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| 01:28.37 | linuxfool | Hey everyone |
| 01:31.19 | TazzNZ | hi |
| 01:37.03 | WIMPy | lo |
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| 02:43.02 | linuxfool | What are some candidates for asterisk-based open source gui managers for a pbx? |
| 02:43.11 | linuxfool | totally free and open source ones |
| 02:43.17 | linuxfool | no warranty, no support |
| 02:45.03 | WIMPy | If you don't want support you can take all of them. |
| 02:45.44 | WIMPy | I heard with FreePBX there's even some chance for support, though not in here. |
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| 02:49.53 | linuxfool | Are you rinning Asterisk WIMPy? |
| 02:50.47 | WIMPy | Rinning? Is that a term for letting things go down the gutter? ;-) |
| 02:50.59 | WIMPy | Yes, I'm running serveral instances. |
| 02:54.39 | TazzNZ | lol |
| 02:54.57 | linuxfool | Ha! Oops. How are you using it? Just for fun, or for business? |
| 02:55.17 | WIMPy | both |
| 02:56.10 | WIMPy | Where fun is more like "fun". |
| 03:21.18 | linuxfool | Are you running virtual instances or do you have a dedicated box for them? |
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| 03:21.45 | WIMPy | Only one of them is virtual. |
| 03:25.01 | linuxfool | How many extensions does your largest instance have? |
| 03:28.24 | WIMPy | 5 permanent ones, which is my home box. |
| 03:29.58 | WIMPy | Most are used as add-ons to PBXs. |
| 03:32.52 | linuxfool | How so WIMPy? You're running them on top of existing systems? |
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| 03:52.02 | WIMPy | "on top"? More like alongside. |
| 03:54.42 | linuxfool | What systems are they running alongside of? |
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| 04:00.27 | WIMPy | Mostly HiPath. |
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| 08:27.41 | bleikrone | Hi |
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| 08:31.19 | t4nk707 | hello everyone |
| 08:32.00 | t4nk707 | I am new to asterisk. I kind of jumped on it a couple of years back |
| 08:32.35 | t4nk707 | I just install asterisk 12 and I'm having issues with making calls |
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| 08:33.15 | t4nk707 | is there anyone who can answer a couple of questions for me |
| 08:38.17 | t4nk707 | anyone here? |
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| 08:42.59 | t4nk707 | hi |
| 08:44.06 | t4nk707 | can anyone tell me why I can make calls from one of my endpoints, but can't receive. everything is setup the same way as another one which can do everything |
| 08:44.36 | t4nk707 | on one of my endpoints can make and receive calls |
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| 09:09.35 | sekil | probably registration problem |
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| 10:31.25 | gavimobile | I've been having a lot of trouble with making calls lately. I'm not sure if the issue is me or my tisp. here is a call I made which took a minute in a half to initiate. usually the call will go through within 10 seconds. http://pastebin.com/qmuvwSSc |
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| 10:57.06 | michael_work | file, we are from different timezones :( |
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| 11:57.55 | SirLagz | ok, so with pfSense - static port didn't work. |
| 11:58.05 | SirLagz | or at least pfSense with static port setup by itself didn't work |
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| 12:15.41 | jameswf | https://doc.pfsense.org/index.php/Asterisk_VoIP |
| 12:15.57 | SirLagz | jameswf: yep, I've already done most of that. |
| 12:16.08 | jameswf | "Most of" |
| 12:16.18 | marsmars | Hi, i have got some troble with my gsmgateway. I used raspbx on a Raspberry Pi with deactivated freepbx. The Channel chan_dongle seems to be runnig. The Incomming calls comes from an Fritz!Box Router modified with Freetz an Asterisk 11 via IAX2: While debug i found only "No such context/extension" |
| 12:16.19 | SirLagz | jameswf: the only thing I haven't done is using the SIP proxy |
| 12:16.46 | SirLagz | jameswf: There's a few things that some other people pointed out that I'm doing first like the static port mappings |
| 12:17.52 | jameswf | I think pfsense does some stupid stateful crap that hoses things which is why they do sipprox |
| 12:18.40 | SirLagz | jameswf: yeah, there are ways around it apparently. I don't particularly want to run the sip proxy on my pfSense box but if I have to, I will |
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| 12:30.08 | [TK]D-Fender | marsmars: So go point it to a context that has the extension it's looking for. |
| 12:30.49 | [TK]D-Fender | SirLagz: And you've diabled port randomization? |
| 12:30.53 | [TK]D-Fender | disabled* |
| 12:32.20 | marsmars | [TK]D-Fender: I have some Config here http://pastebin.com/i0tFGHdX posted; The context seems to be OK |
| 12:32.51 | [TK]D-Fender | marsmars: Show us the call debug |
| 12:33.05 | SirLagz | [TK]D-Fender: yes i did |
| 12:33.14 | SirLagz | [TK]D-Fender: but perhaps I did it wrong, so I updated it and I'm trying again for another day |
| 12:33.19 | [TK]D-Fender | marsmars: And no, ther is no match from what I see |
| 12:33.28 | [TK]D-Fender | marsmars: Look at the exten you told it to dial in. |
| 12:33.30 | marsmars | at pastebin or here? |
| 12:34.25 | [TK]D-Fender | marsmars: pastebin gives some clear signs. Pastebin the call... |
| 12:36.18 | marsmars | http://pastebin.com/1QKX5kNE |
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| 12:39.12 | [TK]D-Fender | marsmars: CALLED NUMBER : 02104991140 USERNAME : to_Freetz |
| 12:39.33 | [TK]D-Fender | marsmars: [tomobile] <--- what line in http://pastebin.com/i0tFGHdX is supposed to match that number? |
| 12:41.46 | gavimobile | I've been having a lot of trouble with making calls lately. I'm not sure if the issue is me or my tisp. here is a call I made which took a minute in a half to initiate. usually the call will go through within 10 seconds. http://pastebin.com/qmuvwSSc |
| 12:46.18 | marsmars | [TK]D-Fender: exten => s,2,1,Dial(DONGLE/dongle0/${EXTEN}) should, but I think the s is wrong. is "_." better ??? |
| 12:46.27 | [TK]D-Fender | marsmars: It isn't looking for "s" |
| 12:46.40 | [TK]D-Fender | marsmars: it's looking for a match for 02104991140 |
| 12:47.20 | [TK]D-Fender | marsmars: "_." is a terrible pattern. It'll match all sorts of dangerous things. |
| 12:47.51 | [TK]D-Fender | marsmars: Make a sane match for what appropriate number patterns you should actually actu on. |
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| 12:54.41 | marsmars | [TK]D-Fender: Thanks. Pattern is now _0XXXXXX |
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| 13:23.22 | Katty | MARSHMALLOW FLUFF |
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| 13:49.26 | widler | can anyone shine some light on why only one of my end points can receive calls |
| 13:49.40 | widler | asterisk 12 |
| 13:49.46 | [TK]D-Fender | Show us the call |
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| 13:52.03 | anonymouz666 | I am back |
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| 14:00.55 | widler | GNU nano 2.0.9 File: extensions.conf [users] exten=>6002,1,Dial(PJSIP/HPL,20) exten=>2506,1,Dial(PJSIP/2506,20) exten=>2003,1,Dial(PJSIP/GT3,20) [ Wrote 4 lines ] [r |
| 14:01.35 | [TK]D-Fender | widler: pastebin the actual call and device status dumps, etc... |
| 14:01.37 | [TK]D-Fender | ~pb |
| 14:01.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:01.40 | [TK]D-Fender | ^^^ |
| 14:01.48 | widler | sorry about that. on my mobile |
| 14:03.14 | gavimobile | [TK]D-Fender: could you have a look at my pastebin. the call in that output took 1 minute and 30 seconds to go through. I just need to verify its it's a local or provider issue |
| 14:05.45 | [TK]D-Fender | gavimobile: You can clearly see them giving you a "trying" very quick. Delay is all on their side based on their setup and where you're calling |
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| 14:09.12 | gavimobile | [TK]D-Fender: thanks |
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| 14:18.21 | widler | sorry it took a while. I had to get my pc. http://pastebin.com/J78HupKz |
| 14:18.36 | widler | here is the pastebin to my problem |
| 14:21.22 | widler | on one channel and can make and receive calls, but the others can only make calls |
| 14:22.19 | file | you need to do "pjsip set logger on" and pastebin it with an attempt, so it shows the SIP signaling and where the traffic is going |
| 14:22.43 | file | and the pjsip.conf for the device(s) in question, including AORs |
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| 14:35.43 | anonymouz666 | anyone in here using pjsip heavily in production? |
| 14:35.55 | anonymouz666 | I would like to hear a feedback about it |
| 14:36.37 | file | a better way of asking that would be "anyone in here using chan_pjsip heavily in production?" |
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| 14:36.42 | file | I'm sure pjsip itself is |
| 14:37.17 | anonymouz666 | s/pjsip/chan_pjsip |
| 14:37.27 | anonymouz666 | done! |
| 14:37.30 | anonymouz666 | heh. |
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| 14:54.07 | P-NuT | Hey all, I'm using the ast_tls_cert script to generate the TLS CA and server cert, but the CN is always "Asterisk Private CA". This is not good for clients that need to verify the cert's CN validity, is there a way to change this? |
| 15:01.50 | widler | AORs pastebin is http://pastebin.com/55pNDGP7. there was a lot of info I wasn't sure how much would be needed |
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| 15:07.42 | c|oneman | what are some tiny business class VoIP phones? for desk real estate or portability etc |
| 15:08.12 | mbowie | Softphones... super small surface area. |
| 15:08.28 | mbowie | Virtually non-existent. |
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| 15:18.12 | [TK]D-Fender | c|oneman: Polycom IP 335, etc |
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| 15:20.48 | gavimobile | [TK]D-Fender: I just tried with another provider and the same thing is happening. 54 seconds for a call to go through http://pastebin.com/AfrZJgyd |
| 15:21.00 | gavimobile | is there any possiblity that the information you gave me was wrong? |
| 15:21.11 | gavimobile | two providers with problems? |
| 15:21.16 | [TK]D-Fender | gavimobile: Check anything between your server and your provider.... |
| 15:21.25 | [TK]D-Fender | gavimobile: because we see their ack early on... |
| 15:21.36 | gavimobile | [TK]D-Fender: such as my router? |
| 15:21.55 | [TK]D-Fender | Is that between your server & your provider? |
| 15:22.00 | gavimobile | yes |
| 15:22.21 | gavimobile | the weird thing is that land line calls work instantaniously |
| 15:22.40 | gavimobile | ill reboot my router |
| 15:22.44 | gavimobile | its a brand new cisco routetr |
| 15:23.13 | c|oneman | polycom provisionning is a real pita |
| 15:23.37 | gavimobile | nevermind.. land is also taking time |
| 15:23.45 | gavimobile | im suspicious its something to do with the war |
| 15:23.49 | gavimobile | here in the middle east |
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| 15:25.58 | [TK]D-Fender | Which one? There are so many these days.... |
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| 15:26.17 | [TK]D-Fender | c|oneman: I've been doing it for just under a decade.... works fine for me... |
| 15:26.48 | c|oneman | takes me about 15 minutes to setup a phone with polycom |
| 15:27.04 | [TK]D-Fender | Takes me about 1 |
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| 15:27.35 | [TK]D-Fender | There are also scripts to generate those if you want to go that route. |
| 15:27.40 | [TK]D-Fender | I do mine all by hand |
| 15:28.13 | c|oneman | do you use the webui or do you push via ftp or seomthign |
| 15:28.37 | [TK]D-Fender | provisioning exclusively means "no webui" |
| 15:30.04 | [TK]D-Fender | I make un-box my phone, grab the MAC, make my configs, assemble the phone, plug it in, grab a coffee, and come back to a fully functional phone. |
| 15:30.10 | [TK]D-Fender | 1 minute of caring involved |
| 15:30.23 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
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| 15:34.39 | widler | would anyone look at this http://pastebin.com/55pNDGP7 and tell me what's wrong with my AORs? |
| 15:40.16 | c|oneman | [TK]D-Fender: I'm using them with digium switchvox so I don't have much control over the provisionning server |
| 15:40.36 | c|oneman | in any case, I can't seem to get them to connect to any server half the time |
| 15:40.52 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 15:41.22 | [TK]D-Fender | c|oneman: Set up your own then |
| 15:41.39 | c|oneman | how does the phone know where the server is out of the box? |
| 15:41.57 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:43.27 | [TK]D-Fender | the phone has specific settings from factory |
| 15:43.39 | [TK]D-Fender | Which you can of course change immediately on plugging them in |
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| 15:57.14 | gavimobile | [TK]D-Fender: I think I may have narrowed down the issue. the problem seems to be only happening on 1 of my peers. this peer is a linksys ata box with 2 lines. I disabled the enabled line and enabled the disabled line and configured it to my sip settings. the call immeditly went through. recently I was trying to disable the *77 feature on my sipura device so asterisk could use the ext *77 from the peer. I was unsucessfull and the setting |
| 15:57.40 | gavimobile | would the block annonymous callers setting have anything to do with my pbx? |
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| 16:19.15 | mjordan | widler: without seeing your configuration and a DEBUG log with the messages received from the PJSIP stack, it's pretty hard to say. |
| 16:26.55 | tom______ | I have a strange question. I have a scenario where a call comes in an plays a recording, than dials a number to listen to another recording. I need the ability to get that call back when the callee hits the #. I have tried using features.conf and can see it activate but it does not end the dialed call and move on. Any advice would help |
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| 16:41.09 | bkruse | @tom______ why not just have the # sign hangup the call on the second box? |
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| 16:42.46 | tom______ | that would work. How would I do that? I single direction Bye message? |
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| 17:00.20 | bkruse | @tom______ just hangup() after your playback in the dialplan |
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| 17:01.24 | tom______ | It isnt a playback. It is a dialed call. the recording is offsite and requires a sip call. I would need to bye just one leg of the call |
| 17:02.53 | [TK]D-Fender | tom______: Have your feature call a macro that looks for the bridgechan and kills it. |
| 17:03.39 | mbowie | Anyone want a refurb'd Cisco SPA514G-WS? Just ordered it two weeks ago for eval and no longer need it. |
| 17:03.41 | tom______ | ok. I can try that. Not sure how to kill just 1 side. |
| 17:04.38 | [TK]D-Fender | kill that channel... that is killing only 1 side.. since the caller isn't hanging up, "g" should work. |
| 17:05.03 | tom______ | ok. I will try that |
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| 17:20.09 | hexanol | I have a question about CCSS |
| 17:20.51 | hexanol | it only seems to works when the Dial application is used in the "original" exten/context that was called |
| 17:20.55 | hexanol | for example, if I have an extension |
| 17:21.11 | hexanol | that do a "GoSub" into another extension |
| 17:21.25 | hexanol | and the Dial application is called in this extension |
| 17:21.56 | hexanol | then when call completion is requested, it rings the caller phone correctly, but it then call directly the "gosub" extension |
| 17:22.21 | hexanol | which is not working for my dialplan, because the gosub extension expects some argument |
| 17:22.25 | hexanol | so, how do you work around this ? |
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| 18:01.39 | voip | doot |
| 18:01.52 | carrar | hh |
| 18:01.53 | carrar | shhh |
| 18:01.59 | *** part/#asterisk tom______ (~tom@wsip-24-120-113-148.lv.lv.cox.net) |
| 18:02.04 | carrar | be warry warry quit |
| 18:02.06 | carrar | quite even |
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| 18:22.33 | TazzNZ | morning all |
| 18:31.15 | voip | hrmmmmmmmmmmmmmmmm |
| 18:31.21 | voip | carrar: greetings |
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| 18:54.35 | MasterSenpai | I am trying to figure out how to configure Asterisk so when I dial from the Asterisk box, the call would be routed to a Freeswitch server? There is documentation from Freeswitch that talks about it, but it only mentions the configuration for the Freeswitch box. I am wondering if there is configuration I need to setup on the Asterisk box in order to do this? |
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| 19:06.11 | P-NuT | Hey all, is it possible that you can have a username of [1000] but have a SIP phone number of 12345? If so what do you put in users.conf? |
| 19:08.54 | TazzNZ | P-NuT: you are mixing up routing and registration |
| 19:09.02 | [TK]D-Fender | Users.conf = trash |
| 19:09.04 | [TK]D-Fender | Stop using it |
| 19:09.33 | TazzNZ | routing is where you take a number or sip address and route it to an end point or destination |
| 19:09.59 | TazzNZ | registration is to let Asterisk know where an end point is |
| 19:10.44 | P-NuT | Not really, because you don't need to add a registered user to extensions.conf if you have them in users.conf. They can all call each other based on username. |
| 19:11.00 | [TK]D-Fender | P-NuT: Seriously... ditch it. |
| 19:11.26 | [TK]D-Fender | P-NuT: And run your own dialplan |
| 19:11.33 | P-NuT | So you think its always better to register with the number? |
| 19:11.49 | [TK]D-Fender | P-NuT: No. A number you dial is EXTENSIONS.CONF |
| 19:11.59 | [TK]D-Fender | Users.conf thinks it manages BOTH device config AND dialplan |
| 19:12.00 | [TK]D-Fender | this is BS |
| 19:12.11 | [TK]D-Fender | Stop using users.conf and make your own peers and dialplan |
| 19:12.55 | [TK]D-Fender | The username and what you dial from a device need not have anything to do with each other. |
| 19:13.02 | P-NuT | really? This is news to me everyone recommends using users.conf |
| 19:13.11 | [TK]D-Fender | Who is this "everyone"? |
| 19:13.25 | TazzNZ | [TK]D-Fender: the *everyone....you know :D |
| 19:13.27 | [TK]D-Fender | Pretty much nobody here |
| 19:13.50 | [TK]D-Fender | ~users.conf |
| 19:13.52 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 19:14.02 | P-NuT | Hmmmm.. So just use sip.conf and extensions.conf instead? |
| 19:14.05 | [TK]D-Fender | yes |
| 19:15.33 | P-NuT | Hmmm, ok then. I'll take that advice onboard. |
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| 19:43.56 | Katty | repeatedly pokes Qwell |
| 19:44.05 | Qwell | dies |
| 19:44.12 | Katty | YOU"RE BACK! |
| 19:44.15 | Katty | how was your bday |
| 19:44.17 | Qwell | I am! |
| 19:44.32 | Qwell | I was on a boat. |
| 19:44.41 | Katty | with no alchohol i hear. |
| 19:44.48 | Qwell | indeed. stupid TN law |
| 19:44.51 | Katty | but /how/ was the bday? |
| 19:44.56 | Katty | besides alchohol free. |
| 19:44.58 | Qwell | I was on a boat. |
| 19:45.06 | Katty | were you going fast, too? |
| 19:45.13 | Qwell | like 20mph :p |
| 19:45.23 | Qwell | We got a pontoon, because, boat. |
| 19:45.26 | Katty | lol |
| 19:45.43 | Katty | lots of lounging? or did you actually fish? |
| 19:45.51 | Qwell | no fishing |
| 19:46.17 | Qwell | if you ever go back ~to Gatlinburg, I suggest going to Douglas Lake. |
| 19:46.35 | Katty | i'm sure we will |
| 19:46.44 | Katty | is that where you rented the pontoon? |
| 19:46.48 | Qwell | also, Nascar Speedzone. |
| 19:46.57 | Qwell | $18 for riding go-karts, ALL DAY LONG |
| 19:47.02 | Katty | nascar....doesn't.... |
| 19:47.06 | Katty | ah, go karts. |
| 19:47.24 | Katty | i don't think i can go kart if it's not related to mario. |
| 19:47.33 | Qwell | They probably sell bananas there. |
| 19:47.38 | Katty | teehee |
| 19:47.46 | Qwell | I didn't check. Why did I not check? I'm slacking. |
| 19:47.51 | Katty | you so are. |
| 19:47.56 | Katty | how much was the pontoon rental? |
| 19:47.57 | file | Qwell, SLACKER |
| 19:48.00 | Katty | and did you sleep on the pontoon? |
| 19:48.22 | Qwell | surprisingly cheap. 4hr @ $100, but since we got there early, they gave us another hour for $20. |
| 19:48.28 | Qwell | plus gas was like $30? |
| 19:48.40 | Katty | not bad. |
| 19:48.55 | Katty | do you think the lake is ok for kayaking, too? |
| 19:49.09 | Qwell | Probably! It's connected to the TN river. |
| 19:49.30 | Qwell | it's huge. lots of distance available to travel. |
| 19:49.34 | Katty | and how far is it from gatlinburg, you think? |
| 19:49.55 | Qwell | umm, I want to say it was like a 40 minute drive from Pigeon Forge? |
| 19:50.08 | Katty | south.....east? |
| 19:50.15 | Qwell | northwesterly |
| 19:50.24 | Katty | sieverville? |
| 19:50.27 | Katty | or whatever it is |
| 19:50.58 | Katty | googles |
| 19:51.04 | Qwell | err, no, northeast I guess. it's the big body of water nearby. You can't miss it on maps |
| 19:51.52 | Katty | oh wow that's huge! |
| 19:51.59 | Qwell | twss |
| 19:52.32 | Katty | we might go next year |
| 19:52.41 | Katty | of course we were going to go somewhere this year, but with everything that happened... |
| 19:52.53 | Katty | derik's brother died, my dad had open heart surgery. |
| 19:53.02 | Katty | his dad has been diagnoised with cancer |
| 19:53.14 | Katty | i don't think we're going anywhere anytime soon :/ |
| 19:54.24 | Katty | if anything we might go to pennsacola or something |
| 19:54.55 | Katty | i'm really glad you had a good bday tho!!! |
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| 23:13.48 | marsmars | hi, I have a problem that my gsmgateway (raspbx with deactivated freepbx) sends a hangup via IAX2 to my asterisk server runnig on a modded Router (Fritz!box modded with Freetz and Asterisk 11 with chan_capi); It seems, taht the IAX2 Channel direct hangup without lookup for dialplan... I could post some conf-Files at pastebin ... |
| 23:24.43 | justdave | so if you have two devices connected to an asterisk server, and one of them is video-capable and the other isn't, is it possible to specify a static image to use as the representative video content from the non-capable one when sending video to the one that is capable? |
| 23:24.59 | justdave | (on a call placed between them) |
| 23:25.20 | justdave | if it is, my google-fu apparently isn't picking the right search terms to find it |
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| 23:36.09 | marsmars | if you mean with "video-capable" a video conference then I have to corect me, that is not a feature I want to use. I want to make a call from ISDN over IAX2 to mobilefunk |
| 23:36.37 | justdave | nah, this is for a call between two phones |
| 23:37.32 | justdave | one is a desktop Polycom phone without a camera and the other is an Eyebeam softphone, for example |
| 23:38.24 | justdave | would be nice to put a silhouette or something on the screen instead of just having it blank if the caller isn't sending any video |
| 23:59.20 | marsmars | chan_iax2.c:4829 __auto_congest: Auto-congesting call due to slow response |
| 23:59.20 | marsmars | <PROTECTED> |
| 23:59.21 | marsmars | <PROTECTED> |