| 00:00.58 | slav3_kitten | one step at a time right? |
| 00:02.39 | drmessano | Yep |
| 00:03.04 | drmessano | I wonder if I should be concerned that posting to FB via command line is harder than Twitter |
| 00:03.14 | drmessano | errrr, thats backwards |
| 00:03.20 | drmessano | I wonder if I should be concerned that posting to FB via command line is EASIER than Twitter |
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| 00:12.57 | slav3_kitten | i'm concerned you use either an both |
| 00:15.12 | slav3_kitten | in my mind, if you have my phone number you're in my social network... otherwise you're not |
| 00:19.15 | drmessano | I hate telephones |
| 00:20.16 | sawgood | sometimes I really hate telephones |
| 00:23.59 | resist0r | slav3_kitten: word |
| 00:24.17 | resist0r | I dont have a facebook or twitter account either |
| 00:24.26 | slav3_kitten | i <3 phones |
| 00:24.31 | slav3_kitten | that's why i'm in here |
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| 02:01.30 | volga629 | I still have troubles with dial local extensions with ldap, just can't find any information what need to be in extensions and what options |
| 02:04.05 | carrar | need more googlefoo |
| 02:13.11 | volga629 | I am all last week tried find some, I got ldap working user registration working on 85% and dilaplan completely now going anywhere |
| 02:13.29 | volga629 | not |
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| 02:22.55 | JamKo | Is there a way to tell which version of 1.6.x this patch started going into? https://reviewboard.asterisk.org/r/310/diff/#index_header .. I am using 1.6.2.12 installed a couple of years ago, and would like to know if it installed with that patch. |
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| 02:43.40 | JamKo | I am having some issues negotiating a T.38 passthrough. My provider is telling me "Can you change the behavior of the T.38 reINVITE whereby it only sending the T.38 reINVITE when it is the MG (e.g. UAS) per ITU specifications on T.38 negotiation?" |
| 02:43.55 | JamKo | "The mechanism to handle the T.38 negotiate is handled in the core of Asterisk and your dial plan." |
| 02:44.44 | JamKo | Anyone know what it is I am after to change? |
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| 03:32.20 | Yourname` | Hello. I have a sip trunk that uses eth1 transport (VPN link). In the SIP packets, the IP that's going is our public IP (set using externip), but I want it to show the VPN IP assigned to us (192.168.200.2 =eth1 and 192.168.1.252=eth0).. can I set something on the trunk that shows our internal IP for the calls that go through it? |
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| 04:52.23 | apc-sphilp | Having a problem with Asterisk 1.8.12 / Exchange 2010 SP2 Unified Messaging that I'm hoping someone has run across before. Every 4 minutes, the Exchange box is logging a UMCore event 1400 -- The UM IP gateway did not respond as expected to a SIP OPTIONS request. Response Code = 404, Message = Not Found. Any ideas how to resolve? |
| 04:57.39 | kaldemar | apc-sphilp: how do you want it resolved? do you not want to see logs or do you not want to send OPTIONS messages? |
| 04:58.41 | apc-sphilp | Good question :) I'm not sure why it's sending OPTIONS messages, so I don't know what the impact of not sending them would be. But, turning off the logs is probably quicker! |
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| 05:02.47 | [TK]D-Fender | apc-sphilp: pastebin the SIP DEBUG from the * |
| 05:02.49 | [TK]D-Fender | side |
| 05:05.00 | apc-sphilp | Here's the output from "sip set debug peer ExchangeUM": http://pastebin.com/b5drsVYB |
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| 05:12.14 | kaldemar | apc-sphilp: you probably have qualify=yes enabled for the peer. that's an option that makes asterisk send OPTIONS messages for keep-alive and monitoring purposes. |
| 05:13.41 | apc-sphilp | That peer doesn't have a qualify=, but the [general] section of sip.conf does have a qualify=3000 statement. Does it have the same effect? |
| 05:18.02 | kaldemar | apc-sphilp: almost. you can put qualify=no for the peer. |
| 05:18.51 | apc-sphilp | Okay, I've made that change and reloaded sip. Waiting for the 4-minute mark to see if the Exchange message shows up again |
| 05:21.51 | [TK]D-Fender | apc-sphilp: that is not a 404 |
| 05:22.04 | [TK]D-Fender | I think you got the debug direction wrong enable GLOBAL sip debug, not restricted |
| 05:22.37 | apc-sphilp | Okay, will give that a try next |
| 05:27.23 | apc-sphilp | [TK]D-Fender: Ah, think I see what's happening... Helps when I'm looking at the right debugs :) |
| 05:27.32 | [TK]D-Fender | Usually |
| 05:27.56 | apc-sphilp | I see the OPTIONS message from the Exchange server, then a "Looking for s in internal (domain 10.1.4.9)", then the 404 going back out. |
| 05:28.10 | apc-sphilp | Guessing I'm missing the "s" extension in [internal] |
| 05:28.52 | [TK]D-Fender | So make a dummy one for it there |
| 05:29.07 | [TK]D-Fender | It doesn't have to do anything in particular |
| 05:29.46 | apc-sphilp | Cool, I set it up with a NoOp for now |
| 05:30.42 | apc-sphilp | Yep, that fixed it. Exchange no longer logs the 1400 warning and instead logs a 1401 "The gateway responded promptly to a SIP OPTIONS request." |
| 05:30.45 | apc-sphilp | Thank you! |
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| 05:33.48 | volga629 | I am looking for info on extensions options ldap just can't make local call between extensions to work from ldap |
| 05:36.24 | volga629 | dial plan just not working for me luck of information and asterisk constantly crush when can't determine extensions |
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| 05:44.51 | WIMPy | volga629: You don't perhaps have both a switch and an include in that context, do you? |
| 05:45.25 | volga629 | no only switch |
| 05:46.31 | volga629 | I need first do some priority through ldap which will say what does each extensions in in next step |
| 05:48.45 | volga629 | that as far I got local extensions not match http://fpaste.org/0Gek/ |
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| 05:52.37 | volga629 | that current setup http://fpaste.org/aNPa/ |
| 05:53.15 | volga629 | I just stack what need to be next and which form need define |
| 05:55.02 | WIMPy | It was just an idea. I just fixed my dialplan issues by replacing an include => but an lswitch => Loopback/. |
| 05:55.21 | WIMPy | But I've got NFI about LDAP, sorry. |
| 05:55.52 | WIMPy | s/but/by/ |
| 05:57.46 | volga629 | If I will find the way properly put all extensions through ldap and dial between it, I will share the solution, thank you. You always trying to help. |
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| 07:14.52 | bulkorok | hi |
| 07:16.49 | ChannelZ | whips off his pants |
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| 11:16.42 | Ice_Strike | Hello |
| 11:17.12 | WIMPy | awhat? |
| 11:18.53 | Ice_Strike | Asterisk currently running 20 channels (concurrent calls) on VoiP1 but other sip user cant make any call via Voip2 - keep getting hangup. |
| 11:19.40 | Ice_Strike | If I lower the concurrent calls - it seem to work fine for other sip user to make a call voa Voip2 |
| 11:19.45 | Ice_Strike | What is going on? |
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| 11:59.45 | slav3_kitten | zombie shufffles |
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| 12:27.19 | Gugge | Ice_Strike: you dont have nearly enough info there to get help. |
| 12:27.40 | Gugge | Ice_Strike: info about what voip1 and voip2 is, and how they relate to eachother is missing |
| 12:28.07 | Gugge | Ice_Strike: and an verbose log of the failed call would help too. |
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| 12:40.29 | slav3_kitten | needs to nuke linux at the moment to feel not so frustrated |
| 12:51.33 | acidfoo | does anyone use cel_pgsql and have experienced memory leak ? |
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| 14:06.03 | slav3_kitten | ok.... i got one of the cisco phones provisioned |
| 14:06.31 | zxvff | cool |
| 14:06.35 | zxvff | now provision another |
| 14:07.29 | slav3_kitten | well my log is like filled with "registered SIP 'mac addy' at 192.168.10.10:53144 and 53143 |
| 14:07.33 | slav3_kitten | is that normal |
| 14:07.41 | slav3_kitten | i am running asterisk -cvv |
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| 14:23.08 | edge | Our towns power got hit pretty good last night, and my phone system got rebooted. Now the queue shows everybody as unavailable but they are available because i can call them. How do i reload them into the queue? |
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| 14:25.21 | JamKo | How can I tell if this patch was included in my version of 1.6.2.12-rc1: https://reviewboard.asterisk.org/r/310/ .. ? |
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| 14:29.07 | [TK]D-Fender | JamKo, Go find it on the tracker and check the commit dates |
| 14:29.20 | slav3_kitten | why on earth does this phone keep registering! |
| 14:29.23 | [TK]D-Fender | JamKo, And that is an old release of a dead branch. |
| 14:29.39 | [TK]D-Fender | slav3_kitten, go change the interval |
| 14:29.53 | slav3_kitten | [TK]D-Fender, in what? asterisk or the phone? |
| 14:30.01 | slav3_kitten | the first ? should be a , |
| 14:30.13 | [TK]D-Fender | on the phone |
| 14:30.39 | slav3_kitten | yea i did that via the config files i was given last night. it seems to be just giving me the finger |
| 14:30.57 | slav3_kitten | phone work, can't call itself though |
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| 14:31.32 | [TK]D-Fender | Talking to your self is typically considered to be insane.... |
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| 14:32.12 | slav3_kitten | very true but the soft phone i had could ring itself but would hangup when you go to answer line 2 |
| 14:32.21 | slav3_kitten | which i found funny |
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| 14:34.49 | slav3_kitten | lack of documentation = frustrating |
| 14:35.08 | edge | Its odd sip sees the phones connected the queue says they aren't |
| 14:35.21 | *** join/#asterisk Thorn (~Thorn@unaffiliated/thorn) |
| 14:35.37 | edge | and only by rebooting the phones is this fixed |
| 14:35.55 | slav3_kitten | Thorn, you're not in wisconsin are you? |
| 14:36.31 | Thorn | huh? not even close |
| 14:37.37 | [TK]D-Fender | edge, Any odds of your showing us the backup for your problem? |
| 14:38.18 | edge | [TK]D-Fender: the backup? |
| 14:38.40 | [TK]D-Fender | yes. Showing us your queue dumps, etc... |
| 14:38.51 | edge | sure |
| 14:39.06 | edge | i manually walked around the building and reset half the memebers and they joined |
| 14:39.14 | edge | the other half i did not reboot (so i could attempt to resovle it) |
| 14:39.38 | edge | i can pickup my phone and call those people, but when i do a 'queue show' they are listed as being unavailable |
| 14:39.58 | edge | [TK]D-Fender: doesn't it have to crash to get a dump? |
| 14:40.53 | [TK]D-Fender | show us the queue from CLI. attempted calls in, channel dump, etc |
| 14:42.27 | edge | [TK]D-Fender: actually in a twist, sip show peers, has them listed as 'unknown' perhaps the phones are within their registration window and think they're still connected while asterisk does not. Which would explain why they are missing from the queue until reboot and then they re-register |
| 14:42.48 | [TK]D-Fender | indeed |
| 14:43.24 | edge | [TK]D-Fender: sorry to waste your time. I was a little nervious that no calls were comming in and was a little short of breath trying to figure it out (no oxygen to my brain) |
| 14:43.41 | edge | I think i will shortten the registration peroid on the phones |
| 14:44.20 | [TK]D-Fender | Step one : collect all the evidence and shove it in a pastebin. Step two : let other eyes look at it before you drive yourself crazy. |
| 14:45.27 | slav3_kitten | mother fuc..... |
| 14:45.40 | slav3_kitten | nope still crazy registering... |
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| 14:49.50 | edge | [TK]D-Fender: actually people were calling me and i was calling them. Does this happen without registration. And i think to make matters worse, since the power was out the phones had the right time, and the server was back in 2006 (gota replace battery) and while asterisk was running i fixed the time. which i'm sure had a good deal of effect on it |
| 14:50.45 | [TK]D-Fender | Phones don't need to be registered to place calls, and on some phones calling out may trigger re-registration.... |
| 14:51.08 | edge | [TK]D-Fender: ah. Thank you very much. I feel better now. |
| 14:51.53 | edge | [TK]D-Fender: , actually, one more quick question. Right now when i'm talking on the phone and another person calls me, it rings. Is that a change i'd have to make to the dial plan with hints to give them a busy tone, vs ringing me? Or is that a sip.conf change? |
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| 14:53.06 | d_preston215 | Chanspy spygroups can automatically listen to two channels in a walkie-talkie like format, right? |
| 14:53.18 | [TK]D-Fender | edge, You don't want to ever be able to take a call while being on a call? |
| 14:53.28 | [TK]D-Fender | edge, You know even analog has "call waiting" .... |
| 14:53.58 | [TK]D-Fender | edge, If you're referring to queue purposes, then there are Queue parms to limit this |
| 14:55.04 | edge | [TK]D-Fender: A couple of people had asked me about this because with our old key system you couldn't call a line that was already on the phone and for some reason they missed that. They also liked being able to see a triangle over their extension showing they were already on the phone. VS right now they call the # and it rings rings and rings and then goes to voice mail |
| 14:55.27 | Ralcala | its Asterisk 1.8.11 on fc16 a good distro to work or not? |
| 14:56.12 | *** join/#asterisk GameGamer43 (~users.553@gateway/web/irccloud.com/x-nhwghowzylpqesvn) |
| 14:56.16 | [TK]D-Fender | edge, Seems they want a very dumbed down system. You could try limiting on the phone, or adapting your dialplan to check first. |
| 14:57.10 | [TK]D-Fender | Ralcala, Whatever distro you can properly manage is fine. 1.8.13 is out however so..... |
| 14:57.23 | Ralcala | cause i cant register a 4 FXO GW |
| 14:57.46 | [TK]D-Fender | Ralcala, The versions aren't your problem. |
| 14:57.54 | [TK]D-Fender | Ralcala, Your configuration is |
| 14:58.02 | Ralcala | i installed CentOS on another pc with no problems |
| 14:58.19 | [TK]D-Fender | Ralcala, pastebin your sip.conf entries for it along with the SIP DEBUG from * CLI for the failed attempts. |
| 14:58.21 | [TK]D-Fender | ~pb |
| 14:58.21 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:58.23 | [TK]D-Fender | ^^^ |
| 14:58.27 | [TK]D-Fender | Ralcala, It isn't the distro. |
| 14:58.38 | Ralcala | ok |
| 14:59.12 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 15:00.41 | Ralcala | err one question how i get SIP DEBUG? |
| 15:00.49 | Ralcala | sorry |
| 15:00.56 | Ralcala | i newbie |
| 15:01.02 | [TK]D-Fender | "sip set debug on" |
| 15:01.12 | Ralcala | thanks |
| 15:01.53 | slav3_kitten | http://pastebin.com/TG3BKC6p |
| 15:03.03 | [TK]D-Fender | slav3_kitten, Looks like it should be hourly |
| 15:03.08 | slav3_kitten | it should be |
| 15:03.15 | slav3_kitten | it's like random secondly |
| 15:04.28 | slav3_kitten | seems to work though, but when i end a call i get a retransmission timeout |
| 15:04.43 | JamKo | [TK]D-Fender: There is a revision number 206279 attached to it, within the patch. I coming up empty searching on the revision number, at the old and new tracker site. |
| 15:05.44 | [TK]D-Fender | JamKo, Then check by date. And then reconsider the fact you're asking about an old RC. You should upgrade to the latest as it is... |
| 15:08.28 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:08.30 | Ralcala | Fender i got sip.conf but hwo extract sip debug its a log file? |
| 15:08.40 | Ralcala | where is? |
| 15:08.59 | JamKo | [TK]D-Fender: You are right, I should. But upgrading breaks sht, and this one runs correctly for the most part. |
| 15:09.21 | [TK]D-Fender | Ralcala, * live CLI only, no logs. |
| 15:09.32 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:09.33 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:09.52 | Ralcala | * CLI shows nothing... |
| 15:10.18 | [TK]D-Fender | Ralcala, then packets aren't even arriving or you didn't properly set sip debug |
| 15:10.32 | [TK]D-Fender | Ralcala, an if they aren't arriving.. you've misconfigured a device or firewall |
| 15:11.01 | Ralcala | ok |
| 15:11.12 | Ralcala | checking |
| 15:11.27 | *** join/#asterisk jbolo (~jonathan@200.48.64.2) |
| 15:11.32 | *** part/#asterisk jbolo (~jonathan@200.48.64.2) |
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| 15:13.35 | slav3_kitten | lol @ hotel connect |
| 15:13.49 | slav3_kitten | jitter is 6249.0 ms |
| 15:15.34 | *** part/#asterisk sickgrinder (~MattB@rrcs-97-76-33-146.se.biz.rr.com) |
| 15:17.51 | *** join/#asterisk sickgrinder (~MattB@rrcs-97-76-33-146.se.biz.rr.com) |
| 15:18.23 | slav3_kitten | [TK]D-Fender, you ever deal with a cisco CCIE? |
| 15:18.36 | [TK]D-Fender | nope |
| 15:19.01 | slav3_kitten | i'm currently discussing the phone thing with one... they can't think outside the box it seems |
| 15:19.14 | kanelbullar2 | Hi guys. Are there any plans to backport the ICE/STUN/TURN support to asterisk 1.8? |
| 15:20.49 | [TK]D-Fender | kanelbullar2, Last I heard it's in there already... |
| 15:21.06 | [TK]D-Fender | kanelbullar2, Not that more than 1% need it |
| 15:21.51 | kanelbullar2 | Ok, thanks D-Fender. I thought it was only in trunk. I will check it out. |
| 15:23.59 | [TK]D-Fender | And that isn't "trunk". |
| 15:24.21 | [TK]D-Fender | Trunk is what will become * 11. no 8 or 10 branch counts as that |
| 15:24.28 | [TK]D-Fender | 1.8 / 10 |
| 15:26.12 | kanelbullar2 | Yes, that was I meant. I thought it had only been implemented in trunk as a new feature and not in 1.8. |
| 15:29.01 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 15:29.02 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:31.58 | slav3_kitten | head desks repeatedly |
| 15:34.44 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:38.55 | Qwell | woah it's an angler |
| 15:41.20 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v001-081.mobile.uci.edu) |
| 15:55.33 | slav3_kitten | hrmmm |
| 15:55.44 | slav3_kitten | why is this not properly hanging up |
| 15:57.59 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 15:58.59 | *** join/#asterisk dorphalsig (be9399cc@gateway/web/freenode/ip.190.147.153.204) |
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| 16:00.18 | *** join/#asterisk din3sh (din3sh@41.136.83.138) |
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| 16:19.37 | *** join/#asterisk twodogs (~twodogs@telok.vitiate.me) |
| 16:26.25 | slav3_kitten | GOD I'M GOING TO KILL IT! |
| 16:26.37 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:27.04 | d_preston215 | Likewise. |
| 16:28.52 | slav3_kitten | oh? |
| 16:29.16 | d_preston215 | I want to monitor two channels at once. |
| 16:29.34 | slav3_kitten | sounds tricky |
| 16:29.47 | slav3_kitten | i want my cisco phone not to register every few seconds |
| 16:30.05 | Qwell | so change the registration timer |
| 16:30.16 | slav3_kitten | Qwell, like i didn't try that first :| |
| 16:35.11 | slav3_kitten | i've got a defaultexpiry=120 in sip.conf |
| 16:35.32 | slav3_kitten | changed the registration interval to 120 in the cisco config files.. |
| 16:36.31 | [TK]D-Fender | That is seconds... |
| 16:36.38 | [TK]D-Fender | every 2 minutes = a bit much... |
| 16:36.55 | slav3_kitten | err 3600*** |
| 16:37.05 | slav3_kitten | < reading docs |
| 16:39.53 | *** join/#asterisk thehar (~thehar@173.255.210.136) |
| 16:40.56 | slav3_kitten | sorry i was reading the asterisk docs that sate 120 as default |
| 16:42.14 | slav3_kitten | and the register flood pours in |
| 16:54.45 | *** join/#asterisk AaronCharpentier (~IceChat77@pool-71-162-114-189.bstnma.fios.verizon.net) |
| 16:55.42 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 16:55.47 | AaronCharpentier | Hi guys, huge emergency here - I'm the IT for a medium sized company, we rely on our asterisk pbx box (Cbeyond service, Verizon T1 line) for our entire phone system, it's been down since around 9 am. The T1 line is damaged and won't be returning, we're losing out on massive amounts of time right now due to this. |
| 16:56.08 | AaronCharpentier | We have a FIOS connection, the phone server(asterisk server, centos) is also connected to the fios line for internet for system updates and local routing. |
| 16:56.38 | AaronCharpentier | Is it at all possible for us to change the routing to not go through our T1 Wildcard and to use our fios instead for communication with the Cbeyond servers to temporarily restore our service? |
| 16:56.44 | Qwell | sure |
| 16:57.03 | AaronCharpentier | Please, enlighten me - I can offer web development, hosting and many other services free of charge for your assistance. |
| 16:57.13 | Qwell | Change from Dial(Zap/g1/${EXTEN}) to Dial(SIP/myitsp/${EXTEN}). YMMV. |
| 16:57.28 | Qwell | (remember that we know nothing of your config, at this point) |
| 16:57.45 | nny | AaronCharpentier: Cbeyond should have a SIP connection as well. I work with them on some of the asterisk projects |
| 16:57.47 | AaronCharpentier | Where would I find this code, would it be in the GUI or a local file on the asterisk server, and definitely - any information you need I can try my best to provide. |
| 16:57.53 | Qwell | GUI? |
| 16:57.56 | nny | AaronCharpentier: is this freepbx? |
| 16:57.58 | [TK]D-Fender | cbeyond is an ITSP, and the T1 was data, not voice |
| 16:58.06 | AaronCharpentier | Graphical user interface |
| 16:58.10 | nny | AaronCharpentier: nm I know it is |
| 16:58.13 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 16:58.20 | AaronCharpentier | Yes, freepbx/asterisk |
| 16:58.20 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 16:58.32 | Qwell | [TK]D-Fender: They don't offer physical T1s? |
| 16:58.34 | nny | AaronCharpentier: you'll have to have their engineers connect your PBXinaflash install to Cbeyond via SIP |
| 16:58.55 | [TK]D-Fender | Qwell, Just a question of knowing what these guys deploy from experience |
| 16:58.59 | Qwell | Nothing to do! |
| 16:59.09 | AaronCharpentier | So no quick fix then, we do have a physical T1 from verizon |
| 16:59.16 | Qwell | AaronCharpentier: Voice or data? |
| 16:59.19 | nny | AaronCharpentier: connectng via SIP is a quick fix imho |
| 16:59.30 | AaronCharpentier | they contract out to verizon, the t1 is verizon |
| 16:59.47 | [TK]D-Fender | AaronCharpentier, IIRC Cbyeond uses IP auth. You'd have to contact them to let them know the IP of your FIOS and change your trunk accordingly. |
| 16:59.50 | paulc | SIP outbound is easy, with any provider - the bigger issue is your inbound calls |
| 16:59.52 | Qwell | well, hell, call cbeyond then. |
| 16:59.59 | AaronCharpentier | Sorry I think I'm too inexperienced to take your advice, I don't want to waste your time with more questions, I'll continue looking into it, gotcha |
| 16:59.59 | Qwell | This is a 2 minute fix. >.> |
| 17:00.00 | AaronCharpentier | oh we have |
| 17:00.09 | AaronCharpentier | we've bitched at them all morning |
| 17:00.34 | [TK]D-Fender | and? |
| 17:01.04 | AaronCharpentier | theyre repeatedly giving us the runaround, saying nothing can be done and theres no eta on when the service can be restored |
| 17:01.16 | AaronCharpentier | im calling them now to inquire about switching our box to use the fios connection its already also connected to |
| 17:01.29 | *** join/#asterisk dorphalsig (be9399cc@gateway/web/freenode/ip.190.147.153.204) |
| 17:02.10 | nny | AaronCharpentier: good luck |
| 17:04.22 | [TK]D-Fender | AaronCharpentier, I was asking about calling them about the FIOS |
| 17:04.40 | [TK]D-Fender | AaronCharpentier, Let us know when you've got the details ready for that |
| 17:04.44 | nny | i've been trying to get them to host a centos install for one of their clients over FBXinaflash for weeks, it's sullied the whole sales process :S |
| 17:04.49 | AaronCharpentier | ah gotcha havent asked them yet, im on hold right now |
| 17:05.08 | AaronCharpentier | they likely dont want to pick up because they've gotten so much heat from my supervisor and myself today lol |
| 17:05.19 | AaronCharpentier | we've had no phones all day, we rely on them intensely lol |
| 17:08.05 | nny | ouch |
| 17:11.17 | *** join/#asterisk jsjc (~Adium@82.158.173.96.dyn.user.ono.com) |
| 17:15.51 | *** join/#asterisk jsjc (~Adium@82.158.173.96.dyn.user.ono.com) |
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| 17:28.46 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
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| 17:31.34 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:33.36 | Qwell | oej: You appear to be coming from an IPv4 address. :( |
| 17:33.54 | AaronCharpentier | been transfered to 3 different people, told its not possible, told its possible, told its not possible and unsupported, told its possible but unsupported...rofl |
| 17:33.59 | AaronCharpentier | on hold again |
| 17:34.24 | _Corey_ | AaronCharpentier: "Call forwarding" + Any ITSP |
| 17:34.38 | Qwell | _Corey_: number port would take several days |
| 17:34.45 | Qwell | err, nevermind |
| 17:35.44 | _Corey_ | Qwell: Depending on the carrier, call forwarding can take days... ;) (That's right AT&T... I'm talking about YOU) |
| 17:36.57 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:40.54 | AaronCharpentier | haha |
| 17:41.58 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:42.43 | [TK]D-Fender | AaronCharpentier, If you're getting the runaround then you aren't asking clear questions. |
| 17:45.05 | AaronCharpentier | Well, I can assure you I am asking very clear articulate questions, they're just very unhappy with us, and the indian guy I spoke with could not understand anything I was asking him. |
| 17:46.05 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 17:48.19 | _Corey_ | AaronCharpentier: If you used the word FiOS with Cbeyond you probably confused them. Ideally, you want to establish a SIP connection via the Internet with them to route calls. Next best thing is to get service with someone else and just forward all calls. |
| 17:48.57 | AaronCharpentier | Yeah, we're looking into alternate solutions. |
| 17:49.41 | nny | AaronCharpentier: I have a contact there, want me to try him? |
| 17:51.36 | AaronCharpentier | Just got off the phone, and hey yeah if you can get any info as to what needs to be done for us to route it through our internet then certainly. |
| 17:52.33 | Qwell | and today you learned the importance of SLAs. ;/ |
| 17:52.42 | AaronCharpentier | lmao |
| 17:53.06 | AaronCharpentier | Today I learned the importance of too many cigarettes to calm myself down and a lot of advil for the splitting headache I got dealing with some of these guys, especially at verizon. |
| 17:54.02 | nny | AaronCharpentier: ouch. Yeah i assume you have access to the Fpbx control panel? |
| 17:54.23 | AaronCharpentier | yeah |
| 17:55.12 | nny | you can setup a second trunk through the Web UI, as far as providers, I can suggest a couple... |
| 17:55.20 | nny | that would provide outbound calling instantly |
| 17:55.45 | AaronCharpentier | But inbound calls would be a different story, yeah? |
| 17:56.20 | AaronCharpentier | Also, which providers would you suggest? |
| 17:56.25 | nny | you would need to setup an inbound DID from the provider, set that up in freepbx and forward the cebyond number to it through their support |
| 17:56.34 | nny | this btw is a good backup plan to have around almost always |
| 17:56.44 | AaronCharpentier | We're currently in the market, we looked up our contract and we can currently break it. |
| 17:57.26 | nny | AaronCharpentier: backups never hurt, you'll find most providers have a better cost point |
| 17:57.50 | l0st-soul | AaronCharpentier: i dont know anything about cbeyond, but cant you configure your ip in their web interface ? |
| 17:58.09 | l0st-soul | (if they use ip auth) |
| 17:58.17 | nny | AaronCharpentier: I have used vitelity and voipstreet so far. Disclaimer, one of my clients is having some disconnects with vitelity (only one, i have used them for years). voipstreet is a new testing ground for me, their webui is kinda scary |
| 17:58.46 | nny | i have also use flowroute but it seemed more geared towards home and soho |
| 17:59.17 | AaronCharpentier | Gotcha |
| 18:00.35 | nny | both voipstreet and vitelity offer configuration help |
| 18:00.43 | AaronCharpentier | nice |
| 18:00.51 | *** join/#asterisk suedoh (~suedoh@cpe-98-149-151-89.socal.res.rr.com) |
| 18:00.59 | AaronCharpentier | and voipstreet could run through our existing 35/35 fios line, yes? |
| 18:01.03 | nny | so once you have an account setup you can use their "wizards". Over all i am trying voipstreet in hopes of moving to them |
| 18:01.16 | nny | AaronCharpentier: yes, i am assuming some things here |
| 18:01.20 | nny | 1.) your pbx is in house |
| 18:01.27 | AaronCharpentier | yes it is |
| 18:01.31 | nny | 2.) you have a public interface or proper ports forwarded |
| 18:01.35 | AaronCharpentier | indeed |
| 18:01.41 | nny | then yeah you should be good. |
| 18:02.47 | AaronCharpentier | awesome thank you i am giving them a call right now. |
| 18:02.52 | nny | AaronCharpentier: you also can use the #freepbx channel to dig down on conifguration. If you ask about it here you'll find that this channel doesn't deal with freepbx as much so won't have answers |
| 18:03.04 | nny | AaronCharpentier: but generally their support should be very helpful, has been for me |
| 18:03.22 | nny | then all you need is to get cebyond to forward the inbound DID to your new DID. |
| 18:05.07 | Qwell | ~itsplist-us |
| 18:05.07 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:05.08 | Qwell | fwiw |
| 18:06.44 | nny | ew broadvoice |
| 18:06.57 | Qwell | didn't make the list |
| 18:07.01 | nny | hahaha |
| 18:07.10 | nny | i know, just configuring for broadvoice is like pulling teeth ;) |
| 18:07.17 | nny | the rest i agree with |
| 18:07.26 | nny | i am testing voipstreet, i'll let ya know |
| 18:07.36 | zxvff | one time I set up an asterisk server |
| 18:07.54 | Qwell | hey me too! |
| 18:08.11 | zxvff | it was fun and now i am here on asterisk irc |
| 18:08.23 | zxvff | hi Qwell |
| 18:09.12 | slav3_kitten | configuring cisco.... also pulling teeth |
| 18:09.29 | zxvff | dentistry can be like pulling teeth sometimes |
| 18:09.39 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:09.43 | slav3_kitten | nah they drill more then pull these days |
| 18:09.54 | zxvff | true |
| 18:10.03 | zxvff | i wouldn't know, i've not been to a dentist in something like 8 years |
| 18:10.16 | slav3_kitten | i foresee a lot of drilling |
| 18:10.19 | zxvff | i should go since i have had benefits for the past 2 years |
| 18:10.27 | zxvff | insurance i mean |
| 18:11.01 | slav3_kitten | you get a teeth cleaning with happy ending? |
| 18:11.07 | *** part/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 18:11.15 | zxvff | i do? brb, dentist |
| 18:11.31 | slav3_kitten | LOL |
| 18:12.33 | _Corey_ | happy endings... now if my dentist looked like jennifer aniston in 'horrible bosses'... |
| 18:13.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:15.08 | Qwell | _Corey_: my dentist has good taste in hygenists. |
| 18:15.47 | Qwell | I wonder if he uses Asterisk... I should get on that |
| 18:18.10 | _Corey_ | Qwell: medical/dental offices are pleasant places to deploy pbxs in that regard |
| 18:19.25 | Qwell | My dentist is really awesome too. He knows tech stuff (he told me he stopped paying the phonebook people, and went exclusively to Google Adwords, and it has paid off). |
| 18:19.34 | _Corey_ | I spent a week deploying/training at a big OBGYN center once... between the staff and the patients coming through, let's just say it wasn't a bad week |
| 18:19.44 | *** join/#asterisk sarsaeol (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 18:19.50 | Qwell | He knew of Digium when he asked where I worked on my first visit |
| 18:20.18 | Qwell | He's also the only Dr. I know that answers his own phone sometimes. |
| 18:20.18 | zxvff | do you work for Digium? |
| 18:20.19 | _Corey_ | very nice |
| 18:20.25 | Qwell | zxvff: Who? Never heard of them. |
| 18:20.30 | zxvff | ok |
| 18:20.36 | Qwell | (yes..) |
| 18:20.51 | zxvff | ha |
| 18:21.29 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-zancxltxwsvjolfu) |
| 18:21.44 | *** join/#asterisk tamiel (~tamiel@c-69-181-167-169.hsd1.ca.comcast.net) |
| 18:22.03 | Qwell | That was the weirdest experience.. I called his office one day, and he answered the phone. I was taken aback for a second, but then when I asked my question, he immediately knew who I was - by voice alone. |
| 18:22.54 | _Corey_ | my dentist primarily knows me by the mumbling noises |
| 18:23.01 | Qwell | heh |
| 18:24.00 | sarsaeol | i had a similar experience recently with a new eye doctor. answers all his own calls… and when i called to see how my lenses were coming he gave me an update with me just having to say "hi" the whol phone convo took 20 seconds |
| 18:24.35 | Qwell | sarsaeol: very nice |
| 18:24.57 | sarsaeol | yeah, although i initially thought he jsut wanted me off the phone ;) |
| 18:27.14 | slav3_kitten | my doctor knows me best by "needless" sti screenings |
| 18:27.28 | zxvff | i should go get one of those |
| 18:28.03 | slav3_kitten | i'm a bit paranoid |
| 18:28.12 | zxvff | i'm a bit lazy |
| 18:28.16 | zxvff | hence not going to doctors or dentist |
| 18:29.09 | slav3_kitten | so it normaly goes "so doc, need a sti screening" then "have you been sexually active since the last one?" then i go "no" and i get a 20 minute basically "then why are you getting one" to which i respond "no longer recent, there is a woman i'm possibly going to do something with an i require her to have recent test results, i need to supply the same" |
| 18:29.17 | slav3_kitten | and i get an eye roll an such |
| 18:29.36 | zxvff | lol |
| 18:29.47 | zxvff | yes that's silly |
| 18:29.51 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:30.04 | slav3_kitten | zxvff, i've never had a sti or even a scare. thus it's not silly |
| 18:30.15 | zxvff | if a girl told me that i had to have test results to sleep with her i would tell her to gtfo |
| 18:30.25 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:30.38 | slav3_kitten | well that's why you're going to contract herpes an pass it around one day :P |
| 18:30.38 | sarsaeol | i requre a resume and three references before making love |
| 18:30.48 | zxvff | lol |
| 18:31.46 | slav3_kitten | sarsaeol, british chick i'm picking up from the airport sunday came with a resume and references |
| 18:32.08 | zxvff | well i suppose it makes more sense if you're into international sex trafficking |
| 18:32.14 | sarsaeol | LMAO |
| 18:32.40 | sarsaeol | slav3_kitten: were they actual like past-relationship references? |
| 18:32.41 | slav3_kitten | pffft if only |
| 18:33.01 | slav3_kitten | she's been a friend for years and we are hitting up anthrocon |
| 18:33.07 | zxvff | :/ |
| 18:33.09 | zxvff | i don't like you. |
| 18:33.21 | slav3_kitten | sarsaeol, actually i wouldn't really call them relationships but ye they were past flings |
| 18:33.26 | slav3_kitten | zxvff, why not?! |
| 18:33.31 | zxvff | you're a furry |
| 18:33.37 | slav3_kitten | and your point? |
| 18:33.41 | zxvff | i don't like you. |
| 18:33.54 | zxvff | I think you're a 'bad person' (TM) |
| 18:33.58 | sarsaeol | slav3_kitten: thats amazingly thorough |
| 18:34.02 | slav3_kitten | way to be open minded |
| 18:34.05 | zxvff | i'm not |
| 18:34.56 | slav3_kitten | i see that |
| 18:35.15 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:35.35 | slav3_kitten | zxvff, fyi furries are not what the media makes them out to be. well some are but i dislike those |
| 18:35.48 | zxvff | let's chat about asterisk |
| 18:36.11 | slav3_kitten | you mean instead of your intolerance? |
| 18:37.37 | slav3_kitten | you're not allowed to dislike me until you have a real reason to, my subculture affiliations are immaterial in a room such as this. |
| 18:39.14 | seanbright | is lumenvox as much of a fly-by-night company as they appear? |
| 18:39.29 | Qwell | seanbright: They've been around forever |
| 18:39.59 | seanbright | they change product offerings every 3 weeks |
| 18:39.59 | brettnem | wtf, am I in the asterisk chat room? |
| 18:40.06 | seanbright | brettnem: no |
| 18:40.15 | seanbright | this is the asterisk chat room |
| 18:40.20 | seanbright | not the asterisk one |
| 18:40.37 | brettnem | oh.. i see... |
| 18:40.53 | brettnem | good to see things haven't changed much |
| 18:42.25 | seanbright | Qwell: have you ever heard of anyone integrating anything other than lumenvox & sphinx with asterisk? |
| 18:42.40 | seanbright | sorry, or, not & |
| 18:43.52 | slav3_kitten | btw i'm totally going to have my friend record a bunch of voice prompts for my asterisk box, her voice is amazing |
| 18:44.27 | slav3_kitten | later all, heading to a party |
| 18:44.47 | zxvff | "if you're a cat, press 1" |
| 18:45.36 | *** join/#asterisk phanohanover (phanohanov@modemcable096.110-80-70.mc.videotron.ca) |
| 18:45.50 | slav3_kitten | zxvff, seriously... how old are you? i thought this place was full of adults who could act like adults... |
| 18:46.04 | *** part/#asterisk phanohanover (phanohanov@modemcable096.110-80-70.mc.videotron.ca) |
| 18:46.21 | zxvff | no just adults who role play animals |
| 18:56.30 | *** join/#asterisk michael-i (~anonymous@204.11.231.178.static.etheric.net) |
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| 19:05.01 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 19:06.18 | volga629 | Is some body have experience with LDAP extensions ? |
| 19:13.48 | *** join/#asterisk serafie (~erin@nat/digium/x-rjzqaygpyerpgwav) |
| 19:15.17 | volga629 | I am trying set ldap extensions dial between each other, but it crashing asterisk when extension not match ldap search |
| 19:15.46 | [TK]D-Fender | volga629, What version of * are you running? |
| 19:18.59 | volga629 | asterisk Asterisk 1.8.11-cert1 |
| 19:19.26 | [TK]D-Fender | 2 releases behind... |
| 19:19.54 | [TK]D-Fender | I would search the tracker for issues regarding this |
| 19:20.38 | volga629 | yes, IMAP and SRTP LDAP is that why, I am trying make stable version for environment |
| 19:21.58 | volga629 | pbx01 kernel: asterisk[13649] general protection rip:3c45878480 rsp:4195ab28 error:0 |
| 19:28.04 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.172) |
| 19:32.35 | volga629 | I so bug reports, but it all related to earlier versions of ldap module |
| 19:33.45 | michael-i | Using 2 identical installations of 10.2.1, I've set up an encrypted IAX trunk between the two boxes. My first call works great but subsequent tries result in "packet decrypt failed." Any thoughts? |
| 19:35.03 | michael-i | The link may lose a few packets here&there but is the encryption state machine really so sensitive to that? |
| 19:35.19 | volga629 | And in log I see (AstExtension=*98) why is search this extensions where context set from-internal |
| 19:37.19 | [TK]D-Fender | micthought : current ver is 10.5.0 |
| 19:41.17 | michael-i | [TK]D-Fender: I checked through the change log up to 10.5 and didn't spot any real changes to iax. More curious about how sensitive encryption is to packet loss I guess in general. |
| 19:49.13 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 20:21.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:26.35 | dorphalsig | ~wiki |
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| 20:49.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 21:04.16 | uskerine | hi |
| 21:04.20 | uskerine | how can i force a given codec? |
| 21:04.30 | uskerine | and which is less intensive in CPU A-law or u-law? |
| 21:09.10 | [TK]D-Fender | they are the same |
| 21:09.20 | [TK]D-Fender | And you force it by only allowing it in your peer |
| 21:11.19 | uskerine | in sip.conf? |
| 21:12.07 | uskerine | do you know any softphone for android working fine with asterisk? |
| 21:12.49 | *** join/#asterisk crienzo (~crienzo@66-87-124-209.pools.spcsdns.net) |
| 21:13.28 | [TK]D-Fender | Sipdroid, 3CX has one, ICS has one built-in |
| 21:14.49 | uskerine | i am having a though time with sipdroid |
| 21:14.50 | uskerine | :( |
| 21:15.16 | uskerine | i can not get asterisk working fine |
| 21:17.47 | uskerine | any suggestion for troubleshooting audio issues? |
| 21:23.10 | [TK]D-Fender | uskerine: try actually describing and showing them. |
| 21:26.50 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 21:29.29 | leifmadsen | uskerine: you could try (probably not really public yet...) http://www.zoiper.com/downloads/beta/ |
| 21:30.03 | uskerine | should a couple of sipdroid's work over wifi? |
| 21:30.25 | leifmadsen | they probably "should" |
| 21:30.37 | *** join/#asterisk albertoandrade (~albertoan@186.206.16.212) |
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| 21:33.15 | *** part/#asterisk jsjc (~Adium@31.4.238.100) |
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| 21:34.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 21:39.44 | Sjors | is it possible to record calls in Asterisk, by making the Asterisk server an RTP gateway between a connected call and saving all traffic that goes through? |
| 21:40.01 | Sjors | or is this another layer of traffic that Asterisk does not know/care about? |
| 21:41.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:41.59 | [TK]D-Fender | Sjors: this is what * normally does. |
| 21:42.05 | [TK]D-Fender | Sjors: * is not a SIP proxy |
| 21:42.09 | [TK]D-Fender | ~b2bua |
| 21:42.09 | infobot | it has been said that b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 21:42.21 | [TK]D-Fender | Sjors: And yes you can have * record your calls |
| 21:42.36 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 21:43.30 | Sjors | [TK]D-Fender: ahh, so the RTP traffic already passes through it? |
| 21:43.31 | Sjors | cool |
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| 22:25.40 | *** join/#asterisk gusto (~gusto@tmo-082-81.customers.d1-online.com) |
| 22:26.23 | gusto | hi |
| 22:26.29 | gusto | what's up? |
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| 22:55.46 | sarsaeol | anyone got tips on troubleshooting OWA? |
| 22:55.58 | sarsaeol | some useful documantation perhaps? |
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| 23:20.34 | *** join/#asterisk lanken (47a272bd@gateway/web/freenode/ip.71.162.114.189) |
| 23:21.14 | lanken | hi all! My company is working on getting a business contract for cell phones (single payer, contracts linked to company's Tax ID#) |
| 23:21.31 | lanken | I want to do background research. is there a forum for this type of thing? |
| 23:21.39 | lanken | (I'm just not sure what to google) |
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