| 00:03.17 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 00:06.48 | stoked | anyone know where to look for Singapore DID's? |
| 00:08.02 | *** join/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 00:08.33 | dan__t | Singapore. |
| 00:08.51 | stoked | nevermind |
| 00:08.54 | dan__t | :) |
| 00:08.56 | dan__t | I don't know. |
| 00:09.00 | dan__t | ~providers |
| 00:09.01 | jbot | well, providers is http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
| 00:09.25 | dan__t | That might work, I don' tknow |
| 00:09.26 | RichardLynch | Has anybody successfully installed this: http://blog.tmcnet.com/blog/tom-keating/asterisk/voice-changer-for-asterisk.asp soundtouch is erroring out in "make" with complaints about memcmp not defined in scope. Googling tells me it's gcc-4.2 versus gcc-4.3 issue. Trying to apt-get remove 4.3 and apt-get install 4.2 and then ./configure tells me I have no compiler at all. :-( |
| 00:11.56 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 00:12.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:14.11 | doolph | there's any script to configure the dahdi channels in AsteriskNow? |
| 00:14.36 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 00:17.48 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 00:40.42 | dan__t | Hrm. |
| 00:40.50 | dan__t | Any FastCGI hackers about? |
| 00:43.53 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 00:45.01 | doolph | hwo to configure dahdi |
| 00:45.10 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 00:48.28 | RichardLynch | doolph: As I understand it, you do it just like Zapatel, as it's mostly just a name change... |
| 00:48.42 | RichardLynch | Never done it myself, mind you. |
| 00:50.33 | RichardLynch | Can anybody recommend something like this http://blog.tmcnet.com/blog/tom-keating/asterisk/voice-changer-for-asterisk.asp but which actually compiles? |
| 00:52.29 | doolph | ok |
| 00:52.31 | doolph | I found it how |
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| 01:18.32 | doolph | omg |
| 01:19.37 | NovceGuru | OMFG |
| 01:20.04 | doolph | freepbx doesn't show reports :( |
| 01:23.05 | jaytee | it doesn't? |
| 01:23.16 | jaytee | is it supposed to? |
| 01:28.38 | eppigy | hello |
| 01:28.42 | eppigy | i am dave |
| 01:29.24 | eppigy | doolph: you will need to look in to asterisk+cacti etc |
| 01:30.37 | doolph | [Sat Mar 07 20:20:12 2009] [error] [client 192.168.5.76] PHP Notice: Undefined variable: tostatsmonth_sday in /var/www/html/admin/cdr/call-log.php on line 450, referer: http://192.168.5.77/admin/reports.php |
| 01:31.37 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 01:35.59 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 01:38.04 | NovceGuru | k |
| 01:39.39 | doolph | any idea somebody? |
| 01:39.52 | kb3ien | i just noticed i'm getting no cdrs in /var/log/asterisk/ even though i left [csv] uncommented in cdr.conf is there something else needed to get basic logging? |
| 01:41.25 | kb3ien | i tried decommenting the other fields, to no avail. is there a cdr debugger? |
| 01:41.38 | rob0 | kb3ien, did you restart? |
| 01:42.36 | rob0 | If you're running as non-root, does the asterisk user have write permissions to the log directory? |
| 01:42.46 | rob0 | (rwx actually) |
| 01:43.02 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 01:43.29 | drmessano | BTW |
| 01:43.36 | drmessano | Asterisk = Boolean Fail |
| 01:43.59 | [TK]D-Fender | drmessano: Say it ain't not never no true! |
| 01:44.19 | kb3ien | no, its wasnt restarted. but why [cdr] alone wasnt enough, i'm not sure. oh well. |
| 01:45.31 | drmessano | I request all boolean config options IMMEDIATELY be changed |
| 01:45.45 | rob0 | yes/no/maybe |
| 01:45.55 | drmessano | I expect a reversal of the base logic involved in the options |
| 01:46.09 | drmessano | canreinvite becomes nocanreinvite |
| 01:46.25 | drmessano | allowguest to noallowguest |
| 01:46.27 | drmessano | etc |
| 01:46.38 | [TK]D-Fender | drmessano: And people say you're too negative.. |
| 01:46.44 | drmessano | lol |
| 01:46.44 | [TK]D-Fender | LIES |
| 01:47.06 | doolph | hey anyone using AsteriskNow here? |
| 01:47.07 | [TK]D-Fender | rob0: I hold the patent on "illogical operators". |
| 01:47.18 | [TK]D-Fender | rob0: X = maybe Y |
| 01:47.29 | RichardLynch | doolph: Somebody is using $tostatsmonth_sday in a PHP script, and not initializing it. Either it's a typo of the var name, or they only initialize it sometimes. You can probably ignore the error, if things are working otherwise. If you have issues, pastebin the script and I'll look at it. |
| 01:47.36 | drmessano | Also, I want options that become more inclusive to use the format veryno, no, yes, veryyes |
| 01:47.58 | rob0 | doubleplus |
| 01:48.02 | RichardLynch | drmessano: wayno |
| 01:48.10 | rob0 | doubleplus yes |
| 01:48.13 | rob0 | doubleplus unyes |
| 01:48.19 | drmessano | or sike, lame, false, true, trudat, and doubleplus (I like that one) |
| 01:48.35 | doolph | RichardLynch I am using AsteriskNow beta 1.5, the Reports on Freepbx just not working, where should I see? |
| 01:48.37 | drmessano | Hang on |
| 01:48.37 | rob0 | Thank Mr. Orwell, I'm just passing it along. |
| 01:48.52 | drmessano | sike needs to be reserved for options that do nothing like they claim |
| 01:49.01 | drmessano | allowguest=sike |
| 01:49.04 | drmessano | noallowguest=sike |
| 01:49.06 | drmessano | ARGH |
| 01:49.56 | drmessano | Oh, and time to clear up this g726 crap too |
| 01:49.59 | drmessano | For now on |
| 01:50.06 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 01:50.44 | drmessano | allow=g729 |
| 01:50.44 | eppigy | YOINK |
| 01:50.52 | drmessano | allow=oldg726 |
| 01:50.56 | drmessano | ITS SIMPLE |
| 01:51.43 | RichardLynch | doolph: Pastebin this sucker: /var/www/html/admin/cdr/call-log.php on line 450 |
| 01:52.01 | drmessano | Also, need a trap in the default configs to keep people from using them |
| 01:53.10 | drmessano | noyoudont=yes |
| 01:54.28 | RichardLynch | ohnoyoudont | dontyoudare |
| 01:54.52 | drmessano | Ohhh |
| 01:56.42 | doolph | RichardLynch: http://pastebin.com/m5d8509b8 |
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| 02:44.50 | mmlj4 | bah, I'm having trouble compiling wanpipe drivers... include/linux/wanrouter.h:344: error: expected specifier-qualifier-list before âget_info_tâ # and google only has a couple of pastebins |
| 03:00.08 | RichardLynch | doolph: d7d9f12cb |
| 03:00.21 | RichardLynch | doolph: http://pastebin.com/d7d9f12cb |
| 03:00.41 | RichardLynch | Sorry - Went off to dinner. |
| 03:01.34 | drmessano | You can take the chump outta Compton, but you ain't eva takin Compton outta the chump |
| 03:02.50 | RichardLynch | thinks drmessano has been replaced by a bot |
| 03:03.08 | drmessano | Word |
| 03:03.16 | RichardLynch | OpenOffice |
| 03:03.34 | drmessano | Open to the source, yo |
| 03:04.16 | RichardLynch | Speaking of source, is ANYBODY willing to give this a go and tell me if they can compile this stuff, or is it just me that can't: http://blog.tmcnet.com/blog/tom-keating/asterisk/voice-changer-for-asterisk.asp |
| 03:04.44 | drmessano | No mo hi stepping with my ho's to the MS WORD |
| 03:04.51 | drmessano | Can I hear a "Oh yell yeah"? |
| 03:05.39 | RichardLynch | I had to install OOo 3 yesterday, as somebody sent me a job offer in .docx format. Sigh. I took the job, though. :-) |
| 03:05.43 | eppigy | YEAH COME ON |
| 03:05.49 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 03:06.03 | RichardLynch | Amen! |
| 03:06.11 | drmessano | PREACH IT |
| 03:06.39 | RichardLynch | Or can I do this with AGI and pipe audio in/out of sox??? |
| 03:06.55 | drmessano | WHITE SOX YO |
| 03:07.23 | Qwell | I need a physicist. |
| 03:07.35 | rob0 | drmessanobot=youbetcha |
| 03:07.38 | drmessano | MAYBE I CAN HELP, YO |
| 03:07.43 | drmessano | IM GELLIN LIKE MAGELLAN |
| 03:07.50 | NovceGuru | o qwell </stuey> |
| 03:08.00 | NovceGuru | fail |
| 03:08.07 | RichardLynch | I can see how I could get AGI to record, and there's playback, but can I pipe one in/out and do something with conference calling to mute one channel (the original voice) while piping the sox output in on a different channel? |
| 03:08.11 | drmessano | You said a physicist |
| 03:09.19 | drmessano | I AINT NO PHYSICIST YO, BUT IM GLOBAL LIKE CHERNOBYL |
| 03:09.25 | drmessano | CALL ME DA FALLOUT |
| 03:10.15 | drmessano | Im meltin down like a reacta |
| 03:11.05 | drmessano | makes an LOLcat with a cat laying in a puddle |
| 03:11.20 | drmessano | IM IN UR WATER TABLE, POISININ UR PEOPELS |
| 03:11.25 | jaytee | ah, I see we're off our meds again! tsk, tsk. |
| 03:11.34 | drmessano | I CAN QUIT AT ANY TIME |
| 03:11.41 | jaytee | lol |
| 03:12.42 | drmessano | Dont you have a pride parade to organize, Rainbow Brite? |
| 03:12.58 | drmessano | Wait, she didnt have rollerskates |
| 03:13.18 | drmessano | OK, SO I CANT ONE-MEME THIS REFERENCE |
| 03:14.41 | RichardLynch | mmlj4: What version are you compiling? |
| 03:14.43 | drmessano | Damn, made coolthreads quit |
| 03:14.55 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 03:16.46 | eppigy | lol |
| 03:16.48 | eppigy | lollin |
| 03:17.26 | RichardLynch | drmessano: You got a brand new key? |
| 03:17.34 | drmessano | Oh you bastard |
| 03:17.38 | RichardLynch | :-) |
| 03:17.41 | drmessano | Now that song is stuck in my head |
| 03:17.49 | RichardLynch | Now that damn thing is stuck in your head, isn't it? |
| 03:17.54 | drmessano | I hope you die on the toilet |
| 03:17.54 | RichardLynch | LOL async |
| 03:18.05 | drmessano | Embaressingly |
| 03:18.25 | drmessano | Cause of Death: "Straining from a hard poo" |
| 03:18.29 | drmessano | YEAH |
| 03:18.48 | drmessano | Good think I bought a Janes addiction CD |
| 03:18.55 | drmessano | thing |
| 03:19.00 | drmessano | Like 2 hours ago |
| 03:19.02 | RichardLynch | When I was a kid, my mom found the old lady next door dead on the toilet. I am not making this up. |
| 03:19.02 | drmessano | I can drown it out |
| 03:19.10 | drmessano | Ouch |
| 03:19.12 | drmessano | No shit? |
| 03:19.21 | RichardLynch | Whistle the theme from "Benson" tv show, if you can remember it. It'll drive anything out. |
| 03:19.25 | drmessano | Ba-dump-ching |
| 03:19.26 | RichardLynch | For real. |
| 03:19.39 | drmessano | Good ole Benson |
| 03:19.52 | RichardLynch | She wanted to move her before calling the morgue, but decided against. |
| 03:19.54 | drmessano | Robert Guilliaume or however he spelled it |
| 03:20.18 | drmessano | I wouldnt move her.. what if she had a clingon |
| 03:20.25 | drmessano | Could have gone airborne |
| 03:20.50 | drmessano | Wiping a dead old ladys behind is NOT IN MY JOB DESCRIPTION |
| 03:20.53 | RichardLynch | So, please, can AGI read/alter/write the audio for a channel, bouncing it through sox? Sample code/example? |
| 03:21.21 | drmessano | Not sure, not much of an AGI'er |
| 03:21.32 | RichardLynch | Or any other way to achieve that? |
| 03:21.43 | mmlj4 | RichardLynch: wanpipe-3.2.7.1 |
| 03:21.51 | mmlj4 | sorry, was flogging my box |
| 03:23.32 | mmlj4 | also, ./Setup dahdi doesn't seem to be a valid target, it only wants "install" |
| 03:23.39 | drmessano | Are you a he/she? |
| 03:24.08 | mmlj4 | drmessano: who? |
| 03:24.31 | RichardLynch | mmlj4: It would appear to be something provided by /proc stuff... get_info_t*, that is, from the comments I can find in another version of wanrouter.h |
| 03:24.48 | RichardLynch | Does your box have the usual /proc stuff? |
| 03:25.00 | mmlj4 | yep |
| 03:25.02 | mmlj4 | suse 11.1 |
| 03:25.12 | drmessano | I was thinking about flogging and box |
| 03:25.26 | mmlj4 | heh, ok |
| 03:25.38 | drmessano | Jaytee usually refers to his "BRB time" as "Flogging the dolphin"... and tried to play it off as "work stuff" |
| 03:25.42 | drmessano | But I watch HBO |
| 03:25.46 | drmessano | I am no dummy |
| 03:26.02 | RichardLynch | Is 3.2.7.1 incredibly new or old? |
| 03:26.14 | mmlj4 | lemme gander |
| 03:26.22 | jaytee | hahaha |
| 03:26.32 | jaytee | 3.2.7.1 what? |
| 03:26.47 | *** join/#asterisk jetlagmk2 (n=jetlag@70.104.80.249) |
| 03:28.03 | mmlj4 | STABLE Voice & Data Drivers ............ wanpipe-3.2.7.1.tgz (2008-08-21) \n (Zaptel/WAN/API) |
| 03:28.14 | mmlj4 | you know what, that'll never work with dahdi anyhow |
| 03:28.19 | mmlj4 | lemme get the beta |
| 03:28.32 | RichardLynch | mmlj4: get_info_t is defined in /include/linux/proc_fs.h, line 48, according to Google... You may be able to upgrade your kernel, install proc_fs, or just add an include to the wanrouter.h to get that proc_fs.h file loaded in ... |
| 03:29.29 | RichardLynch | This is kernel 2.6.8 I'm reading. |
| 03:29.54 | RichardLynch | So it's fairly old... |
| 03:30.22 | mmlj4 | ok, so... |
| 03:30.28 | NovceGuru | queso |
| 03:30.41 | mmlj4 | I'm at 2.6.27.19-3.2-default |
| 03:30.49 | mmlj4 | in other words, not too old |
| 03:31.05 | *** part/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
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| 03:31.41 | mmlj4 | xchat-- |
| 03:32.27 | RichardLynch | mmlj4: If you still want to try to compile it, just cram an <include proc_fs.h> at the top and give it a shot... |
| 03:33.57 | mmlj4 | the beta drivers are cranking and haven't died yet... maybe it'll work |
| 03:35.41 | RichardLynch | crosses his fingers for mmlj4 |
| 03:35.42 | mmlj4 | I think it's happy now |
| 03:35.48 | *** join/#asterisk JuStIcIa_ (n=john@cbl-sd-74-96.aster.com.do) |
| 03:36.02 | RichardLynch | Does anybody do AGI? Am I in the wrong channel or something? |
| 03:38.11 | mmlj4 | i'm interested in some perl agi thingies at some point... but no, sorry |
| 03:38.37 | RichardLynch | alright, time to start with "Hello World" and see what I can wring out of this sucker. |
| 03:40.08 | jaytee | RichardLynch, y'know there's a whole chapter on AGI in "the book"? |
| 03:40.18 | mmlj4 | true |
| 03:43.16 | mmlj4 | wanpipe setup complete && /me happy |
| 03:43.33 | mmlj4 | now comes the harder part :-) |
| 03:52.52 | RichardLynch | jaytee: Yes, but it doesn't make it at all clear if/how I could read the audio, pipe it to sox, and push back out the altered audio. |
| 03:54.08 | jaytee | hmmm, maybe you could use Asterisk's built in file convert feature instead? |
| 03:54.13 | NovceGuru | cat stuff | sox - blah |
| 03:54.14 | NovceGuru | ah! |
| 03:54.46 | drmessano | cat sox? |
| 03:54.50 | drmessano | How fucking adorable |
| 03:55.02 | jaytee | wasn't the Clinton's cat named Socks? |
| 03:55.14 | drmessano | They're mittens, you cat hating bastard, not sox |
| 03:55.23 | drmessano | MR WIGGLESWORTH WEARS MITTENS |
| 03:55.40 | NovceGuru | http://rachelhulin.com/blog/image/sox.jpg |
| 03:55.40 | NovceGuru | meow |
| 03:55.41 | drmessano | pets Mr Wigglesworth |
| 03:55.59 | drmessano | Nice |
| 03:56.02 | NovceGuru | think of all the things that cat has witnessed |
| 03:56.11 | NovceGuru | how many "cousins" it has seen |
| 03:56.12 | mmlj4 | we used to have a cat named ben hur |
| 03:56.18 | drmessano | Well |
| 03:56.26 | mmlj4 | well, we had named it ben, but then it had kittens |
| 03:56.50 | drmessano | Think about the Bush's dog, and imagine it laying the oval office, licking its balls, while its master orders the death of tens of thousands |
| 03:57.12 | NovceGuru | imagines |
| 03:57.16 | drmessano | pours socks a saucer of milk |
| 03:57.20 | mmlj4 | i wonder how many zero will starve to death? |
| 03:57.41 | drmessano | zero? |
| 03:57.47 | mmlj4 | 0 |
| 03:58.07 | drmessano | Im lost |
| 03:58.09 | NovceGuru | cat giveashit > /dev/null |
| 03:58.20 | NovceGuru | me to :( |
| 03:58.27 | NovceGuru | too, actually |
| 03:58.30 | drmessano | I wonder how many 9 will drink tea |
| 03:58.33 | drmessano | ???? |
| 03:58.34 | mmlj4 | my point being, if you make comments about bush, there are ample obama-scoffers in attendance |
| 03:58.46 | NovceGuru | I wonder if 62 will come over tomorrow |
| 03:58.52 | drmessano | I dont need a lecture, douchebag |
| 03:59.17 | rob0 | 42 |
| 03:59.19 | mmlj4 | and I don't want to listen to BDS |
| 03:59.33 | drmessano | So put me on ignore or leave |
| 03:59.39 | drmessano | But please, dont cry |
| 03:59.40 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-104-80-249.pskn.east.verizon.net) |
| 03:59.43 | drmessano | I cant take the crying |
| 03:59.46 | mmlj4 | cry? hardly |
| 03:59.49 | drmessano | Its TOO much |
| 04:00.10 | drmessano | hands mmlj4 a pussykerchief |
| 04:00.29 | drmessano | It will be ok, one day another republican will be elected and you can have another recession |
| 04:00.31 | drmessano | Dont worry |
| 04:01.14 | drmessano | Im sure in 8 years there will still be foreigners that disagree with us that some republican can try to eradicate |
| 04:01.47 | drmessano | I dont see a shortage of non-white rich people that republicans can try to eradicate |
| 04:02.12 | drmessano | Keep ya chin up, and ya nutz down |
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| 04:33.15 | mazpe | is there a way to change an agi variable from extension.conf? like the agi_extension variable |
| 04:37.53 | drmessano | Holy cow, Brian James is dead |
| 04:39.05 | drmessano | I wonder if he did any voice work for phone systems |
| 04:41.40 | mazpe | any special needs to happen to change a variable like agi_extension? |
| 04:42.16 | mazpe | Set(agi_extension=0000000) doesnt seen to be working |
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| 05:06.08 | [TK]D-Fender | mazpe: What is an "agi variable". You seem to be setting a perfectly boring normal channel variable there |
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| 05:15.17 | siera08 | i have MoH problem in asterisk 1.4.18 now. When i call external phone using cmd dial with 'r' option, i can listen 'connecting...blabla' and call is successed. |
| 05:15.32 | siera08 | When dial with 'm' option, MoH music plays. but i can't hear 'connecting...'.and when called party is busy, i can't hear busy message from telephone provider. |
| 05:15.36 | siera08 | anyone had problem yet? |
| 05:16.14 | siera08 | when i dial internal phone with 'm' option, it's okay... |
| 05:16.17 | siera08 | and i applied "internal_timing = yes" in asterisk.conf. but same problem. |
| 05:16.47 | [TK]D-Fender | siera08: is MoH playing when the busy message should come in? |
| 05:17.04 | siera08 | yes. |
| 05:17.14 | siera08 | i can't listen busy message. |
| 05:17.43 | siera08 | but when i call internal phone, i can hear that. |
| 05:17.51 | [TK]D-Fender | siera08: Sounds more like you have call-progress issues and * doesn't know when the other side answers |
| 05:18.16 | [TK]D-Fender | siera08: And calling other phones conencted to *, it at least knows what the call status is. |
| 05:18.23 | [TK]D-Fender | siera08: What are you calling OUT on? |
| 05:19.02 | siera08 | i call out external phone through isdn trunk. |
| 05:19.53 | siera08 | and using 'r' option instead of 'm' option, this problem has not occured. |
| 05:20.14 | [TK]D-Fender | siera08: you should never use "r"... |
| 05:20.20 | doolph | asterisk-addons-mysql doesn't work with asterisk 1.6? |
| 05:20.49 | siera08 | yes, i don't use both 'r' and 'm' option. |
| 05:22.06 | [TK]D-Fender | siera08: you should NEVER use "r" |
| 05:22.11 | siera08 | yes. |
| 05:22.23 | siera08 | using sip or iax trunk, it's not problem. |
| 05:22.37 | siera08 | but using zap or isdn trunk, it has problem. |
| 05:24.32 | siera08 | i think that asterisk doesn't catch call status... |
| 05:25.11 | siera08 | using sip or iax trunk, if called party is busy, call status is answer status and play busy message. |
| 05:26.44 | siera08 | but using zap or isdn trunk, if called party is busy, the telephone provider make call status to ringing status(maintain ring), not answer and play their busy message. |
| 05:29.27 | doolph | omg after upgrade this asterisk 1.6 dahdi stop working |
| 05:31.42 | siera08 | [TK]D-Fender: my asking is failed?? |
| 05:32.21 | siera08 | [TK]D-Fender: please, give me some clue to solve it.. |
| 05:36.46 | RichardLynch | siera08: I could be wrong, but I think when he said never use "r" he meant it... :-) But what do I know? |
| 05:39.41 | siera08 | RichardLynch: thank u. i gonna use 'm' option instead of 'r' option for MoH now. |
| 05:44.19 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 05:51.40 | siera08 | asterisk 1.4.18 enables to config "silence suppression"? |
| 06:05.50 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 06:12.24 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 06:30.54 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 06:37.47 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:40.30 | doolph | ok asterisk 1.6 sucks |
| 06:40.43 | doolph | dahdi got freeze |
| 07:19.01 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176206596.dsl.bell.ca) |
| 07:23.03 | *** join/#asterisk dandate2 (n=dandate2@c-67-169-101-44.hsd1.ca.comcast.net) |
| 07:23.09 | dandate2 | how do I use chanspy to whisper to my reps? |
| 07:27.32 | *** join/#asterisk _omer (n=_omer@119.152.5.183) |
| 07:27.34 | _omer | hi |
| 07:27.42 | *** part/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 07:28.48 | _omer | anyhelp on CALLBACK billing formula to hangup a call according to the dialed destinations and user balance? |
| 07:30.27 | _omer | a user has $2 in his account......he triggers a callback ... Destination one: Saudi Arabia ($0.15 p/m) ..destination two: USA ($0.02 p/m) ....then how to do billing and let the asterisk know the time of hangup.... |
| 07:35.04 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 07:44.59 | Qwell | stupid question for anybody listening.. what time is it currently? |
| 07:45.12 | Qwell | My phone and desktop have both freaked out. |
| 07:47.28 | rob0 | Sun Mar 8 07:47:28 GMT 2009 |
| 07:48.00 | rob0 | still CST |
| 07:59.46 | *** join/#asterisk HoverHell (n=hell@91.146.50.221) |
| 07:59.53 | drmessano | Oh man |
| 08:00.01 | drmessano | I found a bug in Dahdi.. |
| 08:00.08 | Qwell | user error |
| 08:00.14 | drmessano | Time change happened, and it reverted back to Zaptel |
| 08:00.16 | drmessano | !!!! |
| 08:00.26 | drmessano | I'll post to the tracker |
| 08:00.37 | Qwell | oh, hey, neat.. my computer didn't crash with the DST change this time. |
| 08:00.45 | Qwell | It did last time >.< |
| 08:01.06 | drmessano | heh |
| 08:01.19 | Qwell | err, wait |
| 08:01.26 | Qwell | no, it was the leap second |
| 08:01.33 | drmessano | DST seems to be one of the damn friggin hardest problems for programmers to overcome |
| 08:01.45 | drmessano | Ive seen more software glitches due to DST |
| 08:01.52 | Qwell | well, I told you what my G1 did, heh |
| 08:01.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:02.02 | Qwell | It didn't change a third time though like I expected. |
| 08:02.20 | Qwell | now I need to figure out how to get it to figure out what time it actually is.. |
| 08:02.48 | drmessano | At my old job, the radio automation systems used to go apeshit when the time changed.. it was like Y2K twice a year |
| 08:03.05 | drmessano | and of course, they ALWAYS had bugs at the first of the year due to some legacy code |
| 08:03.12 | Qwell | my friend was telling me that they had real problems with security alarm systems in Indiana |
| 08:03.25 | drmessano | Apparently If < 2009 is better than some Year + 1 crap |
| 08:03.48 | drmessano | Indiana would piss me off |
| 08:28.08 | SparFux | Is there a way for asterisk to make an analogue telephone hangup, by sending a special dtmf code or something? |
| 08:32.05 | SparFux | BTW I found something about the sending other than the own MSNs to external parties, there is a feature "CLIP - no screening" and it does allow exactly this I suppose. And it is only available for some kind of PMX connection, which I think I don't have. So I can only send one of my own 10 MSNs to the called party. |
| 08:38.06 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 08:38.55 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 08:40.22 | SparFux | But it's even an opportunity. I plan to set things up so that a caller keeps getting the dialtone even when I pick up the ongoing call, I am presented with audio reading the MSN, then I can decide on wether I want to connect, if so I press some digit and get connected, otherwise I hang up. Even when I hang up, perhaps, the caller should still get a ringing tone. So it won't be detected I didn't pick up :-D |
| 08:49.04 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 08:55.21 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 08:55.31 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 09:05.20 | SparFux | Can I interact with the called party without connecting the call? As to for example let the called party decide wether to pickup the call at all? |
| 09:21.37 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 09:29.40 | joako | SparFux: Yes |
| 09:30.44 | SparFux | joako: would be needed for my "no callerid workaround" :-) Plus, it is even better than callerid, because people won't see I simply don't pick up the call ;-) And nobody can peek my mobile for the number called. I could even setup a PIN request to be presented with the Callerid ;-) |
| 09:32.29 | joako | SparFux: see the CLI command "show application dial" M(x[^arg]) - Execute the Macro for the *called* channel before connecting........ |
| 09:32.43 | SparFux | ah. |
| 09:34.26 | SparFux | the docs say otherwise here. "M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers)." |
| 09:34.55 | SparFux | at least on voip-info.org, but you are RIGHT! The asterisk says what you wrote. |
| 09:35.00 | *** join/#asterisk zafar_ (n=IceChat7@116.71.217.229) |
| 09:35.01 | joako | I sitll use asterisk 1.4.... |
| 09:35.05 | SparFux | seems like voip-info.org isn't up to date. |
| 09:35.12 | joako | LOL voip-info still probably has it from 1.2 |
| 09:35.19 | SparFux | I have 1.4 too. the asterisk only doc says what you say. |
| 09:35.22 | SparFux | ok. |
| 09:35.27 | SparFux | that's great. :-) |
| 09:35.34 | joako | Well still, it would be after the called party answers the call, but before the calling party gets connected |
| 09:36.20 | SparFux | really. ok. |
| 09:36.21 | zafar_ | my IAX truck is saying unreachable to another asterisk box which is behind firewall |
| 09:36.50 | zafar_ | can anyone help with it, m in deep sh*t for almost two days now |
| 09:37.04 | Maliuta | IAX truck??? |
| 09:37.08 | Maliuta | is that b-double? |
| 09:38.12 | zafar_ | in mean trunk |
| 09:38.15 | mosty | zafar_, do you control the firewall? |
| 09:38.29 | zafar_ | yes |
| 09:38.35 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 09:38.44 | mosty | zafar_, does it work when you disable the firewall? |
| 09:39.03 | zafar_ | let me give u the picture |
| 09:39.33 | zafar_ | asterisk box has its own firewall, but asterisk box is behind another firewall |
| 09:40.20 | SparFux | joako: Great, this seems to work! ;-) |
| 09:40.40 | zafar_ | i can access it through web interface |
| 09:41.39 | SparFux | Now I only have to implement this in dialpaln. |
| 09:41.40 | zafar_ | i have open 4569 5004 5060 and some custom ports that i am using |
| 09:44.35 | SparFux | I can use espeak for saying the number. I think this was the best option with asterisk. |
| 09:45.21 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 09:46.00 | mosty | zafar_, but does it work when you disable the firewall? |
| 09:50.03 | joako | zafar_: See if your firewall has the option "ÃMZ" and enter the asterisk ip there |
| 09:56.46 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 09:59.06 | SparFux | I remember a way to use espeak in the dialplan without additional modules, I guess by using pipes. But I cannot remembe exactly. How could this be done? |
| 10:05.37 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 10:06.52 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 10:10.25 | SparFux | How can I connect the sound output of an apppplication run by System() to the sound output of the calling channel? |
| 10:13.38 | mosty | you'll probably need to look into chan_alsa |
| 10:16.18 | SparFux | Well, there is the problem that I want to forward the sound to the extension, not the sound card. I guess I have to create a wav file and play it. |
| 10:16.56 | mosty | what are you trying to do exactly? |
| 10:20.45 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 10:21.46 | SparFux | I want to Playback() the output of espeak. |
| 10:23.30 | SparFux | http://pastebin.com/d7b13f1ef doesn't work. And it is really complicated. I think i remember a really easy solution of this, but it might be it was only System(espeak xxx) and what I did was I heard the output on the local soundcard. But that's not what I want. The output should go to the phone of course. Same as output of Playback(). |
| 10:23.59 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 10:28.26 | SparFux | It's a pity I can't simply connect stdout of a program to asterisk input. |
| 10:30.41 | SparFux | perhaps something is in the book. |
| 10:32.34 | SparFux | or perhaps this agi stuff is what I need. |
| 10:34.07 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 10:34.56 | SparFux | Ah, I got a workaround. simply remove -w from http://pastebin.com/d7b13f1ef at sox line. |
| 10:36.19 | SparFux | I am stupid! SayDigits can be used, too :-) |
| 10:38.45 | SparFux | Ok, but it is too slow. |
| 10:55.53 | SparFux | That's it working so far :-) |
| 10:56.20 | SparFux | Only I have to manage the 2: "keep giving caller the ringing tone while the called party hangs up" :-D |
| 11:00.33 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
| 11:04.25 | SparFux | And I have no idea how to do this. Perhaps I can transfer the call to a NULL extension and let it ring on this one. |
| 11:07.40 | mosty | perhaps you could send the caller to a conference with moh that sounds like ringing |
| 11:08.14 | mosty | then if you want to take the call, you just join the conference |
| 11:08.33 | SparFux | mosty: ah, that would be nasty, you know they would have to pay fee for some connections, you know. |
| 11:08.50 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 11:09.04 | SparFux | So far everything works without actually picking up the call. That's great. Only letting it go on ringing is the last thing to do. |
| 11:09.23 | mosty | what are you trying to do, exactly? |
| 11:09.25 | SparFux | People will not even recognize denied calls then. I think this is great. |
| 11:09.36 | SparFux | mosty: look at http://pastebin.com/d320e6d1d |
| 11:09.57 | SparFux | http://pastebin.com/d2af98243 even |
| 11:10.23 | SparFux | When call-beaker-mobile is called, it does not pick up the call, but go to macro. :-) |
| 11:10.41 | SparFux | then it presents caller id with name and number, then it asks for dtmf. |
| 11:10.58 | SparFux | accept already works fine. replay callerid works fine, too. |
| 11:11.23 | SparFux | note that even a PIN is asked in the beginning. Ppl won't even be able to check for calls I get when they steal my mobile. |
| 11:11.27 | SparFux | or peek on it. |
| 11:11.45 | SparFux | Now, when I deny the call with "2", the caller gets disconnected. |
| 11:11.56 | SparFux | This is not what I want, I want the caller to not notice a thing. |
| 11:12.17 | SparFux | the called phone should be disconnected and the caller should still get the ringing tone :-) |
| 11:13.08 | mosty | i don't quite understand what you're trying to do, but just put Wait(999) then then Goto the wait priority after that exits |
| 11:13.30 | mosty | instead of ExitMacro |
| 11:13.59 | SparFux | really nice feature. |
| 11:14.13 | SparFux | hm... I will try. |
| 11:14.41 | mosty | how does the present-callerid macro get called? |
| 11:15.04 | SparFux | What I am trying again: in the present-callerid macro the call hasn't been picked up yet. And if I press 2 then (the called party) then the call should keep ringing on the caller side, but the called phone should be disconnected. |
| 11:15.22 | SparFux | mosty: check http://pastebin.com/d2af98243. It is called in the M() |
| 11:15.44 | SparFux | That's why in asterisk 1.4 the call isn't being picked up then. |
| 11:15.52 | SparFux | only if the macro reaches Answer() |
| 11:17.40 | mosty | what happens if you just hangup if you don't want to answer the call? |
| 11:18.15 | SparFux | I think it gets disconnected, too. Let me try. |
| 11:19.09 | SparFux | Yup, gets disconnected. I need to have a hangup extension which will do the unknown trick of letting it ring even if I hangup. |
| 11:20.51 | mosty | exten => h,1,Wait(999) ; exten => h,n,Goto(h,1) |
| 11:21.06 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 11:21.13 | mosty | tried that? |
| 11:21.27 | SparFux | Yes, it doesn't work. |
| 11:22.31 | SparFux | It seems in the macro it doesn't even go to h extension :-( |
| 11:22.31 | mosty | hmm, how about changing your Dial command to add LOCAL/s@wait-forever |
| 11:22.48 | SparFux | sure I have to define it in the dial command in call-beaker-mobile context. |
| 11:23.48 | mosty | and do the Wait thing in wait-forever |
| 11:24.00 | *** join/#asterisk dmcn (i=david@fatpaq.mcnally.dk) |
| 11:24.46 | SparFux | perhaps MACRO_RESULT is the key. |
| 11:25.11 | dmcn | hi - i'm getting the following error when doing DigitTimeout,10 in extensions.conf: No application 'DigitTimeout' for extension - what could be the reason? |
| 11:26.58 | mosty | dmcn, http://www.voip-info.org/wiki/view/Asterisk+func+timeout |
| 11:27.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 11:28.56 | SparFux | doesn't work so far: http://pastebin.com/d1eab13a4 |
| 11:29.07 | SparFux | It seems the h extension isn't used at all in a macro? |
| 11:31.35 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 11:32.27 | SparFux | GOT IT! |
| 11:32.53 | Gido-E | what did you do wrong? |
| 11:32.56 | SparFux | Check this out: http://pastebin.com/d3d8d6848 |
| 11:33.08 | SparFux | I use Set(MACRO_RESULT=CONTINUE) and THEN I Wait forever. |
| 11:33.23 | SparFux | CONTINUE :CONTINUE - Hangup the called party and continue on in the dialplan from where you called Dial |
| 11:33.36 | SparFux | is eXACTLY what I want. hangup the CALLED party and GO ON :-D |
| 11:33.57 | SparFux | Only that whan I don't press 2 but just hang up, it won't work. But perhaps I can figure that out, too. |
| 11:34.25 | SparFux | http://pastebin.com/d3d8d6848 is the working thing so far. Really great stuff. |
| 11:36.24 | SparFux | Even better wait_forever sthing: http://pastebin.com/d5e981010 |
| 11:36.47 | dmcn | mosty, thanks, i'll check it out :) |
| 11:37.31 | mosty | SparFux, have you tried Dial(LOCAL/${CALL_MOBILE}&LOCAL/s@wait-forever,...) ? |
| 11:37.43 | SparFux | no. |
| 11:37.47 | mosty | along with MACRO_RESULT=CONTINUE |
| 11:38.00 | SparFux | I have a context wait-forever? |
| 11:38.07 | SparFux | ah! |
| 11:38.09 | SparFux | I get you. |
| 11:38.38 | mosty | you can create one, and put Wait(999) in it |
| 11:38.50 | SparFux | yes, nice idea. |
| 11:40.50 | SparFux | It should not work, I have other extensions ringing, but the call was terminated without this wait_forever behind the macro call. |
| 11:41.24 | SparFux | the continue continues the former channel. This one has to wait. Still calling an extension does not help, if this channel which took over the call does terminate in some kind. |
| 11:41.53 | SparFux | anyways. |
| 11:44.41 | SparFux | sorry my fault. it is a test env in which I DONT have other extensions ringing. so your idea should work. |
| 11:46.46 | mosty | i'm not sure if it will work- but it's worth trying |
| 11:46.49 | *** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) |
| 11:47.21 | SparFux | I have a big problem. When calling from bri the whole thing doesn't work. The call gets connected immediately. |
| 11:49.45 | SparFux | strange. |
| 11:50.22 | SparFux | does the call get bridged or someting? |
| 11:50.51 | SparFux | ah, no. |
| 11:50.56 | SparFux | that's complicated. |
| 11:52.09 | mosty | what kind of bri card are you using? and are you using bristuff? |
| 11:52.14 | SparFux | ok, from my sip phone I call out to pstn in pstn-dialout context, which means, the call does not go to call-beaker stuff, but Dial is called in pstn-dialout immediately :-) |
| 11:52.53 | SparFux | mosty: I am using LCR, which is great, I have to say. |
| 11:52.53 | SparFux | Ok, but this is a dialplan issue, I suppose. |
| 11:53.26 | SparFux | I tried from one of my sip phone and these directly call Dial form pstn-dialout context. |
| 11:53.27 | mosty | sounds more like a channel driver isse to me |
| 11:53.32 | mosty | issue, even |
| 11:54.05 | SparFux | Oh wtf, I am so stupid, I called my mobile phone number directly. This does not eeven go thru asteirsk :-: |
| 11:54.14 | SparFux | no no, wrong number. |
| 11:54.37 | SparFux | I directly called my mobile phone. But I of course have to call my asterisk box's number which is forwarded to the mobile phone. :-D |
| 11:56.51 | SparFux | anyway , the mobile doesn't ring. |
| 12:04.33 | SparFux | is it critical to have more Dial() commands in a row, like Dial(Local/*) |
| 12:06.22 | mosty | huh? |
| 12:08.47 | Maliuta | gets critical on peoples communication skills |
| 12:09.00 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 12:09.07 | *** part/#asterisk dmcn (i=david@fatpaq.mcnally.dk) |
| 12:10.04 | SparFux | Calling s@call-beaker-mobile together with other sip extensions in one Dial() doesn't seem to work. |
| 12:10.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:12.06 | SparFux | Dialing "sip/Ekiga&sip/Twinkle&sip/Yate&iax2/Yate&sip/SPA2&Local/017648313720@call-beaker-mobile|60||TW" doesn't seem to work. |
| 12:12.23 | SparFux | all extensions ring, but not Local/017648313720@call-beaker-mobile |
| 12:13.18 | mosty | what do the asterisk logs say about the local channel? |
| 12:13.31 | SparFux | ah, I think again my fault. I am using three lines on a bri! STUPID! |
| 12:14.23 | SparFux | have to get a second mobile. hold on. |
| 12:16.42 | SparFux | works. :-) |
| 12:16.58 | SparFux | But I have a callerid clash, I think. Somewhere I set callerid to a deafult. this isn't wanted here :-) |
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| 12:39.46 | SparFux | *sigh* the typos... |
| 12:47.52 | SparFux | in the macro called by M(), the callerid seems to be wrong. |
| 12:48.21 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
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| 12:49.58 | *** join/#asterisk op3r (n=op3r@114.108.201.205) |
| 12:50.16 | op3r | hello is it possible to have a separate asterisk recording server? |
| 12:51.32 | mosty | op3r, for recording what? |
| 12:54.40 | op3r | call recordings |
| 12:54.55 | op3r | cos it takes a lot of resources |
| 12:55.01 | op3r | and I want to offload it to another server |
| 12:55.05 | op3r | is that possible? |
| 12:57.05 | mosty | that's possible i believe |
| 12:57.57 | *** join/#asterisk wonderworld (n=ww@ip-62-143-20-187.unitymediagroup.de) |
| 13:00.54 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 13:08.21 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 13:10.40 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 13:11.11 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-237-246.lns10.mel6.internode.on.net) |
| 13:12.20 | SparFux | Why aren't my NoOp() s being printed on log for debugging? |
| 13:16.37 | SparFux | Anyways, my feature is ready for use. |
| 13:20.52 | op3r | errr |
| 13:21.30 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 13:26.10 | *** join/#asterisk zenfox (n=steven@c-98-213-240-12.hsd1.il.comcast.net) |
| 13:26.28 | *** join/#asterisk possy (n=npossy@smtp.theinternet.de) |
| 13:26.39 | possy | good day |
| 13:27.40 | ghenry | hi |
| 13:28.34 | possy | I downloaded the current 1.4.23.1 to get asterisk to work with the lcr (mISDN v2) channel. After completion of configure && make install in both lcr and asterisk, trying to start asterisk -cvvvv I get an error in line 3306 re DAHDI support not correctly configure. Exact error is http://pastebin.com/m73cd29cd |
| 13:29.27 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 13:29.41 | possy | The modules.conf already contains lines to noload => anything with dahdi in there, i.e. http://pastebin.com/m19194b7b |
| 13:30.09 | possy | Is there anything else that I need to do? |
| 13:32.40 | *** join/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 13:32.53 | dlynes | Is asterisk 1.6 stable yet? |
| 13:33.40 | dlynes | I'm thinking about installing it on my home phone system, but it'll be connecting to predominantly 1.4.22 and 1.4.23 asterisk boxes |
| 13:43.12 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 13:53.23 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 13:56.30 | possy | There is NO more dahdi module in the /usr/lib/asterisk/modules asterisk:/etc/asterisk# ls /usr/lib/asterisk/modules/|grep dah |
| 13:56.30 | possy | asterisk:/etc/asterisk# |
| 13:56.51 | possy | and it still complains about a wrongly configured dahd |
| 13:56.57 | possy | dahdi |
| 14:01.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:04.07 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 14:05.39 | SparFux | Can I Dial() into a macro or extension and give arguments to it? |
| 14:07.37 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:08.25 | jjshoe | SparFux http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 14:09.10 | possy | Is my DAHDI error part of a FAQ, or is just everybody asleep, that might know the answer ;)? |
| 14:09.20 | SparFux | I don't mean the M() thing, but the main Dial() argument. |
| 14:13.28 | SparFux | dialplan stuff can get really complicated. |
| 14:14.22 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 14:17.05 | dlynes | SparFux: No, you do a Macro(macroname,arg1,arg2,arg3,arg4,...) |
| 14:17.42 | dlynes | SparFux: i.e. use the macro application not the dial application |
| 14:18.40 | SparFux | Ok, I have this nice feature implemented which will let me choose wether to answer a call or not after espeak read the CID to me. Here it is: http://pastebin.com/d489adda3 |
| 14:18.54 | *** join/#asterisk Great_Anta_Baka (i=c419ffd2@gateway/web/ajax/mibbit.com/x-d7bf8f3f04903a15) |
| 14:19.14 | SparFux | Now I would like to use this not only for ${DEV_BEAKER_MOBILE} which is LCR/somenumber , but even for my SIP channels. I am looking for a more general approach. |
| 14:19.56 | SparFux | But I use the feature with DIal() like this: Dial(${BEAKER_SOPH}&${BEAKER_ATA}&${CALL_BEAKER_MOBILE}) |
| 14:20.01 | SparFux | So other extensions are involved, too. |
| 14:21.32 | SparFux | So, ${CALL_BEAKER_MOBILE} is s@call-beaker-mobile and in this extension DEV_BEAKER_MOBILE is used to call actually. But how can I accomplish some other channel like SIP/example is used? I am thinking of something like Dial(${BEAKER_SOPH}&${BEAKER_ATA}&s@call-beaker-mobile^SIP/example) or the like. |
| 14:21.56 | dlewis | nice, even the president uses a Cisco 7970: http://www.nytimes.com/slideshow/2009/03/07/us/0307OBAMA_10.html |
| 14:23.18 | SparFux | Basically what I want to do is, I want to do a Dial(ext1&ext2&ext3&...,M(..)) but the M macro should only be used for some of the extensions. |
| 14:23.21 | SparFux | not for all. |
| 14:23.31 | mvanbaak | dlewis: you think he uses it with asterisk+chan_skinny ? |
| 14:26.22 | [TK]D-Fender | SparFux: then call multiple local channels and have those implement your macro |
| 14:26.58 | [TK]D-Fender | (or not, as applicable) |
| 14:27.00 | SparFux | So I need a local channel context for every device I want to call. |
| 14:27.12 | dlynes | SparFux: for every device you want to call through a macro |
| 14:27.29 | SparFux | Yes, somehow overhead, but I think that's the only way. |
| 14:27.49 | SparFux | Otherweise I would have to set a variable for the device to use. |
| 14:27.53 | dlynes | SparFux: eg: Dial(Local/ext1&SIP/ext2&SIP/ext3&IAX2/ext4&Local/ext5&Local/ext6) |
| 14:28.25 | SparFux | Yes, and then in the local chanels ext1, ext5 and ext6 I do the M() call. |
| 14:28.37 | dlynes | SparFux: correct |
| 14:28.57 | [TK]D-Fender | SparFux: A local channel for each dial you want to use the macro |
| 14:29.32 | SparFux | I think that's the easiest way, unfortunately. |
| 14:29.39 | dlynes | SparFux: exten => ext1,1,Macro(...) and same for ext5 and ext6 |
| 14:32.18 | dlynes | [TK]D-Fender: Do you happen to know if 1.6 has stabilized yet? |
| 14:32.29 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 14:32.40 | dlynes | [TK]D-Fender: or is still just as flaky as 1.4 used to be? |
| 14:33.00 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 14:33.08 | [TK]D-Fender | dlynes: NO |
| 14:33.20 | [TK]D-Fender | [10:29]<dlynes>SparFux: exten => ext1,1,Macro(...) and same for ext5 and ext6 <- to this |
| 14:33.33 | joako | How do I update polycom phones for the change in DST? I had this same issue last year.... |
| 14:33.44 | [TK]D-Fender | dlynes: 1.6 is moderately stable... still some quirks, but it matured much faster than 1.4 |
| 14:34.14 | [TK]D-Fender | joako: Upgrade toa firmware that knows of the change |
| 14:34.20 | SparFux | Fender: to this? |
| 14:34.27 | [TK]D-Fender | joako: Or search for "day" in sip.cfg |
| 14:34.56 | [TK]D-Fender | dlynes: Why are you showing him how to call a macro manually? He does not want this. |
| 14:35.07 | [TK]D-Fender | dlynes: it is a DIAL MACRO to auth the call. |
| 14:35.31 | [TK]D-Fender | SparFux: He's backwards on this. |
| 14:35.41 | SparFux | Fender: to auth the call AND to let the called party ACCEPT the call. Otherwise the call will not even be picked up. |
| 14:36.21 | dlynes | [TK]D-Fender: ah |
| 14:36.36 | [TK]D-Fender | SparFux: thats the auth I'm talking about |
| 14:36.49 | joako | [TK]D-Fender: Yep it was in sip.cfg. Firmware IIRC is pretty new |
| 14:36.57 | dlynes | [TK]D-Fender: thanks |
| 14:37.05 | dlynes | SparFux: sorry for confusing you |
| 14:37.11 | SparFux | dlynes: it's ok. |
| 14:37.38 | SparFux | But the said things apply anyway, I guess. I will have to have one context like [call-beaker-mobile] for every extension I want to use this for. |
| 14:37.40 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 14:38.02 | possy | is away: Doing girls stuff |
| 14:38.21 | dlynes | [TK]D-Fender: Will running 1.6.0.6 on one machine affect other machines that it's talking to, if they're running 1.4.22 or 1.4.23? |
| 14:38.59 | SparFux | Otherwise I think I cannot tell the [call-beaker-mobile] what device to use. I would have to give it arguments from within the Dial() statement. |
| 14:39.00 | dlynes | [TK]D-Fender: shouldn't affect them adversely, right? |
| 14:39.13 | [TK]D-Fender | SparFux: http://pastebin.com/m279cebdd |
| 14:40.10 | [TK]D-Fender | SparFux: I don't know why you made that context the way you did. Looks like half a macro. |
| 14:40.32 | SparFux | Perhaps it's because I am an asterisk noob. |
| 14:41.19 | SparFux | actually, the way you do it in the [sample] is the way I do it. But the question is, wether I can use [call-beaker-mobile] for other extensions, too. |
| 14:41.39 | SparFux | Half a macro? |
| 14:41.57 | SparFux | But I have a small update. |
| 14:42.37 | SparFux | http://pastebin.com/d46f149da |
| 14:42.45 | SparFux | there now is a context wait-forever |
| 14:44.10 | SparFux | Why is it half a macro? |
| 14:45.13 | SparFux | I call this pcid = private cid feature. It is a password protection for cid of incoming calls :-) |
| 14:46.32 | [TK]D-Fender | http://pastebin.com/m7fde9bb5 |
| 14:47.26 | [TK]D-Fender | this "wait forever bit" makes no sense |
| 14:48.13 | SparFux | Yeah fender! |
| 14:49.11 | SparFux | That looks nicer. |
| 14:49.30 | SparFux | dialwithaccept is a more general approach alrady. :-) |
| 14:50.06 | SparFux | Fender: wait-forever is needed to keep ringing on the calling party while the called party is already disconnected. |
| 14:51.33 | [TK]D-Fender | SparFux: Whats teh point? |
| 14:51.48 | SparFux | The caller shouldn't notice anything of the denial of call. |
| 14:52.13 | [TK]D-Fender | SparFux: You should cap it globally however |
| 14:52.28 | SparFux | cap globally? |
| 14:55.34 | [TK]D-Fender | SparFux: You shouldn't ring forever... |
| 14:55.48 | SparFux | aha. |
| 14:55.56 | SparFux | ok, there should be a timeout. |
| 14:57.26 | SparFux | should dialwithaccept use ${ARG1} ? |
| 14:57.51 | *** join/#asterisk ingenius (n=alektro@host16.190-136-29.telecom.net.ar) |
| 14:58.15 | [TK]D-Fender | SparFux: http://pastebin.com/m731794ce |
| 14:58.18 | [TK]D-Fender | SparFux: oops |
| 14:58.29 | SparFux | ok. |
| 14:59.14 | SparFux | extensions have to be numbers, right? they can't be names. |
| 15:00.15 | [TK]D-Fender | SparFux: Sure they can |
| 15:00.35 | [TK]D-Fender | SparFux: exten => fred,1,NoOp(zomg!) |
| 15:00.38 | SparFux | then it's quite comfortable. I can rename sample to pcid-extensions and name them sanely. |
| 15:01.14 | *** join/#asterisk takashi_85 (n=glory@196.219.89.79) |
| 15:01.16 | SparFux | Macro(macro-dialwithaccept,SIP/200) becomes Macro(dialwithaccept,SIP/200) |
| 15:01.44 | *** part/#asterisk takashi_85 (n=glory@196.219.89.79) |
| 15:02.44 | [TK]D-Fender | SparFux: mORE "OOPS" - http://pastebin.com/m3663a6ce |
| 15:02.56 | SparFux | ok ok :-D |
| 15:04.17 | SparFux | http://pastebin.com/d7524b4be |
| 15:04.35 | SparFux | removed the wait-forever and used dummy-ring :-) |
| 15:05.12 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 15:05.27 | [TK]D-Fender | SparFux: Now you forgot the ARG1, and I don't see the sample of you calling anything |
| 15:05.51 | SparFux | The calling stuff is way up in my dialplan in some mailbox macro :-) |
| 15:06.00 | [TK]D-Fender | SparFux: ok, well play with it a bit... |
| 15:06.12 | SparFux | no, I just haven't added ARG1 yet. I just saw it was missing, but never added it. |
| 15:06.29 | SparFux | But true, it has to be there. |
| 15:06.57 | SparFux | the pastebin reader has to just assume he can Dial(Local/beaker-mobile@pcid) |
| 15:08.53 | brunner | SendDTMF seems to always drop the first digit... is there a way to remedy that? |
| 15:09.42 | [TK]D-Fender | SparFux: ... *I'm* the reader and if you think I'll assume anything I don't see is actually done correctly, then Put. Down. The. Crack. Pipe (c) JerJer |
| 15:09.50 | SparFux | brunner: perhaps voice channels aren't setup fast enough, add a Wait(2) in front perhaps? |
| 15:09.57 | [TK]D-Fender | brunner: Send it twice, or wait |
| 15:10.20 | brunner | I'm waiting 5 seconds before sending anyhting |
| 15:10.32 | brunner | it still drops it |
| 15:10.34 | [TK]D-Fender | ok, BBIAB |
| 15:11.26 | brunner | even if I put a w in front of the digits in the dial command, it still drops it |
| 15:11.54 | brunner | exten => s,n,Wait(3) |
| 15:11.54 | brunner | exten => s,n,SendDTMF(w1#########1) |
| 15:12.04 | brunner | the first 1 never sends |
| 15:12.09 | SparFux | Fender: I bet the missing /n in my dialplan is what made my CALLERID be wrong! |
| 15:13.23 | SparFux | http://pastebin.com/d1739bbe4 |
| 15:13.34 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:14.10 | brunner | sending an "a" first seems to work |
| 15:19.05 | SparFux | Fender: but nice thing, right? I like the pcid feature. :-) |
| 15:19.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 15:20.18 | SparFux | Fender: but nice thing, right? I like the pcid feature. |
| 15:21.00 | [TK]D-Fender | SparFux: I didn't read the full logic that close, but yeah, a somewhat normal thing to do. |
| 15:21.03 | SparFux | What I still have to implement is erasing the CALLERID and just giving it as audio. |
| 15:21.21 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 15:29.31 | [TK]D-Fender | arg, back later still |
| 15:31.13 | SparFux | With hiding CID: http://pastebin.com/d3e1a7c2 |
| 15:33.54 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 15:34.11 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:36.41 | NovceGuru | <PROTECTED> |
| 15:38.31 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:38.52 | [TK]D-Fender | Oh God... running Windows at home... I feel... dirty.... |
| 15:42.01 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:42.34 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:44.51 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 15:45.31 | SparFux | What do I have to set CALLERID(num) to, to actually have an unknown callerid? |
| 15:48.07 | [TK]D-Fender | SparFux: core show application setcallerpres" |
| 15:48.49 | brunner | can I concatenate values like this? a${EXTEN} |
| 15:49.01 | brunner | to prepend "a" to ${EXTEN} |
| 15:50.42 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:52.48 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 15:54.46 | *** join/#asterisk doolph (n=doolph@190.141.68.31) |
| 15:59.15 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 16:00.17 | *** join/#asterisk af_ (n=getsmart@88-149-230-241.dynamic.ngi.it) |
| 16:00.45 | riddlebox | do you guys sell any other pbx's beside asterisk? |
| 16:01.37 | [TK]D-Fender | riddlebox: Katty's company will whore any product you'll buy from them :) Most recently Panasonic PBX's |
| 16:02.36 | riddlebox | yeah our company is even worse avaya,win,vodavi,samsung,panasonic,nortel,hitachi....... |
| 16:03.37 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 16:04.44 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 16:08.39 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 16:09.05 | axisys | this does not look right http://pastebin.com/d2262e114 |
| 16:09.06 | axisys | why is it looking for zaptel? |
| 16:10.54 | [TK]D-Fender | axisys: because it SUPPORTS it perhaps? |
| 16:12.46 | axisys | [TK]D-Fender: how do I make it look for dahdi instead ? |
| 16:13.00 | axisys | i am trying to use it with asterisk-1.6.0.6 |
| 16:13.20 | [TK]D-Fender | axisys: how about asking in THEIR channel. |
| 16:16.46 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:17.38 | *** join/#asterisk luca`gervasi (n=ashura@host218-170-dynamic.16-87-r.retail.telecomitalia.it) |
| 16:17.41 | luca`gervasi | Hallo |
| 16:17.49 | axisys | i installed a digium like card from openvox.. two of them fxo and two of them are fxs ports.. is there a way i can tell which one is which ? |
| 16:18.13 | luca`gervasi | i need to debug my extensions, is there a cli command to show the contexts called by an agent? |
| 16:18.18 | axisys | i tried to plug a phone in one port.. i was hoping the asterisk console would display a message |
| 16:19.18 | [TK]D-Fender | luca`gervasi: core set verbose 10 |
| 16:19.53 | luca`gervasi | tnx, tring now :D |
| 16:20.24 | luca`gervasi | not working |
| 16:20.47 | luca`gervasi | it just says "== Using SIP RTP CoS Mark 5"... nothing more |
| 16:21.20 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 16:21.41 | [TK]D-Fender | luca`gervasi: then your description was a little poor and its not even hitting the dialplan. |
| 16:21.52 | [TK]D-Fender | luca`gervasi: you need to debug your SIP PEER |
| 16:21.59 | [TK]D-Fender | luca`gervasi: "sip set debug on" |
| 16:23.29 | dlynes | Does asterisk 1.6 not have a res_features.so module? |
| 16:23.29 | luca`gervasi | too much data... |
| 16:23.29 | dlynes | Just asking, because it ships with a features.conf file |
| 16:23.37 | luca`gervasi | some times ago, someone suggested me a way to have like a call trace for contexts |
| 16:29.32 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 16:31.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:34.59 | *** join/#asterisk Flare_123 (n=flare@189.101.252.221) |
| 16:39.32 | [TK]D-Fender | dlynes: Of course it does |
| 16:40.07 | [TK]D-Fender | luca`gervasi: if you set verbose to 10 and saw nothing then you aren't even executing dialplan. there are no contexts in play |
| 16:40.09 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 16:41.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:41.14 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:48.27 | luca`gervasi | i have a client, with a "context=internal" field. I have a [internal] context, with some extensions on...what could lead my agent not to go inside this extension? |
| 16:50.02 | [TK]D-Fender | luca`gervasi: I'm not going to waste time guessing until you show us the SIP debug. |
| 16:50.06 | [TK]D-Fender | luca`gervasi: PASTEBIN |
| 16:50.08 | [TK]D-Fender | `pb |
| 16:50.10 | luca`gervasi | yessir |
| 16:50.15 | luca`gervasi | immediately :D |
| 16:50.56 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 16:58.35 | luca`gervasi | http://rafb.net/p/11N1Zg43.html |
| 16:59.07 | *** part/#asterisk possy (n=npossy@smtp.theinternet.de) |
| 16:59.23 | [TK]D-Fender | luca`gervasi: where's the rest? |
| 16:59.36 | [TK]D-Fender | (not that I need it) |
| 17:00.01 | luca`gervasi | is there missing parts? |
| 17:00.06 | luca`gervasi | tryiong again |
| 17:00.53 | [TK]D-Fender | luca`gervasi: We didn't even get to the end of your call attempt. Also include your dialplan. |
| 17:03.23 | luca`gervasi | do you know how to instruct gnu screen to save the current buffer? :D |
| 17:04.27 | SparFux | Fender: next thing for me is to write a spam killer dialplan. The caller should be prompted with an announcement about forbidden spam, then some keypad codes are offered for the different concerns. :-D |
| 17:06.56 | *** join/#asterisk murraytm (n=murraytm@wsip-68-224-219-238.br.no.cox.net) |
| 17:12.46 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 17:13.23 | murraytm | when i use the Dial command with multiple numbers (i.e., Zap/r0/5551212&Zap/r0/5551213), is it possible to find out which of the numbers actually connected from inside the dial macro? |
| 17:16.04 | luca`gervasi | finally :| |
| 17:16.13 | brookshire | hi! |
| 17:16.24 | luca`gervasi | sorry, i was unable to strip down the console colors |
| 17:18.03 | luca`gervasi | http://rafb.net/p/nUerct87.html <---- stripped version |
| 17:22.30 | luca`gervasi | [TK]D-Fender, you there? |
| 17:22.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:23.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 17:26.28 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 17:29.16 | keith4_ | oh jeez. why is it triple-spaced? |
| 17:30.15 | [TK]D-Fender | luca`gervasi: and where's the DIALPLAn? |
| 17:31.34 | luca`gervasi | http://rafb.net/p/2H2Dm319.html |
| 17:31.36 | luca`gervasi | here :D |
| 17:32.30 | luca`gervasi | http://rafb.net/p/kZpCs439.html <--- agent definition |
| 17:33.03 | mvanbaak | [default] |
| 17:33.13 | [TK]D-Fender | luca`gervasi: Looking for 222 in inbound (domain 10.0.0.1) SIP/2.0 404 Not Found |
| 17:33.23 | mvanbaak | exten => _X.,1,Verbose(1,Upset [TK]D-Fender) |
| 17:33.32 | mvanbaak | ^^ my dialplan |
| 17:33.45 | [TK]D-Fender | luca`gervasi: do YOU see a "222" in [inbound] ? I know *I* don't |
| 17:34.25 | luca`gervasi | but... it should search for 222 in "internal" context... am i wrong? |
| 17:34.34 | luca`gervasi | look in the third paste pls |
| 17:34.38 | luca`gervasi | the phone definition |
| 17:34.45 | [TK]D-Fender | luca`gervasi: perhaps you forgot to actually apply your changes |
| 17:34.57 | luca`gervasi | realoaded astersisk 2 times |
| 17:35.08 | luca`gervasi | once i killed the process for sure |
| 17:35.08 | dlynes | [TK]D-Fender: hrm...don't see anything about the res_features.so file in the 'make menuselect' menu, either |
| 17:35.41 | mvanbaak | dlynes: res_features.so has been moved to main/features in trunk (and prolly some 1.6.X version |
| 17:36.20 | dlynes | mvanbaak: ah...so why is it still looking for res_features.so, then? |
| 17:36.36 | dlynes | mvanbaak: oh...nvm |
| 17:36.43 | [TK]D-Fender | luca`gervasi: you've clearly got something screwed up there. |
| 17:36.44 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:37.03 | dlynes | mvanbaak: I'm still using an asterisk 1.4 modules.conf file |
| 17:37.25 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 17:38.33 | mvanbaak | dlynes: :) remove the res_features.so line and you should be all set |
| 17:40.55 | op3r | how can you send a call to an extension of an asterisk server for example extension 1234 from another asterisk server without having an account? |
| 17:41.10 | op3r | some sort of like sip://1234@192.168.0.12 ? |
| 17:41.23 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 17:41.34 | mvanbaak | op3r: only if you allow guest calls |
| 17:42.12 | op3r | yeah I know but how can you send a guest calls? |
| 17:42.25 | op3r | from an asterisk server? |
| 17:42.45 | [TK]D-Fender | op3r: Dial(SIP/1234@otherserver) |
| 17:42.55 | mvanbaak | Dial(SIP/ip.of.other.asterisk/1234) |
| 17:43.02 | op3r | ok gotcha |
| 17:43.07 | op3r | thanks thanks |
| 17:43.17 | mvanbaak | both ways will work |
| 17:43.59 | op3r | thanks! |
| 17:44.43 | DarKnesS_WolF | [TK]D-Fender: hey my old friend :) how are u doing ? |
| 17:45.15 | [TK]D-Fender | DarKnesS_WolF: Still breathing |
| 17:46.18 | DarKnesS_WolF | [TK]D-Fender: ya i can see and kicing n00bs butts about reading manuals and the book :P |
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| 17:48.36 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 17:53.55 | [TK]D-Fender | BRB |
| 18:02.28 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 18:08.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:20.25 | luca`gervasi | what could give a "401 Unauthorized" ? |
| 18:23.56 | luca`gervasi | why would my phone get a "SIP/2.0 401 Unauthorized" when dialing out? |
| 18:24.03 | [TK]D-Fender | lucawrong user/pass jus like the error says |
| 18:26.22 | *** join/#asterisk possy (n=npossy@smtp.theinternet.de) |
| 18:26.53 | luca`gervasi | the agent registers...shouldn't it mean that the agent usese user/password correctly? |
| 18:27.06 | possy | good evening |
| 18:27.16 | luca`gervasi | hi possy |
| 18:27.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:27.53 | possy | I am using Asterisk 1.4.23.1 with mISDN v2 and LCR 1.3 - Call comes in, Asterisk answers, but the remote side does not hear a thing. |
| 18:27.56 | possy | hey luca`gervasi |
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| 18:28.35 | luca`gervasi | possy, behind nat? |
| 18:28.52 | possy | no. It is an ISDN card that is in the asterisk box (hfc) |
| 18:28.58 | luca`gervasi | i had similar problems till i setup correctly "externhost and localnet" |
| 18:29.09 | possy | luca`gervasi, yep, for SIP that is needed |
| 18:29.21 | luca`gervasi | sorry, read now |
| 18:29.33 | possy | thanks for trying :) |
| 18:29.57 | possy | I wonder if any of the bri ISDN using folks from Europe are alive |
| 18:30.19 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 18:32.31 | possy | and calling out on the ISDN channel does not work either. The ISDN side of things (lcr) shows a dialout attempt (and keeps the line busy until the timeout), but the other end does not see an incoming call |
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| 18:40.48 | luca`gervasi | [TK]D-Fender, found the glitch! :D |
| 18:45.03 | [TK]D-Fender | and...... |
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| 18:50.12 | luca`gervasi | sorry, here i am :D |
| 18:50.27 | luca`gervasi | domain=mydomain,WRONGEXTENSION |
| 18:50.50 | luca`gervasi | ...the domain context override the agent context context :| |
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| 19:15.08 | *** mode/#asterisk [+oo Qwell Corydon76-dig] by irc.freenode.net |
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| 19:39.44 | tlyng | is there any howto/tutorial on developing plugins for asterisk? (Using C) |
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| 19:45.59 | keith4_ | "plugins"? |
| 19:46.14 | keith4_ | ~followme |
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| 20:12.13 | keith4_ | what's the status of FollowMe in 1.4? the wiki has conflicting information |
| 20:13.14 | mvanbaak | core show application followme |
| 20:13.53 | keith4_ | hmm. helps to load the damn module |
| 20:14.23 | mvanbaak | run make menuselect |
| 20:14.29 | mvanbaak | it will show you what it needs |
| 20:14.50 | mvanbaak | prolly needs meetme and dahdi |
| 20:15.46 | [TK]D-Fender | mvanbaak: Differentiate between "load" and "compile".... |
| 20:15.53 | [TK]D-Fender | mvanbaak: And no, certainly not. |
| 20:15.59 | keith4_ | just wasn't loaded |
| 20:16.12 | [TK]D-Fender | app_followme=BLEH. Nothing you couldn't do in dialplan easily already. |
| 20:16.30 | [TK]D-Fender | And HAS been done a dozen times over. |
| 20:16.50 | keith4_ | yah, i've had a very basic "followme" for a long time.... |
| 20:17.06 | keith4_ | but people bitched about having to press a button to accept a call |
| 20:17.08 | mvanbaak | [TK]D-Fender: my idea. but ppl seem to like it ... |
| 20:17.38 | keith4_ | so, i made it connect immediately. which then breaks for people who turn their cellphones off, because it goes immediately to voicemail |
| 20:18.44 | keith4_ | I think this is a "can't please everyone all the time" situation, so I'm going to go back to how it was. and in looking for more robust implementations, I ran across the "official" followme app. but it's sequential, which I don't like |
| 20:19.28 | keith4_ | yet, the potential integration with AstDB is enticing. *if* it's true. there seems to be some dissenting opinion about it in the wiki, however |
| 20:20.21 | mvanbaak | you can use the astdb i your dialplan followme setup as well |
| 20:20.32 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 20:20.48 | keith4_ | I'm not sure I trust people to remember to "log in" to the follow me system |
| 20:21.21 | [TK]D-Fender | I'm pretty sure I don't trust people to dial normal PSTN phone #'s, but I try to let go a little... |
| 20:22.08 | keith4_ | thoughts on the first example that's listed here? http://www.voip-info.org/wiki/view/Asterisk+tips+findme |
| 20:23.49 | keith4_ | meh. I'll give app followme a shot, before I judge it |
| 20:25.29 | keith4_ | it can't be *that* bad |
| 20:26.26 | mvanbaak | it isn't. but it's also something that can be done in dialplan |
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| 21:17.17 | keith4_ | ugh. F this |
| 21:17.20 | keith4_ | i'm doing my own |
| 21:17.44 | keith4_ | writes "I will not question the wisdom of #asterisk" 100 times on a chalkboard |
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| 21:18.55 | keith4_ | what's the best way to test if an optional ARG is defined? compare it to ""? test for length > 0? or... is there an "isdefined" test? |
| 21:19.38 | thehar | depends if you're using LEN or not |
| 21:20.24 | keith4_ | eh? |
| 21:23.15 | [TK]D-Fender | keith4 : its a variable. Test it like every other |
| 21:23.38 | haps | any common 'gotchas' for checking voicemail? my sip phone (gs budgetone) is registered to the right voicemail, since it knows there's a message waiting, but when i dial voicemailmain() it doesn't recognize user/pass |
| 21:24.07 | haps | although i'm sending dtmf via rfc2833 |
| 21:24.23 | haps | (before I did that it would time out on user/pass, now it just says they're wrong) |
| 21:25.27 | haps | the cli says: [Mar 8 17:24:21] WARNING[954]: app_voicemail.c:6874 vm_authenticate: Couldn't read username |
| 21:26.07 | jaytee | you need mailbox=username@default in your sip.conf |
| 21:26.38 | jaytee | and the username should be your extension |
| 21:27.30 | keith4_ | [TK]D-Fender: does [${ARG4}] evaluate to true for non-null ARG4? |
| 21:27.55 | keith4_ | i can only find the opposite. e.g., that ! is valid as a unary operator |
| 21:28.04 | joako | any advice how can I get apache2 to allow the polycom phones to use http upload of the log files? |
| 21:28.15 | *** join/#asterisk jchonig (n=jch@firewall.honig.net) |
| 21:28.49 | keith4_ | ["${ARG4}" != ""] seems so... inelegant |
| 21:28.50 | jchonig | Stupid AEL question time. |
| 21:29.48 | jchonig | Is there a way to recursively expand variables (X=A; Y=B; Z=${X}${Y}; so that Z ends up being AB)? |
| 21:29.51 | jchonig | In 1.4.23.1 |
| 21:29.56 | mvanbaak | keith4_: inelegant but working |
| 21:30.27 | keith4_ | mvanbaak: true. inelegant-but-working trumps elegant-but-not-working, any day |
| 21:30.51 | mvanbaak | yup |
| 21:31.26 | keith4_ | I found an example of using ["${ARG4}3" != "3"]... which seems like something a VBscript programmer would do |
| 21:31.43 | keith4_ | ... but it would be <>, I guess |
| 21:32.05 | jchonig | keith4_ are you talking about sh scripts and test? |
| 21:32.14 | keith4_ | nope |
| 21:33.02 | [TK]D-Fender | keith4 : $["${ARG4}" != ""] or ${ISNULL(${ARG4})} |
| 21:33.32 | [TK]D-Fender | jchonig: Set(Z=${X}${Y}) |
| 21:33.39 | [TK]D-Fender | jchonig: And that isn't recursion. |
| 21:33.52 | keith4_ | [TK]D-Fender: perfect! (except that those two are logically opposite ;-) |
| 21:33.57 | jchonig | [TK]D-Fender Can I do that in a global? |
| 21:34.13 | jchonig | somehow? |
| 21:34.20 | [TK]D-Fender | keith4 : Yeah yeah.. you know what I mean... |
| 21:34.28 | keith4_ | [TK]D-Fender: Set(Z=${Z}${X}) ! |
| 21:34.30 | brookshire | ~ |
| 21:35.39 | haps | jaytee: I have that, only the context in voicemail.conf is 'home' so I have mailbox=101@home |
| 21:36.11 | jchonig | [TK]D-Fender: I mean, is there a place to set that globally? wiki says [globals] can only have VAR=VALUE |
| 21:38.18 | jchonig | I meant 'globals {' |
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| 21:41.36 | [TK]D-Fender | jchonig: Have you TRIED? |
| 21:42.21 | jchonig | nope, will do it right now |
| 21:42.33 | [TK]D-Fender | jaytee: Not quite... |
| 21:42.46 | [TK]D-Fender | jaytee: You are mixing VM notifications with his call to voicemailmain.... |
| 21:43.14 | jchonig | [Mar 8 17:42:48] ERROR[6965]: ael.y:812 ael_yyerror: ==== File: /etc/asterisk/extensions.ael, Line 162, Cols: 9-9: Error: syntax error, unexpected '(', expecting '=' |
| 21:43.22 | [TK]D-Fender | haps: Do you see it acknowledging the # you entered? |
| 21:43.38 | [TK]D-Fender | jchonig: Show us what you did, not jsut the error it generated |
| 21:43.38 | keith4_ | if his MWI is working, he must have the voicemail set correctly in sip.conf |
| 21:43.47 | [TK]D-Fender | keith4ARGH. NO |
| 21:43.52 | keith4_ | ducks |
| 21:43.53 | jchonig | <PROTECTED> |
| 21:44.02 | jchonig | PHONE_JEFF=SIP/ext-jch; |
| 21:44.05 | jchonig | reverse that order |
| 21:44.12 | [TK]D-Fender | jchonig: You don't use Set() is globals <- |
| 21:44.13 | jchonig | Both in the globals { section |
| 21:44.21 | [TK]D-Fender | jchonig: No apps in there |
| 21:44.25 | jchonig | Right. |
| 21:44.41 | jchonig | So I need to use the Set command where I reference that variable |
| 21:45.05 | [TK]D-Fender | jchonig: No. You reference it by REFERNCING IT. ${somevar} |
| 21:45.17 | [TK]D-Fender | jchonig: You use Set() to SET a variable |
| 21:45.30 | jchonig | If I use this: |
| 21:45.50 | jchonig | globals { PHONE_JEFF=SIP/ext-jch; PHONES_JEFF=${PHONE_JEFF} } |
| 21:46.13 | jchonig | When I reference $PHONES_JEFF in a DIal I get a value of $PHONE_JEFF |
| 21:46.42 | haps | [TK]D-Fender: I don't know what you mean there; now that i have the dtmf set to rfc2833 if i do "mailbox#" it goes straight to the password prompt... before it didn't matter if i pressed # or not it would timeout before giong to password prompt |
| 21:46.53 | haps | is there a voicemail debug setting i should turn on? |
| 21:47.08 | [TK]D-Fender | haps: pastebin the entire failed attempt |
| 21:47.12 | [TK]D-Fender | ~pb |
| 21:47.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 21:47.33 | haps | [TK]D-Fender: there's just the one line, unless you want me to turn on sip debugging too? |
| 21:47.42 | *** join/#asterisk double_cheesburg (n=chatzill@ip68-98-36-177.ph.ph.cox.net) |
| 21:47.43 | [TK]D-Fender | jchonig: haps there is MORE |
| 21:47.54 | [TK]D-Fender | haps there is MORE |
| 21:48.07 | [TK]D-Fender | jchonig: And is that not what you are looking for? |
| 21:48.33 | jchonig | I wanted to the dial command to set SIP/ext-jch |
| 21:49.19 | [TK]D-Fender | jchonig: Please learn to pastebin this stuff to. |
| 21:49.35 | jchonig | OK |
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| 21:57.30 | haps | [TK]D-Fender: what debugging options should I have on in the CLI? |
| 21:58.31 | [TK]D-Fender | haps: You telling me you see NO CALL to Voicemailmain? No output of the sounds its playing? NOTHING? |
| 21:58.56 | haps | unless i have sip set debug |
| 21:58.59 | haps | otherwise no |
| 21:59.10 | [TK]D-Fender | haps: then you don't even have BASIC VERBOSE up yet |
| 21:59.10 | *** join/#asterisk SparFux (n=raoul@e182020143.adsl.alicedsl.de) |
| 21:59.12 | [TK]D-Fender | &#$ |
| 21:59.18 | [TK]D-Fender | haps: "core set verbose 10" |
| 21:59.25 | [TK]D-Fender | haps: You are running completely blind |
| 21:59.30 | haps | yup |
| 21:59.35 | haps | thanks for that hint |
| 21:59.56 | haps | was watching dtmf via the sip debug but there is a *lot* of output |
| 22:00.05 | haps | sorry for being a dumbass n00b |
| 22:00.18 | [TK]D-Fender | haps: Want to prove DTMF, jsut doo a bloody READ. |
| 22:02.26 | haps | oooh dude, your advice is fantastic |
| 22:02.41 | haps | now I see that it's looking at 'default' context instead of 'home' context |
| 22:03.03 | [TK]D-Fender | haps: as I figured baseed on your call to Voicemailmain <- |
| 22:03.09 | [TK]D-Fender | haps: you have to specify where |
| 22:06.05 | [TK]D-Fender | ok, out for a while. |
| 22:06.10 | [TK]D-Fender | BBL |
| 22:06.14 | haps | thanks |
| 22:06.33 | *** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar) |
| 22:08.46 | haps | Maybe someone here could help me narrow this down further: |
| 22:08.47 | haps | http://pastebin.com/d4a838355 |
| 22:12.05 | jaytee | it's not passing a mailbox number |
| 22:12.20 | jaytee | how about a pb of extensions.conf |
| 22:12.53 | haps | exten => 700,1,VoiceMailMain(@home) |
| 22:13.13 | haps | i included '101@home' and tried that but got the same result |
| 22:13.43 | haps | but it's weird because in sip.conf i have mailbox=101@home and it flashes now because there's a message |
| 22:14.38 | haps | oooh i did a 'reload' and it's working now |
| 22:14.58 | haps | thanks all |
| 22:15.19 | SparFux | ah, was looking for it. But ok, if it now works :-) |
| 22:15.37 | SparFux | I did the sip mailbox stuff to have my analogue phone connected via ATA display the MWI :-) |
| 22:16.12 | haps | nice |
| 22:16.37 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:16.40 | haps | working through these configs is amazingly rewarding |
| 22:16.59 | haps | i feel like i just level up'd - new skill: voicemail! |
| 22:17.12 | jaytee | yeah, especially rewarding when you COMMIT the changes :-) |
| 22:17.45 | *** join/#asterisk doolph (n=doolph@190.141.68.31) |
| 22:17.47 | SparFux | still can't track down the callerid problem with the missing leading zero and sometimes 49 instead of zero. Anyway, it seems to be a lcr problem. |
| 22:19.27 | *** join/#asterisk jcoffi (n=jcoffi@190.213.22.5) |
| 22:20.07 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 22:23.24 | doolph | if I want to record a call when dial *1 is not working... |
| 22:25.32 | SparFux | doolph: there is a Dial() option which allows for recording. Have you set that? |
| 22:25.32 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 22:25.33 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:26.59 | double_cheesburg | I'm getting errors when I dial into my box. I've been putting the errors into search queries but don't seem to get much |
| 22:27.01 | double_cheesburg | http://pastebin.com/m4c3d5670 |
| 22:27.09 | double_cheesburg | Can anyone advise on this? |
| 22:28.52 | jaytee | do you have an extension s with priority 1 in the [from-zaptel] context in your extensions.conf file? |
| 22:29.11 | *** join/#asterisk asdf- (n=x@unaffiliated/asdf-) |
| 22:29.13 | asdf- | can anyone recommend an IAX2 compatible softphone for windows? |
| 22:30.23 | jaytee | Zoiper, MediaX, take your pick |
| 22:30.48 | asdf- | thanks... didnt care for zoiper's setup... will try mediax |
| 22:31.31 | *** join/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 22:35.24 | dan__t | Hello. |
| 22:36.43 | SparFux | hi |
| 22:36.44 | double_cheesburg | jaytee: nope |
| 22:36.59 | double_cheesburg | I didn't know I need a [from-zaptel] context |
| 22:37.01 | SparFux | anyone using linux-call-router? |
| 22:40.01 | SparFux | If a mobile calls, the callerid should be 0177... if this is the number of the mobile, right? |
| 22:40.42 | SparFux | Because I get this log: [Mar 8 23:28:48] NOTICE[3497]: chan_lcr.c:750 lcr_in_setup: [call=10 ast=NULL] Incomming setup from LCR. (callerid 17XXXXXXXXX, dialing XXXXXXX) |
| 22:40.59 | SparFux | the 17XXXXXXXXXX seems to be wrong! there is a 0 missing. |
| 22:41.04 | jchonig | Can anyone point me to how to diagnose this error message: |
| 22:41.06 | jchonig | <PROTECTED> |
| 22:41.06 | jchonig | Really destroying SIP dialog '47f5dded0625c65b77646d653f462401@172.25.29.121' Method: INVITE |
| 22:41.06 | jchonig | [Mar 8 18:40:18] WARNING[16849]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 22:41.34 | double_cheesburg | Starting simple switch on 'DAHDI/4-1' |
| 22:41.39 | double_cheesburg | What does this mean?? |
| 22:41.48 | double_cheesburg | I'm running my fxo channel on 3 |
| 22:44.16 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 22:45.29 | jchonig | Ah, a bogus qualify parameter.... |
| 22:49.47 | double_cheesburg | ?? |
| 22:58.35 | *** join/#asterisk pdfhacker (n=dd@38.104.98.118) |
| 22:59.24 | pdfhacker | Is there a free app that will let me do reasonably accurate, simple voice recognition? ("Press or say 1") |
| 23:01.14 | SparFux | There is http://www.isdn4linux.de/pipermail/isdn4linux/2009-January/003782.html this message and this: http://www.isdn4linux.de/pipermail/isdn4linux/2009-February/003788.html. And the guy seems to have solved the leading 0 problem, but I don't understand the second posting and what he exactly did better. Any idea? |
| 23:02.44 | double_cheesburg | http://pastebin.com/mc1a57ad |
| 23:02.46 | *** join/#asterisk luca`gervasi (n=ashura@host218-170-dynamic.16-87-r.retail.telecomitalia.it) |
| 23:02.48 | luca`gervasi | hello |
| 23:03.29 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 23:04.30 | luca`gervasi | i have some custom .gsm files. in asterisk 1.4 i copied them in /var/lib/asterisk/sounds. Doing the same copy-paste on my new 1.6 installation gives no results. is there a way to understands where asterisk is looking for the sound file i call through "PLAY" ? |
| 23:06.50 | pdfhacker | luca: If you know the name of a file that it can play, you find search to find the directory the working file is in... |
| 23:07.45 | luca`gervasi | ...i created the file, i know its name :) but i need to find the directory where asterisk looks for it, because it doesn't get played |
| 23:08.29 | drmessano | luca`gervasi: Permissions, perhaps |
| 23:08.42 | luca`gervasi | all the permissions seems correct |
| 23:11.08 | luca`gervasi | good night everyone :D |
| 23:11.14 | luca`gervasi | goes to sleep |
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| 23:12.04 | asdf- | i have a 450mhz computer with 512mb of ram... and i have a 1ghz computer with 512mb but with 500gb.... i want to run pfsense, NAS and asterisk... should i put asterisk and the nas on the same server? |
| 23:12.09 | asdf- | and pfsense on the 450mhz? |
| 23:12.27 | asdf- | or just jam it all together on the same server? |
| 23:14.28 | pdfhacker | Assuming low call volume, just use Asterisk on the 450mhz computer. Doesn't use all that many resources unless your transcoding lots of audio |
| 23:14.53 | asdf- | and pfsense and freenas on the other computer? |
| 23:15.05 | haps | asdf-: i run pf, nfs, and wifi (AP) on a 500mhz p3 |
| 23:15.35 | haps | i'd run * on there too but the p3 bus is slow and it's not good at dealing with high-speed ssh traffic |
| 23:15.38 | rob0 | If you're planning to have a Digium DAHDI card, you might have trouble on older machines. |
| 23:15.58 | asdf- | yeah... 4 maximum concurrent calls |
| 23:16.02 | haps | but you could run all that on 1ghz, assuming it's a 'home network' setup |
| 23:16.18 | haps | nice thing about pf is you can use altq |
| 23:16.29 | haps | but if you are new to the game i'd start with * on a separate box |
| 23:16.34 | asdf- | haps, are you running pf or pfsense? |
| 23:16.36 | haps | just until you get the router configured as you like it |
| 23:16.44 | haps | runs pf on freebsd7.1 |
| 23:16.48 | *** part/#asterisk tehfox (n=tehfox@adsl-dyn-165.95-102-35.t-com.sk) |
| 23:16.48 | rob0 | (the telephony cards need PCI v. 2.something, which older motherboards might not support.) |
| 23:16.59 | asdf- | i am not going to use telephony cards |
| 23:17.02 | haps | pfsense is a distro, pf is the firewall |
| 23:17.13 | haps | pfsense has all this gui crap afaik |
| 23:18.27 | *** join/#asterisk maddog01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 23:18.41 | asdf- | yeah, i am trying to get everything setup asap... i dont have the time to tweak every little thing |
| 23:20.03 | *** part/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 23:21.06 | asdf- | haps, rob0 & pdfhacker... thanks for the help |
| 23:24.08 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:24.08 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:26.44 | *** join/#asterisk jjshoe (n=jjshoe@CPE-69-23-98-163.new.res.rr.com) |
| 23:28.55 | dan__t | What's goin' on |
| 23:29.32 | SparFux | I did it! |
| 23:30.44 | SparFux | now my lcr doesn't omit the leading zero for incoming calls. |
| 23:33.28 | dan__t | heh |
| 23:34.06 | SparFux | Whatever reason there is for it to remove it. |
| 23:34.10 | dan__t | So, still got some lingering questions about AGI. Do I understand this correctly that I could make Asterisk speak AGI to a web application that I have set up beind Apache or lighty, and work in that manner? |
| 23:36.03 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 23:36.10 | pdfhacker | dan__t: essentially AGI lets you transfer control of a call to any command (or -- in the case of fastagi -- tcp/ip port) |
| 23:37.16 | dan__t | I understand what AGI is, sure |
| 23:37.28 | dan__t | Just wondering if my proposed practice was an acceptable route |
| 23:37.43 | dan__t | This new Asterisk AGI Programming book is really bad-ass, but its not entire clear on that. |
| 23:37.47 | pdfhacker | You could make it work -- have you looked at the AGI libraries available? |
| 23:38.08 | dan__t | PHPAGI and friends? Yep. |
| 23:38.19 | dan__t | Just, like I said, wanting to do this over HTTP as much as I can |
| 23:38.20 | pdfhacker | I'd recommend using one of those |
| 23:38.22 | dan__t | Well. Kindof. |
| 23:38.31 | dan__t | I can just set the content type or something. |
| 23:38.34 | possy | is away: Doing girls stuff |
| 23:39.34 | pdfhacker | dan__t: AGI doesn't operate over HTTP -- that's another layer of abstraction that you'd have to deal with |
| 23:39.47 | dan__t | I understand that much as well. |
| 23:39.59 | pdfhacker | dan__t: so why do you want to do it? |
| 23:40.01 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:40.02 | dan__t | Guess I should just stick to like xinetd as proposed in this book. |
| 23:40.27 | dan__t | Because I can debug 100x better when developing AGI that runs in that manner. |
| 23:41.24 | pdfhacker | Okay. For the sake of simplicity and speed I'd recommend you strongly consider a FastAGI server though |
| 23:41.43 | dan__t | Oh, there are specific FastAGI servers? |
| 23:41.50 | dan__t | Like purpose-built FastAGI servers? |
| 23:41.59 | pdfhacker | http://www.voip-info.org/wiki/view/Asterisk+FastAGI |
| 23:42.36 | dan__t | Ooh I misunderstood. |
| 23:42.37 | dan__t | Ok. |
| 23:42.38 | pdfhacker | Personally I use Asterisk.NET , which works really well, though I've had to make more than a couple updates (I guess I should submit the patches...) |
| 23:43.34 | dan__t | Awesome, Firefox just took a shit. |
| 23:43.58 | drmessano | That never happens |
| 23:44.03 | drmessano | Firefox is perfect |
| 23:44.04 | dan__t | I guess when it comes down to it, something as simple as xinetd still might work just fine. |
| 23:44.16 | dan__t | Perfectly imperfect. |
| 23:44.42 | drmessano | brings in the FBI (Firefox Bureau of Intelligence), clubs dan__t, drags him off, and removes all evidence of his existence |
| 23:44.44 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 23:44.46 | drmessano | Nothing to see here |
| 23:44.57 | drmessano | Move along |
| 23:45.04 | dan__t | heh |
| 23:45.08 | drmessano | Firefox = perfect |
| 23:45.13 | drmessano | Carry on |
| 23:45.15 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-237-246.lns10.mel6.internode.on.net) |
| 23:45.29 | jblack | I wish firefox were perfect. |
| 23:45.39 | drmessano | huffs |
| 23:45.39 | *** join/#asterisk sergey (n=sergey@sergey-home.iks.ru) |
| 23:45.50 | drmessano | brings in the FBI (Firefox Bureau of Intelligence), clubs jblack, drags him off, and removes all evidence of his existence |
| 23:46.00 | drmessano | Really, nothing to see here.. again |
| 23:46.11 | jblack | damn. outfoxed. again. |
| 23:46.42 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:46.56 | drmessano | Oh screw it |
| 23:47.17 | jblack | beastiality? |
| 23:47.37 | drmessano | drags the Asterisk devs off to a secret bunker (the ones we like, anyway), locks the #asterisk doors and chains them, and sets the room on fire |
| 23:47.56 | drmessano | Nothing to see here.. Firefox is perfect |
| 23:48.10 | drmessano | Now to go take care of FreeBSD.. those guys THINK they found a bug.. |
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