| 00:17.51 | vader- | quick opinion question. We are currently running an older version of Asterisk like 1.2.x... All dial plans have been done by hand, all sip connections, etc. If you had to move to a GUI type interface for configuration what would you recommend? Our enviroment is about 70-100 SIP Phones, Digium TDM2400 and a Digium PRI card... Right now it's easy for me to provision a phone with a cisco config |
| 00:17.52 | vader- | file, and the asterisk settings, etc. But for other people in my department and anyone new we hire they need point and click... Just looking for any suggestions you guys might have. |
| 00:18.19 | vader- | would freepbx be the best option you guys think? |
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| 01:36.33 | eXcAliBuR | how do i show what sip conf is loaded? |
| 01:36.38 | eXcAliBuR | i tried show sip |
| 01:36.40 | eXcAliBuR | that didn't work |
| 01:36.40 | eXcAliBuR | :/ |
| 01:37.23 | paulc | try: sip show peers |
| 01:37.28 | paulc | or sip show registry |
| 01:37.33 | paulc | depending on what you're looking for |
| 01:37.45 | eXcAliBuR | i want to see if the include file is loading |
| 01:38.51 | paulc | What's in the include file? if it's peers, you should be able to see them in "sip show peers" (I do that, and it works great) |
| 01:39.01 | eXcAliBuR | it's general stuff |
| 01:39.22 | eXcAliBuR | and i don't know if i need to put it with [general] or if it takes that from the main sip.conf |
| 01:40.08 | paulc | I'd lean towards general stuff in [general], and peer-specific stuff in peers, with or without includes (but with is nice for externally generated stuff) |
| 01:40.49 | paulc | I gotta take off, time to go battle traffic.. good luck though! |
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| 02:04.20 | [TK]D-Fender | [19:18]vader-would freepbx be the best option you guys think? <- yes |
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| 02:59.02 | eXcAliBuR | i'm running freepbx (trying it) fresh install... i get this |
| 02:59.04 | eXcAliBuR | *CLI> reload |
| 02:59.04 | eXcAliBuR | No such command 'reload' |
| 02:59.10 | eXcAliBuR | nothing seems to work |
| 02:59.12 | eXcAliBuR | :{ |
| 02:59.56 | Docfxit | Is Penguin awake? |
| 03:00.12 | eXcAliBuR | i think i killed it |
| 03:00.13 | eXcAliBuR | :{ |
| 03:07.58 | Docfxit | I have an Asterisk box that ran out of room on the hard drive. I think the asterisk.ctl file has disapeared. Could I get some help in getting Asterisk up and running? |
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| 03:26.41 | mnathani | how do I put a dialplan in that recognizes + then country code rather than 011 + country code |
| 03:27.01 | mnathani | I mean the actual "+" prepended to the number |
| 03:37.31 | mnathani | I currently get: to extension '+1NNNMMMMMMM' rejected because extension not found in context 'mycontext'. |
| 03:42.50 | D30 | hi all |
| 03:42.51 | D30 | http://codepad.org/YSzojlgY |
| 03:43.07 | D30 | does channel 1 - 4 represents the FXO? |
| 03:43.13 | D30 | good day btw.. |
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| 03:46.54 | eXcAliBuR | is 64bit and less stable than 32 bit installs? |
| 03:46.59 | eXcAliBuR | any* |
| 03:55.46 | leifmadsen | not that I've ever experienced |
| 03:55.53 | leifmadsen | if anything, more stable due to ability to use more memory |
| 03:56.01 | leifmadsen | (excluding PAE kernels) |
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| 04:27.17 | eXcAliBuR | thanks senior leifmadsen |
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| 04:57.38 | mnathani | I am trying to make sense of this forum post: http://www.freepbx.org/comment/14473#comment-14473 |
| 04:57.46 | mnathani | which file do those commands go in? |
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| 06:01.39 | [TK]D-Fender | mnathani: that post tells you.... |
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| 06:14.53 | vedic | Can you tell me if there are any advantages of CRC4 enabling on E1 line? |
| 06:15.09 | vedic | My operator says he can enable it if requierd. By default, CRC4 is disabled |
| 06:16.50 | vedic | Note that I am trying to use Speech Recognition on Asterisk |
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| 08:05.20 | Bearish | Sooo, i've got this problem, we are using Asterisk as a SIP gate for our Avaya CM, and we have some incoming calls through SIP which i route to a custom extesnion like 4000@AvayaH323 |
| 08:05.25 | Bearish | 4000 is our Avaya-based call center number, where a secretary picks up a phone and then transfers the call to its final destination (sales people, for example) |
| 08:06.03 | Bearish | The problem is, i don't see the "final destination" in my CDR logs, and i'd like to |
| 08:06.17 | Bearish | Is it possible in any way? |
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| 08:20.40 | Blashyrkh | i just installed iaxmodem, created peer entries for every modem i configured in iax.conf, |
| 08:20.40 | Blashyrkh | but when i start iaxmoden they dont connect |
| 08:22.41 | Blashyrkh | when i start iaxmodem manually iaxmodem config, i get a Registration timed out |
| 08:22.42 | Blashyrkh | http://pastebin.com/iuWG2YaR |
| 08:27.11 | mirela666 | Blashyrkh: andin your iax.conf |
| 08:27.42 | mirela666 | calltokenoptional= is it set to 127.0.0.1 |
| 08:28.03 | mirela666 | make sure asswords and ports are correct |
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| 08:36.44 | Blashyrkh | passwords and ports are correct |
| 08:36.53 | Blashyrkh | do i need to specify the ports in the iax.conf too? |
| 08:37.02 | Blashyrkh | i just did but i doesnt make no difference |
| 08:37.22 | Blashyrkh | calltokenoptional = 127.0.0.1 should b in there? |
| 08:37.51 | Blashyrkh | that just removes the need for calltoke for localhost right? |
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| 08:45.31 | D30 | hi all what usually the cause of having no audio during calls? |
| 08:45.52 | D30 | i can that the call went through but not heard anything including ringing |
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| 08:50.11 | iulhk | i have installed asterisk 10.7.0 , does asterisk support instant messaging without integration of openfire or opensips, kamailio? |
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| 09:44.21 | iulhk | nobody is there ? |
| 09:44.33 | iulhk | i have installed asterisk 10.7.0 , does asterisk support instant messaging without integration of openfire or opensips, kamailio? |
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| 09:57.56 | Blashyrkh | is there some way inside asterisk to see if my iaxmodems are trying to register? |
| 09:58.11 | Blashyrkh | i set iax2 set debug on |
| 09:58.59 | Blashyrkh | and when i do an iax2 reload, shouldnt it show me the extensions asterisk just parsed? |
| 09:59.31 | Blashyrkh | in this case peers |
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| 10:00.45 | Blashyrkh | on another machine with the exact same configuratino but a different asterisk version it tells me |
| 10:00.45 | Blashyrkh | <PROTECTED> |
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| 10:28.08 | mirela666 | Blashyrkh: what does iax show peers tells you? |
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| 10:58.17 | mirela666 | Hi, VERBOSE[16738] rtp.c: -- Packet2Packet bridging SIP... is a message that says connecting 2 sides to direct RTP stream? |
| 10:58.58 | mirela666 | cause my calls drop after answer and that is the last msg |
| 10:59.40 | mirela666 | and endpoints can't reach them selfs directly |
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| 11:01.10 | WIMPy | Calls drop or you just don't hear anything? |
| 11:01.30 | mirela666 | drop in same miisec\ |
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| 11:13.10 | mirela666 | aha found good info here http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
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| 12:55.48 | wonderworld | hi, trying to port an asterisk instance from one machine to another. compiled ast12 on the new machine and copied the config from the old machine. now i get Cannot update type 'bucket' in module 'core' because it has no existing documentation! when staring |
| 12:56.16 | wonderworld | could this be related to not having made progdocs on the new instance? |
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| 14:03.01 | devil_evoxxx | hi all guy |
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| 14:06.46 | BeachBall | whats the cheapest toll-free provider by minute? I've seen $0.04 --> to canada |
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| 14:17.49 | Katty | BeachBall: tin can and a string. |
| 14:19.55 | BeachBall | time to Rrrroll up the rim |
| 14:20.30 | BeachBall | didn't win :{ |
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| 14:29.36 | devil_evoxxx | what's the best way to pickup call from queue? I receive calls from my telco provider and send to a queue |
| 14:29.48 | devil_evoxxx | if another phoe, out of the queue want to pickup the call, what's the best way? |
| 14:29.57 | Katty | with a handset. |
| 14:30.04 | Katty | you lift the phone and go, herro! |
| 14:30.14 | Katty | Agent Devil here! |
| 14:31.14 | WIMPy | devil_evoxxx: The opnly option I see is via an external application using AMI. |
| 14:31.36 | WIMPy | That's the way I pick up calls that have gone to VM. |
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| 14:32.24 | Katty | hugs sruffell |
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| 14:34.06 | mirela666 | devil_evoxxx: you can set up a speed dial button on the phone which will do the AddQueueMember() app |
| 14:35.08 | [TK]D-Fender | Or you could just use the standard PICKUP tools.... |
| 14:35.41 | WIMPy | Wow |
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| 14:36.33 | WIMPy | Oh, I see. You don't pick up the call from the queue, but from the agent. |
| 14:36.57 | WIMPy | So it would only work while at least one agent is ringing. |
| 14:38.07 | WIMPy | But if there is no ringing agent, mirela666s version would work without waiting. |
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| 14:43.17 | devil_evoxxx | WIMPy: there always a ringing agent ( queue with rrmemory strategy) |
| 14:44.14 | mirela666 | you can use pickup group for example |
| 14:44.20 | WIMPy | I don't know your situation, but all agents being busy seems likely to me. |
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| 14:47.48 | devil_evoxxx | so i can add group to each sip-configuration in sip.conf in.example callgroup / pickupgroup |
| 14:47.52 | devil_evoxxx | and then pickup? |
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| 14:48.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:50.34 | jsarrel | Hi all, I am having a difficult time with a dialplan (Asterisk 11). I am trying to extract a substring out of a string containing quotes. I was hoping someone here could point me in the right direction. Dialplan snippet: http://pastebin.com/cuvQ8PBC Log: http://pastebin.com/aLd673sF |
| 14:51.07 | jsarrel | It seems Asterisk is not recognizing the escaped quotes? |
| 14:51.20 | jsarrel | or rather is interpreting them anyway |
| 14:56.03 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 14:56.26 | Katty | hugs russellb |
| 14:57.50 | russellb | ohai |
| 14:58.05 | Katty | herro. |
| 14:58.54 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 15:02.31 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:08.41 | leifmadsen | jsarrel: you gotta escape the escape probably |
| 15:08.46 | leifmadsen | \\\[ |
| 15:08.47 | leifmadsen | for example |
| 15:08.57 | *** join/#asterisk zoid_ (~awainer@181.29.129.12) |
| 15:09.25 | Katty | how's little leifmadsen jr this morning? |
| 15:09.31 | leifmadsen | Katty: off at the gym :) |
| 15:09.40 | Katty | bit young for cardio? |
| 15:09.42 | leifmadsen | he was good today, teeth not bothering him too much so far |
| 15:09.47 | leifmadsen | Katty: nah, he loves running now |
| 15:09.50 | Katty | excellent! |
| 15:09.52 | leifmadsen | just a little drunken |
| 15:09.57 | Katty | hehe |
| 15:10.06 | leifmadsen | (wife at the gym, he stays in the play area :)) |
| 15:10.09 | Katty | well i'm glad you're getting him started early |
| 15:10.27 | Katty | gym++ |
| 15:10.33 | Katty | soon the weather will be agreeable with running! |
| 15:10.37 | leifmadsen | indeed |
| 15:10.39 | leifmadsen | I was thinking that today |
| 15:10.43 | leifmadsen | it's been a terrible winter |
| 15:10.49 | Katty | nods |
| 15:10.50 | leifmadsen | I should really try and get out running this year |
| 15:10.51 | mirela666 | Can someone confirm to me with more experiance here |
| 15:10.52 | mirela666 | http://pastebin.com/jHyrc6Wm |
| 15:10.54 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 15:10.55 | zoid_ | HI, I'm trying to soft hangup some stuck channels in my asterisk, but the channel denomination used in "sip show channels" is not the same expected in request hangup, is it possible to translate one to another? |
| 15:11.07 | leifmadsen | zoid_: core show channels |
| 15:11.14 | leifmadsen | zoid_: channel request hangup <foo> |
| 15:11.18 | Katty | leifmadsen: what temp do you usually consider your Bare Minimum for running? |
| 15:11.24 | leifmadsen | Katty: 10C |
| 15:11.25 | mirela666 | Packet2Packet bridging from remote end meanse they have directmedai on yes |
| 15:11.43 | leifmadsen | mirela666: not quite... I believe it just means asterisk doesn't have to transcode |
| 15:11.51 | leifmadsen | so doesn't go as far up the stack |
| 15:12.04 | Katty | leifmadsen: sounds reasonable. mine's about 15C |
| 15:12.16 | leifmadsen | Katty: yea, 10C around here is shorts and t-shirt weather mostly |
| 15:12.21 | Katty | nods |
| 15:12.27 | Katty | silly canadians. hehe |
| 15:12.29 | leifmadsen | people drive with the top down in the car, with the heat cranked :) |
| 15:12.35 | leifmadsen | Katty: we have to take what we can get :) |
| 15:12.39 | mirela666 | you mean codecs and technology are the same |
| 15:12.49 | Katty | leifmadsen: what temp is Too Hot to run in for you? |
| 15:12.54 | zoid_ | leifmadsen: the problem with that, is that core show channels doesn't seems to be showing the stuck channels |
| 15:12.56 | Katty | s/too/to |
| 15:12.58 | leifmadsen | Katty: I 35C |
| 15:13.04 | leifmadsen | zoid_: then they aren't stuck |
| 15:13.05 | Katty | jeebus |
| 15:13.08 | Katty | that's pretty warm. |
| 15:13.18 | Katty | are you running in not bot your knickers?! |
| 15:13.21 | Katty | bbut |
| 15:13.22 | leifmadsen | zoid_: stuff in 'sip show channels' with no active calls is perfectly normal |
| 15:13.34 | mirela666 | leifmadsen: you mean codecs and technology are the same. But the calls drop after that |
| 15:13.36 | leifmadsen | Katty: yea, 35C I would try to run in a gully surrounded by trees |
| 15:13.43 | leifmadsen | mirela666: that's a different problem then |
| 15:13.47 | leifmadsen | sounds like dialplan error |
| 15:13.54 | Katty | leifmadsen: i stop at about 26C or so |
| 15:13.56 | zoid_ | leifmadsen: even hour after? It stays in BYE |
| 15:14.04 | leifmadsen | shrugs |
| 15:14.07 | Katty | leifmadsen: but i probably don't drink as much water as i should |
| 15:14.15 | leifmadsen | ok, that's all I got in me today, I got work to do :) |
| 15:14.23 | leifmadsen | Katty: yea on the hot hot days you gotta drink lots of water |
| 15:14.30 | zoid_ | leifmadsen: thanks |
| 15:14.35 | Katty | shoos leifmadsen back to work |
| 15:14.41 | leifmadsen | Katty: I would also run without a shirt (but that was back when I was fit and people wouldn't puke :)) |
| 15:14.47 | zoid_ | the real problem I'm having is RTP port exhaustion |
| 15:14.58 | Katty | i don't think that's much of an option for women. |
| 15:14.59 | *** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net) |
| 15:15.02 | zoid_ | and I believe is because of stuck channels |
| 15:15.09 | leifmadsen | Katty: it's an option in Canada |
| 15:15.32 | leifmadsen | literally... there is a law passed back like... 15 years ago that says women can be topless |
| 15:15.37 | leifmadsen | I have yet to see a topless woman though |
| 15:15.52 | Katty | i can't imagine why not ;) |
| 15:15.55 | leifmadsen | although for running, might be slightly uncomfortable :) |
| 15:15.59 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:15.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:15.59 | leifmadsen | Katty: ikr?! |
| 15:16.03 | Katty | hehe |
| 15:16.07 | Katty | waves to putnopvut |
| 15:16.48 | putnopvut | waves back |
| 15:16.57 | Katty | putnopvut: is today goodly? |
| 15:17.11 | leifmadsen | putnopvut: you missed all the topless woman talk |
| 15:17.21 | Qwell | can confirm |
| 15:17.29 | Katty | hugs Qwell |
| 15:17.42 | putnopvut | I learned yesterday that 33 states in the US permit toplessness in public. |
| 15:17.48 | *** join/#asterisk navaismo (~navaismo@201.124.139.163) |
| 15:17.55 | Katty | putnopvut: i hope missouri ISN'T one of them |
| 15:17.58 | putnopvut | And I suppose today is goodly so far. |
| 15:18.01 | Katty | putnopvut: excellent! |
| 15:18.04 | Katty | hugs navaismo |
| 15:18.13 | Katty | Qwell: how's your ladyfriend? |
| 15:18.48 | Katty | Qwell: stuffed full of tasty cajun foods, i hope? |
| 15:18.48 | Qwell | Katty: well. Been training a new puppy. |
| 15:18.55 | Qwell | quite! |
| 15:18.57 | Katty | Qwell: oooh. a new puppy! :> what kind? and are there photos? |
| 15:18.59 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 15:19.01 | leifmadsen | totally read "new puppy" as "her" |
| 15:19.20 | Katty | TIL leifmadsen is a kinkster! :P |
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| 15:19.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:19.24 | navaismo | hugs Katty |
| 15:19.29 | Katty | navaismo: how'rechu :> |
| 15:19.36 | Qwell | Maybe publicly visible? https://scontent-b.xx.fbcdn.net/hphotos-ash4/t1/1507627_10152249520359207_352527634_n.jpg |
| 15:19.45 | Qwell | mjordan: I blame you. |
| 15:19.52 | leifmadsen | Qwell: yea, if you get to the jpg link everything is public |
| 15:19.57 | leifmadsen | stupid facebook |
| 15:20.03 | Katty | Qwell: aww. |
| 15:20.28 | Katty | Qwell: it's so cute n spotty :> |
| 15:22.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:25.21 | mjordan | Qwell: new dog? |
| 15:25.50 | Qwell | mjordan: yar. Not what I blame you for though. |
| 15:26.02 | mjordan | oic |
| 15:26.09 | mjordan | I'd take the blame for that though. Cute dog. |
| 15:26.27 | Qwell | Went to Cajun Cafe this weekend. It's < 5 minutes from me, across the river. I'd somehow never heard of it before. |
| 15:27.51 | navaismo | Katty, fine fine |
| 15:28.00 | Katty | navaismo: excellent :> |
| 15:28.26 | navaismo | and you? |
| 15:28.34 | Katty | oh just peachy, so far |
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| 15:30.11 | navaismo | peachy like the 3d printer? |
| 15:30.12 | mjordan | Qwell: I've driven by, never been; heard good things |
| 15:32.28 | Katty | navaismo: exactly! |
| 15:32.30 | Katty | hugs Penguin |
| 15:32.36 | *** join/#asterisk vlad_starkov (~vlad_star@81.213.74.232) |
| 15:32.43 | Penguin | squirms |
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| 15:46.26 | Pooh | Hi. I have a question about the pakages at http://packages.asterisk.org/centos/5/asterisk-1.8/ (yes, I know CentOS 5 is old, but it's not my choice to be using that system...) |
| 15:46.50 | Pooh | I want to build some modules which are not part of the standard RPMs (but are available in the source) |
| 15:47.12 | Pooh | how can I find out what compile-time options were used to build the binary RPM package? |
| 15:47.28 | Qwell | Pooh: look at the SRPM |
| 15:47.30 | Pooh | if I build my own modules from the SRPMS, and then try to load them into Asterisk, I get: |
| 15:47.59 | Qwell | Use the entire package you built, not just a single module. |
| 15:48.11 | Pooh | Module XXX was not compiled with the same compile-time options as this version of Asterisk |
| 15:48.57 | Pooh | hm, isn't it feasible to build the modules I need using the same options as the binary packages I've already installed, and then just drop them in place? |
| 15:49.02 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 15:49.49 | Pooh | I don't know how to find out what compile-time options were used (but this is presumably possible, because Asterisk can tell me they don't match) |
| 15:50.27 | *** part/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:50.30 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:50.41 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 16:28.11 | Farkie | Pooh, I'm using fedora 12 :P |
| 16:28.42 | Farkie | apparently that's because of a similar reason |
| 16:29.41 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 16:31.46 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 16:32.10 | Pooh | maybe I'm naive, but I expected that downloading the SRPM, and using buildrpm on it would generate the same as I get from downloading and installing the binary RPM |
| 16:32.46 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 16:32.55 | Pooh | (provided I get them both from the same repo, of course - in this case http://packages.asterisk.org) |
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| 16:42.37 | BeachBall | the music on hold is so soothing |
| 16:43.40 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
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| 17:08.11 | *** join/#asterisk traph (~traph@unaffiliated/traph) |
| 17:08.19 | traph | hi |
| 17:09.10 | traph | what are the downsides of installing asterisk from an official distro repository, then compiling it? |
| 17:09.38 | [TK]D-Fender | that isn't 2 things to compare so far. |
| 17:09.47 | [TK]D-Fender | distro repo = typically binary. |
| 17:09.57 | [TK]D-Fender | for which there is no "then compiling it" |
| 17:10.25 | traph | sorry, bad English, I guess |
| 17:11.13 | [TK]D-Fender | If it's "repo" vs "download source + compile", then there are clear reasons for either |
| 17:11.14 | traph | is it possible for all asterisk's functions to work when installed from a binary package? |
| 17:11.43 | [TK]D-Fender | When you get a package, who's to say it was BUILT with all the available functions? |
| 17:11.55 | traph | package maintainers |
| 17:12.12 | [TK]D-Fender | Package maintainers often suck. HARD. |
| 17:12.26 | [TK]D-Fender | I'd be pretty picky about the source |
| 17:12.55 | traph | what about official repositories? |
| 17:13.14 | [TK]D-Fender | Whose? |
| 17:13.54 | traph | ubuntu's |
| 17:14.18 | [TK]D-Fender | that isn't ASTERISK's official... go see what version they have. They are NOT "up to date" |
| 17:14.32 | [TK]D-Fender | Chech the topic.... |
| 17:15.16 | [TK]D-Fender | check* |
| 17:18.22 | mjordan | traph: Ubuntu packages have not been updated for some time. Additionally, Asterisk is a bit hard to package with "everything" available, since some options are mutually exclusive (for example, various voicemail storage backends) |
| 17:19.23 | traph | I see. That's why it's best to install and maintain asterisk manually. |
| 17:19.40 | mjordan | well, or at least know what your packages are :-) |
| 17:20.09 | mjordan | traph: we do make packages for CentOS, and I think there's usually a package variant for *most* people's needs, but I'm sure there are some people's needs that aren't met by them. |
| 17:20.53 | traph | I also see there is a debian repository, but it's not documented in the wiki |
| 17:21.17 | mjordan | traph: nope. The package maintainer hasn't chosen to document it on the wiki. |
| 17:21.40 | [TK]D-Fender | [12:12][TK]D-FenderPackage maintainers often suck. HARD. |
| 17:23.08 | traph | Got your point. Thanks guys! |
| 17:24.51 | [TK]D-Fender | traph: Now not being "bleeding edge" can be a good thing. |
| 17:25.36 | [TK]D-Fender | traph: Also packages may take you forward when you don't have a pressing reason to do so and risk introducing issues into a previously working environment. |
| 17:25.57 | [TK]D-Fender | traph: Source puts you in ful control, but a lot more manual to maintain. |
| 17:26.37 | jeffspeff | on a 30 minute call, these channelstats don't seem quite right. only 16 packets received and 24 packets sent? what causes this? 192.168.168.141 b750c747-85 00:30:31 0000000016 0000000000 ( 0.00%) 0.0000 0000000024 0000000023 (95.83%) 0.0002 |
| 17:28.59 | workingcats | hi, i was just checking the open ports on my asterisk server and i noticed 5000 is open for asterisk's UNISTIM... could someone give me a pointer how to turn it off? |
| 17:29.18 | Qwell | workingcats: Don't build that module. |
| 17:29.42 | workingcats | so since i already built it can i just delete the appropriate file? |
| 17:30.10 | Qwell | sure |
| 17:30.15 | [TK]D-Fender | workingcats: yes, or better still : "noload => chan_unistim.so" <- modules.conf |
| 17:31.08 | workingcats | ah great, thanks |
| 17:31.14 | workingcats | i'm guessing i can use that to turn off iax as well |
| 17:32.08 | [TK]D-Fender | yup |
| 17:32.25 | JeffC_NN | Does anyone know about function_DENOISE() https://wiki.asterisk.org/wiki/display/AST/Function_DENOISE |
| 17:32.27 | [TK]D-Fender | disabling instead of killing lets you change your mind later easily |
| 17:32.58 | JeffC_NN | specifically, would it work to remove quiet audio from a ulaw channel? |
| 17:33.10 | workingcats | yep that worked, thanks |
| 17:33.20 | Qwell | JeffC_NN: not really possible to answer. You'll have to try it. |
| 17:33.27 | workingcats | yeah and it just seems cleaner than "randomly" deleting/moving files |
| 17:34.25 | JeffC_NN | I'm not really asking about the quality/success of the function, but if it's worth using in general, and if there are any technical reasons why it should/shouldn't be used for my codec (ulaw) |
| 17:34.39 | *** join/#asterisk biox_ (~jesseolso@173-165-238-158-minnesota.hfc.comcastbusiness.net) |
| 17:35.49 | [TK]D-Fender | JeffC_NN: Quality is the point. If your signal is clean, it'll denoise something that isn't "noise and you might not hear the ocean on te other end of the line. |
| 17:36.25 | JeffC_NN | does anyone use AGC in combination with DENOISE? |
| 17:36.27 | [TK]D-Fender | JeffC_NN: If things are bad then you'll want to. BUt this sounds only valid if it's YOUR lines that are the issue, unless you make flipping it on/off a user choice. |
| 17:36.42 | JeffC_NN | oooh, good idea :D |
| 17:40.41 | [TK]D-Fender | DYNAMIC_FEATURE <- |
| 17:43.07 | JeffC_NN | ah, makes sense. |
| 17:48.50 | *** join/#asterisk xaristax (~xaristax@fixed-203-0-89.iusacell.net) |
| 17:49.17 | *** join/#asterisk Tokeiito (~quassel@78-58-144-254.static.zebra.lt) |
| 17:49.36 | xaristax | Hi :D i have a question if you can help me i have ${EXTEN:1} and i eat one digit from the begginin but what if i want the first 3 digits what can i do? |
| 17:52.34 | BeachBall | what is the cheapest toll-free DID provider? |
| 17:52.39 | BeachBall | i'm seeing 4.5cents |
| 17:52.47 | BeachBall | I want less cents |
| 17:53.23 | [TK]D-Fender | xaristax: ${var:TrimFromFront:TotalCharsTorReturn} |
| 17:53.36 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 17:53.42 | [TK]D-Fender | BeachBall: It's hard to imaging you making less sense :p |
| 17:54.05 | BeachBall | rolls quickly over to [TK]D-Fender |
| 17:54.13 | xaristax | Fender THANKS :D |
| 17:54.19 | BeachBall | glares |
| 17:54.35 | BeachBall | was kinda funny tho |
| 17:54.36 | [TK]D-Fender | quickly rolls over BeachBall, then backup over him to be sure |
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| 18:13.51 | *** join/#asterisk XandriX (xandrix@gateway/shell/anapnea.net/x-zjrygpfmfqobpueo) |
| 18:14.57 | XandriX | if i use old cisco phones like the 7940's on sip firmware and i set them to communicate with an asterisk server that is outside of my lan i will most likely encounter nat issues (is this statement correct ?) |
| 18:15.10 | *** join/#asterisk _MrB_ (~user@router2.hsdev.com) |
| 18:15.58 | workingcats | XandriX, only if there's a NAT inbetween |
| 18:16.59 | XandriX | well i mean voipserver ---> internet ---> firewall in question ---> lan ---> cisco phones |
| 18:17.10 | XandriX | so id think there would be a nat between the phones tne the voip server |
| 18:17.42 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
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| 18:19.49 | workingcats | it depends, really |
| 18:20.05 | workingcats | if its a private home, almost certainly |
| 18:20.41 | workingcats | if its a business... many businesses have enough "normal" routable IPs for every device so they don't need to do NAT (those lucky *******) |
| 18:21.05 | XandriX | home |
| 18:21.07 | XandriX | so yeah nat |
| 18:21.20 | *** join/#asterisk ionface (~ionface@unaffiliated/ionface) |
| 18:21.58 | workingcats | so you definitely have NAT. whether that often causes problems in your planned setup i'm afraid i can't answer but someone else can probably help |
| 18:22.08 | workingcats | *almost definitely |
| 18:22.10 | XandriX | cools |
| 18:22.15 | XandriX | baah worst case |
| 18:22.31 | XandriX | ill vpn between the voip server and my lan and or sort of bridge it |
| 18:23.52 | workingcats | if done properly that can be a clean solution |
| 18:23.59 | workingcats | unlike NAT, which by definition is FILTHY ;) |
| 18:24.09 | XandriX | workingcats: aggreed |
| 18:24.22 | XandriX | i hope it doesnt come to that but if i must will setup a vpn bridge and do it that way |
| 18:24.27 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:25.15 | XandriX | ill test to see |
| 18:25.22 | XandriX | once i get to the client tonight |
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| 18:38.07 | *** join/#asterisk u0m3 (~u0m3@109.96.184.215) |
| 18:38.19 | ipengineer | Is there a way to see if an exten exist for a given realtime context? |
| 18:43.59 | *** join/#asterisk u0m3 (~u0m3@92.80.94.155) |
| 18:47.11 | ChannelZ | does the DIALPLAN_EXISTS function not work for realtime? |
| 18:47.16 | ChannelZ | (I don't know, I don't use RT) |
| 18:47.33 | *** join/#asterisk calum_ (~calum_@host109-153-55-187.range109-153.btcentralplus.com) |
| 18:48.15 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:48.40 | ipengineer | ChannelZ: Not sure I need to take a look at it |
| 18:54.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:59.35 | rrittgarn | Shot in the dark: Anybody in here in the Chicagoland Area looking for work? Full time based out of the western burbs. |
| 19:02.51 | ChannelZ-Wk | Shoveling snow? |
| 19:10.30 | hardwire | heh |
| 19:10.36 | Katty | frowns |
| 19:10.44 | Katty | cdr-stats installs on a script now :< |
| 19:14.12 | Katty | sighs, digs through script |
| 19:15.11 | *** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com) |
| 19:17.49 | BeachBall | Q. what term would i search for, for automated calling thingy... for example a storm closes the office, I want to issue a call all employees and play the message |
| 19:18.44 | rrittgarn | you can either look into call files, or the AMI interface for originate depending on how you want to generate the calls |
| 19:19.13 | rrittgarn | and ChannelZ: We're having a heatwave didn't you hear?! its 37 outside! stuff is melting! |
| 19:19.30 | Docfxit | Penguin: Can I pm you? |
| 19:19.38 | BeachBall | Yes |
| 19:19.53 | *** join/#asterisk troyt (~troyt@2601:7:6200:1382:44dd:acff:fe85:9c8e) |
| 19:25.07 | *** part/#asterisk issackelly (sid18035@gateway/web/irccloud.com/x-makjwlrktsikacxj) |
| 19:26.07 | Docfxit | I had a hard drive fill up with Asterisk. Asterisk won't start now. I have cleared some space. What can I do to figure out what is stopping it from starting? |
| 19:26.51 | BeachBall | did u blow the dust off the hard drive? |
| 19:27.22 | Docfxit | BeachBall: It is an old install. |
| 19:27.33 | BeachBall | it will give error messages |
| 19:27.37 | BeachBall | check the log files |
| 19:27.46 | BeachBall | it will say why it can't start |
| 19:28.16 | Docfxit | BeachBall: I will see if I remember how to get into the log file. |
| 19:28.48 | navaismo | execute: asterisk -vvvvvvcg and see where it stop it |
| 19:29.15 | navaismo | for further info check /var/log/asterisk/full log |
| 19:30.56 | Docfxit | It starts with /var/run/asterisk/asterisk.pid no such file. |
| 19:31.29 | Docfxit | Unable to bind socket to /var/run/asterisk/asterisk.ctl. |
| 19:31.31 | navaismo | running asterisk -vvvvvvvcg?? |
| 19:31.44 | Docfxit | Yes. |
| 19:31.59 | Katty | i think. |
| 19:32.11 | Katty | this script is irritating. |
| 19:32.19 | Katty | whatever happened to doing things nonscripty :< |
| 19:32.49 | navaismo | which script? |
| 19:33.06 | Katty | asterisk-stat installs itself via a script now |
| 19:33.41 | Katty | and the script fails |
| 19:33.50 | navaismo | ah the cdrs stats! Yeah it sucks if you are not in centos or the other not shure if debian |
| 19:34.01 | Katty | it claims to not support debian |
| 19:34.06 | Katty | but it's deceit and lies |
| 19:34.12 | navaismo | yep |
| 19:34.28 | Katty | and i will make it work |
| 19:34.37 | Docfxit | navaismo: The folder /var/run/asterisk is not there. How can I get it back? |
| 19:35.16 | navaismo | i dont know, ow did you installed asterisk from sources? Which OS? |
| 19:35.34 | Docfxit | navaismo: Ubuntu. |
| 19:35.58 | navaismo | Katty, i was doing that but on the raspberry give up after 3 days compiling mongodb |
| 19:36.09 | Katty | well i will get it done. |
| 19:36.09 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:36.16 | Katty | and then i will make a length, cranky blog post about it |
| 19:36.23 | navaismo | Docfxit, from sources? |
| 19:36.24 | Katty | maybe even some fish shaking. |
| 19:36.26 | Katty | i mean fist. |
| 19:36.28 | *** join/#asterisk [RAG]Sharky (ragsharky@541CAD92.cm-5-5c.dynamic.ziggo.nl) |
| 19:36.36 | Katty | we've leaving the fish out of it, this time. |
| 19:36.51 | Docfxit | navaismo: All my tar files only have etc. I don't see any with var. |
| 19:37.18 | Chainsaw | Docfxit: So mkdir -p /var/run/asterisk && chown -R asterisk:asterisk /var/run/asterisk |
| 19:37.21 | Chainsaw | Docfxit: And then try again. |
| 19:39.40 | Docfxit | Chainsaw: Will that automaticlly put the proper files in the folder or should I try to undelete the folder? |
| 19:39.48 | Docfxit | Chainsaw: It's not in the trash. |
| 19:40.15 | protocoldoug | I've got a situation where I can see incoming RTP packets in a packet capture, but... I don't see them in a "rtp set debug on" (nor do I hear them / can I record them) any idea where to look next? |
| 19:40.22 | Chainsaw | Docfxit: It will write the PID file by itself, but the directory needs to be there and the ownership needs to be right. |
| 19:40.49 | Docfxit | Chainsaw: Great. I'll try it. |
| 19:41.18 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 19:42.29 | navaismo | protocoldoug, iptables maybe |
| 19:42.59 | protocoldoug | thank you navaismo -- however, I cleared out my iptables for this test |
| 19:43.14 | *** join/#asterisk [RAG]Sharky (ragsharky@541CAD92.cm-5-5c.dynamic.ziggo.nl) |
| 19:43.25 | Docfxit | Chainsaw: it came back with chown: 'asterisk:asterisk': invalid user. |
| 19:43.35 | navaismo | the packet capture was in the pbx? |
| 19:44.15 | protocoldoug | yep, did the packet capture on the same box. (about to post the pcap / sip.conf / rtp.conf on a pastebin, just a few sanitizing it) |
| 19:45.09 | Chainsaw | Docfxit: Try asterisk:dialout instead. |
| 19:45.28 | Chainsaw | Docfxit: If that doesn't do it either you'll need to pastebin me the init script. I am not fully familiar with how Debian package things. |
| 19:46.19 | Docfxit | Chainsaw: Invalid user. |
| 19:46.41 | Docfxit | Chainsaw: where can I find the init script? |
| 19:46.54 | Chainsaw | Docfxit: Hopefully in /etc/init.d, a text file by the name of asterisk. |
| 19:47.11 | Chainsaw | Docfxit: Upload it on the web somewhere and link me to it though, it's going to be 10+ lines. |
| 19:47.25 | *** join/#asterisk spillere (~spillere@molus.co) |
| 19:48.16 | spillere | I just installed for the first time a asterisk server, i'm trying to use linphone to connect to the server. On sips.conf I created something like register =>name:1234@myServerIp/s, how do I connect to the server after that? |
| 19:48.20 | Docfxit | Chainsaw: I have inetd.conf Would that be it? |
| 19:49.03 | protocoldoug | Here's my sip debug / rtp debug / cli output: http://pasteall.org/49677 |
| 19:49.07 | protocoldoug | pcap is at line 230 |
| 19:49.18 | protocoldoug | also sip.conf, rtp.conf and a "route -n" at the bottom, too |
| 19:49.41 | navaismo | WIMPy, did you know if its possible to have the queue_log in the file and in the DB(using realtime)? |
| 19:50.47 | WIMPy | I have never used realtime. |
| 19:51.14 | WIMPy | I stil try to get the basics working. With limited success. |
| 19:51.29 | navaismo | lol |
| 19:51.50 | WIMPy | is dead serious about that. |
| 19:52.01 | navaismo | O_O |
| 19:52.11 | Chainsaw | Docfxit: No. |
| 19:52.19 | Chainsaw | Docfxit: /etc/init.d/asterisk |
| 19:54.13 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 19:58.10 | Docfxit | Chainsaw: I have it for you at http://pasteall.org/49678. |
| 19:58.46 | Katty | navaismo: script altered! |
| 19:58.54 | Chainsaw | Docfxit: Could I have the output of "id asterisk" please. |
| 19:59.06 | navaismo | Katty, nice! |
| 19:59.43 | Katty | think it will bomb? |
| 20:00.03 | Chainsaw | Katty: No, I predict a flawless victory. |
| 20:00.06 | Docfxit | Chainsaw: Sure. Where would I find it? |
| 20:00.16 | Chainsaw | Katty: Nothing is as infuriating as a badly written script. |
| 20:00.22 | Katty | hehehehe |
| 20:00.26 | Katty | true story |
| 20:00.27 | Chainsaw | Docfxit: It's a command for you to run. I'm expecting numeric output. |
| 20:01.20 | Katty | MongoDB is now installed! |
| 20:01.32 | Docfxit | Chainsaw: It returned No such user. |
| 20:04.34 | Chainsaw | Docfxit: If they are running Asterisk as root then I am running away. That way lies madness. |
| 20:04.59 | Chainsaw | WIMPy: Weren't you a Debianite? |
| 20:05.31 | WIMPy | Is "ite" a pro or a con term? |
| 20:05.39 | Docfxit | Chainsaw: Please don't run away. I don't know how it was setup. |
| 20:05.57 | WIMPy | I'm usually on Slackware. |
| 20:06.32 | Chainsaw | WIMPy: It can be either. |
| 20:06.49 | Chainsaw | Docfxit: If it runs as root then just the mkdir -p I had you run should suffice. |
| 20:08.39 | Docfxit | Chainsaw: So I should run sudo mkdir -p /var/run/asterisk. |
| 20:08.49 | Docfxit | ? |
| 20:08.53 | Chainsaw | Docfxit: Yes. |
| 20:09.13 | Chainsaw | Docfxit: And you can omit the chown, because Asterisk runs as root. |
| 20:10.03 | spillere | i'm setting up a asterisk sip server, can I use ip's instead of domain names? for all configuration and real? |
| 20:11.06 | Docfxit | Chainsaw: I ran asterisk -vvvvvvcg what came back is Unable ot open pid file /var/run/asterisk/asterisk.pid Permission denied. |
| 20:11.29 | protocoldoug | navaismo: got it working -- it was bad routing tables. I wasn't using the proper gateway for the destination address. Thank you for putting in some thought, helped me out when I was feeling at the end of my rope :) |
| 20:11.29 | Chainsaw | Docfxit: Then you need to look at /etc/conf.d/asterisk and tell me what the asterisk user variable is set to. |
| 20:12.00 | navaismo | good to hear is fixed |
| 20:12.04 | navaismo | and no problem |
| 20:12.33 | Penguin | docfxit: Go ahead. It better not be spam. |
| 20:13.14 | Docfxit | Chainsaw: I don't have a folder called /conf.d in the etc folder. |
| 20:14.40 | Chainsaw | Docfxit: Pastebin of the asterisk output, if there is more then that one line? |
| 20:14.59 | Docfxit | Chainsaw: Ok. |
| 20:15.54 | *** join/#asterisk Guest44661 (~root@24-197-236-46.static.stpt.wi.charter.com) |
| 20:16.57 | Docfxit | Chainsaw: http://pasteall.org/49679. |
| 20:17.21 | Chainsaw | Docfxit: You need to sudo /etc/init.d/asterisk start |
| 20:17.30 | Chainsaw | Docfxit: Instead of trying to run asterisk as your own user. |
| 20:19.29 | Guest44661 | running trixbox ce here, trying to install rhino r4fxo and zaptel. I think I installed the driver properly, but zaptel always has make errors.. I have the latest rhino and zaptel versions..?? |
| 20:19.56 | navaismo | grabs popcorns |
| 20:20.24 | Katty | oh boy rhino |
| 20:20.30 | Katty | gets the pretzels |
| 20:20.44 | Guest44661 | am i dealing with incompatability here? |
| 20:21.06 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 20:21.09 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 20:21.24 | Katty | HHANNNNnnnn |
| 20:21.30 | Guest44661 | let me ask you this, if you where working with trixbox ce, what would be the easiest hardware to get functional on it? |
| 20:21.34 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 20:21.55 | Katty | Guest44661: i think i'd just install asterisk and be done with it |
| 20:22.01 | Katty | Guest44661: personally... |
| 20:22.22 | Docfxit | Chainsaw: It came back with Starting Asterisk PBX: asterisk but the extensions are not ringing. |
| 20:23.16 | Guest44661 | katty, i thought i had asterisk installed...?? |
| 20:23.28 | Katty | Guest44661: did you now? |
| 20:23.30 | Penguin | ~trixbox |
| 20:23.30 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 20:23.37 | Katty | Guest44661: i guess thoughts can be deceiving |
| 20:24.46 | Katty | now then, minster Penguin |
| 20:24.51 | Katty | Penguin: how're you, dear? |
| 20:25.04 | Guest44661 | katty, ok interesting. I somehow came under the assumption that trixbox was the linux distro that ran asterisk... so is asterisk a linux distro unto itself? |
| 20:25.06 | Penguin | I'm good. |
| 20:25.22 | Penguin | I'm shoving some French bread pizza in my face before I have to take off. |
| 20:25.55 | Katty | Guest44661: neither are distributions. |
| 20:25.55 | Penguin | ~asterisk |
| 20:25.55 | infobot | Asterisk is an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org/ |
| 20:26.05 | Katty | distrobutions. |
| 20:26.08 | Katty | however you spell that word. |
| 20:26.19 | Katty | Penguin: mmm, pizza. where are you going? |
| 20:26.30 | Guest44661 | katty let me phrase that better. what OS runs asterisk the best..?? |
| 20:26.32 | chuckf_ | Guest44661: asterisknow is probably what you are looking for if you want a 'distro' |
| 20:26.43 | Katty | Guest44661: the one that you are most familiar with |
| 20:26.54 | Penguin | I've got an afternoon meeting. |
| 20:26.58 | Katty | eww. meetings :< |
| 20:27.01 | Katty | hugs chuckf_ |
| 20:27.15 | chuckf_ | hugs Katty |
| 20:27.21 | Katty | how're you dear? and the wifey? |
| 20:27.53 | chuckf_ | We're doing pretty good. She's got a job interview Friday |
| 20:27.53 | Penguin | guest44661: I believe Digium compiles packages for both CentOS and Debian, but it will run on almost any Linux-based OS. |
| 20:27.54 | Guest44661 | katty ok, so if i dump ubuntu 12.04 server on this machine, asterisk is going to run without difficulty? i apparently have my wires crossed, i somehow thought that trixbox was asterisk's special linux distro..?? |
| 20:28.23 | Qwell | "special", as in the Olympics. |
| 20:28.25 | Katty | chuckf_: oooh! wish her luck for me, if you think it appropriate |
| 20:28.32 | newtonr | Penguin, only CentOS packages https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 20:28.36 | chuckf_ | I'll do that |
| 20:28.51 | chuckf_ | How are things in your world? |
| 20:29.06 | Katty | Guest44661: I think perhaps the key thing that you're overlooking, is that this isn't going to be Easy(tm) |
| 20:29.21 | Katty | Guest44661: you will have to learn some things, and no doubt read. |
| 20:29.55 | Guest44661 | katty ok, let me say "easiest"... in other words, as less "hard" as is possible. |
| 20:30.07 | Penguin | I know for a fact that someone provides debs... I just don't know who. |
| 20:30.08 | Katty | Guest44661: well the easiest way is probably to look into the Digium appliances. |
| 20:30.32 | Guest44661 | penguin, that would make my life much easier. food for thought. |
| 20:30.50 | Katty | chuckf_: oh, you know. busy |
| 20:30.54 | Katty | chuckf_: better than dull tho, i suppose! |
| 20:31.37 | Katty | does digium have hosted solutions yet? |
| 20:31.48 | Guest44661 | katty thanks, i will dig around some more. I only installed trixbox because I somehow thought this was the standard in asterisk installations. (dont ask me how I managed this.) |
| 20:32.19 | Katty | Guest44661: I would say the "standard" is to use asterisk on your platform of choice |
| 20:32.30 | Katty | Guest44661: but that will no doubt take time to learn. |
| 20:32.52 | Guest44661 | katty good to know. i hate centos based distros with a passion. give me apt-get debian nased goodness any day of the week.\ |
| 20:35.22 | Katty | well, bye then :< |
| 20:38.47 | chuckf_ | he didn't want to learn anyway |
| 20:41.45 | Katty | he could have tried osmosis at least! |
| 20:41.48 | Katty | puts a book on chuckf_'s head |
| 20:41.58 | Katty | chuckf_: do you feel smarter? :> |
| 20:48.51 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 20:52.26 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 20:53.01 | chuckf_ | I always feel smarter around you Katty |
| 20:53.59 | *** join/#asterisk MaliuatLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 21:00.22 | JeffC_NN | When someone transfers an incoming DAHDI call from one SIP extension to another (mixed locally I think), my console gets about 10 of these per second: WARNING[5864][C-00000b77]: abstract_jb.c:284 ast_jb_put: SIP/7001-00000dcc received frame with invalid timing info: has_timing_info=0, len=0, ts=0, src=dahdi_read |
| 21:00.39 | JeffC_NN | Is that normal, or something I should be concerned about? |
| 21:19.00 | Katty | chuckf_: that's cause you're a smart cookie! |
| 21:21.07 | *** join/#asterisk xaristax (~xaristax@fixed-203-0-89.iusacell.net) |
| 21:21.15 | xaristax | if i set the phone to re register every 2 minutes is a problem? |
| 21:21.18 | xaristax | or why every phone that i see have for default 3600? |
| 21:21.45 | spillere | i have to users in sips.conf user001 and user002, in extensions, how do i make them connect to talk? |
| 21:24.09 | JeffC_NN | xaristax: 3600 is seconds, I believe. I don't think 120 (2 minutes) would be a problem, assuming you don't have too many extensions, and your * server is running on decent hardware. |
| 21:24.15 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:25.02 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:25.09 | JeffC_NN | spillere: check out Chapter 5 of the Asterisk book: http://oreilly.com/catalog/asterisk/chapter/ch05.pdf |
| 21:26.22 | xaristax | thanks JeffC_NN im a little bit newbie on this but if the re register do not affect the call i mean if im trying to locate a user in that re register it will be available |
| 21:26.23 | xaristax | ??? |
| 21:27.33 | spillere | JeffC_NN: thanks |
| 21:27.58 | JeffC_NN | re-registering is just checking in with the phone server. It doesn't affect the extensions ability to be located, unless you don't re-register in the timeout specified in Asterisk. What are you trying to accomplish? |
| 21:29.03 | JeffC_NN | xaristax: what are you trying to gain by setting a fast register timeout? |
| 21:30.12 | Docfxit | I don't seem to have a user named asterisk on my system for some reason. If I create a user named asterisk should it be a desktop user or administrator? If I assign it a password what file/files needs that updated? |
| 21:30.51 | WIMPy | Are you running Asterisk on Windows? |
| 21:31.03 | Docfxit | In ubuntu. |
| 21:31.04 | Docfxit | Ubuntu. |
| 21:31.25 | WIMPy | makes a mental note to avoid ubuntu. |
| 21:32.13 | xaristax | JeffC_NN: im trying to do this because sometimes i cannot locate the extension even if is register |
| 21:32.24 | Docfxit | WIMWY: It's an old system that I didn't setup. I do need to update it. |
| 21:33.02 | [TK]D-Fender | [16:30]DocfxitI don't seem to have a user named asterisk on my system for some reason. If I create a user named asterisk should it be a desktop user or administrator? If I assign it a password what file/files needs that updated? <- "desktop" |
| 21:33.04 | JeffC_NN | Docfxit: This guide shows how to setup asterisk using an account that has no password: http://wiki.freepbx.org/pages/viewpage.action?pageId=1409028#InstallingFreePBXonUbuntu12.04Server(PrecisePangolin)-NowcreatetheAsteriskuserandsetownershippermissions. |
| 21:33.25 | [TK]D-Fender | Docfxit: You should understand what that implies and clear given the reason for running * as its own user |
| 21:35.08 | Docfxit | Thanks. |
| 21:36.39 | [TK]D-Fender | The whole point of running * as non-root .... is so they CAN'T administer and screw-up your system |
| 21:37.08 | [TK]D-Fender | checkout time, BBIAB |
| 21:40.58 | Docfxit | I followed the instructions on the link: I'm getting an error. Sudo adduser --disabled-password --no-create-home --gecos "Asterisk User" |
| 21:40.58 | Docfxit | adduser: Only one or two names allowed. |
| 21:47.52 | MaliuatLap | Docfxit: well there is no username for starters, and not enough commas in the gecos :) |
| 21:48.12 | MaliuatLap | Docfxit: man adduser |
| 21:48.23 | MaliuatLap | Docfxit: or "man useradd" |
| 21:48.32 | MaliuatLap | they are different utilities |
| 21:49.45 | Docfxit | MaliuatLap: I was just following instructions at: http://wiki.freepbx.org/pages/viewpage.action?pageId=1409028#InstallingFreePBXonUbuntu12.04Server%28PrecisePangolin%29-NowcreatetheAsteriskuserandsetownershippermissions. |
| 21:50.06 | Docfxit | I'm running Ubuntu. |
| 21:50.22 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:50.23 | MaliuatLap | Docfxit: RTFM |
| 21:50.48 | MaliuatLap | Docfxit: and by that I mean the "adduser" and "usseradd" M's |
| 21:51.22 | *** join/#asterisk danjenkins_ (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:51.22 | MaliuatLap | Docfxit: I'm also sorry you're running Ubuntu ... man up and run real Debian ;P |
| 21:51.35 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 21:51.44 | vastina | MaliuatLap-- |
| 21:51.53 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 21:52.16 | MaliuatLap | vastina: I have issues with Canonical making money off my packaging efforts :) |
| 21:52.22 | vastina | disdain of an OS distribution is childish |
| 21:52.23 | WIMPy | Are you saying ubuntu is even worse than debian? |
| 21:52.23 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:52.44 | leifmadsen | MaliuatLap: then you have an issue with open source in general |
| 21:52.54 | xaristax | im getting this error with an external phone chan_sip.c:25557 handle_request_invite: Failed to authenticate device |
| 21:52.56 | Docfxit | MaliuatLap: Great suggestion. When I create a new install I'll keep that in mind. |
| 21:52.58 | xaristax | someone can help me? |
| 21:53.46 | MaliuatLap | Docfxit: knowing how to add users to your system is a pretty basic sysadmin thing ... should be one of the first things you figure out :) |
| 21:54.01 | WIMPy | xaristax: Use a valid account. |
| 21:54.15 | xaristax | im using a valid account |
| 21:54.17 | MaliuatLap | WIMPy: Debian is the better of the Linux distro's I've worked with |
| 21:54.29 | xaristax | i even register and all goes right but when i dial i get this |
| 21:54.47 | WIMPy | And about the distro thing I have a very simple idea: If the kind of Distro you use makes a difference, it's the wrong one. |
| 21:55.14 | Docfxit | MaliuatLap: I didn't creat this install. I don't claim to be a sysadmin. I'm just in charge of it now. I'm trying to learn as fast as I can. |
| 21:55.14 | vastina | agreed |
| 21:55.23 | WIMPy | xaristax: That message states the opposite. |
| 21:55.48 | xaristax | WIMPy yep and only happends with external phones |
| 21:55.58 | MaliuatLap | WIMPy: I have moved away from various distros as they made it harder to manually conf things (i.e. tried pushing people to their GUI tools) |
| 21:56.14 | MaliuatLap | WIMPy: Debian was the one I found that didn't really do that |
| 21:56.15 | WIMPy | xaristax: Maybe the account you use is IP restricted? |
| 21:56.38 | xaristax | nop permit=0.0.0.0/0.0.0.0 |
| 21:56.44 | xaristax | its very extrange |
| 21:56.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:56.54 | WIMPy | MaliuatLap: That's why I'm on Slackware. It has become worse as well, but nit as bad as others. |
| 21:56.55 | vastina | MaliuatLap: that's a failure of your intimacy with $distribution, and is absolutely irrelevant in here besides. |
| 21:56.58 | xaristax | its there a way to make a sip debug only from one extension |
| 21:57.25 | WIMPy | xaristax: No, but you can restrict it to one IP. |
| 21:57.35 | MaliuatLap | Docfxit: well blindly following instructions off of a random website is not good practice ... cookie cutter sysadmins tick me off. :) Learn to read the man pages, and read as much as you can about something before dicking with it :) |
| 21:57.36 | [TK]D-Fender | xaristax: "sip set debug peer [thepeer]" |
| 21:57.37 | WIMPy | And that can be done via the name of a peer as well. |
| 21:57.38 | [TK]D-Fender | xaristax: Never call it "extension" |
| 22:08.43 | spillere | I managed to connect two user using Dial(), but when connection is done, I can hear anything. ANy pointer on what could be the problem? |
| 22:11.48 | ChannelZ-Wk | networky |
| 22:12.26 | [TK]D-Fender | netnoworky :) |
| 22:12.29 | ChannelZ-Wk | Your packets are either going to undesireable places or being blocked by firewalls, either case possibly by configuration. |
| 22:12.41 | ChannelZ-Wk | (mis)configuration |
| 22:13.29 | spillere | thanks! |
| 22:15.11 | *** part/#asterisk jsarrel (~jsarrells@24-158-61-198.static.hckr.nc.charter.com) |
| 22:17.19 | spillere | ChannelZ-Wk: need to add some NAT configuration?! |
| 22:17.33 | [TK]D-Fender | If involved, clearly |
| 22:19.18 | spillere | i have a VPN running asterisk, and im trying to connect two computers @ my home, in which I have a modem and router, so I could guess that it could be a problem right?! |
| 22:19.24 | spillere | anything else I should look into? |
| 22:19.54 | [TK]D-Fender | that actual comms <- |
| 22:19.57 | [TK]D-Fender | the* |
| 22:20.56 | spillere | comms? |
| 22:21.58 | [TK]D-Fender | "sip set debug on" <- Go look at what it is actually DOING |
| 22:26.11 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:26.42 | spillere | oh, great, ill try |
| 22:34.11 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:34.56 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 22:36.21 | spillere | [TK]D-Fender: http://pastie.org/8750070 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) |
| 22:36.27 | spillere | is it a problem? |
| 22:36.45 | [TK]D-Fender | Don't focus on little things like that. You need to see thowe whole thing |
| 22:36.53 | ChannelZ-Wk | you cut out all the most important bits |
| 22:36.59 | [TK]D-Fender | And that means looking at the WHOLE CALL |
| 22:37.07 | spillere | one sec, let me try to get it |
| 22:38.11 | spillere | is this enough? http://pastie.org/8750078 |
| 22:41.16 | [TK]D-Fender | That is not a call |
| 22:41.22 | [TK]D-Fender | that is a bunch of options packets. |
| 22:41.46 | [TK]D-Fender | Where do you see "oh, call from outside, trying to match"? |
| 22:41.54 | [TK]D-Fender | And stuff actually processing? |
| 22:42.45 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:43.35 | spillere | it rings the other phone |
| 22:43.40 | spillere | just no audio is transfered |
| 22:43.48 | spillere | i managed to play the "hello world" audio |
| 22:44.15 | [TK]D-Fender | that is not call debug |
| 22:45.37 | Docfxit | I am having trouble getting the phones to register. My log showing the startup is at http://pasteall.org/49681. |
| 22:46.03 | spillere | [TK]D-Fender: how can I make a log to show to you? |
| 22:46.26 | *** join/#asterisk jpoz (~jpoz@184.169.152.1) |
| 22:46.41 | [TK]D-Fender | place an actual call. Get the output from beginning to end of that call. |
| 22:47.33 | spillere | ok, let me try |
| 22:47.36 | rrittgarn | @Docfxit looks like you already have an asterisk process running. pkill asterisk |
| 22:47.55 | rrittgarn | or something else bound to those ports |
| 22:58.56 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 23:02.01 | Docfxit | rrittgarn: Thank you. I ran pkill and then started asterisk again. I have the output here http://pasteall.org/49684. |
| 23:09.08 | rrittgarn | Docfxit looks like you have errors in your sip.conf, as well as your extensions.conf |
| 23:10.23 | Docfxit | rrittgarn: Would you take a look at them for me? |
| 23:15.04 | [TK]D-Fender | Docfxit: I'm not seeing you look at actual registration attempts. |
| 23:15.52 | *** join/#asterisk lorsungcu (~anonymous@74-36-135-149.dr02.brvl.mn.frontiernet.net) |
| 23:16.53 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:17.33 | Docfxit | <[TK]D-Fender>: How can I fix that? |
| 23:17.42 | [TK]D-Fender | LOOK <- |
| 23:17.47 | [TK]D-Fender | "sip set debug on" |
| 23:20.37 | Docfxit | <[TK]D-Fender>: it returned sip: usage: and a bunch of possibilities non with debug. |
| 23:20.51 | Docfxit | Non = none. |
| 23:21.12 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 23:21.37 | [TK]D-Fender | then pick the equivalent |
| 23:22.17 | Docfxit | sip: Usage: sip [-h] [-V] [-a file] [-c dir] [-d file] [-e] [-g] [-I dir] [-j #] [-m file] [-p module] [-r] [-s suffix] [-t tag] [-w] [-x feature] [-z file] [file] |
| 23:22.50 | [TK]D-Fender | ..... that is not Asterisk CLI |
| 23:23.25 | Docfxit | [TK]D-Fender: How can I get into CLI? |
| 23:23.28 | [TK]D-Fender | ... |
| 23:24.01 | [TK]D-Fender | asterisk -rvvvvvvvv |
| 23:25.19 | Docfxit | I get an error saying: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 23:25.43 | *** part/#asterisk mjordan (~matt@nat/digium/x-swnqsdshiuegqvrf) |
| 23:26.07 | WIMPy | You probably just don't have permissions to access it. |
| 23:27.29 | Docfxit | I tried with sudo. Same results When I look in /var/run/asterisk I don't find the file asterisk.ctl. |
| 23:29.06 | WIMPy | Then 'find' it. |
| 23:29.32 | WIMPy | Anyway it looks a little messed up if it doesn't know where it is. |
| 23:29.33 | ChannelZ | (just popping in) is asterisk even running? |
| 23:29.42 | WIMPy | Unless your Asterisk isn't running. |
| 23:30.06 | ChannelZ | sounds not |
| 23:30.08 | ChannelZ | ps ax |grep asterisk |
| 23:30.26 | ChannelZ | or pidof asterisk |
| 23:31.03 | [TK]D-Fender | http://pasteall.org/49684 <- where did THIS come from if you can't get to CLI? |
| 23:31.33 | [TK]D-Fender | Asterisk Ready. |
| 23:31.34 | [TK]D-Fender | *CLI> |
| 23:31.36 | [TK]D-Fender | You're AT CLI there |
| 23:31.47 | [TK]D-Fender | ]How did you ge there the first time and NOT know just now? |
| 23:31.59 | [TK]D-Fender | Let alone fail... |
| 23:33.06 | Docfxit | [TK]D-Fender: I started it. Now it's running. |
| 23:33.18 | Docfxit | I'm in the CLI. |
| 23:34.09 | [TK]D-Fender | So you have plenty of dialplan errors, cdr config errors, users.conf seems to have plenty of issues ( reeks of AsteriekGUI |
| 23:34.22 | [TK]D-Fender | Go look at actual comm attempts |
| 23:35.20 | Docfxit | With sip set debug it is asking for either sip set debug IP or sip set debug peer. |
| 23:35.34 | WIMPy | or "on" |
| 23:36.03 | Docfxit | I tried on. This is the result. |
| 23:37.03 | [TK]D-Fender | if you're on anANCIENT version it would be "sip debug on". |
| 23:37.20 | WIMPy | How ancient would that be? |
| 23:38.24 | Docfxit | 1.4.22. |
| 23:38.34 | [TK]D-Fender | GAH |
| 23:38.45 | [TK]D-Fender | so use the old syntax |
| 23:38.52 | WIMPy | I guess, you'd better hire a historian. |
| 23:39.04 | sruffell | smirks |
| 23:39.44 | WIMPy | Better don;t mess with that installation. It might have listed status. |
| 23:41.03 | Chainsaw | I'm aware of 1.2 installations. |
| 23:41.23 | Chainsaw | I've been bribed not to tell. |
| 23:41.39 | WIMPy | Yes, I know someone still using 1.2 as well. He hasnt touched it since it was current. |
| 23:43.03 | Docfxit | WIMPY: This is a business that is trying to get the phones working. I'm trying to get them to update to a new PC. Until I can get that accomplished I need to keep this install up and running. I'm sorry for all the headache of questions from an old install. |
| 23:43.48 | [TK]D-Fender | Docfxit: I gave you the old syntax. get debug |
| 23:48.26 | Docfxit | [TK]D-Fender: I have debug on now. It keeps flashing new lines. I will try to grab something and past it for you. |
| 23:51.47 | *** join/#asterisk xaristax (~xaristax@fixed-203-0-89.iusacell.net) |
| 23:52.59 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 23:53.22 | xaristax | Hi i have the next problem i have some external phones connected to my pbx with nat at some point they change the status no UNREACHABLE so i force the phones to register every 120 seconds but is a game between asterisk and the phone asterisk set the status to unreachable and the phone reconects and its OK |
| 23:53.29 | xaristax | why is this? |
| 23:54.10 | Docfxit | [TK]D-Fender: I have a sample of the debug output at http://www.pasteall.org/49685. |
| 23:54.34 | Docfxit | [TK]D-Fender: Thank you very much for trying to help with this. |
| 23:55.01 | [TK]D-Fender | get more |
| 23:55.17 | Docfxit | K. |
| 23:57.19 | Docfxit | [TK]D-Fender: this is everything in the cash http://www.pasteall.org/49686. |
| 23:57.31 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:58.36 | [TK]D-Fender | looking clearly bad auth.... |
| 23:58.45 | [TK]D-Fender | And a single device |
| 23:59.01 | [TK]D-Fender | oops, multiple now |
| 23:59.16 | [TK]D-Fender | go prove your peers are there and loaded, go check all your configs |
