IRC log for #asterisk on 20110910

01:16.17*** join/#asterisk infobot (~infobot@rikers.org)
01:16.17*** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
01:28.52*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
01:39.34*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
01:56.09*** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com)
02:02.20ChannelZIt's The Law
02:02.55*** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com)
02:09.45*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
02:15.46luke-jrI don't see how to limit calls to/from extensions using GROUPs… all the examples break as soon as someone transfers
02:16.47carrarSIP Devices?
02:16.54luke-jrye
02:16.56luke-jrs
02:17.29p3nguinSet the group, then check the count.
02:17.32luke-jrPolycom SIP devices, I think
02:17.33carrarCould use ChanIsAvail
02:17.40luke-jrp3nguin: that works, until it gets transferred
02:17.58p3nguinWhat happens when you transfer?
02:18.15luke-jrp3nguin: it leaves the original caller in the group for the original destination
02:18.35luke-jrso the transferred destination is left with none in the group, and the original destination is callless and "busy'
02:19.00carrarYou just want to know if they are on the phone right?
02:19.11luke-jryeah
02:19.17carrarUse ChanIsAvail
02:19.24p3nguinWhy would you use groupcount for that?
02:22.00luke-jrp3nguin: that was the suggestion on the note deprecating call-limit
02:22.13carrarGet Coding!
02:22.16p3nguinBut call limit isn't for checking if someone is on the phone, either.
02:22.19carrartime is wasting
02:22.22p3nguinSo there's some confusion.
02:23.07p3nguinUsing GROUP_COUNT is the new way to limit calls, but that doesn't have anything to do with checking if someone is on the phone.
02:25.23p3nguinEither you've gone about something the wrong way, or you gave the wrong answer when you were asked if you just wanted to know if someone is on the phone.
02:28.29luke-jrChanIsAvail always reports it as available ;)
02:28.52carrarSo find out what you are ding wrong
02:28.59p3nguinprobably because it's available.
02:29.21luke-jrp3nguin: I'm just trying to send to busy-voicemail if they're on the line
02:29.34carrarread up on that application
02:29.35p3nguinturn off call waiting on the phones.
02:29.54carrarcore show application ChanIsAvail
02:30.13p3nguinUse DIALSTATUS to send to the appropriate place after they are found to be unavailable.
02:30.17luke-jrDEVICE_STATE seems to be usable
02:30.29luke-jrp3nguin: DIALSTATUS is never BUSY on these phones
02:30.35p3nguinTURN OFF CALL WAITING
02:31.01luke-jrp3nguin: I don't know how.
02:31.03p3nguinIf you have call waiting, the phone is obviously not going to be busy.
02:31.09p3nguinThat's the point of call waiting.
02:31.19luke-jrp3nguin: the phones themselves don't make any indication to the user that there's an attempted call
02:31.31p3nguinNo call waiting tone?
02:31.34luke-jrnope
02:31.50p3nguinWhat kind of phone doesn't have call waiting tones when there is a second call?
02:31.53luke-jrno idea
02:32.10p3nguinYou're actually working on this system, aren't you?
02:32.31carrarluke-jr, perhaps you should just hire someone to do this for you
02:32.41p3nguinIf so, surely you can figure out what kind of phones they are.
02:33.18p3nguinEven the shittiest of phones send an intelligible user agent info.
02:33.33carrarI'm able to do exactly what you want with ChanIsAvail just fine
02:33.45carrarI'll do it for $5,000,000
02:34.13p3nguinYou shouldn't have to, though, and if the channel is available (which it apparently is), ChanIsAvail will always return that it is available.
02:34.22carrar*sigh*
02:34.34carrarok both of you
02:34.50p3nguinIf I have call waiting enabled, my channel will always be available unless I enable DND-busy.
02:35.21carrarcause thats how you wrote your dialplan
02:35.21luke-jr[22:30:17] <luke-jr> DEVICE_STATE seems to be usable
02:35.26carrardidn't need too
02:35.36p3nguinCall waiting has very little to do with dial plan.
02:36.04carrarThough most people like the call waiting tone
02:36.05p3nguinThe same dialplan that is used to send one call is used to send every subsequent call.
02:36.09carrarI do
02:36.22carrarbut some don't want to be bothered by a second call
02:36.32carrarso ChanIsAvail solves that
02:36.49p3nguinIf my phone is able to accept a second call, I'd like to know I am getting a call to be able to answer it, divert it, or ignore it.
02:37.16p3nguinBut if I don't want a second call, I have to turn off call waiting.
02:37.32p3nguinOr I can rewrite the dialplan to use GROUP_COUNT for each device.
02:37.42p3nguinand limit it to 1
02:37.48carrargroupcount is not the way to do that
02:38.05p3nguinChanIsAvail isn't either, since the channel is always available if call waiting is enabled.
02:38.10carrarChanIsAvail IS
02:38.14carrarit works great
02:38.17carrarI use it
02:38.32carrarI've done this before
02:38.41carrarI have customers use it
02:38.44carrarSo yes
02:38.46carrarit works
02:38.49p3nguinExplain to me how ChanIsAvail is going to report back that the phone is busy when there is another "line" on it able to accept a call.
02:38.52carrarit does exactly what he is asking for
02:39.12carrarChanIsAvail can tell you if the device is in use
02:39.24carrarif it is, then don't send a call
02:39.48carrarI invite you to read:
02:39.48carrarcore show application ChanIsAvail
02:40.33p3nguinYou're apparently missing the key point I'm trying to make.  If the phone has call waiting, the channel is always available.
02:40.36carrarand I think your both capable of figuring out how to use it correctly
02:40.52carrarHe does not want to use callwaiting on the phone
02:41.07p3nguinBut he said he doesn't know how to turn it off.  So it is on until then.
02:41.21p3nguinSo it's on, and the channel will always be available.
02:41.22carrarOk whatever, I'm done
02:42.10carrarHe can read his phone manual if he wants
02:42.41p3nguinHe said he can't even figure out what kind of phones they are, so don't be too sure.
02:42.41*** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins)
02:42.48carrarheh
02:43.16p3nguinI've never before seen anyone that can't figure out what kind of SIP phones are in use.
02:43.19carrarVery capable phones he has
02:43.19carrar<luke-jr> Polycom SIP devices, I think
02:43.23p3nguinOr at least have some idea, even if it's wrong.
02:43.41p3nguinHeck, if he has Polycoms, there's a setting for almost everything.
02:44.23carrarsip show peer <peer name>, look at Useragent
02:44.33carrarwhat does it say luke-jr
02:44.53p3nguinBut why would there be no callwaiting tones on additional calls?  There's a setting to suppress that?
02:45.12p3nguinNo beep, no flashy-flashy?
02:47.35carrarcall.callWaiting.ring
02:47.53carrar(beep, ring, silent)
02:48.10carrarbeep is not set
02:48.12carrarif
02:48.48carrar(for polycoms)
02:49.56p3nguinI wish my phones could ring instead of beep.
02:50.03*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
02:50.09carrarHow annouying would that be
02:50.19p3nguinquite
02:51.57p3nguinMy main reason is for when I am listening to a conference (muted, of course), and I get another call.
02:52.04p3nguinI sometimes don't notice the tone.
02:52.20carrarand then what?
02:52.24p3nguinEspecially if I walk away from my desk for a minute and there's a call.
02:52.27carrarput the conf on hold?
02:52.31p3nguinYes.
02:52.35carrarplaying on hold music?
02:52.40p3nguinI'm muted.
02:52.46carrarno
02:52.50p3nguinyeah, I am.
02:52.54carrarmuted via the conf muting?
02:52.56p3nguinyes
02:53.02carrarnot the phone mute, ok
02:53.14p3nguinIf the admin unmutes me while they are on hold, that's their problem.
02:53.57p3nguinNah, I don't rely on phone mute on the conf because it always goes unmute when I switch from speaker to headset or something like that.
02:54.06p3nguinI just mute/unmute with dtmf as needed.
02:54.24carrarUse a cisco phone so you have a shinny red button lite up
02:55.01p3nguinI have a Cisco phone, but that doesn't help when I go away from my desk with the conf on.
02:55.11p3nguinI can't hear the beep or see the light.
02:55.22p3nguinIf it would give a normal ring, that would be perfect.
02:56.09p3nguinBut I don't really want to get a Polycom phone just for that.
02:56.38luke-jrPolycomSoundPointIP-SPIP_330-UA/2.2.0.0047
02:56.42carrarWell
02:56.44carrarThere you go
02:56.52carrarYou have lots of call waiting options
02:56.58luke-jrDEVICE_STATE is easier
02:56.59carrarnone, beep or a ring
02:57.31carraror if you really just don't want to give the end user a option for call waiting use ChanIsAvail
02:57.59p3nguinJust disable call waiting first.
02:58.15p3nguinThen you have the choice of ChanIsAvail or using DIALSTATUS.
03:00.42p3nguinHmm.  I guess now we have to carry guns when we travel by bus.
03:01.05carrarShould always asert your right to carry
03:01.27p3nguinSome goofball in Springfield, MO, shot and killed someone on a Greyhound.
03:01.40carrarYou need a permit to carryconcealed there?
03:01.52p3nguinYeah
03:01.57carrarsame here
03:02.09carrarWashington state
03:03.05carrarwe have a Reciprocity agreements with MO
03:03.16carrarso my CPL is good there also
03:03.42p3nguinIn MO it's not hard to get the permit, but on the IL side of the river, the lawmakers have been fighting over right to carry for some time now.
03:04.14carraryeah none with IL
03:04.42p3nguinWe have very strict rules concerning carrying weapons in IL.  I think they call it like Six Seconds to Safety or some bullshit like that.
03:05.17p3nguinIt has to be in a special type of carrying case on your person (fanny pack-like), and it can't be loaded.
03:05.25p3nguinLot of good that'll do when needed.
03:06.00carrarhaha
03:06.37p3nguin"Oh hang on for a second while I pull out my disassembled firearm, put it together, and load it up... then you can try to mug me."
03:07.37ChannelZBut it's for the children!
03:08.24carrarhow did children survive up to this point in time anyhow!
03:08.33p3nguinOne of the rules for CCW in MO that I found very amusing is that you are required to carry your permit with you when carrying your weapon, but if you don't have your permit, it's not a crime.
03:08.41ChannelZMore of them drown in buckets of water
03:09.09carrarshouldn't be acrime
03:09.39carrarThey should outlaw buckets of water!
03:09.44carrarpermit required
03:09.57carrarWBP
03:10.10ChannelZYes.  Home Depot is a death factory
03:10.16p3nguinI think there is a 3 or 6$ fine for not carrying your permit when you are carrying a weapon.
03:10.39carrarcan always open carry in our state without a permit
03:10.49carrarthats fun to do
03:10.56carrarcops love it
03:11.06ChannelZAnd soccer moms
03:11.20p3nguinIt's silly.  They put a CCW endorsement on drivers license, so I think that should be good enough.
03:12.08ChannelZWhat's silly is thugs that don't give a shit about gun-free zones or permits or anything.
03:12.20ChannelZMeanwhile I have to pay $150 for a permit
03:12.22carrarhttp://www.king5.com/home/related/Raw-Conversation-recorded-by-man-carrying-gun-into-coffee-shop-104215743.html
03:12.28carrarhttp://www.youtube.com/watch?v=DL7gB-M61MI
03:12.28carrarhttp://www.youtube.com/watch?v=3fbKZ2kJp8s
03:12.31carrargood stuff
03:13.15p3nguinIn order to carry openly in Washington, do I have to be a resident or the state or be a resident of a state which has CCW permits?
03:13.26carrarJust a resident
03:13.41p3nguinSo I couldn't get away with it if I go visit you.
03:13.47carrarhaving a CCW is just if it's concealed
03:13.56carrarbut it's just nice to have it anyways
03:14.13p3nguinI don't know which is more desired, concealed or open.
03:14.23carrarwell I would have to re-read the law on open carry if you are not a resident
03:14.53ChannelZ"IANAL"
03:15.00carraropen is better if you want to let people know that they probably are not going to robb you with a knife
03:15.10p3nguinI'd think concealed would potentially get you into a place where you needed to use the weapon, where open carry would be more of a deterent.
03:15.24ChannelZ"Guns make people nervous and when people have them, we're going to follow up an ask questions."
03:15.35carraryeah, screw their rights!
03:15.37carrarheh
03:15.49ChannelZYeah lady, well your Suburban XLT that you can barely see over the steering wheel of makes me nervous, lady
03:15.57p3nguinBut then again, the possibilty of someone having a concealed firearm is also a deterent for some of the inexperienced criminals.
03:16.13p3nguinAnd why do I keep misspelling deterrent?
03:16.19ChannelZDetergent
03:16.23p3nguin:/
03:16.35carrarDeoderant
03:16.44carrarthats misspelled too
03:19.20ChannelZSpelling is for chumps!
03:22.56ChannelZif anyone wants Googly Plus invites, yell.  Not that they're hard to get
03:23.25p3nguinI thought we already had everyone on it that wanted to be on it.
03:23.29carrarI don't have a faceplace login, not sure I need google either
03:23.50ChannelZyeah it's kind of pointless why it's "closed"
03:23.52ChannelZstill
03:23.55carrarBesides, I have my own social web site
03:24.25p3nguinI don't do any other social networking stuff, but I decided to do G+ and be a rebel.
03:24.25ChannelZratemyrack.com
03:24.53ChannelZoh wow that site still exists
03:25.19carrarCan't really see that going away anytime soon
03:26.41ChannelZthere is always a fresh supply of racks
03:28.29ChannelZoh, here.  http://www.nraila.org/Legislation/Federal/Read.aspx?id=7072
03:34.09ChannelZouch
03:39.17*** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com)
03:44.55*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
04:05.37*** join/#asterisk coppice (~chatzilla@116.92.16.50)
04:26.16*** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca)
04:26.33dijibanybody running * on centos in here?
04:27.37p3nguinDoes the OS have that great of an impact on asterisk that you won't accept help from those who are not running CentOS?
04:30.17ChannelZI run .50 CentOS
04:31.12p3nguinGet two hundred of them and you can have a DollarOS.
04:32.04ChannelZand some shitty auto-tune rap
04:33.00coppicemost people with a strong opinion run 2CentOS
04:33.54dijibno im having an issue running * as a daemon still
04:34.14dijibit runs on boot. but i cant call in out internal or anything.
04:34.20dijibuntil i run a -c
04:34.31dijibthen it spawns a new instance
04:34.55dijibsafe_asterisk starts as root, asterisk starts as user asterisk
04:35.06dijibive done all the chkconfig stuff
04:36.03p3nguinStart over.  The problem is that it *does* work?
04:38.22dijibwhen i start from cold boot. safe_asterisk & asterisk both run. but phones have no action. i call my line and its dead air...
04:38.36dijibnot sure what else to say. no worky but the process is running
04:40.32p3nguinIf you kill off any and all asterisk-related processes, can you start it with  asterisk -G asterisk -U asterisk -vvvvddddddddg  ?
04:41.18dijibwhats g u d?
04:41.44p3nguin"man asterisk" doesn't know?
04:42.38dijibapparently not
04:43.08dijib[root@trunk ~]# man asterisk
04:43.08dijib-bash: man: command not found
04:43.09dijib[root@trunk ~]# man /usr/sbin/asterisk
04:43.09dijib-bash: man: command not found
04:44.19dijib-d debug, -g dump core if crash and.... u>...?
04:44.41dijibthere isnt a u in the online man page but i assume is unnatend?
04:44.45dijibits
04:45.00dijibam i flooding?
04:46.22p3nguinI didn't say to use an option u.
04:46.22dijibuser
04:46.27p3nguinbecause there isn't one.
04:47.01dijibsaying g requires an argument. .. dump path?
04:48.03p3nguinMust be a new feature; mine does not require any path.
04:49.29dijibin lower case it doesnt your right
04:50.31dijibwhats L in in ps -L
04:51.50dijibUnable to access the running directory (Permission denied).  Changing to '/' for compatibility.
04:51.57dijibwhat dir's do i have to chmod? i
04:52.04dijibi figured it was a permission issue
04:52.15dijibanything *
04:52.35dijib/var/spool/asterisk /var/lib/asterisk
04:52.43dijibwhat else am i missing?
04:55.11dijibnevermidn let me figure this out
04:55.14p3nguinchown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk
04:55.18p3nguinoops
04:55.25p3nguinI retract that statement.
04:55.35p3nguinYou'll figure it out.
04:58.45dijibk rebooting to see
05:08.13dijibim still getting that permisson error
05:08.20dijibafter those chown
05:09.53*** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com)
05:11.50p3nguinI'm guessing that means asterisk is trying to access your current working directory, and it can't.
05:12.06p3nguinI'm not sure why 1.8 does that -- 1.4 has never done that to me.
05:12.21p3nguin1.8 always seems to, though.
05:12.50p3nguinBut that message has never prevented asterisk from running for me.
05:19.28*** join/#asterisk nix8n82-phone (~hmg@75-174-132-115.chyn.qwest.net)
05:20.30dijibstill not working
05:20.59dijibit looks like my dialplan has been loaded cuz i try my extensions and it tries to call and if i use a non defined extension i get an error
05:21.07dijibmy voicemail immediatly hangs up though
05:21.08p3nguinWhat happens when you start it with asterisk -G asterisk -U asterisk -vvvvddddddddg  ?
05:21.13dijibit runs
05:21.18p3nguinPerfect.
05:21.37dijibits running right now though, it just doesnt work
05:21.56p3nguinI don't know what "doesnt work" means.  You have to be very specific.
05:21.59*** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6)
05:22.09dijibi can register sip clients
05:22.18p3nguinThat's good.
05:22.28dijibbut all i get is dead air
05:23.07p3nguinIf you pick up a phone and call an extension which dials another phone, you hear nothing as the other phone rings?
05:23.25dijibphone doesnt even ring
05:23.28dijibi just hear nothing
05:23.54p3nguinSo nothing happens at all.  Show me the extension you're calling.
05:23.56*** join/#asterisk Syrex (~syrex@dsl-146-17-198.telkomadsl.co.za)
05:24.16*** join/#asterisk joako (~joako@opensuse/member/joak0)
05:24.36dijibok yes it rings
05:24.58p3nguinThe target phone rings, but you hear nothing while you wait.  Is that right?
05:25.04dijibyes
05:25.05joakoI just want to say Asterisk works well for me. 2231343 calls processed System uptime: 4 weeks, 1 day, 10 hours, 59 minutes, 48 seconds
05:25.15dijibnot while i wait, call established... i hear nothing
05:25.18p3nguinWhat kind of phone are you using to call from?
05:25.34dijibsoftware to software
05:25.40p3nguinWhile you were waiting on someone to answer, was there a ringing sound on your side?
05:25.57dijibhow do you have 2million calls?
05:26.18p3nguinlots of calls per hour
05:26.28dijibwhat are you a callcenter for?
05:27.02p3nguinWhile you were waiting on someone to answer, was there a ringing sound on your side?
05:27.25dijib3156 calls per hour
05:29.54joakoI use alarm panels and the AlarmReceiver app as a sort of telematics system
05:37.16*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
05:45.29*** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6)
05:50.52dijib4 S asterisk  1308  1304 19  80   0 -  7668 -      14752   0 01:41 ?        00:00:55 /usr/sbin/asterisk -f -U asterisk -vvvvg -c
05:51.01dijiband i still cant asterisk -r
05:51.05dijibhowcome?
05:51.11p3nguinDid you run it like I said?
05:51.46dijibim running it how safe_asterisk runs it on boot
05:52.33p3nguinWhat is your astrundir set to in asterisk.conf?
05:53.14dijib/var/run/asterisk
05:53.38p3nguinWhat files are in that directory?
05:54.04dijibasterisk.pid
05:56.16p3nguinIf you run asterisk the way I said, do you still only have that one file in /var/run/asterisk?
05:58.46dijibno i get an additional asterisk.ctl
05:59.59p3nguinGood.  So asterisk is not broken.
06:00.12p3nguinIf the ctl exists, you should be able to connect with asterisk -r
06:00.48dijibi can but not running through safe_asterisk
06:02.00p3nguinI'd like to see the output of ls -dl /var/run/asterisk
06:02.25dijibrunning how?
06:02.46dijib770
06:02.49dijibis the permission
06:02.52p3nguinJust the output of that command.
06:03.17dijibowen and asterisk:asterisk
06:03.20dijibis owener group
06:03.36p3nguinI just want to see the output of the command.
06:03.47dijibdrwxrwx---. 2 asterisk asterisk 4096 Sep 10 01:53 /var/run/asterisk/
06:03.56dijib770 500:500
06:04.04p3nguinI realize it's hard, but I knew you could do it!
06:04.27dijibim telling you what it said with 770 500:500
06:04.38p3nguinI just asked for the output.
06:04.46dijibits the same things bro
06:05.01p3nguinIf you don't want to cooperate, that's fine.  Good luck!
06:08.22dijibare you normally this short with people?
06:08.46carrarStep 1) FIX
06:08.48p3nguinShort?  I asked you THREE TIMES before you gave me the output.
06:08.49carrarStep 2) IT
06:08.53carrarStep 3) FIX IT
06:09.03p3nguinStep 4) Profit?
06:09.16carrarwell he can't get to setp 1
06:09.17carrarstep 1
06:09.25dijib770 500:500 was the output.. it just wasnt in a format you were willing to accept
06:09.28p3nguinoh
06:09.30p3nguinGood point.
06:09.37carrarWho the hell says "770 500:500"
06:09.41carrarhahah
06:09.46p3nguinThere is absolutely no possible way that command outputted that data.
06:09.49p3nguinNone.
06:09.51p3nguinAt all.
06:10.11dijibthen whats the drwxrwx--- mean then?
06:10.18dijib=770
06:10.23carrarSMRT
06:10.28p3nguinBut that's fine, you aren't required to accept my help.
06:10.36p3nguinI'm also not required to provide it.
06:10.46dijibalright then man.
06:10.47carrar0770
06:10.58dijibwhy 0?
06:11.32p3nguinman ls?
06:12.32p3nguinNope, I was mistaken.  It isn't in there.
06:12.37carrarMaybe you should run Asterisk from rpm
06:12.45carrarman, I can't believe I just said that
06:13.09p3nguinHis asterisk seems to run fine the way I told him to run it, but he wants to use safe_asterisk instead.
06:13.36carrarsafe_asterisk should work great on CentOS
06:13.46p3nguinHis must be a special case.
06:14.28dijibsafe_asterisk monitors if asterisk is running. for some reason i cant -r into it and the system doesnt work ok.
06:14.36dijibdont worry about it, im too much trouble
06:15.54p3nguinIf I used safe_asterisk, I would be more likely to understand why it doesn't create the socket... but I don't, so I don't.
06:16.29p3nguinI should look into it, though.  It is apparently a liked way to run asterisk.
06:18.17dijibi think i found my error
06:18.46dijibnope nevermind
06:26.50ChannelZPEBKAC?
06:27.23p3nguinIt looks like safe_asterisk should be run as "safe_asterisk -U asterisk -G asterisk" at least.
06:28.35p3nguinOr change ASTARGS in the script itself.
06:31.16p3nguinIt's not a bad looking script, but I don't know that it will give me any real benefit over how I currently run asterisk.
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06:41.10dijibi changed ASTARGS with -G asterisk -U asterisk still with same result. no .ctl
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06:43.27kvad12Hey guys anyone have any exp running asterisk on XEN? If so how does it run any issues. Looking to migrate cluster.
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08:20.45justdaveanyone know if the kernel modules in the asterisk yum repo are ever going to get recompiled against a recent kernel?
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08:42.42devil_evoxxxhi all
08:44.12devil_evoxxxa question, in other country ( no italy ) operator that provide E1/T1 access is able to permit incoming calls from different district (for example a E1 for local prefix 06 and another E1 for local prefix 05)
08:52.13tzafrirdevil_evoxxx, and the question is?
08:56.54devil_evoxxxwe are a wisp in italy and we have a TDM stream into our farm. We splice a 1gbit ethernet and a E1 stream. But for incoming calls on E1 the only prefix is the local prefix where we are
08:57.05*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
08:57.11devil_evoxxxand wee need more local prefix for neighbor city for offer voip services
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10:49.34TomTomhi there, i have a beginners problem. i want that asterisk does react on capi/isdn calls, but it says "no extension found". i've made a rather simple setup. any idea what may be still wrong? config -> http://nopaste.info/d23861ce00.html
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11:30.03pahi, question: i updated asterisk some time ago, now it is 1.8.6 (on ubuntu natty)
11:30.17pai also installed the pkg asterisk-mp3 to be able to play mp3s
11:30.43pabut i am not sure where to place my files: if i try to reach my old working extension now, i get:
11:31.07pa<PROTECTED>
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11:53.53wdoekes2pa: /var/lib/asterisk/sounds[/en] ?
11:55.44paah yes
11:55.46paright
11:55.54pabut then the problem was that it doesnt read the mp3
11:55.55pai think
11:56.01pai converted to gsm and it worked
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12:21.48mtbfI'm trying to troubleshoot some issue with app_rxfax on some old box with 1.2.0, while analysing logs I found out, since some day the only message in the log related to the app_rxfax is : app_rxfax.c: Got hangup , no configuration files were modified since that time, anyone saw simmilar problem?
12:44.38*** part/#asterisk StaRetji (~BigAll@80.93.240.171)
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13:10.59eduzimrshi, im having problem with my sip realtime cache... anyone can help?
13:11.07eduzimrsusing mysql
13:12.57eduzimrswhen my mysql database is down, all my sip clients get disconnect from *
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13:16.49*** mode/#asterisk [+o blitzrage] by ChanServ
13:20.00dwayneeduzimrs, just a guess, but it is probably not a good idea to bring your database down while using realtime
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13:22.43eduzimrsok, but im doing tests, it should be in cache no? when rtcachefriends=yes is set.
13:24.18eduzimrsdwayne: my sip clients shouldnt get disconnect from *
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13:50.46krish|wired-inguys, does this work - https://issues.asterisk.org/jira/browse/ASTERISK-3435
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13:51.58krish|wired-inhow can I get the uptime in unix format
14:01.52krish|wired-inguys, how to query dialplan variables
14:01.56krish|wired-inlike ${EPOCH}
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14:05.19_omeris there any specific room for a2billing ?
14:08.24_omerhttp://212.108.128.50/   <---- how to add such kind of POPUP window security on A2Billing Admin Folder ....
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14:23.25_omeranyone ???
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14:52.16simplydrewdoes anyone have experience with working with chan_sccp? I have it installed, but am having some issues
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15:35.20k3asd`hi
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15:55.24_omerhttp://212.108.128.50/   <---- how to add such kind of POPUP window security on A2Billing Admin Folder ....
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15:56.03WIMPy_omer: Connection refused
15:56.45WIMPyFor popups I use System().
16:02.25mtbfI'm trying to troubleshoot some issue with app_rxfax on some old box with 1.2.0, while analysing logs I found out, since some day the only message in the log related to the app_rxfax is : app_rxfax.c: Got hangup , no configuration files were modified since that time, anyone saw simmilar problem?
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16:47.02p3nguinsimplydrew: What's wrong with it today?
16:47.21HenriqueAtilaHello :)
16:47.53simplydrewp3nguin: I've checked and rechecked everything, but when I actually use the command you recommended of "module load chan_sccp.so" I get an error of "WARNING[771]: loader.c:446 load_dynamic_module: Error loading module 'chan_sccp.so': /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: manager_event"
16:48.26pabelangerwhat version of Asterisk?
16:48.34p3nguinOkay, so that's still the same problem as before.  Did you try using either of the other two branches instead of the stable branch?
16:48.59simplydrewp3nguin: I wasn't sure how exactly to go about that, but I remembered you saying something about it
16:48.59p3nguinI suggested you use stable branch 3.0, but there is dev branch 3.1 and dev branch 4.0.
16:49.11simplydrewp3nguin: how would I go to the dev branch of 4.0?
16:49.49simplydrewpabelanger: 1.6.0.26
16:50.17p3nguinsvn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/branches/V3.1/ chan-sccp-b_V3.1
16:50.36p3nguinand
16:50.42p3nguinsvn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/trunk chan-sccp-b_V4.0
16:50.51p3nguinfor those two branches
16:50.59pabelangermy bad, I just noticed it was chan_sccp
16:51.06HenriqueAtilaWhat distro is better for run asterisk?
16:51.20p3nguinhenriqueatila: Spin the wheel.
16:51.26simplydrewokay, I'll give 4.0 a try. should I go back and remove the previous version?
16:52.01p3nguinsimplydrew: You only need to remove it if you want to.  When you install the new one, it will overwrite the existing files.
16:52.13simplydrewp3nguin: gotcha, okay. just wanted to confirm
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16:55.18simplydrewp3nguin: after doing ./configure, make clean, I did a make install and get these errors: http://pastebin.com/Z7aWLtkY
16:57.09p3nguinI guess I need to install some asterisk testing instances so I can see what works where for me.
16:57.32p3nguinI use Asterisk 1.4 branch and the stable 3.0 branch doesn't give me any problems.
16:57.53simplydrewhmm
17:06.16pabelangersimplydrew: you are missing the Asterisk header files
17:06.53simplydrewpabelanger: how would I go about resolving that?
17:07.14pabelangerdownload asterisk
17:09.55simplydrewpabelanger: okay, I assume i need to go up to 1.8 since I'm currently on 1.6?
17:10.11simplydrewwasn't sure of the best safe way to do that without disturbing anything
17:10.39pabelangerhow did you install asterisk?
17:11.11simplydrewwell, this is a trixbox install, so it technically did it
17:11.50pabelangerso, you need to figure out how trixbox installs the header files, or get the source code for that version of Asterisk installed
17:13.38simplydrewthose would be located in /usr/include/asterisk, correct? it appears that a lot are in there. not sure what I'm looking at in there though. lots of .h files
17:13.52pabelangeryes, .h are header files
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17:52.44carrarY*A*W*N
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17:54.41AshuttoHello
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17:58.31Ashuttoi'm behind several nat levels i cannot control, my only "public" interface free from nat&firewalling is IPv6 only. Is it possible to have a working asterisk in this conditions?
17:59.40WIMPySure. But using SIP to the outside will at least be tricky.
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18:11.28p3nguinI'd still like to know why when I get voice mail, I get msg0000 in the voicemail directory, but msg0001 is the file that is emailed.  There is always an offset of 1 between the message files in asterisk and the message file that is emailed.
18:16.02ChannelZit's wrong?
18:16.17p3nguinUh yeah it's wrong.
18:16.38samandirielbut it feels so right...!
18:16.40ChannelZI mean it's actually mailing you the wrong file, it's not just the name in the email?
18:17.03p3nguinThe content of the sound file is right, but the name of the file is wrong.
18:18.02p3nguinIf I get a voicemail, it will be recorded as msg0000.wav, for example.  When completed, it will be emailed to me as msg0001.wav rather than the correct file name.
18:18.53p3nguinIt has been this way through several versions.  I've asked about it multiple times, but no one has ever been able to tell me how to fix it.
18:19.13p3nguinI'll patch it locally if someone can tell me where the problem lies.
18:19.33ChannelZI've never had that.
18:19.48p3nguinI'm looking at app_voicemail.c right now hoping something pops out at me.
18:19.56ChannelZI'd have to look at the source, I wonder if you have an old .txt file lying around causing it to get confused
18:20.01samandirielwhy does it matter, so long as it's consistent?
18:20.10p3nguinbecause it's wrong.
18:20.14ChannelZsamandiriel: don't even ask
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18:20.26ChannelZAre you using database storage?
18:20.39p3nguinno
18:23.31p3nguinI guess I need to check others' mail boxes before I make the assumption that it is broken globally.
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18:28.41p3nguinOkay, I found a user with no mails in the INBOX.  I called and left a message.  I see msg0000.WAV and msg0000.txt in the INBOX.
18:29.07p3nguinNow to determine what file name is emailed.
18:30.27ChannelZI can't even trick mine on purpose so far.
18:33.04anonymouz666using Local/ channels as a realtime queue members is weird
18:33.16anonymouz666it does not enter in the context you write in Local/
18:34.19p3nguinIt works fine from the conf; I've never messed with realtime.
18:35.15anonymouz666indeed, it works fine from the conf.
18:37.10ChannelZWell now I've screwed up my whole mailbox
18:37.17p3nguin:(
18:37.41ChannelZwell in so much as I deleted all my messages but there are still files there
18:37.58ChannelZI confused it by making empty .txt files
18:38.16ChannelZthere now it's right
18:39.24ChannelZYou mentioned on the test box it recorded msg0000.WAV but before you said the attachments were named .wav
18:39.51p3nguinI used msg0000.wav as the example to describe the problem.  I'm using WAV only.
18:40.56p3nguinI seem to remember a time where I actually used wav and WAV, and both files would be emailed with a name offset of 1.
18:44.43ChannelZdunno.  If your INBOX is completely empty of files and you do a test and it's offset, I have no idea what is happening
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18:47.29p3nguinI'm still waiting to find out what file was emailed.  msg0000.txt and msg0000.WAV are in the INBOX, but I bet msg0001.WAV is what shows up in email, since that is what happens to mine every time, regardless of the msg index number.
18:48.34p3nguinI don't even remember if there has ever been a version I used that didn't do this.
18:49.01p3nguinWhen I first noticed it, I rolled back a few asterisk versions and it was still doing it, so I assumed it was doing it all along and I just didn't pay attention.
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19:11.54rotten777anyone awake?
19:13.50WIMPy*zzZzZZ*
19:14.01rotten777darn :-X
19:15.45rotten777i'm just a beginner with some stupid questions... what harm would it do to be awake? ;)
19:16.37rotten777i have a DID at a voip provider, a single sip phone (polycom ip 330), and a single debian based server. trying to get dialing working and evidently i'm not very smart
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19:20.00WIMPyThat was not a question.
19:20.31WIMPyBut I've got one: Why do you want to add a server between your ITSP and your phone?
19:20.46rotten777I don't but I can't get broadvoice to work with the polycom ip 330
19:21.02rotten777i'm brand spanking new to voip but not to linux or networking by any means
19:21.18rotten777so i'm probably just reading the conf instructions wrong in the polycom gui
19:23.03p3nguinNew message 1 in mailbox.  Attached is msg0001.WAV.  So it's not just my box.
19:34.55ChannelZis your email command doing something insane?
19:35.42p3nguinmailcmd = /usr/bin/msmtp -t
19:35.52ChannelZhm.
19:35.53ChannelZshrugs
19:36.02p3nguinIt's beyond me.
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19:50.20rotten777is there a way if my polycom or softphone are even registering on my asterisk box?
19:50.33p3nguinIs there a way for it to do what?
19:50.54rotten777is there a way to check the active sip devices on asterisk
19:51.05p3nguinsip show peers
19:52.20rotten777hmm ok they both show up
19:52.37rotten777i can't dial either extension though... i hate being the noob
19:52.56p3nguinWhat context did you assign for the phones?
19:53.36rotten777[testing]
19:53.36rotten777exten=>1001,1,Dial(SIP/byrdits)
19:53.37rotten777exten=>1002,1,Dial(SIP/softphone)
19:53.46rotten777is that what you're asking?
19:54.07p3nguinIn sip.conf, you assigned a context for each phone.  What context did you assign?
19:54.20rotten777testing
19:54.24p3nguincontext= something
19:54.55p3nguinIn sip show peers, does byrdits show an IP address?
19:55.02rotten777no
19:55.08p3nguinThen it isn't registered.
19:55.10p3nguinAnd you cannot call it.
19:55.37rotten777Name/username              Host            Dyn Nat ACL Port     Status
19:55.37rotten777byrdits/byrdits            (Unspecified)    D          5060     Unmonitored
19:55.37rotten777softphone/softphone        192.168.77.2     D          5060     Unmonitored
19:55.37rotten7772 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
19:55.46rotten777so it isn't seeing my hardware phone...
19:56.22p3nguinbyrdits isn't registered.  Can you use byrdits to call softphone by extension 1002?
19:56.30rotten777i can't call either from either
19:56.40rotten777do i need to set the outbound proxy on the polycom to the ip of the asterisk box?  this is all on a private subnet btw.. no nat or anything
19:57.15p3nguinYou have to set the server address, but I don't know if you have to specify it again in the outbound proxy field.  I don't think you do.
19:57.39rotten777ok under lines i have line 1 with the address user id and password completed
19:58.19rotten777ahh i didn't have it in server 1 or server 2... let me try server 1 with the ip
19:58.25ChannelZcan't figure out why he keeps getting "Correct auth, but based on stale nonce" from Zoiper lately
19:59.45rotten777[Sep 10 15:59:11] NOTICE[25588]: chan_sip.c:21763 handle_request_register: Registration from '<sip:192.168.77.21@192.168.77.21>' failed for '192.168.77.22' - No matching peer found
19:59.51*** part/#asterisk didnot (~didnot@unaffiliated/didnot)
19:59.53ChannelZta-da!
20:00.05rotten777what is that?
20:00.19p3nguinDid you specify your sip user name in two places?
20:00.31p3nguinaddress and auth name
20:00.54rotten777address was the ip i thought
20:01.00rotten777but i am wrong
20:01.01rotten777haha
20:01.09p3nguinI'm pretty sure address is your sip name.
20:01.20rotten777ok putting in the username in address and auth name as the username as well
20:01.31rotten777yeah i'm a complete noob to this. thanks for the help btw
20:02.28rotten777Name/username              Host            Dyn Nat ACL Port     Status
20:02.28rotten777byrdits/byrdits            192.168.77.22    D          5060     Unmonitored
20:02.28rotten777softphone/softphone        192.168.77.2     D          5060     Unmonitored
20:02.28rotten7772 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
20:02.36p3nguinThat's good!
20:02.56rotten777ok when dialing from the hardware phone to the softphone i get circuit busy tone
20:03.17p3nguinand the other way?
20:03.35rotten777no ring or circuit busy or anything dead air
20:03.59p3nguinAre those the correct addresses?
20:04.04ChannelZprobably dialplan hasn't even sent the dial
20:04.05p3nguinfor the phone
20:04.07rotten777oh snap wait
20:04.08p3nguins
20:04.09rotten777found it
20:04.38rotten777sweet! it works
20:04.45rotten777both ways
20:04.52p3nguinWhat was the problem?
20:04.58ChannelZcongrats - now you're bi!
20:05.19rotten777softphone wasn't placing call it was just synth tone & not actually talking to asterisk
20:05.47rotten777well crap
20:05.52rotten777now get circuit busy again
20:06.34rotten777[Sep 10 16:05:36] WARNING[25633]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
20:06.50p3nguinThe device seems to be gone.
20:08.42rotten777ok now that's reliable
20:08.47rotten777not sure why it failed the once
20:09.02rotten777now if i can figure out how to use my broadvoice account to dial to the ptsn
20:09.06p3nguinnetwork issues?
20:09.30rotten777i guess... the asterisk box, my softphone and my hardphone are all in the same gigabit switch
20:09.36rotten777nothing layer 3 in between them
20:09.48p3nguinThat's easy.  Start by rethinking context hierarchy.
20:10.07rotten777any pointers?
20:10.10p3nguinCreate a new sip peer for the ITSP.  Create a new context for outgoing calls via that peer.
20:10.10rotten777or places to read up
20:10.12*** join/#asterisk Fritz09 (~Adium@pop1-2174.catv.wtnet.de)
20:10.17p3nguin~book
20:10.17infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook
20:10.27p3nguinThere's a huge book with all the details.
20:10.42p3nguinOr I can share some examples with you.
20:10.53rotten777yeah examples sound good
20:11.48rotten777i basically have the 1 broadvoice did via sip and 1 server and 1 hardware phone
20:11.49p3nguinexample sip.conf:  http://pastebin.com/tER2jGnY
20:11.59rotten777k reading now
20:14.23p3nguinexample extensions.conf:  http://pastebin.com/Piqv4Egj
20:15.00rotten777ok to get the asterisk server to talk to itsp over nat do i need to specify the external ip?
20:15.08p3nguinIn these examples, voipms is the peer I'm using for outbound calls.
20:15.12p3nguin~sipnat
20:15.12infobot[~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet.
20:16.00p3nguinWhich asterisk version are you using?
20:16.32rotten777latest with ubuntu server
20:16.36rotten777let me check
20:16.39p3nguincore show version
20:16.50rotten7771.6.2.9
20:17.12WIMPyThat is not very new.
20:18.27p3nguin15 months old
20:18.42p3nguinLots of bug fixes in 15 months.
20:19.10rotten777its ubuntu natty
20:19.13rotten777they're that far off eh?
20:19.48p3nguinI don't know when Ubuntu Nasty was released.
20:23.39rotten777alright i'm kind of confused...
20:23.45rotten777i'm reading the dialplan basics
20:23.54rotten777i have the broadvoice peer loaded
20:24.03*** join/#asterisk dlisenby (4bb63d33@gateway/web/freenode/ip.75.182.61.51)
20:24.23p3nguinDid you use a register statement for broadvoice?
20:25.01rotten777[broadvoice]
20:25.02rotten777type=peer
20:25.02rotten777disallow=all
20:25.02rotten777allow=ulaw
20:25.02rotten777contet=from-broadvoice
20:25.02rotten777dtmfmode=rfc2833
20:25.04rotten777host=sip.broadvoice.com
20:25.07dlisenbyp3nguin, thanks for your help the other night with Callcentric.  I went out and acquired a sip trunk with Nextiva.
20:25.08rotten777nat=yes
20:25.09p3nguinDon't flood us.
20:25.13p3nguin~pb
20:25.13infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
20:25.48p3nguinBroadvoice is not behind nat, so nat=yes is wrong.
20:26.27rotten777ok the phone is do i need to append that?
20:27.09p3nguinIf they are on the same network segment as asterisk, you can probably omit the nat line in the phone entries.
20:27.20rotten777gotcha so they should be fine
20:27.33p3nguinBut you need to finish configuring your broadvoice peer entry.
20:27.53p3nguinYou didn't specify your username or password for them, so you won't be able to authenticate when sending calls.
20:28.02p3nguinSee my example.
20:28.49p3nguinAnd you need to send a registration unless you are doing IP auth, which I doubt you are.
20:30.30rotten777looking at the register statement in that example.. what is the data prior to the dns entry of the sip server
20:30.39rotten777register => 105245:2FsuonGrOuq4r@chicago.voip.ms
20:31.01p3nguinregister => username:password@host
20:31.53rotten777ok so i have to fille the username variable, the secret variable and the register variable?
20:32.11p3nguinWhere did variables come into play?
20:32.43rotten777gah yeah i'm reading it as a script or something i mean the statements for username secret and register
20:32.46p3nguinIf you are not doing IP auth, you have to register (by using a register statement) to be able to receive calls.
20:33.10p3nguinAnd several providers require that you are registered before you can send calls.
20:33.20rotten777they hold redundant data and i wanted to make sure that i wasn't adding username and secret statements in there if it was only needed in register
20:33.25rotten777gotcha
20:33.54p3nguinThe peer entry below is used to match the calls coming in as well as "route" calls going out.
20:34.15p3nguinusername/password will be required in it for sending calls out, because they will want authentication.
20:35.36rotten777ok so i need the 2 peers in sip, 1 for the itsp, 1 for the phone... i think i'm good. let me see if the peers show up in asterisk
20:35.56p3nguinAnd the register statement in the general section
20:36.10p3nguinYou can see if you are registered to the ITSP with sip show registry.
20:36.27rotten7770 sip regs
20:37.34p3nguinThe register statement has to be in the [general] section of sip.conf, before any peer entries, and before [authentication] if it exists.
20:38.20rotten777ok it is showing in the regs now
20:38.33p3nguinYou forgot to run sip reload?
20:38.40rotten777i restarted the asterisk service
20:38.49p3nguinThat's a lot of work just to reload sip.
20:38.52p3nguinsip reload
20:38.55rotten777haha ok
20:39.10p3nguinAnd when you changed extensions.conf, use dialplan reload.
20:39.21rotten777ok
20:39.24p3nguins/changed/change/
20:40.04p3nguinSo you've registered to the ITSP, and you're ready to look at the dial plan?
20:40.09rotten777yup!
20:40.32p3nguinHave you created your [from-broadvoice] context in extensions.conf?
20:40.54rotten777doing that now
20:41.37*** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106)
20:42.00rotten777is that the DID# on the exten statement on the from-ipkall ?
20:42.31p3nguinYes.  My ITSPs send calls to my phone number, so the extension is the DID number.
20:43.24rotten777gotcha. the "30"  behind the sip address. what does this signify
20:43.39p3nguinSIP/jack,30?
20:43.48rotten777yes
20:43.51p3nguin30 second timeout for dialing a phone named "jack"
20:44.11p3nguinAfter 30 seconds, it'll progress to the next line if there is one.
20:44.42rotten777ok
20:45.32*** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net)
20:45.32rotten777so 4 exten statements
20:45.52p3nguinwhere?
20:45.59rotten777from-broadvoice
20:46.18p3nguinYou probably just need one extension, but that one extension will have several priorities.
20:46.28rotten777i see some statements with verbose playback congestion playtones... are these needed?
20:47.17p3nguinI have no idea what I was doing there.
20:47.27rotten777lol k
20:47.30p3nguinBut now I have to go look at my actual dial plan to see if I am doing that in there too.
20:48.25p3nguinOh, I see what I was doing.
20:48.51p3nguinI was trying out both ways of providing congestion tones and forgot to decide on one when I pasted that.
20:49.02rotten777gotcha
20:49.16p3nguinSometimes I do silly things like that.
20:49.19rotten777ok i have the from-broadvoice finished
20:49.33rotten777hey at least you can understand what you're looking at
20:50.35rotten777isn-outbound is the context for the outgoing to the itsp
20:50.36rotten777?
20:50.54p3nguinIf you didn't figure it out already, I am defining any used phone number explicitly and then any number I am not defining will end up on the pattern _X. to play a message saying the number is not in service.
20:50.59p3nguin~isn
20:50.59infobotmethinks isn is ITAD Subscriber Number (see 'itad'). An ISN is a method of dialing SIP URI's via a standard keypad on a telephone. Because of the alphanumeric nature of SIP URIs, it is difficult to dial them via the keypad on your phone. The use of ISN numbers simplifies this by utilizing DNS lookups to map the ISN number to a domain. See http://www.freenum.org for more information.
20:51.24rotten777ah
20:51.30p3nguinYou don't need ISN for broadvoice.
20:52.10rotten777ahh voipms-outbound?
20:52.20p3nguinThat's where I dial out to the PSTN.
20:52.31rotten777i think i'm able to see the logic in the statements a lot better now
20:52.56rotten777so you have an optional 1 at the beginning of the number... it adds the local area code if needed
20:53.03p3nguinI'm supporting 11-digit, 10-digit, 7-digit in the local area code, plus international dialing.
20:54.15rotten777wow asterisk is a lot more powerful than i thought
20:54.20p3nguinSo if you dial 1+area code+ 7 digits, it matches.  If you don't dial the 1 but just the area code and 7 digits, it matches.  If you just dial 7 digits, it dials those 7 digits in your local area code that you write into that line.
20:54.39rotten777my plan with broadvoice is unlimited in the state of FL
20:54.56rotten777i'll have to find a list of FL area codes and put it in here... anything outside of florida i'll use my cell
20:55.44rotten777hmm ok dialplan reloaded
20:55.50rotten777i have the broadvoice-outbound in there
20:55.59rotten777i don't get anything when i dial
20:56.02p3nguinhttp://www.50states.com/areacodes/florida.htm
20:56.12p3nguin16 of them
20:57.18rotten777cool
20:57.23p3nguinYou probably didn't configure the rest of the context hierarchy well enough to match the extensions in your broadvoice-outbound context.
20:57.38p3nguinI prefer to assign a context 'phones' to any phone.
20:58.02p3nguinThen in the phones context in extensions.conf, I use includes to specify what those phones can do.
20:58.30p3nguinSee line 157-161 in my extensions.conf example.
20:59.02rotten777ahh my internal is a different name and i have the outbound now in there
20:59.41p3nguinYou have to build a good hierarchy so that no incoming calls ever have the ability to dial back outbound.
21:00.08p3nguinOtherwise, you run the risk of someone running out your prepaid balance or running up a huge bill if you pay for calls you already made.
21:00.40rotten777oh nice haha
21:00.52rotten777ok i've got that stuff done and reloaded the dialplan
21:00.57p3nguinIn most cases, phones will never have a reason to call your own DIDs, so they don't have a reason to be able to call your inbound context.
21:01.09rotten777yeah
21:01.29p3nguinIn cases where you do have to call your own numbers, we re-think the way to accomplish it.
21:02.07p3nguinOkay, so includes are in place and calling a phone number does what?
21:02.13rotten777dead air
21:02.19rotten777i did show channels
21:02.23p3nguincore set verbose 4
21:02.28rotten777k
21:02.33p3nguincall again.
21:02.57rotten777> doing dnsmgr_lookup for 'sip.broadvoice.com'
21:03.03rotten777sticks there
21:03.17rotten777broadvoice does say to use a proxy... i'm assuming i should have added that
21:03.47rotten777outbound proxy server
21:03.51p3nguinThey didn't offer you a configuration sample so you'd know how to use their service?
21:04.18rotten777checking right now
21:04.40p3nguinIf you didn't see any call progression appear in the console when verbose was turned up, the next thing to do is check the sip debug to see why there is no call starting.  sip set debug on
21:11.18rotten777they have an asterisk conf example of sorts on the site
21:12.03p3nguinIs it where I can see it?
21:12.33rotten777http://pastebin.com/SinFTbmE
21:14.52rotten777http://www.broadvoice.com/support_install_asterisk.html
21:15.22p3nguinSo they want you to register using a proxy that is the same as the host.  How weird.
21:15.42p3nguinproxy sip.broadvoice.com and host sip.broadvoice.com
21:15.49p3nguinDoes not make sense to me.
21:15.53rotten777yeah i have the feeling the service might suck but it was a quick google... nobody i know has any idea about voip
21:16.03p3nguin~itsp-us
21:16.15p3nguin~itsplist-us
21:16.15infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
21:16.27rotten777yeah now the sip registration is failing too with their instructions
21:16.32rotten777ugh
21:16.59p3nguinI'd put it back like it was and then check the sip debug to see what is going wrong.
21:17.44p3nguinThey may require you to use fromdomain and fromuser in the peer entry.
21:17.55rotten777yeah i have those there
21:18.11rotten777fromdomain is sip.broadvoice.com fromuser is my #
21:18.24rotten777but username is also my #
21:19.39p3nguinsip set debug on
21:19.41p3nguinmake a call.
21:20.02rotten777[Sep 10 17:19:47] NOTICE[26257]: chan_sip.c:11696 sip_reg_timeout:    -- Registration for '<8636584102>@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again (Attempt #19)
21:20.02rotten777<PROTECTED>
21:20.03p3nguinWhen it fails, hang up.  Copy everything and paste in the pastebin.
21:20.07rotten777ahh ok
21:20.36p3nguinFix the register statement first.
21:20.51p3nguinThey'll most likely want you to be registered in order to send calls.
21:22.52rotten777[Sep 10 17:22:32] NOTICE[26257]: chan_sip.c:18392 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
21:23.02rotten777it seems like the hosts file addition they asked for killed the sip registration success
21:23.22p3nguinI have no idea why they would have asked you to do that.
21:23.37p3nguinBut that message does not say it failed.
21:23.47p3nguinIt says the expiration is in 30 seconds.
21:24.02rotten777ok i tried to make a call and get nothing
21:24.18p3nguinIf you have sip debug enabled, you should see something.
21:24.23rotten777request sent is the state it is in
21:24.26rotten777yeah i see  a ton of stuff
21:24.32rotten777i still have the verbose to 4
21:25.44rotten777http://pastebin.com/ua1gUu3t
21:27.02p3nguinIt says it is transmitting nat to a public IP address.  Did you remember to put nat=no in your broadvoice peer?
21:27.35rotten777i just added nat=no
21:27.37rotten777again
21:27.44rotten777reload and looks like it quieted down
21:28.05rotten777now the same thing
21:28.05p3nguinIt's doing a lot of REGISTER stuff.  I'd undo whatever I did to make it not register.
21:28.19p3nguinYou had it registering before, so revert back.
21:29.21p3nguinI've never seen an ITSP that knows how to configure asterisk for the end user.  It's kind of ridiculous.
21:29.30ChannelZIf it ain't broke, try harderf
21:29.46rotten777i dont know what else i can change back
21:29.52rotten777i thought it was back how we had it
21:30.18p3nguinYou said something about /etc/hosts.  Did you undo the changes there?  You changed the register statment.  Did you undo that, too?
21:30.27rotten777yes i removed the hosts entry
21:30.38*** part/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net)
21:30.42*** join/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net)
21:30.48rotten777whoops
21:30.49rotten777lol
21:31.49rotten777does the register statement whitespace matter?
21:32.17p3nguinThe only whitespace should be around the =>
21:32.21*** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net)
21:32.27rotten777yeah thats where it is
21:32.31p3nguinregister => user:pass@host
21:32.44p3nguinor sometimes...
21:32.47p3nguinregister => user:pass@host/extension
21:33.15rotten777it all looks fine to me
21:33.28*** join/#asterisk rdegges (~rdegges@69.164.197.143)
21:33.50p3nguinAfter sip reload, sip show registry still says request sent?
21:34.13rotten777ahh registered now
21:34.17p3nguinIt usually only takes a few seconds for the registration to succeed and that status changes to registered.
21:34.35rotten777ok so it shows registered now
21:34.52p3nguinThat should quiet some of the REGISTER packets in the sip debug.  Enable sip debug and make a call.
21:36.42rotten777http://pastebin.com/tVaKNtDX
21:37.40p3nguinStill just a crapload of register stuff.
21:37.43p3nguinNo call at all.
21:38.48rotten777http://pastebin.com/uSc96J7t
21:38.53rotten777did i screw something up here
21:40.06p3nguinThat actually looks quite nice.
21:40.18p3nguinBut you're missing some parts of extensions.conf.
21:40.27rotten777that was just the bottom
21:40.31p3nguinDid you omit them for the paste, or did you leave them out?
21:40.37rotten777yeah just for the paste
21:40.41p3nguinokay
21:41.07p3nguinNow your phones' entries in sip.conf... they have context=phones?
21:41.26rotten777context=bits
21:41.31rotten777no wonder no calls are being placed...
21:41.31p3nguinThat's a problem.
21:42.39rotten777do they need multiple context statements or just the 1
21:43.04p3nguinYou can only have one.
21:43.14rotten777yeah they're set to context=phones
21:43.21rotten777dialplan and sip reload
21:43.53rotten777seems the same
21:44.36*** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106)
21:45.08rotten777http://pastebin.com/FG1H4PkU
21:45.19rotten777it is showing registered
21:45.48p3nguinStill just a bunch of register crap.
21:45.55p3nguinNo calls in that paste either.
21:46.47rotten777user=phone on the peer for broadvoice is fine?
21:47.19p3nguinYou don't need to set that explicitly, but if you are seeing that info in the debug it is fine.
21:48.44rotten777ok weird
21:48.50rotten777softphone works fine
21:49.00rotten777polycom doesn't
21:51.44rotten777I don't get that. they're on the same subnet calling into the same asterisk server calling to the same number
21:52.11p3nguinAre the peer entries for the two phones pretty much the same?
21:52.33rotten777they're exactly the same
21:52.36rotten777other than the username
21:56.34*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
22:01.18*** join/#asterisk felimwhiteley (~quassel@46.7.101.58)
22:01.18rotten777bam fixed
22:01.18rotten777p3nguin you're the man
22:01.43p3nguinWhat was causing the Polycom to fail?
22:02.11rotten777there's 2 pages of network/sip stuff in it's gui. i removed it all and re-added everything
22:02.15rotten777something was goofed up and not sure what
22:02.18rotten777but it was in the polycom
22:02.22rotten777not in asterisk
22:04.21rotten777now i have another problem though... outbound works great but i get the broadvoice voicemail when i dial in from the outside
22:04.42p3nguinsip set debug on
22:04.45p3nguinmake a call.
22:05.39rotten777http://pastebin.com/GdenCYHK
22:05.47rotten777i see the invite
22:05.48rotten777but
22:06.54p3nguinThey are sending to extension s rather than to your phone number.  Change your register statment to include /extension on the end of it.  The extension in the register needs to be your phone number.
22:07.05rotten777k
22:07.10p3nguinI don't know what they expect you to do in the case of more than one phone number.
22:07.15p3nguinI hate ITSPs that do that.
22:08.20rotten777hmm ok i have the /npanxx1234 on the extension
22:08.32rotten777on the register statement rather
22:08.50p3nguinAs long as it's your phone number and the extension is also your phone number, that should solve it.  sip reload, make a call.
22:09.21rdeggesHey all.
22:09.56rotten777http://pastebin.com/bp4694Dy
22:10.06rdeggesHave any of you got experience using MeetMe under high loads with DAHDI dummy?
22:10.27rdeggesI'm trying to find out if using dahdi dummy is a sufficient timing backend for high-volume meetme usage
22:10.56rdeggesI'm trying to support 100+ callers per meetme room, but having a really hard time doing that currently using dahdi dummy.
22:11.07rdeggesI'm not sure if it's dahdi dummy that's the problem, or my version of asterisk, or my hardware.
22:11.15rdeggesI'm using 1.6.2 atm.
22:11.24rdeggesAny advice or pointers I could use to help narrow it down a bit?
22:11.30*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
22:15.03*** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com)
22:19.10rotten777p3nguin: the inbound context from broadvoice has just the phone number.. the sip invite shows "phone#@ip:port".. does that need to be reflected in the context?
22:19.53p3nguinNo.  The extension can only be the phone number.
22:20.12p3nguinWell, it could be anything you make it, but it does not include the IP address or the port number.
22:20.30p3nguinI'm still waiting to see debug of a failed call.
22:22.28rotten777http://pastebin.com/s5xRMEK4
22:22.36rotten777thats straight to voicemail at the broadvoice switch
22:23.00p3nguinThat's not a call.
22:23.48p3nguinI don't understand what's going on.
22:24.29rotten777when i call from my cell phone to the did # i go straigh to voicemail
22:24.37rotten777it doesn't come to asterisk
22:24.41rotten777or to my polycom
22:26.15p3nguinI'm sure it's something going on with broadvoice, but I don't know what needs to be done to overcome it.  I'm glad I have an ITSP that cooperates.
22:30.44rotten777i disabled their voicemail and now get a busy tone when calling in
22:32.55rotten777now outgoing doesn't work
22:32.55rotten777http://pastebin.com/jpLKf7dL
22:50.08*** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com)
22:57.45p3nguinIt looks like it is finally at least trying to start a call.
22:58.04rotten777what was the tilde command you did earlier that showed itsp's
22:58.13p3nguinDid you forward the RTP port range in addition to the SIP port?
22:58.13rotten777i'm about done with broadvoice
22:58.17rotten777whats the rtp port range?
22:58.18p3nguin~itsplist-us
22:58.18infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
22:58.46p3nguinIt is normally UDP 10000-20000.  Check rtp.conf to verify.
22:59.54p3nguinOf the ones in that list, I use voip.ms and flowroute.
23:01.21rotten777yeah now i'm getting errors with the sip registry
23:02.07p3nguinMy primary is VoIP.ms, because it just works.
23:04.29p3nguinIn the several years I've been using them, I have had very few problems.  The biggest problem is typical of every service provider -- they don't know shit about how to do anything.
23:06.55rotten777yeah i'm pinging their servers and somehow i get less latency to new york than i do to miami... miami is about a 3 hour drive from here... amazing
23:07.07rotten777i wonder if i can get a refund for the crap services
23:07.13rotten777you would recommend voip.ms
23:07.14rotten777?
23:07.23p3nguinAbsolutely.
23:07.37p3nguinI buy all my DIDs from them right now.
23:08.53p3nguinI'm thinking about getting a DID from Flowroute pretty soon, though.  I use them for outbound calling from time to time, but don't have a phone number on them.
23:10.34*** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca)
23:10.50blitzragep3nguin: good to know -- I use Unlimitel exclusively right now, but with the 3 outages I've experienced lately, it looks like I'm going to be (at the very least), adding another provider to the mix. voip.ms is used by a company a buddy of mine works for in Vancouver, and I've had some good success with them in the little bit of trials I've done
23:11.09p3nguinI like 'em.
23:11.16p3nguinGood rates, decent quality.
23:11.36p3nguinI have both regular and toll-free DIDs with them.
23:11.55p3nguinAnd they support IAX2 if you're into high volume calling and want to trunk.
23:12.35rotten777of course voip.ms has local did's on backorder...
23:12.36rotten777lol
23:12.42rotten777*sigh*
23:12.57p3nguinAnother good thing I like about VoIP.ms is that they offer a refund of unused funds if you are not satisfied.
23:14.11dijib& more then 2half months only has cost me $14.69
23:14.25dijib2 and half
23:14.43p3nguinYou must have paid for the 3000 minute package.
23:14.52dijibnope. toll free did
23:14.54dijibthats it..
23:15.13dijibi dont use the phone much.. ive got more testing calls then any real ones
23:15.20p3nguinGood grief.  I don't spend that much in 6 or more months.
23:15.20dijibp3nguin, feeling better today?
23:15.32dijibhow do u do it then?
23:15.38p3nguinI wasn't feeling poorly before today.
23:15.52dijibi thought i had unlimited incomming but it doesnt look like it
23:15.58dijibor unlimited outgoing... or something
23:16.08p3nguinMy DID is $0.99 per month, so that's just $12 per year.
23:16.16p3nguinPlus inbound calls.
23:16.58p3nguinIn their toll-free numbers, they only do pay-per-minute.
23:17.05p3nguinsame for outgoing calls.
23:17.39p3nguinOnly regular DIDs have unlimited (limited to 3000 minutes) calling.
23:25.58p3nguinI guess the way I get by so cheap is by having several DIDs, some of which are free, and don't have a lot of inbound calling on the pay-per-minute numbers.
23:26.15p3nguinAlso, a lot of my calling is to toll-free numbers, so those don't cost me anything.
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23:33.08rotten777well i'll take a toll free did but i'm on manual activation from voip.ms
23:33.13rotten777so i have to wait for a human
23:38.39p3nguinpewp
23:38.48rotten777flowroute looks good though
23:38.50rotten777pretty slick interface
23:38.53rotten777decent prices
23:39.10p3nguinslightly higher than voipms, but still very reasonable.
23:39.27p3nguinActually they might be less on per-minute rates.
23:39.32p3nguinBut the DIDs cost more.
23:39.37rotten7771.2 cents/m
23:40.18p3nguinfor what?
23:40.27rotten777DID
23:40.29rotten777flowroute
23:43.05p3nguinOkay, so they are slightly higher on DIDs and minutes.
23:43.16p3nguinMaybe it's the termination rate that is slightly lower.
23:43.30rotten777in comparison to using my cell 100% of the time, they're exponentially cheaper
23:43.40rotten777and seeing as i'm on sprint, more reliable
23:44.40*** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6)
23:55.49*** join/#asterisk mindCrime_ (~chatzilla@static-50-52-147-222.drhm.nc.frontiernet.net)

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