| 00:03.51 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 00:04.59 | Rahail | HI i need little idea i would like to know how can achive this after every single call it need to send reset to the channel ( I am using some sort of 3g modem) |
| 00:07.23 | Penguin | So how do I configure asterisk 11 with multiple google voice accounts so that it behaves the same way it did in 1.8? I want incoming calls to go to extension myusername@gmail.com instead of extension s. |
| 00:08.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:38.02 | [TK]D-Fender | ~book |
| 00:38.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:38.06 | [TK]D-Fender | yoshie902a, ^ |
| 00:38.30 | yoshie902a | thanks |
| 00:43.21 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 00:45.56 | teloniusz | Hello. Strange problem. I'm inspecting "triangle" configuration: asterisk A -> asterisk B -> asterisk C, directmedia on A<->B and B<->C trunks |
| 00:48.35 | teloniusz | I'm checking with wireshark what goes between them in SIP headers. And it looks like this: INVITE A -> B with SDP pointing to A:someport, INVITE B -> C with SDP A:someport, 200 OK C -> B with SDP C:otherport and 200 OK B -> A with SDP B:anotherport |
| 00:49.36 | teloniusz | so... it looks like "central" asterisk tries to make RTP from A to C go directly |
| 00:50.04 | [TK]D-Fender | it will |
| 00:50.17 | teloniusz | but doesn't send back C:otherport to A |
| 00:50.21 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-26-148.dynamic.qsc.de) |
| 00:50.22 | [TK]D-Fender | if you allowed A & C to use direct media they will reinvite between each other |
| 00:51.12 | teloniusz | the effect: A sends RTP to B, no RTP between B and C, no media, connection is broken after some 30 secs because of no media |
| 00:52.56 | [TK]D-Fender | I'd start providing debug... |
| 00:53.51 | teloniusz | from B, I presume? |
| 00:54.19 | teloniusz | hm, there is one differencje between the three, no directrtpsetup enabled on C |
| 00:55.03 | teloniusz | I'll try enabling it on C, maybe it'll help... |
| 00:57.04 | [TK]D-Fender | It has to be allowed on all sides |
| 00:57.33 | Penguin | And your Dial() options must not require asterisk to stay in the media path, or directmedia won't work. |
| 01:11.39 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 01:11.39 | *** mode/#asterisk [+o file] by ChanServ |
| 01:22.39 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 01:22.39 | *** mode/#asterisk [+o file] by ChanServ |
| 01:26.27 | yoshie902a | Can someone help me figure out the system requirements (hardware) I would need? I donât want anything too big, It would be for my home to play with. one line active line at a time, but 10 numbers in the system. I would want to set it up to record conversations |
| 01:29.54 | Penguin | Got an Atom 1GHz box with 256M RAM? |
| 01:31.09 | Penguin | Anything Pentium 3 or higher should be acceptable for your situation. |
| 01:32.26 | yoshie902a | is Atom the CPU? |
| 01:32.55 | yoshie902a | Iâm looking for something small that does not make a lot of noise or take a lot of energy |
| 01:33.27 | Penguin | Yes, Atom is the CPU. Most are fanless. |
| 01:33.31 | *** join/#asterisk epinky (~epinky@unaffiliated/trismegisto) |
| 01:34.09 | yoshie902a | would I need to build the box or are ther boxes prebuilt that are cheap |
| 01:34.21 | Penguin | There are cheap ones prebuilt. |
| 01:34.29 | Penguin | Are you in the US? |
| 01:34.34 | yoshie902a | yes |
| 01:34.43 | Penguin | I'd recommend looking on ebay. |
| 01:37.25 | [TK]D-Fender | You could pick up a 100$ netbook and it'll come with a free battery backup |
| 01:38.16 | epinky | I have this problem, when I answer up the phone I say hello, but I cannot hear the voice of the other endpoint after a few seconds, the same happens the other end, the caller continues to hear the ringing even though the other end has answered already, what could be the cause? |
| 01:38.27 | yoshie902a | I wanted to avoid running a laptop all the time. I was thinking something small and quite . would this work? http://www.ebay.com/itm/DELL-OPTIPLEX-FX170-THIN-CLIENT-ATOM-1-60GHZ-1GB-DDR2-1GB-FLASH-DRIVE-GRADE-A-/121486873861?pt=US_Thin_Clients&hash=item1c492e9505 |
| 01:39.41 | Penguin | Do you want something with a read hard drive? |
| 01:39.48 | Penguin | That says 1G flash |
| 01:41.07 | Penguin | Maybe this? http://www.ebay.com/itm/Intel-Atom-E3815-1-46GHz-Fanless-NUC-Barebone-Kit-DE3815TYKH0E-Single-Pack-/111421232255?pt=Desktop_PCs&hash=item19f139147f |
| 01:41.19 | Penguin | It seems like it accepts a 2.5" hard drive. |
| 01:41.52 | [TK]D-Fender | It's not like an Atom laptop is really going to take much more than any of those |
| 01:42.05 | [TK]D-Fender | And it will have a battery backup, bulting in console for maintenance, etc. |
| 01:43.05 | Penguin | Laptops are pretty okay servers for those very reasons. |
| 01:43.11 | *** join/#asterisk Maxxed (~root@ec2-107-20-252-119.compute-1.amazonaws.com) |
| 01:43.16 | yoshie902a | the barebones after I add a harddrive and memory would cost $250, |
| 01:43.26 | yoshie902a | maybe a laptop if itâs cheaper |
| 01:44.02 | Penguin | Do you have to have fanless? I have a Dell ultra small form factor PC that I'm using now. |
| 01:44.23 | yoshie902a | Iâd prefer it b/c of the extra noise |
| 01:45.12 | Penguin | Here's a perfect example: http://www.ebay.com/itm/Dell-Optiplex-760-SFF-Intel-Celeron-440-2-00GHz-2G-mem-80-GB-HD-DVD-DRIVE/271631374065?_trksid=p2047675.c100010.m2109&_trkparms=aid%3D555012%26algo%3DPW.MBE%26ao%3D1%26asc%3D27538%26meid%3Daaa93caefee149f692b1c1fa29a5ef03%26pid%3D100010%26prg%3D11353%26rk%3D4%26rkt%3D24%26sd%3D151488493436 |
| 01:45.26 | Penguin | $50 ready to run. Comes with hard drive and RAM in it. |
| 01:47.19 | yoshie902a | good option |
| 01:47.24 | Penguin | Here's one for $46 but you need to add your own hard drive: http://www.ebay.com/itm/Dell-Optiplex-745-Intel-VPro-1-8GHz-2GB-RAM-w-AC-Adapter/121492417381 |
| 01:47.39 | Penguin | That's the model I have. Dell Optiplex 745 |
| 01:48.08 | yoshie902a | only thing itâs bigger than I wanted, Iâm really looking for something like the mac mini, but cheaper |
| 01:48.39 | yoshie902a | whatâs the issue with the flash drive? Itâs still permanent storage, isnât it? |
| 01:48.39 | Penguin | I got a 10000 RPM 2.5" hard drive on a 3.5" heatsink adapter for like $30 to put in mine. |
| 01:49.15 | Penguin | You can use a flash drive, but you should tune the system to reduce the number of disk writes. |
| 01:49.26 | Penguin | Logging should be minimal or off-system. |
| 01:50.48 | Penguin | I have been running a 4G flash in my Neoware 800 MHz box, and I've run out of space. That one only had 1G flash, so I can't imagine how cramped that would be. |
| 01:51.01 | yoshie902a | true |
| 01:51.09 | yoshie902a | I should have a real drive |
| 01:51.21 | yoshie902a | how big is Astrisk as a program? |
| 01:51.34 | Penguin | That's why I've been moving to this new Optiplex with space in it for a real hard drive. |
| 01:51.39 | Penguin | 15MB maybe? |
| 01:51.46 | Penguin | Asterisk itself is small. |
| 01:52.17 | Penguin | Then you've got sound files and other things that take up space. |
| 01:53.09 | yoshie902a | I figure the biggest item will be the recording of calls |
| 01:53.20 | yoshie902a | any idea how much space it needs per minute? |
| 01:53.43 | Penguin | Using WAV, mine use 1MB for every 10 minutes. |
| 01:54.01 | Penguin | WAV, not wav |
| 01:54.58 | yoshie902a | whatâs the difference between WAV and wav? |
| 01:55.19 | Penguin | It's a different format. |
| 01:56.05 | Penguin | somefile.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz |
| 01:56.45 | Penguin | somefile.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 01:57.09 | Penguin | So PCM vs GSM |
| 01:57.35 | yoshie902a | interesting, thanks! I never knew there was a difference |
| 01:57.43 | Penguin | In asterisk, there is. |
| 01:58.08 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 01:58.08 | Penguin | Since Linux file systems are case sensitive, asterisk uses WAV for a different format from wav. |
| 01:58.17 | yoshie902a | good to know! |
| 02:01.37 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 02:01.46 | yoshie902a | what do you think of a SSD drive? |
| 02:01.52 | yoshie902a | same issue as the flash? |
| 02:02.09 | [TK]D-Fender | SSD's have garbage collection, wear leveling, etc |
| 02:02.53 | Penguin | SSDs have the ability to take more writes than what we refer to as flash drives. |
| 02:03.28 | yoshie902a | so better than flash, but not as good as a mechanical drive? |
| 02:05.59 | [TK]D-Fender | perhaps better than mechanical |
| 02:07.59 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 02:09.47 | *** join/#asterisk stevesmename (~stevesmen@71-223-80-102.phnx.qwest.net) |
| 02:11.17 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 02:14.56 | yoshie902a | I have this.. http://www.toshiba.com/us/computers/laptops/satellite/C850/C855D-S5320, do you think I would be able to install linux on it and get it working? |
| 02:15.38 | yoshie902a | I was actually going to return it, and get something else, but if it works, maybe I should just use it |
| 02:16.00 | yoshie902a | I just feel like running it all the time would be bad for some reason |
| 02:17.07 | Penguin | I run my laptop all the time. |
| 02:17.23 | Penguin | <PROTECTED> |
| 02:17.35 | yoshie902a | ok, I think thatâs what Iâll do then |
| 02:17.45 | yoshie902a | Ubuntu? |
| 02:17.54 | Penguin | I run Arch. |
| 02:18.12 | yoshie902a | why arch? |
| 02:18.40 | Penguin | I tried it sometime back around 2006-2007 and liked it. |
| 02:19.13 | Penguin | Although my new Dell box I'm using for Asterisk is running CentOS. |
| 02:19.44 | yoshie902a | I heard the two most popular were ubuntu and centOS |
| 02:20.51 | yoshie902a | does arch use rpm or apt-get? |
| 02:20.56 | Penguin | no |
| 02:26.26 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-26-148.dynamic.qsc.de) |
| 02:30.35 | yoshie902a | I think Iâm going to run Asterisk on Ubuntu, Iâm more used to is, but only from my experience with AWS. |
| 02:30.58 | Penguin | The OS you choose doesn't make a lot of difference to asterisk. |
| 02:31.05 | Penguin | Use whatever you want to use. |
| 02:35.00 | *** join/#asterisk KNERD (~KNERD@23.227.189.101) |
| 03:05.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:37.45 | yoshie902a | I have a desktop computer I use all the time, would it be bad to install asterisk on it for my daily use? |
| 03:38.03 | yoshie902a | should I have asterisk on a completely seperate computer? |
| 03:38.14 | WIMPy | Only if it's bad/old. |
| 03:40.03 | yoshie902a | itâs a nice computer |
| 03:40.22 | yoshie902a | xeon process 16gb ram etc |
| 04:02.25 | [TK]D-Fender | massive overkill |
| 04:45.05 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 04:54.00 | *** join/#asterisk timahvo1 (~rogue@196.216.71.122) |
| 05:05.50 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 05:06.36 | Rahail | HI i need help from expart please some one.. I need to build some sort of script where after every single call hang up it need to send 3 command IE:/usr/sbin/asterisk -rx "iax2 reload" etc simular to this |
| 05:07.03 | Penguin | Why would you need to reload iax2 after every call? |
| 05:07.16 | Rahail | its just example |
| 05:07.28 | Penguin | Use extension h and hangup handlers. |
| 05:07.46 | Penguin | ~asteriskwiki |
| 05:07.46 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
| 05:07.57 | Rahail | i am using 3g modem for call and after each call hang up i want to reset the modem ie /usr/sbin/asterisk -rx "dongle reset donglename" |
| 05:08.54 | Rahail | Penguin do you think you can help me on this please .. if you have time |
| 05:10.07 | [TK]D-Fender | <Penguin> Use extension h and hangup handlers. |
| 05:10.16 | [TK]D-Fender | Rahail, He just GAVE you the way to do it |
| 05:10.37 | [TK]D-Fender | Rahail, So go set up your "h" Asterisk Standard Extension. |
| 05:11.06 | Rahail | [TK]D-Fender how can you give me 1 example please i am not that good on this every week learning new thing |
| 05:11.20 | [TK]D-Fender | GO READ |
| 05:11.37 | [TK]D-Fender | You have all the keywords to raed up on these |
| 05:11.42 | [TK]D-Fender | read* |
| 05:11.55 | yoshie902a | I reviewed a number of tutorials on asterisk and one mentioned that if I use Virtualbox that it would create a memory leak and run slower, any body agree or disagree? |
| 05:12.13 | [TK]D-Fender | yohVirtualBox is the wrose option for this |
| 05:12.17 | [TK]D-Fender | worst* |
| 05:13.43 | yoshie902a | yoh? |
| 05:13.52 | yoshie902a | yoshie? |
| 05:14.00 | [TK]D-Fender | yoshie902a, VirtualBox is the worst option for this |
| 05:14.22 | Rahail | [TK]D-Fender i did lot of reading there are some thing out off my knowledge this why asking expert help |
| 05:14.43 | [TK]D-Fender | "h" Asterisk Standard Extensio <- |
| 05:14.48 | [TK]D-Fender | GOOGLE IT |
| 05:15.03 | [TK]D-Fender | This doesn't require "expert" This is the basics |
| 05:17.34 | yoshie902a | I got a windows machine, that xeon, 16gb, etc that I thought I could run Asterisk in the background. if not Virtualbox, any other method? or should I just install linux on an old laptop and run asterisk from there? preferrably like to limit the electricity usuage by limited my equipment |
| 05:19.04 | [TK]D-Fender | Almost any other method |
| 05:19.35 | yoshie902a | is there another option like virtualbox where I can run it from windows 7? |
| 05:19.55 | [TK]D-Fender | bare metal linux, ESXi, KVM. Xen is a slightly lower rating from what I've heard |
| 05:23.13 | Rahail | [TK]D-Fender someting liek this http://the-asterisk-book.com/1.6/besondere-extensions.html |
| 05:23.16 | Rahail | are you referring |
| 05:23.34 | [TK]D-Fender | yes |
| 05:23.57 | Rahail | how will it rember to put +1 on every call each hour |
| 05:26.15 | yoshie902a | with ESXi or KVM, would I need to reconfigure my windows machine or can I run them without redoing everything? |
| 05:26.27 | [TK]D-Fender | <Rahail> how will it rember to put +1 on every call each hour <- what does this even mean? |
| 05:26.38 | [TK]D-Fender | What is putting "+1" on a CALL mean? |
| 05:27.29 | Rahail | if i do set 1 this time after call hang up in next call it need to set 2 next call it need to set 3 like that |
| 05:27.29 | [TK]D-Fender | yoshie902a, Only thing UNDER windows that might be viable is HyperV or perhaps VMWare WorkStation |
| 05:27.52 | Penguin | And what does that have to do with exten h and hangup handler? |
| 05:27.59 | [TK]D-Fender | yoshie902a, Barring that FORGET WINDOWS. Asterisk is NOT Windows app. |
| 05:28.08 | Rahail | i need way to control that numbers |
| 05:28.20 | Penguin | You want to increase a counter? |
| 05:28.24 | [TK]D-Fender | Rahail, Go set something that * will remember |
| 05:28.29 | Rahail | yes |
| 05:28.44 | Penguin | I would suggest writing a value to the asterisk database. |
| 05:28.46 | Rahail | upt to 20 after 20 then start again from 1 |
| 05:28.47 | [TK]D-Fender | Rahail, Either a GLOBAL VARIABLE or an ASTDB VALUE |
| 05:28.50 | yoshie902a | would you recommend a direct linux install on a weaker laptop or a hyperV install on windows? |
| 05:29.09 | [TK]D-Fender | Direct install is always preferrable |
| 05:29.10 | Penguin | "weaker laptop" |
| 05:29.33 | Penguin | If it's over 500 MHz and more than 128M RAM, it's fine. |
| 05:29.44 | [TK]D-Fender | For a single-call setup? Sure |
| 05:29.59 | Penguin | That'll handle at least three at a time. |
| 05:30.13 | yoshie902a | great, thanks! |
| 05:30.23 | Penguin | That's at least three times more than the requirement. |
| 05:30.24 | yoshie902a | I very much apprecaite all the advise |
| 05:30.37 | yoshie902a | advice |
| 05:32.17 | Rahail | i think i would have to build maby some sort of php or other program and read write data from db . |
| 05:32.21 | Rahail | only if i know who can do that for me |
| 05:32.49 | Penguin | totally irrelevant |
| 05:32.59 | Penguin | The asterisk database is BUILT IN. |
| 05:33.07 | Penguin | Asterisk already knows how to talk to it. |
| 05:33.14 | Penguin | core show applications like db |
| 05:33.16 | Penguin | Go read. |
| 05:33.28 | Penguin | core show functions like db |
| 05:33.32 | Penguin | Go read that too. |
| 05:33.53 | [TK]D-Fender | Rahail> i think i would have to build maby some sort of php or other program and read write data from db . <- not from what you told us |
| 05:34.06 | [TK]D-Fender | Rahail, and I keep giving you the things to look for. |
| 05:34.19 | Rahail | i am looking but getting more confuse |
| 05:34.42 | Penguin | Write some lines of dialplan to write to and read from the asterisk db. |
| 05:35.07 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:39.44 | Penguin | Go to your asterisk CLI and run "core show function DB" and read what it says. |
| 05:40.15 | Rahail | Penguin not eassy for me to understand i am reading every where i find info's but not getting in my head |
| 05:40.33 | Penguin | "core show function DB" |
| 05:40.37 | Rahail | can I pm you ? if is ok |
| 05:40.42 | Penguin | No. |
| 05:40.44 | Penguin | "core show function DB" |
| 05:40.59 | Penguin | You do that and then we'll talk more. |
| 05:41.38 | Rahail | the thing i need i am 100% my knowledge not enough to build it |
| 05:41.42 | Penguin | "core show function DB" and tell me what it says. |
| 05:41.55 | Rahail | ifn about funciton db |
| 05:42.01 | Penguin | What does it say? |
| 05:42.13 | Rahail | info about functions 'db' |
| 05:42.26 | Rahail | syntax argurments see also dbdel, deb delte etc |
| 05:42.33 | [TK]D-Fender | Rahail, that saves/gets values from the internal Asterisk DB |
| 05:42.45 | [TK]D-Fender | Rahail, Use it to save a value for your COUNTER |
| 05:42.54 | [TK]D-Fender | Rahail, This means go TRY IT NOW. |
| 05:43.33 | [TK]D-Fender | Rahail, Go set a value. Then look for it. Then get it in your dialplan. |
| 05:43.45 | [TK]D-Fender | Rahail, Once you can do this then you can increment it when needed |
| 05:44.27 | Rahail | once it reach to 20 how do i start from 1 again |
| 05:44.37 | [TK]D-Fender | Set it to 1 |
| 05:44.52 | Penguin | ExecIf() can do that. |
| 05:45.53 | Penguin | or GotoIf() in combination with a Set() |
| 05:46.14 | [TK]D-Fender | or just IF() |
| 05:46.28 | [TK]D-Fender | within a set |
| 05:46.41 | Penguin | Several options await! |
| 05:47.36 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 05:47.52 | Rahail | not getting my head u guys will lough |
| 05:47.59 | Rahail | ~pastbin |
| 05:48.05 | Rahail | !pastebin |
| 05:48.11 | Rahail | ?pastebin |
| 05:48.35 | [TK]D-Fender | Rahail, You don't seem to have spent even 5 minutes on this. |
| 05:49.07 | Rahail | i am just did 1 thing just want know if i am heading to right direction beforee the db |
| 05:49.25 | [TK]D-Fender | Rahail, Try more |
| 05:49.38 | [TK]D-Fender | Rahail, You're not goingt to learn until you see it |
| 05:49.41 | [TK]D-Fender | Go try. Go look. |
| 05:49.59 | Rahail | i wish i had confidence like your |
| 05:50.15 | [TK]D-Fender | Rahail, You aren't giving yourself the opportunity to gain confidence. |
| 05:50.30 | [TK]D-Fender | Rahail, You are just asking for others to literally do your programming for you |
| 05:51.07 | Penguin | I'll probably do part of it. :/ |
| 05:51.12 | Rahail | true but this not for eveyr one cup of tea |
| 05:51.18 | Rahail | specialy me |
| 05:51.21 | Penguin | But I've got to see some effort. |
| 05:51.27 | [TK]D-Fender | Rahail, Then don't use Asterisk |
| 05:51.48 | [TK]D-Fender | Rahail, The dialplan is 90% of Asterisk. It is ALL of the brains of processing your calls |
| 05:52.04 | [TK]D-Fender | Rahail, If you're not going to learn it then you are headed for a dead-end |
| 05:53.13 | Rahail | i cant stop using asterisk :( no option for me (i will need help from expart to build this) however i dont think i can do it.. |
| 05:53.17 | Rahail | still trying |
| 05:54.15 | Penguin | I've got a command ready to send, but I need to see some effort that you're trying. |
| 05:54.16 | [TK]D-Fender | Option is to actually sit down with the book and learn it. |
| 05:54.27 | Penguin | Show me a command that you're trying. |
| 05:54.53 | Rahail | <PROTECTED> |
| 05:55.12 | Penguin | How does that relate to the what we've been talking about? |
| 05:55.19 | Penguin | I don't see anything about the asterisk db. |
| 05:55.39 | [TK]D-Fender | rehSet also does NOT call some external command |
| 05:55.44 | Penguin | And you need System() not Set(). |
| 05:55.46 | [TK]D-Fender | Rahail, Set also does NOT call some external command |
| 05:55.52 | Rahail | ouch |
| 05:56.27 | [TK]D-Fender | Rahail, "core show application set" <- You don't seem to have read the instruction that say what the app even does |
| 06:00.08 | Rahail | 1st i get the call then i use h extenions with DB then I use system to exucte the externel command |
| 06:00.17 | *** join/#asterisk stasdizzi (~stasdizzi@143-27-133-95.pool.ukrtel.net) |
| 06:01.01 | Penguin | Set(DB()=${IF()}) |
| 06:01.08 | Penguin | There's your hint. |
| 06:01.18 | Penguin | This is for a counter. |
| 06:03.43 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 06:04.19 | Rahail | i use set 1st then h,System |
| 06:04.20 | Rahail | right |
| 06:05.11 | [TK]D-Fender | Rahail, What order do YOU think they should be in? |
| 06:05.21 | Rahail | 1st counter |
| 06:05.23 | Rahail | then sytem |
| 06:05.28 | [TK]D-Fender | Rahail, Then do it |
| 06:05.35 | Penguin | I would do it the other way. |
| 06:05.42 | Rahail | which way |
| 06:05.42 | [TK]D-Fender | I would do it both ways |
| 06:06.20 | [TK]D-Fender | loads up the Innuendo Cannon... |
| 06:06.51 | ChannelZ | kaBOOM |
| 06:07.26 | Rahail | is there way to limit it by hour |
| 06:07.42 | Rahail | or i need it more advance filter |
| 06:07.54 | Penguin | There's always a way if you're willing to learn how to do it. |
| 06:08.28 | [TK]D-Fender | Rahail, Its your dialplan go make it check those other factors |
| 06:09.14 | [TK]D-Fender | Rahail, I'm not sure you have fully grasped that dialplan = programming. It'll do what you tell it to do. You want conditions? You want to look things up to make those decisions? then do it. |
| 06:10.07 | Rahail | i need to add 2 conditon call intervel and total duration per hour If it reach that duration not to run the command this hour |
| 06:10.11 | Rahail | so much to learn |
| 06:10.13 | [TK]D-Fender | Rahail, Time to sit down with the book and learn the basics of dialplan flow, applications, variables, functions and expressions. |
| 06:14.32 | Rahail | Penguin i am still bit confuse on the SEt part if option |
| 06:14.57 | [TK]D-Fender | Rahail, Go learn how variables and functions work. |
| 06:25.38 | Rahail | trying |
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| 06:35.23 | Rahail | [TK]D-Fender i dont think i can do this can you guide soem one who can do it free or paid eather way please i go tno opitons |
| 06:38.15 | [TK]D-Fender | Rahail, It's far too late for me to take this on tonight, perhaps in the morning. |
| 06:39.23 | Rahail | what time in morning you will be here.. so I can bug you again .. |
| 06:39.50 | [TK]D-Fender | 8 hours |
| 06:40.11 | Rahail | means 9.40AM Estern time |
| 06:40.13 | Rahail | cool thank you |
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| 09:04.22 | wasanzy | hello |
| 09:05.20 | wasanzy | I have used Confbridge to setup conference, but the difficult am having is, I want confbridge to call an external application every 10mins whiles the user is in the conference |
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| 10:10.12 | epinky | I have this problem, when I answer up the phone I say hello, but I cannot hear the voice of the other endpoint after a few seconds, the same happens the other end, the caller continues to hear the ringing even though the other end has answered already, what could be the cause? |
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| 10:37.55 | eirirs | epinky: you have any SIP functions enabled on your router? |
| 10:38.04 | eirirs | epinky: you behind NAT ? |
| 10:39.24 | epinky | eirirs: it's intranet, pbx to pbx communication, one pbx per vlan throughout multilayer switches, no firewall or nat in these networks |
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| 10:40.27 | RadJackson | Hello , is there a way to have the number of calls running on a trunk sip ? |
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| 11:05.54 | teloniusz | RadJackson: asterisk -r -x 'core show channels' |grep -c 'SIP/trunk_name' ? |
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| 11:58.31 | tosar | We've got two ips from our SIP provider. How do we configure sip.conf to use them for outgoing calls? |
| 11:58.39 | tosar | any ideas will be appreciated |
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| 12:12.15 | Eric-K | do they have a domain with DNS SRV available tosar |
| 12:12.52 | tosar | no |
| 12:13.01 | tosar | they just provided two ips |
| 12:13.06 | Eric-K | hm ok |
| 12:13.17 | tosar | said one was for signaling and the other for media |
| 12:13.35 | tosar | we couldn't find much info online |
| 12:13.47 | tosar | if anyone knows some resources, we would appreciate |
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| 12:17.19 | Eric-K | Well if one is for signalling, I don't think you'd have to configure the media IP. |
| 12:17.36 | Eric-K | Try setting up a call using only the signalling IP and see if RTP automatically connects to the media IP. |
| 12:17.56 | Eric-K | You can also check the SIP packages, it should contain media information including the IP. |
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| 12:21.08 | tosar | thanks, we'll try again and will post results |
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| 12:21.38 | Eric-K | Sure, no problem. If you check the SIP signalling, the packets should show SDP with media information. |
| 12:21.48 | Eric-K | I bet that includes the IP of the media server :) |
| 12:22.07 | Eric-K | Just connect to the signalling IP in your sip.conf |
| 12:22.40 | tosar | do we need to set the ip to outbound proxy? |
| 12:22.56 | Eric-K | The signalling IP, yes, give that a try. |
| 12:23.07 | tosar | will do, thanks |
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| 12:31.49 | tosar | we got this error: |
| 12:31.54 | tosar | ERROR[29268]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo( |
| 12:32.11 | tosar | when using the ip address as outbound proxy |
| 12:33.17 | Eric-K | Are you sure you formatted the IP correctly? No extra characters? |
| 12:33.48 | tosar | yes |
| 12:34.35 | Eric-K | Could be many things in the config leading to a problem like this I think. |
| 12:34.45 | Eric-K | Which Asterisk version? |
| 12:34.49 | tosar | 13 |
| 12:34.57 | tosar | should we be able to ping the provided ips |
| 12:34.58 | tosar | ? |
| 12:35.38 | Eric-K | I would expect they would respond to ping, yes. |
| 12:35.57 | tosar | ok |
| 12:36.03 | tosar | we'll start with that |
| 12:36.08 | tosar | might be something network related |
| 12:37.13 | Eric-K | Indeed |
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| 12:38.29 | duff_ | Hello guys, I need some help with asterisk + sendmail |
| 12:38.44 | duff_ | I need it to be forced to use port 587 |
| 12:38.45 | Eric-K | tosar I'd try something like this in your sip.conf; http://pastebin.com/raw.php?i=rRt7XxB5 |
| 12:38.55 | duff_ | anyone can help me out |
| 12:38.58 | Eric-K | quickly wrote that from scratch, might have a mistake |
| 12:39.14 | Eric-K | You should ofcourse tweak it to your needs. |
| 12:41.25 | duff_ | Eric-K, can you help me out ? |
| 12:41.45 | Eric-K | I'm sorry, I'm not familiar with sendmail. |
| 12:42.10 | duff_ | :/ |
| 12:42.31 | duff_ | Im setting up a voicemail, and I need all audios/msg to be send via email |
| 12:42.50 | duff_ | and Im having a hard time using sendmail, the log shows connection refused all the time |
| 12:43.26 | Eric-K | I tried a Google search, maybe this can help? http://pbxinaflash.com/community/index.php?threads/verizon-smtp-port-587.5420/ |
| 12:44.13 | Eric-K | Or Google: asterisk AND sendmail AND 587 |
| 12:45.00 | duff_ | okay |
| 12:45.11 | duff_ | I already did this that is described on this link |
| 12:45.13 | duff_ | didnt work |
| 12:45.22 | duff_ | but Im going to keep searching. thanks Eric-K |
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| 12:47.37 | Eric-K | Good luck :) |
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| 13:46.16 | yoshie902a | Hi Iâm trying to install the FreePBX distro from a USB drive and keep getting a kickstart error. Any good instructions on how to get it installed using a USB? I donât have any DVDâs I can use |
| 13:47.27 | yoshie902a | I found this http://www.objectpac.com/2012/11/19/of-centos-and-freepbx-part-1-bootable-usb/, but it shows the freepbx does not get installed |
| 13:47.38 | yoshie902a | wondering if there is a better solution |
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| 13:56.06 | [TK]D-Fender | ~freepbx |
| 13:56.06 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:56.08 | [TK]D-Fender | ^^^ |
| 13:56.31 | yoshie902a | thanks, sorry about that |
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| 14:14.07 | malcolmd | Penguin: you're probably going to have the best luck trying to use "google" instead of "google-v1" there's no direct mapping |
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| 15:12.13 | Maxxed | im having a weird issue.. in my gtalk.conf, i have two accounts, but they are not falling into the right context on inbound calls |
| 15:12.16 | Maxxed | context=google-in |
| 15:12.31 | Maxxed | it keeps failing back to the default context |
| 15:12.34 | Maxxed | any ideas? |
| 15:13.21 | Maxxed | == Starting Gtalk/+17138752545-63bd at ,mygmailaccount1234@gmail.com,1 failed so falling back to exten 's' == Starting Gtalk/+17138752545-63bd at ,s,1 still failed so falling back to context 'default' |
| 15:13.59 | WIMPy | That looks like the answer to your question. |
| 15:14.20 | Maxxed | Ey? |
| 15:14.43 | Maxxed | I have the correct context in the dial plan, but it seems to just fail back to the default, idk why |
| 15:15.17 | [TK]D-Fender | Asterisk is telling you that you don't have a match there |
| 15:15.23 | [TK]D-Fender | And I doubt it's lying. |
| 15:15.33 | [TK]D-Fender | I'd look VERY closely at your configs again... |
| 15:15.41 | Maxxed | I get that, but I should have a match.. let me check again, maybe I'm missing something |
| 15:15.55 | [TK]D-Fender | It's telling you exactly that. |
| 15:16.27 | Maxxed | The inbound conext it pretty stright forward.. |
| 15:17.01 | Maxxed | ; * From Google |
| 15:17.01 | Maxxed | [google-in] |
| 15:17.01 | Maxxed | exten => s,1,Answer() |
| 15:17.01 | Maxxed | exten => s,n,Wait(2) |
| 15:17.01 | Maxxed | exten => s,n,SendDTMF(1) |
| 15:17.03 | Maxxed | exten => s,n,Dial(SIP/2600&SIP/2601, 180) |
| 15:17.06 | Maxxed | exten => s,h,Hangup |
| 15:17.36 | [TK]D-Fender | ~pb |
| 15:17.36 | 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. |
| 15:17.38 | [TK]D-Fender | ^^^^ |
| 15:17.40 | [TK]D-Fender | do not flood in here |
| 15:17.46 | Maxxed | Sorry |
| 15:17.57 | [TK]D-Fender | Second, nowhere in that debug line does it mention this context |
| 15:18.01 | wasanzy | how do I exit conference after specified minutes? |
| 15:18.09 | wasanzy | am using Confbridge |
| 15:19.48 | Maxxed | Yeah, I know thats the part thats confusing me, the context is set in the gtalk.conf |
| 15:19.59 | Maxxed | I would expect it to say, google-in context is jacked or something |
| 15:20.00 | [TK]D-Fender | Maxxed: You should probably be showing us.... |
| 15:20.11 | [TK]D-Fender | [10:13]Maxxed== Starting Gtalk/+17138752545-63bd at ,mygmailaccount1234@gmail.com,1 failed so falling back to exten 's' == Starting Gtalk/+17138752545-63bd at ,s,1 still failed so falling back to context 'default' |
| 15:20.17 | [TK]D-Fender | And it clearly isn't looking there |
| 15:20.17 | Maxxed | yep, let me get tthe configs in paste bin |
| 15:20.30 | Maxxed | I have to be missing something silly |
| 15:20.44 | [TK]D-Fender | Almost certainly. |
| 15:20.53 | [TK]D-Fender | Let's take a look |
| 15:23.40 | Maxxed | http://pastebin.com/6kUJG6sv |
| 15:24.07 | Maxxed | the extentions, gtalk, and jabber conf. i suspect something is wrong that i'm over looking |
| 15:26.34 | Maxxed | oh, btw, the default context, i placed google-in just to get it to work, its not ideal that it be that way |
| 15:27.00 | [TK]D-Fender | Looks like you've masked what it was actually looking for in your debug line there. |
| 15:27.02 | Maxxed | default should be empty, its just rigged that way to keep things working |
| 15:27.11 | [TK]D-Fender | So we can't prove what it thinks it's matching |
| 15:27.18 | Maxxed | yeah, i didnt sensor out anything other than the passwd's |
| 15:27.22 | Maxxed | just a sec |
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| 15:28.27 | Maxxed | http://pastebin.com/raw.php?i=h90uX8a1 |
| 15:28.45 | Maxxed | i just called in via my mobile, that is what asterisk was doing ^ |
| 15:29.17 | Maxxed | is the context supose to start with something like s,jivebarandlounge@gmail.com,1 ? |
| 15:29.51 | Maxxed | exten => s,1,Answer() |
| 15:30.12 | Maxxed | er, exten => s,jivebarandlounge@gmail.com,1,Answer() ? |
| 15:30.35 | Maxxed | that makes no sense :/ |
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| 15:31.42 | [TK]D-Fender | it's missing a context there |
| 15:31.53 | [TK]D-Fender | And the only section missing one is [general] |
| 15:32.01 | [TK]D-Fender | Which leads me to believe it isn't amtching any of the others |
| 15:32.08 | [TK]D-Fender | For whatever reason |
| 15:32.13 | [TK]D-Fender | So I'd go set it there as well. |
| 15:32.19 | Maxxed | shouldnt it match on the [google-in |
| 15:32.21 | Maxxed | ] |
| 15:32.51 | Maxxed | on a inbound call, the should start at [google-in] right? |
| 15:32.59 | [TK]D-Fender | I'm talking about GTALK |
| 15:33.08 | [TK]D-Fender | THAT is not matching the sections you have defined |
| 15:33.13 | Maxxed | yep |
| 15:33.15 | Maxxed | thats what it seems |
| 15:33.23 | [TK]D-Fender | So it's using the [general] settings... which you didn't define. |
| 15:33.33 | Maxxed | it is using the general settings? |
| 15:34.11 | [TK]D-Fender | That's what it appeasr to be doing |
| 15:34.15 | [TK]D-Fender | Go change it and see |
| 15:37.14 | Maxxed | i comented out everything [genral] in the gtalk.conf, includiong the line [general], same behavior |
| 15:37.51 | [TK]D-Fender | Did you ADD a "context=" line for [general] to see that it IS taking effect? |
| 15:40.14 | wasanzy | any help? |
| 15:40.29 | Maxxed | ah hah |
| 15:40.30 | Maxxed | http://pastebin.com/raw.php?i=UcWtUwNH |
| 15:40.46 | Maxxed | so, i can have only one conext for gtalk? |
| 15:41.10 | Maxxed | i have two accounts, i would have thought i could have a context for inbound on one account, and a seperate conect for the other account |
| 15:41.38 | Maxxed | if account1 is called, ring ext 2610, if account2 is called, ring ext 2600 |
| 15:42.14 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 15:42.54 | *** join/#asterisk potatoface (~potatofac@unaffiliated/sonshizuke/x-48977) |
| 15:43.02 | [TK]D-Fender | Maxxed: So far it looks like the username as the extension dialed |
| 15:43.12 | [TK]D-Fender | so that would be the way to separate teh calls |
| 15:43.37 | Maxxed | ah, i see that! |
| 15:43.48 | Maxxed | let me poke around a bit and see if i can get that going |
| 15:43.56 | potatoface | hi, i read everywhere that rtpholdtimeout must be bigger than rtptimeout... what happens if the two are set to the same value? anyone a clue? didnt find anything googling.... |
| 15:43.56 | Maxxed | thanks a ton for your help [TK]D-Fender :) |
| 15:44.07 | Maxxed | sometimes ya just need a 2nd set of eyes on these things :) |
| 15:44.38 | Maxxed | btw do you work for digium or something? i sware i recall your handle from like 5 years ago :p |
| 15:44.53 | Maxxed | hell, prob further back than that! |
| 15:45.39 | [TK]D-Fender | Maxxed: I've been here for a little over 10 years now. |
| 15:46.03 | [TK]D-Fender | Maxxed: But no, not employed by an IT related business at all |
| 15:46.42 | Qwell | Maxxed: He's just a sadist. |
| 15:47.28 | Maxxed | haha :p |
| 15:47.40 | Qwell | You think I'm joking. |
| 15:47.41 | Qwell | >.> |
| 15:52.52 | [TK]D-Fender | It's like a kind of tortue... |
| 15:52.56 | [TK]D-Fender | TO HAVE TO WATCH THIS SHOW! |
| 15:57.46 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-dogpfssxuwjvghna) |
| 15:58.13 | Maxxed | sweet, i have my dial plan sorted out, directing calls by username, woohoo :D |
| 15:58.25 | Maxxed | thanks again man, i owe you a soda :) |
| 15:59.21 | [TK]D-Fender | Might as well |
| 16:11.28 | pa | tzafrir, on gitorius i see that dahdi-linux-extra has been cloned by someone (not sure whom) into dahdi-linux-extra-zaphfc_fix |
| 16:11.35 | pa | did you merge that back, by chance? |
| 16:12.17 | pa | oh i think so |
| 16:12.20 | pa | october 2011 |
| 16:17.04 | tzafrir | pa, do you have such a card? |
| 16:17.31 | tzafrir | If so, such a card that can be down for testing? |
| 16:18.38 | pa | yes |
| 16:18.46 | pa | i was going to clone and try it, now |
| 16:19.00 | pa | let's see if i manage to build it |
| 16:22.36 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:22.41 | pa | tzafrir, i'm checking the readme. It says that dahdi requires BKL. but is there such thing anymore in modern 3.1x kernels? |
| 16:22.58 | pa | at least in my config i can't find it anymore |
| 16:23.46 | tzafrir | do you have a kernel git tree and can spot at which version it stopped working? |
| 16:24.10 | pa | you mean BKL? |
| 16:24.15 | tzafrir | Generally you only need to build a kernel tree as far as 'make modules_prepare' |
| 16:24.40 | pa | i have my own kernel compiled, so i think that should be ok |
| 16:24.42 | tzafrir | and then use 'make KSRC=/full/path/to/kernel/tree' in the dahdi source tree |
| 16:29.37 | pa | tzafrir, it looks like i have some compile errors: http://codepad.org/L4UWLdWI . As you can see in the paste, i'm using kernel 3.13 sources from ubuntu 14.04, compiled by myself |
| 16:30.09 | pa | am i missing some step? |
| 16:30.26 | pa | do i have to unload/uninstall ubuntu's dahdi? |
| 16:30.29 | pa | (first) |
| 16:30.57 | tzafrir | which tree do you try to build? |
| 16:31.15 | tzafrir | no, this is unrelated to the Ubuntu package |
| 16:31.20 | pa | it's the dahdi-extra/dahdi-linux-extra on your gitorious |
| 16:31.42 | tzafrir | it's an error in dahdi-base.c, so it's not related to the zaphfc driver |
| 16:32.41 | pa | hm.. right. but what does that mean? that there's an error in that tree? |
| 16:34.29 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 17:00.18 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
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| 17:26.31 | *** join/#asterisk FuriousGeorge (6bbc1aee@gateway/web/freenode/ip.107.188.26.238) |
| 17:29.34 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zowzrtunxyumjfvq) |
| 17:29.44 | FuriousGeorge | hey all, im in a tight spot here |
| 17:30.04 | FuriousGeorge | i brought some electricians out to rip out and old intercom system to replace it with asteriosk |
| 17:30.28 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 17:30.37 | FuriousGeorge | and i bring a server i set up today, and not only will the channel banks not connect to it, nothing will. tried x-lite and other clients too. i see nothing on the console |
| 17:31.19 | FuriousGeorge | maybe this is related, but asterisk -rvvv cannot connect to running asterisk, even tho the pid and ctl files exist under var run, and i can see that the process is running |
| 17:31.37 | FuriousGeorge | so right now im paying these guys to do nothing while my server is not accepting connections |
| 17:32.01 | Qwell | Kill Asterisk, start it with asterisk -vvvvc. What does it say is happening? |
| 17:32.52 | FuriousGeorge | Qwell: trued that. it says nothing is happening. i try to connect x-lite and i see nothing on the console to indicate a connection came in, failed or otherwise |
| 17:33.16 | Qwell | I don't see "nothing is happening" anywhere in the source. |
| 17:33.31 | Qwell | pastebin a full log of starting Asterisk with -vvvvvc |
| 17:33.43 | FuriousGeorge | Qwell: good idea |
| 17:38.40 | FuriousGeorge | Qwell: http://pastebin.com/LeiKpcp1 |
| 17:39.11 | FuriousGeorge | Qwell: sip listening on port 0.0.0.0:5060.... didn't see that before. that can't be right can it? |
| 17:39.19 | [TK]D-Fender | FuriousGeorge: I'm not seeing calls... I'm seeing * loading its configs and that's it |
| 17:39.37 | [TK]D-Fender | [12:39]FuriousGeorgeQwell: sip listening on port 0.0.0.0:5060.... didn't see that before. that can't be right can it? <- that's the norm |
| 17:39.59 | [TK]D-Fender | FuriousGeorge: Where's the SIP debug for any kind of comms to/from anything? |
| 17:40.30 | FuriousGeorge | [TK]D-Fender: all i have are sip clients, and since asterisk seems not to see any connection attempts, i can[;t make calls |
| 17:40.33 | FuriousGeorge | ill try sip debug |
| 17:40.39 | ChannelZ-Wk | Does this thing know its correct IP address? |
| 17:40.52 | ChannelZ-Wk | (or at least the one you think it is?) |
| 17:41.00 | [TK]D-Fender | doesn't matter yet |
| 17:41.03 | ChannelZ-Wk | ifconfig |
| 17:41.15 | [TK]D-Fender | we aren't looking at traffic and see nothing at the level of debugging we're at |
| 17:41.26 | ChannelZ-Wk | If he sees nothing, it sort of implies the traffic isn't making it to where he thinks it is. |
| 17:41.31 | [TK]D-Fender | Go check for traffic. then when you see none, go check your firewall. |
| 17:41.41 | [TK]D-Fender | Then when you see none go check your clients themselves |
| 17:42.02 | ChannelZ-Wk | (and I was referring to the machine its self, not anything Asterisk thinks) |
| 17:42.11 | [TK]D-Fender | ChannelZ-Wk: no |
| 17:42.29 | [TK]D-Fender | ChannelZ-Wk: If you're referring to WAN IP, knowing that does not change the fact of nothing coming IN. |
| 17:42.44 | *** join/#asterisk luizmaia (~lcm@177.66.104.81) |
| 17:43.23 | *** join/#asterisk Navion (~Navion@rivendell/users/navion) |
| 17:43.45 | FuriousGeorge | ChannelZ-Wk: that could be it. i set static ip on it |
| 17:43.45 | ChannelZ-Wk | I'm saying if all his phone are trying to talk to some hostname or IP that isn't actually the IP of his box because it's misconfigured. |
| 17:44.13 | FuriousGeorge | sip debug gets me nothing |
| 17:44.25 | [TK]D-Fender | FuriousGeorge: Check your firewall |
| 17:44.26 | FuriousGeorge | ifconfig shows correct ip. i can ping from asterisk and i can be pinged |
| 17:44.32 | [TK]D-Fender | FuriousGeorge: Then your client |
| 17:45.08 | FuriousGeorge | [TK]D-Fender: there is none (for instance) between the gxw54216 channel bank and asterisk... unless centos comes with something out of the box |
| 17:45.08 | FuriousGeorge | but i can ssh in |
| 17:45.29 | ChannelZ-Wk | iptables -L -v -n |
| 17:45.34 | [TK]D-Fender | FuriousGeorge: "unless" tells me you aren't looking |
| 17:46.13 | luizmaia | Hi guys, i'me new on voip Stuff, i managed to install an elstix with extensios talking to each other. I have a E1 trunk, i have configured it with DAHDI using some internet documentation, but i cant call out and receive connection throw this trunk, some information: |
| 17:46.22 | luizmaia | Elastix 2.5 |
| 17:46.38 | luizmaia | pci:0000:04:01.0 wcte11xp+ e159:0001 Digium Wildcard TE110P T1/E1 Board |
| 17:47.10 | luizmaia | my channels: |
| 17:47.11 | luizmaia | piranha*CLI> dahdi show channels |
| 17:47.11 | luizmaia | <PROTECTED> |
| 17:47.11 | luizmaia | <PROTECTED> |
| 17:47.11 | luizmaia | <PROTECTED> |
| 17:47.11 | luizmaia | <PROTECTED> |
| 17:47.21 | [TK]D-Fender | BAI BAI |
| 17:48.08 | *** join/#asterisk luizmaia (~lcm@177.66.104.81) |
| 17:48.13 | luizmaia | sorry |
| 17:48.14 | luizmaia | :D |
| 17:48.15 | [TK]D-Fender | ~pb |
| 17:48.15 | 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. |
| 17:48.19 | [TK]D-Fender | luizmaia: ^^^^ |
| 17:48.20 | [TK]D-Fender | Do NOT flood in here |
| 17:48.31 | luizmaia | ok, my bad. |
| 17:48.36 | [TK]D-Fender | luizmaia: Also, FreePBX is not supported in here. |
| 17:48.38 | [TK]D-Fender | ~freepbx |
| 17:48.38 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:48.40 | [TK]D-Fender | ^^ |
| 17:49.13 | luizmaia | ok, but i think my problem is with asterisk |
| 17:49.36 | ChannelZ-Wk | Well your problem is probably freepbx isn't configuring asterisk right |
| 17:49.53 | luizmaia | i got this error: [2014-11-25 14:10:03] WARNING[7719][C-00000002] app_dial.c: Unable to create channel of type 'DAHDI' (cause 6 - Channel unacceptable), its FreePBX? |
| 17:50.01 | ChannelZ-Wk | We can tell you how to fix asterisk but freepbx will generally stomp on top of configs and do whatever the hell it wants |
| 17:50.21 | ChannelZ-Wk | What channel is it trying to dial? |
| 17:50.36 | luizmaia | i understand, i'll try on frepbx channel. thanks anyway. |
| 17:50.52 | ChannelZ-Wk | oh.. although you got kicked out, we only saw 2 |
| 17:51.54 | [TK]D-Fender | [12:49]luizmaiaok, but i think my problem is with asterisk <- you didn't show us the dial command it even tried to execute, only the fact it thought iwas bad. These commands are build by FreePBX based on your configs as well. Take it up there. |
| 17:51.58 | *** join/#asterisk FuriousGeorge (6bbc1aee@gateway/web/freenode/ip.107.188.26.238) |
| 17:52.20 | FuriousGeorge | sorry, connection dropped. i see nothing in the iptables output about 5060, but im not too familiar with it |
| 17:52.28 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:53.22 | luizmaia | ok, here is the log: http://pastebin.com/eja50uRh |
| 17:54.16 | ChannelZ-Wk | Are your E1 channels in a group? (dunno how this manifests in FreePBX) |
| 17:54.45 | ChannelZ-Wk | (though I'd assume it does this automatically.. but this is why it's hard to help with FreePBX) |
| 17:55.13 | [TK]D-Fender | luizmaia: take this to #freepbx |
| 17:55.47 | FuriousGeorge | i have a suspicion that somehow my network is misconfigured |
| 17:56.07 | FuriousGeorge | but i can ping asterisk server from this laptop and connect over ssh |
| 17:56.11 | MasterChen | hello everyone |
| 17:56.22 | [TK]D-Fender | furYou're only talking about "output", you're showing us nothing, and you've given us no details. |
| 17:56.28 | [TK]D-Fender | FuriousGeorge: You're only talking about "output", you're showing us nothing, and you've given us no details. |
| 17:57.03 | ChannelZ-Wk | FuriousGeorge: what was the result of "iptables -L -v -n" ? Is the default policy ALLOW or is something else going on? |
| 17:59.03 | FuriousGeorge | ChannelZ-Wk: http://pastebin.com/7ga92AjV |
| 17:59.14 | FuriousGeorge | i wouldn't know if there's a problem there |
| 17:59.21 | FuriousGeorge | but i suppose it looks good to me |
| 17:59.25 | FuriousGeorge | nothing ibvious |
| 17:59.45 | ChannelZ-Wk | yikes |
| 17:59.54 | luizmaia | <[TK]D-Fender>, ok i will! |
| 18:00.18 | FuriousGeorge | ChannelZ-Wk: that doesn't sound good |
| 18:01.35 | FuriousGeorge | this is an out of the box centos net install. all i did wsa set ip to static, so i have no idea what setting (if any) could be causing this |
| 18:01.41 | ChannelZ-Wk | there's just a lot going on in this thing. I don't know if this is a Centos thing or some firewall script |
| 18:02.06 | [TK]D-Fender | FuriousGeorge: Trash that firewall. |
| 18:02.10 | ChannelZ-Wk | I'm confused by "Chain INPUT (policy ACCEPT 0 packets, 0 bytes)" |
| 18:02.41 | FuriousGeorge | [TK]D-Fender: sorry, could you be more specific, ive never worked with iptables. is there a way to just allow all? |
| 18:02.53 | FuriousGeorge | this thing is not even to be connected to the internet |
| 18:02.58 | [TK]D-Fender | FuriousGeorge: TRASH IT |
| 18:03.04 | [TK]D-Fender | iptables --flush |
| 18:05.34 | FuriousGeorge | that worked |
| 18:06.17 | ChannelZ-Wk | ..it caused things to work? |
| 18:06.40 | FuriousGeorge | ChannelZ-Wk: yes, i can see registrations, sorry i wasn;t more specific |
| 18:06.59 | ChannelZ-Wk | which would make sense the more I look at that firewall.. which seemed to only allow icmp (why your pings worked) and ssh. |
| 18:07.34 | FuriousGeorge | unfortunately there's some script setting this. i guess ill take it to #centos |
| 18:07.46 | FuriousGeorge | unless you guys have a suspect in mind |
| 18:08.11 | ChannelZ-Wk | Sorry I don't run centos |
| 18:08.59 | FuriousGeorge | ChannelZ-Wk: np, and i appreciate your help very much |
| 18:09.14 | ChannelZ-Wk | "firewalld" perhaps |
| 18:11.07 | FuriousGeorge | ChannelZ-Wk: u got it (i think) firewakd is running |
| 18:12.37 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 18:12.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:13.51 | ChannelZ-Wk | dunno what init script system centos uses but you probably want to disable it |
| 18:15.04 | FuriousGeorge | firewald was under the systemctl command, asterisk is seen with chkconfig and services blah start/stop |
| 18:15.08 | FuriousGeorge | ChannelZ-Wk: you got it again |
| 18:15.22 | ChannelZ-Wk | Or dig into its config to set it up correctly to work for you, but if it's really a completely internal box running things you know about and you don't care to/need to firewall it specifically, I'd just shut it off |
| 18:15.22 | FuriousGeorge | i disabled it and everything is fine |
| 18:15.44 | FuriousGeorge | ChannelZ-Wk: that's what i did. there is no wan access here |
| 18:15.47 | Kobaz | you would think after like more than 5 years in development you would be able to hot-reload the entire config for a polycom without rebooting |
| 18:16.08 | ChannelZ-Wk | It seems like most phones are like that. |
| 18:16.39 | Kobaz | i know, it's terrible |
| 18:16.43 | ChannelZ-Wk | You changed the voicemail button extension! Rebooting! |
| 18:16.53 | Kobaz | haha |
| 18:16.53 | Kobaz | yeah |
| 18:17.02 | Kobaz | give me the source and i'll fix it |
| 18:17.10 | FuriousGeorge | later all and thanks all again |
| 18:18.06 | [TK]D-Fender | See I just get it right the first time , every time, and never have to worry about things like that... |
| 18:18.32 | ChannelZ-Wk | See ya Furious |
| 18:21.35 | *** join/#asterisk n3tctrl (~n3tctrl@12.218.71.199) |
| 18:46.23 | Kobaz | [TK]D-Fender: and then when requirements change? |
| 18:48.04 | *** join/#asterisk areski (~areski@80.174.128.55.dyn.user.ono.com) |
| 18:48.06 | [TK]D-Fender | Kobaz: Hasn't yet... |
| 18:48.29 | Kobaz | hah |
| 18:48.33 | [TK]D-Fender | Kobaz: And having to do that after months and months... sure doesn't make me care about the minor inconvenience |
| 19:00.15 | *** join/#asterisk aruntomar (~arun@123.252.214.209) |
| 19:01.07 | *** join/#asterisk u0m3 (~u0m3@92.80.86.86) |
| 19:01.12 | *** join/#asterisk adeel (~adeeln@fw1.ridgeway.scc-zip.net) |
| 19:01.55 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 19:02.06 | *** join/#asterisk hfb (~hfb@pool-71-119-57-68.lsanca.dsl-w.verizon.net) |
| 19:04.32 | adeel | is there a recommended way to emulate a for loop inside of the asterisk dialplan? i would like to iterate over a set of variables, with varying names |
| 19:04.57 | Kobaz | emulate a loop? |
| 19:05.03 | Kobaz | why would you want to emulate a loop, when you can just.... loop |
| 19:05.16 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 19:05.23 | Kobaz | adeel: http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
| 19:05.36 | adeel | Kobaz: ah, i was trying to avoid AEL |
| 19:05.46 | Kobaz | adeel: scroll to the bottom where it says Loops |
| 19:05.50 | Kobaz | adeel: why? |
| 19:06.06 | [TK]D-Fender | adeel: increment counter. check counter. Ok to continue? continue. loop back. Wash. Rinse. Repeat |
| 19:06.08 | adeel | Kobaz: maintenance reasons...no one else here knows AEL...they barely know the basic dialplans |
| 19:06.15 | Kobaz | it gets converted to dialplan anyway, there's a negligible speed hit |
| 19:06.35 | [TK]D-Fender | And that isn't emulating a loop... it IT one. |
| 19:06.36 | Kobaz | adeel: well, if someone knows at least the tinyest little bit of scripting, ael will make more sense than regular dialplan |
| 19:06.38 | [TK]D-Fender | IS* |
| 19:06.40 | adeel | [TK]D-Fender: unless there's a way to dynamically generate a variable name, that doesn't really work |
| 19:06.48 | Kobaz | adeel: you can generate a variable name |
| 19:06.52 | [TK]D-Fender | adeel: Perhaps you should be more specific |
| 19:06.54 | adeel | Kobaz: how? |
| 19:06.57 | Kobaz | Set(varname=foo); |
| 19:07.06 | Kobaz | NoOp(${${varname}}); |
| 19:07.17 | [TK]D-Fender | adeel: ALL dialplan variables are dynamic |
| 19:07.26 | Kobaz | adeel: note the double ${ |
| 19:08.18 | Kobaz | adeel: loop in regular dialplan... line 1: set loops = 10; do stuff; loops = loops - 1; if loops > 0; goto line 1 |
| 19:08.23 | Kobaz | adeel: have you ever done BASIC programming? |
| 19:08.29 | Kobaz | dialplan is the same thing |
| 19:08.32 | [TK]D-Fender | Nope |
| 19:08.35 | [TK]D-Fender | far from |
| 19:08.59 | Kobaz | it's the same language form |
| 19:09.02 | [TK]D-Fender | * Dialplan is much more like assembler. |
| 19:09.05 | Kobaz | is what i meant |
| 19:09.06 | adeel | Kobaz: have you ever not been condescending? i appreciate the help, but way to be a douche about it |
| 19:09.12 | *** part/#asterisk adeel (~adeeln@fw1.ridgeway.scc-zip.net) |
| 19:09.15 | [TK]D-Fender | with As for as flow is concerned |
| 19:09.23 | Kobaz | when was i condescending? |
| 19:09.40 | [TK]D-Fender | You weren't |
| 19:09.45 | [TK]D-Fender | (that I could tell) |
| 19:09.51 | Kobaz | it was all basic facts |
| 19:10.05 | [TK]D-Fender | He was unfortunately poor in his description of his actual need. |
| 19:10.24 | Kobaz | assembler/basic |
| 19:10.37 | Kobaz | i mean like... it's all unstructured linear programming |
| 19:10.39 | [TK]D-Fender | Perhaps frustrated and maybe unreasonably off-put by the suggestion to use AEL. |
| 19:10.43 | Kobaz | :( |
| 19:10.45 | Kobaz | i love ael |
| 19:10.50 | Kobaz | as you already know :) |
| 19:10.58 | [TK]D-Fender | Crap we never needed... |
| 19:11.03 | Kobaz | it's good crap |
| 19:11.18 | [TK]D-Fender | No, it's really an ugly patch. |
| 19:11.19 | Kobaz | it's actually not crap, if you really start to use it and appreciate the structure |
| 19:11.31 | Kobaz | okay yes, it's an ugly way to do it |
| 19:11.35 | Kobaz | transpile to dialplan |
| 19:11.35 | [TK]D-Fender | PBX_LUA and similarly direct methods are much more like what the goal should have been |
| 19:11.42 | Kobaz | yeah, okay, i would go with that |
| 19:11.55 | Kobaz | lua is an nice approach, i just wish the variable syntax wasn't so verbose |
| 19:12.08 | [TK]D-Fender | I never touched either as standard dialplan works fine for me and just about everyone. |
| 19:14.46 | *** join/#asterisk FuriousGeorge (6bbc1868@gateway/web/freenode/ip.107.188.24.104) |
| 19:14.53 | FuriousGeorge | hey all. |
| 19:15.13 | FuriousGeorge | make config does not make a working init script for sentos 7. it just hangs |
| 19:15.25 | FuriousGeorge | can someone recommend a more simple more generic one that they use? |
| 19:15.31 | Kobaz | [TK]D-Fender: i 100% agree it works |
| 19:16.21 | Kobaz | if you want to do any kind of remotely complicated scripting that has a readable structure..... then a structured language will help with that |
| 19:16.31 | Kobaz | that's my main point in the ael/lua discussions |
| 19:17.09 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 19:18.40 | FuriousGeorge | exec //usr/sbin/safe_asterisk works well enough |
| 19:19.12 | Kobaz | [11 25 14:10] <adeel> kobaz: adeel: have you ever done BASIC programming? --- that's not condescending? |
| 19:19.12 | Kobaz | [11 25 14:10] <adeel> but just fuck off dude, thanks for whatever "help" you've given.. |
| 19:19.13 | Kobaz | poor guy |
| 19:22.49 | [TK]D-Fender | Kobaz[11 25 14:10] <adeel> kobaz: adeel: have you ever done BASIC programming? --- that's not condescending? <- Nope |
| 19:22.51 | pjensen00 | eh, you have to have a thick hide when asking for help |
| 19:23.09 | Kobaz | what probably happened |
| 19:23.16 | Kobaz | is he didn't know BASIC was a language |
| 19:23.30 | pjensen00 | Ha, I hadn't even thought of that |
| 19:23.37 | Kobaz | if you take it literally, then that would be kind of insulting |
| 19:23.44 | [TK]D-Fender | Kobaz: That'd be kinda sad/unfortunate |
| 19:23.52 | Kobaz | but i did explain to him that BASIC was a language |
| 19:23.54 | Kobaz | he didn't care |
| 19:24.03 | pjensen00 | wouldn't worry about it |
| 19:24.05 | Kobaz | it's like asking "have your used html" |
| 19:24.06 | Kobaz | haha |
| 19:24.24 | Kobaz | yeah, i'm not going to lose sleeo over it |
| 19:24.24 | [TK]D-Fender | WHAT'D YOU CALL ME?!?! |
| 19:24.26 | Kobaz | haha |
| 19:24.30 | Kobaz | I CALLED YOU AN HTML |
| 19:24.46 | pjensen00 | ... yeah well you're a dongle. |
| 19:24.51 | [TK]D-Fender | runs in circles screaming |
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| 20:35.26 | linuxgeek | ~ask Can I set the useragent string in chan_pjsip ? |
| 20:35.26 | infobot | Nope, linuxgeek! I won't ask "Can I set the useragent string in chan_pjsip" |
| 20:36.28 | linuxgeek | Hi, Can I set the useragent string in chan_pjsip ? |
| 20:36.56 | malcolmd | yes. declare it in a global type: [global] type=global user_agent=My Fancy Thing |
| 20:37.36 | linuxgeek | ah , thanks malcolmd , I'll test that and report back |
| 20:40.19 | linuxgeek | malcolmd, is there a complete configuration reference anywhere ? The documentation has very little 1-to-1 reference |
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| 20:48.34 | ChannelZ-Wk | So with PJSIP when I make a call, it seems to be returning 'Ringing' to my device immediately before it even attempts to dial the remote end. Is that a pjsip thing or an asterisk thing? |
| 20:52.21 | [TK]D-Fender | ChannelZ-Wk: Show us the call |
| 20:55.31 | malcolmd | linuxgeek: there are some good examples of basic things on the wiki. pjsip itself is self-documented though. rather than compiling a burdensomely large pjsip.conf.sample file, try "config show help" from the asterisk cli. you'll then see that you can get help for lots of loaded modules, including the pjsip ones |
| 20:57.20 | linuxgeek | malcolmd, thanks , will look into that |
| 20:57.43 | Graiden | I checked in over at #freepbx and haven't seen an answer yet, figured I'd run it by the folks here. Does anyone have a working XML page that displays voicemail count, caller id, etc... and allows the user to delete from their SIP device, or have a suggestion on where I might look? |
| 21:01.37 | [TK]D-Fender | Graiden: There are a few generic web ones already. Feel free to convert those to XML |
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| 21:17.40 | dan_j | Hi. I'm having a problem with IFTIME. For some reason, even though the time matches, the IF statement is still returning false. |
| 21:17.42 | dan_j | Any ideas? |
| 21:17.42 | dan_j | http://pastebin.com/XYybCNDk |
| 21:19.30 | [TK]D-Fender | dan_j: invalid function syntax |
| 21:19.44 | [TK]D-Fender | dan_j: $IFTIME <- no |
| 21:20.27 | dan_j | Ah. Gotcha |
| 21:20.41 | [TK]D-Fender | You also have nested a ton of expressions you really don't need |
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| 21:20.58 | [TK]D-Fender | <PROTECTED> |
| 21:21.43 | dan_j | I'm just used to putting " around things because sometimes, one side is totally empty. IE if one side contains an empty string. |
| 21:21.43 | grog | newbie general VoIP question alert... I'm in Thailand, my provider's (VoIP.ms) nearest server is in London, does that mean my incoming calls hit the PSTN there, or are routed over the interpipes and terminate within Thailand? |
| 21:22.03 | dan_j | grog: London pstn |
| 21:22.34 | dan_j | Incoming calls from pstn lines will come through pstn to VoIP.ms |
| 21:23.31 | [TK]D-Fender | dan_j: that isn't going to be blank and already returns what it has to. no need for an expression at all |
| 21:23.31 | grog | right, but how do they get to Thailand? |
| 21:23.41 | grog | because thailand's routing and international gateways are horrid (300-400ms, roundabout thru SG or HK) |
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| 21:28.10 | [TK]D-Fender | heads home... |
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| 21:36.48 | Penguin | I still need to know how to configure asterisk 11 for multiple google voice accounts so that it behaves the same way as 1.8. I want incoming calls to go to extension username@gmail.com instead of exten s. |
| 21:37.34 | Penguin | In 1.8, each incoming call goes to username@gmail.com and I created an extension for each account. In 11, calls are going to s every time. |
| 21:38.29 | Penguin | I asked about the transport 1.8 used since 11 offers several transports. Does the transport affect the extension inbound calls are sent to? |
| 21:41.11 | Penguin | What controls which extension calls are going to? In 1.8, calls go to username@gmail.com first, and if that extension doesn't exist, it will fall back to exten s. |
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| 21:47.48 | Micc | It seems like hadoop would be perfect for storing asterisk voicemail. Anyone know if this has been done before? |
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