IRC log for #asterisk on 20070423

00:00.46*** join/#asterisk LetsTalkPBX (n=LetsTalk@pool-71-182-164-82.pitbpa.east.verizon.net)
00:04.59*** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net)
00:12.19danicholsonAny-one have the *.conf files that they are using with a Cisco AS5400 that they are willing to share?
00:18.31khronosAnybody ever connected an Asterisk system to an Altigen?
00:18.31aydiosmiokhronos: clever.
00:18.31aydiosmiolet me know how that works out
00:18.55khronosWhat I want to do is link my exten on the work pbx to my phone at home since I work from home a lot.
00:19.43khronosRight now I have ti breaking out a line and call my home phone, but I'd like to take that out of the loop.
00:21.19*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:33.15khronosAh, looking around and talking to my boss I see I've got to use h323 to talk to the Altigen.
00:33.46khronosNow just have to figure out how to format the h323 config file.
00:34.14khronosA question I have though, should I use the h323 driver that comes with Asterisk or openh323.
00:36.52bkruse_homeewwwwww
00:36.55bkruse_homefile: <3
00:42.49*** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
00:42.57*** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net)
00:43.21*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
00:46.18*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)
00:47.36paavumcan I integrate any fax 2 mail solution (spanDSP/HylaFax/Asterfax) with Asterisknow B4?
00:47.41paavumb4 = beta 4
00:51.00jovannottisomething here knows what version I should use for pwlib and openh323 in fedora core 6 ?
00:52.51ManxPowerjovannotti: only the version listed in the readme or install file will work
00:56.12jovannottithen I need to download and install Open H.323 version v1.17.1, PWLib v1.9.0 ?
00:56.25jovannottiI tried to install these, but it looks so older to fedora core 6
00:56.32jovannottiand I have problems compiling it
00:56.57*** join/#asterisk hmm-home (n=hmm-home@24-117-131-41.cpe.cableone.net)
00:57.08hmm-homeI forgot how colorful gaim is
01:04.22*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
01:05.52*** join/#asterisk JT_ (n=jon@unaffiliated/jt)
01:11.46*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)
01:16.39*** join/#asterisk jcaceres (n=josexato@190.40.70.66)
01:16.54jcacereshello does anybody used RAGI?
01:18.56danicholsonHello, any-one here with a Cisco AS5400 connected to *1.4?
01:23.37ManxPowerjcaceres: RAGI lets your app LISTEN for audio, it does not allow your app to SEND audio.
01:31.33ManxPowerjcaceres: Sorry, I was thinking of EAGI.  What I said does not apply to RAGI
01:32.32jcaceresoka ManxPower,
01:32.38*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
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01:39.22*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
01:44.16danicholsonHello, any-one here with a Cisco AS5400 connected to *1.4?
01:47.34blitzragedanicholson: there wasn't 30 mins ago, and there still isn't
01:53.21*** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net)
01:57.34*** join/#asterisk digiterata (n=digitera@bas1-montreal02-1096716595.dsl.bell.ca)
01:59.03digiteratahey folks
01:59.29digiteratavery frustrated, looking for some advice.
02:00.11jovannottiidem
02:00.29digiterataI'm not a linux pro but I'm trying to get Asterisk 1.4 up and running on a virtual machine.
02:01.02*** join/#asterisk remmo (n=junk@smack.isp.net.au)
02:01.47khronosI've actually thought about running Asterisk in a vm, but I think the hardware modules for interface cards you have will need to be compiled in the the host machine's kernel as well as the ztdummy module if you don't have any cards.
02:02.21digiterataI've played with AsteriskNow but I'm looking to get a clean installation up and running and I'm not sure where to start. Trying to find a good simple distro that works nicely with Asterisk
02:02.39*** join/#asterisk bkw_ (i=brian@adsl-70-142-43-193.dsl.tul2ok.sbcglobal.net)
02:03.12digiterataI've played around with AsteriskNow and it's pretty nice actually. just that it's a bit locked down and I'd like to be able to talk to Asterisk through the AGI
02:03.45digiterataI think for anything real, I'd still rather have asterisk on bare metal, but at the moment I'm purely in development.
02:04.11digiteratawhat distro do you use to base your Asterisk machines on?
02:04.43*** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au)
02:04.56NuggetI run asterisk in Slackware, FreeBSD, Mac OS X.
02:05.13phix<PROTECTED>
02:05.32digiterataI've heard good things about Slackware. Is it a good distro to learn on?
02:05.51phixIhave two asterisk servers, A and B, I am trying to call a user on A from B
02:06.06phixthe user on A has an extension of 3, and is using SIP
02:06.11digiterataSlackware, it's known for stability, yes?
02:06.19NuggetLinux is Linux.
02:06.22phixhello
02:07.10digiterataok, so it doesn't really matter. except it does. they all seem to use different package managers.
02:07.14NuggetSlackware is not a particularly good distro to learn on, though.
02:07.18NuggetI use it because I hate it the least.
02:07.27*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
02:07.27*** mode/#asterisk [+o mog] by ChanServ
02:07.30NuggetIt's the least Linuxy Linux.
02:07.44phixcan any one give me a hand?
02:07.57bluelinqis there a way to increase the volume using some configuration ina 5960 on sip?
02:08.06digiterataI have a VMware image running Ubuntu 6.10 server.
02:08.50digiterataIt seems pretty good except for the fact that I can't seem to wget asterisk from digium.
02:08.58phixgrrrrrrrrrrrr
02:09.22bluelinqweird part is that under the sccp firmware the volume is a lot louder...
02:09.28digiteratahey phix, I'm sorry. wish I knew.
02:10.24phixdigiterata: ok
02:10.30phixdigiterata: you know anyting about iax?
02:11.57bluelinqsip volume anyone?
02:16.56bluelinqphix: here you go...http://www.pastebin.ca/453539
02:19.30digiteratayeah I know about iax (sorry was on the phone)
02:19.44digiterataenough to be dangerous anyway
02:22.12phixgrrrr
02:22.16phixwhy am I getting rejected for?
02:23.58*** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net)
02:25.24nDuffAnyone know how to modify the dialplan on a GXP-2000?
02:26.57nDuffI'm looking at the web interface, and I don't see anything at all about the dialplan (except for a field for a fixed prefix to be added to outgoing calls)
02:26.58phixok so why am I getting rejected for?
02:27.13phix......
02:28.15nDuffphix: That's typically an authentication-related message, which implies some obvious initial places to look.
02:28.34nDuffoh, wait, you posted more above.
02:30.50nDuffphix: How is the host defined in your iax.conf? A user, a peer, or a friend?
02:32.01*** join/#asterisk Fieldy (i=6nD2Oy7n@gentoo/contributor/Fieldy)
02:32.35phixfriend
02:32.45phixnDuff: on some servers A and B
02:33.01phixas I would like either end to make or receive calls
02:33.12phixsome = both
02:33.18phixtypo :P
02:34.09nDuff...and your secrets match, and your host entries match the actual IP the remote traffic is coming from, and the IAX module has been reloaded since you did all this?
02:34.14phixyes
02:34.37phixI have /etc/init.d/asterisk restart on both servers
02:34.53phix(both are debian system running asterisks 1.2)
02:34.54nDuffHuh. Dunno. Last time I did that, it Just Worked.
02:35.00phixhmmm
02:35.38phixnDuff: I want to dial server A extension 3 from server B
02:36.24phixI have added in exten => 501,1,Dial(IAX2/name/3) on server B
02:36.57phixdo I need to add anything on Server A to give server B permission to ring that extension?
02:37.49phixdo I need a hostfrom directive?
02:38.17nDuffIIRC, I ended up specifying everything (hostname, password, etc) in my outgoing Dial path. (Okay, it didn't Just Work -- there was a little trial/error/adjustment on the outgoing side)
02:38.37phixdo I need to put it in an outgoing dial plan?
02:38.49nDuffnot to say it should be necessary to do that -- I was just in a hurry, and that's what worked for me.
02:38.59phixI have setup a SIP user which is in context sip, the extension I want to call is in context sip too
02:39.03phixis that all I need to do ?
02:39.09phixhmm
02:39.49*** join/#asterisk CunningPike (n=CunningP@204.239.8.149)
02:40.26digiteratahey guys, I've been trying all weekend to get up and running on Asterisk 1.4  My latest attempt is from a VM running Astlinux. Could anyone tell me if it's possible to upgrade the Asterisk 1.2 on this distro to 1.4?
02:40.57nDuffphix, the whole thing is more like IAX2/username:password@host:port/number?context
02:41.14phixhmmm
02:41.27phixdo I need to specify username and password?
02:41.33nDuffdigiterata: Of course it's possible, as long as you've got a compiler; it's just a matter of how *hard* it is.
02:41.35phixI have guest enabled on both servers
02:42.03nDuffphix: I had to. Shouldn't be necessary, then or now, but I was in a hurry and it worked for me.
02:42.41phixok
02:42.58nDuffphix: If that doesn't work, I'm not sure what to tell you -- I'd probably turn on debug logs or (if they're inadequate) start instrumenting the relevant source to figure out what's going on, but that's just me.
02:43.05digiterataright, about that *hard* part.
02:43.47nDuffdigiterata: I don't know the first thing about Astlinux, so I can't tell you.
02:43.58digiteratai'm a complete linux n00b but I'm good with google - haven't found the answer yet though
02:44.50*** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com)
02:47.31VJFROMGTknock knock
02:56.05*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
02:58.12*** join/#asterisk jovannotti (n=jovannot@190.84.99.36)
02:58.26jovannottisomething has tested TC400B card from digium >
02:58.27jovannotti?
03:08.23*** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net)
03:10.40*** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195)
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03:16.05piper69is LINKSYS PAP2 locked to vonage?
03:16.26jqltypically
03:16.33jqlPAP2-NA is the unlocked variety
03:18.46piper69where can i get that , all what is in the market is for vonage
03:19.07piper69aren't vonage suppouse to go out of busniess
03:20.37khronosPossibly, but we don't know yet.
03:21.09khronosFrom what it looks like we'll have to wait til the courts decide the pattent issues that are being faught over.
03:21.46mogvonage wont go under
03:21.54mogverizon might buy em for their customers
03:21.59mogafter they crush them of course
03:22.11mogor someone who has indemnity from verizon on said patents
03:22.18jqlvonage has liabilities, though. bad purchasing decision
03:22.57jqlall that cash in the bank still doesn't make them worth it...
03:22.58piper69http://shop4.outpost.com/%7BVq1PeD5uecMRKyYm34Ehcw**.node2%7D/product/4175693;jsessionid=Vq1PeD5uecMRKyYm34Ehcw**.node2?site=sr:SEARCH:MAIN_RSLT_PG
03:23.04fileyou silly goose
03:23.09mogyeah i know
03:23.13mogi am retarded
03:23.52danicholsonhttp://www.voipnow.org/vonage/index.html
03:25.00khronosIn the next couple months I will be setting up a voip over satellite system for a couple of locations.
03:25.14khronosWhat are my options for doing fax over ip?
03:25.31Corydon76-homeLOL... oh, wait, were you serious?
03:25.54khronosI was thinking of doing some sort of trunking transcoding server at the satellite company that will take the calls and transcode them to gsm off the pri.
03:26.43khronosIf I set the asterisk servers at the remote locations to use gsm how would I be able to have the clients pass faxes over their atas at the different locations and still have them work?
03:27.04nDuffkhronos: T.37 and T.38.
03:27.10CunningPikekhronos: What he said
03:27.14fileyou were thinking of sending faxes over that?
03:27.19VJFROMGThow can i tell what codec is been used on a particular iax2 call?
03:27.19CunningPikeI can't type fast enough, obviously
03:28.06khronosWould I have to set certain extensions up as fax lines or will I be able to have the atas at the homes pick which protocol to use?
03:28.12CunningPikekhronos: T.38 is your only chance, and even then, you're getting involved in a whole heap of hurt, my friend
03:28.48nDuffkhronos: See http://www.soft-switch.org/foip.html
03:28.52*** join/#asterisk mekong (n=josh@cpe-69-203-218-147.nyc.res.rr.com)
03:35.27mekongs
03:37.32nDuffkhronos: given you enough to think about?
03:38.36*** join/#asterisk darkladywolf (n=root@wolf.tpgi.com.au)
03:38.54nDuffkhronos: If I were in your shoes, I'd rig up something T.37-style (maybe even real T.37), using iaxmodem or something like it at each POTS connection and then sending the content as email between them.
03:40.55*** join/#asterisk thoughtpolice (n=austin@c75-111-137-154.plaicmtc01.tx.dh.suddenlink.net)
03:41.04nDuffkhronos: that said, I'm very happy to not be in your shoes right now.
03:42.40darkladywolfHi folks.  Anyone here a DUNDi expert?  Or at least willing to help out someoene who knows they've got it wrong, but can't work out where?
03:44.19khronosProblem is I'm basically being a service provider to homes.
03:44.26khronoson islands.
03:44.39khronosThe only way they have to get to the world is over satellite.
03:48.04khronosBasically here's the situation.
03:48.39khronosWe've got a bunch or rich people who come from the us and they have homes on this island and they want to have the same services there as they do when their in the US.
03:49.13CunningPike~wglwat
03:49.18jbotwell, wglwat is well, good luck with all that
03:49.22CunningPike:)
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03:53.57*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
03:54.59jqlit's good to be rich
03:57.15*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
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04:03.07*** join/#asterisk JerJer[mobile] (n=jj@199.45.11.90)
04:05.10JerJer[mobile]anyone seen a compile error in menuselect/mxml   ?
04:05.34JerJer[mobile]make[3]: Entering directory `/usr/src/asterisk/menuselect/mxml'
04:05.34JerJer[mobile]autoconf
04:05.35JerJer[mobile]autoconf: no input file
04:07.04mostyhow can i get asterisk to transcode my prompts to g729 and save the files?
04:07.40nDuffkhronos: would they be actively unhappy about getting and sending their faxes via computer?
04:07.53nDuffkhronos: because if you can give them a TCP connection to a hylafax server, that's much much less trouble.
04:08.19nDuffkhronos: if that wouldn't fly -- then it's store-and-forward.
04:11.40[TK]D-Fendermosty, you mean to transcode prompts you already have?
04:12.22mosty[TK]D-Fender, yes
04:13.50mostyi have lots of files, i don't want to use that online util
04:14.08[TK]D-Fendermosty, don't know a really "convenient" way, sorry
04:15.25*** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca)
04:16.14ChkDigitHey guys, I had Asterisk<->Polycom presence working when I let a client.  I walked out the door, and it stopped.
04:16.34mostythere are links to a dead website which had a "convert" cli command, this is as close as i have found to a good solution
04:16.54ChkDigitWhat is the reason for Asterisk to report that a set is Idle, despite the fact the CLI says it is rining?
04:17.02ManxPowerconvert is a 1.4+ feature, I believe
04:17.35ManxPowerChkDigit: the hint and exten do not match
04:17.49mostymanxpower: do you have 1.4 installed? could you see if it includes res_conv.so ?
04:17.58ManxPowerChkDigit: What 1version of Asterisk?
04:18.11ChkDigitThey match.  However, I just did /etc/init.d/asterisk restart and things look okay...
04:18.19ManxPowermosty: Gads no, I currently have no timetable to update my machines to 1.4
04:18.32ChkDigitIt is Asterisk Bus.Ed. 1.3.0.b1
04:19.17ManxPowerChkDigit: you'll have to talk to Digium for support of that
04:20.47*** part/#asterisk darkladywolf (n=root@wolf.tpgi.com.au)
04:22.41ChkDigitBelieve me. I did.,
04:23.09ChkDigitSpent 2 hours on tier 1, and 3 on 2 with angler.
04:23.20ChkDigitSolved the problem, then it crept back.
04:25.43*** join/#asterisk JT (n=jon@unaffiliated/jt)
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04:45.41threatwell that was fun
04:47.33threatiax refused to work
04:49.45threatis there a iax work directive for asterisk?
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05:01.39ChkDigitIn VoiceMail() what would cause it to not respond to the caller pressing 0 or *?
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05:06.42*** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus)
05:07.15Keltusis there an actual hardware limitation to regular computer modems that make them not work with asterisk? or is it purely software
05:08.41mitcheloci imagine it's software
05:16.23nDuffI've seen a discussion of why a voice modem is inadequate for telephony, but that was long ago and far away, and I certainly couldn't come up with a link right now.
05:16.46nDuff...for that matter, it might have been in something published by way of dead tree.
05:17.01Keltushah
05:17.13KeltusI'm trying to avoid buying new hardware for my voip stuff
05:17.23Keltusso I want my phone line to hit my computer
05:17.32nDuffI mean, there *is* chan_modem
05:19.36JTyes
05:19.36nDuffahh. *was* chan modem, but it was removed because the hardware generally couldn't do what chan_modem asked it to.
05:19.36JTcheap modems are cheap shit
05:19.37JTwith poor components like hybrids
05:19.37JTand you need a driver for the device
05:19.37JTand it must be a soft modem for it to be even possible to make a driver
05:19.37Strom_C~ygwypf
05:19.50jbotsomebody said ygwypf was You Get What You Pay For.  If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.
05:19.50JTjust spend a couple of dollars and get something proper
05:19.50*** join/#asterisk Bazy (n=bazy@exodus.upctm.ro)
05:29.20*** join/#asterisk JoelSolanki (i=Joel@202.160.161.94)
05:29.33threatJT, hi
05:29.33JThi
05:29.33threatJT, I require your assistance
05:29.33threatI am trying to setup IAX between two asterisk servers, however I keep getting authorisation denied messages
05:29.33threatis there some type of ACL settings I need to configure?
05:29.34threatwould you like a pastebin dump of both of the servers iax.conf and extensions.conf file?
05:29.34KeltusJT: what do you guys recommend?
05:29.34JTKeltus: what is the scenario?
05:29.34nDuffthreat: picking out a single individual and demanding assistance is not exactly good ettiquette. Actually, neither of those actions is.
05:29.35KeltusJT: I want to run a customer service center
05:29.35nDuffKeltus: How many lines?
05:29.35Keltuswe have about 10 CSRs rotating, and we get about 200 calls a day
05:29.35Keltuswe have 1 toll free number
05:29.35Keltusanything else is up in the air
05:29.35nDuffKeltus: I'd be getting a proper PRI and a T1 card (or E1, if appropriate for your region) at that point.
05:29.36Keltuscan you explain what those things are and what they ?
05:29.36Keltusthey do*
05:29.36nDuffKeltus: a PRI is where your phone company gives you a single drop with 24 voice channels (or maybe fewer voice channels and runs data over the rest)
05:29.36nDuffKeltus: a T1 card goes in the Asterisk box; you plug the PRI into it.
05:29.36threatnDuff, ok
05:29.36threatnDuff, how do you suggest I go about this then?
05:29.37threatI singled JT out since he helped me earlier today
05:29.37JTnDuff: ask the channel
05:29.37JTerr
05:29.37JTthreat:
05:29.39JTpeople don't like being singled out
05:29.39JTor dobbed in
05:29.59Keltushmmm okay, what would the data channel be for?
05:30.05Keltuswe have standard internet access
05:30.13KeltusI think T1 lines already, over ethernet
05:30.19Keltuscan I just use those?
05:30.28*** join/#asterisk dhakatel (n=ashrar@58.65.224.5)
05:30.32nDuffthreat: about what -- asking for help? There's plenty of documentation on that; although it's not quite IRC-centric, http://www.catb.org/~esr/faqs/smart-questions.html is a good place to start.
05:30.54nDuffKeltus: who are you getting your T1 lines from? Are they also a telco?
05:31.17nDuffKeltus: They might be able to split it up and put some voice channels in, but it very much depends on who they are; I'd expect they'd want to sell you something slightly different.
05:31.48nDuffKeltus: If the full T1's bandwidth is more than you need, you might talk to them about splitting it into 12 voice lines and the other half data.
05:32.10threatnDuff, ok great
05:33.09Keltusour T1 line is from XO communications
05:33.22Keltusokay, I didn't know about the voice + data package
05:33.23nDuffI don't know them.
05:33.26KeltusI'll check that out
05:33.57jqlit's usually listed under "fractional" service
05:34.41Keltusgotcha
05:34.47threatok here we go, general questions 1) how do I setup IAX asterisk - asterisk 2) How do I stop POTS incoming calls from ringing analog phone when the caller has hanged up
05:38.26nDuffthreat: (1) - that's a pretty general question. Have you read http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers?
05:39.13nDuffthreat: (2) - err... they *do* stop. For me, anyhow. Presumably something's broken on your site if they don't, and that needs to be localized before it can be debugged very effectively.
05:42.04Keltuswhat card would you guys recommend for just some testing? ie. something I can do at home and configure asterisk until I want to make the big step?
05:52.45threatnDuff, hmmm
05:53.15threatnDuff, well if I ring the asterisks server on the POTS line then hang up the asterisk server keeps ringing the internal analog lines
05:53.18JTKeltus: you don't need any cards to do testing
05:53.23JTKeltus: you can just use VoIP
05:53.24threatwhat setting would control this?
05:53.27threattimeout?
05:54.50nDuffKeltus: the only thing you might buy hardware for in test-phase is if you want to decide on what kind of phones to use.
05:56.22*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:57.08*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
05:57.22JoelSolankiHi all. was reading on asterisk 1.4 and found that it doesnt required re-invites.  it tell like this. ' if we know at call setup  that we can release media, we will do that directly '
05:57.34JoelSolankiis this early media disable stuff ?
05:57.44THX2000Is there a way to get asterisk to store voicemail wav's and info w/ r/w access rights for users?
05:58.10mostyTHX2000, what are you trying to do?
05:58.51THX2000get php to parse the info stored in msg0000.txt
06:00.11Keltuswell I want to be able to test calling my toll free number --> ringing out yahoo messenger over gizmo
06:00.35mostyTHX2000, running php from apache, cgi, cron, what?
06:00.40Keltushow would I do it without any hardware? the toll free number needs to redirect to an actual phone #
06:00.45THX2000thttpd
06:01.02THX2000php-cli
06:01.11JTKeltus: you can get DIDs over voip
06:01.24JTnot as reliable as a PSTN one, but good enough a lot of the time
06:01.26mostyTHX2000, you could make the voicemail dirs setgid
06:01.54mostyTHX2000, and put them in group that the thttpd user is a member of
06:02.28nDuffKeltus: does it need to be the same toll-free number you'll be using in production?
06:03.02Keltusyea, I want to make sure it works end-to-end
06:03.08THX2000that kinda makes sense. Im in a bit over my head here, but i guess thats how ya learn :P
06:03.24Keltusbasically, be able to handle 5 calls per day to our customer service rep, as a beta test
06:04.02Keltuswhat kind of hardware would I need to just have a very small number of calls? I have an old linux computer I can set up asterisk on
06:04.13*** join/#asterisk SwordManX (n=sword@ip70-161-179-101.hr.hr.cox.net)
06:04.20Keltusand then I'll be using gizmo -> yahoo messenger for the VoIP for a free test call
06:05.09nDuffKeltus: ahh. For just one line, an IAXy might be your cheapest bet.
06:05.59nDuff...actually, wait, that's the wrong end.
06:06.18nDuff...that'll talk to a phone, not to an outside line. (I never keep "FXO" vs "FXS" straight, but that's the distinction)
06:06.21JTIAXy isn't that cheap anyway
06:06.39nDufftrue.
06:06.49nDuffSomething from Sipura, then.
06:07.03nDuffI think they've got an ATA that does both FXO and FXS.
06:07.24Keltusis a IAXy a FXS device?
06:07.26JTSPA-3102 has 1 FXS and 1 FXO port
06:07.30JTKeltus: yes
06:07.34KeltusI just need a FXO don't I
06:07.42*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:07.46nDuffKeltus: depends, what kind of phone(s) will you be using?
06:07.50Keltussince for outbound, I will use gizmo and yahoo messenger for testing
06:07.53nDuffahh.
06:07.55KeltusI'll use speakers and a mic
06:07.56Keltusfor now
06:08.36nDuffI don't exactly advise that. Something built for telephony will give you proper echo cancellation, and otherwise sound a whole lot better to folks on both ends.
06:08.52JTnDuff: he's just testing
06:09.34nDuffwell, yeah, but if your support staff hate it during the test...
06:09.43Keltusshould I get them a headset?
06:10.01nDuff...but then, I guess whether it's a sell-it-to-the-staff test or a get-asterisk-working test.
06:10.09nDuffKeltus: I'd recommend it, yes.
06:10.21Keltusgotcha
06:10.30Keltusit's more get-asterisk-working but it would be nice if everyone was excited about it
06:10.52JTKeltus: yeah you must use headset, pc speakers and mic pretty much won't work
06:11.25Keltusand then you recommend SPA-3102 for the incoming call part?
06:11.40JTfor business a card is probably better
06:11.44JTlike a TDM400P
06:12.11JTbut as someone else suggested, if you have quite a few agents, you'll want PRI ISDN T1/E1
06:12.23Keltusyeah I want to just test a few calls on one line first
06:12.34Keltusto make sure this is what we want
06:13.08JTthe problem is you'll need a totally different card, but i guess an spa-3102 might be cheap enough to test
06:13.18JTotherwise you can get phone service over the Internet
06:14.05Keltusthe different between the two cards is the spa-3102 does incoming and outgoing right? so wouldn't the headset be good enough
06:14.13Keltusthe CSR will have computers, and plug their headsets in them
06:15.08JTnot really
06:15.20JTthe audio quality from a SIP hardphone like a Polycom is much better
06:15.45Keltusok
06:15.57KeltusI'll put in orders for those tomorrow. any other equipment we'll need? I read about the X100P card for asterisk calls.
06:15.59flendersKeltus: I have an SPA-3000 at home, and sound quality is not that good when using the FXO channel
06:16.20JTKeltus: headsets to suit the phone as well i guess
06:16.38JTKeltus: you should probably do a bit more research and testing before buying a whole bunch
06:16.42Keltusright. so SPA-3102 and a headset
06:16.57Keltuswell it prices to about $100 right now so it's not much
06:17.01JTyeah
06:18.21Keltusgreat
06:18.42Keltusthanks for all the advise
06:39.09*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
06:43.20*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
06:43.29mostyi have a strange problem with one-touch recording. it works with some calls but not others. sip->sip calls work, but sip->iax (->sip) calls don't work from the originating sip end
06:43.45mostyall my dial commands have wW in the options
06:43.52mostywhat could be wrong?
06:52.19*** join/#asterisk Ast001 (n=uros@77-105-44-230.adsl-2.sezampro.yu)
06:52.32Ast001hello
06:52.43Ast001I need your advice
06:53.23Ast001is it good to set priority to -19 to asterisk
06:53.30Ast001with renice -19 asterisk pid
06:53.35*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
06:54.00mostyare you having trouble with running asterisk at the default nice level?
06:54.07Ast001well I have
06:54.30Ast001My operators who works from hove and have adsl 512/128 can not hear properly musiconhold
06:54.48Ast001they can hear but it stops for 1-2 sec etc...
06:55.00JTyou must have rtp silence supression or a highly compressed codec like g.729
06:55.17Ast001no i dont use g729
06:55.19mostyi doubt that's load-related
06:55.32Ast001i use ulaw alaw gsm ilbs
06:55.33Ast001ilbc
06:55.43Ast001operators have xlite
06:55.56mostyturn off silence sippression in x-lite
06:56.07JTmosty: load related?
06:56.13*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.voop.net)
06:56.20*** join/#asterisk hellop (n=hellop@udp112969uds.hawaiiantel.net)
06:56.23JTyes but silence supression in x-lite only controls its stream to asterisk
06:56.35Ast001well ok
06:56.43Ast001i will do it
06:56.55mostyjt: i doubt the issue would have anything to do with the nice level
06:57.12JTmosty: ah yeah
06:57.17mostyAst001, does your pbx have a zaptel timer?
06:57.17hermulidoes anyone have an idea why app_mysql would leave connections open (sometimes) even when i call the closing thing from dialplan?
06:57.19Ast001is adsl 512/128 is enough for ulaw alaw
06:57.30JTyes unless it's over a bad dsl network
06:57.40JTyou need 85kbit/s
06:57.45mostyAst001, try with gsm
06:57.46Ast001i have pri isdn
06:57.57JTmosty: he's already tried though
06:57.58Ast001i dont know about zaptel timer
06:58.05Ast001I tryed gsm
06:58.16Ast001They can hear gsm but moh stops and continue
06:58.37JTmoh does the exact same thing?
06:58.37mostyhow long do the pauses last?
06:58.48Ast0011-2 sec
06:58.51Ast001to 4-5 sec
06:59.27Ast001silence suppresion
06:59.30mostyis it just moh that does that? does it happen when you're speaking?
06:59.32Ast001can not find that
06:59.35Ast001no
06:59.40Ast001they can speak well
07:00.29Ast001is that excacll named SILENCE SUPRESION ?
07:00.56JTmight be something similar
07:01.07Ast001use concealment ?
07:01.09mostyi don't know what x-lite calls it. don't worry, it probably wont help
07:01.15Ast001transmit silence ?
07:01.24JTyes
07:01.36Ast001transmit silence=yes
07:01.44Ast001at the moment
07:02.22Ast001now its now
07:02.24Ast001now
07:02.37Ast001transmit silence=no
07:02.57*** join/#asterisk af_ (n=getsmart@81-174-47-36.f5.ngi.it)
07:03.13Ast001it did not help
07:03.30Ast001pauses on moh continues on ilbc and gsm
07:03.43JTAst001: so there is a PRI on the same machine that provides MoH?
07:03.50mostywhat version of asterisk?
07:03.55Ast0011.2.17
07:04.01Ast001yes everything on the same machine
07:04.06JThrm
07:04.11JTweird
07:04.18JTmany channels?
07:04.35Ast001well about 10 channels work at the same time
07:05.09Ast001is this SIP related problem ?
07:05.16Ast001would it be better with mozphone and iax2 ?
07:05.22JTmaybe, who knows
07:05.28*** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
07:05.33JTrun rtp debug on a peer
07:05.42Ast001i tryed that
07:05.46JTand have a call on that peer, then put it on MoH
07:05.55JTdoes the flow or RTP packets slow during MoH?
07:06.07Ast001didnt noticed that
07:06.15Ast001enormous number of lines of type
07:06.29Ast001send rtp packet recieved rtp packet ...
07:06.42JTyes it sends a line for every packet
07:07.15Ast001rtp debug ip xx.xx.xx.xx ?
07:07.21JTyes
07:07.43JTsee if it sends less rtp packets during MoH silence
07:08.50Ast001only this line Sent RTP packet to 77.105.44.230:50419 (type 98, seq 44253, ts 2830616, len 50)
07:09.37JTthere should be more than one line, just take note of if the flow of packets is reduced during the problem silence
07:10.03*** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.3.121.revip2.asianet.co.th)
07:10.06Ast001this line continue to repeat indefinitly without stop
07:10.23Ast001until i do no debug
07:10.43JThrm ok, so the flow is probably unchanged by silence
07:10.52Ast001yes
07:11.00JTtry a different SIP client or phone
07:11.00BugKhaMhi, what's the best package used to recieve/send faxes? spandsp?
07:11.12JTBugKhaM: explain the problem a little more
07:12.07BugKhaMJT: I wanna receive faxes and send it out automatically  by email, just wondering what to use
07:12.30BugKhaMJT: http://www.voip-info.org/wiki-Asterisk+Fax+to+email is what I found
07:12.39JTi guess either spandsp or hylafax should work
07:13.18BugKhaMJT: application spandsp used to be ported with *, but not anymore?
07:13.51JTyeah only guaranteed to work with certain versions
07:13.56Ast001does it have something with irq priorities
07:14.07Ast001because eth0 is sharing resource
07:14.17JTAst001: with what?
07:15.49Ast001no
07:15.54Ast001its at  23:   66489101   IO-APIC-fasteoi   eth0
07:16.16JTerr so what's it sharing with?
07:16.16Ast001but its on 23 do i need to move it ti 2 or 3 ?
07:16.28JTno why would you
07:16.28Ast001no eth1 is sharing it is for LAN
07:16.32Ast001sorry
07:16.37Ast001ok
07:16.47JTi see no evidence of irq sharing from what you've pasted
07:17.01JTnow try running zttest whilst doing problem MoH calls
07:17.02Ast001yes eth0 is not sharing
07:17.12JTsee if the scores drop below 99.97%
07:17.16Ast001i made error confused eth0 with eth1
07:18.20Ast001-bash: zttest: command not found
07:18.32JTyou should install it then
07:19.06Ast001where can i found it ?
07:19.14Ast001is it part of zaptel ?
07:19.19*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
07:19.19JTyes
07:19.41Ast001well I did ./configure make make install during compiling zaptel
07:20.04Ast001why did it install that zttest ?
07:20.09Ast001didn't ?
07:20.16JTit might not by default
07:20.23JTjust check the documentation on how to do it
07:21.27*** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com)
07:21.43mattwj2005anyone of the 265 people in this room own a solar panel
07:21.45mattwj2005?
07:22.04mattwj2005:)
07:25.17*** join/#asterisk squall (n=squall@ns2.squallnetwork.net)
07:26.39Keltuswhat's the difference between a x100 and a x100p?
07:27.03JTone has a p missing
07:27.06mattwj2005digium cards?
07:27.28Keltusyea
07:27.50JTKeltus: one has a letter p missing
07:27.51mattwj2005no idea off hand
07:28.01Keltusfunny JT
07:28.02mattwj2005have you googled it?
07:28.04Keltusyeah
07:28.05mattwj2005:)
07:28.06Keltusnothing
07:28.15KeltusI'm thinking x100 is just shorthand for x100p
07:28.17mattwj2005www.voip-info.org ?
07:28.18JTKeltus: and it's the CORRECT ANSWER :)
07:28.25Keltusah
07:28.27JTthere is no such thing as an X100
07:28.44mattwj2005the p guy was off somewhere
07:28.45mattwj2005:)
07:28.47mattwj2005lol
07:28.52JTthe X100P is a crappy discontinued FXO card based off an old discontinued Intel winmodem chipset
07:29.18mattwj2005I think he had to use the bathroom
07:29.51Keltusyea. the one I purchased for testing is called  TDM01B
07:30.04mattwj2005sorry 2:30 am humor
07:30.11JTwhich is a form of the TDM400P
07:30.18Keltusyup. I know
07:30.28Keltusit was $150
07:35.14Ast001rtp debug ip does not give me anything when I use iax2 and mozphone
07:35.22Ast001and with mozphone its much better
07:35.31JTwell of course not, iax2 doesn't use rtp
07:35.45Ast001so maybe its rtp thing ?
07:36.03JTrtp or sip maybe or xlite
07:36.11Ast001i see
07:36.18Ast001then I will migrate to mozphone
07:36.18JTthat's a minimum off all the variables that have changed :P
07:36.49*** join/#asterisk santoshr (i=1063@203.199.110.93)
07:37.06mattwj2005I am thinking about using the natural Minneasota sunlight in the Summer
07:37.28mattwj2005thinking about buying a solar panel....assuming my apartment doesn't mind
07:37.32santoshri am using asterisk 1.2.9 .. is thr a way to tell a call to goto a context,s,1 after the "S"  time in Dial
07:37.47JTmattwj2005: for what purposes?
07:38.19santoshrso when the call disconnects it goes to a specific playback or something
07:38.29mattwj2005charge batteries
07:38.47mattwj2005be kind to the earth
07:38.50mattwj2005that kind of thing
07:39.00JTmattwj2005: batteries for anything in specific?
07:39.17mattwj2005well I own a 80 GB iPod
07:39.45mattwj2005it would be cool if I had battery power for it this summer
07:39.57mattwj2005I have a few other usb devices
07:40.10mattwj2005I have a nice radio that runs on AA
07:40.51mattwj2005I also have a gameboy that takes AA
07:41.18mattwj2005energy cost in the US are always so expensive during the summer
07:41.45mattwj2005and I am a hobbist at heart
07:42.26flendershaha, a solar panel to recharge your ipod
07:42.29flendersthat was a good one
07:42.38mattwj2005lol
07:44.30JTthe solar panel would probably cost more, but it could be a fun project :)
07:44.32Keltusanother newbie question - if I have 2 FXO ports, do I need 2 wall jacks that each have a different line on it? or is there some sort of splitter
07:45.06flendersKeltus: you need 2 wall jacks
07:45.12mattwj2005~$100 get some good voltage
07:45.18mattwj2005that is usd
07:45.27santoshr. is thr a way to tell a call to goto a context,s,1 after the "S" time in Dial
07:45.30JTmattwj2005: wattage you mean?
07:45.39Keltusflenders: do telcos usually give you two different numbers for each jack? or one number for the two jacks
07:45.51mattwj200515 W
07:45.54flendersyou want to split one single line?
07:45.59JTKeltus: that would depend if they're different lines
07:46.00flendersor you want 2 lines?
07:46.02mattwj2005http://www.amazon.com/Sunforce-50032-Solar-Battery-Charger/dp/B0006JO0X8/ref=pd_bbs_sr_4/002-9570840-7785656?ie=UTF8&s=automotive&qid=1177312165&sr=8-4
07:46.05Ast001its much better on iax2
07:46.06*** join/#asterisk Bananaskin (n=Banana@81-86-102-88.dsl.pipex.com)
07:46.18KeltusI have a 1 800 number
07:46.29JTAst001: must be a bug in something, somewhere.....
07:46.33Keltusif I have one line and split it, will it alternate the jacks?
07:46.43KeltusI mean, I want to be able to take 2 calls simultaneously
07:47.09Ast001well it was default isntalation of Asterisk 1-2-17 and zaptel for 1-2 and libpri
07:47.27mattwj200541 inches might be too big though
07:47.40Ast001it is strange it worked perfecly before I moved to optical cable
07:47.41mattwj2005~ 1 meter
07:48.10JToptical cable where?
07:48.30Ast001at headquarters where server is
07:48.51flendersKeltus: you need 2 lines
07:49.17flendersKeltus: and 2 FXO channels
07:49.53JTAst001: yeah, what does the fibre do?
07:49.55flendersmattwj2005: massive panel
07:50.06Keltusalright, so I guess I need to call the telco to set this up. it's not something I can do in hardware, right?
07:50.08JTthat's like saying "I connected a piece of copper to my server"
07:50.37mattwj2005maximum output at 12 V....1.5 A
07:50.42Ast001I had cable link 2mbit/sec
07:50.48Ast001cable internet
07:50.56JToh, INTERNET, ok
07:50.59Ast001but link was down to meny times
07:51.14Ast001and ISP said optical cable is much better
07:51.18JTmaybe your new isp does something funky to rtp, who knows
07:51.18mattwj2005using ohms law
07:51.20JTsame isp?
07:51.21*** join/#asterisk qdk (n=qdk@213.150.62.32)
07:51.21JThrm
07:51.21Ast001stable
07:51.36Ast001no isp is the same
07:51.43Ast001new is just optical cable
07:51.53JTAst001: is it optical ethernet?
07:52.02Ast001they say no barrieras no firewall nothing
07:52.18Ast001some box is connected to server with ethernet
07:52.25JTbut you still can't tell me where along the line the fibre comes into it
07:52.48mattwj2005I am guessing that'll be too big
07:53.04Ast001other wire is tv cable signal and goes to tv cable box
07:53.19mattwj2005here is a little more conversative solution
07:53.21mattwj2005http://www.amazon.com/Sunforce-50022-Battery-Trickle-Charger/dp/B0006JO0TC/ref=pd_bbs_12/002-9570840-7785656?ie=UTF8&s=automotive&qid=1177312165&sr=8-12
07:53.33Ast001don't know really
07:53.45Ast001but server can go to internet ok and ping is ok too
07:53.49*** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com)
07:53.52Uatec_Greetings
07:54.11Ast001no firewalls
07:54.18Ast001connection speed is ok too
07:54.24Ast001said speakeasy.net/speedtest
07:54.40JTAst001: have you even seen the fibre cable?
07:55.11mattwj2005416 mA
07:55.26mattwj2005@ 12 V
07:55.31*** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it)
07:57.03flendersAst001: so, using cable it worked, with "fiber" it doesn't?
07:57.26Ast001with cable it worked great
07:57.39Ast001now it is working with problems
07:57.40*** join/#asterisk psk (n=psk@golia.caltanet.it)
07:58.05*** join/#asterisk bintut (n=bintut@203.125.63.150)
07:59.49JTAst001: is the fibre the ethernet connection between the modem and the PC or what?
08:00.04mattwj2005Ast001 do you have any network certs?
08:00.44Ast001between pc and that box is ordinary ethernet cable
08:00.57Ast001used before
08:00.57JTso you have no idea where the fibre is? awesome
08:01.03Ast001no
08:01.09mattwj2005there is a short list of things that kill voip quality
08:01.26nemskilatentcy
08:01.32JTadd magic optical cable to it
08:01.32nemskicpu load
08:01.33Ast001well isp sets all of that
08:01.51mattwj2005lack of bandwidth......people that set up bad qos or traffic shaping......greedy telephone companies.......and bent cables
08:01.59Ast001I just put ethernet cable in that box and configured it on server
08:02.22JTso you have no clue about how you connect to the Internet? great
08:02.33mattwj2005you can also add poor cables...loose cables....cables without clips etc
08:03.05Ast001throug tv signal I guess
08:03.10JTi saw a cable in a datacentre that was a flat 8 core cable between switch and server
08:03.17JTall i thought was "what an idiot"
08:03.40mattwj2005I make 46.1k usd doing networking
08:03.47flendersAst001: do you have a cable modem?
08:03.49mattwj2005I love to help for free if I can
08:03.52JTAst001: so the coaxial cable tv network
08:03.59JTthat's not fibre :)
08:04.28flendersmattwj2005: is that your salary?
08:04.54mattwj2005indeed
08:05.15flendersis that a lot in the US?
08:05.20Ast001i have some strange box wich is bigger that cable modem i had for previous connection
08:05.29mattwj2005I am not rich
08:05.32mattwj2005I am not poor
08:05.42flendersand you're not starving I guess
08:06.04mattwj2005nope
08:06.25mattwj2005every time I go to the grocery store I try to give to the hungry
08:07.21mattwj2005no man on this planet should go hungry if he has neighbors and a city or town
08:07.43flenderstrue!
08:07.48flenderswhat state are you in?
08:08.04mattwj2005Minneasota....we touch Canada!
08:08.37JTdoes canada enjoy being touched?
08:08.54flendershahahha
08:09.09mattwj2005I am not Canada
08:09.11*** join/#asterisk izaak (n=izaak@modemcable097.151-202-24.mc.videotron.ca)
08:09.20mattwj2005you'll have to ask them
08:09.24JTheh
08:09.35*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:11.11mattwj2005we are pretty safe...we touch just the top part
08:11.24mattwj2005*with the top part
08:12.04mattwj2005http://en.wikipedia.org/wiki/Minnesota
08:12.14mattwj2005http://upload.wikimedia.org/wikipedia/commons/thumb/5/54/Map_of_USA_MN.svg/286px-Map_of_USA_MN.svg.png
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08:18.30mkl1525Hi, I've got agents and I'd like to distinguish if a direct call is made or the queue application is calling so that I can set different Dial() parameters. But haven't found any variable that I could use for this, setting my own variable didn't help either - any hints?
08:20.59dhakatelyap
08:21.42dhakateli used it
08:22.37dhakatelwhat type of reduncy u need
08:22.55JTdhakatel: wrong channel?
08:22.59dhakateljust fail over and take over
08:23.06dhakatelor data replecation
08:23.16JTdhakatel: WRONG CHANNEL
08:24.06mattwj2005anyone have any good suggestions for AA to USB adapters?
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08:26.47santoshrcan the call flow be controlled after the call disconnects after S(time) in Dial
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08:28.33flendersmattwj2005: wanna charge your ipod using AA?
08:28.33*** join/#asterisk saftsack (n=oliver@p54a72e07.dip0.t-ipconnect.de)
08:28.45mattwj2005yup
08:29.02flendersI think apple has something
08:29.22flendersI think it plugs into the ipod connector
08:29.45mattwj2005see here is my idea
08:29.57mattwj2005solar panel charges AA batteries
08:29.59tzafrirI saw some in ebaw. But I really don't understand how is this related to *
08:30.07mattwj2005when done I charge my iPod
08:30.12tzafrirUnless * is considered to be a wildcard
08:30.35flenders:D
08:30.43mattwj2005I am sorry I am off topic
08:30.44mattwj2005:(
08:31.22JTnot a bad off topic though
08:31.51mattwj2005I am saving money on energy though
08:31.51mattwj2005probably shutdown some computers too
08:31.51mattwj2005:(
08:32.27mattwj2005I will be running a form of an asterisk server...even if it is virtual
08:33.25mattwj2005speaking of Asterisk...
08:33.35mattwj2005I can get one up and running without using a make samples
08:33.38mattwj2005:)
08:35.23*** join/#asterisk Nickle (i=Loots@c-71-204-146-212.hsd1.ca.comcast.net)
08:35.58NickleCan anyone recommend a low cost yet good quality outbound service?
08:36.23NickleUS maybe Canada/unlimited calling.
08:36.23mattwj2005www.teliax.com is my favorite service provider
08:36.33NickleAhh, good price?
08:36.43mattwj2005~$25 usd
08:36.47NickleNot bad
08:37.12mattwj2005good quality...they are looking to hire people too
08:37.17mattwj2005ATTN ROOM
08:37.21NickleHeh
08:37.26NickleYou work for them?
08:37.29Nickle;P
08:37.47jqlshill!
08:38.08mattwj2005nope
08:38.43mattwj2005just say if your between Diet Mt and a desecent Internet connection....
08:38.48Nicklethat $24 plan is unlimited outboud and incomming eh
08:39.10mattwj2005yeah...but read the fine print
08:39.17Nicklemy AIX server is in LA
08:39.34NickleIAX
08:39.35mattwj2005they give you so many mins and then they charge over that amount
08:39.36Nicklerather
08:40.23mattwj2005they have been pretty reliable and good quality
08:40.25Nickleis there a free service which will allow me to call toll free numbers?
08:40.36Nickleactually, here's the scoop
08:40.43mattwj2005not sure
08:40.54Nicklei recently was promoted, and i work with a global team. i just need to dial ATT teleconference
08:40.57Nicklebut i have a CELL phone
08:41.07Nickleand i work out of my house 4 days a week
08:41.13Nicklehaha
08:41.31Nicklebut i have my asterisk server, so i figure, what the heck
08:43.48mattwj2005ISBN 0-596-00962-3
08:43.54mattwj2005read that
08:44.35mattwj2005it has a lot of good info on what asterisk can and cannot do
08:44.48*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
08:45.10mattwj2005I have a copy...haven't read it yet
08:46.04Nickleso there's no toll free forwarding service
08:46.06Nickle:\
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08:46.59mattwj2005http://www.tollfreegateway.com/
08:47.12NickleNice
08:47.14NickleThanks!
08:47.23mattwj2005your welcome
08:48.51mkl1525Is there any way to proceed with a queue call after the agent hangs up? tried the h extension -> works but can't use the channel with an ivr anymore, Dial with g parameter executes the parameters immediately after the caller gets in the queue and not when he leaves the queue - any further suggestions?
08:57.36*** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it)
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09:15.21FreezeSI've got a problem with a TE110P. I'm getting: ZT_CHANCONFIG failed on channel 25: No such device or address (6)
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09:24.07tzafrirFreezeS, is it T1 or E1?
09:24.35uskihi, i have a very stupid question for you; when using authenticate in a dialplan, how the hell do i enter the code from my phone ? it says "enter the code followed by square" but there is no "square" key ! and it's not # or * so what is it ?
09:24.36tzafrirsounds like you have T1 and trying to ocnfigure it as E1
09:24.53FreezeStzafrir: the jumper is set for E1
09:25.26tzafrirFreezeS, and is that reflected in /proc/zaptel/1 ?
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09:26.21uskinoone uses Authenticate ???
09:27.18tzafriruski, "#"?
09:27.28uskididn't work, but i can try again...
09:27.42tzafrirI don't use Authenticate, BTW
09:27.55hermulii did, for a moment, and it was #
09:28.53FreezeStzafrir: can I paste it to you on private ?
09:29.00tzafriryes
09:29.20tzafrirFreezeS, youy can also pasebin it
09:29.55uskihmm ok, that's strange, it seems that no audio is received from my SIP channel in fact, so that'd explain why it doesn't work, i think it's just timeouting
09:29.56*** join/#asterisk nettie (n=nettie@ns.coolgadgets.it)
09:29.58uskithanks ;)
09:30.20FreezeStzafrir: is there a way to check from software the jumper setting ?
09:30.21nettiehey tzafrir at the end I wasnt able to fix it at the end :(
09:30.36nettiestill getting IRQ #16 disabled and crash
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09:32.10zeeeshhi
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09:35.50uskiany idea of the way i can check if my asterisk server received something from the sip server
09:38.45uskiis there something special that i should use to enable DTMF recognition ?
09:42.34uskifixed: i needed to use dtmfmode=inband
09:42.52Uatec_uski, thankyou for sharing your solutions with us.
09:44.06hermuliI'm still wondering the mysql connections don't time out :P
09:44.14hermuliwhy*
09:45.37hermulii have no idea what to try next...
09:47.17uskianyone knows why I can't place an outgoing call with DISA ? if I use Channel: SIP/provider/number in a call file it works so the setup is good, but I can't place a call with DISA. I have the tone, but as soon as I enter a number the tone changes to a "busy" tone or sth like that. Any example extensions.conf file for this ?
09:47.47uskimaybe i need to prefix the number with 9 or so ?
09:51.33*** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no)
09:56.06FreezeStzafrir: that was it, the jumper was set in the wrong way
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09:57.42sebadelhello, is this the right room foor AADK issues ?
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10:12.16creativxwoah is this a room
10:12.29sebadelhello
10:12.36sebadelany experience with AADK ?
10:12.55neoalexdoes anyone know of any service allowing unlimited calling to the US for the same price as skypeout (29.95/year)
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10:22.17sebadeldoes anyone know why save_config could hang on a brand new AADK
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10:36.08uskiany idea of how I can place an external call using a SIP channel with DISA ?
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10:58.59mostyanyone good at debugging one touch recording? it works on some calls for me, but on internal -> external calls only the external end can turn it on/off. all my dial commands have wW though
10:59.10*** join/#asterisk Fibres (i=Fibres@cpc2-leic3-0-0-cust157.lei3.cable.ntl.com)
10:59.13FibresHi all
11:03.11*** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net)
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11:16.06FibresIm wondering can anyine help me. I need to provide a gateway to allow a PRI ISDN based pbx to connect to voip.
11:19.27*** join/#asterisk SoftIce (n=bongo@dsl-242-115-118.telkomadsl.co.za)
11:19.28SoftIcehi
11:19.49SoftIcelisten, if I have asterisk realtime, and I just want sip.conf in the db, do I also need extensions.conf
11:19.54SoftIceor can extensions.conf be read locally?
11:20.17Uatec_AADK?
11:25.02DrukenHMESoftIce: nothing has to be in realtime...
11:25.15DrukenHMEit's whatever you want... realtime is just a choice... an option
11:25.37SoftIceDrukenHME: yes but what im asking is can just sip sit in realtime
11:25.40nemskiit's all relative
11:25.43SoftIceor if i wanted sip would i need the extentions too ?
11:25.55SoftIceall i want is sip.conf
11:26.02SoftIcecan that be done by itself, yes/no ?
11:30.28DrukenHMEyes/no
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12:02.08uskiassuming that my SIP provider supports emitting more than one call simultaneously, is one context enough in SIP.conf, or should i specify one context per outgoing slot ?
12:03.17*** join/#asterisk sebadel_ (n=seb@162.36-246-81.adsl-static.isp.belgacom.be)
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12:05.53*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
12:07.41DrukenHMEuski: one context is fine...
12:08.29sebadel_hello, I have a brand new AADK but the webinterface is very unstable
12:08.48sebadel_I'd like to know how to save my config via the SSH console
12:08.48DrukenHMEwtf is aadk ?
12:08.59sebadel_Asterisk appliance Develompment Kit
12:09.11sebadel_save_config is hanginf
12:09.14sebadel_hanging
12:09.37sebadel_is there any better web interface for it ?
12:09.42uskiso... "SIP/xxx is circuit-busy" means that my operator refuses to send another call, right ?
12:09.44*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:09.51DrukenHMEoh... the nice sexy machine that's way over priced....
12:09.57sebadel_indeed
12:10.32DrukenHMEsebadel_: i know nothing about it, except that tried to get me to buy one... and i wasen't havin no part of it
12:10.46DrukenHMEuski: or the line your calling is busy...
12:11.11sebadel_I had t buy one for a customer
12:11.22sebadel_and it doesn't work as expected
12:11.31sebadel_it's supposed to be out-of-the-box
12:11.35DrukenHMEdoes anything really work as expected?
12:11.37uskino it isn't, this happens immediatly after a Dial :( so they don't want me to send another call, sh** (thx DrukenHME ;))
12:11.56sebadel_not really
12:12.12GeertI've got a call (in show channels)
12:12.17sebadel_I get the web interface, the SSH prompt but I can't save anything
12:12.24Geertwhich is stuck, how do I stop/kill it?
12:12.33DrukenHMEgeert, restart
12:12.35sebadel_and the webinterface crashes every time I change anything
12:12.44Geertnot an option, there are 23 current calls :p
12:12.54DrukenHMErestart when convient?? :)
12:13.16Geertoh, that'll work :p
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12:13.49jovannottisomeone has tested TC400B with G723 succesfully ?
12:14.19GeertDrukenHME: doesn't work, the call hangs so asterisk won't restart :)
12:14.29DrukenHMEsebadel_: you'd probably be better to go into the ast-dev or call asterisk tech over that thing??
12:14.33*** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)
12:14.44DrukenHMEoh yes... too true...
12:16.17DrukenHMEGeert: is it a zap interface?
12:16.31Geertyes
12:17.42[[blah]asfdgot a new error message today:
12:17.42[[blah]asfdConnected to Asterisk <Version Unknown> currently running on No more connections                                                                 allowed
12:17.42[[blah]asfd<PROTECTED>
12:17.42[[blah]asfdNo more connections allowed
12:17.42[[blah]asfd*CLI>
12:17.48jovannottiI have tested TC400B, it only works for me in G729, in G723 only in one direction, could someone help me please
12:17.56[[blah]asfdanyone know what happened
12:18.00[[blah]asfd?
12:18.08DrukenHMEGeert: zap destroy channel work ?
12:18.16*** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net)
12:19.37[TK]D-Fenderjovannotti: Virtually noone here has a TC400.  Go call Digium support for it.
12:19.44GeertDrukenHME: I just did "restart now"
12:20.00DrukenHMEthat'll work :)
12:20.44jovannottithanks Fender, I am trying to tall with them
12:25.54JTtalk?
12:26.43jovannottitalk, sorry, but since yesterday I am trygin to call but anyone answer the phone
12:27.01JTkeep in mind US business hours :)
12:27.05JTit was the weekend
12:28.04frigidzephyrthey are open 7:00AM to 7:00PM ,  CST
12:28.19frigidzephyror CDT right now i think
12:29.52*** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr)
12:30.01jovannottilet me try right now , thanks
12:30.11*** join/#asterisk ecze (n=ecze@eczema.ecze.com)
12:34.17jovannottithis is the address
12:34.17jovannottiDigium, Inc.
12:34.17jovannotti150 West Park Loop, Suite 100
12:34.17jovannottiHuntsville, AL 35806
12:34.17jovannottiUnited States
12:34.47jovannottiwhat rime do you think is ther right now ? because I am in the line , waiting for assistance ,,, some minugtes ago
12:35.34nasls_lsado you know any good vo-ip provider in Greece ?
12:35.49nasls_lsaon in Europe ..
12:36.25florzwell, in .de, there are quite a few "acceptable" ones ...
12:37.03florzhave a look at the voip-info.org wiki
12:38.31frigidzephyri am in huntsville where digium is, its 7:37 AM
12:38.32nasls_lsathanks
12:38.32eczere
12:38.32nasls_lsanai  ?
12:42.32jovannottiok, nobody pickup the phone, I am waiting for aprox 10 minutes on line :(
12:42.43eczeYesterday I was talking about a bug in the old deprecated chan_modem channel. Finaly I have posted the bug in the bug systems from digium but right now I need to apply the best solution for testing purpose...
12:51.15Uatec_jovannotti, it's 7.50 abouts i think
12:51.20Uatec_maybe earlier
12:52.06Uatec_i checked
12:52.15Uatec_it's 0751 hours there
12:53.16JTmaybe he doesn't understand
12:53.16JTthere's probably 1 person there
12:53.18JT:P
12:53.22Uatec_and they start at 7am
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12:57.18jovannottiok ok I'll try later, I already posted in digium support he problem too
12:57.49mostystay on the phone
12:58.09mostyi've emailed digium's support a number of times, and never received any response
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13:05.13uskianyone using the "failann" macro to say the reason of a failed call ? it seems that my version of asterisk (1.2) doesn't have this macro builtin
13:05.22uskias a lot of people are using it i assume it's builtin
13:05.34blitzrageuski: nothing is "built in"
13:05.39blitzrageyou have to create your dialplan
13:06.20uskiyea ok, thanks
13:06.29blitzrageyou may be able to find a copy of that macro somewhere on the Internet
13:07.18uskii tried, everyone uses it, and no one seems to have it, thus my question ;)
13:07.27blitzragewho is everyone?
13:07.29blitzrageI've never heard of it
13:07.29uskii bet that they all copied the line from somewhere and they didn't test
13:07.34uskiwell at least 20 people
13:07.42blitzragefrom where?
13:07.57blitzragesounds like something that probably came from trixbox or something
13:08.04uskiah yea that's possible
13:08.23[TK]D-Fender~trixbox
13:08.29jbotTrixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons.  It is these things on top of  which make it seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/
13:08.29blitzrageits the most probable
13:09.10uskiok thanks, i'll try to see if i can find that macro, else i'll create it myself with several Goto and Playback
13:09.26uskiBTW, the link provided by jbot is out of date (404)
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13:19.23[TK]D-Fender~trixbox
13:19.25jbotTrixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons.  It is these things on top of  which make it seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/
13:19.36[TK]D-Fenderhrm
13:19.57[TK]D-Fender~trixbox
13:19.59jbotTrixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org
13:20.01[TK]D-FenderThere
13:22.02JT[TK]D-Fender: you had to allows the unclean presence of trixbox.org grace your pc? :P
13:22.20*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
13:23.00[TK]D-FenderJT : No, I jsut fix the bot messages on where to go for stuff so we can more efficiently dispatch trolls.
13:23.45*** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk)
13:23.45JTheh fair enough
13:24.21*** join/#asterisk nettie (n=nettie@ns.coolgadgets.it)
13:25.04[TK]D-FenderJT : And I have considered getting a spare PC to test things like Trixbox / * GUI, etc on that is not my home server for the purpose of perhaps being able to support them (business is business).
13:25.38JTwglwat ;_
13:25.41JT;)
13:26.03*** join/#asterisk Muhadib (n=passaro@83.240.150.187)
13:26.37nettiehi guys, what's the most common sip users / extensions dialplan naming strategy please? how are you naming sip phones, users and extensions? you just user numerical values or there's a known/suggested "sanity checked" :) strategy please?
13:26.39[TK]D-FenderJT : Yeah, you know you won't see ME around here asking for help with it :)
13:26.54JTheh
13:26.57*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
13:28.12blitzragenettie: In my vPBX environment I just name it [username#vpbx]
13:28.39blitzragenettie: don't use extension numbers though -- you could use MAC addresses, or some unique username, or a combination of things
13:28.58nettieso, phones mac address as username?
13:29.02nettiein sip.conf
13:29.13blitzrageit really depends on what you're doing and what makes sense... there isn't really any official way of naming your extensions
13:29.24nettiethen variables with name of the users in extensions.conf and related mac address
13:29.38nettieand then real extension pointing to the variables?
13:29.38blitzragenettie: sure, you could... I personally don't because I find a mac address conveys very little information when looking at it
13:29.47blitzragenettie: sure, you got the idea
13:30.13blitzrageas long as it's a unique value, that's pretty much all that matters
13:31.58*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
13:32.22defsdooranyone use aastra phones know how to stop it recording missed calls for group rang calls ?
13:32.58*** join/#asterisk Muhadib (n=passaro@83.240.150.187)
13:33.40DrukenHMEi use aastra phones, but don't record anything...
13:33.41*** join/#asterisk shinao1 (n=shinao1@196.1.179.225)
13:34.38defsdoorit displayed n missed calls on the lcd display
13:34.47DrukenHMEor... you mean the caller id...
13:34.54defsdoorI'd rather it didn't if the calls was a group calls
13:35.17*** part/#asterisk captiancrash (n=jonmoore@70.159.118.70)
13:35.19MuhadibI need help with echo-cancellation issues... Is there anyone that could help me? please...
13:35.35[TK]D-Fenderdefsdoor: You can't  Period.  It doesn't know anything about calls it receives.  SIP phones are smart & don't assume anything about the servers sending them calls.
13:35.36mostydefsdoor, the phone probably can't tell the difference between "group" calls and singlular calls
13:35.42DrukenHMEdefsdoor: i don't think that can be accomplished
13:35.53defsdoorI thought so :(
13:36.23defsdoorI'll take it off the display altogether if I can
13:36.33defsdooreveryone has voicemail so important missed calls will be there
13:37.09DrukenHMEewhh... no caller id.... i don't like how you think...
13:37.30*** part/#asterisk LouieDog (n=louiedog@pool-71-164-37-158.chrlwv.east.verizon.net)
13:37.40defsdoorDrukenHME: not cancel cli
13:37.49defsdoorDrukenHME: just dont display missed calls on the lcd
13:38.08DrukenHMEoh, well that's a phone option....
13:38.26DrukenHMEyou can probably fix that in the cfg files
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13:41.36blitzrageSwK: oh no you di-ant!
13:41.48SwK?
13:42.09SwKi di-ant whut?
13:42.38blitzrageoh you know
13:45.53*** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net)
13:46.21elriahHi all.  Anyone using (successfully) the Cisco 7941/7961's behind NAT going to a public Asterisk box?
13:47.17blitzrageelriah: yep -- all I did was set the NAT stuff on the phone, and it worked
13:47.28elriahDid you have to make any firewall changes?
13:47.31blitzragenope
13:47.41elriahThat's on the 7941's, right?  Not the 7940?
13:47.49blitzrage7960 actually
13:47.55nasls_lsaI am little bit confused about mISDN and dial plan ... how to configure msns , and which line rings to a phone ..
13:47.55elriahYea, different firmware.
13:48.03blitzrageoh yah -- uses the java stuff right?
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13:48.12elriahThe 79x1's use different firmware than the 79x0's. (sigh) thanks anyway.
13:48.17blitzrageI have a 7970 here, but couldn't get it to register anywhere but locally
13:48.33mostynasls_lsa, msns are configured in the misdn.conf from memory
13:48.46blitzragebut I gave up on it because it was taking too much of my time trying to get it to run
13:48.47elriahOur 7940's work great.  But we bought 25 7941's and have to use them via a VPN.
13:48.51nasls_lsaand how do I get them to my dialplan ?
13:48.52blitzrageI should really just sell it...
13:49.19elriahblitzrage: Your probably like us, the phone is so cool you don't want to get rid of it.  Plus the quality is top-notch.
13:49.19blitzrageanyone want a 7970? :)
13:49.41blitzrageelriah: indeed -- and I love my 7960, except it stopped taking incoming calls for some reason...
13:50.00blitzrageall the other phones on my network are fine though
13:52.14nasls_lsa[studio]    ports=3    context=studio   msns=210515       with that will I take in my dialplan a call doing     studio,1,Dial(SIP/110)     ?
13:52.30elriahAnyone using 1.4.2 w/asterisk realtime in production?  If so, what's the results?  Any major issues?
13:53.21mostynasls_lsa, no. do you know what a context is in extensions.conf?
13:53.28nasls_lsaelriah: I am setting it up , works very good at the moment , and saw my isdn card and worked fine .. I have some problems with the configurations ( dial plan - misdn ) but that is my problem :)
13:53.35blitzrageelriah: I am -- works great
13:54.10blitzrageelriah: I load tested a 2x quad-core Xeon box and got at least 500 calls with media... with like... < 20% CPU usage
13:54.26elriahCool.  Looks like it's time to consider upgrading from 1.2.16 ...
13:54.29blitzrageusing realtime, func_odbc, lots of stuff
13:54.40blitzrageI found a lot of bugs and had them fixed in my testing
13:54.57blitzrageso you're welcome :D
13:55.13elriahHey, thanks!
13:55.14elriah;)
13:55.30blitzragemake sure you setup a separate development platform and test first
13:55.36elriahAlways...
13:55.41blitzragedon't just upgrade your production server, or you're gonna cry :)
13:55.48nasls_lsamosty: yes ..
13:56.01elriahlol, we process about 18k calls a month, I wouldn't dare...
13:56.31mostynasls_lsa, well context=foo in misdn.conf refers to a context in extensions.conf, not an extension
13:57.55nasls_lsaaah ! mISDN/studio ?!?!?!?
13:58.18GreyFoxxelriah: hehe we're paranoid too. We're hovering around 78k a month so upgrades are very strictly planned :)
13:59.17nasls_lsamosty:  I quit ...  out of ideas
13:59.31GreyFoxxthey want me to setup Openser infront of the asterisk boxes now, partly so I can redirect traffic away from a box getting updates/maintenance
14:03.00blitzrageGreyFoxx: that's what I just setup -- OpenSER is the registration point, and it can just randomly drop calls to any Asterisk box in the cluster, and its all good
14:03.13blitzragepractically have a self-healing network now :)
14:06.08*** join/#asterisk marexz (n=marexz@marexz.mil.lv)
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14:09.21GreyFoxxblitzrage: Yeah, that's pretty much what they want me to setu. Authenticating to our existing SQL database as much as possible. I just need the time to sit down and really look over the openser documentation
14:10.51blitzrageGreyFoxx: yah... its a pain in the butt :)
14:11.10blitzrageI setup views in our pgsql table to do the authentication and subscriber tables, etc...
14:11.17blitzrageso Asterisk and SER both know about the same users
14:11.38blitzragethen the GUI can configure it, change something, and both SER and Asterisk know about it at the same time, with no reload scripts or anything
14:19.50GreyFoxxThat's basically what I planned to do using views.
14:20.35GreyFoxxBasically a view for the voicemail and sip users
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14:22.57blitzrageGreyFoxx: yeppers, it works very well
14:27.08nettieuhmm blitzrage I keep getting extension XXXXXXXX in context 'bri' does not exist. Rejecting call on channel 0/1, span 2
14:27.13nettiewhere the extension is there
14:27.17nettieI quadchecked eheh
14:27.33nettieany idea what coul dbe the problem please?
14:27.49blitzragesounds like you don't have a pattern match that matches the number coming in
14:27.52[TK]D-Fendernettie: Your dialplan does NOT match.  It does not lie about this.  Go pastebint he whole mess including CLI output so we can find out what you di wrong.
14:27.54blitzrageor the bri context doesn't exist
14:28.06[TK]D-Fender~pb
14:28.10jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
14:28.15nettieit's pretty basic
14:28.21nettiewill pastebin
14:28.21blitzragestill have a bug though
14:28.27[TK]D-Fendernettie: PASTEBIN
14:28.35nettieeheh sure
14:29.13[TK]D-Fendernettie: its not that we don't trust you..... actually... yeah that really does sum it up well....
14:29.16[TK]D-FenderPASTEBIN! ;)
14:29.22nettiehttp://pastebin.ca/454283
14:29.23nettie:)
14:29.33nettieI'm sure I' missing something
14:29.44blitzrageI'm sure you are too :)
14:29.47blitzrageAsterisk is positive of it
14:31.09[TK]D-Fender[zero_local}
14:31.42[TK]D-Fenderfix your braces, and do a reload to make sure what you're showing us is actually in effect
14:32.37DrukenHME[TK]D-Fender: are you being your typical self today ?
14:33.01[TK]D-FenderDrukenHME: Think so...
14:33.10*** join/#asterisk denzs (n=denzs@carbon.gonicus.de)
14:33.25[TK]D-FenderDrukenHME: Yup.. chan_bile.so is indeed loaded ;)
14:33.43*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:34.01MuhadibDespite having configured in zapata.conf file "echocancel = yes" and "echotraining = yes", when I write in * CLI "zap show channel 4 it displays "echo cancellation currently off"!!! Anyone that passed the same problem????
14:36.01denzshi, iam trying to limit the call time and let asterisk play a warning after 10seconds with Dial(IAX/xxx/${EXTE:1}|30|ttL(20000:10000)), but after 10secs the call gets hanged up with the following message in the CLI: Apr 23 16:33:49 DEBUG[32746]: channel.c:3361 ast_generic_bridge: Nobody there, continuing...
14:36.01denzsApr 23 16:33:49 DEBUG[32746]: channel.c:3361 ast_generic_bridge: Nobody there, continuing...
14:36.32denzsis there something to mention when using the L option?
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14:37.21nettieIt's a whiargh
14:37.24nettiethanx TK !!
14:37.26nettiedoh
14:37.28nettielemme retry
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14:40.01nettieworks..
14:40.02nettiedamn
14:40.06nettieeheh
14:40.08nettietypo
14:40.41nettieevil typo :) .. I was sure it was a stupid thing.. I configured a quite complex IVR a cuple of months ago with macros and stuff
14:40.42nettieeeh
14:41.10[TK]D-Fendernettie: I'm sure it was stunningly complicated :)
14:46.36nettieTK :)))))))))))0
14:46.58nettiephear me ehehe
14:47.41MuhadibAsterisk sucks... It's echo cancellation doesn't work by far... Anyone that configured echo-cancellation properly?
14:48.11GreyFoxxWe just use hardware that does echo cancellation well
14:48.21GreyFoxxYou aren't using a X100P/clone are you ?
14:48.23*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)
14:48.41Muhadibnops...I have a digium TDM400P
14:49.27Muhadiband I'm trying to use zaptel echo canceller
14:49.50GreyFoxxWe didn't like the ones on the TE405. We found it wasn't all that great, so we have an army of Telabs 81-2572 cards
14:49.52*** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195)
14:50.07mostybah, i hate when google thinks my queries look like virus/worm queries
14:50.20GreyFoxxthe 2572's work very well, they onboard support 64ms, but with a daughter card will do 128ms
14:51.14*** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91)
14:51.16MuhadibTDM400P is an analogue card with 4 fxo/fxo ports
14:51.49MuhadibI only have echo issues with analogue interfaces
14:52.14GreyFoxxWe've never had any luck with the zaptel echo cancelling here with the zaptel stuff
14:52.48[[blah]asfdcould anyone recommend a way to sync my linux laptop to my works MS exchange calendar? I have been using thunderbird for my email, that works just fine.
14:53.34[TK]D-Fender[[blah]asfd: Evolution
14:53.47Muhadibso which software echo-canceler do you recomend using with zaptel stuff?
14:54.04[[blah]asfdi installed and am trying that, is there something more i need to do to configure that? i dont see where to set up any server information.
14:54.41[TK]D-Fender[[blah]asfd: Sorry, that answer is about as much as you should expect here.  I think you might be "lost" :)
14:54.44ChkDigitIsn't evolution-connector required to hook into exchange?
14:54.54*** join/#asterisk heison (n=heison@gw-yyz1.somanetworks.com)
14:55.50ChkDigitAnyway, I have an Asterisk question: What would prevent VoiceMail() from receiving 0 or * when the user presses them?
14:55.59GreyFoxxMuhadib: I don't recommend any, and I've never seen any t6hat worked all that well myself. ymmv
14:56.05GreyFoxxooh lunch time
14:57.27mostyChkDigit, are both directions of audio actually working?
14:57.50Muhadibthanks Greyfoxx
14:58.14ChkDigitmosty: I'd assume so.  Directory, VoiceMailMain, and other apps work.
14:58.21[[blah]asfd[TK]D-Fender: dammit... habit, i meant to post that in #linux ;-)
14:59.10[[blah]asfdjoin /#linux
14:59.12[[blah]asfderrrrrrr
14:59.18[[blah]asfdleaving now, sorry all.
14:59.20*** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91)
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15:03.38[TK]D-FenderChkDigit: you have to configure your mailbox's to accept "0", and need to have extens to support "*" and "0"
15:03.49paavumhello, I am using asterisk GUI, however when I try to log in I get a "404 Not found" error
15:04.11paavumI've tried to look in the logs but I cant see any error there
15:04.12*** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk)
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15:04.12paavum:s
15:04.37tsurkopaavum, check the topic:)
15:04.38paavumcan anybody gimme a hand?
15:05.08paavumThey are Fun Channels, not support channels :P
15:05.10paavum;)
15:05.13*** join/#asterisk karlhaines (n=karl@209.12.254.71)
15:05.56[TK]D-Fenderpaavum: GUI's are not supported here.
15:06.01[TK]D-Fenderpaavum: ANY of them.
15:06.58paavumone final q... can I get spanfax to wrk with * 1.4?
15:07.06*** join/#asterisk astawerksdotcom (n=astawerk@cpe-75-179-164-7.woh.res.rr.com)
15:07.21[TK]D-Fenderpaavum: Never heard of spanfax....
15:07.28*** join/#asterisk akitogo (n=chatzill@213.221.85.2)
15:07.34paavumits spandsp's fax thingy
15:07.46[TK]D-Fenderpaavum: get your names right :)
15:08.06[TK]D-Fenderpaavum: And not sure... I've seen some talk about real problems with 1.4
15:08.13tsurkodoes anybody have experience with softphones on thin clients?
15:08.24paavumtsurko... I'm trying to get that working
15:08.25[TK]D-Fenderpaavum: I think IAXModem + Hylafax is working properly un 1.4 though
15:08.42astawerksdotcomi got xlite to work on a wyse box before
15:08.48tsurkopaavum, what terminals are you using?
15:09.02paavumtsurko... old pentium II/III
15:09.16tsurkoi mean the software part:)
15:09.24paavumltsp
15:10.05tsurkopaavum, me too:)
15:10.09akitogoHi, anybody here who succeeded to installed a digium B410p with asteriskNow?
15:10.16tsurkoand how it's going?
15:10.20*** join/#asterisk nf1 (i=nf1@77.70.24.142)
15:10.41*** join/#asterisk astawerksdotcom (n=astawerk@cpe-75-179-164-7.woh.res.rr.com)
15:11.06paavumtsurko...  I worked on it untill last week, this week I'm trying to get asterisk now in virtual machines
15:11.18paavumtsurko... there are a couple of howtos around
15:11.23tsurkopaavum, I'm trying to do something similar
15:11.45tsurkobut I have some porblems with the softphones - the sound is a little mettalic
15:12.08tsurkoWhy you're installing * on virtual machines?
15:12.27paavumcuz I need 4 independent installations for 4 different companies
15:12.42paavumand I dont wanna use 4 servers :P
15:12.53tsurkoI see
15:13.18tsurkoit's just for testing purposes I suppose?
15:13.21paavum[TK]D-Fender --> cant you help me with the GUI thing? seems like nobody's around in the other channels
15:13.34[TK]D-Fenderpaavum: No, I can't
15:13.38paavum:(
15:14.22*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)
15:14.29putzzGUI = for Noobs that dont want to learn. LOL
15:14.49paavumputzz... GUI makes things quicker
15:15.05[TK]D-Fenderpaavum: I can mount entire sysems in the time you've asked about that ;)
15:15.35paavumand what makes it hurt is that its true
15:16.06paavumone more thing... asterisknow does not support mysql cdr?
15:16.47putzzI dont think u really understand that GUI is not supported here
15:17.01paavumI'm not speaking of guis now ...
15:17.24paavumI just wanna know if I have to compile the damn thing myself
15:17.36putzzwell asterisknow is what?
15:17.49paavuma distro
15:17.54paavumasterisgui is a gui
15:18.12paavum^^
15:18.57[TK]D-Fenderpaavum: Again, this is not the pace to ask about either.
15:19.37paavumsigh
15:21.06*** join/#asterisk naitram (n=ttech@216.77.58.40)
15:21.53naitramhow do you force asterisk to unregister a sip client
15:22.51[TK]D-Fendernaitram: typcailly if you comment out the register & reload your sip config I believe * will send the unregister
15:23.48naitramok, thkns
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15:28.54uwehello, i have asterisk and sip clients connected to it, and when i do sip show peers the status shows time ~150 ms , this is a lot i think ... but when i ping the clients i get very good response time (<2 ms), i think this can be something in the network filtering non icmp traffic or something, so i was wondering if there is a way to test that time at any given point ? an application like ping but for sip :)
15:30.34*** join/#asterisk agile (n=mike@63.98.55.146)
15:30.52JTgirigood morning, I hope somebody could help me with this...i want to use as my pstn gateway..i can call out from asterisk without any problem..but if i call from another  sip server i get following error message
15:30.56JTgiri<PROTECTED>
15:30.56JTgiri<PROTECTED>
15:30.56JTgiri<PROTECTED>
15:30.56JTgiri<PROTECTED>
15:30.56JTgiri<PROTECTED>
15:30.58JTgiri<PROTECTED>
15:30.59mostyuwe: is it a problem when you call?
15:31.00JTgiri<PROTECTED>
15:31.11putzzomg
15:31.15putzz~pb
15:31.25jbot[pb] a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
15:31.48JTgiriok sorry
15:32.54uwemosty, i have not very good sound qulity and sometimes the sound skips a little
15:35.55[TK]D-FenderJTgiri: Looks like the call is coming in just fine.  Its your dialplan, go make it do what you want.
15:36.06*** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net)
15:36.12mostyuwe: you will need to be more descriptive, what do you mean by "not very good sound quality" ?
15:38.44uwei mean the sound is slightly inturrupted
15:43.25nettie<PROTECTED>
15:44.15joe--fhey, what do you guys think of voxbone for voip forwarding?
15:44.57*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
15:48.45*** join/#asterisk Mad|Cow (n=madcow@000-202-109.area3.spcsdns.net)
15:49.32Mad|CowDoes anyone have a an examples on how to match international numbers? I'm in the US and having issues dialing the UK and other countries.....
15:49.56*** join/#asterisk `pariah (n=josh@unaffiliated/pariah)
15:50.48mostywhat do you want to match these numbers for?
15:51.17KighMad|Cow: exten => _00XXXX.,s,NoOp(${EXTEN:2:4} is the country prefix)
15:51.27Mad|Cowmosty: in my extensions.conf file... so I can dial international
15:51.33*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-9bcafbb44472e732)
15:52.05Kighyou need to dial 00XXYYYYYYYYY .. where XX is the country prefix an YYYY the parties number, without leading zero
15:52.11Mad|CowKigh: I dont need 011?
15:52.18Kighwhats 011
15:52.41MercestesPrefix for an international call in the US
15:52.42Mad|CowKigh: I thought 011 was how you dialed international
15:52.51Kighim from germany .. i.E. a german number in internation format: 00491511 where 1511 is the city
15:53.09KighMad|Cow: urm dunno what it is in your country, in europe that "00"
15:53.17Kigh*thats
15:53.32putzzus to UK is: 01144
15:53.50putzz*U.S to UK is: 01144
15:53.55Kighthen you need to match exten => _011XXX.,s,NoOp()
15:54.38*** join/#asterisk Exstatica (i=exstatic@redline.mednor.net)
15:54.39Kighdidnt know its 011 in U.S., i've never been there  sorry
15:54.55Mad|CowKigh: would I still need the ${EXTEN:2:4} in your example above?
15:55.33putzzno you wont need it
15:55.59KighMad|Cow: no you will just dial(${EXTEN}) on a normal line.
15:56.21jm|laptop~enum
15:56.31jbotsomebody said enum was http://www.voip-info.org/wiki-Enum
15:56.40mvanbaakenum rox !
15:56.47Kighack
15:56.57putzzew
15:56.59mostyMercestes, it depends what your upstream provider requires
15:57.11jm|laptopmvanbaak: am I right in thinking that I register my phone number in reverse or something and I get calls routed via IP where possible?
15:57.15Mad|Cowgot it... thanks guys... I have to run but I'll try it in a bit
15:57.17Mad|CowThanks again!
15:57.40mostyer, stupid nick-completion
15:57.45mvanbaakjm|laptop: something like that yeah. You put your phone number in reverse and you can add different records to it
15:57.51jm|laptopmvanbaak: weird.  What stops me putting our competitors numbers in the directory so that anyone trying to contact them will get patched to MY voip?!
15:58.04mvanbaaklike: preference 10: call IAX2/jm@my.voice.box
15:58.16mvanbaakpref20: call SIP/jm@my.voice.box
15:58.30mvanbaakjm|laptop: because of validation calls and stuff
15:58.35jm|laptopoh
15:58.37jm|laptopby whom?
15:58.44jm|laptoppaid administrators?
15:58.49mvanbaakthe enum registry
15:58.50mvanbaakyeah
15:58.56jm|laptopstill sounds a little dodgy
15:59.00jm|laptopI might try it; all the same
15:59.03mvanbaakevery country has a delegation
15:59.04jm|laptop~dundi
15:59.06jbotextra, extra, read all about it, dundi is http://www.dundi.com
15:59.21mvanbaakthe delegation is based on country code
15:59.22jm|laptopoh dundi is something different
15:59.22blitzragedundi is pretty sweet
15:59.32mvanbaakdundy is very sweet :)
15:59.40jm|laptopleast cost routing, right?
15:59.51blitzrageyou can use it for that, ya
15:59.52mvanbaakjm|laptop: here in .nl you have to give your landline nr for enum
15:59.55jm|laptopoften free when the local calls are for that local code
16:00.00blitzragedundi is how you pass information between two Asterisk boxes
16:00.04mvanbaakthat number gets a call and reads a pin
16:00.10mvanbaakyou have to provide that pin
16:00.23mvanbaakthat will give you access to the enum records of your phone nr
16:00.29mvanbaakthat way you can attach info to it
16:00.37jm|laptophmm
16:00.40jm|laptopI'll read up later
16:00.48jm|laptopyou understand my concern, though?
16:00.52jm|laptopre: lying
16:00.55mvanbaakif the enum nr is already taken (think, I try to get my neighbours nr) they gonna verify it manually
16:00.58mvanbaakuhhuh
16:01.02mvanbaakI do
16:01.09mvanbaakbut the risk is the same as with domain names
16:01.21jm|laptopI suppose
16:01.26mvanbaakbrb, going to visit neighbour
16:01.28mvanbaaklatero
16:01.32paavumtsurko ...
16:01.34paavumu there?
16:01.56*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
16:03.12*** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-9e33bd3ecc6c6953)
16:04.10sivanadoes anyone know of a way to check to see if a channel is in use (ie: SIP/chan1234)
16:04.20sivanausing 1.4
16:04.43sivanaIsChanAvail is broke it seems
16:04.47*** part/#asterisk deoptima (n=deoptima@c-71-228-222-87.hsd1.tn.comcast.net)
16:04.51*** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net)
16:05.14[TK]D-Fendersivana: Works for me... show how you're using it.
16:06.12sivanahttp://www.pastebin.ca/454394
16:08.01brad_msswi just upgraded from asterisk 1.2 to 1.4.2 ... I take it using jbforce=yes in sip.conf is not recommended? on incoming calls, it appears to cut the channel off in about 3 seconds with a critical jb error ...  though I can get some calls through, randomly it seems.  Also, setting jbimpl=adaptive appears to not work at all .. i get no sound from either end
16:08.59brad_msswjust wondering if having jb enabled is worth it at all
16:09.59brad_mssw... that said, I should have kept the log from when this occurred, i had to disable it real fast because it was on my office's production phone system, so i can't give the exact errors until after business hours
16:11.11mostybrad_mssw, 1.4 isn't really ready for production use yet, in my experience
16:12.26brad_msswhmm, nice
16:12.51brad_msswspent a couple hours trying to port my extensions.ael over ... thought that'd be all there was to it
16:14.24mostyif you stick with 1.4 now, expect a few things to be broken :/
16:15.26*** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl)
16:15.58brad_msswi was mainly upgrading to get access to the jitterbuffer as we have some quality issues from time to time
16:17.25*** join/#asterisk StealthHe (n=StealthH@151.204.60.4)
16:17.47*** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com)
16:17.51sivanalots of core dumps in 1.4
16:18.43rg1_question: does anyone know if there is a way when asterisk receives a call, if the caller-id that is presented is "spoofed" from someone?
16:18.57brad_msswi'm not running a high-load system at all, max is usually 3-4 active calls ... hopefully i won't have too many issues
16:20.46brad_msswactually, found something on bugs.digium.com that looks to be the exact error message I was getting: [Apr 12 09:26:32] WARNING[26247]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission a11ce-10262-461e3382-1cff-216.109.205.115 for seqno 7771 (Critical Response)
16:21.02brad_msswwith jbforce=yes ...
16:21.13brad_msswit would hang up after that
16:21.37mostywell that warning message appears in other situations too
16:22.44StealthHeIs anyone aware of the existence of any helpful documentation for incorporating Asterisk voicemail into an existing Cisco CM architecture?
16:22.50rg1_anyone know anything about caller-id's out there?  Specificially, about the caller-id of incoming calls
16:23.23mostyrg1_, you can set the callerid that local extensions see
16:23.43brad_msswrg1_: as far as I know, there's no way to tell if the callerid was spoofed for not ...
16:23.51rg1_incoming calls that hit the PRI trunk line
16:24.14mostyrg1_, what are you trying to do exactly?
16:24.19rg1_brad_mssw-thats what i thought too, but when i spoofed an out-going call to my bank to have them think i was calling from my house, they could tell
16:24.45rg1_yet, when i call a "normal" phone, it looks like the # I am spoofing
16:25.09rg1_so I'm wondering if there is another variable that the switch is getting (NI2) that might be available on asterisk to check
16:25.22brad_msswrg1_: was the number you were spoofing a legitimate number?
16:25.33mostyrg1_, no this is most likely done by your service provider
16:25.48rg1_yes
16:25.52rg1_legitimate
16:26.07rg1_mosty - don't understand your comment
16:26.15[TK]D-Fendersivana: You are calling it wrong.
16:26.36rg1_XO is my service provider and I can make calls with the callier-id/name I set in asterisk
16:26.56mostyrg1_, your service provider can pass on the callerid you provide them, but it's their option wether to trust it or not
16:27.05rg1_same provider sends both calls (XO) - 1) to normal phone customer looks like my home#
16:27.17rg1_mosty, i understand that, but they could tell it was NOT my home phone
16:27.38rg1_so they must be able to see the actual phone# (at least that's what i'm thinking)
16:27.51mostyplus, big businesses and government agencies get more callerid data than regular telephone service
16:27.53rg1_maybe a more sophisticated card/switch gets both numbers?
16:28.00brad_msswrg1_: guarantee your provider doesn't allow you to set the name ... at least not in the US
16:28.03rg1_yeah, thats what I was thinking
16:28.11brad_msswrg1_: it may populate the name calling to another phone on their own network, but not across networks
16:28.15rg1_brad_mssw - true
16:28.40rg1_I think mosty's latest comment must be correct
16:28.58*** join/#asterisk FreezeS (n=bla@82.208.157.125)
16:29.11rg1_I'm wondering what kind of switch/card/service can tell that, because for another application I'm doing I really need to KNOW that the caller-id is 100% from that phone#
16:29.53mostyrg1_, talk to your service provider. you can probably only get what you want by forking out a lot more money for your service
16:30.03rg1_right
16:30.23mostyas an alternative, you can setup a callback service
16:30.39rg1_right, thats what we're doing - but outgoing costs us call charges
16:30.45rg1_so we're trying to avoid that if possible
16:30.54rg1_and we don't want to have to ask for passwords
16:30.57mostydepends how secure you want to be
16:31.00rg1_right
16:31.42StealthHeIs there a hardware compatibility list for Asterisk? More accurately, is it compatible with a Cisco 1760 using VIC2-4FXO cards?
16:32.14Qwell[]StealthHe: wouldn't the cisco just be speaking SIP?
16:32.28StealthHeI believe so.
16:32.45Qwell[]then there isn't much for the hardware to be compatible with...
16:33.00Qwell[]asterisk doesn't know/care what hardware the cisco has
16:33.11aydiosmioI have all sorts of junk talking SIp to our AS5300 (IOS 12.2)
16:33.24aydiosmiono big deal
16:33.26StealthHeMy apologies, I've pretty much been thrown into this by my company. I am just starting to research all this information, so if I seem a little less knowledgable, it's because I am at the moment.
16:33.35Qwell[]!book
16:33.36Qwell[]erm
16:33.38Qwell[]~book
16:33.41jbotit has been said that book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
16:33.41Qwell[]~wikis
16:33.42jbotwell, wikis is http://www.voip-info.org
16:33.46aydiosmio~consultant
16:33.48jbotHire a consultant.
16:33.49Qwell[]StealthHe: start there
16:33.52aydiosmiohahaha
16:33.54Qwell[]...or there
16:33.57aydiosmionice one jbot
16:34.36*** join/#asterisk jeffik (n=Valued@h-64-105-236-252.chcgilgm.covad.net)
16:34.59StealthHeI believe I have that in pdf format. Hopefully, I'll be given the time to actually get through it.
16:35.16Qwell[]option 3: quit
16:35.20StealthHeIndeed!
16:35.45aydiosmiogood way to learn though
16:35.56aydiosmiothight deadline on something you know nothing about
16:36.09aydiosmioit may not be right, but damned it it doesn't work
16:36.40*** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com)
16:37.00aydiosmiojust don't bother us with things that can be answered with a little bit of research
16:38.23aydiosmioand by "us" I mean these fine men and women with large repositories of asterisk knowlege
16:38.36Sweeperholy fuck is openser a nasty nasty thing to configure
16:40.44StealthHeThe deadline is really the only reason I bothered. I'll refrain from asking further questions until I've had the chance to read through the book.
16:41.09*** join/#asterisk jovannotti (n=saravia@190.144.48.1)
16:41.35*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
16:42.33joshaidanI have a simple question... What does PVT stand for?  Such as in SIP_PVT.
16:44.39*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
16:47.10*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
16:47.37puzzledhi
16:48.47jm|laptopdoes enum work in 1.2 ?
16:49.00*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
16:49.51jm|laptopand anyway - surely p2p > enum ?
16:50.18*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
16:52.24jovannottihelpppppp, I need to use the TC400B card with G723. I wrote to support@digium.com and call to the phone numbers, but nobody answer to me !!
16:54.25aydiosmiodamnit
16:54.29*** join/#asterisk riksta (n=rick@rhamnett.plus.com)
16:54.33aydiosmioI told you to stop whining yesterday
16:54.39aydiosmiosame applies today
16:54.57rikstaHi can someone tell me how i can find out about sending SMS over my E1 channel? I can't seem to find anything useful via google
16:55.16ManxPowerjovannotti: when did you contact them?
16:55.42jovannottiI wrote an e-mail this morning. 5 hours ago aprox
16:55.50ManxPowerriksta: did you try voip-info.org or the mailing list archives?
16:56.01rikstaManxPower, yeah, went there first!
16:56.11ManxPowerjovannotti: Digium's offices opened 3 hours ago
16:56.25ManxPowerSupport usually takes 24 - 48 hours to respond.
16:56.34ManxPowerIf you want faster support buy a support contract.
16:57.14rikstaManxPower, i can only really find people wanting to sell me a sms gateway :) I want to find out how to do it myself
16:57.22riksta(if possible)
16:59.13*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
17:01.45aydiosmioriksta: what do you mean sms gateway?
17:01.48aydiosmio(you can build one on linux)
17:02.13aydiosmioheck I built one on windows
17:02.15rikstaaydiosmio, i want to know how to send sms data down my E1
17:02.26rikstai don't want to use an sms gateway from someone
17:02.30aydiosmioall you need is a cell phone with an AT interface and some software to send/recieve
17:02.45rikstano, i dont want this
17:02.55aydiosmio...E1?
17:02.55*** join/#asterisk qdk (n=qdk@193.164.155.27)
17:02.55rikstaread what I am saying ;)
17:03.12rikstaaydiosmio, if you don't know what an E1 is then I'm pretty sure you can't help me.
17:03.13aydiosmioI don't think you understand how SMS works
17:03.31aydiosmioI know what an E1 is, I 'm wondering why the heck you'd want to send SMS with it
17:03.38paavumriksta,... I thnk you need some kind of interface with the cellular network
17:03.46paavumyou cant just send sms over e1s
17:03.53ManxPowerpaavum: that would be called the SMSC
17:03.57aydiosmioYou can only send SMS over the internet in the form of a TCP conneciton to a SMS gateway provider
17:04.03ManxPowerpaavum: are you not in Europe?
17:04.18ManxPoweraydiosmio: you are not in Europe either, are you?
17:04.23rikstahmmm, ok conflicting info :) please can someone clarify
17:04.37rikstaI just heard it's possible to push your own SMS over an E
17:04.38riksta1
17:04.45aydiosmiono I'm not
17:05.04ManxPowerriksta: In the USA you can only send SMS over the internet.  In Europe you should be able to send an SMS from Asterisk to the SMSC of the carrier of the destination.
17:05.04aydiosmioI'm not from the land where the PSTN allows SMS sending to home phones
17:05.12ManxPowerThe SMSC will have a telephone number.
17:05.29rikstaok, and I imagine you need to send it to the correct network's SMSC?
17:05.39ManxPoweraydiosmio: SMS in the USA and SMS in the rest of the world work totally differently
17:05.51rikstaManxPower, also, i'm looking for some documentation, on actually HOW to send this using asterisk
17:06.12ManxPowerriksta: I don't believe they exist
17:06.12ManxPowerthere has been discusstions on the mailinglists, however.
17:06.13rikstaok , question resolved :)
17:06.20rikstayeah, i saw that one thread
17:06.40aydiosmiohttp://lists.digium.com/pipermail/asterisk-dev/2003-December/002425.html
17:08.23rikstayeah i read that one
17:08.36rikstaconfused me ;)
17:09.01Qwell[]ManxPower: discusstions?
17:09.12Qwell[]ManxPower: surely you didn't think we'd let that slide? :)
17:09.16rikstahehe
17:09.48ManxPowerUs people in Alabama ain't good speelers
17:09.55Qwell[]peepol?
17:10.50jm|laptopwhat is the point of free164 numbers?
17:10.55[TK]D-Fenderdisgustion : (n) an unlgy conversation you'd rather not be having.
17:10.59[TK]D-Fenderugly*
17:11.05*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
17:13.31*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com)
17:15.20mvanbaakback
17:17.52*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
17:18.14*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
17:20.25sivana[TK]D-Fender: I changed it.. added sj as the options
17:20.38sivananot sure why the syntax was different from 1.2 when it worked.. but we'll see
17:21.10[TK]D-Fendersivana: Priority jumping is even MORE of a dead item in 1.4
17:21.24[TK]D-Fendersivana: You should not be using that means where at all avoidable.
17:21.59sivanathat's the only call where I use it
17:22.22sivanabut I'll take some time later and incorporate the status variable
17:23.11[TK]D-Fendersivana: Yeah, for a down & dirty 1-shot dea, what the heck..
17:23.14[TK]D-Fenderdeal*
17:23.54jm|laptophmm
17:24.07jm|laptopI should be able to receive calls in the format   IAX2/foo@bar.com   right?
17:24.19jm|laptoprather than   IAX2/guest@bar.com/foo  ?
17:25.39ManxPowerjm|laptop:  IAX2/foo@bar.com will send the call to the "s" extension on the destination server
17:26.00jm|laptopthat's what I thought
17:28.07russellbooh, extensions.com
17:28.07jm|laptopah
17:28.08jm|laptop.conf lol
17:28.08ManxPowerthe context of "s" is whatever is the context= line from the [foo] section of iax.conf
17:28.08ManxPowerThe reason it goes to the "s" extension, of course, is because there is no destination extension on the Dial line.
17:28.09jm|laptopsure
17:28.09ManxPowerIt is a little known fact that "s" is in fact French for "no destination extension"
17:28.09jm|laptopbut I can grab the foo@  ?
17:28.19ManxPower"grab"?
17:28.30jm|laptopApr 23 18:28:14 NOTICE[9373]: chan_iax2.c:6925 socket_read: Rejected connect attempt from 204.55.81.11, who was trying to reach 's@'
17:28.43jm|laptopoh wait
17:28.49ManxPowers@ means "s" at "no context found"
17:29.13jm|laptopwhere did aix.conf go?!  :O
17:29.25Qwell[]jm|laptop: it's still on your IBM
17:29.26ManxPowerthere never was an aix.conf
17:29.32jm|laptopmail:/etc/asterisk# find . | grep aix.conf
17:29.33jm|laptopmail:/etc/asterisk#
17:29.34jm|laptop:(
17:29.55ManxPowerand there never will be an aix.conf in Asterisk.
17:29.59Qwell[]there might
17:29.59jm|laptopok I should have used -name
17:30.11ManxPowerperhaps you are thinking of iax.conf and for some reason lost half your brain?
17:30.20Qwell[]ManxPower: if somebody ports it to AIX, I imagine they'll have AIX specific customizations
17:30.27jm|laptopManxPower: it's been a long day :(
17:30.29ManxPowerLook under the sofa.  That's where I always find mine.
17:30.38ManxPowerjm|laptop: then go to bed and stop wasting our time.
17:30.46jm|laptop:O
17:31.24ManxPowerI'm not joking.  If you are so tired as to make such simple mistakes you will not actually accomplish anything no matter how much you try.
17:31.52jm|laptopmy default context goes in [general] ?
17:36.34*** join/#asterisk jmacz (n=jmacz@201.244.170.241)
17:37.37jm|laptop" ... No application 'Dail' for extension ..."
17:37.47jm|laptopManxPower: I'm beginning to agree with you
17:37.48jm|laptop:|
17:38.44ManxPowerI have been wrong before, but that was years ago.
17:39.36[TK]D-Fender(as measured in may-fly years, not human)
17:39.38[TK]D-Fender;)
17:40.54[TK]D-Fender~book
17:40.56jbotrumour has it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:41.40*** join/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com)
17:43.38anonymouz666vars inside ${CURL must be passed as ${var} or just as var
17:43.48jm|laptopso
17:43.55jm|laptopI appear to have e164.org sort of set up
17:44.33*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
17:44.57jm|laptopalthough I'm still not entirely convinced
17:46.17mvanbaaklol
17:46.20mvanbaakget some sleep
17:46.22mvanbaakand try again
17:46.39*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:53.39*** join/#asterisk candyman50 (n=mdecandi@pool69-59-255-25.kewr1.s.vonagenetworks.net)
17:54.28mvanbaakworks for me
17:55.18candyman50Question: Does anyone know how to mainupulate the Sip TO: Header in *?
17:55.44candyman50I know I can manipulate the FROM with CALLERID(name/num)
17:56.10*** join/#asterisk Dandan (i=dandan@wsip-70-167-100-158.ri.ri.cox.net)
17:56.15Dandanhey all
17:56.17Dandan:)
17:56.21Dandanjust jumped to 1.4
17:56.28Dandanis there any reason why Echo app would be REAAAALY jerky?
17:56.49Dandanas in: the sound is probably 10 seconds long and it takes the app over 1 minute to play it?
18:04.37*** join/#asterisk bkw_ (i=brian@adsl-70-142-43-193.dsl.tul2ok.sbcglobal.net)
18:04.38anonymouz666anyone know how can I debug the function ${CURL()}
18:04.48anonymouz666it's not sending anything to the server
18:06.59*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
18:11.51*** join/#asterisk hrmphh (i=patrick@notchill.com)
18:11.54hrmphh[Apr 23 10:04:28] NOTICE[9062] chan_zap.c: Avoiding deadlock...
18:11.57hrmphhover and over in the logs
18:12.02hrmphhhow can i fix that?
18:12.30*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
18:12.36anonymouz666just a quick question
18:12.46anonymouz666does ${curl()} support https?
18:16.05*** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net)
18:16.19*** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net)
18:16.22ManxPowerhrmphh: stop using chan_spy or Monitor
18:16.35elriahHi all.  Does anyone know if the T.38 patch is in asterisk 1.2.16?
18:16.55ManxPowerelriah: since it is a NEW FEATURE the patch would not be in 1.2.x
18:16.56elriahWould it be in show codecs if it were?
18:17.22elriahAhh.  Didn't know when it was introduced.  Found a mantis thread that was for 1.2...
18:17.35elriahsays committed into trunk 33890.
18:17.39AndrewGearhartanybody here familiar with the Polycom 320?
18:18.01elriahAndrewGearhart: what's the question?
18:18.03hrmphhmanx; i wasnt aware that i was?
18:18.06hrmphhmanx; how to disable?
18:18.21*** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net)
18:18.22ManxPowerelriah: trunk is whatever the next version will be at the time of the comit
18:18.30ManxPowerso if 1.2 was out at the time, then it would be in 1.4
18:18.34elriahGot ya.
18:18.37hmm-homei'm having some string problems  SayDigits(${foo_${bar}}) does not work. ${bar} = 0 and the value for ${foo_0} is 12345
18:18.41elriahIs there a backport that you know of?
18:18.42hrmphhalso i was told that the person making the outbound call heard nothing, but when he later spoke to the person on the receiving end, they said they heard hold music
18:18.47AndrewGearhartelriah: it says that it is a two-line phone... what exactly does that mean? that it can only handle two lines period? or that it only has buttons for two lines...
18:19.00ManxPowerhrmphh: you don't disable.  You would have to be running chanspy or monitor to record or listen to your calls
18:19.03elriahAndrewGearhart: No, that it can display two line labels.
18:19.37ManxPowerAndrewGearhart: A "line" on a polycom can handle more than 1 call, but we always disable that feature because it confuses users
18:19.39elriahAndrewGearhart: Most people just provision a single line lable, i.e., extension 805, but that doesn't limit the number of inbound/outbound calls.  I believe that number is 64.
18:19.49ManxPowerif you want more than 2 "lines/calls" get a different model
18:20.05hrmphhmanx; youre saying i would have had to interactively run it?
18:20.30hmm-homeyeah nevermind i'm retarded
18:20.30trevarthanHello. I have an IVR running on an asterisk box connected to a linksys spa3102. After a day or so of up-time, the IVR starts exhibiting an annoying "ringing" noise, sort of like a feedback loop, immediately after DTMF presses. As far as I can tell, this problem is only manifest on the spa3102 <-> SIP/asterisk link. Other SIP phones connecting to the asterisk box sound fine. Any ideas?
18:20.41ManxPowerhrmphh: NO!  You would have to put it in extensions.conf
18:21.18AndrewGearhartelriah & ManxPower : thanks
18:21.34*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
18:21.44ManxPowerI'm starting to think that nobody actually manages to convert one video format to another.  All docs about this using any tool do not work or are just plain wrong.
18:22.00ManxPoweror are Mac specific
18:22.03Qwell[]ManxPower: that sounds like it's just asking for trouble, really
18:22.12elriahAhh, the trials and tribulations of bridging fax calls on Asterisk 1.2...
18:22.14Qwell[]though, I have thought about using transcode/mencoder to do it
18:22.21elriahAnyone have some chicken bones?
18:22.30Qwell[]but, I don't *ever* violate patents...
18:22.35trevarthanManxPower: yeah, transcode works. I use it with MythTV all the time.
18:22.40syzygyBSD:) Qwell
18:22.56fileQwell[]: you are violating a Telcomjoshvoxmart patent by saying you aren't!
18:23.00syzygyBSDI never knowingly violate patents...
18:23.04Qwell[]file: I have a license, nub
18:23.10ManxPowerQwell: I'm trying to remux .ty (tivo) into a format Quicktime Pro can read (.vob, .mpg) to convert it to mpeg4
18:23.15hrmphhManx; theres nothing ine xtensions relating to chanspy or monitor
18:23.20Qwell[]ManxPower: oh, not with asterisk?  heh
18:23.21hrmphhsomething else must be causing the deadlock
18:23.21ManxPowertrevarthan: every ffmpeg based transcode looks terrible to me
18:23.26Qwell[]transcode works great
18:23.29elriahAt this point, just by getting up in the morning, I'm pretty sure everyone violates some technology patent.
18:23.34*** join/#asterisk oej (n=olle@apollo.webway.se)
18:23.35ManxPowerhrmphh: then you must be having some other problem
18:23.51AndrewGearhartour current phone system shows 12 buttons for the extensions and 6 buttons for outside lines... and I'm trying to figure out what the necessary minimum price would be to carry that kind of feature set over to a new phone system
18:24.00ManxPowerQwell: does transcode support 1.5Mbps - 8.0Mbps bitrates?
18:24.08AndrewGearhartthat's the root of my questions. ;-)
18:24.15Qwell[]ManxPower: it supports whatever you tell it...
18:24.21syzygyBSDAndrewGearhart: what is your question?
18:24.31ManxPowerQwell: ffmpeg is supposed to but, but that never works
18:24.59elriahAndrewGearhart: If you want a good speakerphone, consider at least a Polycom 501.
18:25.08syzygyBSDManxPower: do you have the codec for .ty installed?
18:25.15AndrewGearhartsyzygyBSD: was just explaining more about what I had previously asked in case anybody was interested.
18:25.24syzygyBSDoh, I just joined
18:25.30ManxPowerQwell: well it "works" with lots of buffer underruns and if I tell it to use a MASSIVE buffer then the underruns go away but the file cannot be played in quicktime for windoews
18:25.30syzygyBSDor rejoined...
18:25.40ManxPowersyzygyBSD: yes.
18:25.41Qwell[]ManxPower: That's a feature
18:25.46AndrewGearhartelriah: that would be one of the things that would be part of it... at least one of the phones needs to have a nice speakerphone on it
18:25.46ManxPowerit is called vsplit-linux
18:25.53syzygyBSDya, why would you want to use windows?
18:26.02syzygyBSDj/k
18:26.04Qwell[]ManxPower: You just have to give it a good codec..  something windows supports
18:26.14Qwell[](which, by default, is very little)
18:26.18anonymouz666I think app curl does not support HTTPS :(
18:26.20syzygyBSDwill vlc on windows play it?
18:26.23anonymouz666Corydon-w ?
18:26.27anonymouz666am I right ?
18:26.28ManxPowersyzygyBSD: because I believe that a commercial mpeg4 encoder is better than a free one.
18:26.42Qwell[]ManxPower: There's your first mistake ;)
18:26.49Corydon-wanonymouz666: it depends
18:26.50ManxPowerQwell: Uh, by the time the file gets to windows is it a "standard" mpeg2 file.
18:27.06Qwell[]ManxPower: mpeg2 is just a wrapper, isn't it?
18:27.14syzygyBSDlol...
18:27.17ManxPowerI can PLAY the files all day long in VLC or WMP
18:27.27ManxPowerQwell: both a wrapper and a codec
18:27.28anonymouz666Corydon-w: on what?
18:27.30Qwell[]oh
18:27.33syzygyBSDwell that sounds like a quicktime problem
18:27.46Corydon-wanonymouz666: depends on the installation of libcurl
18:28.03ManxPowersyzygyBSD: My point still stands.  I cannot get from point A to point B
18:28.11syzygyBSDif you can use vlc, why would you want to use quicktime?
18:28.19syzygyBSDoh.. you can... just no reason to...
18:28.23ManxPowersyzygyBSD: because I do not want to make 500 people install VLC
18:28.28syzygyBSDhence, the lack of tutorials
18:28.43syzygyBSDoh, so this isn't for personal....
18:28.53anonymouz666Corydon76-home: thanks again!
18:29.00syzygyBSDwait.. you want to make 500 people install itunes, but not vlc?
18:29.03ManxPowerAnd since WMP will only play streams that are in the MS version of mpeg4 in an .ASF container if I do that then I lock out all the mac users
18:29.08anonymouz666I will rebuild the lib
18:29.26DandanHey Guys? Anyone might know why Echo app (in Ast 1.4.2) is slow ancd choppy?
18:29.48DandanI have ztdummy, uhci, everything done properly...
18:29.49ManxPowersyzygyBSD: I figure quicktime will be more useful than vlc for the majority of "uh, where is the any key" users.
18:30.18ManxPowerand honestly VLC's seeking features suck
18:30.21syzygyBSDI stopped supporting those users a long time ago
18:30.29syzygyBSDmy answer is go ask your 5 year old
18:30.44ManxPowersyzygyBSD: no one under 21 is allowed on the property
18:31.01syzygyBSDoh.. so it is a business... changes even more...
18:31.10syzygyBSDyou work at a brewery?
18:31.11syzygyBSDmmmmm
18:31.18syzygyBSDcan i get a job?
18:31.30ManxPowersyzygyBSD: gay campground.
18:31.41ManxPowerThe interview process is somewhat non-tradisional
18:31.43Dandanlol, even better prolly :>
18:31.45*** join/#asterisk Strom_M (n=strom@135.196.213.180)
18:32.05elriahManxPower: Eh? What's a gay campground?
18:32.15elriahManxPower: Only gay couples allowed?
18:32.16illsciword of life
18:32.35ManxPowerelriah:  A campground where everyone is happy.
18:32.46syzygyBSDvery happy...
18:33.02trevarthanI have an IVR running on an asterisk box connected to a linksys spa3102. After a day or so of up-time, the IVR starts exhibiting an annoying "ringing" noise, sort of like a feedback loop, immediately after DTMF presses. As far as I can tell, this problem is only manifest on the spa3102 <-> SIP/asterisk link. Other SIP phones connecting to the asterisk box sound fine. Any ideas?
18:33.08elriahWho would have thunk it...  A quick google search yieled: http://www.campgayusa.com
18:33.09syzygyBSDexcept for the IT support...
18:33.15elriahIt's official, there's a website for everything.
18:34.15syzygyBSDtrevarthan: bad zap connection or echo canceler issues
18:34.43trevarthansyzygyBSD: not a zap connection. It's a terminal adapter. a linksys 3102.
18:35.03trevarthanecho canceler on the asterisk end? Or the spa3102 end?
18:35.04anonymouz666Corydon-w: I need to use something like that in app_curl.c: curl_easy_setopt(curl, CURLOPT_SSL_VERIFYPEER, FALSE);
18:35.10ManxPowertrevarthan: change the packet side on the SPA from .3 to .2
18:35.26anonymouz666Corydon-w: you authorize me to do that in app_curl.c ?
18:35.27ManxPowerI don't recall what the exact option is called.
18:35.34ManxPowermaybe ms per packet or something like that
18:35.47trevarthanManxPower: ok. I'll give that a shot. Thanks.
18:35.58hrmphhwhat ist he setting that determines when it will start dialing?
18:36.05hrmphhi.e. no more DTMF
18:36.05hrmphhtones
18:36.07syzygyBSDtrevarthan: either, maybe both (the way they cancel out the echo the other already canceled....)  the analog connection to your phone
18:36.49elriahhrmphh: Dial Plan, usually on the phone.
18:36.49syzygyBSDhrmphh: what do you mean
18:37.00hrmphhits an analog phone
18:37.02hrmphhconnected to fxs port
18:37.26syzygyBSDdigittimeout?
18:37.32elriahThe # key will do it.  Dunno how you would control a dial plan for a phone, digittimeout maybe.
18:38.21*** join/#asterisk wundaboy (n=look@adsl-68-122-118-112.dsl.pltn13.pacbell.net)
18:38.27ManxPowerhrmphh: your dialplan in extensions.conf if you are on an FXS port.  the dialplan on the ATA if you are using an ATA
18:38.35ManxPowerdigittimeout only works for IVR, not dialing
18:38.51ManxPowerand # will NOT do it if you are on an fxs card
18:39.04naitrami have a linksys ATA, anyone know what to set so that i get a notification of an off hook when the phone is lifted off the hook?
18:39.11*** join/#asterisk Meaty` (n=meaty3@office.abi.ca)
18:39.11elriahWell, then I stand corrected.
18:39.32hrmphhhmm ok
18:39.34hrmphhso im sol then
18:39.36hrmphhno way to change
18:39.38wundaboyI am using a hosted pbx that requires an 11digit dialstring, is there anyway i can setup some kind of proxy to modify the dialstring?
18:39.40hrmphhseems like it takes a while to start ringing
18:39.42hrmphhafter it dials
18:40.49ManxPowerhrmphh: that will only happen if you have a badly designed dialplan
18:41.18ManxPoweri.e. you are dialing 5551212 but you have another exten line that can match 2125551212
18:41.28ManxPowerhow does asterisk know that 5551212 is the full number?
18:42.27*** join/#asterisk Strom_C (n=strom@135.196.213.180)
18:43.24ManxPowerQwell: According to a FAQ from apple.com Quicktime Pro with the MPEG2 support can transcode MPEG2 video to other formats, but not MPEG2 audio.
18:43.43Qwell[]ManxPower: lame
18:44.01*** join/#asterisk DoDaT69 (n=dodat69@internal.digitalson.com)
18:44.05ManxPowerhrmphh: my users never have to wait for more than 1/10th of a second for the call to be processed after they dial
18:44.09ManxPowerQwell: I agree.
18:44.21Qwell[]ManxPower: can it do different audio in an mpeg2 wrapper?
18:45.54ManxPowerQwell: Should be able to, working on that now
18:46.10*** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net)
18:47.22hrmphhmanx; i dont know if the call is waiting to be processed
18:47.25hrmphhor just processing takes a while
18:47.27hrmphhhttp://pastebin.ca/454623
18:47.30hrmphhtheres the dial plan
18:48.23tzafrir_laptopflash works nicely on an FXS card
18:48.41hrmphhflash?
18:48.44hrmphhto transfer?
18:48.46hrmphhor conf?
18:50.42tzafrir_laptoptransfer
18:51.35anonymouz666Corydon-w! I broke the app :) niiice
18:51.40hrmphhok heres a really strange problem description: user makes outbound call (e.g. dials 91800WHATEVER), hears nothing but silence on the phone after dialing, hangs up after a while. later gets in touch with person he thought he was calling and person said their phone rang, they answered, and they heard the music on hold!
18:52.23ManxPowerhrmphh: put the contents of extensions.conf not the output of "show dialplan"
18:52.32hrmphhits extensions.ael
18:52.35hrmphhthats what i put
18:52.41ManxPowerhrmphh: can't help you with that
18:52.50*** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org)
18:52.50hrmphhbasically the same thing
18:52.53hrmphhjust diff syntax
18:53.17AmbroseAnyone know how to change the limit of the Dial command? I have a really long dial command with a bunch of w's and it seems like it's getting cut off and not sending the whole ting
18:53.59AmbroseIn the cli it show "Executing Dial <....>" but the next line says "Called <....>" and that's where it's cut off
18:54.10ManxPowerhrmphh: correct.  I don't really feel like learning an entire new syntax for extensions today.
18:54.18ManxPowerhrmphh: what FXS card are you using?
18:54.29hrmphhTDM400P
18:54.34hrmphhits the 1 fxs, 3 fxo
18:54.48joe--fhey, what do you guys think of voxbone for voip forwarding?
18:57.10ManxPowerAmbrose: you would have to modify the asterisk source for get more than: channel.h:#define AST_MAX_EXTENSION    80
18:57.28AmbroseManxPower : Thanks.
18:58.08trevarthanManxPower: so.... change 'RTP Packet Size:' under the SIP tab from .03 to 0.2?
18:59.07trevarthanany idea why that is necessary?
19:00.30ManxPowertrevarthan: because The SPAs default to 30ms packets and Asterisk is hardcoded and designed for 20ms packets.  So if you leave the defaults  the last 10ms of each packet is dropped.
19:02.04trevarthanoh. ok. yeah, that's good to know, eh?
19:02.42naitramanyone know how to enable hook flash on linksys voip gateway
19:02.45ManxPowertrevarthan: of it is .03 in the SPA then change it to .02  if it is .3 in the ATA then change it to .2
19:03.33*** join/#asterisk Goodjoke (n=Goodjoke@74.202.86.23)
19:04.12AmbroseManxPower : When I uncomment that and changed it, now I get an error when I try to run make. Any ideas?
19:05.50*** join/#asterisk sysreq (n=sysreq@193.245-ppp.3menatwork.com)
19:07.50Hmmhesaysis a variable set anywhere that tells you what context your gosub came from?
19:10.08ManxPowerHmmhesays: See README.variables
19:10.28ManxPowerAmbrose: # is not a comment
19:10.33ManxPowerput the # back in
19:11.03*** join/#asterisk JerJer[mobile] (n=jj@199.45.11.90)
19:11.17AmbroseManxPower : Oh, ok
19:16.48*** part/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net)
19:18.23*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
19:22.22AndrewGearhartis there a feature in asterisk that, if the extension that you're calling is busy, you can set it to ring both of you when they become free?
19:26.11*** part/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org)
19:29.59*** join/#asterisk nasls_lsa (n=chatzill@athedsl-136017.home.otenet.gr)
19:31.15JerJer[mobile]AndrewGearhart:  In the telco world this is called CampON
19:31.22JerJer[mobile]and i have not seen that feature
19:36.14AndrewGearhartJerJer[mobile]: thx.
19:36.31AndrewGearharthow does one become a contributor to asterisk?
19:38.21Dandan~choppy
19:38.23Dandan~slow
19:38.25jbotno I'm not
19:39.17DandanI will answer myself: you you EVER get a SLOW and CHOPPY sound, make sure, you add "noapic" to the kernel boot.
19:39.33Dandanjbot: choppy:you EVER get a SLOW and CHOPPY sound, make sure, you add "noapic" to the kernel boot.
19:39.38Dandan~jbot
19:40.12Dandanhm, how do you train him?
19:41.16redaxhi,
19:41.49Corydon-wjbot: choppy is <reply> If you ever get a slow and choppy sound, make sure, you add "noapic" to the kernel boot parameters.
19:41.51jbotCorydon-w: okay
19:41.56redaxis the bristuffed zaptel give the DID number ?
19:42.03Corydon-w~choppy
19:42.05jbotIf you ever get a slow and choppy sound, make sure, you add "noapic" to the kernel boot parameters.
19:42.07redaxseems like I miss the DID now
19:42.22Corydon-wjbot: no, choppy is <reply> If you ever get a slow and choppy sound, make sure you add "noapic" to the kernel boot parameters.
19:42.24jbotCorydon-w: okay
19:42.30joe--fhas anyone used VoxBone with astericks?
19:43.04Corydon-wYes, but it's Asterisk, not astericks
19:43.15joe--fha
19:43.16joe--fsry
19:44.09Corydon-wvoxbone works fine
19:44.09joe--fCorydon-w: Would you recommend going with VoxBone for VOIP origination?
19:44.09Corydon-wI don't recommend any particular service in my role here.
19:44.38joe--fI'm planning on having asterick manage a bunch of conference calls, and was wondering what would be best..
19:45.33joe--fEach conference call will have a unique PIN, that will only be available for 'X' hours..  and i'm going to parse the Call Records to verify that the two people made the conference call based on the pin# they were assigned..
19:46.05joe--fjust wondering if voxbone is a good choice to start this idea with.. and potentially move over to a large hosting provider..
19:46.18_VoiceMeUp_COMhmm
19:46.44_VoiceMeUp_COMjoe depends on where your clients from
19:46.53_VoiceMeUp_COMUSA/CAN or EUROPE
19:46.55joe--fthey'll be primarily within the US
19:46.58joe--ffor now
19:47.04_VoiceMeUp_COMand concurent cannels
19:47.26_VoiceMeUp_COMwe have a few Datelines clients and call center clients that scaled up to 300 calls on a toll free number without isues
19:47.29*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
19:47.31joe--fyeah, isn't it cheaper to just get a bunch of DID's instead of a bunch of channels for one DID?
19:47.34_VoiceMeUp_COMhow many people per conf and how many confs ?
19:47.36Defrazshouldn't this work exten => _12083467090,1,Dial(IAX2/12083467090@wc-pbx.fuzecore.com)
19:47.42_VoiceMeUp_COMno
19:47.48_VoiceMeUp_COMcheaper per channel
19:47.58_VoiceMeUp_COMsince you can resuse channel banks for active calls
19:48.09_VoiceMeUp_COMand nothave lets say the 4 channels per number sitting and idling
19:48.11joe--f_VoiceMeUp_COM: we're just going to be having... 2.  (the reasoning is that the people aren't allowed to call directly to one another, since we need to log everything)
19:48.16_VoiceMeUp_COMper channel is usually best on volume
19:48.32joe--f_VoiceMeUp_COM: hmm ok, that sounds good
19:48.39CunningPikeWould there be any detrimental effects of storing voicemail on a remote NFS export with nolock?
19:48.40_VoiceMeUp_COMso 2 channels toll free USA ?
19:49.20joe--f_VoiceMeUp_COM: yes, i mean we're potentially hoping to have a couple dozen conferences..
19:49.22*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-72.ph.ph.cox.net)
19:49.33_VoiceMeUp_COMok at same time ?
19:49.36joe--fyes
19:49.42joe--fbut just 2 people per conference
19:49.45DandanCorydon-w: thanks :)
19:49.49_VoiceMeUp_COMyour best bet is hosting these on different weekdays to scatter the volume
19:49.53cr4z3dis 5060 the only port asterisk binds too when starting up?
19:49.55redaxI don't have incoming DID with Zaptel (bristuff) what's did I missconfigured?
19:50.25_VoiceMeUp_COMtelse youll have 24 channels idling on 99% of the time.. ( bette ryet .. 6/7 of the time ) if ocnf are weekly
19:50.35_VoiceMeUp_COMcr4z3d no
19:50.37Dandananyone can recommend voipdiscount? anyone using them?
19:50.55_VoiceMeUp_COMcr4z3d these lots
19:50.57cr4z3d_VoiceMeUp_COM, what else does it bind too? cuz i keep getting a bind error.. never happened until i actually had to restart my computer
19:51.08_VoiceMeUp_COMdundi/iax/sip/manager/etc
19:51.15_VoiceMeUp_COMwell it shoudl tell you the service in the error line
19:51.18cr4z3d_VoiceMeUp_COM, seems to be while starting manager.c
19:51.22_VoiceMeUp_COMlike..DUndi cant bind to XXXXX
19:51.26_VoiceMeUp_COMthen its manager
19:51.32cr4z3dit didn't say what por though
19:51.36_VoiceMeUp_COMnano /etc/manager.conf
19:51.48Dandanuse VI, Luke!
19:51.50_VoiceMeUp_COMport = 5038
19:51.56_VoiceMeUp_COMyou must have another process runing
19:52.01_VoiceMeUp_COMOr another app
19:52.13_VoiceMeUp_COMif you need to run multipleinstances increment that port by 2
19:52.15Dandanor another instance of asterisk (<= done that)
19:52.16_VoiceMeUp_COMto be sure
19:52.42_VoiceMeUp_COMi meant /etc/asterisk/manager.conf
19:52.43_VoiceMeUp_COMsorry
19:52.53joe--f_VoiceMeUp_COM: what do you mean by hosting these on different weekdays.. like use a different DID for each day of the week?..
19:52.56cr4z3dyeah _VoiceMeUp_COM 5038 but netstat -p shows nothing on that port
19:54.00*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
19:54.37JerJer[mobile]Dandan:  perhaps take a look at discountvoipoutlet.com
19:54.53JerJer[mobile]unless you need something in .EU
19:54.54DandanJerJer[mobile]: thanks, I need something with 0 to Poland
19:54.59DandanI do :/
19:55.03JerJer[mobile]ah
19:55.08_VoiceMeUp_COMnah
19:55.09JerJer[mobile]then i can't help
19:55.09DandanWell, I need termination in poland for free
19:55.14_VoiceMeUp_COMi meant host conferences on each day..
19:55.20_VoiceMeUp_COMconf 1 on monday..conf2 on tuesday etc
19:55.21Dandanbut I will originate from the US
19:55.36JerJer[mobile]nothing is free
19:55.46Dandanwell, flat fee
19:55.47_VoiceMeUp_COMcr4z3d telnet localhost 5038
19:55.47Nuggettelnet is eeeeeeevil!
19:56.03cr4z3dyes yes it is
19:56.05cr4z3dbut ok
19:56.09_VoiceMeUp_COMtelnet
19:56.11_VoiceMeUp_COMlol
19:56.20*** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca)
19:56.21_VoiceMeUp_COMok hmm tought nugget ws a chicken mcnugget bot
19:56.21joe--f_VoiceMeUp_COM: well, so, i have no control over when the conferences take place.. they'll be fairly random, since caller1 and caller2 will both select a time that's good for them..
19:56.29_VoiceMeUp_COMoh
19:56.30_VoiceMeUp_COMi get it
19:56.33_VoiceMeUp_COMthen you are ok
19:56.42Defrazexten => _12345564343,1,Dial(SIP/12345564343@blah.blahblah.com)
19:56.43_VoiceMeUp_COMyou can sassume its scattered around the biz week
19:56.46joe--fso potentially one day there could be a ton of conferences, and the next day none.
19:56.50joe--fya
19:57.24DefrazOkay that line work with SIP but when I replace the SIP with IAX2 it doesn't' work.
19:57.30DefrazWhat am I missing?
19:58.24Defrazfrom what I have found I can't see any difference between sip and iax on the dial command.
19:59.08cr4z3d_VoiceMeUp_COM, hmm interesting i actually was able to get in
19:59.26cr4z3dnow how do i kill that
19:59.33cr4z3dand why didn't it show up in netstat
19:59.59JerJer[mobile]Defraz:  matching without any N or X  ?
20:00.49DefrazWhen a call comes into my [from-pstn-custom] context it will forward the call to the other asterisk box (blah.blahblah.com)
20:01.03Defrazand it works with SIP but when I replace sip with iax2 it doesn't work.
20:01.13DefrazI have incoming trunks setup for both sip and iax2
20:06.17*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
20:07.03_VoiceMeUp_COMnetstat -anl
20:07.16_VoiceMeUp_COMnetstat -an |grep 5038
20:13.01cr4z3d_VoiceMeUp_COM, found it using ps -ef.. wonder why i couldn't asterisk -r
20:14.59*** join/#asterisk ReD-MaN (n=redman@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com)
20:15.46*** join/#asterisk Bobocop (n=Bobocop@83.168.90.41)
20:16.32_VoiceMeUp_COMah
20:16.34_VoiceMeUp_COMhmm
20:16.39_VoiceMeUp_COMkillall -9 asterisk
20:16.44_VoiceMeUp_COMthen restart it
20:17.00cr4z3di just did a pkill asterisk and started it up and worked fine
20:17.01*** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)
20:17.13_VoiceMeUp_COMperfect
20:17.21Bobocopwhat's wrong with RxFAX and detecting end of incoming fax transmission? Why don't RxFAX hang up?
20:17.30*** join/#asterisk Buglouse (n=Buglouse@adsl-69-215-134-89.dsl.milwwi.ameritech.net)
20:17.53*** join/#asterisk bkruse (i=bkruse@nat/digium/x-4c440cc43b449661)
20:18.04syzygyBSDthat isn't rxfaxes job?
20:18.27syzygyBSDit should return after finished, just not hangup
20:18.56Mercestes<PROTECTED>
20:19.18*** join/#asterisk thoughtpolice (n=austin@c75-111-146-82.plaicmtc01.tx.dh.suddenlink.net)
20:19.32jm|laptopsomeone with a FWD account want to help me?
20:19.44jm|laptopwith some test calls
20:21.53Bobocopyou're right :) but right now RxFAX stays on the line, waiting for something, even after remote fax machine disconnected... So line stays busy forever. Is the AbsoluteTimeout only solution?
20:21.56*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
20:22.02BobocopMercestes: thx :)
20:24.19BobocopI'm wondering if this is normal behaviour of RxFAX, or it's just my broken setup....
20:25.45*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
20:26.22_VoiceMeUp_COMBobocop add ====   exten => h,1,Hangup()
20:26.42*** join/#asterisk malcolmd (i=malcolmd@pdpc/sponsor/digium/malcolmd)
20:31.18*** join/#asterisk X-Rob_ (n=rob-x@CPE-58-167-128-40.qld.bigpond.net.au)
20:34.29Bobocop_VoiceMeUp_COM: no change :(
20:36.24*** join/#asterisk CyberPony (n=CyberPon@66-194-25-11.static.twtelecom.net)
20:36.27*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:37.11BobocopI'm assuming, that * gives total call handling responsibility to RxFAX, so it seems that RxFAX is the one who should release line... or at least exit after successfull transmission, and return control to *
20:37.26Bobocopor am I totally wrong? ;)
20:39.48*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
20:44.01_VoiceMeUp_COMhmm
20:44.07_VoiceMeUp_COMsip debug it
20:44.15_VoiceMeUp_COMmake it verbose and ebug 999
20:44.39_VoiceMeUp_COMno idea , its really easier when you see the logical aspects
20:44.52Bobocopsip debug?.... this is handled by FXO on local machine
20:45.54_VoiceMeUp_COMhmm didnt know
20:45.55_VoiceMeUp_COMok
20:46.43Bobocopyup, I wish I knew more about its logic too :)
20:46.45*** join/#asterisk hallship (n=hallship@70.103.238.2)
20:47.36Bobocopthe worst thing is, that I have to in fact limit maximum fax size because of setting resonable timeout value
20:48.42Bobocopso when I set timeout for lets say 2 minutes, the maximum fax length will be no more than 2 pages. or even 1.
20:48.51hallshipI am stumped.  I am trying to setup a sip call relay through my asterisk server.  Call comes in on a sip trunk and is then rerouted right back out to a different number.  The issue is with callerid.  I can't seem to get the originating number to show on the final destination.
20:49.07hallshipIt seems be sending the sip account info for the second leg of the call.
20:49.38hallshipDoes anybody know how I can edith the sip From: section on an outbound sip call?
20:51.03*** join/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br)
20:51.24_VoiceMeUp_COMfromsuer
20:51.30Bobocopanyway, thx for help _VoiceMeUp_COM :) cu
20:51.31_VoiceMeUp_COMfromuser = blah
20:51.39_VoiceMeUp_COMno prob
20:51.40*** part/#asterisk Bobocop (n=Bobocop@83.168.90.41)
20:52.02hallshipand blah can be whatever?
20:52.18_VoiceMeUp_COMits gonna be username
20:52.22_VoiceMeUp_COMthe from:BLAH
20:52.25_VoiceMeUp_COMis the callerid
20:52.35_VoiceMeUp_COMyou culd set it on the phone
20:52.39_VoiceMeUp_COMor use SER to rewrite
20:52.50_VoiceMeUp_COMone way or another youll use trustrpid
20:52.52_VoiceMeUp_COM=yes
20:54.21_VoiceMeUp_COMso from:"BLAH "<554544333> or from;user@server or whatever you need to pass it .. but best bet is the SER soluition , you could add headers and or modifiy the full line to whatever you need it to be
20:54.53_VoiceMeUp_COMexample : SER is user based.. bob@domain.com , asterisk is extension based.. NXXNXXXXXX@domain.com
20:55.25_VoiceMeUp_COMthats when you ccan use SER to actually poll a db to figure the did to show for bob and replace all this by from:"BOB"<5554443333>
20:57.45hallshipHmm, Okay that's interesting.  Let me give it a try.  Thanks!
20:59.08_VoiceMeUp_COMk
21:02.01*** join/#asterisk bkruse (i=bkruse@nat/digium/x-87cbb2a170f6d9ea)
21:03.55Hmmhesayshmm can you use # with read?
21:04.18Hmmhesaysi mean can cmd read read the # key and return it
21:04.41_VoiceMeUp_COMeads a #-terminated string of digits a certain number of times from the
21:04.42_VoiceMeUp_COMuser in to the given variable.
21:04.48_VoiceMeUp_COMoh
21:04.50Hmmhesaysapparently not
21:04.51_VoiceMeUp_COM#.. hmm
21:05.05_VoiceMeUp_COM<PROTECTED>
21:05.05_VoiceMeUp_COM<PROTECTED>
21:05.11_VoiceMeUp_COMno since its a end of line terminator
21:07.28*** join/#asterisk hrmphh (i=patrick@notchill.com)
21:07.29hrmphh[Apr 23 14:03:07] DEBUG[9751] chan_zap.c: Got event Wink/Flash(3) on channel 1 (index 0)
21:07.33hrmphh[Apr 23 14:03:07] DEBUG[9751] chan_zap.c: Winkflash, index: 0, normal: 20, callwait: -1, thirdcall: -1
21:07.35hrmphh[Apr 23 14:03:07] DEBUG[9751] chan_zap.c: Already have a dsp on Zap/1-2?
21:07.40hrmphhwhat does that mean exactly?
21:10.27*** join/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl)
21:10.35*** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195)
21:11.08khronosHmm, having a small problem stripping off a number in my dial string.
21:11.31khronosWhat I want to do is hit 9 as a prefix and everything after that excluding the 9 get sent to the other system.
21:11.34khronosI have
21:12.15khronosexten => _9.,1,Dial(IAX2/me@my_friend/{$EXTEN:1})
21:12.22*** join/#asterisk |BLiX| (i=asa@12.192.197.15)
21:12.45khronosOn the other system it seems to not be stripping off the 9 at the beginning of the dial string.
21:14.44*** join/#asterisk danicholson (n=danichol@203.89.191.222)
21:17.20*** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
21:22.40*** part/#asterisk danicholson (n=danichol@203.89.191.222)
21:26.03*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
21:29.40*** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-140-134.ny325.east.verizon.net)
21:30.07|BLiX|anyone know where to find documentation on users.conf?
21:38.41*** join/#asterisk SomethingISODD (n=dan@xplr-ts-v10-208-114-188-192.barrettxplore.com)
21:38.49VecDo iax2 friends authenticate eachother in both directions i.e. incoming and obviously outgoing calls ?
21:39.11SomethingISODDhello all question using G729 how many concurrent calls can be on a 10MG line
21:39.12Vec|BLiX| : the documentation within users.conf is pretty good, its basically iax.conf and sip.conf in one
21:39.14SomethingISODDMG=MEG
21:39.45jm|laptopI have updated to 1.4 and now I don't see call progress despite core set verbose 999 and core set debug 999
21:39.47jm|laptopwhy is that?
21:40.06VecSomethingISODD : I would say 1000 max
21:40.20SomethingISODDok Vec thanks
21:40.31*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
21:40.46Vecjm|laptop : no idea, works fine for me
21:40.54jm|laptopoh
21:40.57Vecjm|laptop : try core set verbose 10
21:40.59jm|laptopok thanks :)
21:41.09Vecand take off core set debug
21:41.15VecI think its core set no debug
21:41.22Vecor core set debug 0
21:41.34jm|laptopCore debug is now OFF
21:41.43Vecset core verbose 10
21:41.47Vecthen see what happens
21:41.49jm|laptopdid that
21:41.52jm|laptopstill nothing :S
21:41.52Vec999 seems like a little much
21:42.03jm|laptopit's Debian, though - so this package might still be 'unstable'
21:42.38Vecjm|laptop : u can try compiling it from source, also try run it asterisk -cvvv
21:46.04jm|laptopVec: my bad; phone was still connecting to OLD asterisk box - sorry.
21:46.32jm|laptopblimey - lots changed 1.2 --> 1.4 :S
21:49.22Uatec_does anybody use 1.3?
21:50.48*** join/#asterisk danicholson (n=danichol@203.89.191.222)
21:54.14*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
21:55.01elriahDoes SayDigits not take a variable?
21:56.56*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
22:02.38anonymouz666it takes
22:02.53anonymouz666saydigits(${blah})
22:02.54elriahYou guys see anything wrong with this? exten => s,n,Set(FILENAME=${${TIMESTAMP}:-11:6})
22:03.04elriahIt keeps evaluating null.
22:03.33*** join/#asterisk denon (n=denon@tooth.decay.org)
22:03.33*** mode/#asterisk [+o denon] by ChanServ
22:04.21elriahi.e., in the consule it says Executing Set("SIP/whatever-b71c7228", "FILENAME=")
22:04.25*** join/#asterisk allankardec (n=root@189-19-59-138.dsl.telesp.net.br)
22:04.29allankardechello
22:04.54elriahAnd it should set FILENAME to the timestamp seconds.
22:05.15Qwell[]elriah: You're trying to execute a number
22:06.32elriahQwell[]: I'm looking but I don't see it.  Throw me a bone (please)
22:06.44Qwell[]You don't need the outer ${}
22:06.54Qwell[]move the :-11:6 inside the ${TIMESTAMP}
22:07.07elriahhrm.. Ok, thanks.. Going to try it now..
22:08.27elriahThanks, Qwell.
22:09.06*** part/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl)
22:09.38allankardecSomebody could help me with the instalation of g729 of intel?
22:09.46Qwell[]allankardec: no
22:09.58allankardecthanks
22:10.30anonymouz666lol=yes
22:12.48*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
22:12.52FuriousGeorgehey all
22:13.34jovannottidid you pay for the license allankardec ?
22:13.57allankardecis one license free
22:14.12allankardecsorry, I'm Brazillian
22:14.48jovannottig729 is not free for installation allankardec, at least do you have bought TC400B card
22:16.20jovannottior do you pay 10USD for each licence
22:16.29FuriousGeorgedoes anyone use gentoo here?  ive always used the svn builds, but i notice gentoo installs all these distro specific patches, and while i have no idea what they are for exactly, im wondering if they may be a good thing
22:17.06allankardecthe intel fornush
22:17.21jovannottido you mean IPP ?
22:17.31allankardecyes, i do
22:18.02allankardecI get non commerce,
22:18.10allankardecI got non commerce
22:18.14*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
22:18.14FuriousGeorgeanother thing.  i didnt mess with the -march setting when i built it (i have a socket 939 opteron 165) and i noticed "k8" was automatically selected
22:18.26FuriousGeorgebut gentoo recommends using "opteron" for my march setting
22:18.27*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
22:18.49jovannottiok, I tried but I have not deep knowledge in C to create the drivers, I saw a page for a guy from Latvia or England that developed some years ago, but the versions are obsolete now
22:19.01FuriousGeorgethe reason im asking about all this is b/c i have two identical systems.  one deadlocks once a week one doesnt.
22:19.26*** join/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl)
22:19.54FuriousGeorgei built the kernels separately, so maybe there is some difference there.  i could try taking the kernels from the good system.  the other difference is that the bad system has two tdm400p cards
22:20.50*** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net)
22:21.02BrianR___Anyone here use asterisk with vitelity?
22:21.22*** join/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar)
22:21.39dnielchupenme bien un huevo
22:21.51dnielaguante la linea con tono
22:22.03dnielaguante avaya y nortel
22:22.19dnielfile: tocame bien la punta de la pija con tus dientes
22:22.23*** join/#asterisk ardor (n=Miranda@las-static-66.18.135.148.mpowercom.net)
22:22.33ardorHi everyone
22:22.38filedniel: ...hi?
22:22.40FuriousGeorgeheh
22:23.04dnielfile: agarra tu boquita y pegate una flor de lamida de pija.
22:23.12jovannottiits an argentinian guy that needs to be banned
22:23.26fileoh, banning
22:23.27fileI can do that!
22:23.30FuriousGeorge:)
22:23.37*** part/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar)
22:24.38FuriousGeorgeso, any gentoo users in here that dont use the svn packages?  im thinking of doing that myself as I'm having issues with deadlocks
22:25.06*** join/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar)
22:25.09dniel:p
22:25.16dnielfile: :p
22:25.18*** part/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar)
22:25.41filewell!
22:25.56*** mode/#asterisk [+b #asterisk!*@*] by file
22:26.00fileoh crap
22:26.10Qwell[]eh?
22:26.13FuriousGeorgei bet this kid is in here under another alias and wants nothing more than some extended attention
22:26.23filemanually typing out ban statements sucks
22:26.28filewhen you haven't done it in years
22:26.39anonymouz666they think that we can't understand the damn spanish
22:26.47Qwell[]/mode +b hostmask channel
22:26.48Qwell[]:D
22:26.56FuriousGeorgeyeah! damn spanish speakers de la verga
22:27.06Qwell[]file: or just /kickban <user>
22:27.29Qwell[]/quote PONG
22:27.37anonymouz666telnet sucks. BitchX was nice
22:27.48Qwell[]erm, I guess you wouldn't /quote that on telnet, heh
22:28.17*** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167039254.pppoe-dynamic.nb.aliant.net)
22:28.35fileit's my room mate!
22:29.02ardorI'm trying to get some Music inside of my app_conference, Is this possable?
22:31.08*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
22:32.37allankardecjovannotti, I installed the g729 codec, after the concluid download, if i run it, i do receive the info: permission denied
22:33.22Qwell[]allankardec: I was serious earlier.  No, you won't get help with that here.
22:33.26allankardecjovanotti: sorry i am slow in tzping in english
22:34.25*** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net)
22:34.46*** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net)
22:36.29digiterataHi all, I'm doing some Asterisk testing with Jabber/Jingle clients and was wondering if anyone can recommend a decent voice client that runs on linux. At the moment my test subject is running GoogleTalk in a Virtual Machine. Hoping theirs a native solution.
22:36.51allankardecquell, sorry i missed your info, can zou repead that again pl
22:37.29anonymouz666allankardec: ele não vai te ajudar pq esse codec é patenteado e vc precisa comprá-lo.
22:37.43Qwell[]anonymouz666: something like that
22:38.10allankardecanonymouz666, ehhehehehehehe, valeu pelo português, agora conseguir ler rapido
22:38.28Qwell[]no, no rapido
22:39.05jm|laptopholy crap: I have really broken IAX :s
22:42.55allankardecspeaking serious, do you know this site www.readytechnology.co.uk/open/ipp-codecs-g729.1 ?
22:45.08bulleallankardec: the situation is a bit complicated when it comes to g729
22:45.21bulleallankardec: as the codec has parts of it being patented
22:46.04bulleallankardec: now, in some countries, like the u$a, there are software patents, and laws against aiding people to commit crimes, eg, helping people to use patented source without the correct licence
22:46.18*** join/#asterisk mholman (n=mholman@203-206-107-167.dyn.iinet.net.au)
22:46.23jovannottiactually you can do it, but downloading IPP from Intel Page, and then trying to compile allakardec, I tried but I prefer pay 10 USD for each license
22:46.25bulleallankardec: so, depending on where you are, you might or might not be required by law, to have a licence, in order to run g729
22:46.59jovannottiI was looking for codecs g729 and g723,  I tried paying to digium 10 USD for each licence. and it works fine
22:47.01bulleallankardec: i would suggest you buy a licence, if you live in a country where its required, but if your country doesnt require it, dont pay for the licence
22:47.44bulleallankardec: anyway, as many people here are american citizens, they are not going to discuss any unlicenced software with you
22:48.45JTerr what the hell
22:48.55JTthere is nothing wrong with discussing patented software
22:49.34bulleJT: im of the same opinion, but i got to take lots of flack, for helping out with non licenced stuff
22:49.38allankardecbrazil don't require to buy the licences
22:49.52bulleallankardec: ye, brazil is a sane country in that respect
22:50.18bulleallankardec: you guys also got wanderlei silva =)
22:50.46anonymouz666rodrigo minotauro is a better fighter
22:50.48JTbulle: there's a differennce between helping someone out with stolen code and discussing code that a patent is required due to something it provides, in a business environment :)
22:51.24bulleJT: well, this chap seems to want to get help on how to use this code without having a valid licence
22:51.26anonymouz666allankardec: how do you know that?
22:51.31bulleJT: as i said, i dont disagree
22:51.38bulleanonymouz666: brazil doesnt have software patents ...........
22:51.51JTbulle: so it'd depend if the code was subject to a copyright violation or not
22:52.08JTwhether he gets a patent for a function it provides is something else
22:52.10anonymouz666bulle: where did you read that?
22:52.15bulleJT: well, as i said, last time i tried to help, i got shitloads of rants about how i was a thief and criminal etc
22:52.33JTbulle: i think the intel code might be stolen
22:53.15jm|laptopI have a silly question :/
22:53.30bulleanonymouz666: nowhere, i have been talking to some brazilian friends of mine
22:53.36jm|laptopin 1.2 my verbose and debug were colourful - in 1.4 monochrome.  What can I do?
22:53.36allankardecMy asterisk is logined the of vonage, the out call is perfect, but, coming call is problem with g729
22:54.02anonymouz666bulle: ok.
22:54.26bulleanonymouz666: software patents are realy the exception, not the rule, out in the world
22:55.31anonymouz666there are a lots of companies in here that uses a paid version of g729, must have a reason for that.
22:56.15DoDaT69how can I add enum checking to outbound route?
22:57.58[TK]D-FenderDoDaT69, Its all just dialplan, go look at the apps related to enum
22:58.38DoDaT69okay
22:59.02bulleJT: well, if its all software, then its per definition a software patent
22:59.26bulleJT: atleast here, you cant patent algorithms and software
23:00.29*** join/#asterisk evilbuny (n=flycasua@kirk.ozwide.net.au)
23:01.33bullehmm, time to slepp
23:01.35JTbulle: i believe the g.729 patent covers the algorithm
23:01.55allankardechello, I have friends that actived the g729 of intel, but I dont speak with him
23:03.16jm|laptop~skype
23:03.26jbotmethinks skype is evil. see gizmoproject.com, or offering free landlines: http://share.skype.com/sites/en/2006/05/free_calls_to_all_landlines_an.html
23:03.57Qwell[]jbot: no, skype is  evil. see gizmoproject.com
23:04.00jbotQwell[]: okay
23:04.22Qwell[]jbot: no, skype is evil. see gizmoproject.com
23:04.26jbotQwell[]: i already had it that way
23:04.32Qwell[]No you didn't
23:04.39allankardechello all, I tired my decision is going to buy.
23:04.45jm|laptopQwell: way to argue with a bot :S
23:04.49JT~skype
23:04.51jbotsomebody said skype was evil. see gizmoproject.com
23:04.52jm|laptopbut skype is evil, right?
23:04.55Qwell[]way to argue with an op arguing with a bot
23:04.56allankardecto licences from digium
23:05.06*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
23:05.09JTlooks like the bot did listen the first time
23:05.09jm|laptopQwell: touche
23:05.25allankardecthanks all
23:05.40Qwell[]jbot: a smart bot wouldn't have stripped the whitespace automatically
23:05.59anonymouz666chan_msn.c would be possible?
23:06.03Qwell[]anonymouz666: sure
23:06.11Qwell[]it's just software
23:06.37bullemsn nowadays just uses sip and rtp or ?
23:06.43jm|laptoph.323?
23:06.56jm|laptopsome I'm guessing I can't just add a skype 'trunk' still?
23:07.12jm|laptop(for free)
23:07.14jm|laptops/some/so/
23:07.15allankardecbut, I have losing time
23:07.22Qwell[]sure you can, if you write a module for it
23:07.34jm|laptop(:
23:08.15allankardecqwell, jonajona, anonymou666 and bulle thank for help me
23:08.17jm|laptopI don't think I like Skype users anyway
23:08.38jm|laptopwith enum/p2p sip uri they will soon lose their foothold?
23:08.40anonymouz666I don't like, but my customers do.
23:08.51Qwell[]customers are nubs
23:09.34jm|laptopseems $600 is too little bounty
23:09.41jm|laptopnot surprising with such a weak dollar atm ;)
23:09.52JTthe old windows client that did voip was h.323
23:09.59JTmsn messenger is supposedly sip
23:10.34*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
23:11.02*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:11.45jm|laptopso why *hasn't* someone reverse engineered Skype?  Licensing? or is it just a pita?
23:12.14digiterataSorry to jump in. Did I hear someone say I can add MSN via a standard SIP channel?
23:12.31JTjm|laptop: some people have, but no-one cares about it, it's rubbish
23:12.41jm|laptopJT: agreed
23:12.50jm|laptopwait: I didn't post that line?!
23:13.05digiterataor is this still theoretical. rubbish to nerds like us, but 40 million people use MSN - not rubbish.
23:13.17Qwell[]digiterata: skype is rubbish
23:13.20jm|laptopoh yes I did;  [00:08] <jm|laptop> I don't think I like Skype users anyway
23:13.21Qwell[]MSN just slightly less so
23:13.47jm|laptopJT: isn't it one of the biggest worldwide 'private' PBXes though?
23:14.06digiteratayeah agreed, Skype - closed, proprietary, crappy p2p architecture, too inconsistent to be useful; too closed to be inter-operable
23:14.08paavumHi
23:14.08JTjm|laptop: possibly
23:14.17paavumHow can I disable zttranscode from beeing loaded?
23:14.32Qwell[]paavum: don't load codec_zap
23:14.46*** join/#asterisk hallship (n=hallship@70.103.238.2)
23:15.00digiterataI'm actually working actively on getting gTalk working with Asterisk - but I'm not very experienced and the documentation is a bit sparse at the moment.
23:16.10anonymouz666but it works.
23:16.28anonymouz666I already configure that
23:16.55digiteratait works, just not for me - yet.
23:16.56hallshipHello, I dropped in earlier with a callid issue.  I got some advice but that doesn't seem to be the answer to my delima.  Here is the issue in a nutshell.
23:17.10Qwell[]hallship: I'm allergic to nuts.
23:17.13hallshipI am trying to setup a sip call relay through my asterisk server.  Call comes in on a sip trunk and is then rerouted right back out to a different number.  The issue is with callerid.  I can't seem to get the originating number to show on the final destination.
23:17.31hallship:)
23:17.44hallship<PROTECTED>
23:18.07digiterataanonymouz666: you know how to configure Jabber/Jingle? Could you show me how? I've got a simple AsteriskNow virtual machine running. not sure how to configure.
23:18.18JThallship: what does it show instead?
23:18.25hallship_VoiceMeUp_COM_ suggested setting the fromuser to the callerid I wanted.
23:18.46hallshipIt shows the info that is configured for the sip account that the sip trunks are registerd with.
23:19.13paavumHi.. Im having severe problems with * 1.4
23:19.17anonymouz666digiterata: there is an example in voip-info that can help you
23:19.20JThallship: why dob people in like that?
23:19.23hallshipThe same scenario works fine using IAX or ZAP channels, btw.
23:19.25paavumI have a queue with analog extns
23:19.46paavumbut every time I get into the queue I get a "Cant ring Zap/xx" msg on  the cli
23:19.57paavumand then I get "tt-allbusy"
23:20.06paavumplayed to me
23:20.15paavumas if it couldnt get into the queue
23:20.32paavumcan anybody give me a hand?
23:20.33digiterataI'll go look again. might have questions for you though. Hope you don't mind. It would make my night if I could figure this out tonight.
23:20.35hallshipJT: Sorry I didn't get your question..?
23:20.48JT<PROTECTED>
23:21.00JTpeople will be weary to help you when you say things like that
23:21.16hallshipOh, ya,  I'm sure he meant well.
23:21.22JTthere is no need to bring up who told you something previously
23:22.03[TK]D-Fenderpaavum, pastebin the clie output of the failed call from beginning to end, and then your applicable config files
23:22.16hallshipOh I see.  Well I wasn't trying to bad mouth him.  Mostly just trying to spark some recollection if anyone was still here from the previous discussion.
23:22.52JThallship: so what callerid are you getting?
23:23.06*** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com)
23:23.11[TK]D-Fenderhallship, Do you have something in writing saying that the provider you are calling out through even permits you to set your callerid?
23:23.27hallshipI get the callerid of the SIP account that the trunks ring into and out on.
23:23.46JTplease don't use the word trunks, they're not trunks in that context :)
23:24.23[TK]D-Fenderhallship, who's callerid are you seeing, the INCOMING calls, or the callerid of the account you are using to dial OUT from?
23:24.29hallship[TK]D-Fender: no, althought their support says it should be possible.
23:24.41IOscannerAnyone have good recommendation of good outbound  terminication for US.
23:25.12IOscannerLooking for a carrier that has good rates with good uptime and support.
23:25.16Qwell[]terminication?
23:25.23hallshipJT:  The call is ringing in on what I've come to know as SIP trunks.  What shall I call them instead?
23:25.33*** join/#asterisk uski (n=uski@ALagny-153-1-18-27.w86-198.abo.wanadoo.fr)
23:25.36[TK]D-Fenderhallship,  I did NOT ask you a yes/no question.
23:25.36JThallship: connections, sessions
23:25.54IOscannersorry Termination
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23:26.00[TK]D-Fenderhallship, Get your head on straight and answer the very direct question I asked.
23:26.18IOscannerwow missed that
23:26.19hallshipI am seeing the account I am dialing out with.  I want to see the callid of the account I am dialing in with.
23:26.24uskihi, i have asterisk 1.2.13 and it segfaults... where do i send the core dump ? ;)
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23:27.00[TK]D-Fenderhallship, before you dial out change the callerID inyour dialplan and see if you can see the change.  if you can't, you're up a creak as they say...
23:28.19hallshipYou mean change the CallerID variable to the whatever I want to to see?  If I noOp the ${CALLERID} variable before I dial out it's correct.  But they are keying off of the From: in the sip header.
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23:30.17hallshipI did find a forum talking about setting some sort of custom variable and getting your provider to key off of that for callerid.  Anyone here had any experience with that?
23:30.50JTthe ${CALLERID(num)} function
23:31.05JTit's not ${CALLERID} anymore
23:31.28*** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au)
23:32.01hallshipOkay, that's the varaible I have been playing with.  And no, setting it doesn't do the trick.
23:32.06[TK]D-Fenderhallship, are you setting "fromuser" in your SIP peer?
23:33.41hallshipWell I'm fairly sure that will work but that doesn't address this issue, in my understanding.  becuase I need to send the originating callers callid info.
23:34.32hallshipI can't set the fromuser on the fly right?  I haven't found anything showing how that could be done anyway.
23:35.10[TK]D-Fenderhallship, I just asked if you had a very specific value set in your sip.conf peer entry.  can you just answer the question......
23:36.27hallshipno
23:37.27hallshipI haven't changed it in the sip  peer.
23:37.40hallshipwhy do you ask?
23:39.40hallship[TK]D-Fender and JT thanks, I'll check back after some more digging.  I get the feeling this isn't the best time :)
23:40.06[TK]D-Fenderhallship, I asked if you SET it to something
23:40.25[TK]D-Fenderhallship, you are getting all your answers mismatched to the question asked.
23:41.05hallshipWhat do you mean?  I ran this in my dialplan.  ;exten => 1235,1,set(fromuser=${CALLERID}
23:41.27hallship- the ; of course :)
23:41.31[TK]D-Fenderhallship, again WRONG ANSWER.  sip.conf.  I've said it like 3 times now.
23:41.54[TK]D-Fenderhallship, "fromuser=[somethinghere]" in SIP.CONF.
23:41.54hallshipI did answer that.  I said no I haven't set that in my sip peer.
23:42.22JThallship: it will never be a good time if you don't answer peoples' questions when they attempt to help you for free :)
23:42.29hallshipBecause I don't see how that would help.  This needs to be a dynamic/automatic change that happens on each call.
23:42.38[TK]D-Fenderhallship, No, you said "I haven't changed it in the sip  peer."  changed from WHAT is what I have to ask myself... does this mean it has a value and you haven't been messing with it *lately*?
23:42.57[TK]D-Fenderhallship, I'm trying to pin down all the things you can do that will screw up your attempts
23:43.28[TK]D-Fenderhallship, ambiguity does not help.  and that "set" you jsut showed me is not valid.
23:44.05[TK]D-Fenderhallship, That is not a variable of any significance in the dialplan
23:45.09hallship[TK]D-Fender: that's what I found.  I was coming back to try and clarify why that was suggested.  That's all.  I don't know why we are so far down this road.  I am quite sure that the fromuser isn't the correct solution at this point.  I merely mentioned it to start a dialogue.
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23:45.43hallshipJT: I'm doing my best to answer what you guys are asking but you seem to have a bone to pick with me.
23:46.07hallshipI am not trying to be ambibous at all.  I am just trying to answer what you are asking.
23:46.18[TK]D-Fenderhallship, No, it could have been involder if you were setting that in your peer entry... wasn't a bad thing to consider, but the wording used was garaunteed to confuse all, and your clairifcations  didn't help :)
23:46.27bkruse_homeJT: keeping the nubs under control?
23:46.40[TK]D-FenderAnyways, I've got to be heading out for a bit.
23:47.32mholmanHi Guys, I have a small problem with asterisk 1.4 + TDM400P...
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23:47.46bkruse_homemholman: let me guess...........it doesnt work?
23:47.53mholmanMy ZAP channels seem to be disconnected exactly 30 seconds after they are put on hold.
23:48.03mholmanDoes this sound familiar to anyone?
23:48.21JerJer[mobile]mholman:  any SIP channels involve d?
23:48.28hallshipThis is a new setup.  callerid has never worked correctly.  I have removed the fromuser setting out of the sip.conf table.  didn't help.  I agree that is what is being sent out.  I am looking for some sort of work around that will allow me to send the originating callerid instead.
23:49.18mholmanthe call is coming through as follows: ZAP -> Asterisk -> SIP -> eyeBeam clients
23:49.46hallshipOkay, thanks for your time.  JT and TK.
23:50.15mholmanIve also tested it using an IAX connection instead of the ZAP one, and on hold works fine.
23:50.51flendersmholman: a similar thing happened to me, and it went away when I got rid of echo on the TDM400
23:51.54mholmanflenders: thx, what setting did you change/did you start getting echo as a result?
23:52.16flendersmholman: also try disabling the busydetect on zapata.conf
23:52.33flendersmholman: you have to play around with tx/rx gains
23:52.42flendersmholman: and also use fxotune
23:53.29mholmanflenders: wow great, I'll go and have a play with those settings. thx
23:53.43mholmanflenders: haven't run fxotune at all yet.
23:54.32flendersmholman: hope that helps
23:54.53bkruse_homeflenders: good advice
23:55.47*** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net)
23:55.54elriahWoohoo!  My first wiki contribution: http://www.voip-info.org/wiki/view/Prompt+Recording+with+Unique+IDs
23:56.11bkruse_homeyay
23:56.19mholmanthanks a lot :-)
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23:58.11FuriousGeorgeany gentoo users here?
23:58.25*** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au)
23:58.48flendersasterisk+gentoo? or just gentoo?
23:58.59FuriousGeorgeasterisk plus gentoo of course :)
23:59.26bkruse_homeFuriousGeorge: i HAVE, whats up;
23:59.33FuriousGeorgeim tying to isolate a deadlock problem, and i'm wondering if i should be using the gentoo ebuilds vs svn
23:59.53*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
23:59.55FuriousGeorgebasically i dont know if its software or hadware or what

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