| 00:02.45 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 00:03.20 | Corydon76-dig | buttons840: a T2? |
| 00:03.45 | buttons840 | well, i don't know really, i thought that's what it would be, i have 96 lines |
| 00:03.51 | buttons840 | could be a few t1's |
| 00:03.58 | Corydon76-dig | buttons840: more like 4 T1s |
| 00:04.05 | buttons840 | yes |
| 00:04.24 | Corydon76-dig | It's up to you |
| 00:04.34 | buttons840 | yep |
| 00:04.40 | Corydon76-dig | They both accomplish the same task with the same APIs |
| 00:04.56 | tompaw | if (ast_cdr_disposition(chan->cdr,chan->hangupcause)) ast_cdr_failed(chan->cdr); << maybe here? |
| 00:05.22 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 00:06.15 | buttons840 | with the spool files, must i use a specific extension, or is there a way to say, in essence "just make this call on any available line" |
| 00:06.15 | *** join/#asterisk xzcvczx (n=simon@gentoo/user/xzcvczx) |
| 00:06.56 | xzcvczx | has anyone got the asterisk book in pdf? astbook.asterisk.org seems to be giving a bad pdf file |
| 00:07.26 | buttons840 | i have it |
| 00:07.42 | Corydon76-dig | tompaw: what version? |
| 00:07.43 | buttons840 | http://astbook.asteriskdocs.org/ ? |
| 00:07.48 | buttons840 | that doesn't work? |
| 00:07.52 | tompaw | Corydon76-dig: 1.6.1.1 |
| 00:08.04 | tompaw | sorry, 1.6.1.0 |
| 00:08.09 | xzcvczx | nope, i click on the pdf file link and adobe reader just keeps saying bad dilw |
| 00:08.14 | xzcvczx | s/dilw/file/ |
| 00:08.48 | Corydon76-dig | tompaw: how are you generating this call? |
| 00:09.16 | buttons840 | i will try download |
| 00:09.37 | tompaw | Corydon76-dig: I read some data from Mysql, close the Mysql link and simply Dial(..., timeout, M(foo)) |
| 00:09.59 | tompaw | foo doesn't do anything special, it just uses the data read from Mysql on the channel |
| 00:10.16 | xzcvczx | ah, i didnt realise it was so big, i will try downloading it as well, it may have just been corrupting due to my incredibly slow connection to that server |
| 00:10.18 | tompaw | and the whole thing is about marking this channel free when the call ends, for whatever reason. |
| 00:10.26 | Corydon76-dig | tompaw: no, I meant, how is the originating channel occurring? |
| 00:10.37 | buttons840 | xzcvczx, i downloaded it again and it works |
| 00:10.46 | Corydon76-dig | tompaw: Incoming SIP? Call file? How? |
| 00:10.52 | tompaw | Corydon76-dig: incoming sip peer call. |
| 00:10.57 | xzcvczx | i am only getting 9kb/s so it will take me a while to check it |
| 00:11.17 | buttons840 | your connection slow, or just that server? |
| 00:11.24 | xzcvczx | buttons840: just to that server |
| 00:11.39 | xzcvczx | i can get 1.5MB/s from some servers |
| 00:12.23 | xzcvczx | buttons840: you wouldn't be able to email it to me by any chance could you |
| 00:12.34 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 00:12.52 | Corydon76-dig | tompaw: Are you getting anything on the console at verbose level 2 that indicates that "h" ran, but exited early? |
| 00:13.23 | Corydon76-dig | tompaw: i.e. "Spawn extension (foo, h, 1) exited non-zero on '...'" |
| 00:13.55 | tompaw | Corydon76-dig: no, only s extensions are mentioned. h isn't even touched. |
| 00:14.28 | Corydon76-dig | tompaw: do you have ANY "Spawn extension... exited non-zero" messages? |
| 00:16.20 | tompaw | Corydon76-dig: yes, one for the context defined as incoming sip peer's context, and the other one for the macro that's being used to wrap this all up. |
| 00:16.51 | Corydon76-dig | tompaw: Is this in a Macro? |
| 00:17.21 | Corydon76-dig | tompaw: and your "h" extension is in that Macro? |
| 00:17.37 | tompaw | Corydon76-dig: yes and yes. |
| 00:18.32 | Corydon76-dig | tompaw: Please upgrade to SVN 1.6.1 |
| 00:18.52 | buttons840 | Corydon76-dig, can you create a spool file that will chose from a number of open lines, or do you have to specify a single specific line? |
| 00:19.20 | Corydon76-dig | tompaw: Already fixed in SVN, just not in a release yet |
| 00:19.40 | tompaw | Corydon76-dig: really? trying now. |
| 00:21.52 | tompaw | compiling... |
| 00:23.30 | tompaw | Corydon76-dig: will this work with 1.6.1.0 addons? |
| 00:24.46 | Corydon76-dig | tompaw: Yes, the APIs don't change in the middle of the 1.6.1.x release series |
| 00:25.21 | Corydon76-dig | Well, not without a REALLY good reason that we would state in the UPGRADE.txt file |
| 00:25.53 | Corydon76-dig | Does not appear to be one in that file |
| 00:28.24 | tompaw | Corydon76-dig: installed svn, but it didn't help. |
| 00:28.35 | tompaw | still nothing about "h" mentioned in the console. |
| 00:28.37 | Corydon76-dig | tompaw: really? That's odd |
| 00:28.38 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4df69acb5169c9e9) |
| 00:29.12 | Corydon76-dig | tompaw: if you move the "h" extension to the context from which the macro was called, does it execute then? |
| 00:29.42 | tompaw | let me try. |
| 00:30.17 | tompaw | yes. |
| 00:30.22 | tompaw | well, that might be my answer then |
| 00:30.24 | Corydon76-dig | Aha |
| 00:30.36 | tompaw | I will create an exclusive context for this sip peer |
| 00:30.40 | tompaw | and move it all one level up. |
| 00:31.01 | tompaw | Corydon76-dig: was I supposed to expect it to work as a Macro's "h"? |
| 00:31.11 | tompaw | or is it all my misunderstanding of how things work? |
| 00:31.51 | Corydon76-dig | tompaw: No, it's another bug |
| 00:32.13 | Corydon76-dig | tompaw: It probably used to work, but we fixed another bug |
| 00:32.45 | Corydon76-dig | Having "h" in a Macro was never supposed to work, but it did for awhile and got documented that way |
| 00:33.04 | tompaw | Corydon76-dig: so just for the record, this problem of ignoring "h" occurs every time a caller hangs up (don't know about calee) when a call is not in an established state. |
| 00:33.22 | tompaw | Corydon76-dig: it's not only the initial set up of the call, but for example SendDTMF(), too. |
| 00:33.39 | tompaw | Hanging up during SendDTMF results in the same problem - no "h". |
| 00:33.53 | tompaw | If the call is established, "h" gets parsed perfectly. |
| 00:34.42 | tompaw | Corydon76-dig: thank you very much for your time and help. |
| 00:35.02 | tompaw | I'm gonna get some sleep now, almost 3 AM here, and in the morning I will finish my code. |
| 00:35.19 | tompaw | Take care. |
| 00:35.35 | Corydon76-dig | tompaw: update to SVN and restore your dialplan |
| 00:35.44 | Corydon76-dig | I just committed the fix. |
| 00:35.46 | tompaw | Corydon76-dig: ok |
| 00:40.25 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-131-23.dsl.stlsmo.sbcglobal.net) |
| 00:41.48 | tompaw | Corydon76-dig: sorry, still no luck :( |
| 00:42.17 | tompaw | it only parses the main extension's h. |
| 00:42.56 | Corydon76-dig | Huh, okay |
| 00:43.50 | tompaw | I'll keep my original dialplan, so if you feel like fixing it tomorrow I'll be here to test it. |
| 00:45.09 | Corydon76-dig | tompaw: when you did a 'svn update', did it say that any files were updated? |
| 00:45.25 | tompaw | no, I removed the old dir... |
| 00:45.46 | Corydon76-dig | tompaw: what does it say when you do an 'svn info' as to the revision? |
| 00:46.17 | tompaw | Last Changed Author: tilghman |
| 00:46.17 | tompaw | Last Changed Rev: 201827 |
| 00:46.18 | tompaw | Last Changed Date: 2009-06-18 20:35:18 -0400 (Thu, 18 Jun 2009) |
| 00:46.38 | Corydon76-dig | tompaw: Try again on the svn update |
| 00:46.38 | Corydon76-dig | You need revision 201828 |
| 00:47.00 | tompaw | At revision 201831. |
| 00:47.26 | Corydon76-dig | Oh, nevermind |
| 00:47.44 | Corydon76-dig | tompaw: I'll look to see if I can find another culprit |
| 00:47.47 | tompaw | strange, svn info first said 201827 |
| 00:47.59 | tompaw | then svn update increased that to 201831 |
| 00:48.02 | tompaw | but no files were changes |
| 00:48.02 | tompaw | ok |
| 00:48.19 | tompaw | thanks and see you tomorrow. |
| 00:48.29 | tompaw | over&out. |
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| 02:07.03 | blaxthos | anyone have idea what could cause me to be able to dial internal and external numbers from an extension, but i can't dial to it from any other extension ? |
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| 02:07.55 | blaxthos | <PROTECTED> |
| 02:07.55 | blaxthos | <PROTECTED> |
| 02:07.56 | blaxthos | <PROTECTED> |
| 02:07.56 | blaxthos | <PROTECTED> |
| 02:08.01 | blaxthos | from 201 to 103 |
| 02:10.08 | *** join/#asterisk greekboy26 (n=sdsds@cust-162-199.on4.ontelecoms.gr) |
| 02:15.51 | superbeef | blaxthos: hmm all calls from the same asterisk pbx? |
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| 02:21.31 | superbeef | blaxthos: i sometimes get that, where the pbx thinks the extension is busy |
| 02:21.51 | superbeef | blaxthos: the lazy move is restart asterisk.. |
| 02:22.06 | superbeef | blaxthos: you might be able to unregister hte extension, shut it down and reconnect |
| 02:22.33 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:23.01 | jplank | how can I confirm if I'm using my HW echo can, and not software? It almost sounds like echo training is going on at the beginning of the call, and then the echo never clears, but I thought * wouldn't use echo training if HW echo can was working? |
| 02:23.21 | jplank | HW echo can was available* |
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| 03:01.56 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:02.25 | scooby2 | weird music on hold works when a caller is first put in the queue but when an agent puts them on hold it goes to the default moh |
| 03:03.35 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 03:04.20 | joobie | TK, you alive? |
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| 03:16.11 | jplank | what would cause echo on a card with HW echo can? |
| 03:26.16 | ectospasm | jplank: echo that's so bad it can't be canceled |
| 03:26.32 | ectospasm | ...or maybe out of date drivers/firmware for the echo can |
| 03:26.49 | ectospasm | ...or the echo can isn't activated. |
| 03:27.04 | ectospasm | ...or even the echo can being faulty |
| 03:27.53 | ectospasm | ...or the base card is faulty |
| 03:28.13 | ectospasm | ...or simply the two cards (base and module) need to be reseated. |
| 03:28.23 | ectospasm | jplank: are you getting all of this? |
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| 03:39.21 | jplank | got them all |
| 03:40.22 | jplank | could a card that was pulled out of the slot a little cause echo? |
| 03:45.31 | ectospasm | jplank: it's counterintuitive, but yes that's possible. |
| 03:46.05 | jplank | the only reason I ask is this customer has been known to knock the amphenol cable out a bit |
| 03:47.14 | jplank | and when I think about it, the echo started when he said he "found a way to secure the amhenol cable" |
| 03:47.14 | ectospasm | jplank: yeah, a card that becomes partially unseated can exhibit all sorts of weird behavior. |
| 03:47.58 | jplank | if he "secured" the amphenol cable, but knocked into it, I'm sure theres a good chance it could of been pulled from the PCI slot a bit |
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| 04:10.59 | Nugget | http://www.syswear.com/view/tshirts?d=52 <-- heh |
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| 04:12.30 | drmessano | <PROTECTED> |
| 04:15.01 | carrar | Sure your floor is clean? |
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| 04:25.11 | scooby2 | is it music= or musiconhold= in queues.conf to specify the music on hold for each queue? |
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| 04:41.36 | coolthreads | just brought a voxzone x100p cards, trying to install zaptel |
| 04:41.49 | coolthreads | gets so far and then gives up |
| 04:42.48 | coolthreads | /usr/src/zaptel-1.4.12.1/kernel/ztdummy.c:202: error: struct hrtimer has no member named expires |
| 04:44.11 | coolthreads | is anyone able to give me any walkthroughs |
| 04:59.47 | coolthreads | any help in here that can point me to success |
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| 05:04.40 | coolthreads | ? |
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| 05:28.11 | coolthreads | algood, i solved my problem |
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| 06:58.28 | prxtien | [Jun 19 16:25:21] WARNING[1774]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)... i cant find much details on this error message, im running 1.6.0.5 |
| 07:03.55 | *** part/#asterisk jake[work] (n=Jake@pool-173-52-144-183.nycmny.east.verizon.net) |
| 07:05.03 | kaldemar | prxtien: pastebin a full CLI output of the failed call with iax2 set debug on, and then maybe someone will be able to help you. that warning is quite uninformative on its own. |
| 07:05.08 | kaldemar | ~pb |
| 07:05.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste , or , http://bin.cakephp.org/ |
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| 07:05.52 | prxtien | okay thanks |
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| 07:21.55 | Ryan09 | Hello, I'm just wondering if there's any way to get asterisk to start 'on hold' music from a random point in a file (as opposed to a random file in a folder) |
| 07:22.44 | Ryan09 | (asterisk 1.4 that is) |
| 07:30.40 | *** join/#asterisk casix (n=casix@212.36.71.109) |
| 07:30.50 | casix | hello |
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| 07:40.37 | *** join/#asterisk doug (i=doug@breakout.telerama.com) |
| 07:40.53 | doug | is there a way to display callerid to asterisk console via extensions.conf? |
| 07:41.25 | doug | like, after the connection messages |
| 07:41.54 | doug | verbose view |
| 07:42.16 | doug | hm, guess it'll do that if i just set a variable |
| 07:42.25 | casix | doug: NoOp(The caller id is: ${CALLERID(all)}) |
| 07:43.01 | jplank | NoOp(${CALLERID(all)}) |
| 07:43.04 | jplank | heh |
| 07:43.31 | doug | cool, thanks. |
| 07:45.14 | doug | hm, is there a good way to have a log file of calls? audit trail or somethin.. |
| 07:46.24 | casix | you can use the cdr |
| 07:46.35 | casix | via file or via mysql |
| 07:46.59 | doug | coolio |
| 07:48.11 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 07:49.48 | doug | huh, is it possible to do sms via asterisk in the us? |
| 07:50.05 | doug | i.e. is there an service provider for landline sms? |
| 07:51.21 | devyll | can anyone tell me where is the link from the pri trunk name and the dahdi configuration ? . For example, when I call Dial(DAHDI/g1/telnumber,150) will asterisk know to use a free channel from the Digium TE220 card ? or does DAHDI/g1 has to be defined somewhere in the confs for dahdi ? Everything is working perfectly inbound, but I can't seem to place any outbound calls. |
| 07:51.29 | *** join/#asterisk peterosd (n=Peter@c83-255-69-68.bredband.comhem.se) |
| 07:52.24 | kaldemar | devyll: it has to be define with "group => 1". all channels below that line belong to g1 unless otherwise defined. |
| 07:53.04 | kaldemar | devyll: and yes, asterisk will pick a free channel from the channel group if any exist. |
| 07:53.10 | devyll | in chan_dahdi.conf right ? |
| 07:54.01 | kaldemar | yes. look at the sample config file for examples. and don't mix this up with the trunkgroup parameter. |
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| 08:07.32 | cfh | hi all , where can i find an example of configuration of asterisk with polycom (kirk) dect phone ? |
| 08:10.01 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 08:10.47 | und3r | hello, anyone have a configuration template for Patton SN4634 -> Asterisk ? |
| 08:11.14 | cfh | the kirk server is registered to asterisk but when i try to dial one desct on the asterisk-console I see : Got SIP response 603 "Decline" back from ... |
| 08:12.00 | cfh | und3r : try this http://wildix.com/partner/patton_templates/ |
| 08:21.14 | devyll | kaldemar: does this say anything to you: http://asterisk.pastebin.com/d6438fef0 ? can you tell me if the configuration is still wrong or it's something with my provider ? |
| 08:21.30 | devyll | and more important, is there a list with the description of the "hugup causes" ? |
| 08:21.37 | *** join/#asterisk Enkhmunkh (n=Enkhmunk@202.131.231.19) |
| 08:22.34 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:8558:4990:c45b:7c95) |
| 08:23.07 | Enkhmunkh | Hello Guys! I am having CallerID problem |
| 08:23.11 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 08:23.22 | Enkhmunkh | on my * cli, WARNING[14315]: callerid.c:607 callerid_feed: Caller ID too long??? |
| 08:23.23 | Enkhmunkh | WARNING[14315]: chan_zap.c:6626 ss_thread: CallerID feed failed: Success |
| 08:23.23 | Enkhmunkh | WARNING[14315]: chan_zap.c:6726 ss_thread: CallerID returned with error on channel 'Zap/3-1' |
| 08:23.57 | Enkhmunkh | I don't know What is happening... Please help me |
| 08:24.11 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 08:24.30 | Enkhmunkh | Some FXO lines are detecting CallerID |
| 08:24.37 | *** join/#asterisk jad_jay (n=chatzill@public.axolys.fr) |
| 08:25.56 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:26.41 | Enkhmunkh | Help please, any suggestions! |
| 08:26.56 | AlmightyOatmeal | bea it with a sharp stick |
| 08:26.58 | AlmightyOatmeal | beat |
| 08:40.39 | coolthreads | anyone here familiar with x100p |
| 08:45.02 | *** join/#asterisk Jacke (i=jacke@valhalla.bofh.org) |
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| 09:09.09 | linast | Hi |
| 09:09.34 | linast | Thereis a way for don't allow an user of a manager to send the UserEvents ? |
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| 10:49.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:00.16 | *** join/#asterisk ludan (n=ludan@unaffiliated/ludan) |
| 11:00.16 | ludan | hi guys |
| 11:00.17 | ludan | is it possible to record a meetme conference?? |
| 11:00.17 | ludan | on a file |
| 11:04.07 | casix | ludan: http://tinyurl.com/l87y23 |
| 11:05.32 | ludan | casix: AHAHHAHAHAHAH |
| 11:05.33 | ludan | :D |
| 11:05.49 | casix | :) |
| 11:07.58 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 11:24.58 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 11:27.18 | devyll | can I make a dialplan where the pbx will initiate multiple outgoing calls and every call to be added to a confrence (meetme) ? . Like, a conference which invites (initates the calls) others to join there. |
| 11:32.52 | devyll | imagine that you call extension 1111 and that extension puts you in conference number 22 and initiates 5 predefined numbers and when each answeares it adds them in conference 22. |
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| 12:06.19 | *** part/#asterisk chrisdus (n=chrisdus@217.112.144.109) |
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| 12:27.18 | *** join/#asterisk jeff_phillips (n=ceramics@209-142-149-133.stat.centurytel.net) |
| 12:28.35 | jeff_phillips | I'm trying to setup an extension number I can dial that will just beep and record whatever I say, and then stop recording when I hang up. I want to adjust the levels so I'm trying to use MixMonitor() instead of Record(). How do I tell it to wait until I hang up? |
| 12:30.28 | [TK]D-Fender | jeff_phillips: Wait(50000) should do. |
| 12:31.21 | jeff_phillips | Oh, so if I just tell it to wait a long long long amount of time, it will terminate early upon hangup anyway? |
| 12:31.57 | jeff_phillips | cool. I guess for some reason I had just assumed it would leave it open to the posibility of recording silence after hanging up by doing that |
| 12:32.26 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 12:32.31 | [TK]D-Fender | jeff_phillips: No, when you hangup its game-over and * hits "h" |
| 12:32.34 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 12:33.18 | jeff_phillips | excellent |
| 12:36.04 | gr0mit | anyone used xorcom astribank interfaces? |
| 12:37.25 | [TK]D-Fender | gr0mit: What about them? |
| 12:38.18 | gr0mit | am working on a proposal toi pull out a nortel option 11 |
| 12:38.28 | gr0mit | 100 extens, 1 x pri |
| 12:38.39 | gr0mit | was thinking about using xorcom kit |
| 12:39.03 | gr0mit | any advice? |
| 12:39.16 | gr0mit | like 'NOOOOOOOOOOOOO' ? |
| 12:39.17 | [TK]D-Fender | gr0mit: I have never seen Xorcom anywhere near competitively priced for mass-analog |
| 12:39.33 | gr0mit | recommendation? |
| 12:39.41 | [TK]D-Fender | gr0mit: And I have personal issues against the though of usb, but others may differ on that. |
| 12:39.58 | gr0mit | am nervous - site is 250 acres |
| 12:39.59 | [TK]D-Fender | gr0mit: AudioCodes or Mediatrix SIP gateways. |
| 12:40.06 | gr0mit | lots of underground cables |
| 12:40.50 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 12:41.33 | *** join/#asterisk davevg-btwtech (n=davevg__@nj-67-76-177-147.sta.embarqhsd.net) |
| 12:41.49 | gr0mit | need to make sure lightening protection etc |
| 12:43.50 | jeff_phillips | gr0mit: I have an AudioCodes SIP gateway. Works pretty well but is a pain to configure |
| 12:43.54 | eppigy | hello |
| 12:43.56 | eppigy | i am dave |
| 12:44.12 | gr0mit | hmmm |
| 12:44.25 | jeff_phillips | i still haven't figured out how to get distinctive ring to work on it |
| 12:44.30 | gr0mit | hehe |
| 12:47.37 | jeff_phillips | You have to make a config file from scratch specifying the number of milliseconds of each ring burst in the ring patterns, and define all the regional tones yourself, then compile the file down into some proprietary binary format using a compiler / conversion utility they provide, and then upload that to the device's firmware, and then scratch your head wondering why it still behaves the same as it did previously |
| 12:48.23 | jeff_phillips | but aside from that it works quite well. Even have one of the 24 extensions ran half a mile away |
| 12:57.46 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:59.01 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 13:01.33 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 13:03.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.50 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-079-217.pools.arcor-ip.net) |
| 13:09.59 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 13:10.10 | Zeeek | It's FRIDAY!! |
| 13:10.52 | jaytee | yippie!!! wahoo!!! |
| 13:11.12 | eppigy | i am only working a half day today |
| 13:11.15 | eppigy | ^_________^ |
| 13:11.18 | mort_gib | What's up with you two blabbing idiots! |
| 13:11.39 | mort_gib | Yippie?? Grmpf |
| 13:11.45 | Zeeek | Friday is the day of the ... |
| 13:12.00 | beek | wait for it... |
| 13:12.05 | Zeeek | http://VoIPusersConference.org |
| 13:12.20 | Zeeek | and since every cloud has a silver lining |
| 13:12.30 | Zeeek | today asterisk has one too |
| 13:12.51 | eppigy | propoganda |
| 13:12.54 | Zeeek | Katty: is here |
| 13:13.01 | Zeeek | but where? |
| 13:13.07 | Zeeek | sniff, no hug? |
| 13:14.21 | gr0mit | wonder if vegastream is better? |
| 13:14.45 | gr0mit | is scared. taking my sone driving for the first time |
| 13:15.11 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:15.19 | Katty | pizza was a bad idea for breakfast. |
| 13:16.23 | Katty | hugs Zeeek |
| 13:16.35 | Zeeek | {{{{{{{{{{Katty}}}}}}}}}} |
| 13:16.51 | Katty | so groggy. |
| 13:16.55 | Zeeek | were you able to remove the cigarette butts before eating it? |
| 13:16.55 | Katty | so unfocusued |
| 13:16.58 | Katty | so....blah.... |
| 13:17.08 | [TK]D-Fender | Katty: c'mon... cold pizza & warm beer : breakfast of CHAMPIONS! |
| 13:17.08 | Katty | what? eww. |
| 13:17.19 | Katty | [TK]D-Fender: i'm giving up caffeine for a few weeks. |
| 13:17.19 | Zeeek | butts in the beer cans too |
| 13:17.31 | Katty | but i dont think i'm going to make it |
| 13:18.01 | Katty | eppigy: YOUR FAULT |
| 13:18.12 | Zeeek | get a room |
| 13:18.16 | Zeeek | a chat room |
| 13:18.21 | Katty | we have one. |
| 13:18.23 | Zeeek | oh |
| 13:18.28 | Katty | and you're in it. |
| 13:18.34 | Zeeek | oh. Oooooohhhhh |
| 13:18.42 | Zeeek | (I'll have what she's having) |
| 13:18.44 | gr0mit | had pizza for breakfast once in USA |
| 13:18.58 | Zeeek | Pizza for breakfast is ONLY good cold |
| 13:19.03 | Katty | gr0mit: it wasn't breakfast pizza, just leftovers from last night. |
| 13:19.06 | gr0mit | at 4am - when he was jetlagged |
| 13:19.18 | Zeeek | because there is such a thing as breakfast pizza? |
| 13:19.19 | Katty | lol |
| 13:19.21 | Zeeek | seriously? |
| 13:19.23 | Katty | Zeeek: yes. |
| 13:19.26 | Zeeek | no way |
| 13:19.28 | Katty | Zeeek: and it's yummy. |
| 13:19.34 | Zeeek | that's like breakfast G722 |
| 13:19.37 | gr0mit | sighs. Only in America! |
| 13:20.05 | Zeeek | although... there is a pizza here with foie gras on it. Have you any idea how rich that is? |
| 13:20.18 | Zeeek | and Katty that has every possible substance you hate in it |
| 13:20.18 | Katty | http://img.timeinc.net/recipes/i/recipes/ck/02/10/pizza-ck-522124-l.jpg |
| 13:20.28 | Zeeek | PLUS cruelty to animals |
| 13:20.49 | Katty | orly |
| 13:20.52 | Katty | oh. |
| 13:20.59 | Katty | they care orly at sally hansen by the way |
| 13:21.03 | Zeeek | so today at 12 Noon EDT, Asterisk in an EC2 instance |
| 13:21.08 | Katty | s/care/carry/ |
| 13:21.20 | Katty | ec2 |
| 13:21.22 | Zeeek | Orly? That's the Paris airport, the little one near town |
| 13:21.34 | Zeeek | Asterisk in Orly |
| 13:21.34 | gr0mit | south of paris |
| 13:21.35 | Katty | Orly is a brand of polish. |
| 13:21.44 | Zeeek | just a few miles south, yes |
| 13:21.45 | Katty | with an awesome rubberized bonding base coat. |
| 13:22.01 | Zeeek | WHy do the Poles want runner? |
| 13:22.19 | Katty | http://www.truthinaging.com/wp-content/uploads/2009/03/orly_bonder.jpg |
| 13:22.20 | Zeeek | why would you want to run asterisk in the cloud? |
| 13:22.38 | Zeeek | find the answers out soon enough |
| 13:22.48 | Katty | will there be caffeinated refreshments? |
| 13:22.55 | eppigy | I am listening to the eastern promises soundtrack |
| 13:22.59 | eppigy | it makes my heart hurt |
| 13:23.02 | Katty | oh this isn't going to happen today. |
| 13:23.05 | Katty | eppigy: link. |
| 13:23.15 | Zeeek | there are always refreshments, attutidu adjusting substances |
| 13:23.24 | Zeeek | ah, the plumber is coming I think |
| 13:23.34 | Katty | with caffeinated refreshments? |
| 13:23.41 | eppigy | http://www.amazon.com/Eastern-Promises-Howard-Shore/dp/B000UZ4D1C |
| 13:24.01 | Zeeek | Howard Stern |
| 13:24.10 | Katty | which track |
| 13:24.19 | Zeeek | Katty #voip-users-conference |
| 13:24.26 | Zeeek | see you there |
| 13:24.30 | Zeeek | and here |
| 13:24.34 | Zeeek | and everywhere |
| 13:24.35 | Katty | just here |
| 13:24.58 | Zeeek | that pizza thing made me hungry |
| 13:25.00 | Katty | unless i can get caffeinated refreshments. then i might have the focus to multitask |
| 13:25.06 | gr0mit | mmmmmh pizza |
| 13:25.29 | eppigy | Katty: eastern promises |
| 13:25.37 | eppigy | but pretty much all of them |
| 13:26.23 | Katty | k |
| 13:26.32 | Katty | oh no |
| 13:26.34 | Katty | sleepy musics |
| 13:26.41 | eppigy | D: |
| 13:27.20 | Katty | listens to something else. |
| 13:27.30 | Katty | eppigy: mymymymy |
| 13:28.08 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 13:28.48 | eppigy | :[ |
| 13:29.25 | Katty | pavaroti? |
| 13:29.31 | Pan3D | queues up Marching Band music |
| 13:30.15 | Blackvel | hi all |
| 13:30.48 | Katty | Pan3D: that's good stuff too |
| 13:30.59 | Blackvel | when I use a dial command with DIAL(Local/tech1&Local/tech2) what happens when the Local channel answers (Answer()) |
| 13:31.13 | *** join/#asterisk djMax (n=chatzill@66.92.91.132) |
| 13:31.26 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:31.36 | djMax | Is there a way to stop a "denial of service" attack from a phone number via caller id? |
| 13:31.47 | djMax | Blacklisting doesn't truly work because it still takes up a channel |
| 13:31.55 | Katty | telco |
| 13:32.07 | djMax | get them to block you mean? |
| 13:32.12 | Katty | ya |
| 13:32.13 | eppigy | not get thej |
| 13:32.15 | eppigy | them |
| 13:32.18 | eppigy | to go to their house |
| 13:32.25 | eppigy | and beat them within an inch of their life |
| 13:32.27 | djMax | given how slow they are with everything else, this'll be fun. :) |
| 13:32.29 | Katty | goes to eppigy's house. |
| 13:32.36 | eppigy | I mean what coudl they do but block it |
| 13:32.41 | eppigy | Katty: come one over |
| 13:32.44 | eppigy | *on |
| 13:32.50 | djMax | yeah, it's a very strange one because it's an inbound fax but there's no damn fax. Lots of people complaining about them apparently |
| 13:32.55 | eppigy | it is nice and comfy |
| 13:32.57 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 13:32.58 | djMax | not sure what they're gaining |
| 13:33.15 | Katty | who knows. |
| 13:33.15 | djMax | makes me want to DOS them right back. :) |
| 13:33.21 | Katty | i'm sure people have done more, for less. |
| 13:33.31 | djMax | so true. |
| 13:33.46 | Katty | pavarotti is so relaxing. |
| 13:33.52 | Katty | almost too relaxing. |
| 13:34.04 | gr0mit | does nortel support SMDI for MWI ? |
| 13:34.17 | Katty | why don't you call them and ask. |
| 13:34.27 | Katty | that's what i'd do. |
| 13:34.54 | gr0mit | calling Nortel is like calling my employer- get passed from pillar to post |
| 13:34.57 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:35.03 | Blackvel | will there be some briding (the inbound channel of the IVR is already answered). how many channels are active? 1? 2? what about the dailplan? is there some way to hangup the local channel or send busy back, and jump back to BUSY state of outer DIAL(Local/tech1&Local/tech2) and continue with the outer ivr dialplan? it looks to me that once the local channel is answered, I have to use GOTO for subsequent ivr-dialplan flows as th |
| 13:35.32 | Katty | 18004667835 |
| 13:35.59 | Katty | Blackvel: you lost me at briding. |
| 13:36.11 | Katty | Blackvel: let's back up a step. what are you attempting to accomplish. |
| 13:36.31 | gr0mit | Please hold the line whilst we triy to find an employee who still works for Nortel...' |
| 13:36.36 | Blackvel | sorry to many msg in between ;) |
| 13:36.37 | eppigy | I might go get breakfast |
| 13:36.44 | Blackvel | when I use a dial command with DIAL(Local/tech1&Local/tech2) what happens when the Local channel answers (Answer()) |
| 13:36.47 | Katty | eppigy: and caffeinated refreshment? |
| 13:36.52 | eppigy | negative |
| 13:36.58 | Katty | :> |
| 13:37.04 | eppigy | breakfast burritoes |
| 13:37.10 | Blackvel | the thing is that |
| 13:37.11 | Katty | mcdonalds? |
| 13:37.15 | Katty | from Regan county |
| 13:37.16 | Blackvel | DIAL statement never gets busy |
| 13:37.25 | eppigy | yesh |
| 13:37.25 | Blackvel | only for one phone |
| 13:37.27 | Katty | k |
| 13:37.57 | Blackvel | but when calling 2-3 phones (& syntax), even when the 1st phone is in BUSY state, the others keep ringing |
| 13:38.13 | Blackvel | there is one statement: pickup "wins" |
| 13:38.21 | Katty | timeout? |
| 13:38.32 | Blackvel | when i pickup any one (of 2-3), the others stop and dial ends |
| 13:38.41 | Katty | Dial(local/whoever&local/whoever,15) |
| 13:38.51 | Blackvel | well yes |
| 13:38.57 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 13:39.01 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:39.04 | Blackvel | then it tries 15 seconds the other phones even i am talking one the 1st |
| 13:39.17 | Blackvel | i like to have : busy "wins" |
| 13:39.29 | gr0mit | then you need clever dialplan |
| 13:39.45 | Blackvel | i would like that the dial ends on first busy and goes to other busy handling |
| 13:39.53 | Blackvel | Dial(local/whoever&local/whoever,15) -> DIALSTATUS = BUSY |
| 13:39.59 | gr0mit | so check channel state |
| 13:40.03 | gr0mit | on all phones |
| 13:40.10 | Blackvel | i have that in my dailplan, but it is not used |
| 13:40.11 | gr0mit | before the Dial |
| 13:41.11 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:41.20 | gr0mit | use logic like if SIP/a is available then ifd SIP/B is available then if SIP/c is available goto (ringall phones) else busy |
| 13:41.22 | Blackvel | leifmadsen gave me the tipp: let the Local channel answer |
| 13:41.31 | Blackvel | then it will stop ringing the other phones |
| 13:41.31 | Blackvel | ---incoming---> Answer() ---> Asterisk ---> Local ----> Answer() |
| 13:41.37 | gr0mit | yuk. |
| 13:41.38 | Blackvel | and it does |
| 13:41.49 | gr0mit | never answer a channel unless you need to |
| 13:41.49 | KavanS | Blackvel, I do something similar....if it's busy on the first SIP dial, then go to voicemail....if not ring other SIP extensions and cell phone |
| 13:41.55 | KavanS | the regular follow me function wasn't good enough |
| 13:42.11 | Blackvel | but i do not understand what is going on behind the scenes. will the two channels (inbound ivr already answer, local channel) be bridged? |
| 13:42.23 | Blackvel | what about the dialplan? will it continue in the inner local channel context? |
| 13:42.40 | Blackvel | how do I get back to outer Dial(local/whoever&local/whoever,15) statement and busy extension? |
| 13:42.43 | jeff_phillips | i can't seem to get this mixmonitor command to actually record anything, but it pretends it is |
| 13:42.43 | [TK]D-Fender | Blackvel: 1st answer stops dialing all of the rest. |
| 13:42.56 | Blackvel | and i love that |
| 13:43.04 | KavanS | jeff_phillips, try the monitor command....and then using a post processing script |
| 13:43.14 | Blackvel | but i am too un-clevel to understand what to do with my ivr dialplan |
| 13:43.20 | gr0mit | so check exten status of all 3 before you execute dial |
| 13:43.30 | jeff_phillips | KavanS: But shouldn't mixmonitor at least record something?! |
| 13:43.39 | Blackvel | i mean there is no way to let the Dial(local/whoever&local/whoever,15) statement go into BUSY handling? |
| 13:43.54 | Blackvel | there is "no return" from local channel? |
| 13:44.04 | [TK]D-Fender | Blackvel: What defiens a bunch of people as "busy" if they aren't ALL busy? |
| 13:44.12 | gr0mit | not uness all chas are busy, but ia m guessing here |
| 13:44.20 | Blackvel | would I have to continue coding my dialplan in the local channel context / GOTO to my other ivr dialplan? |
| 13:44.20 | KavanS | jeff_phillips, probably...I had good luck with monitor, not sure about mixmonitor |
| 13:44.24 | [TK]D-Fender | Blackvel: Thats the point of dialing multiple people, so SOMEONE can answer, not jsut so yuo have more reasons to GIVE UP |
| 13:44.26 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 13:44.50 | Blackvel | i see |
| 13:44.51 | gr0mit | [TK]D-Fender, i think i see where he is coming from. |
| 13:44.59 | [TK]D-Fender | gr0mit: Wish i could. |
| 13:45.01 | KavanS | Blackvel, no, I have a setup similar to that... |
| 13:45.09 | KavanS | Blackvel, sometimes it is nice to evaluate if someone is on the phone or not |
| 13:45.09 | Blackvel | so its about 3 ppl sitting in my company, not just me alone |
| 13:45.11 | [TK]D-Fender | gr0mit: Try 10 people and give up because of the 1st... |
| 13:45.16 | gr0mit | so if i am at home, and I am on the phone, i don't want a second call to ring the other phones |
| 13:45.27 | gr0mit | i want it to go to VM 'gr0m is on the phone' |
| 13:45.40 | KavanS | gr0mit, exactly! |
| 13:45.46 | [TK]D-Fender | gr0mit: then he should check the before dialing. |
| 13:45.53 | KavanS | I don't want my cell phone to start ringing while I'm already dealing with another person |
| 13:45.57 | gr0mit | which is what I said!!! |
| 13:45.57 | KavanS | it just adds to the drama |
| 13:45.58 | jeff_phillips | if you are on phone A, you aren't going to answer phones B, C, or D, so eventually those will go to voice mail anyway. :) |
| 13:46.03 | Blackvel | looks like that it is not that too much clever to use Dial(local/whoever&local/whoever,15) or Dial(local/whoever&local/whoever&local/xlite,15) when its about 1 person |
| 13:46.06 | [TK]D-Fender | gr0mit: And jsut because someone is on the phone doesn't make them busy or return a "busy" status. |
| 13:46.14 | KavanS | jeff_phillips, but it sounds really lame if it's ringing in the background :) |
| 13:46.30 | KavanS | one sec blackvel, I'll share some snippets |
| 13:46.42 | Blackvel | so i have to code my own dailing of 3 devices |
| 13:46.48 | [TK]D-Fender | Blackvel: "core show application chanisavail" <--- |
| 13:46.48 | Blackvel | and not use Dial(local/whoever&local/whoever,15).... |
| 13:46.58 | [TK]D-Fender | Blackvel: No. |
| 13:47.14 | [TK]D-Fender | blackBecause I highly doubt any device will report as "busy" to your Dial() |
| 13:47.27 | Blackvel | true |
| 13:47.49 | Blackvel | i "think" (could not verify) it does only , when ALL are busy |
| 13:47.59 | Blackvel | that would make sense |
| 13:48.20 | Blackvel | i read about chanisavail and also read it MUST not be used for BUSY checking |
| 13:48.39 | Blackvel | i really hate to say i am on 1.2 (because had problems with segfaults on 1.4 with patton media gw) |
| 13:48.46 | Blackvel | no devstate |
| 13:48.50 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 13:48.56 | [TK]D-Fender | [09:48]<Blackvel>i read about chanisavail and also read it MUST not be used for BUSY checking <- Incorrect |
| 13:49.07 | Blackvel | others programmed with devstate (and it works) and probably would in my situation |
| 13:49.10 | KavanS | blackvel: check pm, I pasted my setup |
| 13:49.20 | Blackvel | probably there is something available like using GROUP() |
| 13:49.26 | Blackvel | I hate to overcomplicate things ;) |
| 13:49.39 | Blackvel | ip-phone-forum.de |
| 13:49.47 | Blackvel | some more ppl need stuff like me hehe |
| 13:50.03 | Blackvel | a simple option to Dial(local/whoever&local/whoever,15) would have been the solution for me |
| 13:50.09 | [TK]D-Fender | [09:49]<Blackvel>I hate to overcomplicate things ;) <- But you're so good at it! |
| 13:50.39 | [TK]D-Fender | Blackvel: Again the odds of your devices actually returning "busy" are pretty low. |
| 13:50.47 | KavanS | blackvel: my situation is complicated that I have a single PSTN, so I have to include an local context to execute and evaluate if Zap is congested |
| 13:51.16 | Blackvel | [TK]D-Fender maybe, but I try not to be |
| 13:52.12 | KavanS | blackvel: http://www.pastie.org/517563 |
| 13:52.56 | Blackvel | saw it |
| 13:53.01 | Katty | eppigy: i gave in. headaches started. |
| 13:53.03 | Blackvel | dont get it |
| 13:53.09 | Katty | eppigy: must ween self back, it seems. |
| 13:53.17 | Blackvel | you are still ringing three devices in one dial :) |
| 13:53.30 | KavanS | blackvel: it evaluates dial status at step 7 -9 |
| 13:53.34 | KavanS | dials first extension for 5 secs |
| 13:53.40 | KavanS | if it's busy, then goto vm |
| 13:53.47 | KavanS | if not, continue on....and ring the fuck out of everyone |
| 13:54.30 | Blackvel | hanIsAvail is not a solution to tell you conclusively whether the channel is busy or not, it is primarily to tell you whether it would be possible to send a call there |
| 13:54.36 | Blackvel | http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail |
| 13:54.49 | KavanS | blackvel: try dialstatus :) |
| 13:55.15 | eppigy | Katty: yesh |
| 13:56.45 | Blackvel | sorry i understand nada |
| 13:56.54 | Blackvel | Dial(SIP/201,5,${DIALOPTIONS}m) |
| 13:57.08 | Blackvel | if that is busy (you check for it) |
| 13:57.11 | Blackvel | you do this: Dial(LOCAL/1@kavandial&SIP/201&SIP/401,30,${DIALOPTIONS}m) |
| 13:57.16 | KavanS | yep |
| 13:57.25 | Blackvel | what are these devices? |
| 13:57.38 | KavanS | LOCAL/1@kavandial is the [kavandial] context |
| 13:57.48 | KavanS | that dials my cell phone ;) |
| 13:58.01 | KavanS | also included on pastie... |
| 13:58.02 | Blackvel | and hwat is 401? |
| 13:58.16 | KavanS | no, 201 and 401 are SIP extensions of mine |
| 13:58.30 | KavanS | so it rings 3 phones if the primary SIP line (201) is not busy |
| 13:58.36 | KavanS | including 201... |
| 13:58.51 | Blackvel | but 201 was busy, so you ring it again? |
| 13:59.11 | KavanS | exten => s,9,GotoIf($["${DIALSTATUS}" = "BUSY"]?11:10) |
| 13:59.22 | KavanS | ^^^ if it's busy, goto 11 |
| 13:59.30 | *** join/#asterisk dajhorn (n=chatzill@206.16.96.160) |
| 13:59.31 | KavanS | then goto s-BUSY at the bottom |
| 13:59.33 | KavanS | i.e. voicemail |
| 13:59.58 | Blackvel | now i got it |
| 14:00.29 | KavanS | yeah it's hard to read through, [TK]D-Fender is right about making things simple....but I needed something that checked, because I don't like a bunch of phones ringing at once unless it's necessary :P |
| 14:00.35 | *** join/#asterisk gunter (n=user@87.127.97.39) |
| 14:00.43 | Blackvel | you ring it, if you do not pickup you ring cellphone + main phone + 401. if the first dial on mainphone gets busy, you forward to voicemail :P |
| 14:00.56 | KavanS | yep, that's how it works |
| 14:01.08 | KavanS | at the bottom you'll also notice the hangup definition.... |
| 14:01.12 | KavanS | "missedcall" |
| 14:01.16 | KavanS | in case someone hangs up during this process |
| 14:01.20 | KavanS | I get a notification via email :) |
| 14:01.34 | Blackvel | but still that gets my back to my question |
| 14:01.46 | Blackvel | when the local channel answers (even on busy state) |
| 14:01.53 | Blackvel | what happens with the outer channel? bridged? |
| 14:02.02 | Blackvel | will there be only one active inbound (ivr) channel? |
| 14:02.03 | KavanS | Blackvel, that's a question for someone l33ter than I... |
| 14:02.14 | [TK]D-Fender | With multi-dial you aren't going to get a BUSY back because SOMEONE is not going to report "busy". |
| 14:02.23 | [TK]D-Fender | You are running around in circles |
| 14:02.30 | Blackvel | <PROTECTED> |
| 14:02.50 | Blackvel | correct , no busy |
| 14:03.01 | KavanS | [TK]D-Fender, I had to get around that by executing the "local" dial, then you can evaluate with multi-dial |
| 14:03.03 | Blackvel | lets say the multi-dial dails local channel |
| 14:03.09 | Blackvel | and that local dial aborts with busy |
| 14:03.13 | Blackvel | and that local channel answers |
| 14:03.29 | [TK]D-Fender | Blackvel: You can't abort busy & answer. |
| 14:03.54 | [TK]D-Fender | Blackvel: either / or |
| 14:06.04 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 14:06.48 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:10.07 | Blackvel | http://pastebin.com/d2ba02c46 |
| 14:10.26 | Blackvel | so probably the best way is not to use the multi-dial .... or |
| 14:11.16 | Blackvel | stay in the local context (which got answered), play the busy-prompt stuff to the user and continue with IVR stuff (like choose from the menu what you want: callback, voicemail, forwading to mobile phone) |
| 14:11.59 | Blackvel | and i have that already coded in the ivr (BUSY handling for multi-dial - which the flow can't get back to as simple "return") |
| 14:12.23 | [TK]D-Fender | Blackvel: I've told you repeatedly this method is not going to work.... |
| 14:12.36 | [TK]D-Fender | Blackvel: "core show application chanisavail" <--- pay attention when I hand you the answer. |
| 14:13.14 | jeff_phillips | Well this is strange. I have MixMonitor with the command option specified. I call, it records silence (don't know why it isn't recording me talking). Then when I hang up, the command executes as it should. But, why isn't it actually recording? |
| 14:13.16 | Blackvel | ok, i acept and give up |
| 14:13.20 | Blackvel | :) |
| 14:14.03 | Blackvel | why did ppl invent devstate when chanisavail is sufficent? and why is wiki wrong again |
| 14:14.06 | KavanS | Blackvel, use my example...it works and does roughly the same thing |
| 14:14.31 | KavanS | i evaluate dialstatus during a multidial |
| 14:14.37 | KavanS | by executing a "local" dial command |
| 14:14.46 | KavanS | and then in that context, EVALUATE your dialstatus :) |
| 14:14.52 | Blackvel | KavanS: can you tell me what it is a VERY BAD idea to answer the local channel and continue with my ivr local (using GOTO()) on pastebin line #14? |
| 14:15.14 | KavanS | Why would you not answer the call earlier? |
| 14:15.20 | Blackvel | i do the same, already (well not exactly the same way) :) |
| 14:15.48 | KavanS | I don't understand why you wouldn't just answer, play music etc. |
| 14:15.59 | KavanS | I guess I was under a different assumption as to your goal |
| 14:16.23 | [TK]D-Fender | [10:13]<Blackvel>why did ppl invent devstate when chanisavail is sufficent? and why is wiki wrong again <- devstate has NOTHING to do with ChanIsAvail. next the WIKI gets contributions from many well-meaning but incompetent schmucks |
| 14:16.46 | [TK]D-Fender | blackAND... I don't know what page specifically you are referring to so I'll defer passing judgement on them for the moment |
| 14:17.21 | Blackvel | http://www.ip-phone-forum.de/showthread.php?t=193880 |
| 14:17.43 | Blackvel | one user had the problem and used devstate for busy checking |
| 14:17.51 | Blackvel | maybe its the wrong way, as you tell me |
| 14:18.08 | Blackvel | http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail |
| 14:18.11 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:19.00 | Blackvel | ChanIsAvail is not a solution to tell you conclusively whether the channel is busy or not |
| 14:19.06 | Blackvel | does my English suxx? |
| 14:19.07 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 14:19.17 | [TK]D-Fender | Blackvel: That too... |
| 14:19.27 | Blackvel | what does this sentence mean? |
| 14:19.28 | [TK]D-Fender | Blackvel: And that one stupid forum post means nothing |
| 14:19.46 | [TK]D-Fender | Blackvel: CHANISAVAIL <- It works. Read the instructions and go DO IT |
| 14:20.00 | Blackvel | i know...it's just a solution from someone else (and i can not/will not use it anyways) |
| 14:20.40 | Blackvel | ok [TK]D-Fender, before getting you mad..... i will follow your suggestion and programm using it |
| 14:21.44 | eppigy | breakfast burritoes ^_______________^ |
| 14:21.49 | *** join/#asterisk mapoupier (n=marcandr@MTRLPQ19-1279392437.sdsl.bell.ca) |
| 14:22.07 | Blackvel | sorry to ask, has one answerd my question already how the dialplan logic would flow if a local channel has answered with "ANSWER"? code the logic into the context of the local channel? |
| 14:22.11 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 14:22.23 | Blackvel | i really just want to UNDERSTAND it :) |
| 14:22.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:23.01 | Blackvel | btw |
| 14:23.13 | Blackvel | i am using database put whitelist extensivley |
| 14:23.31 | Blackvel | have versions 1.4/1.6 introduced the features of comments? |
| 14:23.43 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 14:23.56 | Blackvel | i do not know the (temporary or sometimes permanent) whitelisted numbers after 1-2 weeks anymore |
| 14:24.12 | Blackvel | would love to be able to see a comment... |
| 14:24.14 | [TK]D-Fender | Blackvel: Multi-device dial only cares if 1 device answeres the call. Any kind of failure from any 1 member it does not give a rat's ass about. So nothing you do in the local channel means shit. |
| 14:24.41 | [TK]D-Fender | Blackvel: there is no "return a give up all other attempts" flag. This is a dead issue. |
| 14:25.15 | Blackvel | yes I understand |
| 14:25.16 | [TK]D-Fender | Blackvel: If you answer then then think you're going to ahng up then you are killing the call. Again a dead issue. Its as easy as it appears. your determination of status must happen before the call. |
| 14:25.55 | Blackvel | exten => s,n,Answer() |
| 14:26.05 | Blackvel | if the local channel does this manually (maybe it should not) |
| 14:26.17 | Blackvel | what happens spoken from a technical side |
| 14:26.25 | Blackvel | briding two channels into one? |
| 14:27.01 | [TK]D-Fender | Blackvel: huh? |
| 14:27.19 | Blackvel | arent there two channels? the pstn sip channel and the local channel? |
| 14:27.50 | [TK]D-Fender | Blackvel: Typically whatever endpoints were involved break out of local unless you use the "/n" option. |
| 14:28.22 | Blackvel | i tried playback(mymessage) |
| 14:28.25 | Blackvel | and it worked :) |
| 14:28.53 | Blackvel | so the local gets somehow "merged" into the outer / exisisting channel (had no /n option)? |
| 14:29.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:30.06 | *** join/#asterisk josemslopes (n=jlopes@adonis.iportalmais.pt) |
| 14:30.19 | [TK]D-Fender | Blackvel: Yes, it will do that, i'm referring to if you Dial() within your local channel |
| 14:30.45 | *** join/#asterisk salzh (n=Administ@122.144.138.5) |
| 14:30.49 | [TK]D-Fender | Blackvel: Normall once tha answers the local is broken off and the interior Dial merges back to the oter |
| 14:30.51 | [TK]D-Fender | outer* |
| 14:31.17 | Katty | :> |
| 14:31.27 | eppigy | TRABAJO |
| 14:31.31 | josemslopes | i am getting the following error to access snmp module: ASTERISK-MIB::astVersionString.0 = No Such Object available on this agent at this OID |
| 14:31.51 | josemslopes | i use snmpget -v 2c -c private localhost ASTERISK-MIB::astVersionString.0 |
| 14:31.57 | Blackvel | and the dialplan logic continues in the local context... |
| 14:32.28 | Blackvel | was /n option only about variables or was it also about how the local channel gets merged into the existing ivr channel? |
| 14:32.40 | Blackvel | didn't understand that completely from docs |
| 14:32.42 | jeff_phillips | Could anyone help me understand what is happening here? http://pastebin.com/d772006f7 |
| 14:33.22 | jeff_phillips | I'm trying to do airport style paging. You would dial 3205, and speak after the beep and hang up. Then the PA loud speakers would play your announcement twice preceeded by a little alert jingle to get people's attention. |
| 14:34.01 | jeff_phillips | it goes through the motions, but the mixmonitor just creates a wav file of silence that is oddly longer than I would have expected, and its playback is skipped (very quickly repeats the alert tones) |
| 14:34.22 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 14:34.48 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:34.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:35.05 | Katty | deeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeewayne. |
| 14:35.23 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:35.24 | Deeewayne | hugs Katty |
| 14:35.34 | Deeewayne | hello :-) |
| 14:36.17 | jeff_phillips | i'm not seeing a reason for it to record silence instead of my voice. |
| 14:37.50 | jeff_phillips | The record() function worked as a substitute to mixmonitor, but I would like to increase the volume levels |
| 14:39.39 | KavanS | jeff_phillips, post processing script :) |
| 14:39.55 | Katty | hugs on Deeewayne |
| 14:40.30 | Deeewayne | puts Katty on a Ferris Wheel |
| 14:40.58 | jeff_phillips | KavanS: Well okay, I guess I could change my approach, but ... is MixMonitor suppose to be non-functional? lol |
| 14:41.28 | KavanS | jeff_phillips, I killed a portion of my life trying to get monitoring to work for all incoming/outgoing calls, from my experience monitor + post processing = the way to go |
| 14:41.38 | KavanS | in the end I have my calls (including conferences) recorded in .ogg format |
| 14:42.11 | Katty | Deeewayne: :< |
| 14:42.32 | jeff_phillips | I had started with having found some bits of code someone posted that used the record() function but required you to press the # key when done recording, otherwise hanging up would just omit the recording entirely |
| 14:43.02 | jeff_phillips | I switched to mixmonitor in hopes of increasing the volume level of the recording, and to have the recording end upon hanging up automatically |
| 14:43.17 | jeff_phillips | unfortunately it seems to have reduced my volume to zero |
| 14:45.43 | jeff_phillips | alright well... what can I use to amplify the recorded file after it has been created? |
| 14:46.08 | KavanS | ffmpeg |
| 14:47.00 | KavanS | jeff_phillips, http://www.pastie.org/517619 |
| 14:47.07 | KavanS | I use a simple script to mix and convert |
| 14:47.15 | KavanS | you could modify that a little bit (ffmpeg) and increase volume |
| 14:47.22 | KavanS | or maybe sox has such options built in, I am not sure |
| 14:47.47 | jeff_phillips | hmm |
| 14:49.10 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 14:50.20 | Zeeek | so we'll be starting in about an hour. Want to test your g722 SIP client? Join us #voip-users-conference any time and on the conference bridge an hour from now |
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| 15:04.43 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:05.22 | rue_mohr | all refs to the hwec say its 1024 taps, but dosn't say if thats fixed or a max, yesterday they say there was terrible echo |
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| 15:05.55 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:06.27 | Katty | jaytee: ! |
| 15:06.39 | Katty | jaytee: we'v enot hugged today :< |
| 15:06.42 | Katty | hugs jaytee |
| 15:07.39 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 15:07.43 | Zeeek | Katty, by hugging others you have devalued my hugs |
| 15:08.04 | Zeeek | I can't cherish them as much now |
| 15:08.05 | flujan | hello guys, I have a sip trunk connecting two asterisk boxes... Call arrive on the machine 2 but is not transfered to the machine one. |
| 15:08.11 | rue_mohr | so digium says "risk free garuntee" all over it, if I cant get the volume and echo issues sorted, wonder if I should call them on that |
| 15:08.15 | flujan | http://pastie.org/517632 |
| 15:08.17 | flujan | any ideas? |
| 15:08.42 | Zeeek | rue_mohr: have you contacted support? Yes, if there's a guarantee, by all means you should call them |
| 15:08.58 | rue_mohr | its getting pretty frustrating |
| 15:10.06 | rue_mohr | I'm wondering if I should go with an ethernet channelbank |
| 15:10.29 | [TK]D-Fender | rue_mohr: How many lines? |
| 15:10.52 | rue_mohr | [TK]D-Fender, I dont mean to sound like a broken record, 4fxo and 2 fxs |
| 15:11.01 | rue_mohr | as per the tdm800 I have now |
| 15:11.45 | [TK]D-Fender | rue_mohr: http://www.telephonydepot.com/Catalog/Sangoma-B-Series/B600D-Analog-Voice-Card |
| 15:12.17 | [TK]D-Fender | rue_mohr: Get an ATA for the remaining port if you really need it |
| 15:12.25 | rue_mohr | no I can get ethernet devices for a fraction of that |
| 15:12.49 | [TK]D-Fender | rue_mohr: PCI with EC for < $500 USD? |
| 15:12.53 | rue_mohr | for under $600 I can get a 4 fxo to ethernet and a 2 fxs from ethernet |
| 15:12.53 | [TK]D-Fender | rue_mohr: Really... |
| 15:13.00 | jaytee | hugs Katty |
| 15:13.02 | [TK]D-Fender | rue_mohr: What moddel? |
| 15:13.04 | Katty | Zeeek: you're weird. |
| 15:13.07 | rue_mohr | not pci, ethernet apliances |
| 15:13.17 | [TK]D-Fender | rue_mohr: Ok, What model? |
| 15:13.19 | rue_mohr | I wont be abel to find it fast enough |
| 15:13.32 | rue_mohr | I found it yesterday, a few of them |
| 15:13.35 | Zeeek | Katty: why? |
| 15:13.38 | rue_mohr | iirc made by cisco |
| 15:13.43 | jeff_phillips | I got a 24 FXS channel audiocodes MP124 ethernet gateway for $220 on ebay |
| 15:13.43 | rue_mohr | (linksys) |
| 15:13.45 | Zeeek | Girls used to say that in high school |
| 15:13.55 | [TK]D-Fender | rue_mohr: Linksys for FXO? Which? |
| 15:14.03 | rue_mohr | something 400 |
| 15:14.17 | [TK]D-Fender | rue_mohr: I wouldn't put SPA-3XXX into production mainline business use... |
| 15:14.20 | rue_mohr | not spa, where are my notes |
| 15:14.24 | [TK]D-Fender | rue_mohr: OMG, the 400?!?! EWW!!!! |
| 15:14.31 | [TK]D-Fender | rue_mohr: DUMB POS device |
| 15:14.36 | rue_mohr | hey, this tdm800 is killing me |
| 15:14.45 | rue_mohr | nobody even knows how to properly adjust the gains |
| 15:14.52 | [TK]D-Fender | rue_mohr: You can't separately address the ports on it and other craziness. |
| 15:14.54 | *** join/#asterisk gunter (n=user@87.127.97.39) |
| 15:15.00 | rue_mohr | and today I'm trying to reverse engineer it to figure it out |
| 15:15.16 | [TK]D-Fender | rue_mohr: You know my verdict on it, I've said it for months. |
| 15:15.25 | Katty | Zeeek: i wonder why.... |
| 15:15.29 | rue_mohr | yea you said go buy a sagnoma |
| 15:15.29 | Katty | eppigy: did you have breakfast? |
| 15:15.32 | eppigy | yesh |
| 15:15.36 | Katty | eppigy: burritos? |
| 15:15.37 | jeff_phillips | reverse engineering a backwards-engineered device ......hmm... wouldn't that make it forwards again? |
| 15:15.38 | eppigy | BREAKFAST BURRITOE |
| 15:15.41 | [TK]D-Fender | rue_mohr: Why you haven't returned it for a refund I jsut don't know... |
| 15:15.44 | eppigy | PLURAL |
| 15:15.56 | Katty | from where |
| 15:15.59 | eppigy | mcdonalds |
| 15:16.01 | eppigy | :D |
| 15:16.07 | Katty | #4? |
| 15:16.11 | eppigy | negative |
| 15:16.12 | rue_mohr | jeff_phillips, nothing seems to say what the adc resolution is on the tdm800 |
| 15:16.15 | eppigy | I already had a drink |
| 15:16.22 | Katty | ohisee. |
| 15:16.28 | Katty | WHAT WERE YOU DRINKING |
| 15:16.29 | eppigy | i have removed my flip flops |
| 15:16.32 | rue_mohr | I think its 12 bit, and that the ulaw codec expects 16 bit, which causes a 12db loss |
| 15:16.45 | eppigy | fuji water |
| 15:16.49 | Katty | oooh :> |
| 15:16.54 | rue_mohr | but ofcourse nobody seems to know this stuff |
| 15:16.56 | Katty | i can't wear flipflops to work :< |
| 15:17.11 | Katty | gots to look all schnazzy |
| 15:17.14 | rue_mohr | offers Katty some shift registers |
| 15:17.21 | eppigy | I can only wear them on friday |
| 15:17.30 | Katty | what's a shift register |
| 15:17.36 | eppigy | all other days I am max_pimpin |
| 15:18.19 | BlargMaN00 | does anyone know if it is normal on a call file, when you place the call to the Channel: XXXX, and you answer it, then when it dials out to the connecting extension, you get no audio until the call is actually connected... any ideas?? |
| 15:18.20 | Katty | you so are. |
| 15:18.21 | Katty | do. |
| 15:18.26 | Katty | whatever. |
| 15:18.34 | Zeeek | Katty: you'd know why only if we could go back in time 50 years to when I was in high school |
| 15:18.40 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 15:18.59 | Zeeek | ok, not quite 50 but nearly |
| 15:19.05 | Katty | Zeeek: i'm sure all of them would be weird. |
| 15:19.07 | Katty | Zeeek: and sexist. |
| 15:19.11 | Zeeek | who? |
| 15:19.12 | Katty | Zeeek: i wouldn't fit in very well. |
| 15:19.19 | Katty | 50 years ago peoples. |
| 15:19.31 | jeff_phillips | BlargMaN00: How would you have audio *before* the call is connected anyway??? |
| 15:19.36 | Katty | 1 sexist comment and i'd go on a killing spree. |
| 15:19.46 | Zeeek | we smoked, we drank, we ***** |
| 15:19.46 | BlargMaN00 | it's almost like the audio is not bridging until almost 1000ms after the two channels are answered, and audio is established... |
| 15:20.04 | BlargMaN00 | jeff_phillips: audio => ringing... |
| 15:20.09 | eppigy | PUPRLE DRANK |
| 15:20.54 | jaytee | TRABAJO, TRABAJO, RAPIDAMENTE!!! |
| 15:20.55 | jeff_phillips | oh |
| 15:20.58 | Katty | http://theinvisibleagent.files.wordpress.com/2008/12/sexist-ad-3.jpg |
| 15:21.09 | Katty | highly inappropriate. |
| 15:21.42 | eppigy | DONDE ESTA |
| 15:21.49 | Katty | el gato con queso |
| 15:21.56 | eppigy | yesh |
| 15:22.01 | Katty | el bano |
| 15:22.02 | BlargMaN00 | jeff_phillips: my employees are not going to want to here silence, and then all of a sudden, half of a "Hello..." |
| 15:22.25 | eppigy | SI ME GUSTA |
| 15:22.29 | jeff_phillips | i don't like silence either. it is making me crazy |
| 15:23.21 | eppigy | lift off in t minus twenty-seven minutes and counting |
| 15:23.25 | rue_mohr | [TK]D-Fender, would you say the majority of sucessfull installations use the sangoma and not digium cards? |
| 15:23.32 | eppigy | how dare you |
| 15:23.53 | russellb | that is not true at all. |
| 15:23.57 | Katty | i would say the majority of successfull installation use smart people. |
| 15:24.02 | russellb | Katty: +1 |
| 15:24.04 | Zeeek | Katty +1 |
| 15:24.16 | rue_mohr | the lines are quiet by 11db |
| 15:24.21 | jeff_phillips | smart people use the majority of installations |
| 15:24.21 | rue_mohr | 11db! |
| 15:24.25 | Katty | what's with the +1? are we keeping score now? |
| 15:24.29 | [TK]D-Fender | rue_mohr: 100% success 90% of my clients with other cards had issues. Replacement = 100% happy instantly. |
| 15:24.29 | Zeeek | Katty +Russellb 1 |
| 15:24.50 | russellb | o.O |
| 15:25.33 | Katty | that does not parse. |
| 15:25.35 | Katty | but thanks, i think. |
| 15:25.47 | Katty | eppigy: t MINUS 29 minutes! |
| 15:26.01 | eppigy | YES |
| 15:26.06 | russellb | O.O |
| 15:26.07 | russellb | until WHAT |
| 15:26.11 | Katty | lunch. |
| 15:26.22 | leifmadsen | more like 34 mins |
| 15:26.28 | russellb | are we going out to lunch?! ^_^ |
| 15:26.30 | Katty | leifmadsen: you're not helping. |
| 15:26.36 | leifmadsen | Katty: I thought I was! |
| 15:26.57 | Katty | leifmadsen: :P |
| 15:27.39 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 15:28.04 | Katty | http://picoolio.co.uk/photos/medium/447-qk5bx.jpg <- oh that is NOT cool. |
| 15:28.45 | Zeeek | lunch it's almost dinner here |
| 15:29.07 | Katty | well you live in your own little world. |
| 15:29.10 | Katty | that's to be expected. |
| 15:29.16 | Zeeek | and more importantly, #voip-users-conference in 1/2 hour |
| 15:29.16 | jeff_phillips | I'm the only one in the office today so if I want lunch I guess I get to forward the incomming calls to my cell phone |
| 15:29.24 | Katty | Zeeek: LUNCH! in half an hour. |
| 15:29.34 | Katty | jeff_phillips: order in? (= |
| 15:29.38 | rue_mohr | see, tk also uses polycom phones, which also means, if Im right about the problem I'm having, he would have experianced volume problems with the diguim cards that caused echo on attempt of fix |
| 15:29.49 | rue_mohr | cause nobody ever changes the gain on the phone |
| 15:29.52 | Katty | also uses polycoms. |
| 15:29.58 | Katty | we have. |
| 15:30.00 | Katty | on occasion. |
| 15:30.04 | Zeeek | no gain without the pain |
| 15:30.05 | rue_mohr | do you use them with a digium card? |
| 15:30.09 | Katty | no. sangoma. |
| 15:30.13 | rue_mohr | aha |
| 15:30.15 | Zeeek | http://VUC.me see you there in 30 |
| 15:30.20 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 15:30.28 | rue_mohr | I had to dial the gain on the polycom up __11db__ for people to hear properly |
| 15:30.32 | rue_mohr | thats 4 bits |
| 15:30.50 | bijit | rue_mohr: the gain affects all phones...we had the same problem with aastra. |
| 15:31.01 | rue_mohr | which is why I think that the tdm800 has a 12bit adc and the ulaw codec expects 16 |
| 15:31.07 | Katty | eppigy: 24 :> |
| 15:31.08 | bijit | had to change default gain so it works good |
| 15:31.16 | Katty | eppigy: where are we goign?! |
| 15:31.19 | rue_mohr | bijit, digium card? |
| 15:31.35 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:31.35 | bijit | rue_mohr: both digium and sangoma |
| 15:31.48 | eppigy | Katty: I am going home :D |
| 15:31.59 | Katty | eppigy: WHAT?! |
| 15:32.11 | rue_mohr | bijit, please tell more |
| 15:32.45 | eppigy | I have a half day :D |
| 15:32.58 | Katty | :< |
| 15:33.00 | Katty | hates you. |
| 15:33.02 | eppigy | HALF DIZZLE |
| 15:33.04 | rue_mohr | fyi, the number "14844" that all reference pages use with dahdi_monitor to calibrate gains came from kb1 watching the signal on a pri |
| 15:33.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:33.13 | Katty | foschizzle! |
| 15:33.56 | rue_mohr | I'v got an opportunity with my T1 channelbank to do a test, I can loop back a channel with 0db and read astersisk against its own 1mw |
| 15:34.00 | bijit | we had to change gain to -10 |
| 15:34.07 | rue_mohr | just though of that on the way to work today |
| 15:34.16 | rue_mohr | bijit, for rx or tx? |
| 15:34.26 | bijit | tx |
| 15:34.28 | rue_mohr | I had to dial the tx down by almost that much |
| 15:34.31 | rue_mohr | OOOoooOOo... |
| 15:34.41 | rue_mohr | interesting |
| 15:35.05 | rue_mohr | I also, using the calibration pages, had to dial the rx to 1.9 |
| 15:36.10 | bijit | we only used tx and it worked fine |
| 15:36.21 | rue_mohr | did putting tx to that help with echo? |
| 15:36.48 | Katty | just upped outgoing audio |
| 15:37.03 | Katty | i just upped the default audio of the polycom phones |
| 15:37.08 | rue_mohr | I know there is about 11db loss somethere, I'v had to dial up the incomming gain by the same amount on the card and the phones, right now I do it on the phones |
| 15:37.09 | Katty | and left the server alone |
| 15:37.19 | bijit | rue_mohr: yes |
| 15:37.39 | rue_mohr | interesting |
| 15:37.43 | bijit | rue_mohr: our tx default was tx gain: 0 so we had to change it to tx gain: -10 |
| 15:38.15 | bijit | and had to set phones to Use Basic Codecs |
| 15:38.24 | rue_mohr | I just use ulaw |
| 15:38.44 | rue_mohr | http://www.pastebin.ca/1466426 <- there is the log of the things I'v tried |
| 15:39.10 | rue_mohr | -10 is crazy, BUT now I know its not me being crazy |
| 15:39.14 | bijit | rue_mohr: that is what we use also |
| 15:39.37 | rue_mohr | -9 is 1/8 |
| 15:39.43 | rue_mohr | k |
| 15:39.48 | rue_mohr | THANKYOU! |
| 15:40.07 | [TK]D-Fender | rue_mohr: Personal account : I had a customer in a similar boat. I swapped for an A200d and got to set 0/0 on them |
| 15:41.20 | Katty | this base coat smells like pineapple. |
| 15:41.36 | Katty | weird. |
| 15:41.48 | jaytee | if the next coat smells like coconut then you're in business |
| 15:42.00 | Katty | the color is called Polo Princess |
| 15:42.03 | Katty | and it's like a ...hmm. |
| 15:42.04 | bijit | lol |
| 15:42.08 | Katty | baby pink color |
| 15:42.18 | Katty | pink frosting color |
| 15:42.47 | Katty | http://neglelakkmani.files.wordpress.com/2009/04/orly-polo-princess4.jpg <- that color |
| 15:42.56 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:43.05 | Katty | i need a server case in that color. |
| 15:43.12 | [TK]D-Fender | Katty: O RLY? |
| 15:43.17 | Katty | [TK]D-Fender: yesrly |
| 15:47.39 | Katty | first coat is streaky and uneven :< |
| 15:49.01 | eppigy | :D |
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| 15:55.37 | *** join/#asterisk digitalirony (n=digitali@shellium/member/digitalirony) |
| 15:56.00 | Katty | it took 3 coats :< |
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| 15:58.25 | Katty | eppigy: BYE |
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| 16:18.15 | rue_mohr | ok, for an extra $50 I can go to a sangoma A200 or for $-220 I can go to the sangoma B600 both with echo cans |
| 16:18.52 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 16:19.18 | Katty | horay! (= |
| 16:19.42 | rue_mohr | the thought still scares me |
| 16:19.51 | [TK]D-Fender | rue_mohr: The B600d is really an awesome deal for 4 FXO. |
| 16:19.58 | rue_mohr | k |
| 16:20.07 | [TK]D-Fender | rue_mohr: And 1 FXS timing-linked for fax reliability |
| 16:20.26 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 16:20.27 | [TK]D-Fender | rue_mohr: makes the A200d really hard to suggest unless you need 8+ |
| 16:20.30 | rue_mohr | if I'm doing larger installs, the A200 is a good card to be familiar with |
| 16:20.36 | rue_mohr | yea |
| 16:20.55 | rue_mohr | I could use a pap2 for the other analog I need |
| 16:21.00 | Katty | quick! where am i going for lunch! |
| 16:21.04 | [TK]D-Fender | rue_mohr: Kinda like Polycom putting out phones that completely devalidate other models while considering both "current" |
| 16:21.06 | rue_mohr | mcdonalds |
| 16:21.12 | [TK]D-Fender | Katty: OUT! |
| 16:21.24 | Katty | mcdonalds? |
| 16:21.35 | Katty | ugah, hello breakout. |
| 16:21.43 | rue_mohr | hah |
| 16:21.50 | Katty | i might as well buy an acne wash while i'm out |
| 16:21.53 | rue_mohr | ok, so |
| 16:21.53 | Katty | as like a side. |
| 16:22.11 | rue_mohr | you really think I should chase after digium to gimme da money back |
| 16:22.20 | Katty | can i get a number one, with a bottled water and an acne wash. |
| 16:22.39 | Katty | that'll be 5.29, please pull around! |
| 16:22.55 | rue_mohr | I thiught it was $1 now |
| 16:22.59 | rue_mohr | across the board |
| 16:23.00 | [TK]D-Fender | rue_mohr: Draw a line somewhere. My point is you always seem unwilling to. |
| 16:23.24 | rue_mohr | yea I'm kinda an overly stubborn determined type |
| 16:23.26 | [TK]D-Fender | rue_mohr: If you say "one more month" well at least that means you aren't stuck on one path |
| 16:23.37 | [TK]D-Fender | rue_mohr: I prefer the term "masochistic" :p |
| 16:23.46 | rue_mohr | I'm waitng for a reply email from digium |
| 16:24.01 | Katty | when you could just call them |
| 16:24.04 | Katty | and have an answer immediately |
| 16:24.05 | rue_mohr | funny, your not the first person to say that in context of me AND this phonesystem |
| 16:24.36 | [TK]D-Fender | Katty: So far answer != solution |
| 16:24.48 | Katty | [TK]D-Fender: i meant about the refund. |
| 16:24.48 | [TK]D-Fender | Katty: Nor has it been for.... well ... months |
| 16:24.58 | [TK]D-Fender | Katty: that is another matter :) |
| 16:25.08 | [TK]D-Fender | ok, lunch time... out for Indian.... |
| 16:25.11 | Katty | you're another matter. |
| 16:25.14 | Katty | kbai |
| 16:25.42 | rue_mohr | well the tech I talked to last time downloaded me the svn drivers, which we couldn't install right there cause the office was using the phonesystem |
| 16:26.04 | rue_mohr | when I did install them, there was audio for outgoing calls, and no audio at all for incomming calls (co) |
| 16:26.16 | rue_mohr | (!??!?!?) |
| 16:26.24 | rue_mohr | I had to revert the drives back |
| 16:26.37 | Katty | weird |
| 16:26.42 | rue_mohr | totally |
| 16:26.58 | rue_mohr | I didn't even have a clue where to start on that |
| 16:27.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:28.07 | Katty | you know, mcdonalds does have some pretty nice salads. |
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| 16:28.22 | Katty | maybe i will go there (= |
| 16:28.23 | Katty | tata |
| 16:28.25 | rue_mohr | and our vendor just told me how to get ahold of their polycom support properly, so that might be solved |
| 16:33.10 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gw.teamgleim.com) |
| 16:35.17 | rue_mohr | I might have screwed this up, I dont know if I registered the tdm800 in time |
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| 16:43.54 | SuPrSluG | coppice: you around? i have a fax question |
| 16:44.45 | devyll | can I make a dialplan where the pbx will initiate multiple outgoing calls and every call to be added to a confrence (meetme) ? . Like, a conference which invites (initates the calls) others to join there. |
| 16:49.59 | thehar | approx how much is a cab per mile in huntsville? anyone know? |
| 16:55.09 | jplank | is digium support serious? |
| 16:55.34 | jplank | They guy on the phone was telling me things in DIRECT contradiction with the comments in asterisk source |
| 16:56.35 | mog | whats the prob jplank |
| 16:57.20 | jplank | I'm having echo on a 2400p with hw echo can ever since I (was told) to up grade to 1.4.25.1 and Dahdi 2.2.0RC5 to get rid of a half duplex issue |
| 16:57.46 | jplank | now the guy is telling me to set echocan=256, turn echo training on, and use fxotune |
| 16:58.03 | jplank | isn't that all irrelevant with a HW echo can? |
| 16:58.21 | jplank | he also wants me to install HPEC |
| 16:58.50 | malcolmd | fxotune isn't irrelevant because it modifies registers on the modules themselves. the value set in echocan for number of taps is irrelevant, as is echo training |
| 16:59.06 | jplank | so fxotune will work with a HW echo can? |
| 17:02.10 | malcolmd | the only thing it has to do w/ the hw echo can is that its use modifies the hw level gains of the daa in order to minimize any mismatch against your lines. so, running it will have some impact, whether you're using a hardware-based ec or a software-based one |
| 17:02.27 | malcolmd | the hw level gains as well as some other stuffs... |
| 17:04.56 | malcolmd | what support's trying to do by suggesting hpec is trying to characterize whether or not your issue is specific to the version of the echo can code that runs in hardware on the vpm vs. the echo can that runs in software in hpec. they're from the same vendor, but produced in one form for the dsp on the vpm and the other in c for x86 platforms. so, if support has to kick it back up to the dsp vendor they can characterize it as a fault i |
| 17:04.56 | malcolmd | n one, the other, or both |
| 17:05.02 | *** join/#asterisk techie33 (n=techie33@c-67-181-236-185.hsd1.ca.comcast.net) |
| 17:05.17 | techie33 | anyone available for a codec question? |
| 17:07.00 | jplank | I dont mind trying HPEC, but from what I understand since this is a 2400p, I can't get it on all the channels |
| 17:07.10 | jplank | just about 1 module or something like that |
| 17:07.19 | malcolmd | they'll give you a key for 24 channels |
| 17:07.56 | malcolmd | you can run it on all of the channels at 128ms (1024 taps), but it's going to crush your cpu to do so. the lower the taps count the less it's going to annihilate the processor |
| 17:08.10 | jplank | never have that many calls up |
| 17:08.32 | jplank | never seen more then 5, and only 1 or 2 tops being a dahdi channel |
| 17:09.44 | malcolmd | 24 channels @ 128ms will burn an entire core of a xeon 5160. there are certainly heftier processors out there now, but the software echo cancellers run on a single core, so you can't take advantage of multi-cores to spread the load |
| 17:10.22 | jplank | is there any way I can pull the serial number of the card from the CLI? I don't see it in dmidecode |
| 17:10.57 | malcolmd | negative. if you know who you bought it from they may be able to give it to you though |
| 17:11.12 | jplank | grrr, I've purchased so many of these |
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| 17:16.16 | mechbangirc | hi any priority in dialplan to jump to when "DIALSTATUS" is "CONGESTION", like n+101 or something |
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| 17:17.40 | tzafrir_laptop | 1024 taps? sounds like a gross overkill |
| 17:22.04 | techie33 | I'm trying to get calls to be send from our cisco router to our asterisk box in g729 format when the call comes in in fails. However if I make an outbound call I see the calls go out as g729 without an issue. Why would this be? |
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| 17:24.11 | *** join/#asterisk ariel_ (n=ariel_@63.214.236.169) |
| 17:24.20 | ariel_ | hello folks |
| 17:24.49 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:25.51 | timeshell_atwork | Happy Fivesday! |
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| 17:26.59 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 17:27.42 | ariel_ | anyone have any good instructions to upgrade from Zap drivers to Dahdi when your also using libpri ? |
| 17:28.17 | mechbangirc | i am trying to implement LCR in portech mv-378, totally lost. any idea how to deal with sip responses??? |
| 17:28.31 | mechbangirc | in dialplan |
| 17:31.39 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 17:32.47 | *** join/#asterisk jeff_phillips (n=ceramics@209-142-149-133.stat.centurytel.net) |
| 17:33.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.22 | *** join/#asterisk xheliox (n=jeff@178.199.8.67.cfl.res.rr.com) |
| 17:33.36 | jeff_phillips | Okay, MixMonitor() doesn't seem to work so I'm using Monitor() instead. Problem is, I need to record until the call has been hung up and THEN launch another command. MixMonitor() has an option to specify another command to launch upon termination, which worked except that it only recorded silence. |
| 17:33.46 | jeff_phillips | So how can I run something else at the end of the call with Monitor()? |
| 17:34.58 | *** part/#asterisk mechbangirc (n=mech@mbl-65-148-115.dsl.net.pk) |
| 17:35.18 | tzafrir_laptop | ariel_, hmm.. how about UPDATE.txt in the dahdi-tools directory? |
| 17:35.33 | tzafrir_laptop | libpri is irrelevant to zaptel vs. dahdi |
| 17:36.23 | jeff_phillips | In other words, how can I insert some code that will launch upon the channel hanging up? |
| 17:37.12 | ariel_ | tzafrir_laptop, read that file but it does not really talk about libpri, I am upgrading a server that is already running on zaptel, with libpri installed. |
| 17:37.41 | ariel_ | tzafrir_laptop, It says to setup libpri first which I did, then dahdi |
| 17:37.52 | ariel_ | but it seems to have ignored the libpri |
| 17:38.41 | tzafrir_laptop | if you want to build chan_dahdi (of asterisk) with pri support, you need to have libpri installed |
| 17:38.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:38.57 | ariel_ | yes which I did |
| 17:39.28 | tzafrir_laptop | So all's well |
| 17:40.11 | *** join/#asterisk Docteh (n=Kyle@allspark.shadowmage.org) |
| 17:40.48 | ariel_ | except my pri spans are not working |
| 17:50.20 | tzafrir_laptop | do you see the PRI channels in 'dahdi show channels'? the spans in 'pri show spans'? |
| 17:50.51 | ariel_ | argh can't believe it, I found that if you follow the directions the dahdi_conf mess's up your settings. |
| 17:51.06 | ariel_ | tzafrir_laptop, I got it working, |
| 17:51.11 | ariel_ | t/y |
| 17:52.04 | tzafrir_laptop | ariel_, which directions? which dahdi_conf? what needs fixing? |
| 17:53.14 | ariel_ | it did not pickup correctly the E1's |
| 17:53.16 | *** join/#asterisk suma (n=root@64.92.218.196) |
| 17:53.31 | suma | What is the best answering machine detection solution for asterisk ? |
| 17:53.37 | ariel_ | it added crc4 to the mix and setup echocancel=mg2 when it had hardware ec |
| 17:53.48 | Katty | uhoh |
| 17:53.50 | Katty | i'm gettin tired :< |
| 17:55.39 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 17:55.56 | jeff_phillips | if MixMonitor() replaces Monitor() then how come they are so vastly different? |
| 17:56.43 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 17:57.10 | suma | jeff_phillips: Monitor generates two files |
| 17:57.19 | suma | Mix Monitor generate a single for a call |
| 17:57.19 | jeff_phillips | i get that |
| 17:57.27 | jeff_phillips | but MixMonitor doesn't seem to work at all |
| 17:57.46 | jeff_phillips | but has the nice bonus of being able to run a command after it has finished recording |
| 17:57.53 | suma | Works fine for us. |
| 17:57.55 | suma | yes |
| 17:58.12 | jeff_phillips | I can't get MixMonitor to record anything but silence, I can record fine with Monitor() but it doesn't allow me to run a command when it's done |
| 17:58.25 | suma | jeff_phillips: you should be able to do that via management interface as of now |
| 17:58.50 | suma | jeff_phillips: what is your asterisk version ? |
| 17:58.58 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:59.13 | jeff_phillips | Well what I'm trying to do is simply create an extension that when called will beep and record whatever you say until you hang up. Then I want to run a command after the call has disconnected |
| 17:59.36 | xheliox | jeff_phillips: Use the m option for Monitor() |
| 17:59.53 | xheliox | show application Monitor |
| 18:00.44 | jeff_phillips | suma: 1.4.20-1 |
| 18:01.17 | jeff_phillips | I only need one leg of the call |
| 18:01.28 | jeff_phillips | but I need to launch a different command upon hang-up |
| 18:01.41 | suma | jeff_phillips: You can do that with the vartiable MONITOR_EXEC |
| 18:01.53 | suma | read the application description |
| 18:02.55 | jeff_phillips | ohhhh |
| 18:03.03 | jeff_phillips | I'll try that |
| 18:03.11 | jeff_phillips | .. still don't understand why mixmonitor didn't work |
| 18:04.47 | xheliox | Me either. But that should be a work around. |
| 18:05.23 | jeff_phillips | ok |
| 18:06.04 | jeff_phillips | when I tried MixMonitor, with WAV as the format, it would create a wav file with about 5 seconds of silence and if I selected MP3 it would not create a file at all. Nothing was ever recorded either way |
| 18:06.07 | ruben23 | hi how do i remove this error log on my asterisk CLI http://pastebin.com/mfc32d90 |
| 18:06.47 | rue_mohr | sounds like your phone is asking for comfort noise |
| 18:07.16 | ruben23 | rue_mohr: how do i correct that..? |
| 18:07.19 | rue_mohr | what kinda phone you using? |
| 18:07.25 | jeff_phillips | comfort noise? |
| 18:07.41 | rue_mohr | its background static to give you the impression the other side didn't hang up |
| 18:07.45 | ruben23 | im using eyebeam softphones |
| 18:08.00 | rue_mohr | well, look for a comfort noise setting |
| 18:08.08 | rue_mohr | turn it to whatever it isn't |
| 18:08.59 | rue_mohr | (when you have a problem, try things that are different then when you had the problem) |
| 18:09.16 | rue_mohr | esp in abscence of proper support |
| 18:09.35 | rue_mohr | if you paid for that program I'd expect you can call them up and complain |
| 18:09.45 | jeff_phillips | oh yeah |
| 18:10.31 | xheliox | rue_mohr: And I expect you'd find out what sort of support a company has before paying for their software. |
| 18:10.53 | jeff_phillips | there were a few episodes of murder she wrote where between each word/phrase the audio track was made to go completely silent. couldn't even watch it because it was so annoying to hear the trailing hiss and then a sharp drop to silence after every bit of dialog |
| 18:12.31 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 18:12.40 | rue_mohr | so to work out tx level setting I'd need to loop an fxo to an fxs, call it with a 1mw and adjust |
| 18:13.10 | rue_mohr | right now, if I do that through the co, I get incredible loss, over 15dbm of gain dosn't come close to fixing it |
| 18:14.33 | *** join/#asterisk propellerhead (n=yogurt2u@host107.190-136-119.telecom.net.ar) |
| 18:15.20 | rue_mohr | xheliox, heh, thats halarous in context of microsoft |
| 18:16.10 | jeff_phillips | woot it works |
| 18:16.33 | rue_mohr | you got your channels to play murder she wrote? |
| 18:17.56 | rue_mohr | anyone know how I can get the serial number of the tdm card from the console? |
| 18:18.22 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:19.18 | [T]ank | Im looking for something similar to snmp monitoring of things like sip registry and sip peer connections, etc. Does anything like that exist? |
| 18:19.23 | rue_mohr | how can I measure the time of the polarity reversal from the co after a calls hing up |
| 18:19.47 | rue_mohr | [T]ank, hmm |
| 18:20.06 | [T]ank | I see things like argus and hobbit. But I am not sure they will really do what I am after. |
| 18:20.07 | rue_mohr | you want events? |
| 18:20.18 | rue_mohr | or you polling? |
| 18:20.22 | [T]ank | I think that is more to make sure that the safe_asterisk processes are running. |
| 18:20.39 | rue_mohr | ah YOU also need a heartbeat |
| 18:20.42 | [T]ank | I simply want to see if sip peers are registered, or unknown for example |
| 18:20.51 | rue_mohr | your the 3rd person to ask for it latley, |
| 18:20.55 | rue_mohr | I'z the first |
| 18:20.57 | rue_mohr | :) |
| 18:21.12 | [T]ank | if peers go "UNKNOWN" I would want a notification |
| 18:21.30 | rue_mohr | aside form a log file processor I dont know how you might do that |
| 18:21.37 | rue_mohr | well C coding aside |
| 18:22.10 | rue_mohr | it would be good for asterisk to have an snmp module, create heartbeat events and stuff |
| 18:22.42 | rue_mohr | I need to know if asterisk goes down so i can flip a relay and switch lines over to backup phones |
| 18:22.48 | *** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 18:22.58 | [T]ank | yours you should be able to do... |
| 18:23.02 | [T]ank | I do that using hobbit |
| 18:23.14 | rue_mohr | maybe I should look up hobbit |
| 18:23.25 | [T]ank | sec... |
| 18:23.38 | [T]ank | http://www.voip-info.org/wiki/view/Asterisk+monitoring |
| 18:23.41 | rue_mohr | is wating for a call back from digium tech support re his bad audio levels and echo problems |
| 18:23.48 | [T]ank | check those links out. There are more than just hobbit |
| 18:25.21 | Joel | rue_mohr there are devices which will automatically flip lines for you |
| 18:25.35 | rue_mohr | yes relays, but triggering the relay is the trick |
| 18:25.53 | Joel | rue_mohr did you see the word "automatically" in my sentence? |
| 18:25.55 | rue_mohr | the alarm signal on the t1 is messed up, idiot engineer |
| 18:26.53 | rue_mohr | when the channelbank is powered up, it signals ok half way thru the boot it says alarm, after booting it says ok again |
| 18:27.13 | rue_mohr | you get ok if the channelbank has no power at all, or is ok |
| 18:27.20 | rue_mohr | :/ |
| 18:27.41 | rue_mohr | but thats another site |
| 18:28.06 | rue_mohr | I think my plywood paint is dry, I'm gonna go screw equipment to it |
| 18:29.18 | Katty | there are 4 people in our office |
| 18:29.19 | Katty | out of 30 |
| 18:29.24 | Katty | maybe i should go home |
| 18:29.58 | rue_mohr | you want to help me dig thru asterisk source? |
| 18:30.11 | beek | Katty: Only 4/30 people? That, to me, sounds like an excellent situation. |
| 18:33.06 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:33.32 | rue_mohr | playtones, are the paramiters freq and amplitiude? |
| 18:34.10 | rue_mohr | man it would really help to be able to monitor levels on the ulaw stream |
| 18:34.15 | ariel_ | argh my system is still not able to detect fax's, |
| 18:34.31 | rue_mohr | do this |
| 18:34.52 | rue_mohr | dial yourself on the line, and hit dtmf keys, see if you get the dtmf properly |
| 18:35.09 | rue_mohr | if its all messed up, might indicate where you have problems |
| 18:35.19 | rue_mohr | ariel_, digium card? |
| 18:35.29 | rue_mohr | analog? digital? |
| 18:35.33 | ariel_ | yes TE220b |
| 18:35.47 | rue_mohr | thats a 2 channel T1 yes? |
| 18:36.07 | rue_mohr | er E1? |
| 18:36.08 | ariel_ | yes, setup as 2 span E1, euruISDN |
| 18:36.14 | rue_mohr | ah |
| 18:36.25 | rue_mohr | your using in it europe, right? |
| 18:36.27 | ariel_ | faxdetech=both |
| 18:36.31 | rue_mohr | yaya |
| 18:36.34 | ariel_ | rue_mohr, no |
| 18:36.37 | rue_mohr | oh |
| 18:36.45 | ariel_ | out in the open seas.... |
| 18:37.01 | rue_mohr | what are you using it with? eupoean equipt? |
| 18:37.36 | ariel_ | depends, but this case it's Cisco 3745 to E1's on Asterisk to E1's on a Mitel pbx |
| 18:37.55 | rue_mohr | ok, so your all matched up in that sense |
| 18:38.02 | ariel_ | yes |
| 18:38.19 | rue_mohr | hmm |
| 18:38.35 | rue_mohr | who does the clocking? |
| 18:38.41 | ariel_ | our setup was working with asterisk 1.2 and 1.09, We have upgraded one to 1.4.25 with Dahdi and pri and it's not detecting the faxes |
| 18:38.55 | rue_mohr | ah |
| 18:39.00 | rue_mohr | I think they have a gain problem |
| 18:39.03 | ariel_ | we do the clocking but I have tried it with the cisco and with the mitel |
| 18:39.10 | rue_mohr | I been bashing my head agaisnt it for days now |
| 18:39.27 | rue_mohr | my suggestion is to not let asterisk do the clocking |
| 18:39.32 | ariel_ | faxdetect ? |
| 18:39.39 | rue_mohr | let the mitel do the E1 clocking |
| 18:39.52 | ariel_ | well right now I have it in our lab |
| 18:40.17 | ariel_ | connected to a Cisco 3645 |
| 18:40.19 | rue_mohr | I'm betting its a gain problem |
| 18:40.27 | ariel_ | E1 which is giving me the clocking |
| 18:40.31 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:40.46 | rue_mohr | yea, but hte clock source shoudl be the mitel, as in, its the master |
| 18:40.55 | rue_mohr | the card is the slave |
| 18:40.57 | Blackvel | what can I do that ChanIsAvail does not give me back 0 AST_DEVICE_UNKNOWN as ${AVAILSTATUS}. I have to check for BUSY / INUSE |
| 18:41.05 | ariel_ | In this case it's not, it's the Cisco |
| 18:41.16 | rue_mohr | ok, that shoudl be ok |
| 18:41.27 | rue_mohr | Blackvel, dunno |
| 18:41.58 | rue_mohr | ariel_, do you have any means of loss testing? |
| 18:42.20 | ariel_ | Blackvel, what is it giving you? have you tried it with core set debug 99 |
| 18:42.27 | ariel_ | it displays allot of info that way. |
| 18:42.28 | rue_mohr | as in, if you had 11db loss somewhere in asterisk, could you tell? |
| 18:43.10 | ariel_ | with PRI E1 we normally don't have to play with the gains at all. |
| 18:43.19 | ariel_ | sound is great for normal calls |
| 18:43.21 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 18:43.27 | rue_mohr | I know |
| 18:43.29 | rue_mohr | but |
| 18:43.36 | rue_mohr | could you tell |
| 18:43.36 | ariel_ | And it's able to go through our ivr and also pin code aceptance |
| 18:44.06 | rue_mohr | yea, my system was ont he virdge or working, some things did, some things didn't, and alot of stuff was intermittent |
| 18:44.50 | Blackvel | ariel: trying to check if I am talking already on the phone. need to use multi-dial. if phone is busy I need to skip the multi-dial |
| 18:45.22 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 18:45.37 | Blackvel | yes i know, wiki says that this is not possible. should be using devstate. but [TK]D-Fender want me to do it that way (i am on 1.2 anyways and cant use devstate). |
| 18:46.59 | Blackvel | just need to test .... |
| 18:47.18 | ariel_ | so your trying to call the phone but it's not telling you it's busy or any other state? What does the phone do? |
| 18:48.14 | Blackvel | not yet |
| 18:48.21 | Blackvel | trying to get status BEFORE the dial |
| 18:48.26 | rue_mohr | so milliwatt isn't neccissarily 1mw, the level isn't specifically set |
| 18:49.10 | Blackvel | if its busy I won't dial / use the multi-dial DIAL(SIP/tech1&SIP/tech2). |
| 18:49.11 | ariel_ | Blackvel for some our queues and agents we use a macro that goes out to the mysql and checks to see if there busy |
| 18:49.27 | ariel_ | we post to the mysql when they make or get a call |
| 18:49.38 | Blackvel | the multi-dial doesn't get me busy state and let's ring tech2, even I am talking on tech1. so I need to check tech1 channel on BUSY/INUSE first |
| 18:49.58 | Blackvel | oh interesting... |
| 18:50.12 | [TK]D-Fender | wasteful overkill. |
| 18:50.16 | Blackvel | so you do it manually for incoming and outgoing calls |
| 18:50.31 | [TK]D-Fender | this is about as bad as taht GROUP() idea |
| 18:50.39 | Blackvel | hehe |
| 18:50.42 | [TK]D-Fender | Actually WORSE |
| 18:51.05 | Blackvel | I believe asterisk has to get that on-the-fly (internally) |
| 18:51.59 | Blackvel | checked sip.conf peer |
| 18:52.06 | Blackvel | it is host=dynamic, qualify=yes |
| 18:52.10 | rue_mohr | asterisk supports midi!? |
| 18:52.33 | suma | What is the best answering machine detection solution for asterisk ? |
| 18:52.44 | rue_mohr | amd ? |
| 18:52.48 | rue_mohr | aka amd.conf? |
| 18:52.53 | Blackvel | any more options to set to go for some different result as ChanIsAvail: 0 AST_DEVICE_UNKNOWN |
| 18:52.57 | rue_mohr | aka /etc/asterisk/amd.conf? |
| 18:54.05 | suma | Are the commercial ones detects more thant what is with amd ? |
| 18:54.43 | suma | I was in an assumption what asterisk has is a basic one |
| 18:55.49 | Blackvel | oh |
| 18:55.50 | Blackvel | found posting |
| 18:55.54 | Blackvel | http://voipusers.org.nz/pipermail/users/2008-January/003492.html |
| 18:56.39 | Blackvel | Andrew says ChanIsAvail supports [TK]D-Fender idea in >=1.4. looks like it does not do in V1.2 what it is supposed to do |
| 18:58.01 | [TK]D-Fender | Blackvel: .... stop "looking" and get off your ass. You've been handed this and you aren't DOING IT |
| 18:58.17 | [TK]D-Fender | Blackvel: Stop waiting for 100 different accounts of success before getting off your ass. |
| 18:58.20 | Katty | hmm. |
| 18:58.47 | Blackvel | ??? |
| 18:58.56 | [TK]D-Fender | Blackvel: And you will find all sort of people who can't get things to work. They are often twits who can't follow directions. |
| 18:58.59 | Blackvel | it is NOT working on V1.2 [TK]D-Fender |
| 18:59.03 | Blackvel | 0 AST_DEVICE_UNKNOWN |
| 18:59.06 | [TK]D-Fender | And we never see THEIR kind in here, do we? |
| 18:59.16 | Blackvel | result is always: 0 AST_DEVICE_UNKNOWN instead of BUSY/INUSE |
| 18:59.23 | [TK]D-Fender | Blackvel: ... |
| 18:59.36 | [TK]D-Fender | ~wmmfpb |
| 18:59.36 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 18:59.41 | [TK]D-Fender | :D |
| 18:59.56 | techie33 | [TK]Defender: I have a question about g729 codec, do you know much about this? I'm trying to get incoming calls to use the g729 codec but the calls fail. They all want to come in ulaw, however I can make calls outbound from the 7960 handsets and they work fine. I've forced the g729 code on the router, but it still wants to pass ulaw. Any ideas? |
| 19:00.05 | [TK]D-Fender | Blackvel: Do not get it in your head for even 1 second that I trust you did this properly |
| 19:00.59 | Blackvel | yeah |
| 19:00.59 | Blackvel | I feel that |
| 19:01.08 | batphone | lol |
| 19:01.29 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 19:01.44 | [TK]D-Fender | techie33: Learn for Blackvel 's example... |
| 19:01.45 | batphone | whenever i am having a bad day, all i have to do is look in here or in #cisco to find someone having a worse time |
| 19:03.19 | ariel_ | I am hungry,,, humm let me go see what the Vending Machines have for snack's..... |
| 19:03.45 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) |
| 19:05.38 | Blackvel | http://pastebin.com/d3cd0ce1e |
| 19:05.54 | hardwire | leifmadsen: poke.. played with clusterip? |
| 19:05.57 | Blackvel | maybe some more config stuff needs to be improved to get it working |
| 19:06.16 | leifmadsen | hardwire: never heard of it |
| 19:07.18 | hardwire | leifmadsen: based on a source ip, source ip + source port, or source ip + port + dest port you can tell a group of machines to share an IP and only answer based on the result of a hash |
| 19:07.20 | Blackvel | techie33: he has no good day with me as I asked too many questions the last days ;) |
| 19:07.21 | leifmadsen | bad day? try importing several megs of data into mysql cluster -- node go boom |
| 19:07.36 | leifmadsen | hardwire: I have no use for that :) |
| 19:07.40 | [TK]D-Fender | [10:19]<[TK]D-Fender>Blackvel: CHANISAVAIL <- It works. Read the instructions and go DO IT |
| 19:07.40 | hardwire | so I have 4 asterisk servers sharing an IP. one can take over for another |
| 19:07.51 | [TK]D-Fender | [09:46]<[TK]D-Fender>Blackvel: "core show application chanisavail" <--- |
| 19:07.51 | hardwire | leifmadsen: you're hip to clusters.. thought I'd ask. |
| 19:07.54 | Blackvel | techie33: therefore he didn't believe I was actually going for it |
| 19:08.03 | leifmadsen | hardwire: no sure what the question was :) |
| 19:08.10 | hardwire | leifmadsen: no question.. just talk. |
| 19:08.12 | Blackvel | I can really understand him (a bit) |
| 19:08.13 | leifmadsen | gotcha |
| 19:08.17 | [TK]D-Fender | Blackvel: almost 5 HOURS later you seem incapable of reading its instructions |
| 19:08.36 | [TK]D-Fender | Blackvel: And if that isn't clear enough : THAT APP HAS OPTIONS. READ THEM |
| 19:09.10 | hardwire | is sharing the reg contexts between a cluster of servers, using cluster ip, and it seems to work very well. |
| 19:09.27 | jeff_phillips | "seems" |
| 19:09.29 | hardwire | at least for transparency to the end user. |
| 19:09.29 | Blackvel | s flag? |
| 19:09.36 | hardwire | jeff_phillips: :) |
| 19:09.41 | [TK]D-Fender | Blackvel: What does it SAY? |
| 19:09.56 | Blackvel | s - Consider the channel unavailable if the channel is in use at all |
| 19:10.21 | hardwire | leifmadsen: you're mysql cluster go boom? |
| 19:10.28 | hardwire | or somebody elses.. I just joined.. sucks if it happened. |
| 19:11.02 | hardwire | speaking of which.. cluster ip and mysql-ndb go hand in hand :P |
| 19:11.04 | hardwire | -> worky worky. |
| 19:12.40 | [TK]D-Fender | Blackvel: A phone isn't "busy" unless it CANNOT take another call. Yuo are working with multi-line phones! Thats what call-waiting is for! |
| 19:13.01 | [TK]D-Fender | Blackvel: so for most purposes, "busy" = doesn't exist |
| 19:14.42 | hardwire | I believe I've only encountered it on some SIP phones when DND is on |
| 19:14.55 | [TK]D-Fender | hardwire: Yeah, some return that status. |
| 19:15.39 | *** join/#asterisk DSpair (n=dphillip@74-130-11-247.dhcp.insightbb.com) |
| 19:15.56 | DSpair | Good afternoon all (or whatever it is where you are!). |
| 19:17.16 | *** join/#asterisk hugorebelo (n=hugo@200-171-132-124.completo.com.br) |
| 19:17.39 | leifmadsen | hardwire: ya... on certain tables |
| 19:18.17 | hardwire | leifmadsen: it's difficult to stop split brains and keep all tables uncorrupted in a cluster. |
| 19:18.26 | hardwire | I don't envy you. |
| 19:18.38 | hardwire | I use master-master and put each master on a beefy serv when I can. |
| 19:18.42 | leifmadsen | hardwire: well... it's just an initial import -- only fails with certain tables |
| 19:18.51 | leifmadsen | brand new servers -- quite beefy |
| 19:18.52 | hardwire | then use pen locally to distributed the load (pen is awesome) |
| 19:19.10 | leifmadsen | I haven't gotten to the distribution part yet |
| 19:19.17 | Blackvel | [TK]D-Fender CW is turned off |
| 19:19.17 | leifmadsen | I'm just leaving out the big tables for now |
| 19:19.23 | hardwire | leifmadsen: ndb or master/master or master/slave ? |
| 19:19.28 | leifmadsen | ndb |
| 19:19.34 | hardwire | 5.1+ mysql? |
| 19:19.41 | [TK]D-Fender | Blackvel: Either way, go call it properly |
| 19:19.43 | leifmadsen | 5.1.32 + 6.3.24 I think |
| 19:19.57 | hardwire | ah.. I haven't gotten to play with ndb that recent. |
| 19:20.11 | hardwire | I can't wait to try it/break it. |
| 19:20.19 | hardwire | but I have other priorities :) |
| 19:20.20 | leifmadsen | pretty good so far other than it kills the ndb daemon on my non-master data node |
| 19:20.32 | leifmadsen | (when I import certain large tables) |
| 19:20.44 | hardwire | if it's only 2 servers I'd of course recommend master/master and using pen to handle failover. |
| 19:20.58 | leifmadsen | well, it's 3 mysqld's at the moment |
| 19:21.19 | hardwire | I never understood the need for a single server to run ndb_mgmr |
| 19:21.20 | leifmadsen | got a tutorial? |
| 19:21.22 | hardwire | mgmd. |
| 19:21.48 | leifmadsen | actually good turned up lots of info |
| 19:22.16 | hardwire | leifmadsen: install mysql, set up initial privileges, set up a replication account on both machines, set the server id's, set the replication params in mysql.conf, start slave on both. :) There's lots of tutorials for master/master off of google. |
| 19:22.21 | hardwire | it's quick to set up |
| 19:22.29 | leifmadsen | hmmmm |
| 19:22.33 | leifmadsen | what does pen do then? |
| 19:22.34 | hardwire | then using pen is easy.. it's a generic tcp load balancer/failover |
| 19:22.36 | leifmadsen | if it's master/master? |
| 19:22.46 | hardwire | pen -f 3306 server1 server2 |
| 19:22.57 | leifmadsen | virtual IP? |
| 19:23.00 | hardwire | that will foreground pen for testing and use either of the servers.. depending on tcp state. |
| 19:23.02 | hardwire | no virtual IP |
| 19:23.08 | hardwire | just run pen on the client systems |
| 19:23.16 | hardwire | then connect to it's local IP vs the servers. |
| 19:23.19 | leifmadsen | what do scripts connect to then? |
| 19:23.21 | leifmadsen | oh |
| 19:23.22 | hardwire | pen gives good info about failovers. |
| 19:23.27 | leifmadsen | okie |
| 19:23.37 | hardwire | you can even make it round robin between each server for every new connection. |
| 19:23.52 | leifmadsen | link to pen? |
| 19:23.53 | hardwire | I use it for mysql, and some other socket services. |
| 19:24.01 | hardwire | leifmadsen: apt-get install pen is all I got. |
| 19:24.10 | leifmadsen | <-- yum |
| 19:24.13 | hardwire | http://siag.nu/pen/ |
| 19:24.15 | Blackvel | wow...tested it out |
| 19:24.19 | leifmadsen | hardwire: I'll google |
| 19:24.20 | hardwire | it's based off of the "balance" program. |
| 19:24.24 | hardwire | which is very similar. |
| 19:24.26 | Blackvel | it seems now really working |
| 19:24.31 | hardwire | pen however can dump out HTML details on load distribution |
| 19:24.38 | hardwire | balance on the other hand can fail over between server groups. |
| 19:24.48 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:25.17 | hardwire | anyhoot.. tootles. thought I'd put a bug in your ear about it.. if you wanna dick around with clusterip at some point let me know.. I have the goods. |
| 19:25.26 | hardwire | I never knew 802.3 had a multicast mac specification. |
| 19:25.28 | hardwire | but it does. |
| 19:25.41 | leifmadsen | hardwire: coolio -- if I ever find some free time... (not gonna happen) |
| 19:25.45 | leifmadsen | maybe at astricon |
| 19:25.46 | Blackvel | i dont care about status 0-5 and AVAILSTATUS. it's okay that ChanIsAvail routes into the proper direction |
| 19:26.08 | hardwire | leifmadsen: word. |
| 19:26.14 | Blackvel | i still can't believe that :) need to try with snom m3 immediately |
| 19:28.06 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:28.16 | hardwire | Blackvel: send me your m3 when you get bored with it. |
| 19:28.18 | hardwire | :P |
| 19:32.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:33.03 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) [NETSPLIT VICTIM] |
| 19:33.16 | [TK]D-Fender | NUT-SPLIT! |
| 19:34.59 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gw.teamgleim.com) [NETSPLIT VICTIM] |
| 19:34.59 | *** join/#asterisk dajhorn (n=chatzill@206.16.96.160) [NETSPLIT VICTIM] |
| 19:34.59 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-079-217.pools.arcor-ip.net) [NETSPLIT VICTIM] |
| 19:34.59 | *** join/#asterisk jtexter3 (n=jtexter3@72.242.229.213) |
| 19:34.59 | *** join/#asterisk jad_jay (n=chatzill@public.axolys.fr) [NETSPLIT VICTIM] |
| 19:34.59 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 19:34.59 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 19:34.59 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM] |
| 19:35.00 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:35.00 | *** join/#asterisk GlobeTrotter (n=GlobeTro@201.218.90.155) [NETSPLIT VICTIM] |
| 19:35.00 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
| 19:35.00 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 19:36.37 | Blackvel | hardwire: why? are you going for one? on xing voip forum ppl discussed that m3 is not the way to go |
| 19:36.45 | Blackvel | but well....i am not talking too much on it |
| 19:36.49 | Blackvel | its ok for me |
| 19:38.05 | *** join/#asterisk dacs (n=piper69@unaffiliated/dacs) |
| 19:38.10 | dacs | Howdy folks |
| 19:38.11 | hardwire | ok |
| 19:40.29 | dacs | i figure out for fully to benefit from the book and * i will need to get a real SIP account instead of free one. what do you guys recommended |
| 19:42.39 | jeff_phillips | dacs: depends on what you want to do with it |
| 19:43.08 | jeff_phillips | I use DIDforSale.com for my inbound and gafachi.com for outbound. |
| 19:43.28 | dacs | jeff_phillips: just to make and receive calls here in cali |
| 19:43.47 | Blackvel | hardwire: do you guys use any better dect voip phone than snom m3? siemens gigaset? |
| 19:44.19 | hardwire | I don't have squat. |
| 19:44.35 | dacs | jeff_phillips: i have a free SIP inbound with my area code |
| 19:44.54 | dajhorn | dacs: The bot that watches this channel keeps a list of recommended providers. (Who is the bot?) |
| 19:45.18 | BlargMaN00 | ~itsp |
| 19:45.18 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 19:45.29 | BlargMaN00 | ~itsplist-us |
| 19:45.30 | infobot | from memory, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:46.18 | ariel_ | wow what a list. |
| 19:46.25 | dacs | ~itsplist-us |
| 19:46.25 | infobot | rumour has it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:50.58 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 19:51.40 | Docteh | ~itsplist-ca |
| 19:51.40 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 19:51.53 | Docteh | talk about a shortlist |
| 19:52.10 | Docteh | ~itsplist-su |
| 19:53.09 | rue_mohr | twinkle|wifi? |
| 19:53.12 | dajhorn | I'll plug Unlimitel (which is on the list) and voip.ms (which isn't). |
| 19:53.40 | dajhorn | Both are based out of somewhere near Ottawa. |
| 19:53.52 | rue_mohr | east, bad |
| 19:53.55 | rue_mohr | west good! |
| 19:53.59 | Docteh | ah, I'm using les since I'm west coast |
| 19:54.27 | rue_mohr | telless? |
| 19:54.47 | Docteh | les.net |
| 19:54.56 | rue_mohr | Telless - "Naaa, You don't need a phone..." |
| 19:55.10 | Docteh | dissing telus or...? |
| 19:55.15 | jplank | anyone ever work with a two-way overhead paging system? |
| 19:55.16 | rue_mohr | totally |
| 19:55.26 | rue_mohr | two way? |
| 19:55.27 | [TK]D-Fender | [15:40]<dacs>i figure out for fully to benefit from the book and * i will need to get a real SIP account instead of free one. what do you guys recommended <- price doesn't mean anything |
| 19:55.44 | jplank | yea |
| 19:55.48 | jplank | talk back I guess yiou could call it |
| 19:55.50 | [TK]D-Fender | dacs: What you get out of * depends on what you want to do, and how you choose to do it. |
| 19:55.53 | jplank | fully duplex |
| 19:55.58 | jplank | full* |
| 19:56.03 | jplank | speak into the speaker ;) |
| 19:56.18 | rue_mohr | for $30/mo I'll give you an account with a number on the westcoast |
| 19:56.38 | rue_mohr | you just have to put up with a little CBC1 in the background |
| 19:56.42 | jplank | I'll do it for $29 |
| 19:56.51 | rue_mohr | $28.50 |
| 19:56.52 | [TK]D-Fender | dacs: And the book is there to give a brief overview of most of * and telecom in general |
| 19:57.04 | *** join/#asterisk SomethingIsodd (n=dan@72.172.174.69) |
| 19:57.12 | SomethingIsodd | HEllo all anyone here use freeradius |
| 19:57.13 | Docteh | my apartments intercom gives me CBC1 stronger than the people calling up |
| 19:57.26 | Docteh | SomethingIsodd: I use it for PPPoE but not asterisk |
| 19:57.33 | rue_mohr | OOooo I'm not hte only one with a cbc problem? |
| 19:57.35 | superbeef | If I loop my t1 card with zttool, and stick a loop plug in the back of hte card, I should see some TX and RX activity on here shouldn't i? http://pastebin.ca/1466716 |
| 19:57.43 | rue_mohr | wonder if they have their transmitters dialed up too much |
| 19:58.02 | SomethingIsodd | Docteh maybe you can still help i am reporting the cdr`s to a mysql database. and its putting everything in the database twice. i can not figure out how to stop it from dupilicating eveything |
| 19:58.06 | rue_mohr | you wouldn't have a red alarm |
| 19:58.14 | rue_mohr | you know how to make a t1 loopback? |
| 19:58.39 | louben | 1-4,2-6 |
| 19:58.41 | jplank | 1-4, 2-5 |
| 19:58.48 | rue_mohr | hah |
| 19:58.51 | louben | s/6/5 |
| 19:58.54 | jplank | is right |
| 19:59.00 | superbeef | rue_mohr: well the gu on the site is gonna stick a plug in |
| 19:59.08 | Docteh | SomethingIsodd: thats odd, is it doing a radius account packet when the call starts and when it ends? |
| 19:59.32 | SomethingIsodd | Docteh it was i have removed that entry so it only does it at he end of the call. |
| 19:59.38 | BlargMaN00 | technically you are both wrong... it's 1-5, 2-4 |
| 19:59.38 | SomethingIsodd | but it still puts the end of the call in twice |
| 19:59.50 | rue_mohr | http://eds.dyndns.org/~ircjunk/images/dscn9333_T1loopback.jpg |
| 20:00.10 | rue_mohr | thats a T1 loopback |
| 20:00.18 | rue_mohr | mkay? |
| 20:00.36 | BlargMaN00 | rue_mohr: that's what i said... |
| 20:00.36 | Docteh | SomethingIsodd: run the radius in debug mode, figure out if asterisk is sending two packets or if radius is writing twice for no real reason |
| 20:00.49 | SomethingIsodd | i already did and its not. |
| 20:01.26 | SomethingIsodd | its almost like i have a second spot in my config. on radius thats telling it to add it the second time. but i have been all over sql.conf and there is only the one entry |
| 20:02.10 | Docteh | if you raise the debug on free radius high enough i think it'll print out both sql queries |
| 20:02.14 | Docteh | well, inserts |
| 20:03.25 | superbeef | rue_mohr: my guy onsite was supposed to have a loopback plug but doesnt lol |
| 20:03.34 | *** join/#asterisk Knoxville (n=Knoxvill@70-90-77-201-BusName-mn.hfc.comcastbusiness.net) |
| 20:04.18 | SomethingIsodd | let me check |
| 20:04.20 | [TK]D-Fender | 1 minute job with a crimp tool |
| 20:04.27 | Knoxville | hey I have a new phone for the office. I edited sip.conf and voicemail.conf. reloaded the modules, and the phone now has voicemail and an externsion, however if anyone calls the phone it gives a busy tone, any ideas? |
| 20:04.29 | dacs | well Price does matter , but i want to experience with setting up * to originate call as well as terminating them! |
| 20:04.31 | superbeef | for some reason my boss doesn't have a loopback plug, but has a smartjack key |
| 20:05.18 | SomethingIsodd | Docteh it only shows one insert let me try a call |
| 20:06.29 | SomethingIsodd | Docteh it shows it being inserted twice. |
| 20:06.34 | [TK]D-Fender | dacs: Typcially there is nothing to "set up" for this really. |
| 20:06.36 | dacs | my main use for * right now, is that i want to have it setup for our church, where the church preist could open his softphone and leave a message " Mass service is canceled for Monday" and * will dial a list of all members phone leaving that message |
| 20:06.36 | SomethingIsodd | it wasnt ealier |
| 20:06.40 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 20:07.00 | [TK]D-Fender | dacs: Your same peer authing incoming calls you jsut do Dial(SIP/mypeer/18005551212,30) and you're done. |
| 20:07.14 | [TK]D-Fender | dacs: I wouldn't really call this an "acheivement" |
| 20:07.44 | dacs | [TK]D-Fender: maybe for you, but for me its really big |
| 20:07.51 | [TK]D-Fender | dacs: Now implementing realtime SIP peers, ODBC VM storage, writing an AGI or AMI application. These would be milestones to look at |
| 20:07.59 | [TK]D-Fender | dacs: Ok, well, its 1 dial command.... |
| 20:08.22 | [TK]D-Fender | dacs: You could get the same level of acheivement jsut by setting yourself up with ekiga.net and calling someone on there. |
| 20:08.34 | dacs | [TK]D-Fender: don't compare me to you...you have what maybe +5yrs exp with * so for you it a cake :) |
| 20:08.37 | [TK]D-Fender | dacs: Same functional difference as far as process goes, but at no cost. |
| 20:09.10 | [TK]D-Fender | dacs: I jsut showed you a single Dial line. How many guides out there for "how to setup broadvoice", etc all do this? DOZENS. |
| 20:09.22 | [TK]D-Fender | dacs: If you hve a peer entry for incoming, dialing = 1 line. |
| 20:09.27 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 20:09.39 | [TK]D-Fender | dacs: Trust me... it really doesn't qualify. |
| 20:10.18 | [TK]D-Fender | dacs: Heck, using AstDB to hold DND status and checking on extens used to dial local phones would be considerably more involving yet still rather petty |
| 20:10.28 | Docteh | mmmmm cake |
| 20:11.01 | dacs | [TK]D-Fender: ok, but i really want to digest * so i can understand whats going on. |
| 20:11.31 | dacs | but for me to make an outbound i will need a place to terminate my calls right |
| 20:11.39 | Docteh | drink plenty of milk, it'll help your body process the vitamin c |
| 20:12.20 | Docteh | yea you will |
| 20:18.37 | [TK]D-Fender | dacs: Of course |
| 20:20.18 | Knoxville | after you create a mac.cfg file do you need to reload any modules? |
| 20:21.11 | [TK]D-Fender | checkout time, later all |
| 20:21.45 | SomethingIsodd | Docteh ok i was wrong asterisk is not sending the entry twice only the once. i have double checked that |
| 20:22.03 | ariel_ | Knoxville, are you talking about setting up polycom phones. If your only editing the mac.cfg and not having to touch anything in asterisk like the sip.conf, then you don't have to do anything else. |
| 20:22.41 | Knoxville | yes it is polycom, I have edited voicemail.conf and sip.conf, everything works bu when calling the extension it gives busy tone |
| 20:22.50 | Knoxville | I figure I need to create a .cfg file? |
| 20:23.43 | e0n` | hmmm, so I figured out the outgoing queue but how do I (when a user picks up) attach it to a phone call |
| 20:26.56 | Katty | wtb pink blackberry case. |
| 20:27.01 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:27.06 | *** join/#asterisk swiftkick (i=eeeeeeee@mail.beanproducts.com) |
| 20:28.53 | KavanS | Big KATO! |
| 20:33.00 | BlargMaN00 | Knoxville: i'm assuming that this is your first run-in with asterisk?? You didn't create an extension for the phone, you just created a sip device... you have to edit extensions.conf to tell the dialplan how to contact the device... |
| 20:33.24 | BlargMaN00 | Knoxville: i.e. exten => 200,1,dial(sip/200) |
| 20:33.38 | Knoxville | do I also need to make the mac.cfg file? |
| 20:34.06 | ariel_ | Knoxville, the mac.cfg are for setting up your polycom's there downloaded from either ftp,tftp or a html server |
| 20:34.22 | BlargMaN00 | Knoxville: yes... you need to make a mac.cfg for the device, and you also need to create a custom <extension>.cfg file for the phone as well... |
| 20:34.25 | ariel_ | Knoxville, but you need more then just the mac.cfg |
| 20:35.09 | BlargMaN00 | Knoxville: otherwise, your phone won't know how to talk to *, and * won't even know your phone exists... |
| 20:35.27 | Knoxville | The phone can dial out and such, works fine |
| 20:35.43 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:35.55 | SomethingIsodd | . |
| 20:36.33 | BlargMaN00 | Knoxville: ok, then don't touch your .cfg files for the phone... you need to add the line that i previously showed you to your extensions.conf file, so that * will know how to contact your phone... |
| 20:39.42 | Knoxville | after I edit the extensions.conf do I need to reload the module, and if so how? |
| 20:40.53 | e0n` | extensions reload |
| 20:40.56 | e0n` | from the asterisk console |
| 20:41.27 | BlargMaN00 | dialplan reload |
| 20:41.31 | BlargMaN00 | from the CLI |
| 20:45.34 | rue_mohr | howcome the asterisk book dosn't say anything about the format for the playtones app? |
| 20:47.20 | dacs | <PROTECTED> |
| 20:47.20 | infobot | extra, extra, read all about it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:48.01 | Knoxville | I need to reload both modules? |
| 20:49.43 | Docteh | odd, voicemailmain() exits if it fails to find an audio file |
| 20:50.40 | rue_mohr | in the source there are about 11 formats for playtones(), but there is no mention of them in the manuals |
| 20:51.46 | rue_mohr | hmm suppose I'll have to add it |
| 20:51.55 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 20:53.12 | rue_mohr | oh thats well hidden |
| 20:53.33 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) |
| 20:54.54 | *** join/#asterisk da__d00d (n=Dana@dsl-vlan422-66-18-194-227.nucleus.com) |
| 20:54.56 | rue_mohr | "Asterisk does not have a parameter to specify the modulation depth and uses 90% by default" |
| 20:55.47 | rue_mohr | where does that put the 1mw signal, 0db isn't 0.9 |
| 20:56.26 | rue_mohr | ultimitly, its what comes out of the codec thats important |
| 20:56.56 | Katty | goes through purse. |
| 20:57.02 | leifmadsen | helps |
| 20:57.10 | Katty | leifmadsen: youowuld not believe what all is in here. |
| 20:57.30 | rue_mohr | http://eds.dyndns.org/~ircjunk/images/ulaw.png |
| 20:57.45 | rue_mohr | 0db is .707v |
| 20:58.03 | leifmadsen | Katty: I have a g/f -- I would believe nearly anything |
| 20:58.08 | rue_mohr | its the 6th point from the middle on the graph |
| 20:58.31 | Blackvel | what mobile voip devices do you use with wlan/asterisk? pda with windows mobile, iphone, blackberry, Nokia Exx series mobile phone? |
| 20:58.56 | rue_mohr | E0 |
| 20:58.57 | Katty | leifmadsen: yeah and it's a big bag too. |
| 20:59.24 | leifmadsen | Katty: always are... all the more space for you girls to carry my stuff in! |
| 20:59.25 | Katty | leifmadsen: payment for my hospital bill and rent. |
| 20:59.42 | Katty | leifmadsen: pink checkbook, with license and cards and what not. |
| 21:00.10 | Katty | leifmadsen: two sets of keys... on those really long uhmm... go around your neck thingies. |
| 21:00.25 | Blackvel | leifmadsen: have no a working multi-dial solution, thanks to [TK]D-fender :) |
| 21:01.00 | Katty | leifmadsen: mp3 player, allergy medication, hand sanitizer, psudeophed, eyedrops. |
| 21:01.15 | Katty | leifmadsen: hair clips! |
| 21:01.40 | Katty | leifmadsen: chapstick type gel stuff. |
| 21:02.28 | Katty | leifmadsen: coinpurse. blackberry, sunglasses. |
| 21:02.40 | Katty | a lighter... |
| 21:02.45 | Katty | i don't smoke tho. it's to light candles at work |
| 21:03.15 | Katty | so that's where my thingy went to rip cable out of 66 blocks |
| 21:03.35 | Katty | 3 pens, another lighter. |
| 21:03.50 | Katty | it was a pretty rainbow one from spencers, but it's empty now :< |
| 21:03.59 | rue_mohr | can I run dahdi_test while a system is in use? |
| 21:04.52 | Docteh | well it wont grab the hardware if its in use |
| 21:04.57 | Katty | leifmadsen: spritsy bottle of nail polish remover and cotton balls. |
| 21:05.44 | Katty | leifmadsen: makeup bag. i won't share what all is in that thing. |
| 21:06.03 | Docteh | woah dude wall of text |
| 21:06.13 | Docteh | ;) |
| 21:06.18 | Katty | leifmadsen: flash drive. |
| 21:06.48 | Blackvel | have a good evening....bye |
| 21:07.08 | rue_mohr | well dahdi_test scores Average: 99.989038 |
| 21:07.16 | rue_mohr | which I think is pretty good |
| 21:07.25 | Katty | leifmadsen: VS perfume. |
| 21:07.50 | Katty | leifmadsen: and some lip gloss. |
| 21:08.08 | dacs | ~DID |
| 21:08.08 | infobot | did is, like, Direct Inward Dialing, or just a phone number |
| 21:10.25 | Docteh | heh heh heh |
| 21:10.42 | Docteh | ~valleygirl |
| 21:11.02 | eppigy | DONDE ESTA |
| 21:14.38 | Katty | eppigy: what are you making me for dinner. |
| 21:15.02 | thehar | russellb: !! |
| 21:15.22 | *** join/#asterisk smps (n=smps@193.170.53.51) |
| 21:15.53 | eppigy | Katty: linguine with red clam sauce? |
| 21:16.03 | eppigy | it is kind of spicey |
| 21:16.32 | russellb | ohai thehar |
| 21:17.05 | thehar | russellb: i will bez in huntsville next month |
| 21:17.16 | russellb | NOWAI |
| 21:17.27 | russellb | what's the occasion |
| 21:17.41 | thehar | dcap/you digium people i pretend to like |
| 21:17.46 | thehar | tee hee |
| 21:17.51 | russellb | ooh |
| 21:17.54 | russellb | cool :-) |
| 21:17.55 | thehar | i joke.. |
| 21:17.57 | thehar | jes |
| 21:18.09 | thehar | somethin like the 19-24th |
| 21:18.12 | russellb | tell Jan that you want me to join the class for lunch one day :-) |
| 21:18.20 | thehar | hehehe |
| 21:18.31 | russellb | and then i get free foodz |
| 21:18.32 | Katty | eppigy: linguine sounds good. not so much clams. |
| 21:18.34 | thehar | okie |
| 21:18.42 | Katty | eppigy: idont' think i've ever had clam sauce :< |
| 21:18.58 | thehar | i'll be like "jan, can you invite aaron, matt and russellb to class today?" |
| 21:19.18 | eppigy | red clam sause is delish |
| 21:19.21 | eppigy | justa little spicey |
| 21:19.27 | eppigy | white cam sauce is really good |
| 21:19.30 | eppigy | clam |
| 21:19.36 | Katty | whitecam sauce. |
| 21:19.40 | Katty | now with extra pixels. |
| 21:19.45 | eppigy | yes |
| 21:19.51 | eppigy | creamy pixels |
| 21:19.55 | thehar | dirty! |
| 21:20.00 | russellb | thehar: yessssssss |
| 21:20.01 | Katty | you are. |
| 21:20.04 | thehar | haha |
| 21:20.10 | eppigy | your face is dirty |
| 21:20.15 | Katty | your mom's face is dirty. |
| 21:20.22 | thehar | sometimes |
| 21:20.29 | eppigy | your elected offical's face is dirty |
| 21:20.33 | thehar | russellb: is the class all yawns? =| |
| 21:20.44 | russellb | thehar: do you know who is teaching? |
| 21:20.46 | Katty | eppigy: that's not all that's dirty! |
| 21:20.54 | eppigy | D: |
| 21:20.59 | thehar | russellb: not yet.. haven't registered but submited budget today |
| 21:21.12 | eppigy | possible illegal campagn contributions |
| 21:21.38 | Katty | he published an internal document to the police department saying people who are liberals, or libertarians, or carrying libretarian items should be arrested for 24 hours on suspicion of terrorism. |
| 21:21.45 | thehar | luckily there are deals goin on with flight/hotel in hsv right now or i'd be stuck in vegas in august |
| 21:21.45 | russellb | thehar: i don't know who all teachers ... Jared usually does, and he's amazing |
| 21:21.56 | eppigy | oh dang |
| 21:22.04 | thehar | hehe |
| 21:22.13 | thehar | that will be funny to be in jared's class if that's the case |
| 21:22.19 | Katty | also that ron paul supports are grouped into that thing. |
| 21:22.22 | Katty | along with Bob Barr |
| 21:22.52 | Katty | finds reddit article. |
| 21:23.17 | thehar | russellb: i'm out but i'll let you know 100% if i'm coming. |
| 21:23.32 | *** join/#asterisk [netman] (n=netman@193.153.152.144) |
| 21:23.34 | russellb | k, have a nice evening |
| 21:23.37 | thehar | you as well |
| 21:23.44 | thehar | have 50+ phones to flash at 3 am tonight! can't wait! |
| 21:23.55 | thehar | & |
| 21:24.07 | Katty | eppigy: http://www.prisonplanet.com/secret-state-police-report-ron-paul-bob-barr-chuck-baldwin-libertarians-are-terrorists.html |
| 21:24.19 | Katty | eppigy: something similiar happened in alabama too, i believe. |
| 21:24.47 | Katty | eppigy: there were a few arrests in KC |
| 21:25.09 | Katty | eppigy: and at least 1 person with fund raising material stopped at an airport, i forget which one. |
| 21:26.18 | thehar | actually russellb is michelle in the office? |
| 21:27.22 | russellb | thehar: no |
| 21:27.33 | Katty | that's your mom. |
| 21:28.01 | eppigy | Katty: thats a shame |
| 21:28.30 | Katty | eppigy: yes. it was fixed tho. |
| 21:28.39 | Katty | eppigy: and a formal appology issued from the governor |
| 21:28.47 | eppigy | haha |
| 21:28.55 | eppigy | apologies |
| 21:29.23 | eppigy | bee rr bee |
| 21:30.27 | ruben23 | hi |
| 21:30.47 | Katty | hi ruben23 |
| 21:30.54 | Katty | what's up with the 23 |
| 21:31.04 | Katty | did your first 22 clones fail? |
| 21:31.07 | *** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net) |
| 21:31.11 | ruben23 | asterisk error logs during calls http://pastebin.com/m11c599d5 |
| 21:31.31 | bryanfe2 | is there a way I can have my asterisk send a SIP client a message (during a call), to basically mute the call so that the SIP client doesn't send any audio? |
| 21:33.50 | ruben23 | hi katty: :) |
| 21:35.45 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:40.03 | *** join/#asterisk tamiel (n=tamiel@85-171-169-103.rev.numericable.fr) |
| 21:40.15 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 21:47.47 | rue_mohr | I want to transfer a 1mw to my telco's silent termination, how do i break that call after I do my tests? |
| 21:47.54 | Katty | oh hey |
| 21:47.56 | Katty | it's almost time to leaf. |
| 21:48.04 | rue_mohr | mae like a tree? |
| 21:48.13 | Katty | something like that |
| 21:51.50 | rue_mohr | k, so recording the local 1mw to a fxs shows a peak of -3.8db... not close enough to the 0db it should be |
| 21:52.02 | rue_mohr | this is just all so wrong |
| 21:57.02 | rue_mohr | ok, I think what I need to do here is the following |
| 21:57.07 | bryanfe2 | Question -- Does Asterisk 1.6.1 still not work correctly when SIP clients have silence suppression enabled? I know this was a problem with prior versions. Is it still? |
| 21:57.44 | rue_mohr | get dahdi_monitor working on my channelbank, set all the gains to 0 on the bank and on asterisk, put a meter on an fxs channel is get a properly calibrated 0db signal level |
| 21:57.58 | rue_mohr | note what it is on dahdi_monitor |
| 21:58.14 | rue_mohr | do an analog loopback and get more numbers |
| 21:58.48 | rue_mohr | use that to adjust the tdm800 card I have here aginst the co 1mw |
| 21:59.13 | rue_mohr | phone my home system sending a 1mw and calibrate our transmit levels |
| 22:05.55 | ehsjoar | I am going through all sip.conf paremeters. It seems the http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf is severely outdated. I am looking at the sample sip.conf in * 1.6.0.10. The problem is that not all parameters are described and their default values are not specified. Does anybody know a good place to find this info? I am looking for |
| 22:05.55 | ehsjoar | 1. What parameters are valid for a User, Peer and in the general section |
| 22:05.55 | ehsjoar | 2. What are their default values |
| 22:05.55 | ehsjoar | I am working on a database schema that covers all parameters. Ultimately I will hook it up to an app server that will feed a Web based GUI |
| 22:08.44 | Docteh | peer/user specific settings overrite whatever is set in general |
| 22:09.36 | *** join/#asterisk [jmc] (n=John@93-45-222-181.ip104.fastwebnet.it) |
| 22:10.14 | Docteh | any setting thats specific to the asterisk server itself, like a bind port, etc wouldn't be usable in a user spot |
| 22:10.26 | ehsjoar | Docteh: Thanks. I got that part. Some parameters are only in the general section though (like bindport) and some are only in the peer/user. I guess I can go through them all by trying to set them and see where they are valid. |
| 22:11.05 | ehsjoar | Docteh: Yes, some of them are easy to understand. Not all though. For instance the directrtpsetup one |
| 22:11.28 | ehsjoar | Why is that guy not valid in a peer/user section? It seems that it should be |
| 22:12.38 | Docteh | http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup |
| 22:12.39 | ehsjoar | I have tried to put it there but a sip show peer <peer> doesn't list it |
| 22:13.34 | ehsjoar | Docteh: Thanks, that explained that parameter really well |
| 22:14.37 | ehsjoar | The reason I am looking for good explanations and where a parameter belongs is that I want to have a "help" button in the Web GUI that would explain it. I guess I just have to go through them one by one. Thanks for your help though |
| 22:15.25 | rue_mohr | "Usage: dahdi_monitor <channel num> [-v[v]] [-m] [-o] [-p] [-l li...." >dahdi_monitor 2 -vv -p dahdi_monitor: invalid option -- p ????? |
| 22:16.00 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:19.51 | comradeb14ck | hey all--anyone know about the RECEIVE CHAR AGI command? |
| 22:20.08 | comradeb14ck | i've been playing with it, and it appears that regardless of what happens, when i use that command it auto-kills my channel |
| 22:20.11 | *** part/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net) |
| 22:20.23 | comradeb14ck | so all agi commands i use after calling RECEIVE CHAR cannot execute unless they run dead |
| 22:20.25 | *** part/#asterisk [jmc] (n=John@93-45-222-181.ip104.fastwebnet.it) |
| 22:21.00 | rue_mohr | does anyone here know anything about telephony audio? |
| 22:21.16 | *** part/#asterisk swiftkick (i=eeeeeeee@mail.beanproducts.com) |
| 22:22.28 | *** join/#asterisk ISO9001 (i=blank@slu.ms) |
| 22:22.44 | Docteh | rue_mohr: might want to ask a more specific question |
| 22:23.13 | rue_mohr | like what value on dahdi_monitor represents 0db |
| 22:23.24 | rue_mohr | I'v been asking for days,nobody knows |
| 22:23.57 | ISO9001 | I'm using FollowMe, but I'd like something like Playtones(ring) instead of musiconhold while it's trying to find an extension that answers. Is that possible? |
| 22:24.22 | rue_mohr | I think it might be about 11000, based on the level readings from recordings made with dadhi_monitor |
| 22:25.22 | Docteh | ISO9001: whats wrong with playing the ringing sound with moh? |
| 22:25.43 | ISO9001 | Docteh: nothing, can I? I thought I was limited to playing mp3s and whatnot. |
| 22:26.56 | ISO9001 | or are you suggesting an mp3 of the ring tone? |
| 22:27.00 | Docteh | yea |
| 22:27.06 | rue_mohr | use playtones |
| 22:28.10 | ISO9001 | rue_mohr: how? |
| 22:28.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:28.44 | rue_mohr | read the wiki pages for playtones and the tone definitions |
| 22:29.10 | ISO9001 | heh. Playtones(ring) gives the sounds I want. I'm not sure how it works with FollowMe though. |
| 22:29.34 | rue_mohr | it'll do it till you say to stop |
| 22:34.21 | ISO9001 | I'm not sure if that's possible here. |
| 22:35.05 | ISO9001 | followme only returns if it can't reach anyone. Maybe if it stops tones if it finds someone... will have to experiment I guess. |
| 22:35.19 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:37.36 | ISO9001 | rue_mohr: yeah that doesn't work. Thanks for the suggestion though. |
| 22:46.29 | rue_mohr | ok off with all the echo suppression |
| 22:53.45 | rue_mohr | 1mw from the co is spilling into the fxo tx REALLY badly |
| 22:54.12 | rue_mohr | try a sip set |
| 22:54.35 | rue_mohr | oh thats better |
| 22:59.57 | *** join/#asterisk juanIMP (n=juan@190.144.243.226) |
| 23:03.34 | *** join/#asterisk mykhyggz (n=mykhyggz@72.11.84.62) |
| 23:05.15 | drmessano | rue_mohr: Is this the same problem you had 4 months ago? |
| 23:05.59 | rue_mohr | YES! |
| 23:06.16 | rue_mohr | I'm really working hard to resolve this, but its really hard when nobody here knows anything about audio |
| 23:06.23 | rue_mohr | or so it seems |
| 23:06.40 | drmessano | What is the issue? |
| 23:07.30 | rue_mohr | a) echo b) volume c) audio muting (fixed by adjusting vmnlpthresh to 4) d) ... what was d |
| 23:07.37 | rue_mohr | let me go get my book |
| 23:08.21 | drmessano | What kind of card are you using? |
| 23:08.39 | lanning | 3) PROFIT! |
| 23:09.14 | drmessano | lanning: Thats always 4 |
| 23:10.00 | lanning | damn, I will have to remember that... :) |
| 23:10.10 | drmessano | 1. Do something, 2. Do something else (which is the nonsensical bit you are mocking), 3. ??????, 4. Profit |
| 23:10.18 | rue_mohr | tdm800 with echo can |
| 23:10.23 | drmessano | 1. Install Asterisk <-- Good start |
| 23:10.44 | rue_mohr | oh we had an issue where the card wouldn't pick up the line, but I think that was our telco |
| 23:10.58 | drmessano | 2. Install ClamAV |
| 23:11.01 | drmessano | 3. ??????? |
| 23:11.05 | drmessano | 4. Profit!!! |
| 23:11.23 | rue_mohr | and the svn driver caused us to have no audio for inbound calls |
| 23:11.33 | rue_mohr | so I'm using the newest stable |
| 23:12.01 | *** join/#asterisk engien (n=mark@c-71-199-107-125.hsd1.pa.comcast.net) |
| 23:12.07 | rue_mohr | I have a call into digium right now, been waiting all day for the tech to return the call |
| 23:12.09 | drmessano | There is no such thing as an "svn driver".. SVN is a conduit, it does not represent a version |
| 23:12.14 | rue_mohr | dont think its gonna happen today |
| 23:12.30 | rue_mohr | vsn head as I understand |
| 23:12.56 | rue_mohr | basically, any calls from the co had no audio, so I worked as quick as I could to get the old drivers back and played with it more later |
| 23:13.26 | rue_mohr | right now, I'm trying to get hte volume levels right, but nobody here seems to know anything about 0db calibration |
| 23:14.15 | drmessano | Well, head means little more than the most recent revision.. that could be across any number of branches.. |
| 23:14.40 | rue_mohr | I know, I dont know how to find out what it was, thats why I fell back on the latest stable |
| 23:14.51 | rue_mohr | which I'd like to point out stil has the broken dahdi_monitor |
| 23:15.40 | rue_mohr | I'm finding I have to adjust gains ont eh dahdi card by like 10db to get proper results |
| 23:16.03 | rue_mohr | and I'm not ht only one, turns out that a few have had to put tx on the fxo to -10db to get it to work right |
| 23:16.18 | rue_mohr | as was the conversation this morning with a few toher tdm800 users |
| 23:16.53 | tompaw | Hi. |
| 23:17.12 | drmessano | So I always pull latest from the most recent branch.. I avoid trunk |
| 23:17.16 | tompaw | That's strange, I got my context with 8 priorities and 4 "h"s. |
| 23:17.43 | lanning | uh, 4h club is a bit different... :) |
| 23:17.43 | tompaw | Now, after I hang up, those 4 "h"s are executed and then it executes "h"s 5-8 that do not exist! |
| 23:17.52 | [TK]D-Fender | tompaw: contexts don't have priorities. |
| 23:18.05 | [TK]D-Fender | tompaw: PASTEBIN is your friend. |
| 23:18.34 | rue_mohr | drmessano, like I say, I'm not sure what it was the digium tech downloaded for me, but when I finally installed it, there was no audio for incomming co calls |
| 23:19.14 | rue_mohr | the gains are odd, as you dial them up, they get louder, but at a point they just start spilling into the opposite path |
| 23:19.57 | lanning | that's normal. |
| 23:20.13 | rue_mohr | and it conflicts, I cant seem to get the proper levels out of it before it spills over |
| 23:20.30 | lanning | at a certain point they are too loud and you get bleed over. |
| 23:20.45 | rue_mohr | yea, but that shoudln't happen till way over 0dbm |
| 23:20.50 | rue_mohr | way over |
| 23:20.57 | lanning | it's a hardware thing, with amplification and close circuits. |
| 23:21.03 | rue_mohr | I'm not getting to 0 and I'm spilling over |
| 23:21.38 | rue_mohr | then again, I'm having to guess where 0db is on the dahdi_monitor cause nobody knows |
| 23:22.09 | rue_mohr | I dont know enough about the audio format conversions going on in asterisk |
| 23:22.18 | rue_mohr | I know where 0db is in a ulaw stream |
| 23:22.45 | engien | I know absolutely nothing about asterisk.. just bought two cisco 7912g ip phones, would like to make them talk to each other - just for a learning experience. Any suggestions on where to start ? |
| 23:22.55 | rue_mohr | http://eds.dyndns.org/~ircjunk/images/ulaw.png its at the 700mv mark |
| 23:23.04 | rue_mohr | which is the 6th point from the middle |
| 23:23.08 | [TK]D-Fender | engien: ... |
| 23:23.09 | drmessano | engien: Putting them on ebay and getting some polycoms is a good start |
| 23:23.09 | [TK]D-Fender | ~book |
| 23:23.09 | 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 |
| 23:23.11 | [TK]D-Fender | ^^^^ |
| 23:23.26 | rue_mohr | I do not advise polycom phones |
| 23:23.36 | *** join/#asterisk datafirm (n=wprater@74-94-79-145-Seattle.hfc.comcastbusiness.net) |
| 23:23.37 | engien | drmessano: why? |
| 23:23.38 | datafirm | Hello. |
| 23:23.45 | rue_mohr | you can get and look at the manual for them, you wont see whats missing |
| 23:24.04 | [TK]D-Fender | rue_mohr: Everyone else is fine, you card is fucked. Please accept the obvious before passing the buck. |
| 23:24.16 | datafirm | Im getting a _NODEST when trying to use direct dial from our IVR.. can someone help me with a few quick debug techniques? |
| 23:24.23 | rue_mohr | they wont show dialed digits during a call |
| 23:24.26 | [TK]D-Fender | rue_mohr: Its not the phone... its your card |
| 23:24.33 | rue_mohr | they lack a descent number of buttons |
| 23:24.48 | rue_mohr | I'm not talking about the volume levels, I'm happy to say thats the card |
| 23:25.13 | rue_mohr | besides there is no sip stream monitor and tools that I can tell if the rtp steams have the amplitude they should |
| 23:25.15 | [TK]D-Fender | rue_mohr: Well few have your insistance on *'s broken SLA so the buttons are just fine. |
| 23:25.33 | *** join/#asterisk secgod (n=secg0d@c-67-184-227-156.hsd1.il.comcast.net) |
| 23:25.40 | rue_mohr | there are not enough buttons |
| 23:25.53 | rue_mohr | aastra comes with 8 |
| 23:26.00 | secgod | what distro seems to be full and features and stable these days ? |
| 23:26.04 | rue_mohr | polycom comes with basically none |
| 23:26.09 | rue_mohr | debian |
| 23:26.19 | rue_mohr | but go guiless |
| 23:26.38 | rue_mohr | ok, this day is over |
| 23:27.15 | rue_mohr | I learned nobody here knows anything abotu audio, somehow I have to prove to digium that there are fundamental flaws in the tdm800 |
| 23:27.17 | drmessano | thinks SLA ranks up there with dinosaurs and key systems in the list of "Top 10 things people see when having sepia flashbacks" |
| 23:27.31 | rue_mohr | I dont have a sla system |
| 23:27.58 | drmessano | You also dont have a working tdm800 card |
| 23:30.55 | rue_mohr | were using it |
| 23:31.32 | rue_mohr | but I have to dial the fxo tx to -10db and I have to dial the polycom handset volume (earpiece) to 11db |
| 23:31.51 | rue_mohr | otherwise the echo can goes balistic |
| 23:33.02 | rue_mohr | we have a supplier with a toshiba keyed system that we get a LOT of echo from, I suspect is has alot of reflection but it shoudlnt be an issue |
| 23:33.21 | rue_mohr | and i have no way to measure or confirm any of this |
| 23:33.43 | rue_mohr | hell I cant even tell if the levels at the ulaw streams are even close |
| 23:34.31 | rue_mohr | at -10db on the fxo tx, I'm surprised that there is any audio left to come back |
| 23:35.00 | rue_mohr | and the day is over, I ahve to go home |
| 23:35.26 | rue_mohr | I'm gonna upgrade my system at home and play with it, it uses a T1 channelbank that I can do some loop tests on |
| 23:35.45 | tompaw | [TK]D-Fender: http://pastebin.com/m1b292629 |
| 23:35.55 | rue_mohr | drmessano, thanks for the intelectual conversation |
| 23:36.40 | [TK]D-Fender | tompaw: because "_," matches ANY exten with 1 or more character INCLUDING "h" |
| 23:36.52 | tompaw | a HA! |
| 23:37.05 | tompaw | me stupid. |
| 23:37.06 | tompaw | thankx. |
| 23:37.13 | drmessano | Who the hell spends 5 months on a TDM card? |
| 23:37.18 | [TK]D-Fender | tompaw: And "_." is an incredibly stupid and dangerous pattern for reasons just like this |
| 23:37.33 | [TK]D-Fender | drmessano: I can give you a nick or two :) |
| 23:39.06 | drmessano | always assumed that was just a rue_mohr |
| 23:45.00 | hardwire | https://dedected.org/trac/wiki/COM-ON-AIR |
| 23:45.02 | hardwire | wtf is that? |
| 23:45.04 | hardwire | neat. |
| 23:46.41 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
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| 23:56.01 | *** join/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
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| 23:57.00 | *** part/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 23:57.28 | *** join/#asterisk misyel (i=misyel@203.177.200.239) |
| 23:58.17 | misyel | can someone tell me about asterisk? |
