively |
| 22:23.51 | miamiseb | but i've spent hours with a debug cable connected to the phone trying to figure out which xml element it wasn't liking, or having problems because of extra white spaces. PITA. |
| 22:24.20 | Jumpie | p3nguin, any recommended phones for egotistical 'upper management' that insist on 'high level' phones? |
| 22:24.21 | Jumpie | lol |
| 22:24.57 | p3nguin | Something that supports half a dozen side cars, I would guess. |
| 22:25.11 | Jumpie | lol |
| 22:25.16 | Jumpie | i like the aastra 6739i |
| 22:25.22 | Jumpie | full color video screen |
| 22:25.39 | miamiseb | I like the snom phones, they look fancy, although I'm certainly not p3nguin. |
| 22:26.21 | Jumpie | i actually havent set up a snom phones but i hear good things |
| 22:26.40 | Jumpie | wow snom meeting point conf phone is like 800$ |
| 22:26.41 | WIMPy | likes them as well. |
| 22:27.20 | Jumpie | i want to be able to do xfer, conf, some speed dials, etc on hard buttons and then map othre functions to others |
| 22:28.59 | *** part/#asterisk alerios (~alerios@190.144.75.22) |
| 22:29.55 | miamiseb | sweet, got both my buttons to popup in fop2. Too bad I lost url string that had the right variable name and had to go hacking through JS to find the problem, but at least that hard part is done. |
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| 22:45.08 | bmoraca_work | ~book |
| 22:45.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 23:34.43 | Janos | hey there, got a small question about queues, queue show my queue, shows me the members of the queue and the status each member is in, where can i find what each status means and how this status is decided ? |
| 23:36.17 | Corydon76-dig | The status is designed to be English readable and is decided based upon the devicestate of the associated channel |
| 23:38.03 | Corydon76-dig | So, main/devicestate.c, plus channels/chan_sip.c for SIP devices |
| 23:38.11 | Janos | the thing is, that sometimes a queue has all members in Invalid state, when they are like that a call to Queue(myqueue,tTn,60) would return right away (which is the behavior i would expect) but sometimes the members are un an Unavailable state in which case even though all the members are Unavailable the Queue application will still wait the 60 seconds before terminating, so, what's the diff between Invalid and Unavailable ? |
| 23:38.24 | sawgood | exten => 123,n,Answer() |
| 23:38.32 | sawgood | what does the () mean in this statement? |
| 23:39.02 | Corydon76-dig | Invalid means that the device does not correspond to an actual device |
| 23:39.15 | sawgood | exten => s,1,Answer |
| 23:39.24 | sawgood | vs this with no () after answer |
| 23:39.26 | Corydon76-dig | Unavailable means that the device is not available to be contacted at this moment in time |
| 23:39.39 | Corydon76-dig | sawgood: no difference |
| 23:39.55 | sawgood | should they be used for good measure? |
| 23:40.12 | Corydon76-dig | sawgood: makes utterly no difference |
| 23:40.19 | sawgood | thank you |
| 23:40.29 | Janos | ok, so is there any way to tell the Queue application that if everyone is Unavailable to terminate right away ? |
| 23:40.50 | sawgood | 123,n,Ringing() |
| 23:40.56 | Corydon76-dig | Janos: you cannot, because a device could become available within that time period |
| 23:40.58 | sawgood | what is the purpose of Ringing? |
| 23:41.20 | Corydon76-dig | sawgood: It's an indication sent to the remote device |
| 23:41.33 | Janos | Corydon76-dig, ok, makes sense |
| 23:41.37 | sawgood | A SIP request? |
| 23:41.58 | Corydon76-dig | sawgood: not a request, usually a 183 |
| 23:42.06 | ManxPower | sawgood, there is not much use for Ringing |
| 23:42.11 | sawgood | got it! |
| 23:42.26 | ManxPower | Asterisk will provide ringing sounds when needed all automatically with no need for you to do anything |
| 23:42.32 | sawgood | exten => s,2,Wait,2 |
| 23:42.41 | sawgood | is there a difference between the above and this |
| 23:42.53 | ManxPower | sawgood, If you want us to hold your hand you might want to provide some dinner and drinks first |
| 23:42.53 | Corydon76-dig | ManxPower: that's not correct. In some cases, people want an explicit 183 to be sent |
| 23:42.59 | sawgood | exten => s,2,Wait(2) |
| 23:43.12 | Corydon76-dig | 180 could be sent, even without a 183 |
| 23:43.29 | ManxPower | Corydon76-dig, and amazingly my users all hear ringing without using the Ringing app |
| 23:43.43 | Corydon76-dig | ManxPower: It's strongly usage-based |
| 23:44.01 | Corydon76-dig | some uses don't need it; others do |
| 23:44.13 | sawgood | I am learning the dialpan process ... fully breaking away from FreePBX in most cases |
| 23:44.22 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 23:44.27 | ManxPower | have you done a "core show applications"? |
| 23:44.39 | Corydon76-dig | Asterisk is a toolkit, not a one-size-fits-all key system |
| 23:45.34 | Janos | Corydon76-dig, i have two separate queues i would like to call, one after another, but would like to skip the first queue if nobody is available, what would be the best approach here ?, create a single queue with everyone on it maybe ? |
| 23:45.42 | boodu | hi |
| 23:45.50 | Corydon76-dig | and some of the previously implicit behaviors need to be explicit to deal with certain situations which could be a security problem |
| 23:46.16 | Corydon76-dig | Janos: queue penalties and weights, most likely |
| 23:47.00 | ChannelZ | Anyone have an issue with the "ksoftirqd" in linux constantly consuming a chunk of CPU time when using dahdi_dummy ? I don't remember my old system doing this |
| 23:48.31 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:48.56 | sawgood | Inside Asterisk, is the portion dealing with conferences called, "meetme", or is this something different? |
| 23:49.35 | Janos | Corydon76-dig, kk thanks, going to read on that |
| 23:49.39 | *** join/#asterisk KNERD (~KNERD@129.113.130.51) |
| 23:49.55 | devmod | anyone has an idea on how to generate a .263 file for asterisk with gstreamer? |
| 23:50.19 | WIMPy | sawgood: meetme or confbridge |
| 23:51.46 | miamiseb | ChannelZ: nope, but you can use many timing sources for dahdi_dummy, are you using the high precision timer or what? |
| 23:51.56 | sawgood | Say I wanted to 'add' or make the meetme module do something 'additional' or diffeerent then normal (for example) if you are the leader of the conference you could enter a feature key to mute all callers |
| 23:52.15 | sawgood | what would it take to make the module for meetme do additional/different things? |
| 23:52.16 | p3nguin | I run Ringing() immediately before the Gosub() that does a CNAM lookup, that way there is a ringing sound rather than dead air for that 1 second that the script is looking up the caller ID info. |
| 23:53.08 | sawgood | p3nguin: thanks for your help ... I spoke with Aastra, and I have DNID working |
| 23:53.10 | sawgood | thanks! |
| 23:54.28 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:54.34 | ManxPower | p3nguin, I'd call that an "edge case" when talking to a n00b |
| 23:54.49 | p3nguin | Just saying a found a use for it. |
| 23:55.00 | p3nguin | s/a/I/ |
| 23:55.03 | ManxPower | p3nguin, I've found a few uses for it, but they are always edge cases |
| 23:55.12 | p3nguin | ah, screw it. |
| 23:55.22 | ManxPower | Ringing is almost as overused as the "r" option to Dial |
| 23:55.56 | p3nguin | I would have guessed that r was more abused, what with FreePBX using r in most of its Dial()s. |
| 23:57.10 | miamiseb | nucking futs |
| 23:57.15 | miamiseb | s/nu/fu |
| 23:57.24 | miamiseb | s/fu/nu |
| 23:57.31 | p3nguin | sed fail |
| 23:57.40 | ManxPower | p3nguin, I was not counting that scourge. |
| 23:57.41 | Janos | is there any way to decrease the verbosity level ? looks like whenever i connect with asterisk -vvvr the verbosity level gets set to 3 and it persist even when i reconnect with just -r |
| 23:57.50 | miamiseb | core set verbose |
| 23:57.55 | Janos | thanks |
| 23:57.55 | p3nguin | s/fail/failure/ |
| 23:58.04 | ChannelZ | miamiseb: Not sure... is it a config option? |
| 23:58.09 | miamiseb | can you chain them? |
| 23:58.20 | miamiseb | ChannelZ: no, part of the compliation of the kernel module |
| 23:58.28 | p3nguin | janos: The -vvv you're using is verbose 3. |
| 23:58.36 | miamiseb | if you cat the proc dahdi it should say |
| 23:58.37 | p3nguin | Don't want verbose level 3, don't use -vvv. |
| 23:58.56 | miamiseb | p3nguin: he just had a prolem with it persisting it seems, not necc. being at 3, just always being at 3. |
| 23:59.03 | p3nguin | Also, asterisk.conf sets the verbose level. |
| 23:59.40 | miamiseb | Am I the only person that find a bgplay beautiful. I could watch those network changes for hours. |
| 23:59.44 | miamiseb | also, I'm easily amused. |