IRC log for #asterisk on 20090622

`00:05.52`	`*** join/#asterisk salzh (n=Administ@61.87.216.5)`
`00:17.30`	`*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk)`
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`00:37.32`	`brunner1`	`does anyone know roughly how much a PRI might cost from AT&T?`
`00:41.30`	`*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)`
`00:42.51`	`russellb`	`brunner: somewhere between $10 and $10000 a month, I would guess`
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`00:43.08`	`brunner`	`russellb: well thanks a lot.`
`00:43.12`	`russellb`	`np`
`00:43.33`	`russellb`	`I have no idea. I just write code...`
`00:43.34`	`brunner`	`russellb: have you ever heard of someone paying $10 or $10000 for a PRI from AT&T?`
`00:43.48`	`russellb`	`nope, usually somewhere in the middle`
`00:44.27`	`coppice`	`$4995 sounds a lot`
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`00:45.38`	`russellb`	`indeed`
`00:45.41`	`russellb`	`waves to coppice`
`00:47.12`	`russellb`	`hm, didn't mean to scare him off.`
`00:53.39`	`brunner`	`how long does it take for CNAM to come through on a PRI?`
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`01:24.17`	`vousb`	`morning folks`
`01:25.36`	`javb`	`Is it true that there are rumors that Cisco have showed some kind of interest on doing business with Digium?`
`01:26.45`	`hardwire`	`sigh`
`01:28.28`	`hardwire`	`res_odbc preconnects fine.. using config name asterisk.. dsn liberty`
`01:28.44`	`hardwire`	`I can do realtime update name george port 666 and have it update the database`
`01:28.56`	`hardwire`	`iaxpeers and iaxusers should be set up fine.. but I can't seem to pull up a specific peer from the db`
`01:29.01`	`hardwire`	`it doesn't even query the db.`
`01:29.20`	`hardwire`	`anybody have this problem with debian lenny + 1.4.21 packages with ODBC?`
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`01:47.56`	`*** join/#asterisk acone_ (n=acone@216.104.45.109)`
`01:48.39`	`acone_`	`can anyone recommend any software for parsing the CDR table in mysql? i am trying to generate statistics about calls. any help is appreciated.`
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`01:52.06`	`ISO9001`	`dunno, most people I know who have that need have just hacked together something with perl to handle reporting.`
`01:53.35`	`acone_`	`ISO9001: thanks. i would be happy to write something myself, but i can't for the life of me find anywhere the cdr table is documented`
`01:53.55`	`acone_`	`i am new to asterisk, and i really dont know which end it up. i just want to generate call stats for my company.`
`01:54.59`	`[TK]D-Fender`	`acone_: funny there are plenty of obvious docs in the source tarball....`
`01:55.16`	`acone_`	`of asterisk?`
`01:55.28`	`[TK]D-Fender`	`acone_: What else do you think?`
`01:55.30`	`joobie`	`TK.. duno if you recall about a week ago I floated a question about if the AMI can report on the longest holdtime for an agent`
`01:55.40`	`joobie`	`-agent +queuemember`
`01:55.56`	`joobie`	`is this only possible to do via editing the app_queue.c code?`
`01:55.57`	`acone_`	`i would expect to find such documentation online. i'm happy to look in the asterisk source tarball`
`01:56.17`	`[TK]D-Fender`	`joobie: Queue members don't have hld times. Queue CALLERS have hold time`
`01:56.45`	`joobie`	`yer sry, queue callers`
`01:56.57`	`joobie`	`currently it only reports the average holdtime`
`02:00.42`	`[TK]D-Fender`	`joobie: "queue show" <- parse it`
`02:01.11`	`hardwire`	`bangs head against wall with odbc and iaxpeers/users`
`02:01.16`	`hardwire`	`it's not even querying the db!`
`02:01.29`	`ISO9001`	`acone_: presumably you could just look at the table schema.`
`02:01.55`	`acone_`	`ISO9001: i have looked at it, but i dont understand what is going on`
`02:01.55`	`hardwire`	`it connects when res_odbc loads.. then just sorta forgets to do anything else. cdr_odbc working fine of course.. res_config_odbc/res_odbc being preloaded as well.`
`02:01.58`	`hardwire`	`wants to die`
`02:02.46`	`ISO9001`	`acone_: which part?`
`02:03.03`	`acone_`	`ISO9001: well, how, for example, would i calculate average hold time`
`02:03.35`	`acone_`	`ISO9001: I want to reconstruct what happened for each call: first he called, then he dialed 3, then he got transferred to tech support, then he held for 2 minutes, etc`
`02:03.40`	`[TK]D-Fender`	`acone_: there is no hold time for CDR`
`02:03.52`	`acone_`	`TK: where would I find this information`
`02:04.00`	`[TK]D-Fender`	`acone_: And there is typically 1 line per CALL, not "pre transaction"`
`02:04.18`	`[TK]D-Fender`	`acone_: What you are looking for does not exist <--`
`02:04.36`	`ISO9001`	`you might be able to piece that together from the management interface, maybe, but I don't see that being pretty.`
`02:04.52`	`acone_`	`TK: I see. i have seen multiple lines in the DB that seem to be for the same call, though. any idea what that is? i can give you an example if you like`
`02:05.04`	`[TK]D-Fender`	`acone_: Might help`
`02:05.11`	`acone_`	`1 sec lemme dig it up`
`02:06.19`	`acone_`	`ok, i have it, but it's a lot of text (3 records, 20ish columns); is it ok if i paste it here`
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`02:06.38`	`[TK]D-Fender`	`~pb`
`02:06.38`	`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/`
`02:06.53`	`buttons840`	`would it be possible (and has anyone achieved) a voip phone which supports and unlimited number of lines?`
`02:07.10`	`buttons840`	`a softphone i mean`
`02:07.13`	`Docteh`	`hardware or software?`
`02:08.02`	`buttons840`	`Docteh, software`
`02:08.08`	`acone_`	`http://pastebin.ca/1469391`
`02:08.14`	`[TK]D-Fender`	`buttons840: Functionally sure. Whats your point?`
`02:08.43`	`joobie`	`TK, what about from AMI?`
`02:08.51`	`joobie`	`trying to avoid asterisk CLI to pull this out..`
`02:09.00`	`buttons840`	`well, i'm just looking for a sip softphone which support many lines, i have tried one called twinkle that supports two lines, but the more the better`
`02:09.44`	`[TK]D-Fender`	`buttons840: how many?`
`02:09.56`	`[TK]D-Fender`	`joobie: You can get CLI from AMI`
`02:10.30`	`[TK]D-Fender`	`acone_: those all look like separate calls`
`02:10.49`	`[TK]D-Fender`	`acone_: Spawned from a multi-dial perhaps`
`02:10.50`	`buttons840`	`[TK]D-Fender, what's the most lines your aware of?`
`02:11.25`	`[TK]D-Fender`	`buttons840: 6 w/ eyebeam`
`02:11.32`	`buttons840`	`with the scalability of softphones, I'm supprised nobody has implemented unlimited line support; of course bandwith would impose a limit, but the software itself wouldn't`
`02:11.38`	`acone_`	`TK: i see. thanks. you mentioned earlier that what i was looking for didn't exist. what is the best way to get as much of that as possible`
`02:11.44`	`[TK]D-Fender`	`buttons840: How many do you need, and why do you believe you need so many?`
`02:12.37`	`[TK]D-Fender`	`acone_: Your inbound channel calling outbound channels each generate their own CDR. 1 record per "dial"-generated call. Thre is no transactional tracking.`
`02:13.03`	`[TK]D-Fender`	`buttons840: Who needs 500 accounts on a stupid soft-phone? Whats the point?`
`02:13.26`	`acone_`	`TK: so there is no way, say, to determine avg hold time?`
`02:13.29`	`*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)`
`02:14.11`	`[TK]D-Fender`	`acone_: There is absolutely NO 'hold-time" concept in CDR <-`
`02:14.13`	`buttons840`	`not accounts, lines, (is there a difference?) what i'm trying to do is stress test my asterisk box by opening and many active calls as possible, I don't have a large number of computers (just 1, maybe 2) to receive the calls`
`02:14.23`	`buttons840`	`thus, 2 active calls max is hardly a test`
`02:14.40`	`acone_`	`TK: i understand that there is no such thing in CDR, but is there any way i can get that information... perhaps through some other DB or log`
`02:14.48`	`[TK]D-Fender`	`buttons840: not "lines", but rather "SIMULTANEOUS CALLS"`
`02:15.07`	`[TK]D-Fender`	`acone_: The information does not exist. Anywhere.`
`02:15.48`	`Docteh`	`acone_: add timestamps to debug information? ;)`
`02:16.12`	`acone_`	`TK: fair enough. is there an API by which i could write something to track it? or would that require completely rewriting asterisk? i'm pretty comfortable in C, and i'm willing to invest some time in this`
`02:17.00`	`acone_`	`i suspect this functionality would be useful to many asterisk users, not just my company. every service-oriented company should want to know stats about its hold times`
`02:17.01`	`[TK]D-Fender`	`acone_: There is nothing to track. This iformation DOES NOT EXIST. There is NOWHERE to look for it. *'s ability to track ANYTHING sucks as it is, and this aspect does not exist anywhere.`
`02:17.25`	`joobie`	`TK, which Action lets you access the CLI from AMI? Cheers man.. good idea`
`02:17.57`	`[TK]D-Fender`	`joobie: get off your ass and read the damn Action: list :p`
`02:18.12`	`Docteh`	`acone_: so you're trying to track basically how long someones in moh while waiting to talk to a person?`
`02:18.45`	`acone_`	`yes`
`02:19.12`	`Docteh`	`sounds like the moh system is where to start then`
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`02:19.23`	`acone_`	`do you think i will have to edit the c code?`
`02:19.44`	`Docteh`	`well based soley on Fender saying the functionality doesn't exist, yes`
`02:20.08`	`buttons840`	`[TK]D-Fender, can you point me in the right direction, is it possible to set up a simulation of a large phone system, without actually having a large phone system?`
`02:20.28`	`Docteh`	`hrm`
`02:20.30`	`[TK]D-Fender`	`buttons840: sipp <-`
`02:20.52`	`[TK]D-Fender`	`acone_: You'll have to rewrite *`
`02:21.21`	`joobie`	`fuk TK.. what is the world coming to.. all this "self-help" crap`
`02:21.21`	`joobie`	`;P`
`02:21.22`	`joobie`	`ok`
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`02:21.28`	`acone_`	`could i put something in mohstream.sh`
`02:21.53`	`acone_`	`i'm thinking that one way to do this is to use the command invoked to play the music`
`02:22.16`	`joobie`	`TK one more Q.. im trying to find a way to clear stats in call queues (specially the AMI QueueStatus Action) .. it tends to build up the stats over a period of itme.. if i do a reload of the module, it flushes it, but sounds harsh.. even asterisk -rx reload flushes it.. but sounds harsh too.. any clean way you can see to do it in 1.4 ?`
`02:22.25`	`joobie`	`thing like "Abanonded calls" for example...`
`02:22.28`	`Docteh`	`acone_: well how does the moh stop? is the process brutally killed by asterisk, or something nicer?`
`02:22.34`	`[TK]D-Fender`	`joobie: Can't help you on that one...`
`02:22.57`	`acone_`	`i'm really not sure. if it's SIGKILLED, then i could do something silly with tracking process numbers`
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`02:23.08`	`acone_`	`but presumably it's SIGHUPed`
`02:23.19`	`joobie`	`ahh k.. thanks neawy.. from what i've read so far 1.6 has a stats reset cmd that you can run.. 1.4 looks like a relaod .. ergh - will let you know if i come across anything`
`02:23.21`	`joobie`	`Cheers`
`02:25.17`	`[TK]D-Fender`	`joobie: Anything in particular holding you back from 1.6?`
`02:26.52`	`joobie`	`just the hassle of a rebuild`
`02:27.20`	`joobie`	`i'm happy with 1.4 overall.. dont really have much pushing to go to 1.6 at the moment.. this is kinda the first issue i've hit`
`02:27.45`	`[TK]D-Fender`	`joobie: rebuild?most key configs port over pretty driect`
`02:29.05`	`joobie`	`i cbf to be honest... have customized my app_queue.c a little which would need to be ported, need to setup my isdn card again too.. and schedule downtime, etc etc.. all the effort to do it doesn't outweigh the beneifits (which for now is just this clear stats)`
`02:29.16`	`joobie`	`im thinking ill just crontab a reload at midnight or something..`
`02:30.33`	`[TK]D-Fender`	`joobie: Sounds like a good reason.; Are your mods in C too personal to propose as permanent patches?`
`02:33.56`	`joobie`	`Not really.. It's just a custom action that gives me a refined result for waht i want to report against.. Didn't think it would be that useful to post because all the info is accessible in existing Actions, just spread across multiple actions...`
`02:39.11`	`[TK]D-Fender`	`joobie: So you AMI broadcast a status update instead of making a client poll for it?`
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`02:48.07`	`joobie`	`yea.. and also refine the data coming back from the braodcast to be just waht i need for my app`
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`05:07.41`	`VaGoNeTaS`	`hi everybody`
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`05:17.53`	`Dovid`	`Good morning ev1`
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`05:35.51`	`acone_`	`hi. i'm trying to get asterisk to use a custom app for musiconhold`
`05:36.16`	`acone_`	`i am trying to use musiconhold.conf , but i cant get it to invoke the application`
`05:36.27`	`acone_`	`mode=custom`
`05:36.27`	`acone_`	`dir=/var/lib/asterisk/mohmp3-empty`
`05:36.27`	`acone_`	`application=/etc/asterisk/mohstream.sh`
`05:36.34`	`acone_`	`any suggestions?`
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`05:58.38`	`aksyn`	`morning :)`
`06:00.53`	`aksyn`	`my company is releasing a (fully indemnified) g729 codec tomorrow for Asterisk & FreeSWITCH (30-50% more channels than Digium, cheaper and more flexible licensing) - if anyone has some spare time, i'd love you to grab a trial copy from http://www.howlertech.com/products/howlets/ and give it a spin (let us know if you have any problems with it, of course!)`
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`06:11.49`	`acone_`	`<PROTECTED>`
`06:11.49`	`acone_`	`<PROTECTED>`
`06:12.05`	`acone_`	`any help is appreciated`
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`07:08.50`	`WeazelON`	`hey guys, does anyone know how to get rid of the annoying " -- Remote UNIX connection -- Remote UNIX connection disconnected " Messages when I activate the Verbose ?`
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`07:13.03`	`Kumba_`	`Anyone ever configured Asterisk with CBeyond's new SBC?`
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`07:41.48`	`dacs`	`all sleepy or what? :)`
`07:41.55`	`angryuser`	`depends`
`07:42.24`	`dacs`	`yeah just been here for a bit and its quite`
`07:42.59`	`angryuser`	`WeazelON, i dont think so`
`07:43.42`	`angryuser`	`dacs, well most of ppls here are from usa i think so they are sleepeing now`
`07:44.09`	`dacs`	`angryuser: am from USA and i am not sleeping`
`07:44.22`	`dacs`	`:P just working a stupid night shift lol`
`07:44.28`	`angryuser`	`dacs, coz you are geek xD`
`07:44.55`	`dacs`	`angryuser: nope`
`07:45.01`	`angryuser`	`dacs, at least they are paying you more ?`
`07:45.56`	`dacs`	`angryuser: yep 15% more , hell i will take that`
`07:46.10`	`dacs`	`angryuser: you at work`
`07:46.25`	`angryuser`	`dacs, i europe its +25% i think (fr)`
`07:46.48`	`angryuser`	`dacs, yes, but day shift`
`07:47.32`	`dacs`	`aha , where u from`
`07:47.54`	`angryuser`	`dacs, so what are you doing at night ?`
`07:48.04`	`angryuser`	`France`
`07:48.28`	`dacs`	`pretty girls eh`
`07:48.29`	`dacs`	`:)`
`07:48.53`	`dacs`	`i work for a mobile company.`
`07:49.29`	`angryuser`	`dacs, pretty girls, bad caracters, yes`
`07:49.33`	`*** join/#asterisk DarkRift (n=dark@65.92.171.252)`
`07:49.59`	`dacs`	`hahaha`
`07:50.01`	`angryuser`	`dacs, well not bad but capricious`
`07:50.38`	`dacs`	`angryuser: we call it here in USA PMSing`
`07:51.34`	`angryuser`	`dacs, PMsing, i will google that`
`07:51.54`	`dacs`	`PMS - a syndrome that occurs in many women from 2 to 14 days before the onset of menstruation`
`07:52.26`	`dacs`	`they will be in a really bad mode`
`07:52.41`	`angryuser`	`dacs, http://www.youtube.com/watch?v=-zb-XyXP4hk xD`
`07:53.02`	`dacs`	`can't watch that here at work`
`07:53.57`	`angryuser`	`dacs, why is that ?`
`07:54.19`	`dacs`	`its blocked`
`07:56.07`	`creativx`	`haha`
`07:56.10`	`*** join/#asterisk war9407 (i=war@liquidswords.org)`
`07:56.11`	`creativx`	`classic pms`
`07:56.15`	`creativx`	`that girl is stable.`
`07:59.52`	`*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)`
`08:03.57`	`*** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex)`
`08:10.05`	`angryuser`	`dacs, have you tryed cgi proxies ?`
`08:10.23`	`dacs`	`angryuser: no`
`08:13.57`	`*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)`
`08:17.05`	`*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)`
`08:19.20`	`*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)`
`08:23.30`	`Kumba_`	`Does the asterisk mysql-cdr need select,insert priviledges?`
`08:23.54`	`barbacha`	`Kumba_: how would it store your CDR into the DB without it ?`
`08:23.58`	`barbacha`	`Kumba_: so YES`
`08:24.16`	`*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)`
`08:24.18`	`Kumba_`	`well, it might use update or lock tables depending upon how the CDR mechanism works`
`08:24.46`	`Kumba_`	`So, I figured i'd ask incase someone was more familiar with it then I`
`08:25.18`	`barbacha`	`in that case my answer is "i don't know". I grant ALL to * on the CDR`
`08:25.20`	`barbacha`	`why bother`
`08:26.18`	`barbacha`	`~Ã¨~_____________`
`08:26.26`	`Kumba_`	`Why give more access then needed...`
`08:27.35`	`*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)`
`08:27.50`	`barbacha`	`(the last line was my cat typing)`
`08:29.42`	`*** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com)`
`08:47.18`	`*** join/#asterisk Silicium (n=Silicium@2001:bf0:c080:200:0:0:0:23)`
`08:47.20`	`Silicium`	`hi there`
`08:47.44`	`Silicium`	`iam searching for a small tapi linux client, for Asterisk or Snom Phones`
`08:48.36`	`Silicium`	`s/tapi/cti`
`08:49.22`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d6730efd1364aea3)`
`08:49.56`	`ISO9001`	`barbacha: if you're using the cdr for billing you might not want the asterisk user to be able to delete lines...`
`08:59.34`	`Kumba_`	`And it makes injection attacks harder if all they can do is inserts`
`09:11.06`	`acone_`	`<PROTECTED>`
`09:11.06`	`acone_`	`<PROTECTED>`
`09:11.06`	`acone_`	`<PROTECTED>`
`09:17.33`	`*** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001)`
`09:25.19`	`*** join/#asterisk AlmightyOatmeal (n=jamie@66-190-59-171.dhcp.mdsn.wi.charter.com)`
`09:28.13`	`*** join/#asterisk intralanman (n=lanman@173-97-82-11.pools.spcsdns.net)`
`09:35.01`	`AlmightyOatmeal`	`grrr, * refuses to register with my sip provider`
`09:35.33`	`AlmightyOatmeal`	`it keeps timing out and doesn't even try to hit my sip provider's proxy`
`09:35.54`	`AlmightyOatmeal`	`and it _just_ started happening when i switched proxies`
`09:36.18`	`AlmightyOatmeal`	`even when i switched back to the old one it still failed and i've reset every network device between my asterisk box and the net`
`09:37.05`	`AlmightyOatmeal`	`my sip provider did see me getting 401 errors for some reason, but my register syntax is spot on and so is my username/password which support verified and tested..`
`09:37.31`	`AlmightyOatmeal`	`i can register manually from a softphone on my laptop but asterisk refuses to register.. _any_ help would be appreciated`
`10:01.41`	`*** join/#asterisk salzh (n=Administ@122.144.138.5)`
`10:04.06`	`dacs`	`where can i locate doc/sip-retransmit.txt`
`10:04.07`	`ickmund`	`if I do a Dial(IAX2/trunkname/extension), how do asterisk know where to send this?`
`10:05.04`	`AlmightyOatmeal`	`dacs: find / -name sip-retransmit.txt`
`10:05.13`	`ickmund`	`In iax.conf, I have a [trunkname], but this doesn't specify an IP?`
`10:05.31`	`dacs`	`shouldn't you define it in your extension.conf and iax.conf ickmund`
`10:07.21`	`ickmund`	`dacs, in extensions.conf I have something like exten => _123,1,Dial(IAX2/trunkname/extension)`
`10:07.53`	`AlmightyOatmeal`	`where you defined 'trunkname' should be where the host/ip address settings are`
`10:09.12`	`ickmund`	`AlmightyOatmeal: I'm looking at the Asterisk book. host is defined as dynamic. Still it works? :P`
`10:09.30`	`AlmightyOatmeal`	`so?`
`10:09.38`	`AlmightyOatmeal`	`my hosts are dynamic as well and i still work`
`10:09.42`	`ickmund`	`AlmightyOatmeal: I'm trying to figure out why it works...`
`10:09.46`	`AlmightyOatmeal`	`well did until last night :P`
`10:10.02`	`dacs`	`ickmund: i am not an expert in * , _123 means anything [0-9]123 extension show be Dial using IAX.. no!`
`10:10.04`	`ickmund`	`AlmightyOatmeal: The question is, if the host is dynamic, how does * know where to send the call?`
`10:10.33`	`AlmightyOatmeal`	`ickmund: you must have some kind of hostname or ip address setup unless you have some magick going on`
`10:10.56`	`ickmund`	`AlmightyOatmeal: Exactly! So, what is it that defines that hostname/IP?`
`10:11.10`	`ickmund`	`Is it the register line I have right above [trunkname]?`
`10:11.15`	`dacs`	`ickmund: what is the question again please`
`10:11.38`	`*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)`
`10:11.50`	`ickmund`	`dacs, When defining a IAX trunk, what is the directive to define the hostname/IP that the calls should be forwarded to?`
`10:12.52`	`*** join/#asterisk KnowWhat (n=KnowWhat@119.153.24.35)`
`10:12.53`	`AlmightyOatmeal`	`ickmund: the directives themselves are in the book, as per your question of why is it working when you haven't configured those, its probably relying on data from when you registered`
`10:12.55`	`KnowWhat`	`hey`
`10:13.29`	`KnowWhat`	`i want some agi programming tutorials in php`
`10:13.45`	`UQlev`	`ickmund: there must be 2 records: 1 for trunk and 2 for registry`
`10:14.54`	`AlmightyOatmeal`	`KnowWhat: meet my friend google! :D`
`10:14.56`	`ickmund`	`UQlev: Ok. Think I got it`
`10:15.15`	`KnowWhat`	`AlmightyOatmeal: hmm i met him thats why i came here`
`10:15.18`	`KnowWhat`	`i need something in detail`
`10:15.25`	`ickmund`	`UQlev: I define the [trunkname] on ServerA, then have ServerB register to ServerA, and thus ServerA know the IP?`
`10:15.29`	`AlmightyOatmeal`	`KnowWhat: meet my friend, a book!`
`10:15.42`	`KnowWhat`	`specially in writing to database, CDR`
`10:15.46`	`dacs`	`KnowWhat: here you go http://lmgtfy.com/?q=agi+programming+tutorials+in+php`
`10:15.55`	`AlmightyOatmeal`	`KnowWhat: actually i dont know of many good websites, you would probably need an ebook or purchase a book from a bookstore/amazon/barnes and noble`
`10:16.02`	`dacs`	`~lmgtfy`
`10:16.17`	`UQlev`	`ickmund: can you pastebin it w/o passwords?`
`10:16.21`	`dacs`	`AlmightyOatmeal: they should add this to the bot me think`
`10:16.36`	`AlmightyOatmeal`	`dacs: me agrees`
`10:16.54`	`ickmund`	`UQlev: I don't have a specific case, I'm trying to understand how it works...`
`10:17.28`	`dacs`	`ickmund: have you gave the book a slight thought!?`
`10:18.12`	`ickmund`	`dacs, c'mon man... I've already said I'm looking at the book and it's examples, and I don't get the underlying parts...`
`10:18.38`	`ickmund`	`I know how to copy / paste already ;)`
`10:18.39`	`wdoekes`	`wouldn't that mean you have too little knowlegde of php?`
`10:18.48`	`*** join/#asterisk CodeWork (n=Miranda@p5083BD22.dip.t-dialin.net)`
`10:19.25`	`AlmightyOatmeal`	`what can cause asterisk to attempt to register with an invalid username/passwd even though the register syntax is perfect with the correct information?`
`10:20.04`	`dacs`	`think he should continue reading the book`
`10:20.19`	`dacs`	`AlmightyOatmeal: might be behind NAT?`
`10:20.35`	`ickmund`	`What I really want to do right now is to have a one way trunk. On the caller side, I'll set up a [trunk] with all the bells and whistles. On the callee side, I do a register => trunk:pass@caller_sides_IP`
`10:20.39`	`AlmightyOatmeal`	`dacs: yes, but it literally suddenly stopped working for some reason`
`10:20.48`	`*** join/#asterisk chendy (n=chatzill@58.251.102.197)`
`10:21.52`	`dacs`	`AlmightyOatmeal: maybe its right in your sip but wrong in your extention.conf`
`10:22.16`	`AlmightyOatmeal`	`dacs: would that have an effect on initial registration?`
`10:23.02`	`dacs`	`AlmightyOatmeal: i am not sure i am just brain storming with you!`
`10:23.17`	`AlmightyOatmeal`	`oh, well extensions.conf is setup just fine, it doesn't reach that point yet`
`10:23.41`	`*** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001)`
`10:23.44`	`AlmightyOatmeal`	`i have a dynamic host setup for my ip address in sip.conf, specified nat=yes, and my local subnet`
`10:24.09`	`AlmightyOatmeal`	`all i did was change proxies last night and hell broke loose.. all i changed was /etc/hosts heh`
`10:24.42`	`dacs`	`ahhhh`
`10:24.58`	`dacs`	`did you do /init.d/networking restart`
`10:25.40`	`AlmightyOatmeal`	`my sip provider is telling me i'm getting 401 errors, and i'm saying that my register syntax is using the same credentials as my peer context, and i verified it with a softphone on my laptop... the softphone works flawlessly`
`10:25.57`	`AlmightyOatmeal`	`dacs: if i do that i will disconnect from here i'm sure`
`10:26.07`	`AlmightyOatmeal`	`i can give that a try, i'll be back`
`10:26.30`	`AlmightyOatmeal`	`and done`
`10:26.40`	`dacs`	`AlmightyOatmeal: do it, because i think it is still caching the old hosts`
`10:26.55`	`dacs`	`and that proves it because your internal network is working`
`10:27.40`	`AlmightyOatmeal`	`i dont see how that proves it, but i verified that /etc/hosts is working by pinging the host in there`
`10:28.01`	`AlmightyOatmeal`	`omfg`
`10:28.08`	`AlmightyOatmeal`	`dacs: i think i'm in love with you`
`10:28.17`	`AlmightyOatmeal`	`i think it registered on the first try`
`10:28.56`	`*** join/#asterisk decimalz (n=pbxk1064@203.171.196.201)`
`10:29.07`	`dacs`	`AlmightyOatmeal: well i think i am as dumb as it gets when it come to * But me think i am go with networking`
`10:29.26`	`dacs`	`think when ever you hear BUT , that means you are FUCKED!!`
`10:29.35`	`AlmightyOatmeal`	`dacs: i've been known to miss the most incredibly simple steps o:)`
`10:29.47`	`dacs`	`you are doing and great BUT`
`10:30.49`	`dacs`	`glad i was able to help AlmightyOatmeal ... may i suggest an advice?`
`10:31.01`	`AlmightyOatmeal`	`i'm always looking for advice`
`10:31.34`	`dacs`	`when ever you ever work with networking , always start at L1 and work your way up <- AlmightyOatmeal`
`10:32.07`	`acone_`	`<PROTECTED>`
`10:32.07`	`acone_`	`<PROTECTED>`
`10:32.07`	`acone_`	`<PROTECTED>`
`10:32.09`	`acone_`	`o`
`10:32.34`	`acone_`	`i've asked this a few times; i dont want to spam the channel. is tehre any more precise information i can give?`
`10:32.42`	`AlmightyOatmeal`	`dacs: i never would have thought that would cache stale data, i was going above and beyond :(`
`10:32.56`	`dacs`	`:)`
`10:34.30`	`AlmightyOatmeal`	`RTP is 10000-20000 right?`
`10:34.48`	`dacs`	`AlmightyOatmeal: you can check that rtp.conf`
`10:34.52`	`dacs`	`but yes`
`10:35.13`	`dacs`	`man where is [T]afender , i want to thank him so much`
`10:36.05`	`dacs`	`he forced me to read the book and i can see it already start benefiting me :)`
`10:39.58`	`*** join/#asterisk miloux (n=KVIrc@milu.rit.se)`
`10:41.58`	`miloux`	`how come my trunk = $AGI->get_variable('trunk_out'); doesnt return anything when i have trunk_out = SIP/test_1000 in dialplan (Its a global)`
`10:42.28`	`AlmightyOatmeal`	`did you sacrifice the required number of kittens first?`
`10:42.40`	`Kumba_`	`crap, I used puppies...`
`10:43.09`	`miloux`	`my $trunk`
`10:43.13`	`miloux`	`ofc`
`10:43.22`	`miloux`	`AlmightyOatmeal: yes :/`
`10:43.42`	`miloux`	`It should work...atleast thats what ive understod from the docs..`
`10:44.30`	`miloux`	`its actually returning as undef`
`10:45.00`	`*** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi)`
`10:47.33`	`*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)`
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`11:47.05`	`*** join/#asterisk ircnickiuse (n=xxx@gw1.mycosmos.gr)`
`11:47.18`	`ircnickiuse`	`is there a mirror for the latest celliax live cd that wasa released?`
`11:47.20`	`ircnickiuse`	`^`
`12:04.53`	`ircnickiuse`	`^^ celliac / skypiax live cd? (any live cd with skypiax / skype integration built in)`
`12:05.25`	`*** join/#asterisk Boardy (n=chatzill@kirakira.xs4all.nl)`
`12:05.55`	`Boardy`	`Hello`
`12:06.03`	`dandre`	`hello,`
`12:06.04`	`Boardy`	`A few days ago I was helped with setting up my MOH. It is working, but not when I call from an external line.`
`12:06.18`	`Boardy`	`Does anybody have a clue what can be the cause?`
`12:06.22`	`*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk)`
`12:06.32`	`dandre`	`I am trying to use immediate=yes in zapata.conf`
`12:07.09`	`*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)`
`12:07.22`	`dandre`	`how can I determine wich extension or device is using that feature in my dialplan?`
`12:08.17`	`dandre`	`or must I define a new context for each device usising immediate=yes?`
`12:08.52`	`dandre`	`the target usage is to predefine a dialed number for thos devices`
`12:09.17`	`[TK]D-Fender`	`dandre: You can do whatever you want.`
`12:09.44`	`dandre`	`how can I know the device then?`
`12:09.50`	`*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`12:09.50`	`*** mode/#asterisk [+o leifmadsen] by ChanServ`
`12:10.05`	`[TK]D-Fender`	`dandre: Considered looking at the CHANNEL?`
`12:11.01`	`dandre`	`I have tried this in the context:`
`12:11.17`	`dandre`	`exten => s/DID,1,...`
`12:11.28`	`dandre`	`sorry`
`12:11.39`	`dandre`	`exten => s/CID,1,...`
`12:11.52`	`[TK]D-Fender`	`dandre: That works if you have a unique CID per`
`12:11.53`	`dandre`	`but that doesn't do what I want`
`12:12.14`	`[TK]D-Fender`	`dandre: Have you already pastebinned your code & failed attempts?`
`12:13.19`	`dandre`	`no`
`12:13.36`	`[TK]D-Fender`	`dandre: We'll never know what little part you did wrong then`
`12:13.48`	`[TK]D-Fender`	`dandre: "it didn't work" doesn't tell us what your mistake was`
`12:14.05`	`[TK]D-Fender`	`dandre: And guessing is a waste of time. Go try something and show us`
`12:14.14`	`*** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr)`
`12:14.17`	`[TK]D-Fender`	`dandre: You should know better...`
`12:14.42`	`*** join/#asterisk simprix (n=simprix@cosmas.supportdept.com)`
`12:15.20`	`dandre`	`http://pastebin.fr/4870`
`12:16.20`	`*** join/#asterisk s14ck (n=s14ck@190-76-126-140.dyn.movilnet.com.ve)`
`12:16.42`	`[TK]D-Fender`	`dandre: and the reason I don't see your zapata.conf in there is...?`
`12:17.20`	`*** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br)`
`12:17.53`	`*** join/#asterisk Meaw (n=dino@188.161.190.185)`
`12:18.17`	`s14ck`	`Good morning *`
`12:18.32`	`[TK]D-Fender`	`dandre: And get rid of that white-space`
`12:19.02`	`coppice`	`isn't space generally black?`
`12:19.43`	`dandre`	`users.conf:`
`12:19.45`	`dandre`	`http://pastebin.fr/4871`
`12:19.58`	`[TK]D-Fender`	`coppice: Yes, it tends to get whiter the dumber the occupant is ;)`
`12:20.30`	`coppice`	`is that the hot heads creating thermal noise?`
`12:20.33`	`[TK]D-Fender`	`dandre: Speaking of "not-bright" you didn't bother settings the callerID <-`
`12:31.58`	`dandre`	`ok now with callerid defined in users.conf it is ok`
`12:32.01`	`dandre`	`thanks;`
`12:34.05`	`*** join/#asterisk Enkhmunkh (n=Enkhmunk@202.126.92.6)`
`12:36.03`	`Enkhmunkh`	`Hi guys, How do i do the measurement of the POTS line on the *?`
`12:36.20`	`Enkhmunkh`	`ring, busy etc cadence`
`12:37.37`	`*** join/#asterisk ariel_ (n=chatzill@63.214.236.169)`
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`12:57.29`	`*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)`
`12:59.23`	`*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)`
`13:01.14`	`beek`	`morning jaytee`
`13:01.25`	`*** join/#asterisk Morghus (n=j@88.82-134-68.bkkb.no)`
`13:01.40`	`Morghus`	`Good afternoon guys. Mind if I ask you for some help? =)`
`13:01.56`	`beek`	`~ask`
`13:01.56`	`infobot`	`Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will.`
`13:02.17`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:02.37`	`*** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled)`
`13:02.58`	`Morghus`	`Allright. Got an asterisk set up. All my phones receive the Caller ID and see the number of the caller, but my E51's don't for some reason. Is there some setting I'm missing on the server, or possibly the phones, to make it show up?`
`13:03.24`	`jaytee`	`morning beek`
`13:03.28`	`[TK]D-Fender`	`Morghus: pastebin your E51's sip peer masking only passwords.`
`13:03.30`	`[TK]D-Fender`	`~pb`
`13:03.31`	`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/`
`13:04.41`	`Morghus`	`I have no idea what you mean by "E51's sip peer"`
`13:04.56`	`*** join/#asterisk ingenius (n=alektro@netsolution.com.ar)`
`13:05.26`	`[TK]D-Fender`	`Morghus: Well how else did you set up your E51 with *?`
`13:05.32`	`*** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com)`
`13:05.36`	`[TK]D-Fender`	`Morghus: You ARE using its SIP client, no?`
`13:06.42`	`Morghus`	`Yes, but I don't understand the question :)`
`13:06.49`	`Morghus`	`*the request`
`13:07.38`	`[TK]D-Fender`	`Morghus: I asked you to pastebin your CONFIG for this sip peer.`
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`13:08.28`	`Morghus`	`You mean the settings on the phone, or the settings on the asterisk server?`
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`13:08.56`	`[TK]D-Fender`	`Morghus: * server ........`
`13:09.14`	`Morghus`	`Allright, hold on`
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`13:51.46`	`j_kroon`	`hi, is there anybody here who would be able to assist me with mISDN stuff?`
`13:51.50`	`ivanvujisi1`	`what's going on with DIALSTATUS variable?`
`13:52.12`	`j_kroon`	`i'm having trouble compiling asterisk with misdn support on asterisk 1.6.1.1`
`13:52.21`	`j_kroon`	`and misdn v2 (in-kernel stuff)`
`13:52.53`	`ivanvujisi1`	`${DIALSTATUS} don't work in 1.4.24.1 any idea?`
`13:53.42`	`j_kroon`	`that's a strange one.`
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`14:03.54`	`[TK]D-Fender`	`ivanvujisiPASteBIN is your friend <- show us the problme.`
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`14:06.36`	`Morghus`	`[TK]D-Fender, sorry, stuff happened. Found the problem. Some dipshit had been clever with things... fixed now :)`
`14:10.28`	`ivanvujisi1`	`anybody to help me with ${DIALSTATUS} ?`
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`14:15.38`	`kaldemar`	`ivanvujisi1: how does it not work? show the problem.`
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`14:18.19`	`talirk81`	`is there a equivenlant to php's is_set($ChanVarName) in an asterisk Dial plan`
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`14:19.57`	`tomodachi`	`how does ip telephony generally play with international country codes?`
`14:20.18`	`tomodachi`	`for example dialing numbers that actually start with a + for example`
`14:20.27`	`tomodachi`	`+46 (for sweden, in my example)`
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`14:24.29`	`mort_gib`	`tomodachi: What do you mean?? That you can't call Sweden from Asterisk :-)`
`14:24.38`	`tomodachi`	`mort_gib: i know that :)`
`14:24.58`	`tomodachi`	`well we have active directory (a ldap server with all of our data) including telephone numbers`
`14:25.06`	`mort_gib`	`tomodachi: Yeah....`
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`14:25.12`	`tomodachi`	`these numbers are stored in the format of +46 XXXXXX`
`14:25.16`	`creativx`	`+ is really 00`
`14:25.29`	`tomodachi`	`its + really just the same as 00?`
`14:25.30`	`tomodachi`	`supersure?`
`14:25.30`	`tomodachi`	`:)`
`14:25.37`	`tomodachi`	`in gsm networks`
`14:25.51`	`tomodachi`	`(phones sync this list through active sync)`
`14:26.07`	`[TK]D-Fender`	`[10:19]<tomodachi>how does ip telephony generally play with international country codes? <- no such thing`
`14:26.22`	`tomodachi`	`[TK]D-Fender: allow me to try to explain`
`14:26.36`	`[TK]D-Fender`	`tomodachi: How does my satellite TV receiver interact with peanut butter?`
`14:26.42`	`tomodachi`	`we have softphones that we also want to sync with this ldap service (a corporate phone directory)`
`14:26.53`	`tomodachi`	`but trying to call +46 doesent really work well from a softphone`
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`14:27.10`	`[TK]D-Fender`	`tomodachi: What you dial on your softphone depends on the server it passes calls to.`
`14:27.20`	`tomodachi`	`asterisk`
`14:27.41`	`tomodachi`	`guess im wonderig if its possible to let asterisk interpret + as 00 for example`
`14:27.45`	`[TK]D-Fender`	`tomodachi: * can handle a "+" in the pattern. Now once you're talking about a ITSP, the format they require can differ as well`
`14:28.13`	`[TK]D-Fender`	`tomodachi: No, * interprets "+" as a "+". What you DO with it it is up to YOU. this is dialplan...`
`14:28.37`	`tomodachi`	`ok so its possible if i just fiddle with my dialplan`
`14:29.11`	`tomodachi`	`that was an anwer that will keep me looking`
`14:29.14`	`tomodachi`	`thanks for your input`
`14:29.26`	`j_kroon`	`tomodachi, yes, like this: exten => _+.,1,Goto(00${EXTEN:1},1)`
`14:30.23`	`tomodachi`	`ok great, ill dig into it thnx for your quick answers`
`14:31.00`	`[TK]D-Fender`	`tomodachi: j_kroon's sample should tell you all you need to start mangling whatever was dialied into whatever form it needs to be.`
`14:31.42`	`tomodachi`	`i saw no refrence to + in the asterisk manual i found so i was unsure, ive done some dialplan editing so i've gotten what i need to get started, thnx again`
`14:31.43`	`j_kroon`	`yes, funny thing is i wrote that prolly just 5 minutes ago before reading this...`
`14:31.55`	`tomodachi`	`j_kroon: ok :) good timing then`
`14:32.35`	`j_kroon`	`yes, some client of mine is clearly insisting on passing +27 instead of 0 ... already had a rewrite for 0027 -> 0 so the logical thing was to rewrite + to 00`
`14:33.43`	`tomodachi`	`the reason we do it is that actve sync (and gsm phones that inherit this global adressbook, plays nicely with it , possibly your customer is in a similar situation`
`14:34.38`	`j_kroon`	`indeed.`
`14:34.56`	`j_kroon`	`now i just need to get misdn working.`
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`14:48.11`	`arsenick-`	`Hi all, we was trying to mark packet comming from our asterisk server to send it via one of our both cable connection`
`14:48.51`	`arsenick-`	`everything we tried failed, so we put things back to normal and now we are unable to make any external call`
`14:49.03`	`arsenick-`	`WARNING[18308]: channel.c:3210 ast_request: No channel type registered for ''`
`14:49.04`	`arsenick-`	`[Jun 22 10:47:06] WARNING[18308]: app_dial.c:1272 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)`
`14:49.24`	`arsenick-`	`the weird part is that there is not channel type, just ' '`
`14:49.53`	`arsenick-`	`asterisk 1.4.25.1`
`14:50.22`	`arsenick-`	`anyone understand this ??????`
`14:50.30`	`[TK]D-Fender`	`arsenick-: And the reason you aren't shoing us the command that CAUSED the error is?`
`14:50.51`	`[TK]D-Fender`	`arsenick-: You have clearly screwed up your DIAL command and what's missing is no doubt completely obvious`
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`14:54.32`	`arsenick-`	`[TK]D-Fender, http://pastebin.com/m397c7948 here it is`
`14:55.32`	`[TK]D-Fender`	`arsenick-: -- Executing [94189302424@DLPN_DialPlan1:1] Macro("SIP/124-b2f15dc8", "trunkdial-failover-0.3\|/4189302424\|\|xxxxxxxxxx\|trunk_1") <- look what this macro is being passed`
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`14:55.58`	`[TK]D-Fender`	`arsenick-: -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/124-b2f15dc8", "/4189302424") in new stack ,-- and here you can see that it doesn't add any tech to it`
`14:56.06`	`[TK]D-Fender`	`arsenick-: Nor any peer, etc`
`14:57.02`	`arsenick-`	`hmm`
`14:57.43`	`arsenick-`	`u mean we're supposed to see something like IAX2/3646324/4189302424 ? or ZAP ?`
`14:59.26`	`[TK]D-Fender`	`asrClearly.`
`14:59.32`	`[TK]D-Fender`	`arsenick-: Clearly.`
`14:59.47`	`arsenick-`	`all my trunk still in the gui and in the user.conf plus I deleted the outgoing rule and reinsert it without success...`
`15:00.53`	`arsenick-`	`I'm not sure if this is related to our modification on the firewall, we just got another connection, so we wanted to forward all the udp 4569 comming from our asterisk server to out voip provider`
`15:01.20`	`[TK]D-Fender`	`arsenick-: This is a config error straight up`
`15:01.31`	`arsenick-`	`but we had few problem and nothing worked.. so we removed ou modification and rebooted both server just to be sure.. and now it's broken :P`
`15:01.41`	`arsenick-`	`ok`
`15:02.07`	`kaldemar`	`sounds like a GUI bug, report it to whom it concerns.`
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`15:03.44`	`adr3nalin3`	`Could someone please give me a hand with Call Recording, I am using snom320's. I enabled automon in features.conf, put DYNAMIC_FEATURES = automon in extensions.conf and put wW in the DIALOPTIONS global variable I am using. I also changed the record button on the snom320 to DTMF with *1 as the snom wiki stated.`
`15:04.09`	`[TK]D-Fender`	`Could be a GUI bug, could be user error....`
`15:06.11`	`kaldemar`	`could be anything, but based on the current information they only played with the firewall by hand, not asterisk configuration.`
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`15:11.37`	`arsenick-`	`[TK]D-Fender, the version of asterisk-gui is the last tarball from digium ftp`
`15:11.46`	`arsenick-`	`is it safe to try updating from svn ?`
`15:12.00`	`[TK]D-Fender`	`arsenick-: GUI's are not supported here, go check out their channel.`
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`15:14.12`	`Joel`	`arsenick- try it?`
`15:14.39`	`arsenick-`	`it's weird the error don't show any type of channel, iax, sip zap or anything else it's just ' '`
`15:14.52`	`Joel`	`arsenick- bugs do exist in software.`
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`15:24.23`	`arsenick-`	`ok so I got the last version of gui and asterisk 1.4, got the same error so it should be an error somewhere in the config..`
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`16:01.30`	`WildPikachu`	`evening guys`
`16:03.22`	`KavanS`	`<adr3nalin3>: do you want to monitor every call?`
`16:03.28`	`KavanS`	`adr3nalin3: do you want to monitor every call? err :P`
`16:03.34`	`KavanS`	`not sure how the < >'s got in there`
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`16:07.14`	`carrar`	`pretty sure if you leave the "<>" in there they will have no idea who you are talking too :)`
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`16:13.44`	`InfoNutz`	`Hello all!... i'm hoping someone has some guru advice on upgrading a production version of asterisk 1.2.0 to the latest version... what is involved in the upgrading taking in considerations that would impact customer service... is it possible to jump from 1.2.0 to say 1.6 or 1.4. Is the most stable version 1.6.1.1 at the moment?`
`16:14.51`	`russellb`	`1.4 is more mature than 1.6`
`16:14.59`	`*** join/#asterisk enzo (n=enzo@extranet.source-rh.com)`
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`16:15.02`	`enzo`	`hi`
`16:15.02`	`russellb`	`for upgrade information, please please please read UPGRADE.txt`
`16:15.21`	`enzo`	`my asterisk opens port 2000, but i don't use this port, how can i ask him to shut it down ?`
`16:16.13`	`InfoNutz`	`thanks russellb!`
`16:17.04`	`enzo`	`skinny.conf opens this port indeed, any idea to close it ?`
`16:17.07`	`WildPikachu`	`so zaptel was replaced in 1.6 with dahdi?`
`16:17.29`	`enzo`	`yes`
`16:17.40`	`russellb`	`enzo: don't load chan_skinny`
`16:17.44`	`enzo`	`zaptel until asterisk 1.4`
`16:17.45`	`*** join/#asterisk CunningPike (n=CunningP@204.239.10.119)`
`16:17.47`	`russellb`	`noload => chan_skinny.so in modules.conf`
`16:17.52`	`WildPikachu`	`is there a doc describing the changes? :) so i don't waste you guys time`
`16:18.00`	`russellb`	`WildPikachu: and dahdi should be used in 1.4 as well, though 1.4 will support both`
`16:18.05`	`russellb`	`but zaptel is no longer maintained`
`16:18.13`	e0n`	`so i am playing with some call files in the outgoing queue (spool files), how do I make the phone ring as well so that when I specify an outgoing call that I create the desk phone rings`
`16:18.18`	`enzo`	`ok thanks russellb`
`16:18.21`	`russellb`	`in asterisk 1.4 - see Zaptel-to-DAHDI.txt`
`16:18.28`	`russellb`	`similar files are also in the DAHDI releases`
`16:18.44`	`WildPikachu`	`thanks russellb`
`16:18.47`	`russellb`	`np`
`16:19.49`	`adr3nalin3`	`KavanS: I would like monitor on demand with the record button on my snom 320s`
`16:20.01`	`WildPikachu`	`googles for where speex went to`
`16:21.10`	`*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)`
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`16:21.24`	`HorizonXP`	`in my dialplan, how can I execute a bash script?`
`16:21.49`	`HorizonXP`	`right now I have exten => _<mynum>,n,System(bash /var/spool/asterisk/future.sh)`
`16:22.27`	e0n`	`hmm`
`16:22.34`	`adr3nalin3`	`KavanS: You from Omaha?`
`16:22.54`	e0n`	`Anyone have any further documentation, the call file should have the SIP phone right yes? Or does it just dial outbound mindlessly?`
`16:23.41`	`enzo`	`i see that asterisk opens 5038 port for Asterisk Call Manager, you know good soft using this feature ?`
`16:24.36`	`joako`	`HorizonXP: Exec can execute your script, but will not return any values expect perhaps 0, -1, etc. If you need to pass variables you need to write an AGI script`
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`16:29.09`	e0n`	`hmm, so call files do not allow the desk phone to ring as well as calling outbound?`
`16:29.41`	`kaldemar`	`joako: what happened to System(/path/to/script ${VAR1}) ?`
`16:30.29`	`kaldemar`	`you don't need AGI to pass variables, getting variable values to the script is another thing.`
`16:30.39`	`KavanS`	`adr3nalin3, ok...I've got some good work for monitoring on my side...took me awhile`
`16:30.50`	e0n`	`Just hangs up the instant I answer`
`16:30.53`	`KavanS`	`adr3nalin3, but mine is automatic....takes place during every phone call`
`16:30.56`	`*** part/#asterisk xheliox (n=jeff@178.199.8.67.cfl.res.rr.com)`
`16:31.21`	`enzo`	`well now, i see port 2727 udp open port by asterisk, but i do't see this port in a conf file in /etc/asterisk, any idea what it is ?`
`16:31.58`	`kaldemar`	`enzo: MGCP`
`16:32.19`	`enzo`	`but i've added noload=> chan_mgcp.so and restarted but still there`
`16:33.06`	`enzo`	`ah it has disappeared now, weird...`
`16:33.12`	`joako`	`kaldemar: You can run it but you won't get back any variables the script generates`
`16:33.51`	`HorizonXP`	`joako: that only executes Dialplan applications`
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`16:34.31`	`joako`	`HorizonXP: System executes a unix command...`
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`16:35.20`	`HorizonXP`	`joako: right, which is what I want. I want to execute a bash script that I have to modify the time of an outgoing call`
`16:35.20`	`joako`	`kaldemar: Here's a REALLY simple AGI script, maybe it helps you: http://pastebin.ca/1470012`
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`16:35.43`	`HorizonXP`	`joako: i want to schedule the call for a minute into the future`
`16:36.42`	`kaldemar`	`joako: to be accurate, you can get back any variable that a script generates. thanks for the example, but i don't really need it. :)`
`16:37.35`	`joako`	`HorizonXP: In that case if you don't need any value return from the script System() will work fine`
`16:38.02`	`HorizonXP`	`joako: is there anything special i need to do to have it actually execute? because it's not working right now..`
`16:38.14`	`joako`	`HorizonXP: Check permissions?`
`16:38.23`	`HorizonXP`	`joako: it's owned by asterisk`
`16:38.29`	`joako`	`Make sure you are using absolute paths, etc`
`16:39.07`	`joako`	`HorizonXP: Is it chmod +x?`
`16:39.18`	`kaldemar`	`you don't need the bash part in front of the script name. if the .sh is executable, it will run. and joako is right about the path, it won't work without a path.`
`16:39.24`	`HorizonXP`	`joako: System(/bin/bash -c /var/spool/asterisk/future.sh)`
`16:39.31`	`HorizonXP`	`joako: it is +x`
`16:39.58`	`enzo`	`asterisk opens 5038, i can unload a module, or i need to put enabled = false in manager.conf?`
`16:40.28`	`joako`	`HorizonXP: Try just System(/var/spool/asterisk/future.sh) and personally even though it doesn't affect much I would put either in a bin/ directory or your agi-bin directory`
`16:41.33`	`joako`	`HorizonXP: I assume /var/spool/asterisk/future.sh at a command prompt works BTW, if it doesn't you need to sort that out first.`
`16:42.07`	`HorizonXP`	`joako: still same thing`
`16:43.16`	`rob0`	`Environment could also be an issue.`
`16:46.43`	`HorizonXP`	`joako: it does`
`16:46.46`	`HorizonXP`	`rob0: how so?`
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`16:47.07`	`*** mode/#asterisk [+o leifmadsen] by ChanServ`
`16:48.39`	`HorizonXP`	`i can't figure out why this isn't working`
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`16:53.28`	`rob0`	`"How so?" What are you asking? Even if I was qualified and inclined to explain all the details of shell command execution to you, it could not be done in IRC.`
`16:54.13`	`rob0`	`I scrolled up to see where you pastebinned the script, did not see it. You want wild guesses?`
`16:54.20`	`rob0`	`~wglwat`
`16:54.20`	`infobot`	`from memory, wglwat is well, good luck with all that`
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`16:56.38`	`*** join/#asterisk ariel_ (n=chatzill@63.214.236.169)`
`16:57.19`	`HorizonXP`	`rob0: well I can pastebin the script if you like :)`
`16:57.58`	`HorizonXP`	`rob0: http://www.pastebin.ca/1470037`
`16:58.09`	`Joel`	`use the full path to date`
`16:58.16`	`Joel`	`and touch`
`16:58.35`	`Joel`	`and the voip call file`
`16:59.07`	`*** join/#asterisk jshriver (n=jshriver@72.240.39.37)`
`16:59.12`	`jshriver`	`greetings`
`16:59.16`	`jshriver`	`anyone recommend easyAsterisk?`
`16:59.31`	`Joel`	`jshriver yes, I recommend asterisk.`
`16:59.44`	`ariel_`	`what is easyAsterisk?`
`16:59.46`	`jshriver`	`what about easyAsterisk for use as a server/asterisk`
`16:59.51`	`Joel`	`jshriver yes, I recommend asterisk.`
`16:59.52`	`HorizonXP`	`Joel: that was perfect, thank you.`
`16:59.53`	`jshriver`	`CentOS 5 based distro I think`
`17:00.02`	`jshriver`	`with web GUI for configuring asterisk`
`17:00.11`	`Joel`	`HorizonXP it's because $PATH isn't what you expect.`
`17:00.21`	`Joel`	`jshriver try #easyAsterisk ?`
`17:01.05`	`jshriver`	`just didnt know if anyone here had tried it or recommended it`
`17:01.25`	`jshriver`	`having a problem with my new box, was on the phone with digium for over an hour and they couldnt figure it out and never had a call back`
`17:01.32`	`ariel_`	`argh easyAsterisk is a limited system which you then need to go to the pro`
`17:01.59`	`Joel`	`jshriver try the freepbx comercial support`
`17:02.07`	`ariel_`	`jshriver: try asteriskNow it's an ISO, Elastix or PBX in a Flash`
`17:02.36`	`jshriver`	`thanks will give it a try. Using asterisk 1.4 now with CentOS 5, works mostly but doesnt do pots calls.`
`17:03.00`	`Joel`	`jshriver like I said, try the freepbx commercial support.`
`17:03.47`	`jshriver`	`how much is that?`
`17:03.50`	`ariel_`	`jshriver: the ones I posted are based on Freepbx and don't have any limits on extensions, pots lines or anything like that`
`17:03.57`	`Joel`	`jshriver google?`
`17:04.03`	`jshriver`	`hrm ok`
`17:04.12`	`ariel_`	`jshriver: ones I posted are free`
`17:04.29`	`jshriver`	`ty will look into asteriskNow as well`
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`17:28.13`	`Katty`	`finally gets to sit down for a minute.`
`17:28.49`	`[TK]D-Fender`	`gives Katty a standing ovation`
`17:28.56`	`*** join/#asterisk BobPierce (n=BobPierc@216.36.132.162)`
`17:29.48`	`*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)`
`17:30.10`	`Katty`	`eyes [TK]D-Fender`
`17:30.22`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`17:30.35`	`rob0`	`has had all he can standsk and can standsk no more! <toot, toot>`
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`17:43.52`	`Katty`	`so... where's everyone at`
`17:44.00`	`jaytee`	`TRABAJO!`
`17:44.16`	`Katty`	`hugs jaytee`
`17:44.36`	`jaytee`	`hugs Katty`
`17:45.09`	`*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)`
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`17:50.58`	`ariel_`	`jaytee: I guess your working like most of us.`
`17:52.04`	`alrs`	`This channel is much quieter than I remember it being a couple of years ago.`
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`17:52.45`	`ariel_`	`alrs: yes it is`
`17:54.03`	`alrs`	`Is it the global economy? Is everyone over in #freeswitch?`
`17:54.44`	`coppice`	`its the economy. the channel needs a bailout`
`17:54.59`	`*** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr)`
`17:55.15`	`ariel_`	`bailout, argh just think of our kids, kids will be paying for all of these bailouts.`
`17:55.50`	`alrs`	`They'll have devalued the dollar by then, don't sweat it`
`17:56.16`	`coppice`	`ariel_: heh neat. I can get the kids to pay for my telecoms for a change`
`18:00.35`	`ariel_`	`just seems our kids expect us to give them more, every year they just want more....`
`18:02.52`	`lanning`	`Please sir. Can I have some more?`
`18:06.10`	`ariel_`	`is having fun configuring yet another 976 number......;-)`
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`18:13.41`	`leifmadsen`	`hardwire: ping?`
`18:14.01`	`hardwire`	`you rang?`
`18:14.08`	`leifmadsen`	`can I pm you? :)`
`18:14.14`	`hardwire`	`pm away.`
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`18:24.50`	`*** join/#asterisk AiZ (n=AiZ@190.2.18.146)`
`18:25.50`	`AiZ`	`hello, i need some help to compile app_switch... it should be easy but i'm getting tons of errors like`
`18:27.18`	`AiZ`	`?`
`18:27.27`	`AiZ`	`' /usr/include/string.h:59: error: declaration for parameter âmemsetâ but no such parameter`
`18:27.32`	`AiZ`	`' /usr/include/string.h:59: error: declaration for parameter âmemsetâ but no such parameter`
`18:27.54`	`AiZ`	`(sorry)`
`18:27.59`	`*** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de)`
`18:30.05`	`tzafrir_laptop`	`AiZ, what version of asterisk?`
`18:30.21`	`AiZ`	`1.4`
`18:30.31`	`AiZ`	`Asterisk 1.4.17~dfsg-2ubuntu1`
`18:30.49`	`tzafrir_laptop`	`app_switch from where exactly?`
`18:31.13`	`tzafrir_laptop`	`do you rebuild the source of that package?`
`18:31.25`	`AiZ`	`app_swift, sorry`
`18:32.32`	`AiZ`	`the module that is used in order to interact with cepstral voices... got it from http://www.darrensessions.com/downloads/`
`18:34.32`	`AiZ`	`tzafrir_laptop come back :)`
`18:35.37`	`Katty`	`blergh`
`18:36.10`	`jplank`	`does nat=yes when type=peer work?`
`18:38.34`	`Defraz`	`I have a new PRI coming in what is the best card to use for that. Should I go Digium or Sangoma`
`18:38.37`	`Defraz`	`?`
`18:38.42`	`Defraz`	`Any ideas?`
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`18:43.29`	`*** join/#asterisk madduck (n=madduck@debian/developer/madduck)`
`18:43.50`	`madduck`	`it should be fairly trivial to Set(Language(),...) based on the first few digits of the caller ID, right?`
`18:43.57`	`madduck`	`I cannot find any recipes though`
`18:44.18`	`madduck`	`e.g. my default language is German, and if an Irish lad calls from a +353... number, I want language==en`
`18:45.24`	`leifmadsen`	`madduck: Set(LANGUAGE()=${IF($[${CALLERID(num):0:3} = 123]?fr:en)})`
`18:45.27`	`leifmadsen`	`madduck: something like that`
`18:45.52`	`leifmadsen`	`exact syntax for LANGUAGE() function (does that exist?) unknown`
`18:46.22`	`madduck`	`ah, and use GotoIf to prevent overriding the default`
`18:46.41`	`madduck`	`will try thanks`
`18:46.54`	`leifmadsen`	`right`
`18:46.58`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`18:49.21`	`madduck`	`http://www.voip-info.org/wiki/view/Asterisk+cmd+SetLanguage is the standard, I think`
`18:50.37`	`AiZ`	can anyone help me with app_swift? ... perhaps it doesn`t work on 64bits
`18:51.32`	`Qwell`	`AiZ: The author needs to update it for the version of Asterisk you're building against.`
`18:53.51`	`AiZ`	`Qwell: thanks :)`
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`19:00.07`	`madduck`	`<PROTECTED>`
`19:00.10`	`madduck`	`[Jun 22 20:57:48] ERROR[24846]: pbx.c:3368 ast_func_write: Function LANGUAGE not registered`
`19:00.13`	`madduck`	`is a bit disconcerting`
`19:00.56`	`madduck`	`https://issues.asterisk.org/view.php?id=9144`
`19:04.32`	`putnopvut`	`madduck: Set(CHANNEL(language)=en)`
`19:04.49`	`madduck`	`CHANNEL(language)`
`19:04.51`	`madduck`	`aaah`
`19:04.55`	`madduck`	`i just found it too. ;)`
`19:04.58`	`putnopvut`	`heh`
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`19:44.43`	`bryanfe2`	`can anyone tell me how to determine for absolutely certain, if "internal_timing=yes" in asterisk.conf really is working?`
`19:44.51`	`rue_mohr`	`so can bugs be filed on stables or only on the latest release?`
`19:45.27`	`rue_mohr`	`bryanfe2, which timing is that for?`
`19:45.38`	`bryanfe2`	`res_timing_dadhi`
`19:45.42`	`rue_mohr`	`ah`
`19:45.52`	`rue_mohr`	`dahdi_dummy?`
`19:45.58`	`bryanfe2`	`yes exactly`
`19:46.38`	`rue_mohr`	`may I ask you what makes you suspect its not?`
`19:46.46`	`bryanfe2`	`indeed`
`19:46.59`	`tompaw`	`Hi`
`19:47.00`	`rue_mohr`	`what makes you suspect its not?`
`19:47.03`	`tompaw`	`2 quick questions:`
`19:47.12`	`bryanfe2`	`I have SIP clients with "silence suppression" enabled. Asterisk isn't sending them any audio while the listener is silent.`
`19:47.22`	`bryanfe2`	`I believe that a switch to "internal_timing" is supposed to alleviate this.`
`19:47.41`	`bryanfe2`	`but I'm not observing that, even though I think I have internal_timing set up correctly.`
`19:47.44`	`rue_mohr`	`bryanfe2, tdm800 with hwec?`
`19:47.51`	`tompaw`	`1) Is there a way to delay the answer of a call in Asterisk? Let's say I want to perform some DB actions that take 5 second and I don't want the remote party to be billed for that (FAS)?`
`19:48.03`	`bryanfe2`	`i have no actual telephony cards installed, I"m just using dahdi_dummy (I believe)`
`19:48.12`	`tompaw`	`2) Are there any tools for sound detection? Like the ringback sound for example?`
`19:48.19`	`rue_mohr`	`there is a flag to not send data when you just have silence`
`19:48.28`	`rue_mohr`	`:/ I cant remember where`
`19:48.30`	`bryanfe2`	`that flag == bad`
`19:48.35`	`rue_mohr`	`its called someting like discontinious`
`19:48.50`	`bryanfe2`	`sip flag?`
`19:49.17`	`rue_mohr`	`tompaw, asterisk asnwers when you do answer()`
`19:49.38`	`rue_mohr`	`tompaw, maybe, check the stuff that TAD uses`
`19:49.46`	`bryanfe2`	`i just found something... codecs.conf "discontinuous transmission" - stops transmitting completely when silence is detected. I have the default setting of "dtx => false"`
`19:49.48`	`rue_mohr`	`bryanfe2, maybe, not sure`
`19:49.52`	`rue_mohr`	`yea`
`19:49.55`	`rue_mohr`	`thats it`
`19:49.57`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`19:50.04`	`bryanfe2`	`i guess maybe I want to try "true"?`
`19:50.17`	`rue_mohr`	`no I think you want it flase...`
`19:50.18`	`rue_mohr`	`hmm`
`19:50.38`	`tompaw`	`rue_mohr: let me pastebin you the scenario.`
`19:51.22`	`rue_mohr`	`[TK]D-Fender, you just gonna sit there or help me out here?`
`19:51.25`	`*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)`
`19:51.30`	`tompaw`	`rue_mohr: TAD?`
`19:51.41`	`rue_mohr`	`er, uh`
`19:51.46`	`rue_mohr`	`answering machine detection`
`19:51.52`	`rue_mohr`	`so I guess its AMD`
`19:52.04`	`tompaw`	`sounds like something perfect!`
`19:52.16`	`*** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it)`
`19:53.17`	`bryanfe2`	`rue_mohr are you talking to me? AMD?`
`19:53.43`	`rue_mohr`	`no state retention, I have no idea what I was talking baout`
`19:53.59`	`madduck`	`likes the result: http://stikked.com/view/91549382`
`19:54.09`	`madduck`	`leifmadsen, putnopvut: fyi ^^`
`19:54.30`	`tompaw`	`rue_mohr: if you'd be kind enough to have a look at http://pastebin.com/md713eea`
`19:55.06`	`*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)`
`19:55.56`	`tompaw`	`rue_mohr: in that scenario, when a remote party calls the [cont_project] content, the call will be answered as soon as the far end from 2nd step answers (Dial)`
`19:56.27`	`tompaw`	`rue_mohr: I want to delay this moment so I can perform some extra actions in 3rd step and only THEN signal the call as answered`
`19:56.53`	`tompaw`	`rue_mohr: what I already found out is this...`
`19:57.29`	`leifmadsen`	`madduck: nice`
`19:57.37`	`tompaw`	`rue_mohr: http://pastebin.com/m68d82317`
`19:58.01`	`tompaw`	`in this case SOME of the Macro actions can delay the Answer, for example SendDTMF()`
`19:58.21`	`tompaw`	`(the call will be signalled as answered to a caller only after SendDTMF() finished sending everything)`
`19:58.24`	`tompaw`	`hey.`
`19:58.26`	`tompaw`	`that's it`
`19:58.34`	`tompaw`	`I can simply send enough "w"s to make it wait`
`19:58.36`	`tompaw`	`thanks!`
`19:58.36`	`tompaw`	`:-)`
`20:00.19`	`*** join/#asterisk errr (n=errr@fedora/errr)`
`20:00.43`	`alonzo`	`how long it wait if you send "w"?`
`20:01.05`	`leifmadsen`	`500ms I think`
`20:01.27`	`alonzo`	`why?`
`20:01.36`	`leifmadsen`	`because that's how it's programmed?`
`20:02.13`	`alonzo`	`it's phone function, or asterisk parameter?`
`20:02.32`	`leifmadsen`	`Asterisk`
`20:02.53`	`*** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`20:03.00`	`tompaw`	`ASterisk`
`20:03.19`	`rue_mohr`	`so just dont put in an answer()`
`20:03.36`	`tompaw`	`it would be super amazing if I could do something like this: http://pastebin.com/m45e4a443`
`20:03.38`	`rue_mohr`	`it wont answer till either you do answer() or dial passes it to someone who does`
`20:04.01`	`tompaw`	`rue_mohr: that's it! I don't WANT it to do that automated answer() when the dial()'ed party answers.`
`20:04.41`	`rue_mohr`	`so have it give them playtones, set up the other side, and give them to each other using a meetme or soemthing`
`20:05.11`	`tompaw`	`hmm.`
`20:05.20`	`rue_mohr`	`used "or somthing" this means he dosn't really know, and thats not surprising cause he isn't an asterisk pro`
`20:05.31`	`bryanfe2`	`rue any other ideas? :(`
`20:05.57`	`bryanfe2`	`as far as I can tell I have internal_timing set up correctly, but my SIP clients with silence suppression still aren't working correctly.`
`20:06.10`	`bryanfe2`	`Maybe I'm mistaken in my assumption that internal_timing is supposed to fix SIP silence suppression in the first place.`
`20:07.38`	`rue_mohr`	`bryanfe2, AGI`
`20:07.41`	`[TK]D-Fender`	`Checkout time, bbiabv`
`20:07.58`	`rue_mohr`	`what!>`
`20:08.25`	`bryanfe2`	`rue what about AGI?`
`20:08.27`	`jaytee`	`sip silence suppression should be turned off on phone`
`20:08.38`	`rue_mohr`	`your just annoyed cause I pointed out asterisk has a 10db loss in its slinear <-> ulaw converter`
`20:08.40`	`bryanfe2`	`jaytee we really want to support that feature, to conserve bandwidth`
`20:09.18`	`rue_mohr`	`jaytee, is it worth me filing a bug about the 1mw being 10db out?`
`20:10.15`	`bryanfe2`	`jaytee can you pls explain, are you suggesting I turn off silence suppression because no matter what I do, Asterisk won't handle it correctly?`
`20:11.35`	`jaytee`	`bryanfe2, just that having silence suppression turned on can cause issues with MOH audio dropping out, disconnects etc.`
`20:12.06`	`bryanfe2`	`jaytee yes that's the kind of problem I'm having, which I was led to believe that proper use of internal_timing (via dahdi_dummy) would fix.`
`20:12.37`	`jaytee`	`dahdi_dummy would never come into play in a SIP to SIP call unless you used MeetMe or Page.`
`20:12.51`	`bryanfe2`	`it's sip to our application`
`20:12.53`	`bryanfe2`	`not sip to sip`
`20:12.55`	`*** join/#asterisk hi365 (n=hi365@94.159.177.240)`
`20:13.57`	`bryanfe2`	`is there any way to fix, for example, MOH to a SIP client with silence suppression enabled, other than by turning off silence suppression? (our goal is to keep it on)`
`20:14.55`	`jaytee`	`play a message to the person being put on hold that they need to continuously blow into the microphone :-)`
`20:15.08`	`bryanfe2`	`any alternative to that? ;)`
`20:15.31`	`bryanfe2`	`our app basically has the caller on mute (more or less), and we have to play stuff out to them.`
`20:15.45`	`bryanfe2`	`we really want to enable silence suppression, to conserve bandwidth`
`20:15.58`	`bryanfe2`	`but our apps seem to break (no audio being sent to the SIP client) under those conditions`
`20:16.12`	`bryanfe2`	`is there any way under the sun that this can be made to work other than having them blow into the mic or disabling silence suppression?`
`20:20.00`	`bryanfe2`	`I guess that's a "no"`
`20:26.23`	`*** part/#asterisk ivanvujisic (n=ivanvuji@91.148.102.205)`
`20:27.26`	`rue_mohr`	`bryanfe2, you have checked all the nlp settings to make sure their off?`
`20:27.32`	`rue_mohr`	`(I dont know wher they are)`
`20:27.39`	`rue_mohr`	`oh you want it on thoguh`
`20:27.59`	`rue_mohr`	`bryanfe2, sorry, that discontinious mode, you do want it true`
`20:28.08`	`bryanfe2`	`in codecs.conf?`
`20:28.22`	`rue_mohr`	`sorry, I got mixed up, most people want it off cause the silence makes them think the calls been dropped`
`20:28.24`	`bryanfe2`	`it's under the [speex] section, and I'm not using speex for far as I know`
`20:28.33`	`rue_mohr`	`sip?`
`20:28.50`	`bryanfe2`	`there is no discontinuous option in sip.conf as far as I can see`
`20:29.55`	`rue_mohr`	`oh turn on nlp...`
`20:30.01`	`bryanfe2`	`nlp?`
`20:30.30`	`tompaw`	`Guys, that AMD is amazing.`
`20:30.30`	`bryanfe2`	`i dont know what that is, nor see an option for it`
`20:30.34`	`tompaw`	`I just need some help with it`
`20:30.55`	`bryanfe2`	`rue whats nlp?`
`20:31.17`	`tompaw`	`AMD says: Changed state to STATE_IN_SILENCE, Too long...`
`20:31.24`	`tompaw`	`Which variable defines that "too long" period?`
`20:31.33`	`tompaw`	`I went through all of them in amd.conf and noone seems to match.`
`20:33.01`	`bryanfe2`	`rue what's nlp?`
`20:34.02`	`*** join/#asterisk jpcansa (n=jpbenavi@ip129-143-122-200.ct.co.cr)`
`20:34.46`	`rue_mohr`	`not sure`
`20:34.52`	`rue_mohr`	`something to do with silence muting`
`20:36.52`	`jpcansa`	`does anybody know how to set far end disconnect on a digium card?`
`20:41.18`	`rue_mohr`	`hu?`
`20:44.17`	`*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)`
`20:45.03`	`*** join/#asterisk Nox93 (i=secret@74.112.32.213)`
`20:45.16`	`*** join/#asterisk autojack (n=owen@nerdnetworks.org)`
`20:46.01`	`*** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net)`
`20:48.04`	`Nox93`	`If I have a sip peer named "xyz" and I want to dial phone number 1112223333 through them is it proper to use Dial(SIP/1112223333@xyz,30) ?`
`20:48.18`	`Nox93`	`I get the same thing(a generic message) as if I use no extension.`
`20:50.41`	`autojack`	`Nox93: the syntax I use is Dial(SIP/xyz/111222333,30)`
`21:03.51`	`autojack`	`I have a setup with a DID calling in over SIP, taking a phone number input from the caller, and then dialing out over SIP via my VOIP termination provider. on answer, my system bridges the calls so they can talk. pretty simple.`
`21:04.08`	`autojack`	`I'm having occasional issues with calls being answered at the other end, but no audio passing.`
`21:04.38`	`autojack`	`it's intermittent and usually works after a couple of tries if it doesn't work on the first one.`
`21:05.38`	`autojack`	`I'm not using NAT or any firewall on the Asterisk end, and I tried switching to packet2packet bridging instead of using SIP reinvite on my calls.`
`21:06.00`	`autojack`	`my termination provider thinks my config is OK, so I guess maybe this is a problem with one of their routes.`
`21:06.07`	`autojack`	`any thoughts?`
`21:09.11`	`*** join/#asterisk cesar_CR (n=cesar@201.192.86.30)`
`21:09.28`	`*** join/#asterisk intralanman (n=lanman@va-67-76-163-226.sta.embarqhsd.net)`
`21:11.21`	`ariel_`	`autojack: how are your did's coming into your pbx?`
`21:15.44`	`*** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com)`
`21:16.57`	`*** join/#asterisk Whitor (n=Whitor@64.128.237.124)`
`21:25.42`	`autojack`	`ariel_: the DID is set up to connect over SIP.`
`21:26.28`	`ariel_`	`So your sip provider is giving you the did and are you also using them for your termination?`
`21:27.34`	`*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)`
`21:27.35`	`autojack`	`well, I bought the DID through my termination provider, and I know they get it from didx.`
`21:27.45`	`*** join/#asterisk lanning (n=lanning@173.8.187.197)`
`21:27.45`	`autojack`	`I'm not sure if that means the DID is fully controlled by them or not.`
`21:27.55`	`autojack`	`I use their site to configure what the DID does.`
`21:28.04`	`*** join/#asterisk xphree (n=seele@unaffiliated/xpider)`
`21:28.29`	`*** join/#asterisk Aiatek (n=Amunoz@190.94.58.62)`
`21:28.50`	`xphree`	`Hello, i'm having a problem with the dialcommand params... i have the following dialcommand param \|60\|HLC(%timeout%:61000:00000)`
`21:29.18`	`xphree`	`where %timeout% is the duration of the call... but the system never hangup the call when achieve the timeout`
`21:29.57`	`xphree`	`only if i remove the C the system hangup the call.. but i need to keep the C param because i need that the call be billed when the party answer the call`
`21:30.02`	`xphree`	`anyone can help me with this?`
`21:33.58`	`*** join/#asterisk scalex000 (n=chatzill@218puntacana02.codetel.net.do)`
`21:34.29`	`scalex000`	`Hi everyone`
`21:35.22`	`*** join/#asterisk botox93 (n=botox93@213.221.82.242)`
`21:37.32`	`blaxthos`	`hi`
`21:37.39`	`xphree`	`hello all`
`21:37.50`	`blaxthos`	`lots of questioners, very few answers today :(`
`21:38.05`	`*** join/#asterisk j_kroon (n=jkroon@dsl-240-128-42.telkomadsl.co.za)`
`21:38.09`	`blaxthos`	`so i added a new extension today (x103), polycom 501 SIP, trixbox/latest`
`21:38.25`	`blaxthos`	`can dial from it to any internal extension (10x), or out a trunk no problem`
`21:38.36`	`blaxthos`	`but can't dial into that extension (103) from anywhere`
`21:38.53`	`blaxthos`	`<PROTECTED>`
`21:38.58`	`blaxthos`	`<PROTECTED>`
`21:39.05`	`blaxthos`	`<PROTECTED>`
`21:39.23`	`ariel_`	`autojack: I would have to say that it could be your reinvite, since you don't know where your provider is behind the read did, you might be having issues as to where there being setup from.`
`21:39.32`	`blaxthos`	`<PROTECTED>`
`21:39.43`	`blaxthos`	`all other extensions work perfectly, always have`
`21:39.45`	`blaxthos`	`any ideas ?`
`21:40.08`	`ariel_`	`it's pass 5:30 pm and I need to head home. See you guys I hope tomorrow.`
`21:42.44`	`j_kroon`	`hi all. i'm looking to make a choice based on whether a certain "sound" exists or not, eg, if foo/bar can be played, play it, otherwize play some other default. I've seen the EXISTS function, but I suspect this more tests for existence of variables, not for existence of files. alternative ideas?`
`21:44.59`	e0n`	`hmm`
`21:45.20`	e0n`	`So how can I track with asterisk to see if a phone number has been disconnected if I am using /outgoing spool files`
`21:45.35`	`j_kroon`	`CDRs?`
`21:46.06`	`j_kroon`	`or do you want to keep a certain concurrency?`
`21:46.26`	e0n`	`Just a way to retrieve the info back from asterisk and store into a file or database of sorts`
`21:46.30`	e0n`	`so I can track it on my own`
`21:46.41`	`j_kroon`	`CDRs is the way to go imho.`
`21:46.57`	`j_kroon`	`cdr_custom is a good choice imho.`
`21:47.18`	e0n`	`ah`
`21:47.31`	e0n`	`can cdr_custom interact with postgres or only mysql`
`21:48.18`	`j_kroon`	`neither.`
`21:48.23`	`j_kroon`	`it dumps you a text file.`
`21:48.41`	`*** join/#asterisk rue_mohr (n=rue@24.207.122.10)`
`21:48.46`	`j_kroon`	`for mysql you can use cdr_mysql (asterisk-addons), and for postgres I believe cdr_odbc should work.`
`21:49.15`	e0n`	`Ok i'll look into it`
`21:50.44`	`blaxthos`	`Two D.C. Metro trains collided during rush hour Monday, CNN reports.`
`21:53.05`	`ajohnson`	`I think it's funny how the whole headline doesn't fit inside of the box on my machine`
`21:53.17`	`ajohnson`	`Watch Now is half cut off`
`21:54.41`	`Deeewayne`	`mog, ping`
`21:55.35`	`*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)`
`21:55.50`	`rue_mohr`	`who here has system running a dahdi card?`
`21:56.22`	`Aiatek`	`you mean dahdi modules`
`21:56.28`	`Aiatek`	`rue_mohr`
`21:56.43`	`rue_mohr`	`dahdi drivers for whatever hardware they have`
`21:56.47`	`drmessano`	`has a zaptel card, not a dahdi card.. :(`
`21:57.00`	`rue_mohr`	`so your using the old drivers`
`21:57.04`	`Aiatek`	`i have a card running with dahdi`
`21:57.19`	`drmessano`	`throws it in a box with his windows video card`
`21:57.21`	`rue_mohr`	`Aiatek, ever had any volume complaints?`
`21:57.32`	`Aiatek`	`nope`
`21:57.45`	`rue_mohr`	`what version of the drivers/asterisk are you using?`
`21:58.27`	`Aiatek`	`DAHDI Linux 2.1.0.4`
`21:58.56`	`rue_mohr`	`hmm`
`21:59.00`	`rue_mohr`	`what asterisk ver?`
`21:59.27`	`Aiatek`	`1.6`
`21:59.37`	`rue_mohr`	`hmm`
`22:01.18`	`rue_mohr`	`could you do me a favor, dial into a miliwatt() app, run dahdi_monitor -vv on the channel and tell me the level it says?`
`22:01.20`	`rue_mohr`	`I want to confirm its about 4600`
`22:01.59`	`Aiatek`	`well i need to turn on my lab pc, because the one that it is in production its in my job`
`22:02.15`	`rue_mohr`	`it wont interrupt anything`
`22:02.29`	`rue_mohr`	`you can do it completely safely live`
`22:02.39`	`Aiatek`	`im not in my job`
`22:02.41`	`Aiatek`	`:)`
`22:02.44`	`rue_mohr`	`oh`
`22:02.50`	`rue_mohr`	`hmmm`
`22:02.59`	`rue_mohr`	`if you get a chance, I'd really like to know`
`22:10.25`	`rue_mohr`	`nobody else running digium hardware?`
`22:10.27`	`talirk81`	`In AGI is the a way i can set the CallerID Name, using Set CallerID sets the number and name both the the number, but i would like to do a nice name, without having to do it in the actual dial plan`
`22:11.27`	`rue_mohr`	`cant help there`
`22:12.54`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)`
`22:12.58`	`russellb`	`set CALLERID(name)`
`22:13.32`	`russellb`	`exec Set CALLERID(name)=newname`
`22:13.35`	`russellb`	`or whatever the syntax is`
`22:33.26`	`*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)`
`22:36.48`	`*** join/#asterisk xphree (n=seele@unaffiliated/xpider)`
`22:36.58`	`xphree`	`Hello people, i need help with the dialcommand params`
`22:37.53`	`xphree`	`when i made a call with the param \|60\|HCL(60000:610000:30000), the system don't hangup the call when the timeout is achieved`
`22:41.11`	`xphree`	`anyone has an idea about my problem?`
`22:43.14`	`*** join/#asterisk voxter (n=voxter@190.241.15.56)`
`22:44.17`	`j_kroon`	`xphree, which timeout?`
`22:44.34`	`j_kroon`	`and why are you using \| compared to , ?`
`22:44.47`	`xphree`	`in the dialcommand param`
`22:45.52`	`j_kroon`	`the L() one?`
`22:46.13`	`xphree`	`DIAL DAHDI/g2/xxxxx:xxxx@server/number\|60\|HCL(60000:61000:30000)`
`22:46.16`	`xphree`	`thes, the L one`
`22:46.23`	`j_kroon`	`does it make a diff at all if you use , instead of \| ?`
`22:46.57`	`j_kroon`	`wtf?!? that DAHDI doesn't look right ... but it could just be me that doesn't know that syntax.`
`22:47.13`	`*** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)`
`22:47.28`	`j_kroon`	`also, i'd wager the fact that y is larger than x might throw L() a bit.`
`22:47.57`	`j_kroon`	`L(x[:y][:z]) <- x is the limit, y is the warn time, thus I'd assume that y needs to be less than x.`
`22:48.18`	`j_kroon`	`so what happens if you have L(60000:30000:10000) ?`
`22:48.53`	`xphree`	`if i use , the system don't call`
`22:49.01`	`j_kroon`	`asterisk version?`
`22:49.20`	`xphree`	`1.4.25.1`
`22:49.40`	`j_kroon`	`ok, i'm using 1.6 generally.`
`22:50.01`	`xphree`	`no, if i use the other conf my system hangup the call inmediatly`
`22:50.09`	`j_kroon`	`extensions.conf:exten => _X.,n,Dial(${ARG1}/${EXTEN},,${trdialopts}) <-- what I use generally.`
`22:50.15`	`j_kroon`	`what?!?`
`22:50.21`	`*** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net)`
`22:50.36`	`j_kroon`	`so with the 60000:30000:10000 you get immediate hangup?`
`22:50.42`	`xphree`	`yes`
`22:50.58`	`xphree`	`let me check`
`22:51.19`	`j_kroon`	`that sounds severely wierd.`
`22:51.44`	`j_kroon`	`finally. after about a week i finally got misdn running on a 2.6.29.4 kernel.`
`22:52.09`	`j_kroon`	`well, compiled at least.`
`22:52.28`	`xphree`	`i'm checking with your suggestion`
`22:53.42`	`[TK]D-Fender`	`xphree: xxxxx:xxxx@server/number <--- this is VOIP FORMATTING, not DAHDI`
`22:54.44`	`xphree`	`[TK]D-Fender: ok my mistake, i confused SIP with DAHDI, sorry`
`22:54.51`	`[TK]D-Fender`	`CRAZY TALK`
`22:55.14`	`*** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)`
`22:55.23`	`xphree`	`i have this: DIAL DAHDI/g1/3006557385\|60\|HLC(60000:30000:10000)`
`22:56.05`	`xphree`	`but the call don't hangup on timeout`
`22:56.37`	`xphree`	`and this is causing problems on my billing system`
`22:57.15`	`seanbright`	`is that being called from an AGI?`
`22:57.20`	`seanbright`	`or from extensions.conf?`
`22:57.34`	`[TK]D-Fender`	`xphree: Since when does the "C" option take the timeout parms?`
`22:57.37`	`xphree`	`from an AGI (a2billing.php)`
`22:57.43`	`seanbright`	`ahhh`
`22:57.47`	`seanbright`	`good eyes, TK`
`22:57.54`	`[TK]D-Fender`	`xphree: Get your order right and go read the instrucitons`
`22:58.45`	`xphree`	`ok, rearrange the dialcomman param to HCL(60000:30000:10000)`
`22:58.57`	`xphree`	`i'm checking.. wait me a minute :)`
`22:59.26`	`*** part/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net)`
`23:00.31`	`xphree`	`is working!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!`
`23:00.48`	`seanbright`	`good!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!`
`23:01.11`	`xphree`	`i was confused by an online tutorial... i will send an email to the author`
`23:01.30`	`j_kroon`	`shees, well spotted [TK]D-Fender ... i would have stared a while.`
`23:01.56`	`[TK]D-Fender`	`xphree: Don't blame him that you aren't even reading the instructions.`
`23:02.08`	`xphree`	`thanks to j_kroon seanbright and [TK]D-Fender and all the Staff for helping us`
`23:02.27`	`seanbright`	`that's what we get paid the big bucks for`
`23:02.40`	`j_kroon`	`speak for yourself :p`
`23:02.59`	`seanbright`	`i'm here for free, too :)`
`23:03.06`	`Qwell`	`~help`
`23:03.11`	`Qwell`	`d'oh`
`23:03.12`	`Qwell`	`~ask`
`23:03.12`	`infobot`	`Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will.`
`23:03.17`	`Qwell`	`the latter part :p`
`23:05.17`	`xphree`	`i'm breathing again... but i lost like 200 usd that i owe my customer because of the timeout.. lol`
`23:05.52`	`ISO9001`	`whoops.`
`23:06.20`	`xphree`	`yes, ups!`
`23:08.56`	`*** join/#asterisk troubled (n=troubled@unaffiliated/troubled)`
`23:10.03`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`23:11.34`	`*** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net)`
`23:13.36`	`bryanfe2`	`does anyone here know if there is some flag I have to set, other than internal_timing = yes, to make sure that RTP uses the internal timer for outbound audio?`
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`23:20.42`	`*** join/#asterisk digitalirony (n=digitali@shellium/member/digitalirony)`
`23:24.12`	`j_kroon`	`bryanfe2, i always play it safe and pass -I in addition to settong internal_timing=yes ... when i set it up there were some conflicting reports as to whether setting internal_timing had any effect,`
`23:24.37`	`bryanfe2`	`j_kroon I am passing -I as well`
`23:24.53`	`bryanfe2`	`my SIP audio behavior is acting as if internal timing is not being used (no audio sent to client, when client is silent)`
`23:25.09`	`bryanfe2`	`but as far as I can tell, I have everything set up correctly for the internal timer to exist.`
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`23:27.01`	`denesh`	`Hi everyone... I am new to Asterisk and I need some one to help me understand the possibilities with asterisk`
`23:27.58`	`denesh`	`I have an office in which an analog phone line needs to be answered by an answering system and route the calls to different rooms based on extension number`
`23:28.04`	`denesh`	`is this possible with asterisk ?`
`23:28.30`	`[TK]D-Fender`	`deneeasily`
`23:28.40`	`[TK]D-Fender`	`denesh: easily`
`23:28.48`	`denesh`	`good awesome....`
`23:28.59`	`denesh`	`now how do i go about it... hardware requirements ?`
`23:29.39`	`*** join/#asterisk gunter (n=user@87.127.97.39)`
`23:30.17`	`denesh`	`do i need any special cards to hook up the phones to the system ?`
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`23:33.08`	`[TK]D-Fender`	`denesh: depends what kind of phones`
`23:33.49`	`denesh`	`regular analog phones`
`23:34.28`	`Pan3D`	`denesh: you`
`23:34.54`	`Pan3D`	`will need a small device that converts your regular phone connection to a network connection`
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`23:36.01`	`denesh`	`hmm if someone could point me to a howto or tutorial...`
`23:36.02`	`Pan3D`	`denesh: http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters`
`23:36.08`	`Pan3D`	`^^`
`23:36.17`	`Pan3D`	`is psychic`
`23:36.22`	`denesh`	`;)`
`23:36.29`	`Pan3D`	`good luck`
`23:36.58`	`[TK]D-Fender`	`~book`
`23:36.58`	`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`
`23:37.00`	`[TK]D-Fender`	`^^^`
`23:37.01`	`Pan3D`	`you might also want to look at the relevant sections of the * book`
`23:37.07`	`Pan3D`	`hahaha`
`23:37.09`	`Pan3D`	`is psychic`
`23:37.11`	`denesh`	`good... i have those... i'll give them a shot`
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